irclog2html for #asterisk on 20050519

00:01.27justnulling2crash3m: http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 There are currently no files for this type.
00:01.31duke149thats just standard mobile phone isnt it? a phone company actually has one of their towers on our site
00:02.11newmedianduke149: that's just for cellular. You set up your own cell repeater, to give yourself perfect or near perfect signal strength; It repeats the closest tower to you.
00:02.24duke149yea
00:02.29crash3mjustnulling2: I dont have a username/password ;)
00:03.06justnulling2crash3m: well i have one and it says There are currently no files for this type.
00:03.10duke149did look at mobile phones once but it just costs 2 much
00:04.01crash3mjustnulling2: an associate just got P0S3-07-4-00 from cisco...I dont know where, but he did
00:04.47justnulling2hmm
00:07.10ariel_ok is anyone here good with the polycom phones. I have a question about there qos settings
00:08.18newmedianjbot seen ManxPower
00:08.20jbotmanxpower <~eric@adsl-6-105-205.msy.bellsouth.net> was last seen on IRC in channel #asterisk, 9h 34m 25s ago, saying: 'you asked about RTP always passing thru Asterisk.'.
00:09.18*** join/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk)
00:09.20bumperhostingHello
00:09.29cursorhello there
00:09.32cursorUK, I see
00:09.35*** join/#asterisk ChkDigit (~mike@static65-87-226-124.regina.accesscomm.ca)
00:09.36cursorYou're up late
00:09.37bumperhostingYeh:D
00:09.40ariel_hello
00:09.41bumperhostinglol
00:09.45*** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net)
00:09.50bumperhostingNot that late its only 1.09
00:10.06cursorwow - that's the same time as on my clock :-)
00:10.13ariel_1 am  hummm I would be sleeping by then.
00:10.32cursorsleep is for the weak
00:10.39*** part/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net)
00:10.43NewSoleMine is Flashing 12:00
00:10.49ariel_cursor, then let me be week.
00:10.54bumperhostinglol
00:10.54ariel_weak...
00:11.05GyrosGeierHrm
00:11.09cursorsleep for a week
00:11.09bumperhostingI dont sleep till about 8am every morning
00:11.10ariel_NewSole, you have my vcr then
00:11.18duke149any recomendations for what distro is good to run Asterisk on?
00:11.25cursorGentoo
00:11.28ariel_CentOS
00:11.41cursorMS Windows XP
00:11.42cursorerr
00:11.43cursoroops
00:11.47cursor:-)
00:11.50GyrosGeierWhat would be a good way to connect a USB phone (HID+Soundcard) to a Linux box?
00:11.58duke149wtf
00:12.01cursorhaha
00:12.21ariel_duke, what OS are you normally used to?
00:12.33cursorGyrosGeier: Via the USB port
00:13.15bumperhostinggod alll so technical
00:13.17*** join/#asterisk theHub (~karl@ool-182cba82.dyn.optonline.net)
00:13.19bumperhosting:( i would not know where to start
00:13.26newmedianNewsole.. there's this .mov, and you watch it, and one week later you get a phone call....
00:13.38GyrosGeiercursor, obviously.
00:13.44cursor:-)
00:14.30GyrosGeiercursor, I was wondering whether there is any soft phones that can deal with this hardware
00:14.41shido6?
00:14.43NewSoleI go to bathroom and do a mov in there and a few min's I get a call for neibors... wondering what died
00:14.56*** join/#asterisk duke149 (~ca07b782@cgiirc-gateway.ictoadd.com)
00:15.00cursorI don't know - sorry
00:15.01newmedianthank you for airing your concerns
00:15.09NewSolelol
00:15.12duke149bah dam CGI IRC client.
00:15.12shido6you did a movie in the bathroom while you were crapping?
00:15.34cursornewmedian: would that be thering.mov ?
00:15.45newmedian:)
00:15.45duke149as i was sayin.. anyone got any recomendations for what distro to run Asterisk on?
00:15.50NewSolethe-bowl-ring.mov
00:16.14shido6redhat works duke149
00:16.32cursorduke: I always recommend Gentoo, so that makes me very predictable
00:16.38duke149hehe
00:17.48newmedianHere's a Linux Distribution Chooser: http://www.zegeniestudios.net/ldc/index.php
00:17.57crash3mjustnulling2: http://www.voip-info.org/tiki-index.php?page=Cisco+Phones
00:18.41justnulling2crash3m: thanks let me take a look at it
00:20.01bmdhey, I have a fairly technical question about HEAD and new features - any -dev types around?
00:21.19cursorask about HEAD and they will come
00:21.30bmdok here's my question
00:21.36duke149newmedian: cool. it says Debian for me.
00:21.44bmdfunctions and agis.  how does one call a function from an agi
00:21.52bmdis there some identity app that I don't know about?
00:21.59jakepdev[work]EXEC
00:22.05cursoryes - exec
00:22.11bmdexec runs apps
00:22.16bmddoes it run functions too?
00:22.26jakepdev[work]functions?
00:22.29cursorfunctions are apps
00:22.31cursorlike dial
00:22.33cursoretc.
00:22.45jakepdev[work]oh - where's that damn translator
00:22.46bmdlooking at pbx.c they look like they're different
00:23.13*** join/#asterisk bofh42 (~bofh42@p548234A3.dip0.t-ipconnect.de)
00:23.35*** kick/#asterisk [cursor!~mark@kram.digium.sponsor.pdpc] by kram (turn off your silly script)
00:23.39*** join/#asterisk hypa7ia (~leigh@HSE-London-ppp291210.sympatico.ca)
00:24.20kramhi file
00:24.25filehi krammy boy
00:24.31filehow are you?
00:25.28*** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz)
00:25.31cursorThat's nice
00:25.34krami'm otay
00:25.47krami've mentioned it to you before, cursor :)
00:25.55kramit was maybe funny 3 or 4 months ago
00:25.56bmdcursor: can I bug you a bit more about apps vs. functions?
00:25.57krambut it's time to move along
00:26.18jakepdev[work]bmd - what are you calling a function?
00:26.44cursorbmd: if you like
00:27.30NewSolehi kram
00:27.33kramhi sole
00:27.36*** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net)
00:27.39bmdjake: anything registered by ast_custom_function_register
00:27.53bmdcursor: looking at the code, it looks like functions and apps differ
00:27.57bmdat least in HEAD
00:28.00NewSolegot a question about your TE405 cards if u got a moment
00:28.02*** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com)
00:28.14drumkillaNewSole: what's the question?
00:28.14cursorCan you give an example of a function vs. an app?
00:28.15bmdfunctions are registered with ast_custom_function_register, apps are registered with ast_register_application
00:28.18*** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-133.modem.logical.net)
00:28.31cursorok
00:28.35NewSoleis there a special setup for Echo cancel on those cards kram
00:28.43*** join/#asterisk Zaw (zaw@zaw.subneural.net)
00:28.45bmdfuntion SIP_HEADER, app SipGetHeader
00:28.56kramsole: you can e-mail support@digium.com
00:29.03kramthey can assist you with configuration issues
00:29.10kramalso we have a new echo can board you can use if you want to offload it
00:29.19jakepdev[work]bmd - you can call SIP_HEADER from the dialplan?
00:29.38bmdjake: yes it's new in HEAD
00:29.41jakepdev[work]ok
00:29.46MikeJ[Laptop]for the echo can board email sales@digium.com ;)
00:29.47bmdfunctions are a new addition
00:29.58NewSolewell.. ok....
00:30.36bmdactually, looking at the code, it looks like nobody has committed any functions yet
00:30.47bmdsorry, I've been following mantis a bit more than CVS
00:30.52bmdthere's a lot of them in mantis
00:31.01cursorI didn't notice the functions stuff go in
00:31.02bmdSIP_HEADER seems to be the only one right now
00:31.06MikeJ[Laptop]bmd, there are a bunch of them, they are in head, not 1.0.x brance
00:31.08kramsole: if you can't get your question answered through tech support, let meknow
00:31.09MikeJ[Laptop]branch
00:31.09cursorit probably had a CVS log like "oops"
00:31.28cursorI'll see if I can find it
00:31.41NewSolewell last time I sent an email to tech... I waited a week almost for anser back
00:31.53kramwhat's your ticket number?
00:31.54drumkillacursor: if you didn't notice, then you weren't paying any attention
00:31.56bmdMikeJ: all I see is SIP_HEADER, grepping for ast_custom_function_register
00:32.03drumkillathere has been a ton of commits realted to dialplan functions
00:32.07cursordrum - possibly :-)
00:32.12bmdok, so back to my question
00:32.13drumkillacertainly
00:32.15bmdagis and functinos
00:32.24bmddoesn't look like you can call a function from an AGI
00:32.27*** join/#asterisk mogorman (~mogorman@207.111.174.1)
00:32.30NewSoleI would have to look up its on office computer
00:32.36bmdunless I'm missing something
00:32.43bmdso there's either two solutions
00:32.55bmdone is an identity app, that just echos what the function args sent to it return
00:33.14bmdthe other is a new agi command outside of EXEC that evaluates functions
00:33.27bmdor am I totally out of it here
00:33.38jakepdev[work]that sounds right
00:33.46kramnewsole: i'll have matt o contact you then
00:34.11jakepdev[work]or implement the function code in your agi app
00:34.11NewSolek
00:34.28bumperhostinganyone wants to do some contracting for me, installing asterisk on a FreeBSD system, hosted remotely.
00:34.32bumperhostingoops
00:34.41bumperhostingdoes anyone want to... " " etc etc
00:34.44kramif you don't get the assistance you need, let me know
00:35.07jakepdev[work]bumperhosting - try shido6
00:35.19NewSoleok thank
00:35.26bumperhostingjakepdev, thanks I will.
00:36.00bmdjakepdev: well, things like SIP_HEADER are impossible to implement in an agi (after SipGetHeader is removed)
00:36.09bmdso there's a gap here in functionality (pardon the pun)
00:36.38*** join/#asterisk lilneon (~tj_r3@cuscon14794.tstt.net.tt)
00:36.41cursorI see a commit notice from 05 May 2005 related to functions
00:36.46lilneonhi everyone good night
00:37.07asteriskn00bevening all, new asterisk install on centos 3 Whenever I try to start asterisk, it randomly dumps and will not load, I have tried asterisk -vv but I get no "error" messeges.  Any ideas on what to look for?
00:37.21*** join/#asterisk hypa7ia (~leigh@HSE-Montreal-ppp143550.sympatico.ca)
00:37.24lilneonanyone can point me to a voip provider who can provide me with a toll free inbound number to my asterisk box?
00:37.32cursornufone.net
00:37.52cursorAssuming USA
00:38.12lilneoncursor: assumption incorrect.. want outside the US, UK..
00:38.34cursorMy mind-reading skills are obviously getting rusty :-)
00:38.56lilneoncursor: nah.. prob no mind to read... i left it at the office
00:39.01cursorhaha
00:39.06lilneon:)
00:39.13lilneonthnx though,... n e others?
00:39.19cursorcalluk sell UK freephone numbers
00:39.26cursorcalluk.com, I think
00:39.33bumperhostinguk
00:39.44bumperhostinggo with sipgate.co.uk they give you free uk numbers.
00:39.46lilneonyeah but that's tol lfree for callers in the UK
00:40.22bumperhostinggo with sipgate... they give u 0845, 0800, 0124 where ever u are... every area in the uk go with sipgate its completly free
00:40.25cursorfreephone numbers are only free within the country the service is provided, unless you get an international toll-free
00:40.28cursor+800
00:40.51lilneonwhich is what i kinda need right nwo
00:41.08cursorI don't know any +800 providers - I haven't looked
00:41.15cursorGoogle should scare up a couple of them
00:41.30cursorThey are probably expensive
00:41.50cursorIt might be cheaper to get a number for each country you want to deal with
00:42.40bumperhostingsipgate is free if u want uk man;)
00:43.49DaminFrom: markster@digium.com "Actually apply timestamp bug (bug #3961)"
00:44.02DaminSo now we're actually ADDING bugs to CVS Head? ;)
00:44.19kramdamin: sorry :)
00:44.25*** join/#asterisk rene- (~root@200.78.176.114)
00:44.38cursorHello rene
00:44.44rene-hey!
00:45.18PatrickDKwe like bugs
00:45.22PatrickDKgives us something to do all day
00:45.33*** join/#asterisk ilium007 (~ilium@203.94.178.214)
00:45.38ilium007hi guys
00:45.51ilium007"Unable to connect to remote Asterisk"
00:45.55ilium007any ideas ???
00:46.06ilium007I am ssh'd into my * box at home form work
00:46.16PTG123cvs head is where bugs live :) its like an ant hill
00:46.21CardoeAny reliable outbound calling providers?
00:46.34PTG123Cardoe, how many minutes per month?
00:47.01Cardoequite a few
00:47.04rene-iirc digium used to sell developer kits based in t100p and zhone zplex back in the day, i have now stumbled on some cheap units on ebay, they claim to have V35 interfaces so how does one connects one to a Digium Rj48 interfase? i have no telecom background so i dont know whether v35 is a protocol or a physical interfase
00:47.20PTG123cardoe: pm so i can help you out..
00:47.57filePTG123: you!
00:48.09PTG123file: you! :)
00:48.18filePTG123: status report.
00:48.28PTG123heh pm me :)
00:49.44*** part/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com)
00:50.54*** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net)
00:53.25*** join/#asterisk shakuhashi (~luciux@200.101.54.31)
00:56.14ariel_rene-, the zhone units were ok but they had major problems.  I had 8 of them and ended up giving them away.
00:58.01*** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no)
00:58.34fileariel_: Got any left?!?
00:59.25*** join/#asterisk TransAM (me3@adsl-68-92-124-249.dsl.lgvwtx.swbell.net)
00:59.29TransAMgood evening
00:59.38TransAMI have a question
00:59.47b0efwhat happened to the transfer command?; I can't find it in recent CVS. I got ztdummy and zaptel on a 2.6 kernel and able to dial into it, but I can't find the transfer command to transfer a call from the asterisk cli. Is this function removed and only found in the manager interface?
01:00.20TransAMwhat is wrong when, I try an echo test from more then one computer and even the phone line and I dont get anything back at all except for the numberpad keypress?
01:00.53TransAMI can call out, hear the called party but I cannot be heard, any idea?
01:01.30mepplgute nacht
01:01.33b0efTransAM: how do you know the keypresses are being echoed?
01:01.55TransAMbecause I press the keys and her it a split second later
01:01.57b0efare you sure they are being echoed from the remote server?
01:02.02mepplgood night
01:02.12TransAMyes
01:02.18b0efmeppl: sleep well, send enemies to hell
01:02.25TransAMIhear them twice
01:02.32meppl;)
01:02.36b0efTransAM: what protocol?
01:02.38*** join/#asterisk Johnsie (~john@acs-24-154-32-12.zoominternet.net)
01:02.41TransAMsip
01:02.45TransAMhmm
01:02.51TransAMwhat ever xlite uses
01:03.03TransAMand when I dail in with my cell phone, same thing
01:03.28b0efhow about when you dial the echo application from the asterisk cli?
01:03.43TransAMcli?
01:03.47TransAMohh
01:03.54bkw_have chan_oss/alsa
01:03.56bkw_and a sound card
01:03.58bkw_:P
01:04.07TransAMI never tried
01:04.13TransAMlet me give it a whirl
01:04.54b0efnow, where did the transfer command go..
01:05.06b0efis it just me?
01:09.00DaminI need chan_pringle-can
01:09.09DaminAnd chan_ting-can-and-string
01:10.13*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
01:10.13*** mode/#asterisk [+o bkw_] by ChanServ
01:11.15*** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org)
01:13.02TransAMI am too new to this, I cant figure it out
01:13.20TransAMI thought I Was doing good when I got it to finally awnser the phone
01:13.49TransAMof course it picks up , then says there has been a terrible error and hangs up
01:14.49b0efwell, you know what you need to do
01:15.07TransAMbuy a bullet, rent a gun?
01:15.09b0efget cli up and dial the echo app locally from the cli
01:15.27TransAMstill trying to figure out how to do that
01:16.18b0efjust execute asterisk -vvvvpc
01:16.45*** join/#asterisk buu (foobar@69.56.150.34)
01:16.55b0efsetup an extension to dial the echo app, I've set mine to extension 600
01:17.15b0efwhen in the cli, just do a dial 600
01:17.30*** join/#asterisk netvulture (0@63.174.172.245)
01:17.31b0efmake sure you've setup asterisk to use a sound api, in modules.conf
01:17.49buuHow do I call a function, like say, "SetLanguage(foo)" from an AGI script?
01:20.08*** join/#asterisk Flav (~jmm@user-10lfdgg.cable.mindspring.com)
01:20.11netvulturecan anybody explain the function of the ./makeconfig.sh 7123 2123 000F905796D9 192.168.1.123 63.174.172.123 "(650) 403-7123 "
01:20.22netvulturewhoops - bad paste
01:20.25*** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com)
01:21.03netvulturecan anybody explain the function of the proxy_backup on a 79xx cisco phone?
01:21.53netvultureat what point would the phone use this proxy instead of the defaul outbound_proxy and line1_proxy?
01:22.05PTG123if it can't reach the primary i am guessing
01:23.00netvulturei gather that would be the idea of it - however cisco has no documentation worth reading that explains it
01:23.06buuBah
01:23.08buuAnyone?
01:23.18netvulturethus cisco proboably doesn't even know what it does
01:24.07*** part/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk)
01:24.41TransAMwhat file holds the SIP passwords?
01:24.56*** join/#asterisk dave2_ (~dave2@leibniz.catalyst.net.nz)
01:25.49Mavviestart with sip.conf, work through all included files.
01:26.54dave2_Hi all, any objections if I launch into a long question? :)
01:27.09Mavviethat was the first.
01:27.31newmediandave2 if you're going to cut and paste, use pastebin
01:27.33newmedian~pastebin
01:27.34jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca
01:27.50dave2_wasn't going to paste a lot, more that I know it's going to be fairly involved..
01:28.09dave2_I have a bunch of UIP200s on the same LAN as my * box..
01:28.28cursorTime for me to go
01:28.32cursorLater, guys
01:29.17dave2_transfers using the XFER hard button on the phone sort of work, but * loses the call into a void if we try and transfer to an extension which ends up in voicemail..
01:29.56dave2_I suspect it's a UIP200 bug, but before making noises in that direction, I hope someone else has run into it.. :)
01:31.12rene-ariel, i have learned that too, is there an alternative that is as cheap to get multiple FXOs in asterisk
01:34.42rene-the cheapest way im aware of getting multiple fxos in asterisk are clipcomm 4 port sip channel banks
01:34.52rene-they are like $350 or so
01:36.24asteriskn00bwhile trying to start asterisk I am receiving the following err "Ouch ... error while writing audio data: : Broken pipe" any ideas?
01:36.29*** join/#asterisk forrestc (~fwc@206.127.78.199)
01:36.50forrestcAre there any SIP protocol level experts on and listening?
01:37.14forrestcI need to find out about the nonce parameter and when it should change
01:37.31forrestcI've got a sip box which is trying to register with the same nonce parameter over and over.
01:38.45*** join/#asterisk jets (~brian@guardian.pmt.org)
01:39.02jetsIs thre major queue development going on?
01:40.35*** join/#asterisk forrestc (~fwc@206.127.78.199)
01:40.55forrestcOooppss.. hit wrong key..   Was there someone one who knew about SIP and the nonce parameter?
01:41.02*** join/#asterisk duke149 (~ca07b782@cgiirc-gateway.ictoadd.com)
01:46.51*** join/#asterisk heison (~heison@216.191.251.226)
01:51.12duke149.
01:51.19Damin--- Results after 594 passes ---
01:51.20DaminBest: 100.000000 -- Worst: 99.987793
02:05.30*** join/#asterisk Cresl1n (~matt@216.207.245.23)
02:06.29*** mode/#asterisk [+o Cresl1n] by drumkilla
02:06.41jetsFYI do an answer before a queue.
02:06.42jetsDuh.
02:06.43jetshaha
02:07.51Cresl1ndrumkilla!!!!!
02:07.55drumkillaCresl1n: !!!!!!!!!!!!!
02:07.59drumkillawe are having a mad party in here.
02:08.09drumkillamusic should be much louder, though
02:08.12Cresl1nyeah
02:08.16Cresl1nhere, I'll turn it up
02:08.23drumkillathere we go.
02:08.31MikeJ[Laptop]ok, that's funny
02:08.41MikeJ[Laptop]it is a mad party over there
02:09.04Cresl1nhey.... is somebody looking at us?
02:09.24Cresl1nwe need to get some mad lights in here
02:09.34MikeJ[Laptop]hehe
02:10.13Cresl1nwoohoo!
02:10.15*** join/#asterisk file[mac] (~jcolp@mctn1-3494.nb.aliant.net)
02:10.20Qwellomg its file!
02:10.21*** join/#asterisk Inv_arp (junya@adsl-3-248-195.mia.bellsouth.net)
02:10.22Cresl1nwe need some lights in #asterisk too :-)
02:10.25file[mac]yes it's me
02:11.49Qwellfile[mac]: Just asked my boss about cluecon...let's see how he takes it. :D
02:12.21Cresl1nhow can we get some lights in #asterisk?
02:12.34Qwellwhat kind of lights?
02:12.42Qwellleds, christmas lights, what?
02:13.46Cresl1ndance club lights
02:13.52Qwellstrobe lights?
02:14.12file[mac]Cresl1n makes me happy!
02:14.21jetsyay for cluecon
02:15.06Cresl1nwoohoo!
02:15.16Cresl1nI'm hyper right now :-)
02:15.21Cresl1nI just drank a can of redbull
02:15.27Cresl1n(a.k.a. happy juice)
02:15.29drumkillaREDBULLREDBULLREDBULL
02:16.01Cresl1nwoohoo!
02:16.02Cresl1nkram!!!
02:16.17drumkillalet's have a dance party
02:16.19Cresl1ngood music
02:16.38Juggiehah
02:16.49drumkillai'll just dance in my chair
02:16.53drumkillawhile I code
02:17.00Juggiea dance party with only guys
02:17.02file[mac]can I dance too?!?
02:17.05Juggiedoesnt sound like much of a party
02:17.10file[mac]unfortunately drumkilla and Cresl1n don't swing that way
02:17.10Cresl1noooohhh.....
02:17.17Cresl1nthere's got to be a chick somewhere here
02:17.27Juggiei dont believe it
02:17.27Cresl1n:-)
02:17.34Cresl1nhrm....
02:17.35drumkillahrm ... it's looking pretty rough
02:17.40Cresl1nyeah, kind of bleak
02:17.42file[mac]Cresl1n: come over to the dark side!
02:17.46Cresl1nno!
02:17.47JuggieCresl1n, i got the quote from Bill Hall, its gone to management now.
02:17.50Cresl1n=)
02:17.52Cresl1ncool
02:18.19*** join/#asterisk Moc[Toronto] (~mochouina@142.131.66.185)
02:18.20file[mac]hippity hoppity!
02:18.36Moc[Toronto]hi everyone
02:18.42file[mac]hey Moc
02:18.45drumkillawheeeeeeee
02:18.47MikeJ[Laptop]hey moc..
02:18.49Juggiefile, you going to cluecon?
02:18.57MikeJ[Laptop]what are you doing in toronto...
02:18.58file[mac]yes
02:19.02MikeJ[Laptop]file, of course he is
02:19.02Cresl1ndrumkilla, are you going to cluecon?
02:19.10Juggiefile, you swinging by to get me on the way?
02:19.12file[mac]look at http://www.cluecon.com/, specifically the schedule
02:19.13Moc[Toronto]doing a national intranet site..
02:19.25file[mac]Juggie: haha
02:19.33Juggiewell your comming from NB :)
02:19.35Moc[Toronto]I gota get brian to change my name on the schedule
02:19.40Juggieyou gotta pass through ontario :P
02:19.58file[mac]I might go through Pearson airport
02:20.14file[mac]Moc[Toronto]: I can get it changed
02:20.16PTG123hey moc hows the fax client coming?
02:20.23file[mac]what do you want it to say?
02:20.33Juggiefile, drive all the way to pearson from nb? why not just fly from nb if your gonna fly
02:20.40file[mac]Juggie: I am :)
02:20.46drumkillaCresl1n: if I am sent there, then yes
02:20.47file[mac]but I will probably have a connection at Pearson
02:20.49Juggieahhhh
02:21.10Juggiei looked @ a flight today, it was 400$ tax in
02:21.18file[mac]mine are coming out to $530
02:22.17Juggiedriving would take 12hrs of straight driving
02:22.21Juggiewhich i'd like to avoid
02:22.44QwellWhats a good travel site?
02:22.47*** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca)
02:22.57Qwellhotels.com is the only one I've ever used...but hotel is included with cluecon
02:23.11Juggiewww.expedia.com www.travelocity.com
02:23.11Moc[Toronto]rr
02:23.17Qwellexpedia, thats the one
02:23.39file[mac]travelocity is what is giving me the best prices
02:23.52QwellWhat is the airport code?
02:24.00Qwellit's like, o'hare airport, right?
02:24.08NewSole2travelocity are assholes
02:24.17file[mac]Moc[Toronto]: I'm getting it fixed for you
02:24.17asterisk99Does anyone here use Sipura 1001? I have a new unit and it will not ring the phone, yet I can go offhook and take the call :(
02:24.37Moc[Toronto]getting what fixed ?
02:24.42file[mac]your name
02:24.48Moc[Toronto]oh ok thanks
02:24.52file[mac]yw
02:25.33Moc[Toronto]Marc Chouinard is my father brother
02:25.43Qwellyeah, is it midway, or o'hare?
02:25.50asterisk99O'Hare = ORD
02:26.57asterisk99Midway = MDW and Meigs (a la Flight Sim) = CGX
02:26.58Juggiei doubt i could get work to pay for cluecon
02:27.07Juggiei dont have passport either
02:27.10Juggiei should do that
02:27.29QwellI'm gonna try, but I need exact costs first...sucks working for a bank
02:27.40MikeJ[Laptop]asterisk99, meigs does not take aircraft anymore.. it got bulldozed
02:27.54Juggiei already have them semi convinced to go to san francisco
02:28.09asterisk99Now ass for ass*****, they are the ones who named the Canadian airports (like YYZ = Toronto?????????????)
02:28.13MikeJ[Laptop]that was a great airport to fly into.. second fav to toronto city senter
02:28.24MikeJ[Laptop]center
02:28.37Moc[Toronto]??
02:28.44Juggieairport codes are fine
02:28.45asterisk99Centre   ---- Canadian spelling ;)
02:28.54MikeJ[Laptop]hehe
02:28.58Juggiedont need to give away the name of the city
02:29.01Juggieits just a code
02:29.22MikeJ[Laptop]great airport, point at the CN tower and go till you almost hit it, then land...
02:29.25Moc[Toronto]toronto is boring...
02:30.12Juggiei like visiting toronto
02:30.14asterisk99true true... but YEG for Edmonton????
02:30.14Juggiebut i woudnt live there
02:30.39Juggieasterisk99, st. john's is YOW its all the same :)
02:31.52asterisk99I worked in the RBC tower in toronto when the Blue Agels came screaming down Yonge St, tunred right on Bay St and headed toowards the CN tower... I was on the 18th floor, they were level with the 17th floor
02:32.24asterisk99Does anyone here use Sipura 1001? I have a new unit and it will not ring the phone, yet I can go offhook and take the call :(
02:32.25*** join/#asterisk kb1_kanobe (~krisbouti@d154-20-43-155.bchsia.telus.net)
02:32.33*** join/#asterisk iq (~iq@70-59-166-10.omah.qwest.net)
02:34.20heisonmoc[toronto]: when are u leaving?
02:35.08*** join/#asterisk IsMe (~me@218.111.224.205)
02:35.48IsMehi, i am looking for a developer that can do CID callback service
02:35.58*** part/#asterisk rene- (~root@200.78.176.114)
02:36.06asterisk99Juggie: "YOW" is the Scam Capital of Canada... Ottawa    St. John's (Nfld) is a really obscure YYT
02:38.57QwellThis site really needs to list what the nearest airport is...
02:39.33Qwelloh, it does :D
02:41.14iCEBrkrbkw_: wake up
02:42.45Juggieasterisk99, ahh YOW is halifax is it
02:43.03asterisk99Ottawa
02:43.13Juggieahh
02:43.22Juggiethats where i am
02:43.24Juggieand i go to st. johns alot
02:43.29Juggiei get the codes mixed up
02:44.09asterisk99I guess all the Sipura users are waitig in lline for the 1st showing of "Revenge of the Sith"
02:44.25asterisk9901:15:00 to go
02:44.45*** join/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net)
02:45.36asterisk99juggie: Been to St. John's --- great town!!!! ---- also been to Memorial U campus
02:46.15Juggieasterisk99, i am from st. johns :) its all about the beer
02:46.27Moc[Toronto]heison: Friday at 5pm
02:46.35Moc[Toronto]btw hi there ;)
02:46.38asterisk99Juggie: Beer?!?!?!?!?!?!?    You mean Screech :)
02:47.02Juggiescreech is just what we give to the people who visit, no one from nf drinks that :)
02:47.28Qwellsweet, $275 round trip
02:47.33Qwellnonstop :)
02:47.39asterisk99Juggie: Just so lomg as you're not a spilly-billy with your beer
02:47.47Juggiei dont spill
02:48.18iCEBrkrAnyone know of any eagi examples that work?
02:51.20mmlj4anyone do wireless VoIP? i have a few questions (I know, just ask, etc.)
02:51.25Juggiewireless how
02:51.30Juggiewireless voip phones?
02:51.57mmlj4i need to bridge 2 offices, 50 meters apart
02:52.16mmlj4thinking of using linksys or d-link wireless bridges
02:52.22asterisk99can someone see if voxilla.com is working???
02:52.28mmlj4WET11, for example
02:52.32Juggiethat should be easy 50 m is nothing
02:53.02mmlj4i'm worried about latency... and also packet aggregation, say with G gear (dunno if it uses that)
02:56.46*** join/#asterisk jape (~jpb@sixshooter.v6.thrupoint.net)
02:57.25*** join/#asterisk DrAk0 (~Luis@200.109.195.19)
02:57.40DrAk0hello, can i make asterisk to use my external modem?
02:57.51DrAk0so recive and make the phonecalls from it?
02:58.04thetalondon't go there DrAk0
02:58.12thetalonyou'
02:58.22thetalonyou'll be better off with pure VoIP solution
02:58.33thetalonget a NuFone account
02:58.42DrAk0is formy house i have no money for special hardware so i would like to use what i have in my hands
02:58.44thetalonor some other guy like teliax
02:59.21*** part/#asterisk jape (~jpb@sixshooter.v6.thrupoint.net)
02:59.37DrAk0thetalon, but it is possible what i want?
03:00.15*** join/#asterisk ilium (~ilium@203.94.178.214)
03:00.19asterisk99DrAk0: Look at the Digium WildCard... it'll do it for ya
03:00.27iliumhi
03:00.49*** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net)
03:01.00mmlj4DrAk0: you can buy specialized modems for $17 USD delivered, on ebay (anyone wanna say how good those are?)
03:01.01asterisk99DrAk0: I;d send you a link, but it looks like digium.com is down
03:01.11implicitwow this is fun
03:01.15asterisk99Drak0: No it's back up
03:01.27iliumcan someone have a look at this output from asterisk -c:    http://pastebin.ca/12121
03:01.37iliumi have just installed freebsd 5.4
03:01.46iliumupdated via cvsup to STABLE
03:01.56iliuminstalled zaptel libpri and asterisk
03:02.07iliumi dont know how to troubleshoot this problem
03:02.17mmlj4hey kram: you *might* see me at the bootcamp in florida in december, if I can find the cash to go
03:02.24*** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org)
03:02.52iliumi have also built a custom kernel to include timing
03:02.56asterisk99Drak0: You would need just 1 FXO daughter-card on a Digium TDM400P ($133 from Digium)
03:03.49iliumanyone ?
03:04.03DrAk0the problem is im in south america that means shipping is very expensive plus my country has problem with us$ and getting them is really hard and well im like out of money too, however i have a motorola external modem thas why i was asking if i could use it
03:04.40mmlj4DrAk0: no, those won't work
03:04.42Nuggetilium: do you actually need skinny or oss?
03:04.48mmlj4the problem is drivers
03:05.00DrAk0aahh
03:05.11Nuggetskinny is for cisco phones and oss is for running a local soundcard.  neither are very worthwhile.
03:05.12DrAk0=/
03:05.14iliumnope
03:05.32Nuggetclean up your modules.conf then.
03:05.42iliumhow do i disable them - i have installed asterisk on this box just last week and it all worked
03:05.46iliumi dont know what chaned
03:06.12iliumactually i have a sneaking suspicition i had 1.0.6 installed last week, and this time round i have ended up with 1.0.7
03:06.19Nuggetnoload => chan_oss.so
03:06.23Nugget^ in modules.conf
03:06.25*** join/#asterisk loud (~ariel@omfg.wtf.no)
03:06.59iliumwhat about skinny
03:07.00Nuggetalthough I'm not sure why you'd need to say that specifically
03:07.20Nuggetdid you install asterisk from ports or build it by hand?
03:07.26iliumfrom ports
03:07.32iliumis that bad ?
03:07.35NuggetI've only ever done it by hand in freebsd, so I'm unsure what the port might do.
03:07.42NuggetI don't know if it's bad or not, I've just never done it
03:07.56iliumive been working woth the guy that ports it to freebsd
03:08.03iliumhe has been helpful, so i was useing his ports
03:08.09Nuggetmakes sense.
03:08.17iliumhow do i remove skinny
03:08.29iliumit starts at least after removing oss
03:08.31iliumthatnks!
03:08.38Nuggetcool.
03:09.04ilium<PROTECTED>
03:09.04iliumodule: Unable to open IAX timing interface: No such file or directory
03:09.14iliumi would assume i should be using IAX
03:09.29iliumis modules.comf the only place this stuff loads form
03:09.42NuggetI don't know what that means, sorry.
03:09.58Nuggeton my freebsd asterisk servers I have never had to change modules.conf at all, so I dunno.
03:10.12iliumwhat versio nr u running
03:10.18iliumim on 5.4 STABLE
03:10.30Nugget5.4 stable, but it's several months old
03:10.39*** join/#asterisk Avrice (~James@mmds-216-19-41-194.mm.az.commspeed.net)
03:10.43Nugget5.3 actually
03:10.53Nuggetand an asterisk stable (1_0) checkout from the same time period
03:10.59iliumhmmmm
03:11.14*** part/#asterisk Cresl1n (~matt@216.207.245.23)
03:11.16iliumthere is nothing in modules.conf about skinny or IAX2
03:11.24NuggetI used ztdummy for a while, but I don't do anything zaptel now.
03:11.37mmlj4DrAk0: look here, at the Intel modems: http://www.voip-info.org/wiki-Asterisk+Hardware
03:11.59mmlj4you can probably find those (be cafeful of the exact chipset model)
03:12.27DrAk0mmlj4, let me check
03:14.48sudhir492help
03:15.01sudhir492oops, wrong window
03:16.52iliumcan someone point me to a known good freebsd build doc
03:17.16NuggetI dunno.
03:17.24NuggetI checkout the source, make, and then make install.
03:17.34NuggetI've never had a problem doing it that way
03:17.38iliumcan you tell me how to do that ?
03:17.40niZon~seen boris
03:17.41jbotboris <boris@S01060040ca1e5b54.wp.shawcable.net> was last seen on IRC in channel #asterisk, 5d 5h 51m 37s ago, saying: 'what?'.
03:17.42NuggetI just did.
03:18.01ilium~seen darwin35
03:18.03jbotdarwin35 is currently on #asterisk (1d 17h 34m 55s).  Has said a total of 35 messages.  Is idling for 1d 5h 43m 31s
03:18.58iliumNugget: can you tell me how you check out the sauce
03:19.17Nuggetset CVSROOT to the asterisk cvs server
03:19.23Nuggetcvs co -r v1-0 asterisk
03:19.51iliumok so you run 1.0 ?? isnt it up to 1.0.7 ?
03:19.59*** join/#asterisk Kumbang (~ecvs@167.205.24.4)
03:19.59Nuggetv1-0 is "stable"
03:20.01Nuggetnot 1.0.0.
03:20.09iliumok cool
03:20.10Nuggetit
03:20.15Nuggetit's like releng_5
03:20.24Avricewhere/what is the best place/doc to read on hooking up sjphone to asterisk over the internet
03:20.28iliumok
03:21.05*** join/#asterisk tessier (~treed@203.210.216.1)
03:21.17Kumbangguys, i have problem sending fax using txfax
03:21.21Kumbang*CLI>     -- Attempting call on Zap/g0/801 for out_fax@ext-fax:1 (Retry 1)
03:21.21Kumbang<PROTECTED>
03:21.21Kumbang<PROTECTED>
03:21.21KumbangDIS with final frame tag
03:21.21KumbangIn state 9
03:21.21KumbangStart tx document
03:21.23KumbangCFR with final frame tag
03:21.25KumbangIn state 4
03:21.27KumbangStart tx page
03:21.31KumbangRTN with final frame tag
03:21.33KumbangIn state 14
03:23.10*** part/#asterisk lilneon (~tj_r3@cuscon14794.tstt.net.tt)
03:28.23DrAk0mmlj4, a friend who lives near me... has a Intel 537EP
03:28.26DrAk0it works?
03:31.34blitzrageKumbang: generally with that number of lines, etiquette says to use pastebin
03:31.42blitzrageKumbang: for future ref.
03:31.53MikeJ[Laptop]~pastebin
03:31.55jbotsomebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca
03:32.44blitzrageilium: checking out stable from CVS is not the same as grabbing a 1.0.x release tarball
03:33.53blitzrageilium: CVS -r v1-0 is the latest patches to the stable branch, post release - but there CAN be problems, so its not considered STABLE - always use the releases for production systems
03:34.09Kumbangsorry
03:34.34MikeJ[Laptop]Nugget, v1-0 is release, not stable... don't call it stable
03:34.48MikeJ[Laptop]it jsut confuses people
03:35.11MikeJ[Laptop]cmon blitz
03:35.14MikeJ[Laptop]:)
03:36.03Nuggetdigium calls it "the current stable release"
03:36.08Nuggetso I do the same.
03:36.58Nuggetit's equivalent to freebsd's "stable", so the term is hardly unfamiliar to ilium.
03:37.48Nuggetperhaps you're just unfamiliar with how us freebsd nuts use the term "stable"
03:37.52MikeJ[Laptop]everyone is too serious in here tonight
03:38.00blitzrageMikeJ[Laptop]: screw you!
03:38.05MikeJ[Laptop]ass
03:38.09blitzragehat
03:38.16MikeJ[Laptop]don't make me kick your tired ass
03:38.27MikeJ[Laptop]you wanna take this outside?
03:38.29blitzrageI'll beat you on my worst day, and your best!
03:38.38blitzrageMikeJ[Laptop]: meet me at the backstop at recess, you're dead!
03:38.43MikeJ[Laptop]alright, cmon over...
03:38.51MikeJ[Laptop]backstop...
03:38.53blitzrageMikeJ[Laptop]: pffft - you come here
03:38.55MikeJ[Laptop]you are tired
03:39.11MikeJ[Laptop]chicken
03:39.20blitzragebackstop - fencing to stop the baseball behind the catcher
03:39.28blitzragemust be a Canadian term... :)
03:39.40MikeJ[Laptop]canuk chicken!
03:40.01blitzrageyou can't even spell canuck right you lousy yank
03:40.15MikeJ[Laptop]1/2 breed, get it straight
03:40.19blitzragelol
03:40.22MikeJ[Laptop]eh?
03:40.45blitzragedamn straight
03:40.50MikeJ[Laptop]my folks grew up in windsor ontario, eh.
03:41.01blitzrageI grew up in Sarnia, ON
03:41.07MikeJ[Laptop]poor bastard
03:41.12blitzragequite
03:41.15*** part/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net)
03:41.16blitzragenow I live in Toronto :)
03:41.21MikeJ[Laptop]eh?
03:41.32blitzrageha
03:41.46Juggieusa sucks :P
03:41.47blitzrageaye
03:41.50blitzrageaye!
03:42.11MikeJ[Laptop]I have lots of family in toronto
03:42.20NewSole2yankies.... yank... yank... yank....
03:42.43Nuxisounds like a lot of usa-envy going on
03:43.04blitzrageenvy? lol
03:43.13MikeJ[Laptop]it's an insecurity thing... it's ok
03:43.14JuggieNuxi, at first glance phpagi 2 may have solved the defuncts
03:43.15MikeJ[Laptop]:D
03:43.17Juggiei woudnt call it envy
03:43.24Juggiei think i like my health care system
03:43.32Juggieand lack of RIAA/MPAA gayness
03:43.40NewSole2there is no envy looking at shot spots on the walls
03:43.43blitzrageok, I have to go before I say something you regret
03:43.48NewSole2yankies.... yank... yank... yank....
03:43.51blitzragelol
03:43.57blitzrageI like that guy :)
03:44.20NewSole2:P
03:44.24MikeJ[Laptop]Juggie, you mean the one that meany my gradmother had to come to the states in order to get compatent care, instead of waiting 9 months for a surgey and duying.. yeah, ontario's medical care rocks.
03:44.49MikeJ[Laptop]till you acutally need care
03:45.01JuggieMikeJ[Laptop], i've been in the health care system its been fine for me
03:45.08MikeJ[Laptop]#asterisk politics ...
03:45.09MikeJ[Laptop]hehe
03:45.31blitzragewoohoo! finally got that damn windows sticker off my laptop
03:45.36MikeJ[Laptop]well, good luck, they have a bad habbit of not having specialist cuz they all moved to the states
03:45.38blitzragethose things are on there GOOD
03:45.45MikeJ[Laptop]where they can get paid.
03:45.49Nuxistudies show that removing the widows sticker makes the hardware more stable.
03:45.54JuggieMikeJ[Laptop], funny, the guy i needed was available no problem.
03:46.14tangelhow can i let people enter an extension after i answer an inbound analog call?
03:46.23MikeJ[Laptop]dunno, she needed spinal surgury, and there was a 9 month+ wait.
03:46.24blitzragetangel: show application background
03:46.32MikeJ[Laptop]she got in in a week in the US.
03:46.41NuxiThis is the same kind of talk my parents have about which state is better.   It's good to see so much patriotism.
03:46.53Juggieof course, because in the usa the people with money get help and the people who have no money die
03:47.06MikeJ[Laptop]not patriotism, really
03:47.06Juggietheres no perfect system
03:47.21Juggieyou can have money rules, or wait your turn
03:47.28Juggieeither way someone wont be happy
03:47.36tangelblitzrage, those are the commands?
03:52.31*** join/#asterisk wvbroadband (~User@pool-151-205-40-176.clrk.east.verizon.net)
03:57.14iliumthis sux
03:57.16*** join/#asterisk onet (~JohnM@VDSL-151-118-4-13.DNVR.QWEST.NET)
03:57.40iliumcan someone tell me how to download asterisk v1.0 stable and compile it on freebsd 5.4 STABLE
03:58.37*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
03:58.50Mavvieilium: cvsup your ports collection, then cd /usr/ports/net/asterisk and make install
03:59.53iliumyeah had that installed but it wouldnot start, errors about IAX2 module not found
04:00.03iliumgrrrr this is gettign so frustrating
04:00.36Mavvieilium: mind your bloodpressure.
04:00.57*** join/#asterisk Kernel_Core (Raph@236.228.dial-up.xter.net)
04:02.38*** join/#asterisk openfly (mjoyce@judecca.aculei.net)
04:02.42openflyhowdy.
04:02.57openflyanyone here ever worked with cisco 7902 ip phones and asterisk?
04:02.57iliumyeah well its hard when youre delaing with freebsd and asterisk !
04:03.11iliumall i want to do is see it work with 1 * box and two SIP sopftphones
04:03.16ilium2 weeks later ........
04:03.34|Vulture|ilium: whats your problem?
04:03.42iliumi just cant get * to compile
04:03.49iliumand now i dont know where i am
04:03.50|Vulture|in BSD?
04:03.55iliumyeah in freebsd
04:04.03|Vulture|sorry Im a linux guy
04:04.09iliumi have made a custom kernel with extra commands to enable timing
04:04.16|Vulture|you tried the walkthrough on the wiki?
04:04.16iliumwhat linux do you reccomend
04:04.25iliumyeah i think so
04:04.28|Vulture|ilium: I use FC3 works wonderful for me
04:04.31iliumi ve tried liek 10 times now
04:04.34openflyicky.
04:04.34Mavvieilium: try contacting the ports maintainer.
04:04.43iliumyeah richard was helping me
04:04.44openflyasterisk compiled fine on debian stable for me...
04:04.48openflyhasn't done much else yet
04:04.50iliumand at one stage i had it sort of workign
04:04.56openflybut seems to be going well thus far =P
04:04.58|Vulture|install... import fedora keys, yum update...
04:05.10MavvieI don't know any richard, I know a maxim as port maintainer.
04:05.15|Vulture|and then the udev tweak for zaptel and your good to go
04:05.22iliumhmmmmm
04:05.27iliumany else knwo freebsd ?
04:05.45*** part/#asterisk mogorman (~mogorman@207.111.174.1)
04:05.59openflyso yeah... i am looking at a cme setup on a 1760 and link to asterisk unless someone can point me to a working cssp driver for a 7902g.
04:06.07Mavvieif you relax for a second, and tell us exactly what yuo have and what doesn't work, it would be much easier.
04:06.21iliumok
04:06.27iliumi have installed freebsd 5.4
04:06.33iliumupdated the ports with cvsup
04:06.43iliumupdated src with cvsup
04:06.55iliummake buildworld
04:06.59iliumamke installworld
04:07.01iliumrebooted
04:07.27iliuminstalled zaptel
04:07.30iliuminstalled libpri
04:07.30Mavviewith regarding to the asterisk port.
04:07.33iliuminstalled asterisk
04:07.42iliumwent to start asterisk
04:07.55iliumand had errors with IAX2
04:08.15Mavviebrilliant! now take the verbosity of the first part of your story and use it on your last line.
04:08.28openflyyou ran in verbose mode asterisk?
04:08.33iliumyes
04:08.36openflyand did you get a copy of said errors?
04:08.41iliumhttp://pastebin.ca/12121
04:09.13Mavviewell, that's a FAQ.
04:09.21Mavvie~ztdummy
04:09.22jbotmethinks ztdummy is zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs)
04:09.22openflydid you "make samples"
04:09.42iliumhuh
04:09.47*** join/#asterisk Defraz (~t0tal@65.103.222.4)
04:09.48Flavwhat cards are people using for their home asterisk setups? tdm400?
04:10.09Mavvieilium: did you google for "Unable to open IAX timing interface" ?
04:10.29openflywell thing is i don't see it loading any valid config files
04:10.29iliumno i cam to the asterisk irc channel to ask if people had seen these error before
04:10.31openflyat all..
04:10.38Mavvieilium: try that first.
04:10.42ilium?
04:10.59iliumisnt this irc channel supposed to be used for help too ?
04:11.10Mavvieno, you came to the asterisk irc channel bitching about that asterisk didn't work on FreeBSD.
04:11.27Mavviewithout giving any relevant information.
04:11.31iliumthis is the exact reason i stay away from these channels
04:11.37iliumbeing flamed by people liek you
04:11.44MavvieI haven't even started flaming yet.
04:11.46iliumi wasnt bitchin i was asking for assistance
04:11.49nine76a valid answer has been given though ilium
04:11.55iliumwhatever
04:12.04openflyilium msg
04:12.05MavvieYou are the weakest link... Goodbye!
04:12.13openflymavvie you are being a jerk.
04:12.17Mavvie(not a real great one, but that was a flame)
04:12.17openflyand antagonizing him
04:15.48iliumhey mavvie - to add insult to injury it would seem you are harrasing a fellow aussie !
04:15.58iliumbarnet.com.au ????
04:16.16MavvieI'm not an aussie, I'm a dutchman, stuck on this goddesserted island.
04:16.17Mavvie:-)
04:16.32iliumthat explains everythign then
04:16.39Mavvieare you in the Sydney area?
04:16.42iliumbrisbane
04:16.57Mavvietoo far, otherwise I would have come over and helped you.
04:17.09Mavvienice counter flame :-)
04:17.11iliumit would have been easy to find
04:17.31iliumdriev along corronation drive and look for the freebsd server lying in the middle of the road
04:18.33*** join/#asterisk Godsey (lanny@goofball.md5.com)
04:19.05MavvieNow that shows you're been in the sun too long... You could have reused the harddisk.
04:19.35iliumno that was destroyed from me punching the box so hard before i threw it on the road
04:20.19openflyright... was that before or after you too hunted crocodiles and fought kangaroos with dead wallabee skins?
04:20.20openfly=P
04:20.32MavvieImpressive. These things are supposed to survive 5G. Shows that you're more muscles than brains.
04:20.42iliumit was a big hit
04:20.50openflythat's not a server....
04:20.54openflythis is a server!
04:20.56openfly=P
04:21.02Mavviehehe
04:21.19iliumactually........i htought it woudl be funny to scare a fellow work mate, and dropped a phone book on his desk
04:21.31iliumclunl cluck clunk
04:21.32openflykill him?
04:21.38iliumhard drive died
04:21.53iliumi fent bad
04:21.54openflythere was a harddrive in the phone book?
04:22.05openflydamn the outback has some crazy technology...
04:22.08Mavvieopenfly: they're very advances there in .au
04:22.31iliumwe only got power on over here last month, so we are still learning all about computers
04:22.41openflymust be those aboriginees i heard they have been stealing alien technology for years.
04:23.35openflyyeah so question.
04:24.06openflyshould i a.  try to link my 7902 ip phones to asterisk directly, or b. run them through cisco call manager on a 1760 and link that to asterisk?
04:24.10openflywhich will be easier?
04:25.04MikeJ[Laptop]openfly, which protocol?
04:25.22openfly7902 has to be skinny
04:25.25openfly=(
04:25.49openflysccp driver is confirmed as far as i know on the 7902
04:25.55openflybut may work on 7905.
04:26.10openflyerr
04:26.14openflyunconfirmed
04:26.15openfly*
04:29.44openflymeh...
04:29.59openflyi guess i should try the cme way first..
04:30.02*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
04:30.12openflygood study for ccnp =P
04:31.12*** join/#asterisk iswm (iswm@iswm.user)
04:32.06*** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net)
04:32.14*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
04:32.21MikeJ[Laptop]the skinny drivers are there, but are immature for some phones, I have been using sip firmware to *, but you loose some of the functionality like that...
04:32.24MikeJ[Laptop]sooo....
04:32.27[hC]Is anyone here using a dial macro that handles both their incoming and outgoing calls? Most peoples examples have one macro for in, and one for out.
04:33.20Mavvie[hC]: for PRI channels?
04:33.26[hC]For anything and everything
04:33.33[hC]I have written one, but ive run into a bit of a snag.
04:34.15Mavvie[hC]: I know it can be done for PRIs, but it's hard for with FXO/FXS's because you don't know where they are coming from except for the context assigned with them.
04:34.26[hC]Right, and thats what i test based on
04:34.26ManxPower.
04:34.28[hC]source context
04:34.39[hC]which is alright, as long as your pbx doesnt originate calls from inside services
04:34.55[hC](call return, call forwarding, etc) cause then you have to keep track of every context that a call can originate from
04:35.17[hC]i might just split it out into two macros again, I was just wondering if people had really done it, and if so, i wanted to take a look at how.
04:38.56*** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au)
04:39.59openflyhrmm...
04:40.01openflyodd...
04:40.34openfly7902 + netgeat fs8p 4 port poe switch != powered 7902...
04:40.42sudhir492how to save a bunch of variables in userfield in CDR?
04:41.23sudhir492documentation is very incomplete in wiki
04:41.44openflyhehe its a wiki when you find out.. complete it =P
04:42.24sudhir492sure. I have updated wiki before
04:47.24niZonopenfly: cisco uses some weird POE protocol
04:48.03Juggienot anymore
04:48.04Juggiethey did
04:48.12Juggiethey use the standard now
04:48.15niZonah ok
04:48.21niZonmaybe he has an old 7902 :P
04:48.36Juggiemy 7960's support both
04:48.43Juggiethey run on cisco poe and normal poe
04:48.50niZonI wouldn't mind a 7960
04:48.50*** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com)
04:48.57niZondo you use chan_sccp?
04:49.11Juggieno
04:49.15niZonSIP?
04:49.17Juggie7960 has a sip firmware
04:49.27niZonah
04:49.31niZonyes
04:49.48Juggiei dont like the 7960 anymore tho
04:49.57Juggiethe mitel 5220's are alot easier to manage/maintain
04:50.13niZondo they have the XML stufF?
04:50.24Juggienope
04:50.29Juggiebut i dont care about that
04:50.36niZonah
04:51.01Juggieif it was my decision everyone would have softphones
04:51.03niZonI want a phone that supports XML stuff
04:51.13Juggiexml stuff isnt that exciting
04:51.21niZonyeah but I have a use for it
04:51.23Juggiethe good thing is it can do a directory
04:51.25Juggieother then that
04:51.30Juggiewhats your use
04:51.46niZonautiomation, lighting controls and stuff
04:51.50niZon-i
04:52.04Juggiewrite an attendant
04:52.22soundguyI didnt know lighting controllers used XML
04:52.26soundguywell..none that I have used have
04:52.31niZonpff
04:52.42niZonsoundguy: I write/build my own controllers :P
04:52.47Juggiewhy would u use a phone
04:52.54niZonwhy not?
04:52.55Juggieto do something you could do on a real computer
04:52.57Juggieand easier
04:53.00soundguyheh..i just use a hog1000 :)
04:53.02ManxPower~docs
04:53.03jbotDocumentation 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
04:53.24niZonwhy would you want to go to a computer, turn on the monitor, pull up IE or a control app and do stuff when you can press 2 buttons on a phone
04:53.35niZon(this is in a home setting btw)
04:53.53Juggieyou could argue either way ;)
04:54.10niZonyeah
04:54.21Juggiei would argue a phone just wants to be a phone
04:56.09niZonor it could be a control terminal and a phone
04:56.10niZonlol
04:56.44niZonit's a matter of prefrence :P
04:58.22|Vulture|anyone know how you use ifconfig to set an adapter to tagged on a VLAN?
04:59.54*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:00.58openflyhrmmm
05:02.53[hC]I thought that if you had a type=user and host=dynamic in iax.conf, that that user HAD to register to you, but that does not seem to be the case. Is this true?
05:04.40*** join/#asterisk HA (~mirc@adsl-70-244-228-14.dsl.tulsok.swbell.net)
05:04.54Mavvie|Vulture|: ifconfig myvlan create, ifconfig myvlan ip.ad.dr.ess netmask ne.tm.as.k vlan tag vlandev interface
05:04.57Mavviebut that's FreeBSD.
05:05.44|Vulture|okay ill try that
05:08.35ManxPowerniZon, I think some people get aroused by using a softphone.  It's the only logical explaination.
05:08.55ManxPowerPersonally I think hardphones are better.
05:09.12Juggiearnt you going to europe :) go and stop bugging us
05:11.15openflyjust spent mad bank on a poe switch...
05:11.26openflyand the darned thing doesn't appear to work with these freaking phones.
05:11.28openfly=(
05:11.32openflyimma go cry i think
05:12.44ManxPowerJuggie, I leave tomorrow night
05:13.11Juggienice..
05:13.13Juggiehave fun
05:14.50*** part/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com)
05:18.22*** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net)
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05:27.30bewestwhich features of asterisk require a working sound card?
05:27.57Qwelldialing from the console
05:28.06bewestdoes playing a sound require alsa/soundcard to be working?
05:28.13Qwelldon't think so
05:29.05Juggieno
05:31.26*** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net)
05:31.45shepherdi think it requires oss /dev/dsp
05:32.54shepherdfor like music on hold maybe
05:33.13*** join/#asterisk felipex (~dsfdsf@host162-91.pool8533.interbusiness.it)
05:34.03Juggieno
05:34.05Juggieit does not
05:34.18Juggieabsolutely no sound card required
05:36.24*** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com)
05:37.00bewestwhen you do something like sip show channels, and it says 0 channels active. what exactly does that mean? that nothing is currently being used or nothing is set up to work?
05:37.11*** join/#asterisk lehel (~lehel@82.79.20.17)
05:47.06*** join/#asterisk lehel (~root@82.79.20.17)
05:47.28*** join/#asterisk rajo (~rajo@bfs.cs.uni-sb.de)
05:47.44*** part/#asterisk lehel (~root@82.79.20.17)
05:54.35nine76bewest: nothing in use
05:54.41Flavcan asterisk record phone calls as .wav files or the like?
05:54.47nine76yes
05:54.52nine76search wiki for "monitor"
05:54.56Flavnine76: thanks
05:55.10nine76np
05:57.03*** join/#asterisk kb1_kanobe (~krisbouti@d154-20-43-155.bchsia.telus.net)
06:04.47*** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com)
06:17.47bewesthow do I know for sure if my sip phone got registered with my asterisk server? and is there a way to do a simple hello world type thing
06:18.09bewestlike, I pick up the phone, and asterisk simply plays a sound and hangs up
06:18.21nine76sip show registry
06:18.33nine76err
06:19.16nine76as for the hello world thing, create an exten which plays a sound. i.e. exten => 1234,1,Playback(beep)
06:19.28nine76then dial 1234
06:19.42bewestok
06:19.48nine76if your connected to your * with verbosity enabled you can see the phones register
06:20.11bewestif it was exten => s, 1, Playback(beep), would that play without dialing anything?
06:20.32nine76"-- Registered SIP '1002' at 192.168.0.1 port 5060 expires 1800"
06:20.43nine76exten => s,no
06:20.49bewestok
06:21.09nine76I do not know of a way to have * do what your saying,without having the phone dial some type of number.
06:21.18bewestI see
06:21.48bewestit's moot cause my phone hasn't registered :-(
06:22.09nine76"sip debug" may help,as does a packet sniffer
06:22.50bewestthe verbosity that you mentioned, is that * setting or a phone setting?
06:23.19nine76that would be starting * with v's. or connecting to an already running * with v's. i.e. asterisk -vvvvvvvvvvvr
06:23.27*** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net)
06:25.06bewestok
06:25.16*** join/#asterisk jo3sm1th (Incogn1t0@200.77.206.43)
06:26.08jo3sm1thDoes Firefly have a version that supports call waiting the one I got now only allows one line at a time which is weird since most VOIP providers allow you more than 3 lines at a time
06:29.19justnulling2can't get sipura spa-1001 connect to astersk, any ideas?
06:30.05nine76you followed the configuration example on the wiki?
06:31.18justnulling2nine76: can't find it any more do yo uhave url?
06:32.59nine76I'm looking. Cant seem to find exact docs relating to the 1001. Have you tried the 2000 example? http://www.voip-info.org/wiki-Asterisk+phone+SPA-2000
06:33.30nine76I would imagine they are very similar...just 1 less line on the 1001.
06:34.39justnulling2makes sense, let me try
06:36.32*** join/#asterisk north_ (faith@once.athief.net)
06:37.04Qwellhmm
06:39.17north_Oh, apparently we share the same ~
06:39.46*** join/#asterisk rue_mohr (~dan@ip-215-222.ppp.ucc-net.ca)
06:39.58rue_mohrkb1_kanobe:  hi
06:40.08rue_mohr?
06:40.42kb1_kanobevening
06:40.51rue_mohrahh, excellent
06:40.55rue_mohrI have words for you
06:41.07rue_mohrwireless context
06:41.26*** join/#asterisk af_ (~af@ip-132-224.sn2.eutelia.it)
06:41.27rue_mohrsoekris net4801   and monowall
06:41.35rue_mohrheard?, know of?
06:41.52*** join/#asterisk af_ (~af@ip-132-224.sn2.eutelia.it)
06:43.43kb1_kanobeback - sorry, was stirring soup. :-)
06:44.00jo3sm1thI can't believe Firefly apparently doesnt have call waiting features
06:48.11*** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk)
06:48.15ellvisgood morning
06:48.47rue_mohrgood night
06:48.58*** join/#asterisk Corydon76-home (grey@pcp08665860pcs.500ash01.tn.comcast.net)
06:49.48ellvisi am experiencing no tones when i call the numbers. no ringing tone, no busy tone and so. where should i look to find a problem? i am using hisax for ISDN
06:51.40nine76console output provide any clues?
06:52.54ellvisnine76: no. when i'll call you, i'll hear nothing, when you'll pick up the phone then the call work normally, we can speak and so.
06:53.27*** join/#asterisk hellop (~hellop@cpe-70-93-40-171.hawaii.res.rr.com)
06:53.32ellvisnine76: audio is just fine, except the the ring/busy tones (is that what "signaling" mean?)
06:54.14nine76strange behavior. I dont know the fix:-/
06:55.09*** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net)
06:55.25ellvisok, i'll keep on digging. thanks for a try anyway
06:57.39bewest<PROTECTED>
06:58.03nine76standard * install, /var/lib/asterisk/sounds
06:58.20*** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com)
06:59.49bewestyay :-) it works
07:00.05nine76progress,excellent:)
07:00.24tzafriranybody using the "classical moh" files from signate? http://www.signate.com/moh.php
07:00.51tzafrirThat pag is not very clear about their copyright/license status: are those files in the public domain?
07:01.05*** join/#asterisk lehel (~lehel@82.79.20.17)
07:01.12lehelhi
07:01.27nine76hello
07:01.42leheldo you use AMP?
07:02.03nine76no sir
07:02.13nine76tzafrir: notice the "Signate Download Terms for Free Music on Hold" paragraph?
07:02.45RaYmAn-Bxtzafrir: it seems to me that they basically don't have any rights for it..So if some of the music/names is otherwise protected by the copyright owners (i.e. it hasn't expired) then you can't use it without license
07:02.47nine76Isnt very clear,but I'm not a lawyer:-/
07:03.38RaYmAn-BxI would guess that the copyright on that music has expired though..
07:05.05tzafrirdownloading them now...
07:05.52hellopOk, I almost got Asterisk Working!
07:06.08hellopBut, this time around my X100P is not responding.
07:06.18nine76"zap show channels" ?
07:06.26hellopHow can I check it?
07:06.27bewestI can't even get to signate
07:06.29bewestI've been trying to all day
07:06.38hellopWho wants to help me?  eh?
07:06.48justnulling2i got sipura spa-1001 to work with fwd but not with astersik, can this be b/c i am behind nat?
07:06.48hellopCome on, whoooo want to help?
07:07.03Jas_Williamshellop: try ztcfg -vvvv
07:07.07hellopOutput of Asterisk here: http://pastebin.ca/12060
07:07.09nine76possibly just,is your * server also behind nat?
07:07.14Qwelljustnulling2: What is behind NAT, ata or *?
07:07.18hellopJas_Williams,   output of ztcfg at that site
07:07.23justnulling2ata
07:07.57hellopJas_Williams,  1 channels configured.
07:08.14justnulling2qwell: sipura is behind nat
07:08.25Jas_Williamshellop: type zap show channels in the cli
07:09.19hellopNo such command 'zap' (type 'help' for help)
07:09.47hellopJas_Williams, so like, zap is not loading
07:10.04Jas_Williamshellop: Zaptel module is not loading compile libpri and zaptel then make clean make install in asterisk to ensure zaptel built into asterisk
07:10.53hellopJas.. ahh  I didn't do libpri, because the howto at Sineapps didn't say to.
07:11.02hellopZaptel howto did...
07:11.16Jas_Williamsyou need libpri
07:11.22hellopwhats it for?
07:11.24brad[]If I send a fax or run a debit machine on the same PSTN line as an asterisk Zap channel, the asterisk server seems to ring all the connected IP phones while the data transmission is taking place. Anyone run into this?
07:11.32nine76I didnt compile libpri either,just zaptel & *. scary
07:12.03hellopnine76, huh?
07:12.05Jas_WilliamsIt is used to set reason codes and decodes for PRI based zap channels without is chan_zap will not compile
07:12.49hellopOk, hey thanks Jas_Williams
07:13.00nine76What functionality have I lost by not having libpri compiled?
07:13.32Jas_Williamsnine76: do you have chan_zap ?
07:13.53QwellI've never installed libpri, and I use zap
07:13.58[hC]I think ive forgotten a step someplace. For some reason when i try to launch MeetMe from my dialplan, it doesnt go anywhere, and a show application MeetMe says there is no meetme application available.
07:14.07[hC]Is there  a trick to meetme?
07:14.16Qwell[hC]: are you noloading some stuff you need?
07:14.24nine76yes,chan_zap is there,and all seems to work fine.
07:14.34kapejod[hC]: do you have a zaptel timing source?
07:14.40[hC]kapejod: yes.
07:14.47[hC]Qwell: i believe im using the default modules.conf
07:14.53Qwell[hC]: check
07:14.53[hC]Maybe i need to specify
07:15.03*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
07:15.03Jas_Williams[hC]: meetme.conf ?
07:15.04kapejod[hC]: ls /usr/lib/asterisk/modules/app_meetme.sol
07:15.34hellopThe FAQ at digium just does cvs checkout zaptel asterisk.. no libpri, thats how I got the X100P working before.
07:15.34[hC]kapejod: negative. Do i have to specially mark it as enabled at compile time?
07:15.50*** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net)
07:16.15Jas_Williams[hC]: Meetme will not copile without zaptel being compiled
07:16.25[hC]Aha. Thats why. I did zaptel after.
07:16.27[hC]Thanks!
07:16.40[hC]I knew it was something silly i'd missed.
07:17.55*** part/#asterisk lehel (~lehel@82.79.20.17)
07:19.06justnulling2any idea how to get sipura work from behind nat with asterisk?
07:19.28nine76mine works fine justnulling2. added nat=yes....thats about all...
07:20.40*** join/#asterisk dg1nsw (~schulte@gate.sympat.de)
07:20.41justnulling2nine76: tried that no luck there is nothing in the * logs so something is really wrong and looks like nat
07:20.46marcus5anyone know of outbound sip/iax providers that support 900 calls?
07:21.12nine76have you enabled "sip debug" and tried to trace down an error?
07:22.29*** join/#asterisk IsMe (~me@218.111.224.205)
07:22.32bewestwhat's the difference between host and permit?
07:22.37IsMehi guys
07:22.44nine76hello
07:23.34IsMeany developer here who can do a SMS callback ?
07:23.54*** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com)
07:24.23justnulling2nine76: on sip debug has something there
07:24.31kapejodIsMe: that is trivial with some little scripts
07:25.03IsMekapejod: i cant do it, not this year nor the next million year
07:25.04IsMehehhe
07:25.33kapejodare you talking about fixed line sms, or sms in general?
07:26.12IsMesomething like GSM user trigger a SMS , * attach to some kinda GSM device, receive the SMS and call back to the originator and wait for destination number
07:26.41kapejodwhy wait for a destination number?
07:26.47kapejodyou can send that in the sms....
07:27.06IsMekapejod: ok, that will do too, but user are generally too lazy
07:27.25justnulling2nine76: it is 401 unathorized, why is that?
07:27.27kapejodbut this way they can use the details from their phonebook
07:27.41nine76perhaps a username/pass mismatch?
07:28.03IsMekapejod: well, i am assuming my users are lazy bunch
07:28.21IsMeeither that or a ANI call back
07:28.24*** join/#asterisk cmk (~cmk_@p54A3BF70.dip.t-dialin.net)
07:29.15*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
07:30.18justnulling2nine76: used the same user/pass ones with xten which works
07:30.20kapejodif you can receive the sms on fixed line than you can do it all with just *
07:30.30kapejodand 1 script
07:32.35*** join/#asterisk at561 (~oishf@68.71.213-37.atlsfl.adelphia.net)
07:33.13IsMekapejod: sms on gsm
07:33.17at561has the e911 crisis been discussed to death already
07:33.18IsMenot fixed line
07:33.36bewestwhat's the difference between host and permit?
07:33.45tengulrewho can give me some tips?
07:34.24hellopWhat's the word on getting some yes/no  0-9 Voice recognition in Asterisk?
07:34.27kapejodIsMe: than it's just a matter of how you connect the gsm devices to your box
07:34.46kapejodhellop: try sphinx
07:35.26tengulresmstools?
07:35.56IsMei just bought a ericsson fixed celluar terminal
07:36.15*** join/#asterisk Cheetah (~Snak@62.217.48.111)
07:36.18IsMebeen googing around, no luck yet
07:36.20hellopkapejod, k..  I've been playing with Voice Recognition for a while..  I have yet to find something that will just match yes/no, instead of looking up entire festival phenome dictionary.
07:36.21Cheetahhey folks :)
07:36.37Jas_Williamsbewest: host=XXX means that is the IP Address of the host, Permit gives a range of permitted IP Adreesses when you don't know what the IP address is.
07:37.01QwellJas_Williams: to expand on that.  When you know what the different hosts the user will connect from
07:37.05Cheetahi just installed the speex codec on my server and was wondering if 1) how much hardware phones are compatible with it, 2) where I can configure the codec on the server side?
07:37.21kapejodhellop: i used to work with some siemens and falcon gsm modems for sms
07:37.39bewestok thanks
07:37.40QwellJas_Williams: unless its one range, cdr...in which case, I'm quite wrong
07:37.44IsMetengulre: sms tool is a sms server, no ?
07:37.52hellopkapejod, does sms have something to do with voice recognition?
07:38.07kapejodhellop: sorry, wrong nick ;)
07:38.25Jas_WilliamsQwell: The range is based on the mask value
07:38.46QwellJas_Williams: ahh, I just assumed it could be comma/space seperated
07:38.51IsMekapejod: nothing related to sms/ani call back anywhere, care to point me to the right direction ?
07:39.45IsMei already have * running as prodution
07:40.29Qwellhmm, if it isn't, maybe I'll make a patch
07:40.40kapejodIsMe: i have some simple, old outdated callback example on my website. with sms it is then just a matter of generating the outbound call (manager interface or pbx_spool)
07:41.05hellop<PROTECTED>
07:41.21hellopI just ran that command without the -lz and then asterisk compiles.
07:41.31hellopOh the error is:/usr/bin/ld: cannot find -lz
07:41.38hellopIs there a more correct way?
07:41.49justnulling2nine76: ok see the problem asterisk is sending packets to local ip of sipura and no wonder they don't get to sipura
07:41.50IsMekapejod: mind showing me ?
07:42.03Flavthere's a list of cheap providers and also a list that usually mentions whether IAX is supported, but "cheapest providers of IAX" doesn't seem to exist at the moment - that'd be a nice list :)
07:42.05Qwellhellop: looks like zlib or something
07:42.18QwellFlav: cheapest is usually crap
07:42.46Qwellhellop: have you tried googling it?
07:42.48kapejodIsMe: http://www.junghanns.net/asterisk/page14.html
07:42.54kapejodIsMe: it's for capi though
07:42.55FlavQwell: hmm, true - cheapest that isn't crap, perhaps?  like, what's used by many/most of the people in here?
07:43.03QwellFlav: I like nufone
07:43.14hellopQwell  yeah.. but..   gcc -lz   doesn't give too good of results
07:43.27ZeeekFlav : nufone, voipjet, voicepulse connect
07:43.29hellopI omit the lz and it compiles.
07:43.36Qwellhellop: well, google it
07:43.42hellopI then noload dundi.. some mayeb for now its not needed.
07:43.56QwellI'd bet you're missing zlib or something
07:44.06justnulling2what ports do i need to open in my firewall for astersik to work?
07:44.17Qwelljustnulling2: depends on what you're doing with it
07:44.18IsMekapejod: site seems to be down
07:44.31nine765060 for sip,10000-20000 for rtp,4569 for iax...probably others
07:44.41kapejodIsMe: nope, it's fine here :)
07:44.54IsMekapejod: is ok now
07:45.20kapejodmind my slow backup system. the main server recently died ;)
07:45.42IsMekapejod: so you are the creator, google always point me to that site together with digium user list.
07:45.59justnulling2qwell: i am trying to get sipura from behind nat to work and asterisk is sending packets to local ip of sipura over the internet so it doesn't work
07:46.05kapejodIsMe: creator of what? :)
07:46.25IsMekapejod: of that callback script
07:46.26justnulling2nice76: all re udp?
07:46.28Cheetahany someone help me with this codec issue? ;)
07:46.33kapejodIsMe: yes
07:46.35nine76yes
07:46.49IsMekapejod: still could not figure out wtf is that. hehhe
07:46.55Zeeekjustnulling2 : did you check the wiki page about NAT?
07:46.57Qwelljustnulling2: sounds like you need to read up on how to get nat working
07:47.09kapejodlol
07:48.44hellopQwell, before, when compiling pbx_dundi, I got error: zlib.h not found.  I fixed that by copying zlib.h from my kernel source to /usr/include/linux.  Now, this -zl error. Any ideas?
07:48.58kapejodRoyK: you are way too friendly this morning
07:49.39Qwellhellop: different file.  install zlib
07:49.40justnulling2yes i have to read lots of stuff on nat and * in general
07:49.53Zeeekjustnulling2 and what have you done so far?
07:49.56hellopQwell, pretty sure it is, maybe I need devel
07:50.05Qwellprobably
07:50.09Zeeekjustnulling2 and with what results?
07:50.11RoyKkapejod: always friendly :)
07:51.03hellopThere is only zliblg-dev...
07:51.22Flavhellop: you installed that already?
07:51.49Qwellhellop: for the 3rd time.  Have you googled the error?
07:51.59justnulling2zeek: changed nat=yes to nat=0 as nine76 said it might help and well it didn't help, any thing else you can think of?
07:52.01at561does 2.6.11.8 lock up for anyone else
07:52.28hellopQwell..  well yes.  But, I'm trying different searchs now.
07:52.50Qwell~google /usr/bin/ld: cannot find -lz
07:52.55QwellI swear to god, if its the first hit...
07:53.16hellopIt sems I need a kernel recompile.
07:53.58Zeeekjustnulling2 : check this out http://willypick.mindsay.com/?entry=10
07:54.32Qwell"you have to install the zlib1g-dev and/or zlib1g package"
07:54.45*** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu)
07:54.59hellopQwell, yeah thats what I figured  thanks for the help.
07:55.06justnulling2zeek: i don't have double nat, * is located on the internet only sipura is behind firewall
07:55.35Zeeekyes, and that page doesn't mention the lines needed in sip.conf
07:55.37Jas_Williamsjustnulling2: enable stun on the sipura
07:56.12CheetahQuestion: Where can I set up the configuration for codecs like speex? ;)
07:56.33Cheetahi read that there was something like codecs.conf, but it seems to have become obsolete
07:58.46Zeeekjustnulling2 to answer your original query, 5060 forwarding should work on the phone end, but what result do you get? Does the phone register, is there one-way audio?
07:59.49Jas_WilliamsZeeek: justnulling2's phone does not post its external IP address in the sip packets, he needs to enable stun correctly on the sipura
08:00.54ZeeekJas_Williams 1, hello, 2, I just realized he said that and it didn't sink in and 3 I don't use STUN but the GS somehow figures it out...
08:01.06Zeeekjustnulling2 sorry as I say I didn't read you right
08:01.57Zeeekother than using STUB there should be a NAT setting of some kind in the phone setup?
08:02.02ZeeekSTUN
08:05.08nine76on my spa-2000 there is a NAT setting,under "line X". stun settings are under "sip". I have not needed to enable stun nor nat on the sipura side to get my setup working. 2000 behind nat -> * no nat. just a nat=yes in * sip.conf :-/
08:06.36hellopQwell, it's compiling now, tks.
08:06.44*** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk)
08:06.47ellvisre
08:08.20ellvisi am having problem with ISDN line, i have no ring/busy tones at all. the voice is good in both direction, the signaling is not working at all. anyone can help me where i should take a look for solution?
08:08.30*** join/#asterisk Martohtar (Martohtar@82.196.218.80)
08:08.53Jas_Williamsellvis: is this PRI ?
08:09.38*** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net)
08:11.01Qwellbed
08:11.07Qwellhellop: I accept paypal
08:11.27*** part/#asterisk buu (foobar@69.56.150.34)
08:11.55hellop$20 if you can get my SIP phone to ring/dial.
08:13.08hellopWow, Asterisk is running with no error msgs..
08:13.39ellvisJas_Williams: no, BRI
08:14.00ellvisJas_Williams: Eicon DIVA 2.01 with HiSax
08:14.41hellopwell it rings but doesn't call out,  yet.
08:14.48Jas_Williamsellvis: use chan_capi or get a card that supports zaphfc
08:15.10*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
08:15.32ellvisJas_Williams: i'd like to, in next box for sure, i am just curious right now if HiSax itself is not supporting it or if i have a mistake somewhere
08:17.22ellvisJas_Williams: anyway, those eicon cards are not usable with capi under linux at all
08:18.47shido6hellop
08:18.49shido6$20?
08:18.51*** join/#asterisk oej (~oej@213.204.186.40)
08:18.55bewest'
08:20.40bewest.
08:20.46hellopshido6, I would be willing to pay for support.
08:20.46bewest7
08:21.09Jas_Williamshellop: what state are we now in ?
08:21.16hellopHawaii..
08:21.29bewestconfusion
08:21.50Jas_Williams:-) no I'm speaking Uk english ;-P what is the state of your X100P now
08:22.29*** join/#asterisk Romik (~romik@1.fix.netvision.net.il)
08:22.37*** part/#asterisk Kumbang (~ecvs@167.205.24.4)
08:22.54ellvis:)
08:23.34bewesthow do you know whether to use sip or iax? I'm trying both and registered for both
08:23.44Flavanyone know of a place that provides listings of voip providers by area codes they have coverage in?  filtering the ones I could use LNP of my existing number manually is slow going :)
08:24.24hellopJas_Williams, oh, it accepts calls to voicemail and the phone, but outgoing just rings busy.  I can probably figure it out.  I need help with some scripting things tho.
08:24.41bewestI mean, I'm paying for a service that I'm trying to connect to using my own *, when I do iax show register and sip show register, both appear to be registered
08:25.05bewesthowever, when I call the number that the service provides, my * says it's rejecting that ip
08:25.26hellopI would be willing to pay (cheaply, I'm just a student)   for figuring out how to do automated outbound calling, determine weather voice or machine, and leave a msg or interact with user.
08:25.36shido6something is really wrong bewest , pastebin.ca your dialplan and iax.conf/sip.conf
08:25.42*** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl)
08:25.55*** join/#asterisk lichen (lichen@aequitas.ipclassless.com)
08:26.23*** join/#asterisk makkia (~pippo@nat.xsec.it)
08:26.25makkiahello
08:27.16hellopBTW, this is not for malicious.  This is for my own business.  When a customer calls.  I'm trying to make an automated voice attendent.
08:29.13hellopIs there a way to have extensions.conf run a shell script?
08:30.06RestLessGeminiHellop: use System command, i've never tried it before though, but i used it to reload asterisk config
08:30.23fenlanderhttp://www.voip-info.org/wiki-Asterisk+cmd+System
08:30.32licheni'm guessing the cisco ata 186's that are specified for use with vonage are locked and cann't be used with any other services?
08:30.41hellopgreat, tks
08:31.06*** join/#asterisk Cheetah (~Snak@62.217.48.111)
08:31.08RestLessGeminilichen: yes you are rite, but you can unlock them if you have the password
08:31.29*** join/#asterisk Heretic666 (~Miranda@195.128.153.131)
08:31.40RestLessGeminiask vonage how to get them unlocked, i think you need to  pay them for ata 186, then they will give you the password to reset it
08:31.59Cheetahwhere do I configure codec settings?
08:32.30lichenif it's cisco equipment why can't you just do a password recovery?
08:32.39RestLessGeminiCheetah: in sip.conf or iax.conf
08:32.44lichenor is such a feature not availble becuse it's "vonage"
08:32.55RestLessGeminilichen: you can not do this until you have the password its locked with
08:33.39Jas_Williamshellop: post your extensions.conf to pastebin.ca for me to look at
08:34.12bewestok shido6: http://pastebin.ca/12131
08:34.14CheetahRestLessGemini, i installed the speex codec, and read that there is some kind of codecs.config; this is not the case anymore, right? because I can't find this file in the asterisk distri or somewhere else anymore
08:34.30lichenahh i gotcha.. also instead of buying an FXS card for my box to plug into my NID could i just buy an ata186 and route calls destined for the analog netowrk through that?
08:35.37RestLessGeminicodecs.conf is there, i've never worked with speex.. sorry :)
08:35.39lichensuch as i have an incoming voip call, could i have a ring a certain call group that would route everything to a 186?
08:36.01RestLessGeminilichen: yes you can do this
08:36.04hellopJas_Williams, http://pastebin.ca/12132
08:36.12RestLessGeminilichen: I do this all the time
08:36.16CheetahRestLessGemini, so, do you have any idea how to configure speex, then? ;)
08:36.27RestLessGeminiCheetah: Nop :)
08:36.32lichenwould i need to have the ata login to the asterisk box?
08:36.36Cheetahsnap :/
08:36.54RestLessGeminilichen: yes .. just like anyother IP-Phone
08:37.14lichengotcha.. it's been such a long time since i've dealt with voip.. i'm trying to pick up all the nuances again
08:37.38RestLessGeminilichen: wishing you good luck
08:37.42RestLessGemini:)
08:37.45IsMeany developer here who can do a SMS callback ?
08:38.09IsMegsm callback
08:38.24Jas_Williamshellop: what device are you using to place the outbound calls and what context is it in ?
08:39.06lichenmy friend has voip service through broadvoice, we're trying to setup something along the lines of having his asterisk box login to the broadvoice server, and then if they're destined for his apt they're route to his ata, otherwise they'll go across a secure tunnel terminating at my router, which then sends it to my asterisk box (the two boxes will be IAX trunked) which will in turn send it to my ATA and then into the NID and ring my analog phones
08:39.09jeffikall: need some help with x-lite
08:39.10*** join/#asterisk Ahrimanes (~aron@hobbes.bsd-dk.dk)
08:41.00RestLessGeminilichen: its very much possible as i've done similar stuff
08:42.19*** join/#asterisk Mimmus (~viggiani@ext.pitagora.it)
08:42.41lichengood deal.. i'm just curious whether or not i'll need to have his ata login to hte broadvoice service or if i should have the asterisk box do it.. i'm guessing we won't have full access to configure his ata, plus i want a voice prompt so the person can choose between either my apt or his, so i guess my best bet is to have the asterisk box pick up the call
08:43.08lichenthen on his end he can either route calls to the ATA or get an FXS port for hte box
08:43.12Mimmushi, I'm just to buy an E1 card for Asterisk and I need some help
08:43.20RestLessGeminino, make ATA login to your *, and * login to Broadvoice
08:43.38lichenokay that's what i figured would be the best bet
08:44.07RestLessGeminiaccept calls from broadvoice to your *, then route them to ATA or to other * box
08:44.09lichenanother area of concern is we can't make multiple concurrent calls.. so i'm hoping we'll be able to stop such a thing in *
08:44.30RestLessGeminilichen: dont even think about it :)
08:44.31Mimmuswhy are different Digium cards for 3.3 and 5 V PCI slots while Sangoma offers a 'universal' card?
08:45.02af_how could I check fo r 3.3 or 5 Volt pci slot?
08:45.08RestLessGeminithey will close your account and send you an overage bill, they've done this with me :)
08:45.27hellopJas_Williams, it's a Budgetone100
08:45.38Ahrimanesaf_: motherboard manual?
08:45.47Jas_Williamshellop: what is it's context
08:45.52hellopcontext is local-access
08:45.52lichenexactly, that's what i'm worried about.. so i'm hoping like if i pick up a line from my * and try to make a call out of our presonal network it will realize that a call to broadvoice is already in progress and not allow it
08:45.52af_Ahrimanes: I don't have it it's an epia 5000
08:45.55RestLessGeminii've tried concurrent calls through my * box and in the end I got my account terminated and they've billed me more then $200
08:46.02Ahrimanesaf_: 2 sec
08:46.40Mimmusrecent motherboards usually have 1 or 0 5V slots :-(
08:47.12af_and a dell 2400
08:47.16af_uhm.
08:47.30RestLessGeminiwell, while making account for outgoing through broadvoice, limit it to just 1 outgoing call, this way it will not let another call go out and gives you a  congestion signal, capture it in dialplan and play appopriate msg
08:47.31hellopJas_Williams, I just got casterisk to compile. so, I haven't had a chance to RTFM on the SIP fone yet.
08:47.31Ahrimanesaf_: dell.com should be able to tell you about the dell
08:47.37hellopwhoops  asterisk
08:47.40af_Ahrimanes: nope :(
08:47.47Ahrimanesaf_: then call support, hehe
08:47.53af_.....
08:48.08Mimmusaf_: Dell 2850 has only 3.3 slots
08:48.27lichennice sounds good.. if he ever gets ip phones i can make it so i can still dial is ip phones even if his fxs is off the hook
08:48.38Mimmusaf_: Dell 2800 (tower) has 1 5V slot
08:48.54makkiammm with MP3Player(...) the quality is of music is good, but with MusiOnOld() is very very bud
08:48.57Jas_Williamshellop: what is the number you are dialling from the phone ?
08:48.59lichenso here's the question.. do you suggest FXS card or ATA186 (or perhaps some other brand of ATA?)
08:49.02makkiais a know problem ?
08:49.02Ahrimanesaf_: via is 3.3
08:49.16af_what happens if I put a board set to 5V in a 3.3 slot?
08:49.21af_thank Ahrimanes
08:49.32hellopJas_Williams, 641-3333
08:49.45jeffikanyody: help with x-lite config
08:50.02MimmusAhrimanes: why do I need to buy a board that fits only in 3.3 or 5V slot (Digium), if I can have one (Sangoma) fitting in both?
08:50.31AhrimanesMimmus: no idea, buy sangoma if this is important to you?
08:50.53RestLessGeminiATA work good, and besides you can not take FXS card anywhere you want, but you can move ATA within the network
08:51.04MimmusAhrimanes: I'd like to buy a Digium card but I'm fighting with this problem
08:51.25AhrimanesMimmus: why, do you need to move the card around between servers alot?
08:51.36lichengood point.. plus i'm partial to just additional network hardware and basically more blinky lights :)
08:51.50Jas_Williamshellop: change exten => _9NXXXXXX,1,Dial(Zap/1/${EXTEN:1})
08:51.50Jas_Williamsexten => _9NXXXXXX,2,Congestion to exten => _9X.,1,Dial(Zap/1/${EXTEN:1})
08:51.50Jas_Williamsexten => _9X.,2,Congestion reload extensions and try again
08:51.52RestLessGemini:)
08:51.55Mimmusmy test server is older and has only 5V slots, production server will be newer and probably it will not have 5V slot :-(
08:52.36AhrimanesMimmus: hm, upgrade test server or buy 2 cards?
08:52.37RestLessGeminibrb
08:53.08MimmusAhrimanes: ok, bit it is annoying (I hope this the right word) :-)
08:53.42AhrimanesMimmus: i know it is.. might face the same kind of problem, but really a new test server can bring more benefits :)
08:55.30hellopJas_Williams, same, busy signal.   But thanks for hint.  It was setup for australia.
08:55.42bewestI have this voip service that I'm trying connect to using *. iax show registry shows that I'm registered, however, when I call the number, I see that * is rejecting the IP
08:55.45MimmusAhrimanes: thanks for support, Sangome offers also a 2-ports card instead Digium only 4-ports
08:56.15Jas_Williamshellop: sounds like your sip phone is not registered
08:56.17MimmusAhrimanes: but I'd like to feel supporting Asterisk creator anyway!
08:56.19AhrimanesMimmus: hehe np.. ok we have 1 4 port digium, but chose to outsource pstn to another company so i just iax now :)
08:56.28*** join/#asterisk tessier (~treed@203.210.216.1)
08:56.36hellopJas_Williams, even tho it rings on incoming?
08:56.44Ahrimanesanyone here doing videotelephony with asterisk?
08:57.21tessierAhrimanes: asterisk has nothing to do with video
08:57.58Ahrimanestessier: oh but it does, videosupport=yes etc
08:58.20fenlanderAhriamanes: yes - we use it for video
08:58.25hellopYannick Tessier?
08:58.34Jas_Williamshellop: do you see any output in the cli
08:58.34Ahrimanesfenlander: ok, with which devices?
08:58.49hellopJas_Williams,  on incoming I do, but not outgoing.
08:58.55*** part/#asterisk dieck (~dieck@saengerschafter.de)
08:59.03fenlanderAhrimanes: softphones - eyebeam and one developed inhouse
08:59.36Ahrimanesfenlander: ah ok, we use eyebeam as well.. but am working on getting some hardphones working now.. but having lots of problems.. heh
08:59.42Jas_Williamshellop: turn on sip debug and then try again post the results to paste bin
08:59.43jeffikfenlander: may i ask you a question avout x-lite?
08:59.56fenlanderjeffik: you can try
09:00.12fenlanderAhrimanes: would be interested to hear how you get on - which phones?
09:00.28Ahrimanesfenlander: innomedia, wooksung and leadtek for now
09:01.03jeffikok, my system sits in chicago, I am i Toronto, i have x-lite and no prob. works fine.  I just set up a friend in Tblisi Georgia with x-lite, he is not behind a firewall or router, he is diredt in
09:01.36jeffikhe can call me, but when i call him i get recording the extension is not availabler
09:02.27fenlanderjeffik: could be a dialplan problem, or maybe his xlite is not registering - what does sip show registry say?
09:02.48jeffiksays logged-in enter number
09:02.54jeffikim runnig aah
09:03.13jeffiki wonder if we should say yes on nat
09:03.26fenlanderI don't know much about aah - you could switch to a shell and run asterisk -r then sip show registry
09:03.43hellopthis bug mean anything? WARNING[9908]: cli.c:967 ast_cli_register: Command 'sipdebug' already registered
09:04.00jeffiki can get to the shell i should run sip registry?
09:04.39fenlanderif there is no nat going on anywhere yous shouldn't need any nat options
09:05.05fenlanderrun asterisk -r from the shell, then "sip show registry" will show you the registered clients
09:05.11Jas_Williamshellop: only a warning do not worry
09:05.27jeffikok
09:06.10fenlanderjeffik: hang on, no it won't ;-) wrong command
09:06.21Ahrimanessip show peers?
09:06.30hellopSIP/2.0 404 Not Found
09:06.52fenlanderthat's what I meant :-) sip show peers
09:07.01Ahrimanesfenlander: priv msg..
09:07.40hellophttp://pastebin.ca/12135  <-Sip fone error.
09:08.00*** join/#asterisk tass (~tassi@ppp163-93.static.internode.on.net)
09:08.56Jas_Williamshellop: proceed the 6407133 with a 9 for an outbound call
09:09.08Jas_Williamshellop: dial 96407133
09:09.18hellopit works
09:09.21tasshello... was wondering if anyone can help me with a digium quad-span E1?
09:09.29hellophehe
09:09.31Jas_Williamstass: in a moment
09:09.38Jas_Williams~jas_williams
09:09.39jboti guess jas_williams is jason@redskycomputing.co.uk, paypal donations accepted
09:09.59Jas_Williamstass: what is your question
09:10.57tasswe have an E1 confirmed working - australian OnRamp10 service, but don't get any lights on the digium card other than the 'knightrider' lights up until the module is loaded
09:10.58hellopJas, can I PM you?
09:11.15Jas_Williamshellop: sure
09:11.15tassas soon as the module is loaded, all lights switch off
09:11.25kapejodtass: did you run ztcfg?
09:12.00*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
09:12.05tassyes... doesn't display any errors
09:12.17tasszttool shows the first 3 spans as 'OK' and no alarms
09:12.26Cheetahi get this in asterisk log:
09:12.27Cheetahspeextolin_framein: Out of buffer space
09:12.35kapejodtass: cat /proc/zaptel/* | grep Span
09:12.42tassasterisk 'zap show channel X' always shows 'InAlarm: 1'
09:13.35tassdo you want me to paste the Grep output?
09:13.38Cheetahany ideas? ;)
09:13.53kapejodtass: yes, sir!
09:13.56tassSpan 1: TE4/0/1 "TE410P (PCI) Card 0 Span 1" HDB3/CCS/CRC4
09:13.56tassSpan 2: TE4/0/2 "TE410P (PCI) Card 0 Span 2" HDB3/CCS
09:13.56tassSpan 3: TE4/0/3 "TE410P (PCI) Card 0 Span 3" HDB3/CCS
09:13.56tassSpan 4: TE4/0/4 "TE410P (PCI) Card 0 Span 4"
09:14.20tassspan 2 and 3 i've tried connecting together with a loopback cable which didn't seem to do anything
09:14.35tassspan 1 has been set up for the PRI
09:14.37Jas_Williamstass: post a pri debug intense span 1 to pastebin.ca
09:14.52*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
09:15.01kapejodtass: did you jumper the card to E1?
09:15.14tassyes, it's jumpered as E1
09:17.14tassunder dev/zap, i can see up to 124 which seems right for E1 which hopefully means its jumpered correctly
09:17.48*** join/#asterisk cjk (~cjk@80.92.64.103)
09:18.38cjkhi, does anyone know a way to get turkish did's ?
09:18.51tassJas_Williams: i've enabled intense debug... is there anything in particular i should do other than putting the cable in?
09:19.18Jas_Williamstass: No just catch some output if any
09:19.28*** join/#asterisk jackthe (~jesse@d594f03e.ftth.concepts.nl)
09:20.48lichenthe hell.. i don't see any of the standard just FXS/FXO cards on digium's site anymore
09:20.55lichennow they're all the modular ones.. did they do away with the others?
09:21.00Flavsorry for a spammy question, but any pros/cons of TelaSIP/VoipXpress?
09:21.47Jas_Williamslichen: 1 port FXO cards AKA X100P is no more. You need to buy the modular cards
09:22.59bewestI don't understand: iax show registry shows that I am registered with my voip provider, yet when I try to make a call to *, it rejects the connection attempt
09:23.03lichenah you're right.. TDM10B is only $125 .. that's nifty
09:27.06Flavhow much was the X100P before?
09:27.17lichenalmost $300 or so i thought?
09:27.22lichenor i could be way off
09:27.30hellopShould my SIP fone be doing 174kbits/s constant to the Asterisk Server?
09:27.53helloplichen, I think it was about 100
09:29.40RoyKhellop: about 80kbps full duplex
09:29.46RoyKso around 160 total
09:30.11RoyKplus, of course, some SIP control traffic, but that'll be minor
09:30.37tassJas_Williams: i haven't seen any debug output at all
09:31.43lichenso i haven't heard much about anything except for the cisco ATA's... are there any others out there or is the cisco the best affordable one
09:32.40Jas_Williamstass: that is strange what is the output of zapcfg -vvv
09:33.48bewestdoes anyone have service with asterlink.com ?
09:33.58Flavstupid question, but any chance you could use a FXO card as an FXS card?
09:34.05hellopRoyK, so I can only have 2-3 calls on my 256kbit upstream?
09:34.23Jas_WilliamsFlav: not without some kind of converter
09:34.24tassjas: it's fairly long, can i pm it to you?
09:34.41Jas_Williamstass: use pastebin.ca
09:35.28FlavJas_Williams: i just noticed the 100X single FXO card is < $10 on ebay
09:35.31RoyKhellop: with g.711, yes
09:35.36Jas_Williamstass: also post zaptel.conf and zapata.conf whilst your there
09:35.57tassjas: done (ztcfg @ http://pastebin.ca/12137)
09:36.04Jas_WilliamsFlav: Yes use a ata adapter for fxs ports
09:36.08bewestin order for * to recieve calls from another iax, do you have to register it?
09:37.12FlavJas_Williams: i was kind of hoping to find a cheapish fxs card so i could do asterisk on the linux box then fxs out to my existing phones
09:37.28lichenflav... i was just looking at that too.  unfortunately i need an FXS card as well.. i think i'm just going to buy an ATA instead though
09:37.37tassjas: config files @ http://pastebin.ca/12138
09:39.33Jas_Williamstass: config files look good
09:39.48Jas_Williamstass: are you connected to the pri at the moment ?
09:39.53tassyes...
09:40.23tassi have the pri through a crossed cable... 1-4 2-5 i think it is
09:40.43tasssame one we have working on a dialogic E1
09:40.48Jas_Williamstass:
09:41.10Jas_Williamstass: is the span 1 going to a carrier or another pbx ?
09:41.18tassjas: carrier
09:41.34Jas_Williamstass: then you need a straight cable to the carrier.
09:41.56tassjas: i'll try a straight cable and check the debug output...
09:43.19*** join/#asterisk kapejod (~kapejod@e178005154.adsl.alicedsl.de)
09:44.47tassjas: swapping the cable didn't change anything, however i did get the following before the swap: http://pastebin.ca/12140
09:46.32*** join/#asterisk Blackvel (~blackvel@dsl-084-057-126-189.arcor-ip.net)
09:46.35jeffikfenlander: it worked
09:46.37Jas_Williamstass: Your PRI is getting nothing from the network....
09:46.48jeffikfinlander: thanks
09:46.55fenlanderjeffik: great
09:47.11jeffikand the connectio is really good,
09:47.21Blackvelmorning
09:47.27tassjas: no configuration could affect this? we had it working just a couple of minutes ago out a separate application (not asterisk) on a dialogic card
09:48.46Blackvelhow can I use xlite to do a direct ip call to *? #2 proxy is configured to use direct ip calling. I type in 192.168.1.2 and also try with 66@192.168.1.2 but I get this error message: Failed to authenticate user Blackvel <sip:192.168.1.3:5070>
09:49.14Blackvelis it possible to set insecure=very in sip.conf [general]? context=directip is there
09:49.18Jas_Williamstass: From what I see in your intense debug there is a cable problem, Has the debug changed with a straight through cable
09:49.58*** join/#asterisk morris (~morris@pcworkshop.plus.com)
09:50.07Jas_WilliamsBlackvel: make sure you have a relm defined in second proxy
09:50.09tassjas: i'll swap the cable and restart asterisk
09:50.12*** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de)
09:50.45*** join/#asterisk strafbomber (~strafi@p54A738F6.dip.t-dialin.net)
09:50.48strafbomberhello
09:52.04BlackvelJas_Williams: oh in xlite? and what should that be? any name? ip of asterisk?
09:52.27tassjas: i received 1 frame that looks the same as the last 2 i pasted
09:53.01Blackvelsame message but now with : failed ... Blackvel...sip:directip:5070
09:53.06Blackvelrealm is now directip
09:53.46BlackvelI can remember there was realm setting also for sip.conf. I don't have that set. is that required?
09:53.52strafbomberis it possible to use the Wildcard TE410P whit linux & capi is there a driver for that?
09:54.01Cheetahhmm
09:54.25Cheetahwhich is the best codec if you want good quality? (except ulaw, that one sucks ;)
09:54.36Blackvelalaw
09:54.41Blackvelit suxx?
09:54.54Blackvelshouldnt be
09:54.55Ahrimanesalaw works fine for me
09:55.08Blackveleverything expect ulaw/alaw is worse
09:55.15Cheetahwell, you can clearly hear that the sound quality has a HUGE loss compared to normal phones
09:55.16hellopass
09:55.37hellopsuch a short cord on the budgetone, it dropped, and now has static..
09:55.55Cheetahi tried speex, but that one doesnt seem to be too compatible with other things
09:55.56Jas_Williamstass: Do you have the cable connected to the correct port You may be plugging int span 4
09:56.22tassjas: it is definately numbered 1...
09:56.25Jas_Williamstass: leave intense debug on and plug straight cable into each in turn
09:56.48AhrimanesCheetah: with alaw i get similar or better sound than wired phones
09:57.17BlackvelCheetah: HUGE loss?
09:57.18bewestcould someone please take a look at this and possibly help me out? http://pastebin.ca/12141
09:57.20CheetahAhrimanes, are there settings I can optimize? because bandwidth is not a problem here (100Mbit connected softphones/phones)
09:57.31BlackvelCheetah: your softphone is the problem
09:57.32CheetahBlackvel, yeah, it sounds worse than GSM
09:57.37Cheetahxlite?
09:57.41Blackvelprolly
09:57.50BlackvelI always have problems
09:57.52Blackveluse hardware
09:58.09tassjas: i didn't get anything from moving the cable
09:58.11Cheetahwell, thats not an option here ;)
09:58.14Blackvelwhy
09:58.23Blackvelget sipura2000
09:58.25Cheetahbecause exactly thats why we move to softphones
09:58.33Cheetahthey need to be mac compatible
09:58.34Blackvelwell
09:58.36Blackvel:)
09:58.44Blackvelbluetooth headsets?
09:58.46Cheetahyep
09:58.49strafbomberis this possible? i want to use this card for an dialin access server
09:58.57Blackvelhmm too bad
09:59.00RoyKCheetah: I don't think there are any good softfons around for mac :(
09:59.10Blackvelxlite works on mac?
09:59.12Blackvelinteresting
09:59.15Cheetahit does
09:59.19Cheetahsjphone?
10:00.07*** part/#asterisk Cheetah (~Snak@62.217.48.111)
10:00.17*** join/#asterisk Cheetah (~Snak@62.217.48.111)
10:00.18Cheetaher
10:00.25Cheetahi should stop pressing red buttons
10:00.27strafbomberhas noone an answer?
10:00.36bewestno one can help me out?
10:00.46Jas_Williamstass: No idea what your problem is then unless you provider has the cables wrong an E1 cross over is 1-5 2-4
10:00.51Cheetahstrafbomber, weird nickname.. especially if you can speak german :D
10:01.15Ahrimaneseek german
10:01.28strafbomber:)
10:01.48strafbomberyes i can speak german, und das besser als englisch ;-)
10:02.22*** join/#asterisk bsunil (~bsunil@202.54.37.182)
10:02.27Blackvelund das typische klische wird wieder wach
10:02.28*** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk)
10:02.33robin_szmeep?
10:02.58robin_szso .. for added fun, I'm back to trying to build ztdummy on debian again ...
10:03.03strafbomberwhich one?
10:03.07robin_szsarge
10:03.16bsunilhello can any onle tell me how to trace a call through asterisk
10:03.47robin_szasterisk-1.0.7 on sarge to be precise
10:03.48Sato1bsunil, with its cdr
10:04.03Blackvelokay, what was now the correct way to do direct ip calling into asterisk?
10:04.06bewestis anyone familiar with this message from *:  Rejected connect attempt from 66.250.68.190
10:04.21Sato1bsunil, over here, please
10:04.34bsunilok
10:04.36bewestI'm registered with a voip provider through iax
10:04.43bewestbut that is the message I get
10:04.53bewestI've tried dozens of things, searching via the wiki and all over
10:04.55robin_szto save me added pain, does ztdummy actually work with a 2.4 SMP kernel?
10:04.56bewestand I am completely stuck
10:04.56Blackvelhm I try to register a xlite to an asterisk behind a firewall
10:05.15Sato1bsunil, check for log files in the directory /var/log/asterisk
10:05.18Blackveldoes xlite use automatically for registering port 5060 or will it use listening sip port 5070 (on my system its 5070)
10:05.40Sato1bsunil, or install the addon to store all call records in mysql or pgsql
10:05.49bsunilSato1 ok
10:05.49bewestmy iax.conf, sip.conf, and extensions.conf are at http://pastebin.ca/12141
10:06.04bewestand again, I am really really stuck, so if anyone could take a look and help me out
10:06.15bewestI would really appreciate it
10:07.27tassjas: RCV_RING<->XMIT_TIP & RCV_TIP<->XMIT_RING ?? this is according to the dialogic instructions.
10:08.36bewestwow no help
10:09.40Jas_Williamstass: sounds about right
10:10.12Cheetahi would help, if I could ;)
10:10.16Jas_Williamstass: should be 1-4 2-5
10:10.17Cheetahbut I'm a bloody newbie
10:10.32Jas_Williamstass: see http://www.gcom.com/home/support/t1crossover.html
10:10.44BlackvelCheetah: you could try to do sip show channels
10:10.52Blackvelto see if you are really using ulaw/alaw
10:11.02Sato1robin_sz, not sure if ztdummy works on SMP kernels, but you can give it a try, just follow the instructions in zaptel directory for compiling ztdummy support, it will take you 5 minutes
10:11.40tassjas: thanks for that... i'm fairly certain that's what we've got
10:11.52tassjas: especially since it works for the other card...
10:12.05tassjas: could it at all be a hardware fault do you think?
10:12.08Zeeekbewest ?
10:12.13*** join/#asterisk Martohtar (Martohtar@82.196.218.80)
10:12.47Zeeekisn't there a little something funny about those first three lines in you context?
10:13.30Jas_Williamstass: possibly.
10:14.03*** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk)
10:14.07ellvisre
10:14.50ellvisanyone have any experience with troubleshooting isdn bri and hisax and signaling?
10:15.16tassjas_williams: Thanks for your help... i might try set up the other spans instead and see if they work, and if i get the same problems i'll see if i can try a new card.
10:15.45tassjas_williams: i think i've tried just about everything I can otherwise!
10:16.13bewestZeeek, dunno, very new at this and been staring at this stuff for hours
10:16.54robin_szSato1: I seem to be having lots pf trouble on debain Sarge compiling zaptel ... it just throws out pages of errors
10:17.27Zeeekbewest, looking at your first three lines, I would have to recommend you do a couple days research and read the doc at http://asteriskdocs.org and the wiki sections about IAX2, extensions etc.
10:18.09Sato1robin_sz, see where it starts throwing errors, and paste the first 2 lines
10:19.10onkeltimmbewest: you have to put in your asterlink number in the pattern section of your first three lines.
10:19.33robin_szSato1: if I use buildpackage it compiles OK, but withut ztdummy .. if I just run make I get:
10:19.39robin_szSato1: In file included from zaptel.c:42:
10:19.39robin_sz/usr/include/linux/kernel.h:72: error: parse error before "size_t"
10:19.39robin_sz/usr/include/linux/kernel.h:74: error: parse error before "size_t"
10:20.43onkeltimmbewest: http://pastebin.ca/12143
10:21.53onkeltimmbut i do not know asterlink.
10:22.00Zeeekonkeltimm whatd'you make of the deny/permit at the end?
10:22.13onkeltimmthey should have config info in their members section for sure
10:22.43onkeltimmZeeek: din't read that far as the first three lines were so dead wrong...
10:22.54Sato1robin_sz, dont know in debian, but in rh, it solves lot of those problems by changing /usr/include/linux/* to the real path to the kernel source (usually in /usr/src/linux-2.4/include/linux)
10:23.07robin_szoh, ok ta
10:23.34onkeltimmoh yeah somebody forgot the netmasks there...
10:23.45*** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net)
10:23.49Mother_greetings
10:24.36ZeeekHi Mom!
10:24.44Mother_heya Zeeek
10:24.53Mother_how's things? coming to Madrid?
10:25.25Zeeekbewest, it took a while for me to realize that many of the files in /usr/src/asterisk/*.sample contain almost all options possible
10:25.40ZeeekMother_ I may be going to Barcelona
10:25.54Mother_well let me know then, I live in Barcelona
10:25.58onkeltimmbewest: go reading. look closely at the examples on voip-info.org
10:26.05Mother_we can have a paella and talk SIP
10:26.07Mother_:)
10:26.16bewestare you referring to the extensions.conf
10:26.19bewestor all of them
10:26.19Zeeekand try to separate the obviously wrong ones from the valid :)
10:26.56Mother_bewest: also look here http://www.loligo.com/asterisk/current/
10:27.08bewestheh I've been looking at all of those
10:27.09Mother_some comprehensive examples for quite a few things * can do
10:27.29bewestI don't go off shooting for help very quickly
10:27.31Zeeekbewest I think your configs are pretty far off; you need to study  harder if you haven't been able to understand them
10:27.33Mother_I found them quite useful when I started - print out the long ones however
10:27.50Zeeekpossible a step by step approach might help
10:27.52ZeeekInstalling linux, asterisk X100P, TDM400, SIP config, Vmail, parking and many other sexy topics
10:27.52Zeeekhttp://automated.it/guidetoasterisk.htm
10:27.58ZeeekThe above is very good
10:28.03Mother_yep, good one too that is
10:28.07bewestok haven't seen that one yet
10:28.11Zeeekaha!
10:28.25Mother_since it has wifi - free calls :)
10:28.49onkeltimmZeeek: uargh decaf...
10:28.53ZeeekCheetah why not IAX if you wanna dream?
10:29.02Mother_well that too
10:29.11CheetahZeeek,
10:29.13Cheetah?
10:29.28Zeeekno that was for Mother_ sorry
10:29.33Cheetahheh
10:29.46Mother_I really hate the problems with hangup detection, I have this callback system working with the slight problem that the Zap channels just hang there forever
10:30.51Mother_Zeeek: well, maybe porting some IAX client would be easier than writing a SIP one from scratch, who knows :)
10:30.56Zeeekannoying and potentially costly
10:31.04Zeeekthe hanging ZAP I mean
10:31.06Mother_indeed, very annoying
10:31.07Mother_yep
10:31.28Mother_does * monitor DTMF on Zap channels while on a call?
10:31.47Mother_maybe I could get it to recognise a # or * to hang up the calls
10:37.55Mother_maybe with the Read command something could be cooked up...grrr
10:39.23fenlanderHow about the h or H option to Dial?
10:39.47onkeltimmfenlander: bad if * does not detect the hengup correctly.
10:40.09onkeltimmMother_: but what about a simple exten => #,1,Hangup
10:40.53*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
10:42.31ellvisCheetah: not really a SIP phone, but who knows, maybe one day... http://www.taptarget.com/products/iphone.html
10:46.12Mother_onkeltimm: I could try that, but I understand that between s,1,Dial..... and s,2,Hangup asterisk is waiting
10:46.31Mother_if I dial # during a call, will it get thrown into the context again?
10:46.35*** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com)
10:47.45Mother_i.e. in [context] I would have s,1,Dial..... s,2,Hangup then #,1,Hangup <- will it ever get here once a call in the context is taking place¿
10:47.45onkeltimmthis is like a menu, isn't it? you have the s extensions. as soon as you dial #, asterisk should execute the hangup there? or am i braindead
10:48.17Mother_yes, but you are dialing # on an active channel - do you mean that if I dialed another number during a call, asterisk will go through the whole smash? it will put the call into a context, start the call, etc???
10:48.18onkeltimmi do something similar to let callers dial a known extension befor sending them to our operator
10:48.38fenlanderWon't the H option to Dial just let you hangup with *? What don't I understand?
10:50.02Mother_fenlander: I see what you mean now, sorry
10:50.19onkeltimmfenlander: misunderstood you, sorry. that may be an option. but you need to use h as this is callback.
10:50.21Mother_adding H/h to the dial command, I misunderstood
10:50.33Mother_let me try that...
10:51.10fenlandernp - it might not work, but worth a try
10:53.42*** join/#asterisk tessier (~treed@203.210.216.1)
10:54.01Mother_no joy :(
10:54.27onkeltimmguys: http://voip-info.org/tiki-index.php?page=Asterisk+cmd+SIPCallPickup can I do something similar with IAX???
10:56.41Mother_what's wrong with pickupgroups etc.?
11:01.03darwin35http://pastebin.ca/12068 if some with knowledge can look at this I have a issue with thr 3010150 and 4715227 not going to the vm  mailbox and just going to falloff .
11:02.11darwin35the 866 nmbr does like it should and goes to the vm
11:03.02onkeltimmMother_: I cannot get it to work with Iax clients.
11:03.08Mother_hmmkay
11:03.30Mother_I think it works OK on one of my boxes with an IAX2 trunk to another *
11:03.32RoyKhttp://news.bbc.co.uk/1/hi/world/americas/4560847.stm <--US mulls ban on women in combat
11:03.37RoyKrotfl
11:03.53Mother_i.e. the calls that come bridged from PSTN on the remote * can be picked up OK
11:04.56onkeltimmactually i am dumbstruck why this does not work. perhaps my config's just wrong... but http://voip-info.org/tiki-index.php?page=Asterisk%20callgroups%20and%20pickupgroups tells me it only works with mgcp, sip and sccp
11:05.23*** join/#asterisk mattbridges (~mattb@mattbridges.plus.com)
11:05.54mattbridgeshello all, can anyone help me to get UKCLID working on my X101P and BT Line?  I've patched it but it's still not playing ball.
11:06.01onkeltimmand zap, of course
11:06.42Mother_ah OK, well I use SIP phones and Zap channels, I've never tried with a native IAX client
11:07.36onkeltimmMother_: seems like nobody else here did. have this problem for the whole week, it's perhaps costing me my job, and i am glad you at least answered.
11:08.19onkeltimmMother_: the chan_iax.c code has nothing of callgroup/pickupgroup so it might not even be implemented
11:08.55onkeltimms/iax/iax2/
11:08.55Mother_that kinda sucks
11:09.09darwin35I cant even get feed back on one small issue dont feel bad
11:10.05onkeltimmand guys, i am ready to DO THIS if anyone of you would just help me get started. (explain the callgroup/pickupgroups internal workings to me)
11:11.23darwin35from reading the wiki its all there
11:11.38onkeltimmdarwin: url?
11:11.55darwin35it talks about using *8 to puckup calls from outher phones
11:12.35Mother_I've never really looked at the innards
11:12.40onkeltimmyeah, as i wrote before, it works only on SIP, SCCP, MGCP, and Zapata
11:12.59*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
11:13.05Mother_darwin35: yes, but if you look here http://www.voip-info.org/wiki-Asterisk+config+iax.conf in the table, it doesn't say pickupgroup can be used anywhere
11:13.22*** join/#asterisk m0f0x (m0f0x@m0f0x.user)
11:13.28darwin35nope not supported IAX@ from what I know
11:13.47darwin35if bkw comes back ask him
11:13.52onkeltimm... why always me... ;)
11:13.58darwin35IAX2 sorry
11:15.26Mother_and you have tried adding pickupgroup and callgroup to iax.conf? just in case...
11:15.35onkeltimmMother_: sure
11:16.23onkeltimmbut it is not in the code. chan_sip.c has callgroup and pickupgroup in the user/peer structs. chan_iax2.c hasn't
11:16.24Mother_damn...
11:16.48Mother_hmkay - maybe I'll have to start learning C for real :)
11:17.33Makenshidoes anyone know of a sip client for windows mobile that supports video?
11:18.09onkeltimm~seen bkw
11:18.10jbotbkw <~bkw@u201.udal.afb.lu.se> was last seen on IRC in channel #debian, 88d 11h 17m 13s ago, saying: 'uhm, the latter one or both of them?'.
11:18.46bewestyay much progress
11:21.11*** join/#asterisk mjlinda (~mjlinda@221.204.104.96)
11:22.28Zeeekit works (back from lunch)
11:22.32Zeeek??
11:24.57*** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net)
11:25.17*** join/#asterisk zotz (~zotz@208.196.247.140)
11:26.21*** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net)
11:26.37tzafrirHow do I play a short "different" sound (to indicate an invalid operation)?
11:26.52tzafrir(in the dialplan)
11:29.17Zeeekyou mean like windows?
11:29.36*** join/#asterisk langals (~icechat5@196.7.14.183)
11:29.49*** join/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl)
11:29.56tzafrirZeeek, I mean like \a :-)
11:31.16Zeeekwhat's wrong with <BELL> ?
11:39.43tzafrirZeeek, how can I use <BELL> from the dialplan?
11:42.14Zeeekhok up a TTY as console?
11:42.15Zeeekthat has the BELL key
11:42.15Zeeekbut I assumed you could play a short sound file?
11:42.15*** join/#asterisk ctooley (~ctooley@pc51.utati.net)
11:42.15onkeltimmtzafrir: what do you mean by invalid operation? in a menu, you could just define a fallthrough exten
11:44.10langalshi there...I am using IAX2 softphones, connecting to Meetme....when there are 3 or more users connecting from behind the same lan (.i.e going through the same NAT), then there are 3 second breaks every 30 seconds or so.....wondering if this problem might be because users all going through the same NAT?...
11:44.20langals...bandwidth should not be a problem....
11:44.54langals...when I have 3 users, but all from behind different LANS, then don't have this problem.....any ideas?
11:45.28tzafrironkeltimm, I want to put it in the i extension of an IVR. However I asked *what* to put and not *where*.
11:46.00ZeeekWhenever I call a friend from ZAP to Firefly (asterisk 1.0.6) the IAX2 the fact that the side answeres is not detected. Asterisk says "no one picked up afetr 25000ms" and goes to the "no one is available..." message. The connection is normal for both ends for the first n seconds, the n being the time in Dial()
11:46.59onkeltimmtzafrir: PlayBack(invalid) ?
11:48.59tzafrirThis is what I currently use. But it may sound strange in an locally-recorded IVR. Oh, well
11:49.26Zeeektzafrir can't you record the file you wish?
11:49.40Zeeek(with <BELL> in it)
11:50.37*** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl)
11:57.16*** join/#asterisk doughecka_ (~Tad@doughecka.user)
11:59.07ctooleylangals, What kind of NAT box do you have?  If it's slightly underpowered it might have problems keeping up with 3 constant streams of data like IAX
11:59.42ctooleyUsing NAT by itself shouldn't be an issue per se, but using NAT can lead to problems if the NAT box can't keep up.
12:04.47Zeeeklangals if you mean all three are behind NAT on the same LAN, it would seem to be a bandidth issue on their side, no?
12:06.50Blackvelhmm question is still open for me. how the heck can I do direct ip calling into asterisk with xlite? :) I get  Failed to authenticate user msg
12:07.24langalsctooley - not sure what kind of NAT it is - will try and find out
12:07.38ZeeekBlackvel - you mean yermutha@123.234.1.23 ?
12:08.12Blackveljepp
12:08.15langalsZeeek - no - the asterisk box is actually also on the same network (altough outside the lan) - so there is plenty of bandwidth
12:08.32Zeeeklangals FWIW, my SPeedtouch router/modem froze when I tried to run asterisk on it. Couldn't keep up apparently
12:08.36robin_szquick question asterisk 1.0.7 on a 2.6 kernel .. do I need ztdummy to run a meetme room or do I remember it not being needed on 2.6 kernels?
12:08.41*** join/#asterisk Dovid (~hirisk@pool-138-89-178-170.mad.east.verizon.net)
12:08.42BlackvelZeeek: my sip.conf [general] contains context=directip
12:08.54Blackvelbut there seems to be still a security issue
12:08.57Zeeekand in that context?
12:09.09Blackveljust failed msg, nothing more
12:09.18Zeeekwhat is in that context?
12:09.26Blackvela dial
12:09.35Zeeekand what is in that dial ? :)
12:09.48Zeeeks,1,Dial... ?
12:09.55Blackvelhaha
12:10.13Blackvela user which asks me what the hell is zeeek asking you :)
12:10.14tzafrirrobin_sz, ztdummy in 2.6 kernels is generally good enough (or at least better than the one in 2.4). But you still need one zaptel timing source as before
12:10.27BlackvelZeeek: s,1,Dial + 66,1,Dial
12:10.34robin_sztzafrir: OH. pooh.
12:10.46Blackvelso I expect that 66@192.168.1.2 and 192.168.1.2 is working
12:10.55tzafrirrobin_sz, those are just modules, they don't require a reboot
12:11.12Blackvelbut it does not execute the dail
12:11.18robin_sztzafrir: but building ztdummy in debain sarge seems next to impossible
12:11.26Blackvelhow can I send for direct ip something like insecure=very
12:11.37Blackvelcan I do that in general section?
12:12.03Zeeekyou have any disallow mask in general?
12:12.33ZeeekI'm pretty sure I can dial anything in my guest context from outside without id
12:12.53Zeeekbut wait, maybe I've never tried
12:13.22Blackvelis the guest context that context=directip thing in sip.conf [general]?
12:13.35Blackvelsure I have disallow but also allow
12:13.39Blackvelg711 should work
12:13.50ZeeekI call it guest - you can call it [oral-sex] if you like :)
12:14.12ZeeekI'm referring to a netmask - sorry meant deny=
12:15.17*** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
12:15.21robin_szhas anyone got any more clues about building zaptel/ztdummy on debian? someone must have doen this before .. I think even I did once ..!! .. sigh
12:15.40Blackvelupps
12:15.46Blackvelnow it works with 66@192.168.1.2
12:15.47robin_szbut that was months ago :( brain fade.
12:15.48Blackveldunno why
12:16.10*** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu)
12:17.16guyeedoes NE knows why my SIP client indicates ringing immediately without waiting for a result from the h323 gw?
12:17.45cjkrobin_sz: i did it
12:17.48cjkits working
12:19.09*** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au)
12:19.34ilium007hi all
12:19.37tzafrirIs there a reason why in the CLI output the 'secret' of a SIP peer not shown and the 'secret' of an IAX peer is shown?
12:19.53kapejodguyee: you dont have the "r" dial option, do you?
12:20.16ilium007i have jst managed to get my * box to route  accall from 1 sip softphone to another. problem is i am getting an error stating:
12:20.27tzafrir(unless you use '<Set>' as your secret, that is
12:20.37ilium007chan_sip.c  process_sdp: No compatible codecs!
12:20.38jakepdev[work]tzafrir - is your secret in IAX.conf or are you specifying it in your dialplan?
12:20.42ilium007can someone help ?
12:21.05tzafrirI refer to the secret entries for peers in sip.conf and iax.conf
12:21.25*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
12:21.32guyeekapejod: Grrrr... I had. thx. problem solved, I owe you a beer :)
12:21.33jakepdev[work]ilium007 - sounds like you need two codecs that are compatible for your two endpoints.  what are your two endpoints?
12:21.50ilium007i have x-lite on my mac osx and firefly on pc
12:21.53ilium007windozwe
12:22.05ilium007both have gsm i assume
12:22.18ellvisre
12:22.24jakepdev[work]do you have allow=gsm in sip.conf?
12:22.39ilium007yes i do
12:22.49ilium007dissallow=all
12:22.52ilium007allow=gsm
12:23.19jakepdev[work]try allow=all (take out the disallow) and see what it comes up with
12:23.34*** join/#asterisk dgillson (~dgillson@rrcs-24-97-7-126.nys.biz.rr.com)
12:24.08ilium007k
12:24.33kapejodguyee: ok. when? where? ;)
12:24.44RoyKjakepdev[work]: no, just do disallow=all, allow=alaw
12:24.48RoyKstart with that
12:24.53ilium007dammm
12:24.54RoyK_always_ start with that
12:25.00ilium007its all scratchy and foul
12:25.04guyeekapejod: anytime... er... what about Budapest, Hungary? :)
12:25.25kapejodguyee: will you cover travel and hotel expenses, too? ;-)
12:25.27jakepdev[work]ilium007 - try Roy's suggestion
12:26.20ilium007same all scratchy and broken
12:26.22ilium007:(
12:26.33RoyKwtf???
12:26.47kapejodRoyK: he already agreed on that.
12:26.51RoyKeven without a good SIP login, this host is able to make calls!!
12:26.51guyeekapejod: not really... I mean... I should look for a few more questions, and then maybe :)
12:27.22jakepdev[work]ilium007 - try ulaw
12:27.29*** join/#asterisk nvrs (RUR@London-HSE-ppp3545191.sympatico.ca)
12:27.34ilium007take out allow=all
12:27.40jakepdev[work]yes
12:27.55jakepdev[work]use disallow=all allow=alaw
12:28.15ilium007ulaw or alaw ?
12:28.16robin_szoh foo. sigh. damn.
12:28.17jakepdev[work]then try disallow=all allow=ulaw
12:28.24ilium007ok
12:28.45*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
12:29.09robin_szso anyother really good suggestions as to how to install a asterisk with a working meetme?
12:29.29robin_szits looking too damn hard on debian (again) sigh
12:29.31jakepdev[work]robin - rapid?
12:29.38jakepdev[work]oh wait - that's debian
12:29.38robin_szrapid?
12:29.45tzafrir~rapid
12:29.49jbot[xorcom rapid] at http://www.xorcom.com/rapid.html . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir
12:30.13robin_sztzafrir: AND IT HAS A WORKING ZTDUMMY?
12:30.14RoyKwtf________
12:30.21robin_szoops, didnt mean to shout :)
12:30.24RoyKI have no peers defined in sip.conf
12:30.26RoyKnone
12:30.28RoyKwhatsoever
12:30.29vaewyntzafrir: bad link dude :}
12:30.44tzafrirrobin_sz, sure. If the installer doesn't detect a TDM card installed, it will modprobe ztdummy
12:30.49kapejodRoyK: but you have a context in the general section.
12:31.26tzafrir<PROTECTED>
12:31.44tzafrirHow do I update jbot information?
12:31.45vaewyn:}
12:31.45onkeltimmvaewyn, tzafrir : s/.html//
12:31.55jakepdev[work]jbot, rapid is  [xorcom rapid] at http://www.xorcom.com/rapid . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir
12:31.56jbot...but rapid is already something else...
12:31.57*** join/#asterisk jetdotnet (jetdotnet@adsl-64-219-216-41.dsl.hstntx.swbell.net)
12:32.06jakepdev[work]jbot - but i don't care
12:32.15jakepdev[work]jbot change it anyway
12:32.26onkeltimmjobot, forget rapid
12:32.31onkeltimmfsck
12:32.34onkeltimmjbot, forget rapid
12:32.34jbotonkeltimm: i forgot rapid
12:32.41jakepdev[work]jbot, rapid is  [xorcom rapid] at http://www.xorcom.com/rapid . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir
12:32.42jbotokay, jakepdev[work]
12:32.50jakepdev[work]tnx
12:33.03onkeltimmnp
12:33.13tzafrirthnkas
12:33.36robin_sztzafrir: so .. the box I have has 2.4.27-2-686-smp already, would the zaptel-modules-2.4.27-2-686-smp_1.0.4-1_i386.deb solve my ztdummy problem you think???
12:33.52tzafrirrobin_sz, it should work, for that specific asterisk/zaptel version
12:34.05*** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net)
12:34.10*** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com)
12:34.18tzafrirMy "unstale" packages are at http://tzafrir.org.il/rapid/
12:34.44vaewynunstale bwahahaha... more correct than unstable these days
12:35.24vaewyncan't wait for the new debian... cause then testing and unstable will get 'unbunched' again
12:36.18tzafrirvaewyn, I currently use Sarge. But there are simply too much stuff waiting for Etch to start
12:37.51*** join/#asterisk GrahamC (hidden-use@213-131-100-29.onyx.net)
12:37.52vaewynI can never remember which names are which...  but I use all the way to testing in 90% of the machines
12:37.55ilium007can anyone suggest a good mac osx SIP client >
12:38.07*** join/#asterisk durex (~ironman@weber.anpa.org.br)
12:40.34GrahamChi everyone. Think I need to load ztdummy on a suse 9.2 system to get meetme working. I get a compile error o rule to make target `modules'.  Stop. Any ideas. I am newbie!
12:41.12GrahamCPS - Am using chan_capi on fritz card
12:42.06darwin35xten works on the mac
12:42.23robin_sztzafrir: thnaks, I installed the modules and was able to insmod zaptel and ztdummy .. now I get to see if they will actually work with my asterisk, its the current debian asterisk, 1.0.7 .. so it might be OK
12:44.11darwin35http://pastebin.ca/12068 if some with knowledge can look at this I have a issue with thr 3010150 and 4715227 not going to the vm  mailbox and just going to falloff .
12:45.17vaewyndarwin35: hehehe... 471 is our local exchange in 269 areacode :}
12:45.38jakepdevdarwin - you say "Goto(autoattend,s,1)", but you don't show s,1
12:45.43darwin35this is in the 479 area code
12:46.25vaewynjakepdev: yeah... is in the includes... but not sure it is correct
12:46.50jakepdevsorry - wrong person
12:47.03jakepdevvaewyn - show the include also
12:47.08sylehttp://www.sipgate.co.uk/catalog/product_info.php?products_id=10
12:47.20vaewynjakepdev: no... right person... I was just replying before he did
12:47.24vaewynand is he showing them
12:47.25darwin35it does
12:47.26sylecan configure 2 sip accounts on same router :)
12:47.40darwin35it shows with a incude day
12:47.58darwin35it reads the tiiem first
12:48.07darwin35to see if they are open or closed
12:48.17vaewyn8667871709 does or does not work also?
12:48.38darwin35the 866 works it goes to the vm
12:48.52darwin35the other 2 nmbrs drop off
12:49.39darwin35thats what confuses me
12:49.44jakepdevdarwin - maybe i'm oversimpliying things here, but it would seem to me that you'd have a main context for the incoming and GotoIf based off of time to your seperate contexts
12:49.59vaewynjust out of curiosity... what does [selection] have in it?
12:50.26vaewynanything with _XXXXXXX?
12:50.27darwin35there is a selection part for at night h
12:50.28vaewynor such?
12:51.33darwin35we have 10 direct nmbrs
12:51.42darwin351051 -0160
12:51.51darwin35they work fine
12:52.20vaewynI was just thinking that since it is included first if there was a matching exten in there it would snatch it before those night/day ones would
12:52.40darwin35but they ring the lines
12:52.50darwin35its just when they hit the vm they fall off
12:53.13darwin35am i going to have to manual set each one
12:53.30*** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it)
12:53.34darwin35to fix this
12:53.48vaewynThat's just odd... it should handle them all the same
12:53.59darwin35its not
12:54.01vaewynvoicemail doesn't care where stuff is coming from
12:54.20vaewynunless you do like   s${EXTEN} or such
12:54.24vaewynbut that ain't in there
12:55.09darwin35whe the 0150 and 5227 hit  the s,3, line they fall off and you get the fall threw message that the call can not be completed at this time
12:55.33mAsH`sorry, anyone used h323 with * ?
12:55.50robin_sztzafrir: are you the sarge asterisk maintainer?
12:56.09*** part/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net)
12:56.09vaewyndarwin35: try putting a  s,104,Goto(4)  in there
12:56.10darwin35if I halt to I can set a dial line for each nmr
12:56.14*** join/#asterisk HA (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net)
12:56.34robin_szhmm
12:56.34vaewynjust out of curiosity
12:56.36robin_szres_musiconhold.c:278 monmp3thread: unable to spawn mp3player
12:56.58robin_szI guess I need an mp3 library, any ideas which one
12:57.16robin_szsorry about he random questions .. stressed here :)
12:57.36vaewyntype   'make mpg123'  in your asterisk source dir.... it will do everything for you
12:58.19vaewyngets the correct version and everything
12:59.29*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
13:00.32robin_szvaewyn: well, it seems to have done someting ...
13:02.13darwin35nope
13:02.29darwin35now it loops and dies
13:02.50vaewyndid it hit the Goto?
13:02.53darwin35am jus going to hard set this one
13:02.59darwin35yes
13:03.29vaewynthat's just freeky
13:03.33darwin35yeah
13:03.39darwin35its pissing me off
13:03.48MikeJ[zzzzzzz]darwin35, did you get your compiling prob fixed?
13:03.50robin_szwee
13:04.58vaewyndarwin35: also... might want to change your comments... you have both a day and night saying 'OPEN w00t!'
13:05.18*** part/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl)
13:05.22vaewynor change the included context :P
13:05.30*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
13:05.52*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
13:07.13tzafrirrobin_sz, no, but I'm on the pkg-voip mailing list and keep in touch with them
13:07.55tzafrirThe "maintainer" of the asterisk packages in Debian is the "pkg-voip" team
13:08.44tzafrirrobin_sz, mpg123 is available from non-free
13:08.54tzafrirhttp://packages.debian.org/mpg123
13:09.15vaewyntzafrir: some people have been having trouble with the newer ones...
13:09.27vaewynis why the make mpg123 thing showed up :P
13:09.33tzafrirThis is the latest 0.59*r(
13:09.39tzafrirs/(/*/
13:10.03tzafrirIt should work fine, from what I understand.
13:10.06Mother_HAH!
13:10.09Mother_it verks!!!
13:10.15vaewynSu Madre!
13:10.17Mother_lol
13:11.06tzafrirrobin_sz, I have also packaged separately the moh files removed from the debian package. They are in my "unstable" directory in the package asterisk-sounds-moh
13:11.24tzafrirI do hope to package alternative sounds soon to avoid this silly license issue
13:11.48*** join/#asterisk PandemiK (~PandemiK@62.2.255.42)
13:11.52PandemiKhi every1
13:15.16Mother_tzafrir: there was a CD in Wired a couple months ago with music one could freely distribute, package, sample, etc.
13:15.36Mother_some of it was OK for moh if you don't want to pay royalties
13:15.36*** join/#asterisk km- (~pgrace@brdgw1.rttx.com)
13:15.52km-howdy!
13:15.55Mother_in any case, it would be überlame if the RIAA started suing people for their moh music :D
13:16.29Mother_hiya
13:16.37km-hey, I hear there are issues with the 7.x series of 7960 firmwares, anyone having the same issues?
13:17.18darwin35thats for testing
13:17.45darwin35so I can dial in at any time
13:18.22sudhir492can someone explain callgroup and pickup group to me?
13:18.27HAis there a dev call today?  if so, what time?
13:18.28Mother_km-: I have 7.3 and no issues so far
13:18.44Mother_I think 7.4 is out but I've not tried it
13:18.47km-mother: cool.  My supplier threw 6.3 onto the phone, I'll give that a go for a while
13:18.53km-if it ain't broke don't fix it, I guess!
13:18.54darwin35around 1 cst 2 est
13:18.57Mother_yep, indeed
13:19.13Mother_gotta go eat, cya all
13:19.15HAty darwin.  i might actually be able to listen in today.
13:19.18km-later Mother
13:19.26darwin35cool
13:21.06*** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:21.27km-I hate it when I misplace stuff!
13:21.40km-does anyone have the program that allows you to convert graphics to logos for the 7960?
13:21.46HAUnless of course we manage to get the financing approved for this new system and then I'll be tied up trying to buy all this crap.  I really do enjoy spending other peoples money but having to spend $21,000 in a single day gets pretty tiring.
13:22.57ariel_hello everyone
13:23.09*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
13:23.09*** mode/#asterisk [+o bkw_] by ChanServ
13:23.27iCEBrkrkm-: You need to start maintaining your own personal wiki
13:23.48iCEBrkrkm-: That way you just dump all your how-to's and other configs/setup files there
13:23.50km-icebrkr: that's not a half bad idea
13:23.55km-bkw: moose penis?
13:24.02*** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com)
13:24.05km-bkw: dude, do you still have the 7960 configuration goodies like the logo creation program?
13:24.15PandemiKHA: what are you calling a dev call ?
13:24.35iCEBrkrkm-: I started one last night  actually, cuz I'm tired of hacking out asterisk confg file and scripts only to forget them 3wks later.
13:24.43km-icebrkr: hahaha, awesome
13:24.48km-I've started hacking on asterisk-java
13:24.57km-it gives me a semi
13:25.01Jas_Williamskm-: it is not required in the latest release it reads standard graphics files
13:25.08iCEBrkrdude, 2.5hrs trying to get my iax.conf stuff working with Voicepulse.. It was working one day and then just stopped.. I dunno WTF I did..
13:25.13km-jas_williams: i.e., version 7?
13:25.16iCEBrkrhahha
13:25.23*** join/#asterisk astoria (~haydenth@66.235.201.217)
13:25.29km-icebrkr: huh, that sucks pretty hard
13:25.33iCEBrkrand the voip-info wiki's iax.conf foe Voicepulse is WRONG
13:25.40iCEBrkrs/foe/for
13:25.48km-I actually can't stand voicepulse
13:25.56iCEBrkrIt works. *Shrug*
13:26.00astoriaHey, i got a stupid problem. two polycom phones, both can make outgoing IAX calls okay, but sip calls between the two bridge okay, but no sound ... sorry to be vague..
13:26.03km-good luck getting support
13:26.03Jas_Williamskm-: POS3-07-4-00
13:26.04iCEBrkrTho, I think my config broke cuz they changed something.
13:26.12km-jas_williams: ahh, thanks for the tip.
13:26.18iCEBrkrkm-: I never call support :P
13:26.19*** join/#asterisk zoa (~zoa@pirus.securax.be)
13:26.26km-jas_williams: I hear there are misc oddities for the 7 series though
13:26.35km-at least, that's what the guy at gts was tellin me
13:26.51iCEBrkrkm-: You ever tinker with eagi?
13:27.01Jas_Williamskm-: works fine for me and has done for a while now. Upgrading can be fun th
13:27.02iCEBrkrkm-: I got sphinx to compile and actually work :)
13:28.20iCEBrkrI'm trying to clean up my asterisk install.. Got voicepulse working again, inbound and outbound FWD working again too.
13:29.13HAPandemiK: IIRC there is a * developers conference call on Thursdays.  And according to darwin, my memory is working fine at the moment.
13:29.25iCEBrkrhaha
13:29.36iCEBrkrDarwin is on vacation.
13:29.37PandemiKok
13:29.46PandemiKthx
13:29.55HAEven though I did stay up all night to go see a movie at 3:30 in the morning.
13:30.11*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
13:30.18km-icebrkr: what's sphinx?
13:30.25km-word, episode 3
13:30.29iCEBrkrkm-: Speech recognition
13:30.33*** join/#asterisk toot (~chris@217.30.126.10)
13:30.33vaewyn*spoiler* Anakin is Vader!!! */spoiler*
13:30.35vaewyn:}
13:30.38km-icebrkr: nice, did it actually recognize speech?
13:30.43km-vaewyn: NOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO!!!!
13:30.46iCEBrkrkm-: I got it working on my workstation.
13:30.50toothey just had 2 call/hangups. then the next call i got the following  Detected alarm on channel 1: Red Alarm
13:30.51km-icebrkr: cool
13:31.03iCEBrkrkm-: I just dunno how I'm gonna get it to integrate with asterisk.
13:31.09mishehu*spoiler* Chewbacca is what a wookie! */spoiler*
13:31.12vaewynand Padme is made out of Hot Grits...
13:31.14tootnow nuffing seems to happen when a call comes in - has it hung the card in some way?
13:31.16iCEBrkrthe eagi-sphinx-test is just a framework
13:31.19PandemiKanyone using budgetone here ? (although my problem is perhaps not from the GS)
13:31.20Pete_Largovaewyn, I think that was spoiled in episode 6 - Return of the Jedi...
13:31.30mishehuvaewyn: those are some hot grits
13:31.35mishehuI'd eat those hot grits.
13:31.37iCEBrkrPandemiK: Yea, I got a BT100
13:31.37HAnow we just need 7, 8, and 9 so we can see if the republic is restored or if it remains an empire.
13:31.37vaewynmishehu: amen!
13:31.54PandemiKiCEBrkr: I have some problem, when I'm called, there's no sound
13:32.00vaewynHA: Anarchy rules...   ;P
13:32.11iCEBrkrPandemiK: NAT'd?
13:32.13mishehubastard friend of mine who studies at the hebrew u had no idea that natali portman was studying there...
13:32.18mishehuheh.
13:32.27PandemiKiCEBrkr: no, my asterisk is on the same subnet
13:32.27sudhir492<mAsH`: I am using H323 with *
13:32.47PandemiKiCEBrkr: it was working great and ... nothing
13:32.48iCEBrkrPandemiK: So Asterisk and your BT have public IPs?
13:33.09PandemiKiCEBrkr: no, they're on the same private address subnet
13:33.10toothmm show channels now shows 0 channels, but ztcfg -v shows 1 channel configured?
13:33.17PandemiKiCEBrkr: a LAN ;-)
13:33.22*** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it)
13:33.31iCEBrkrOh, you're making extension->extension local calls....
13:33.44PandemiKyeah
13:33.49iCEBrkrWell, hell..
13:33.52iCEBrkrThat should be cake.
13:34.01PandemiKI have my BT102 and a softphone (X-lite)
13:34.16iCEBrkrWell, make sure X-Lite is actually 'hearing' your mic. :P
13:34.37PandemiKWhen I call from BT102, it works, and when I call from X-lite, no sound
13:34.54iCEBrkrPretty interesting.
13:34.57PandemiKyeah
13:34.58iCEBrkrI don't think it's your BT tho.
13:35.10PandemiKI forced the codec to ulaw
13:35.16PandemiKand the extensions are the same
13:35.19PandemiKpretty simple
13:35.23PandemiKDial, Hangup
13:35.24*** join/#asterisk doolph (doolph@200.46.148.35)
13:35.36PandemiKhello doolph
13:35.45doolphwhat's new
13:35.59iCEBrkrPandemiK: you try 'sip debug' at the console?
13:36.09PandemiKnot actually
13:36.11iCEBrkrPandemiK: also set verbose 9
13:36.18PandemiKI only run asterisk with verbose 3
13:36.29iCEBrkrI dunno what the upper limit is. :P
13:36.34iCEBrkrBut 9 works for me
13:38.58PandemiKhmm
13:40.25PandemiKI can't get anything from sip debug
13:40.41iCEBrkrIt's showing debug info or no?
13:40.45PandemiKyeah
13:40.50kdayncan somebody give me a sample CDR file, I want to test some things, but I dont have one to do that
13:40.54PandemiKbut reading it don't give me any clue
13:41.08iCEBrkrkdayn: It's just a CSV file.
13:41.15kdayni know
13:41.26iCEBrkrmake one up
13:41.30kdaynbut i would like to see real copy from asterisk
13:42.15kdaynmatbe somebody can send it via dcc? :)
13:42.19PandemiKeeeh, what does that mean : == Spawn extension (sip, 1002, 1) exited non-zero on 'SIP/1001-91a3'
13:42.54iCEBrkrkdayn:
13:42.54iCEBrkr"","kphone","1234","default","""iCE Breaker"" <kphone>","SIP/kphone-1bd3","","VoiceMail","b1234","2004-03-21 22:35:14","2004-03-21 22:35:14","2004-03-21 22:35:33",19,19,"ANSWERED","DOCUMENTATION"
13:42.59iCEBrkrhave fun.
13:42.59kdaynone question, asterrski uses udp, what linux based clients would you suggest?
13:43.05kdayniCEBrkr: thx
13:43.24vaewyniaxcomm for iax connections...
13:43.40vaewynnot sure on sip cause I don't use sip softphones :P
13:43.45PandemiKlinphone, gnophone for sip
13:43.47onkeltimmasterrski, that the russion l10n of * ?
13:43.59onkeltimmsncr
13:44.01PandemiKI've heard that x-lite is available for linux too
13:44.42vaewynIf you have a choice use IAX and iaxcomm though... it will work a ton more places
13:47.27*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
13:48.35zoadont forget idefisk !!!
13:48.45zoaah idefisk is not yet for linux
13:48.56zoaalthough it works with wine
13:50.06PandemiKI have to go, but I'll be back in one hour with my problems ;-)
13:53.43iCEBrkrblah blah blah
14:07.15*** join/#asterisk ManxPwr (~eric@dsl-209-205-172-111.i-55.com)
14:07.34ManxPwrDoes anyone know what company makes the tbird brand of test equipment?
14:08.30*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
14:09.37Pete_LargoActerna
14:09.43Pete_Largouse to be called TTC
14:10.02Pete_Largoand it's T-BERD (Bit Error Rate Detector?)
14:10.12*** join/#asterisk Umaro (~umaro@209.140.74.64)
14:10.42Pete_Largoaren't you supposed to be in europe?
14:10.51Umarohey guys, anyone else having problems with there inbound DIDs from nufone atm?
14:12.24ManxPwrPete_Largo, I leave this evening
14:13.31robin_szhmm m.. weird ... the musiconhold plays for about 0.5seconds and then craps out
14:14.39ManxPwrrobin_sz, That would be expected if your SIP client has VAD enabled.
14:14.44*** join/#asterisk file[class] (~jcolp@66.199.241.90)
14:14.48file[class]meep meep
14:14.56ManxPwrPete_Largo, Do you know if the same company makes test equipment for analog lines?
14:15.02robin_szManxPwr: its an IAX client
14:15.21robin_szfile[class]: meep!
14:15.33file[class]hiiiiiii
14:15.56robin_szand .. if I use a web based client, can I use the same user/pass on all the clients ??
14:16.18Pete_LargoManxPwr - you mean like for your home phone?
14:16.31*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:16.31*** mode/#asterisk [+o anthm] by ChanServ
14:16.40robin_szI mean for like web users calling into our sales team
14:16.50robin_szvia a web phone thing ..
14:17.02file[class]hi anthm
14:17.11robin_szcan I just set a single, hardwired "webuser:webpass" ?
14:17.12ManxPwrPete_Largo, Well yes, but specifically for analog loops (i.e. wall jack to channel bank)
14:17.13anthmhi file with class
14:17.13*** join/#asterisk freat (~freat@c-67-162-109-129.hsd1.il.comcast.net)
14:17.19robin_szor does it need to be unique?
14:17.20freatanyone else Teliax?
14:17.26freatI've got 2 sites down right now
14:17.27ManxPwrfreat, I do.
14:17.35freattry calling in
14:17.48freator out
14:18.03Pete_Largoyou'll need what's called a "Butt Set" don't laugh.  the most common brand is Harris, but I use something called a 'lil Buttie'
14:18.35freatManxPwr: let me know if it's working for you
14:18.40ManxPwrfreat, looks broken
14:18.44freat:(
14:18.47freatcrap
14:19.01file[class]you all need to attend ClueCon
14:19.03file[class]it's where it's at!
14:19.10ManxPwrPete_Largo, A butt set isn't going to measure db levels of audio, loop length (cap kick), ohms
14:19.19robin_szcluecon is gay!
14:19.27ManxPwrfreat, teliax is usually pretty reliable.
14:19.33file[class]and whoever owes me stuff for help better go!
14:19.42ManxPwrfreat, go to their site, go to support, click on the web chat app
14:19.43file[class]so you can buy me drinks :P
14:19.53robin_szits in the USA?
14:19.54HAcluecon is in august right?  if so, then i try to swing by if things work out financially.
14:20.02doolphwhat is Queues
14:20.04file[class]yes
14:20.11file[class]Cluecon is August 3rd to 5th in Chicago, IL
14:20.32ManxPwrrobin_sz, Not everyone in the USA likes bad beer
14:20.34Zeeekteliax is "unreachable" here too
14:20.36*** join/#asterisk Qiang-zh (~Qiang-zh@CPE00112f93a8b1-CM00003964ca63.cpe.net.cable.rogers.com)
14:20.47robin_szManxPwr: true, but its all they can get
14:21.08ManxPwrPete_Largo, Right now we need something to generate 1000hz / 0db test tone, and measure the db level
14:21.22file[class]anthm: I looked at the design studies Chad did :) some of the stuff looks great
14:21.25robin_szManxPwr: in analogue?
14:21.32ManxPwrrobin_sz, no, the isa does have some good beer, but only from the smaller breweries
14:21.43anthmwhich ones you like
14:21.52ManxPwranthm, Sierra Navada is good.
14:21.54robin_szTheakstons Old Peculiar
14:22.01file[class]anthm: I can't remember now, I looked lastnight
14:22.07*** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net)
14:22.19robin_szI fed that to a 'mercan once ... he failed to finish his pint.
14:22.27file[class]anthm: all the entry slips looked great, and the ID tags were great too... hard to decide on the best
14:22.44*** join/#asterisk morris (~morris@pcworkshop.plus.com)
14:22.46anthmoh we talking beer and cluecon logos at the same time lol
14:23.03morrishow can i confirm installation of x100p clone ?
14:23.08file[class]well there was a pic where beer was associated with cluecon...
14:23.12morrisive installed zaptel
14:23.23morrisi have no idea how i check it working
14:23.47robin_szManxPwr: if you are in the UK and want a good solid audio test set, get an ex-bbc EP14 test set. utterley dependable.
14:23.49morrisi have rang the pbx line and asterisk doesnt show anything relating to it being dialed.. i guessed it should say something even if there was no rule
14:23.59Pete_Largohttp://www.acterna.com/global/products/finder/products_by_technology.html#3
14:24.02ManxPwrrob-, usa
14:24.06anthmGrolsch and Hacker Pschorr Weiss !
14:24.09freatrobin_sz: Old Peculier (I believe this is how it's spelled) is quite nice. molassas kind of flavor (mmmmmm....)
14:24.12ManxPwrrobin_sz, usa
14:24.22freatand I'm in Chicago, USA
14:24.25freat;)
14:24.42anthmyah you are in chicago , come to clue con bring your friends
14:24.45file[class]freat: you should attend Cluecon
14:24.53ManxPwrLinksys routers suck
14:24.56freatreally it's in Chicago!!!
14:24.58freat??
14:25.04file[class]yes
14:25.07ZeeekLinksys routers rock
14:25.08freatyes!
14:25.08file[class]http://www.cluecon.com/ for details
14:25.15Ahrimaneswho talked about beer??
14:25.24ManxPwrZeeek, not when they start crashing when put plug a SIP device into them
14:25.26*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
14:25.32freatI'm gonna have to go drinking if Teliax doesn't come back up
14:25.37Zeeekmy routers don't do SIP!
14:25.42freat2 corporate offices are down now...
14:25.46Zeeekfreat have 10 providers
14:25.49ManxPwrZeeek, neither does the linksys
14:25.52file[class]anthm: I wish we had more confirmed people ;(
14:26.06freatZeeek: how do you handle inbound calls though?
14:26.10ManxPwrfreat, Dude, if you rely on VoIP for mission critical stuff...well just kill yourself now.
14:26.21Zeeekah, that's tough. Nufone has some kind of alternate routing
14:26.28freathmm
14:26.36Zeeekbe sure and use voip for all your 911 nees
14:26.43freathaha
14:26.44ManxPwrfreat, What did teliax say when you talked to them?
14:26.50Zeeekheh
14:26.52freatyou can't call them
14:26.53freathehe
14:26.54*** join/#asterisk jbjb (ito@justine.donck.net)
14:26.56anthmwe just gotta keep reminding the ppl with so many questions that we have all the answers ... at , you guesed it! ClueCon!
14:26.56Zeeekemail?
14:27.03freattrying
14:27.10ManxPwrfreat, um, I always got very goos uspport via their web chat support app
14:27.12freatthey just went down maybe 20 minutes ago when I was on the phone
14:27.12vaewynI should come to cluecon even if work won't pay for it...  is so close by and I get to meet more devel types :}
14:27.18freatyeah nobody is online there
14:27.29ManxPwrfreat, ah.  wait a few mins and try again
14:27.32Zeeekmayeb the dialup link went down?
14:27.34syleyeah i had this linksys , netgear router fight with people before, just go dlink all works good
14:27.43jbjbHello ... Ik have build errors on mac os 10.3 ld: Undefined symbols:
14:27.44jbjb_res_9_dn_expand
14:27.44jbjb_res_9_nclose
14:27.44jbjb_res_9_ninit
14:27.44jbjb_res_9_nsearch
14:27.44ManxPwrsyle, naw, get cisco
14:27.44jbjbMakenshi: *** [asterisk] Error 1
14:27.48Zeeeknetgear sucks!
14:27.49jbjbany one ?
14:27.50freatgonna switch outbound to voipjet now...
14:27.57file[class]come to Cluecon where ALL your questions will be answered, such as "Why can't I use _."
14:27.58sylecisco, whos gonna pay that price lol
14:28.06vaewynManxPwr: linksys is cisco now   :}
14:28.08anthmwork should pay for it! , we offer a free service where file and bkw will call your boss and urge him to send you
14:28.13ManxPwrsyle, someone that wants something reliable.
14:28.15file[class]indeed
14:28.23vaewynanthm: hehehehe :}
14:28.23ManxPwrvaewyn, Perhaps I should have said "an IOS router"
14:28.24file[class]we are VERY good at that
14:28.42HAanthm: just how much encouragement are we talking here?
14:29.01Zeeekanyone have any vintage "Cisco Kid" eposodes? It would be a great CluCon opener :)
14:29.10ManxPwrI suspect that a rev2 or rev3 of the linksys would work
14:29.30vaewynHeck... I love WRT54G units... they rock
14:29.34syleoiww fuck i hate this, i get a popup from MS after its update saying would you like to restart now?....i say no...it keeps popping up and saying that every couple minutes, most annoying thing ever
14:29.55Pete_Largothat does suck syle
14:30.08sylehave to reboot i guess what a joke
14:30.15vaewynsyle: Your only 1 knoppix disk away from fixing that permanently :P
14:30.22astoriaIsn't that the slammer virus, syle?
14:30.38astoria:)
14:30.49vaewynnah... just the M$ virus
14:30.54HAis this overkill or just right for doing ds3 to t1 conversions?  http://www.tritechcoa.com/product/291967.html
14:31.04file[class]Cluecon, you MUST and you WILL go to it
14:31.25HAfile: what's the cost?
14:31.59file[class]$650
14:32.05tzangermore than my ass can afford
14:32.05file[class]but that includes your hotel and lunches
14:32.18astoriaanyone going to supercomm?
14:32.31anthmjust email sellmyboss@cluecon.com with the phone # and we will call and ask him to send you
14:33.23astoriadamn, i want to go to cluecon now!
14:33.24ManxPwrThe cost is: one beer for ManxPower
14:33.39ManxPwrIt's one of the lower cost VoIP events
14:33.45ChujiToo many conventions
14:33.59HAanthm:  oh yeah, that's exactly what i need.  i've already asked my boss to spend $21,000 to put in a fancy new * system, and now I'm gonna ask him to pay airfare and con costs for me to go hang out with a bunch if assgeeks.
14:34.26anthmhttp://www.cluecon.com/images/shirtdesign/Carcrash.jpg
14:34.28tzangeranthm: one patch from 4222 got in
14:34.31astoria$21,000 for an asterisk system? thats gotta be some pretty hardcore hardware
14:34.31ManxPwrHA: No, it's "training courses"
14:34.33tzangeranthm: but not hte CNG one yet
14:34.37tzangeranthm: and my bridge is working now
14:34.48anthmyay
14:34.58tzangeranthm: I spoke with Mark; all sounds in zaptel are interruptible, including DTMF.  It's intentional
14:35.01HA'Training Courses'...that might work.
14:35.03anthmhmm who charges 21k to learn asterisk ?
14:35.12jbjbnevermind ...
14:35.13tzangeranthm: so I fix my bridge app such that it doesn't write until I get the DTMF complete event and the bridge works as intended
14:35.21tzangeranthm: but unfortunately it does not exhibit the problem <grrr>
14:35.34anthmstill 200ms shit ?
14:35.48tzangeranthm: so... for my next trick, I am going to dlopen() chan_zap.so and try to recreate it with that
14:36.00ctooleyUh, $21,000 for a complete system, implemented and training courses sounds pretty cheap to me.
14:36.00file[class]tzanger: wow, good luck with that
14:36.00doolph21k?
14:36.05Pete_LargoHA - http://www.interlinkweb.com/systemics/product.asp?sku=ADT+4205290L11
14:36.17anthmstep right up
14:36.20*** join/#asterisk syle (~blah@wnpgmb02dc1-61-40.dynamic.mts.net)
14:36.33doolphctooley what's a good price for you
14:36.43file[class]oh as well at Cluecon there will be uber-secret stuff that anyone who attends will see
14:36.50astoriaIt's the phones that really kill ya on cost..
14:36.52file[class]so see it first, at Cluecon!
14:37.04ctooleydoolph, depends on the number of phones, lines, and level of flexibility
14:37.05doolphi cant go to cluecon
14:37.30tzangeranthm: my bridge app does not show any data loss from zaptel when DTMF is being played
14:37.45vaewynAnyone got experience with checking/setting MWI lights on a norhell system from a * system?
14:37.46Pete_LargoHA - http://www.telepricing.com/wwwboard/messages/11541.html
14:37.49tzangerI get audio from both fds, alternating, every 20ms
14:37.59file[class]c'mon folks - it's not that expensive!
14:38.01*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
14:38.05tzangervaewyn: with an ATA on the nor*, *1<exten>
14:38.07tzangertoggles it
14:38.11tzangerno way to check it though
14:38.24file[class]3 nights hotel, lunches, plus the actual conference for $650!
14:38.25doolphbecause you are a rich guy
14:38.51*** join/#asterisk Andy-AAH (~piekoff59@CPE00508b0d60a7-CM000039b80a5c.cpe.net.cable.rogers.com)
14:39.17HAPete_Largo: In that case...http://www.tritechcoa.com/product/291959.html
14:39.27file[class]j/k
14:39.42*** join/#asterisk mogorman (~mogorman@207.111.174.1)
14:40.13astoriahow long do you have to sign up for cluecon?
14:40.20HAfile:i also have to come up with money for getting there and back from Tulsa, OK...and I'm not in the mood to hitch 700 miles.
14:40.21astoriahow many attendees are there?
14:40.24anthmuntil it arrives!
14:40.28doolphfile can I msg you
14:40.42anthmask bkw for a ride !
14:40.55doolphits for business
14:40.59astoriais the con anywhere near the L?
14:41.07HAanthm: where is bkw coming from and do I have to listen to him scream NEXT! all the way there and back?
14:41.07file[class]HA: you should talk to bkw_
14:41.12anthmelk grove village near the airport
14:41.23file[class]HA: he's coming from Mcallister
14:41.45HASon-of-a...no wonder he keeps yelling NEXT!!!
14:41.48file[class](however you spell it)
14:42.49Pete_LargoHA - did you look for http://www.carrieraccess.com/products/index.cfm/fa/widebank28.htm these as well?
14:42.59*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
14:43.17*** join/#asterisk szw2001 (~vip@web1.ningo.net)
14:43.51HAPete_Largo: What exactly is that?
14:43.57Pete_LargoM13 Mux
14:45.05Pete_LargoDS3 in DS1 (T1)  out
14:45.05*** part/#asterisk Qiang-zh (~Qiang-zh@CPE00112f93a8b1-CM00003964ca63.cpe.net.cable.rogers.com)
14:45.46HAIsn't that the same as this?  http://www.tritechcoa.com/product/291959.html
14:45.55darkskiezif you have a E100P and a tdm400 card, what determines what order the channel numbers are, eg, what makes the pri channels 1-32 and the tdm 33-36 ?
14:45.59morrisAnyone available to spend some time with me to understand what needs doing to get my ptsn line working with asterisk?
14:46.29anthmcome to cluecon and learn how to get your pstn line working with asterisk
14:46.36morrisi would love too
14:46.38file[class]and how to make your asterisk bake muffins!
14:46.39morriswhere is it btw?
14:46.39HAmorris: about $2,000 USD will get it up and running.
14:46.53morrishow about your lick my sweaty nut sack ?
14:47.00darkskiezdeal
14:47.02morrisyou*
14:47.05darkskiez:)
14:47.08morrislol
14:47.10morrisgood lad ;p
14:47.12anthmsee the topic
14:47.17HAwe charge more for that.
14:47.21morrisah i have no passport ;/
14:47.26Zeeekmorris and after all the promise you showed yesterday!
14:47.30anthmPBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm
14:47.37morriszeeek ;p
14:47.43HAthis is America.  You don't need a passport, just come thru Mexico.
14:47.46Pete_LargoHA - that would work fine, Adtran and Carrier Access are direct competitors
14:47.50morrisi have been trying ;p
14:47.54tzangerHA: hahahaha
14:47.58morris*goes back to the books* ;p
14:48.20Zeeekmorris there is a special program I can get you in if you qualify as a triple minority
14:48.37morriswtf lol
14:48.56[TK]D-Fenderfile[class] : Muffins? Eay enough.  I've got my * setup running my X-10 modules, so you'd just need a oven that is 110v switchable like a Barbie EZ-Bake oven and you're set!
14:48.59HAPete_Largo:  OK.  Just making sure it's the right equipment.  Didn't know we needed that until late yesterday right before we were about to tell the boss how much it was actually gonna cost to get his new 'high priority' project in place.
14:49.06file[class][TK]D-Fender: yay
14:49.10Zeeekother possibilities include being in the UK in BT or trying to use Broadvoice
14:49.29Pete_Largofast learning on your part :)
14:49.32[TK]D-Fender<- Acheiving the ludicrous "because it was there"
14:49.59onkeltimmbkw_: someone told me some hours ago that you would be the one that can explain to me what i have to do to get callgroups working on IAX clients?
14:50.17HAPete_Largo: I had my first * box up in under 5 minutes.  Haven't had any issues yet that i couldn't find an answer for.
14:50.42file[class]did I mention you should all come to Cluecon?
14:50.50Pete_Largogeez, make a guy feel good, I'm still trying to figure it out...
14:51.05*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
14:51.27HAi'm hardwired to think like a machine.  unfortunately, i still have to sleep from time to time.
14:51.54Pete_Largowhat's this sleep thing I keep hearing about?  Is it something that happens after your kids grow up?
14:51.55file[class]sleep is overrated
14:52.32*** part/#asterisk szw2001 (~vip@web1.ningo.net)
14:52.37Zeeekwhat and where is ClueCon and why would anyone want to go there
14:52.46HAsleep is very overrated.  that's why i crashed at 5pm yesterday and was up and running by 10pm with plenty of time to take a break and see ep3 before coming into the office.
14:53.09doolphagain Zeeek?
14:53.09robin_szhey, tzafrir!
14:53.09Pete_Largoep3?
14:53.09HAcluecon is da bomb and all the cool kids will be there!
14:53.30morrisroot@tikka:/dev/zap # ls
14:53.30HAStar Wars.  Went to a 3:30am showing.
14:53.30morris1  channel  ctl  pseudo  timer
14:53.35file[class]Zeeek: Telephony Developer Conference, Chicago, LEARN!
14:53.35tzafrirhi, robin_sz
14:53.42anthmwhy cluecon is a collection of all the voip developers in 1 place with lectures and q and a galore
14:53.53Pete_Largohow was it?
14:53.57Zeeeksorry, that text comes from a cronjob
14:54.01robin_sztzafrir: I just wanted to say thanks again for providing those zaptel .debs, saved my ass! :)
14:54.13*** join/#asterisk jamesewing (~James@pbx2.jsci.net)
14:54.20morrisaparently is hould be able to cat 1 and see data when i ring the line.. but i am unable to cat it due to an erro (cat: 1: No such device or addressr )
14:54.33Zeeekanthm have listned to them on the phone for 90 minutes... heheh then I woke up
14:54.37HAPete: the first half is great, the second half is depressing as hell but just because of the story line.
14:54.44tzafrirrobin_sz, you used them with standard debian asterisk debs or with asterisk built from source?
14:55.06robin_sztzafrir: with standard testing debain .debs
14:55.07*** part/#asterisk jamesewing (~James@pbx2.jsci.net)
14:55.10file[class]anthm: what's the Q&A session going to be like?
14:55.13*** join/#asterisk jamesewing (~James@pbx2.jsci.net)
14:55.25robin_sztzafrir: versions are very close
14:55.35vaewynfile[class]: bkw_ yelling 'NEXT!!!' quite a bit  ;P
14:55.36HAsample q & a for cluecon: who the hell are you?  NEXT!!!
14:55.42file[class]lol
14:55.50Zeeeklol
14:55.52anthmlike a college class
14:55.58file[class]whoever makes fun of bkw_ gets kicked out!
14:55.59anthmbig ass theater
14:56.00Zeeekwith or without the sex?
14:56.07morrisreboot brb
14:56.08anthmwith ppl to talk to on a stage
14:56.12file[class]and I HAVE the power
14:56.19vaewynZeeek: you have sex in class?  egads!
14:56.21vaewynhehehe
14:56.22tzafrirrobin_sz, also check out genzaptelconf from the package zaptel
14:56.23file[class]:)
14:56.35Zeeekvaewyn I wished!
14:56.46km-hmm, I wonder if the 7960 is smart enough to ignore ()'s around the area code
14:56.55HAfile: i have the utmost respect for bkw_ but he does yell NEXT!!! alot.
14:56.59Zeeekany live acts or rappers?
14:57.01robin_sztzafrir: I would, but this is running and we have a 5 way conference between USA, athens, geneva and london starting in .. ooh, 5 minutes
14:57.34file[class]NEXT!!!
14:57.42km-file: woot.
14:57.54km-elvis has left the room
14:58.06km-queue the exit music!
14:58.13file[class]can't, queues are broken
14:58.37anthmlmao
14:58.45HAi have put myself into a queue on our test box so i can listen to music.  the text box has good moh.
14:58.50HAs/text/test/g
14:59.00km-I need to get some good moh
14:59.01*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
14:59.07km-maybe I'll get episode 3 theme music!
14:59.19NewSolemorning folks
14:59.20file[class]anthm: I need to buy my plane ticket soon
14:59.24km-howdy
14:59.24*** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net)
14:59.30anthmspeaking of moh guess who added all the native moh support to asterisk so you can say goodbe to mpg123!
14:59.44km-anthm: you wrote a native mp3 player?
14:59.51HAkm-: then your message would be "Please hold while we connect you to Lord Vader." followed by the appropriate theme music.
14:59.57km-HA: hahaha
15:00.21[TK]D-FenderBetter yet, just heavy breathing.....
15:00.26km-yeah
15:00.29anthmc'mon you dont know about that, some communication we have here in this "communication" forum lol
15:00.34vaewyn'Lord Vader will force strangle you you in approximately 5 minutes... thank you for waiting'
15:00.48km-I bet a lot of tech support people wish they could use force strangle
15:00.55vaewynamen!
15:01.02file[class]I wish I could use the cluebat in real life through the monitor
15:01.04file[class]it would be GREAT
15:01.08km-so, how many people saw ep3 last night
15:01.18NewSoleanthm should have seen it the other nite
15:01.25vaewynI am so avoiding it after ep1 & 2 crap
15:01.31km-do not avoid
15:01.31NewSolein other channel
15:01.33km-go see
15:01.46[TK]D-Fendervaewyn : Eveyone is saying Lucas didn't FUBAR this time....
15:01.51bkw_haha
15:01.51anthmfile have dave set you up
15:01.59vaewynI am avoiding... when it hits the dollar theatre I at least won't feel quilty about the money
15:02.04file[class]anthm: oh no... not Dave... nooooo
15:02.05vaewynguilty even
15:02.14NewSolehi bkw
15:02.14km-<yoda>movie enjoyment you will experience</yoda>
15:02.26file[class]anthm: what's his SIP URL?
15:02.38anthmhmm
15:02.38bkw_its not public
15:02.44bkw_but i'll get you into the system like I am
15:02.50file[class]k
15:02.54anthmwe can make an iax url that leads to it
15:02.55bkw_we'll set that up tonight if thats ok
15:03.01file[class]yeah
15:03.04km-where's my damn 7960
15:03.09km-overnight shipment is never fast enough
15:03.14file[class]there's a flight I specifically want... it has, wait for it, SANE TIMES!
15:03.23sylelol
15:03.26km-file: no red-eye express for you?
15:03.30file[class]ha
15:03.38sylei finally got a place in canada i can get my 7960
15:03.41km-newsole: you wont be disappointed
15:03.54[TK]D-Fendersyle : for how much?  And licensed?
15:04.03file[class]bkw_: what times do you suppose you'll arrive/leave?
15:04.06ZeeekAre double-digit ip addresses really sexier or does yours just come up faster in rogue scanners?
15:04.12syle444.05 with PST
15:04.18syleCDN
15:04.19km-WHAT
15:04.22[TK]D-Fenderouch
15:04.23NewSoleack syle
15:04.32vaewynHehehe... show's my age that Ep 4 I saw once before birth... and once after... in the theatre :}
15:04.38bkw_file at cluecon?
15:04.40sylei;m still looking but only place i;v found so far
15:04.48file[class]bkw_: yeah
15:04.54bkw_2nd early
15:04.58bkw_leave friday late maybe
15:05.04km-oh, it's $350 american dollars
15:05.11km-that's not *that* bad
15:05.12file[class]k
15:05.12syleyeah
15:05.17anthmfile, 4990 in the same url as the 42
15:05.18NewSoleI got one of the cisco 7960 here still in box..... been used to make a total of 3 calls
15:05.30file[class]anthm: k
15:05.32km-I wasn't even born when star wars came out
15:05.35syleit needs a license?
15:05.36HAbkw_ you driving or flying?
15:05.38bkw_anthm, you have any special SMS Handlers you want written?
15:05.49bkw_HA depends.. I might fly...
15:05.51file[class]bkw_: we need to test our theory re mobile2mobile
15:05.52[TK]D-FenderI've got a Polycom IP500 & 600 coming in for testing free tomorrow :D
15:06.04vaewynkm-: I was august of 77 so I got both before and after :P
15:06.06[TK]D-Fendersyle : yes, if you ever want to get firmware upgrades....
15:06.15km-vaewyn: hehe
15:06.16sylelicense
15:06.19km-vaewyn: september 81 for me
15:06.22NewSoleno syle... it was fully loaded and given free to me by cisco
15:06.25vaewynyoungin
15:06.27vaewynhehehe
15:06.28syleoww you mean just a login to cisco to get the IOS images?
15:06.28km-yeah
15:06.35anthmdunno do we need one ?
15:06.41bkw_file yes we do
15:06.50file[class]bkw_: are you really going out the 5th late?
15:06.51anthmthe open one doesnt work ?
15:06.59[TK]D-Fenderbasically, then there's a 8$US +/- annual maintenance fee (Cisco Smart0Net)
15:07.04bkw_anthm no we'll use that
15:07.04HAbkw_: i was told if I stand beside the road on I-44 you would pick me up on your way to cluecon.  Is that true?
15:07.11bkw_but I have to build a handler to process and react to the messages
15:07.20bkw_so sms_map will have user_id,number,handler
15:07.27bkw_handler being the code to run the message thru..
15:07.28sylenewsole how much you want for it/
15:07.45km-haha
15:07.48sylevoipsupply.com btw they are 300US
15:07.52km-I wouldnt stand TOO close to the edge of I-44
15:07.56km-bkw may swerve but not stop
15:08.14bkw_so we can map our numbers to a custom handler that does more ;)
15:08.23NewSolewell its somewhat new..... and dust comes with it
15:08.24HAkm-: I'll find a large concrete barrier to stand behind.
15:08.28bkw_km be nice ;)
15:08.29km-bkw: what you guys hackin on?  It sounds interesting!
15:08.31file[class]bkw_: such a marvelous idea
15:08.32NewSolethe dust is free
15:08.44BeirdoHA: you'll get picked up as a hood ornament?
15:09.09sylehow come you don;t want to keep it?
15:09.18file[class]km-: SMS stuff with our platform :)
15:09.41file[class]it's uber sexy
15:09.45km-platform?
15:09.47HABeirdo: If it gets me to cluecon, I'll consider it.
15:09.54Beirdoheh
15:09.58iCEBrkrbkw_: Reminder. km- isn't nice :P
15:10.05km-hahaha
15:10.07km-bkw likes me
15:10.12bkw_ya be nice to km-
15:10.14iCEBrkrkm-: He's the only one :D
15:10.16NewSolewe deal with heavy firewalls..... and sip is not what we need.... have to go though 3 firewalls to internet
15:10.18km-hahahahahaha
15:10.24vaewynkm-: is nice as long as you rub the correct direction
15:10.28iCEBrkrlol
15:10.31HANewSole: Paranoid?
15:10.33bkw_you guys are so wrong
15:10.33km-mmm
15:10.34km-:)
15:10.38vaewynbwahahaha
15:10.39iCEBrkrbkw_: You're one to talk!
15:10.39file[class]everyone, come to Cluecon!
15:10.44NewSolecustomers are....
15:10.53file[class]useless
15:10.54NewSoleand we are vpn linked to customers
15:10.54km-I would have my bosses send me to cluecon if we had enough money
15:10.57HANewSole: understood.
15:11.08file[class]km-: sell your soul, it's worth it!
15:11.10bkw_km- let me at your boss i'll see him
15:11.14bkw_er sell
15:11.16km-where is it?
15:11.21km-ahh chi-town
15:11.22file[class]bbl
15:11.24file[class]heading to the physics lab
15:11.26vaewynbkw_: gettin a bit loose there ;P
15:11.27bkw_chicago
15:11.27NewSolesome are banks and insuracne companies
15:11.38km-august 3-5?
15:11.51km-hrm
15:12.02km-When's the registration deadline?
15:12.08NewSole400 CND... with new IOS flash
15:12.50astoriaAnyone have any good experience with any wireless bridges?
15:12.55km-Yow.  $650?
15:13.07km-OH, it includes the hotel stay?
15:13.08astoriaI got the senao bridge, and it blows. I've used the WET11 and that sucks too..
15:13.18iCEBrkrastoria: Yea, I480 is a wireless bridge.. It's pretty sturdy.
15:13.34vaewynastoria: CB3+ senao?
15:13.39iCEBrkrUnlike the Sunshine Skyway which has wires and they shut it down when it gets too windy.
15:13.43astoriavaewyn: yeah. it sucks.
15:13.45*** join/#asterisk forkqueue (~sam@spc1-ward2-5-0-cust27.bagu.broadband.ntl.com)
15:13.59vaewynastoria: mines rocks solid... and it's on a 4 mile link
15:14.15vaewynastoria: way better than the WET11
15:14.25astoriavaewyn: mine is horrible, has all kinds of problems with DHCP packets, drops the connection every five minutes..
15:14.48vaewynastoria: wierd...  we are using WRT54G nowdays though cause they are so darn cheap
15:14.48astoriado you run SIP over your bridges?
15:15.02*** join/#asterisk hypa7ia (~leigh@serifos.eecs.harvard.edu)
15:15.03vaewynyep
15:15.26astoriawould you use it in a business critical environment?
15:15.50vaewynnot the CB3...  the WRT54G I would with the sveasoft firmware and a few hacks
15:16.02astoriaiCEBrkr: who makes the I480 bridge?
15:16.07vaewynI have them running my link to campu... and that is 6.1 miles
15:16.22*** join/#asterisk Kernel_Core (Raph@101.229.dial-up.xter.net)
15:16.30astoriai'm trying to find a good bridge for a business critical environment..
15:16.36vaewynWRT54G does have a problem with it's auto-gain control though...
15:16.57astoriamaybe it's just not feasible yet to use wireless only
15:17.16*** part/#asterisk feklee (feklee@genba.ffii.org)
15:17.17Kernel_Corehi all ! how do I debug asterisk to show me the DTMF that I issue ?! ( I am useing asterisk as a Proxy betweeen Xten and Cisco )
15:17.29vaewynIf you do choose wireless the most solid units I have had are the older Cisco gear
15:17.37vaewynBR350 and such
15:17.49vaewynrock solid and can hanlde uber noise levels
15:18.11astoriahmmmm... yeah, i'm going wired for business critical..
15:18.13astoriacat5 baby
15:18.20vaewynhehehe :}
15:18.23Kernel_Coreno idea ?!
15:18.31vaewyncheap enough it's a good idea
15:18.57astoriait cost more for the city inspection of my last cat5 install then the hardware cost!
15:19.41pussfellerwhy on earth would the city feel like it needs to inspect cat5?
15:19.54pussfellersorry, im just an easily outraged anarchist
15:20.19vaewynheck... that falls in low-voltage in these parts...  we don't even have to pull permits :P
15:20.42astoriabecause low-voltage in our city is required by city code
15:21.07vaewynunion town?  :}
15:21.24freathey there... if I download HEAD... the new jitterbuffer runs by default right? I noticed some comment somewhere about it being controlled at compile time, didn't see anything in Makefile to indicate a change
15:21.25Moonwickwow, inspections required for cat5 installs?  what a sucky place to live.
15:21.43astoriayeah, it's a very blue collar town..
15:22.02Moonwickwhere is this?  I've never even heard of such a thing before...
15:22.05astoriawe just had the building finished, and the electrical was all inspected okay but the inspector wanted the low-voltage done too..
15:22.09astoriaLivonia, Michigan
15:22.15vaewynsounds almost as bad a McCormick place :}   You can't plug your laptop in... union electrician has to
15:22.21Moonwickah, that figures.  :P
15:22.37vaewynastoria: ohh that would do it!  :}   (Michigander myself BTW... greetz!)
15:22.48astoriaThe inspector failed me the first time because i didn't use enough twist ties on the girders..
15:22.59astoriai had to wire it way up in the girders of the building.. what a hassle..
15:23.02freatjust want to make sure I'm running the new jittebuffer... as I want to put 2 of these in for between sites
15:23.06HAwe had the sbc guy out yesterday to tell us what would be needed to pull in fiber for our ds3.  he has to pull it from 1750 feet away and we just have to file for an easement between the right-of-way and the building and provide a fire-resistant conduit once it's inside the building.  still gonna be 6 weeks before we see it.
15:23.24astoriawow, i thought i had it bad! lol
15:23.29grollojfreat: yes, the new jitter buffer should run by default.
15:23.36freatawesome thanks
15:24.04grollojjust chan_iax uses it
15:24.08freatas of HEAD yesterday, only thing that doesn't seem to be working for me is the SIP notify stuff
15:24.33freatall else seems quite stable. running that box in production today to really test it
15:24.37*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
15:24.55freatabout 15 polycom IP500s run to it, with VoIP termination / origination via Teliax
15:25.09HAso while i wait on the ds3, i have to finish designing the dial-plan and implementing fun things like pay-by-phone and music-on-hold and conferencing and all this other junk.
15:26.14astoriaany news on the FCC E911 ruling today?
15:27.20vaewynnope... but grab your a$$ and says your praryers
15:27.25vaewynprayers even
15:27.29freathehe
15:27.35astoriai'm full of questions today: anybody recommend a good board to use multiple TDM04s on, without an IRQ clusterfsck
15:27.51*** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com)
15:30.52jakepdevastoria - channel bank not an option?
15:31.03astoriatoo pricey for seven lines..
15:31.10jakepdevoh 7 lines
15:31.11astoriai'd rather just slap two tdm04s in it
15:31.12jakepdevok
15:31.38jakepdevi'm using two te100p cards without a problem
15:31.46astoriawhat server/board?
15:31.49astoriai'd prefer a dell..
15:31.56jakepdevit's an hp server
15:32.04astoriawhich one?
15:32.05jakepdevrather pricey :)
15:32.41astoriamaybe i should drop a call down to digium..
15:32.52freatHAHAHHA
15:33.06astorialast time i emailed them and they gave me a quick response.
15:33.07freatI just got a call from a telemarketer, and I heard the asterisk conference join tone
15:33.10astoriamaybe calling would be a mistake.
15:33.16freatlol
15:33.33freattheir caller id was all 000-000-0000 too
15:34.05astoriai want to use asterisk to build a phone sex line :D
15:34.19freatthat's easy enough
15:34.24freatbig conference call
15:34.28astoriayeah, really...
15:34.32*** join/#asterisk hypa7ia (~leigh@serifos.eecs.harvard.edu)
15:34.34astoriathats where all the $$$ is.. :D
15:34.57*** join/#asterisk bannerman (~bannerman@209.216.176.42)
15:35.18astoriaI think i'll drop an email on -users about the TDM04 and dell thing, see if anyone recommends anything...
15:35.29*** join/#asterisk denon (denon@synapse.subneural.net)
15:35.29*** mode/#asterisk [+o denon] by ChanServ
15:35.48jakepdevcan you take out the devices (USB, etc) that you may not need?
15:35.58jakepdevsound
15:36.05freatyeah
15:36.06jakepdevextra serial ports
15:36.14freatdisable all that stuff
15:36.26jakepdevshould free up some irqs
15:36.27freatyou'll need USB if you need ztdummy with 2.4 kernel tho
15:36.29astoriai'd need the serial ports for hylafax..
15:36.38astoriadoes each FXO eat up an IRQ?
15:36.41Qwellno
15:36.44Qwelljust the card
15:36.53astoriaso all i need are two free IRQs?
15:36.54freatbut if you go to 2.6 you can disable usb
15:37.00Qwellat least, thats how it is in my box...
15:37.00astoriathat doesnt' sound like an issue at all...
15:37.03Qwellwith one FXO :D
15:37.18Qwellfreat: Don't let file catch you
15:37.26xeet2any digium people here?
15:38.04jakepdevthey're here - and they could answer - you never know
15:38.08xeet2hehe
15:38.44xeet2ok, better question, anyone here that can give me a quantity price break on iaxy's?
15:38.53Qwellsales@digium.com
15:39.00xeet2besides the obvious =)
15:39.13Qwellkram will say the same thing
15:39.39jakepdevin addition to - if you don't get a response from the e-mail - he'll help you
15:39.49Corydon-wSo... anybody know what NuFone is going to do after the FCC decision this morning?
15:39.54denonyeah .. Ive got some in my trunk here ..
15:39.56QwellCorydon-w: What decision?
15:40.07astoriaI don't think they've made it yet..
15:40.11Qwelle911 bs or something?
15:40.15astoriaYeah.
15:40.20Qwelllame
15:40.20astoria120 days compliance, i hear.
15:40.21Corydon-wFCC ordered all VoIP providers to provide 911 services
15:40.27xeet2corydon: I'd assume nufone and all the other itsp's will sign up with some e911 termination services the ilecs/clecs are offering
15:40.27darwin35when is e-911 going to work
15:40.29astoriaThey haven't ordered it yet..
15:40.38*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
15:40.39QwellThey better open that shit up then, and ALLOW access to it
15:40.41xeet2tis what we're doing
15:40.45*** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca)
15:40.50xeet2verizon has been very helpful
15:40.54xeet2strange of them
15:41.04darwin35xeet you need to document how to do it
15:41.05denonverizon yeah, but hell south has sucked from what I heard
15:41.08astoriaWhy are the ILECs being compliant? are they worried about anti-trust or something?
15:41.17jakepdevhttp://www.tmcnet.com/tmcnet/articles/2005/fcc-voip-911-regulations-120-days.htm
15:41.19*** join/#asterisk cianhughes (~cian@g5.cian.ws)
15:41.22Qwellastoria: no, they never have to worry about that
15:41.25xeet2darwin35: why?
15:41.28denon"help! help! my router has caught on fire and .. hello? hello?"
15:41.29Moonwick"911?  My internets are down!"
15:41.32darwin35so others can do it
15:41.39xeet2mmm
15:41.48xeet2well we'll certainly do that after we're breaking even
15:41.50Corydon-wThe ILECs don't have to worry about antitrust as long as Republicans control the executive branch
15:41.59hypa7iaanyone know about loading SIP firmware onto a cisco 7960?
15:42.13xeet2right now I have about 80k sunk into this startup so I'm not about to share too many secrets
15:42.14darwin35any word
15:42.16Qwell"*hold*hold*hold*hold*snooze*hold*answer*  Yeah, my address is blah, blah, blah, city, state, zip, blah.  Hi, I'm being murdered."
15:42.19darwin35wrong window
15:42.42pussfellerrefusi ng 911 access would be a public relations disaster, thats why
15:43.21astoriawhy do they care about public relations if they're the only guy in town??
15:43.36xeet2that and, who wants phone service from a provider that doesn't do 911 when all the others do
15:43.46xeet2astoria:  there is no town on the net =)
15:43.49ManxPwr~docs
15:43.50jbotDocumentation 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
15:43.52pussfellercause, it could extend to legal liability, given the right media climate
15:44.00QwellManxPwr: Shouldn't you be flying?
15:44.03km-7960++
15:44.10outtolunci was just gonna say taht
15:44.14ronnanyone familiar with this ? app_meetme.c:1149 conf_run: Unable to write frame to channel: No such file or directory
15:44.32ronni get the error when meetme hangs up
15:44.35QwellManxPwr: I'll be at work.  Good luck.
15:44.43ManxPwrQwell, A2) My arms got tired
15:44.46Qwell;]
15:44.52anthmI was just in new orleans a few weeks ago
15:44.56Qwelloff I go
15:45.05ManxPwranthm, and you didn't call!
15:45.16anthmdidnt know the numba
15:45.32ManxPwranthm, nobody does.  that's why I have so much time to IR
15:45.33ManxPwrIRC
15:45.41anthmcheckpoint charlie is my favorite
15:46.12bannermanI've asked before, just looking for second opinions.. I'm putting 4 pots lines in, need an FXO. Should I just get a TDM400P? Or should I be considering alternatives to digium hardware?
15:46.27ManxPwrbannerman, get a channel bank
15:46.27*** join/#asterisk jamesewing (~James@pbx2.jsci.net)
15:46.37astoriaa channel bank for four lines??
15:46.45ManxPwrastoria, why not.
15:46.51astoriabling bling
15:46.55ManxPwrIts reliable, powerful, and expandable
15:47.11ManxPwrAnyway what alternatives ARE there for 4-lines analog
15:47.14outtoluncwell a tdm400 with 4 fxo's is bling bling also <G>
15:47.23astorianot as bling bling as a channel bank!
15:47.28outtolunchehe
15:47.39ManxPwrI've been pretty with the TDM400Ps as of late.
15:47.50km-pretty with them?
15:47.51ManxPwrtzanger, you around?
15:47.52hypa7iacisco phones make hypa7ia want to smashy smashy
15:47.53km-or fed up with them
15:48.02astoriaare there any cheap channel banks for 12 lines or less?
15:48.13ManxPwrI've been pretty fed up with the TDM400Ps as of late.
15:48.20km-manxpwr: how bad are they?
15:48.24bannermanManxPwr: I can't afford a channel bank, we don't have what most people consider phone service out here.
15:48.24km-I just ordered two for clients
15:48.29km-don't tell me they're shit
15:48.37ManxPwrkm-, the one in our corporate server (REV H) locks up about once a month.
15:48.40astoriasays km- in a panic...
15:48.58*** join/#asterisk santiago (~santiago@63.245.86.248)
15:49.01km-ah
15:49.03ManxPwrkm-, FXS modules only.
15:49.06km-my tdm400p does that too
15:49.10km-I set my box to reboot once a week
15:49.11astoriakm-: are you putting them in the same server?
15:49.13km-that seemed to fix it
15:49.19km-astoria: no, two seperate clients
15:49.25astoriadrats!
15:50.13xeet2anyone want to do some iax termination testing?
15:50.32ManxPwrwhen I get back from vacation we'll be working with digium support to try to fix the problem.
15:50.36astoriayou do the termination?
15:50.39xeet2yes
15:50.41astoriasure.
15:50.55astoriai've been testing my pbx all week, i'll test your term while i'm at it
15:53.53km-why is this damn 7960 not registering...
15:53.58km-it can make calls
15:54.04km-but for some reason it's not registering
15:54.17astoriaahhh i spent all day yesterday screwing with SIP subscribe...
15:54.28km-Ah, I remember 7960's now
15:54.29km-heh
15:54.33*** join/#asterisk Cresl1n (~matt@216.207.245.23)
15:54.36km-once you figured out why they were broken they were fine
15:54.42km-but each one suffered from just a slightly different oddity
15:55.01km-usually it's all your fault in the config file
15:55.02km-hehe
15:55.16km-wow, 96 polycoms?
15:55.16Jas_Williamskm-: make sure you have proxy_register: 1
15:55.17Jas_Williams<PROTECTED>
15:55.27*** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
15:55.29ManxPwrkm-, *nod*
15:55.38*** join/#asterisk exonic (~exonic@209.172.11.54)
15:55.42ManxPwr30 of them are deployed, the rest arrived yesterday.
15:55.43km-yeah, I've got that
15:55.43exonicHeya #asterisk
15:56.00km-iI got a message on the console about it trying to register when I had the incorrect password
15:56.06ManxPwrIf you dance would you be an exonic dancer?
15:56.08exonicDoing *67 before the #, for inbound on Zap, doesn't actually block the caller ID
15:56.20km-but now that I've got the credentials right, nothing shows up on console
15:56.35exonicManxPwr, I sure hope so
15:57.53jamesewingWell the FCC just gave the RBOCs a new revenue stream
15:58.08jamesewinghttp://www.tmcnet.com/tmcnet/articles/2005/fcc-voip-911-regulations-120-days.htm
15:59.51denonjamesewing: old news :)
16:00.04km-that's weird
16:00.05*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-208-87.dsl.scarlet.be)
16:00.09jamesewingyes and no, not old since they just announced it
16:00.11jamesewingexpected yes
16:00.20denonjamesewing: well .. I meant that's been the buzz all morning
16:00.22km-I'm getting a 401 Unauthorized from the sip register
16:00.51jamesewingjust be sure to thank them for driving voip underground :)
16:01.12exonicDoing *67 before the #, for inbound on Zap, doesn't actually block the caller ID, Anyone ever have that problem?
16:01.18denonor forcing the telcos to provider easy access to 911
16:01.18*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
16:01.31*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
16:01.45jamesewingwe'll see how easy it is and how cheap
16:02.02km-hah
16:02.07km-I had to unload chan_sip.so then reload it
16:02.57*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca)
16:03.01DaLionhi guys
16:03.03DaLionkernel: Limiting icmp unreach response from 2235 to 200 packets/sec
16:03.11DaLionthink that could make choppy sounds ?
16:03.22DaLionicmp limits ?
16:03.43*** part/#asterisk crash3m (crash3m@crash3m.user)
16:04.15*** join/#asterisk Grooby (~Grooby@66.160.105.186)
16:05.30vaewynegads... what is generating that many ICMP unreachable responses?
16:05.33vaewynand yes... it can
16:06.15DaLioni dont know
16:06.18DaLioni see it in messages
16:06.27DaLioniptables time
16:06.34vaewynsounds like a DoS
16:06.54*** join/#asterisk morris (~morris@pcworkshop.plus.com)
16:07.22vaewynIf you have iptraf or ethereal check out the traffic... should give you a source
16:07.55DaLioncan ethereal run on non gui ?
16:08.22morrisanyone got any suggestions why asterisk doesnt see the phone ringing (zaptel).. but when cat /dev/zap/1 it gets disupted when i call the line
16:08.28vaewynumm... not sure... I just use iptraf :}
16:08.32morrisdisrupted
16:09.01km-tethereal
16:09.05km-is the console version of ethereal
16:09.36DaLioni need to recompile kernel for ipfw ? on fbsd 5.4 ?
16:10.41astoriathe fcc just made their ruling
16:10.44astoriasee fcc.gov
16:10.54denonhours ago :)
16:10.55denonkeep up man
16:11.12astoriawell, they just put it up a few minutes ago..
16:11.15*** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
16:11.32*** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm | The FCC ruled for forced voip E911 - http://www.fcc.gov
16:12.23Moonwickso I wonder if the FCC is going to bang down my door if I don't provide E911 for myself.
16:12.32jamesewingLOL
16:12.44jamesewingonly if its your primary line :)
16:12.46astoriaphew, the ilecs are required to allow access to E911 networks
16:13.02xeet2yeah its actually surprising how they're responding
16:13.09xeet2we'll see what they actually charge for the service though
16:13.32jamesewingyeah, my only worry is high deposits and connection charges
16:13.38astoriathe fcc seems to be taking this obligation pretty seriously..
16:13.57*** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm | The FCC ruled for forced E911 by pstn-terminated voip carriers - http://www.fcc.gov
16:14.00denonthere, happy? :)
16:14.20xeet2happy is relative
16:14.31bkw_does this VoIP thing only count for "Primary" Line providers?
16:14.35hypa7iawow, that's even stricter than the canadian ruling
16:14.36denonif bkw ain't happy, aint nobody happy
16:14.44xeet2bkw: apparently yes
16:14.45bkw_we are by no means a primary line provider
16:14.54denonbkw: I think its really just anyone who provides residential DIDs or somethin
16:15.01xeet2vonage markets themselves a primary line provider
16:15.11xeet2er, as a
16:15.23bkw_denon, we can do business stuffs?
16:15.25denonhopefully carriers will make it cheap enough where theres no reason not to
16:15.26*** join/#asterisk santiago (~santiago@63.245.86.248)
16:15.39denonbkw: no no .. I havent read it, im just guessing
16:16.10denonwhat I have read, implies anything that can make and receive calls to the pstn
16:16.48astoriayep, but there is an exclusion for certain services.
16:17.06denonyou mean the IM stuff they mention?
16:17.28astoriaYeah
16:17.52denonI think thats excluded because you cant "make *and* receive" to the pstn
16:18.02denonie, no DID I guess
16:18.25jamesewingthe new DID will be called "DIM" :)
16:18.48bkw_ya
16:18.53bkw_looks like we are gonna have to do this
16:18.58zoahey ho brian
16:19.10denonbkw: wait a couple weeks, a cheap solution will emerge
16:19.23jamesewingyeah teamspeak :)
16:19.28bkw_zoa what up
16:19.33bkw_denon, doubt it
16:19.37bkw_it will be an expensive solution
16:19.40jamesewingno seriously though i just hope that that happens
16:19.41bkw_unless we can do an open source one
16:20.06zoathat 911 thingie is not good news for small voip providers
16:20.18astoriait's not terrible news though.. could be worse..
16:20.20zoapeople will just go out of the states
16:20.23*** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr)
16:20.31zoathey cant force voiptalk to offer 911 in the states
16:20.33astoriathey could have NOT included the clause about the ILECs have to allow interconnection
16:20.47vaewynhey,.,, they aren't requireing GPS receivers in every VoIP router... we should be cheering!
16:20.59jamesewingWOO HOO
16:21.07bkw_but they will in the future
16:21.11jamesewingactually i think that would be the best id for true mobility
16:21.24bkw_Finally, the Commission stated its intention to adopt, in a future order, an advanced E911
16:21.24bkw_solution that includes a method for determining the customer’s location without the customer
16:21.24bkw_having to self report this information.
16:21.25astoriato protect against terrists of course...
16:21.25denonGPS in every router isnt so bad. . all our cdma phones have it
16:21.35vaewynGPS doesn't work indoors...  ie useless in 90% of voip situations
16:21.55vaewynluckily they thought of that first
16:22.13denonvaewyn: radio triangulation service purchased from a wireless carrier :)
16:22.41jamesewingi dont have an issue with the E911, but i dont think the government should be demanding it, i think that customers and the market should make the demands
16:22.50vaewynareas without cell service but with dialup   :}
16:23.10bkw_jamesewing, yep
16:23.22denonjamesewing: customers are too stupid .. I know several people who have blindly bought vonage, dumped their pstn, and have no cell
16:23.29HmmhesaysI wonder how many sip to sip non transcoded calls a dual 3.0ghz xeon machine would handle
16:23.43bkw_if you handles the media 600
16:23.46bkw_no media 5000
16:23.47bkw_or more
16:24.04HAbkw_: define what media please.
16:24.06vaewynbtw... instant business opportunity...  make a DB of local emergency call center's real numbers...  sell to voip providers for small monthly fee...  profit!!!
16:24.16bkw_it carries the media.. in and out of asterisk
16:24.18jamesewingso require better warning labels, not force monetary benefits and penalties
16:24.18astoriait's not that easy...
16:24.21bkw_if you allow reinvites
16:24.24bkw_it will do more
16:24.30Hmmhesaysi thought asterisk had a limit of 960 calls
16:24.32Hmmhesaysin sip
16:24.41drumkillathere is no hard limit
16:24.46bkw_no
16:24.52bkw_you ulimit might nail it
16:24.52zoathere is no hard limit
16:24.57zoathere was some limit last year
16:24.58bkw_but I had 5550 calls on the box
16:24.58*** join/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk)
16:25.00Hmmhesayswas there previously?
16:25.00zoabut its on longer there
16:25.01xeet2vaewyn: unfortunately the 911 service center numbers are a tightly kept secret
16:25.03Hmmhesaysahh
16:25.17HAso your definition of media is if all traffic is forced to travel thru the * box, right?
16:25.18bumperhostingcat /dev/zap/1
16:25.19bumperhostingspews out loads of characters..
16:25.19bumperhostingmodprobe zaptel doesnt give errors..
16:25.19bumperhostingasterisk doesnt display anything when the line is ringing..
16:25.24bumperhostingwhat can i be doing wrong?
16:25.26denonwar dialing for 911 centers <G>
16:25.31sudhir492vaewyn: How get the local emergency centers phone numbers?
16:25.37vaewynxeet2: until they required this...  :}  now there is an FCC mandate requiring their knowledge :}
16:25.40bkw_sudhir492, you PAY
16:25.44drumkillabumperhosting: check zapata.conf
16:26.01xeet2vaewyn: I didn't see that requirement in the ruling
16:26.09vaewynsudhir492: call them... that's how I got them around here
16:26.17sudhir492to whom? If it is not much, I am willing to pay and publish the list to Asterisk groups
16:26.44sudhir492vaewyn: who do you call? Call 911 and ask for real numbers ?
16:26.49vaewynxeet2: If you must route calls to them the only way you can do that is by having the number... hence... the numbers must be made available in order to comply with the law... hence numbers are now open info...
16:26.51Hmmhesaysgeebus these sysmax boxes at tigerdirect are cheap.
16:27.06xeet2vaewyn: mmm
16:27.07HAhow bout we all push to get an * box installed in each 911 center that allows for direct IAX Communication for VOIP?
16:27.13vaewynsudhir492: That's what I did around here... call the non-emergency number though...
16:27.14xeet2hahaha
16:27.19xeet2HA: good luck
16:27.24km-sip transfer (blind or attended) to a zaptel channel
16:27.26km-does it work?
16:27.32sudhir492vaewyn: Around where?
16:27.40km-I tried doing a blndxfer from the 7960 to a zap channel and the call vanished
16:27.49vaewynsudhir492: Berrien County Michigan
16:28.02vaewyncouple counties in northern indiana also
16:28.17bumperhostingdrum killa check your pm my friend:P it was to big to paste in the channel
16:28.22sudhir492very good. I am going to do that in DC metro
16:28.36km-it was my own stupidity, nevermind
16:29.06astoriait's not as easy as just dumping calls to the local E911 number is it??
16:29.22HAi wonder if digium would consider donating hardware for putting * boxen into 911 centers to allow for direct IAX communication via VOIP.
16:29.33denonHA: are you kidding?
16:29.37denonwtf would you want that?
16:29.43astoriaYou have to remit the location of the customers, as well..
16:29.53jamesewing6000+ PSAPs
16:29.55jamesewinghmmm
16:30.12denonand 20% of them with any kind of broadband connection ..
16:30.14denonnone with QoS
16:30.17bkw_the 911 network can take calls via the PSTN
16:30.20`Saurondenon: Because HA doesn't understand all the implications and requirements of doing E911.
16:30.22bkw_but the operators dont normally do this
16:30.30HAthink about it.  it would make it really easy to comply with E911 if all you needed to do was establish an IAX Connection followed by an IAX Text Message.
16:30.32bkw_each PSAP has a PSTN number to receive calls.
16:30.43bkw_those numbers are not published
16:30.58tzangerManxPwr: I am now but heading out
16:31.01denonHA: you'd not want to rely on local IP connectivity .. use the PSTN, it's built for this
16:31.37denonunless you feel like building a large several-thousand node network, with frame-relay PVCs for each 911 call center
16:31.42vaewynbkw_: this ruling is nice though... makes it so they either have to publish those... or the ruling is unenforceable
16:31.46HAdenon: I Agree that the PSTN is much more reliable.  It's just a thought.  I need more sleep before I can finish the thought thou.
16:31.48*** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca)
16:31.59tzangerManxPwr: /msg me, i'll be back
16:32.12vaewynbkw_: or I should say... 'allow their publishment'
16:32.31denonvaewyn: doesn't say they have to comply for free, does it?
16:33.02vaewyndenon: no... but it makes it available... even if a clearing house does it... it will be doable
16:33.26vaewynup till now that hasn't been available in many areas where they are aparanoid
16:34.00vaewynis stupid that... you can call 911... but you can't call the number 911 calls...
16:34.04vaewynwhere is the logic in that
16:34.29*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
16:34.34denonvaewyn: if you dial 911, its pretty hard to forge stuff .. but a regular number can come from anywhere.. could launch a DoS I guess
16:35.00vaewyncan DoS 911 alos... is the same number
16:35.20denonvaewyn: right, but they can have more info about you
16:35.30vaewyn911 is just a fancy forward
16:35.30denonor the local switch could isolate you
16:35.34denonand lock you out
16:36.00denonhard to lock out 50,000 pstn calls coming in from the outside world with forged ani
16:36.29vaewyndenon: All their info is gotten via the billing info of the number on callrid... that is E911... pretty cruddy ain't it :}
16:36.48denonCID or ANI?
16:36.54vaewynnow providers hav to give them that info... but still..
16:37.05vaewynANI,CID both...
16:37.11vaewynuses ANI first
16:37.26denonnod
16:37.58*** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net)
16:38.38*** join/#asterisk mikewho2 (~Schnappi@ip68-105-227-82.lu.dl.cox.net)
16:39.01mikewho2Hey guys, If i wana get a box running that handles 20+ COs should i just get like 5 diff TDM diguim cards?
16:39.16mikewho2and it seems to me
16:39.18mikewho2CO lines
16:39.25mikewho2have the possibility of haveing a huge surge
16:39.37mikewho2running through and burning my card+pci slot+ 3000$ server
16:39.45mikewho2should i have it through a surge protector?
16:39.48vaewynnope... channel bank with FXO cards... and a T1 card
16:40.08mikewho2vaewyn hrmm... so just order a T1, that has 24 channles
16:40.10HAwhat is the chance of getting a surge from a ds3?
16:40.17mikewho2can you expand on that vaewyn
16:40.35vaewynand if you are getting 20+ lines it might be cheaper to get it directly as a PRI from the phone company
16:40.46mikewho2What is PRI
16:40.54mikewho2do you have to go through the FCC for anything like that
16:40.57mikewho2get any sort of liscense?
16:41.00astoriawhat??
16:41.01`SauronGoogle is your friend.
16:41.07`SauronUse it.
16:41.12mikewho2i was hoping you were my friends too :)
16:41.17astoriaPRI=voiceT1
16:41.18mikewho2maybe thats making too many assumptions
16:41.29newmedianmikewho2 you need to do some more reading so you have some context
16:41.38vaewynPRI = 23 voice channels and 1 data channel...
16:41.55astoriait's not really a data channel per se...
16:41.59astoriai mean, you cant use it for data.
16:42.04astoriacan you?
16:42.09vaewynyou can run as many phone numbers as you want through that
16:42.18mikewho2I would like to know what T1 card what channle bank with FXO is.
16:42.27astoriathat you're going to have to google.
16:42.28vaewynit's a data channel in that it carries the call initiaton data
16:42.31astoriait's extensive..
16:42.35mikewho2i see...'
16:42.39newmedianYou may also want to read some of the docs
16:42.40HAthe data channel is what controls the calls on the other 23 lines.
16:42.40newmedian~docs
16:42.41jbotDocumentation 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
16:42.51*** join/#asterisk strafbomber (~strafi@p54A738F6.dip.t-dialin.net)
16:43.03strafbomberhello
16:43.05vaewynmikewho2: If you are dealing with that many lines from the phone company skip the channel bank and just get a direct PRI line (T1) from them
16:43.13mikewho2would it be completly unwise to run 4 or 5 TDM cards?
16:43.21mikewho2because i want to have the flexibility to plug it in anywhere
16:43.23astoriayes it would be
16:43.24vaewynyes
16:43.28mikewho2and not everywhere you can drop a T1 card in
16:43.31jontowmikewho2; yeah.. you'll probably have some issues there
16:43.41mikewho2Where do the issues come in?
16:43.43astoriathat would be a clusterfuck of IRQ problems
16:43.45jontowinterrupts mainly
16:43.54mikewho2assuming i have enough IRQs
16:43.55*** join/#asterisk Klaus (~klaus@sconk.dk)
16:43.56astorianot to mention pricey
16:44.02mikewho2and i could get all of them working
16:44.04*** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com)
16:44.10vaewynIf you insist on using normal phone lines... use a channel bank and a T1 card...  but try for a PRI directly to your T card
16:44.12astoriawhy can't you get a PRI dropped in?
16:44.13*** part/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com)
16:44.23astoriait will be WAAAY cheaper
16:44.36newmedianmikewho2 are you looking for a mobile solution? i.e. transportable on a day-to-day basis?
16:44.40mikewho2well, this server may change location.... and I dont wana have to explain to the phone comp. why i need a T1
16:44.44jontowhow the hell does one afford to have 23-24 regular phonelines dropped but not a single PRI?
16:44.45mikewho2well, i want it to have that flexibility
16:44.46vaewynVoice PRI are like 450$/month or less...  way cheaper than 20+ real lines
16:45.01astoriathey are just as flexible..
16:45.09astoriabut you're going to have contracts to sign on both..
16:45.15mikewho2but can you get one running to a residential location in a week?
16:45.23*** join/#asterisk amir_ (~amir@195.226.9.186)
16:45.32strafbomberis it possible to use Wildcard TE410P directly whith linux? is there a capi driver available? i want to use this card for an dialin server whit pppd
16:45.44jontowmike; is this illegal? ;)
16:45.51newmedianmikewho2 you're not going to be running sequential diallers, are you? spamming people with calls and then moving on?
16:45.56mikewho2no no no
16:46.28*** join/#asterisk sharprock (~user@lan-gw.fullnoize.com)
16:46.33mikewho2If i did get the PRI solution, what t1 card would work with asterisks?
16:46.38vaewynmikewho2: You can't get 20+ lines to ANY residential home in a week
16:46.44mikewho2(pri does make a hella lot more sense)
16:46.54jontowT100P is what i've been using
16:46.56mikewho2vaewyn the only reason i say that, is cause i've done it before
16:47.03Silik0nmikewho2 any T1 card that works with asterisk will do PRI
16:47.04*** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
16:47.12jontowthey're pricy, at about $500 each
16:47.14*** join/#asterisk denon (denon@synapse.subneural.net)
16:47.14*** mode/#asterisk [+o denon] by ChanServ
16:47.15jontowbut damn do they work..
16:47.20mikewho2heh, a TDM card is like 400
16:47.24mikewho2with FXO and all that
16:47.25Silik0nso a T100P a TE4XXP, or any others
16:47.54faquanyone knows why i get "make[1]: *** [chan_zap.o] Error 1" , when i am compiling asterisk?
16:48.05jontowfaqu; pretty vague error.
16:48.14sharprockHello all, a quick question... I have broadvoice account running with asterisk.  It seems I can call my broadvoice # from multiple (2 tested so far) phones simultaneously...
16:48.18jontowfaqu; pastebin.ca for all of your code-compiling error pasting needs.
16:48.26sharprockI was expecting to get a busy signal on the second call...
16:48.28faqua sec
16:48.30sharprockbut it completed
16:48.34mikewho2When I get this T1 card, do they require FXO and all that, or is that not the nature of a t1
16:48.43jontowmikewho2; that is not the nature.
16:48.51mikewho2indeed
16:48.56mikewho2PRI makes 1000000x more sense
16:48.59mikewho2cheaper
16:49.01jontowyou'll need to know the framing and signalling that your PRI will use
16:49.01mikewho2more reliable
16:49.11mikewho2is that a hardware issue or software issue
16:49.21jontowesf,b8zs + pri_cpe is what i've used most, and what seems to be quite common
16:49.28vaewynT1 is 1 RJ45 cable... and voila... you are set  :}
16:49.41freathey what's that conference coming up in chicago?
16:49.43jontow23 B channels and 1 D channel for signalling
16:50.03newmedianit's more of a provisioning issue, with your service provider who is giving you the PRI;  They'll make assumptions if you don't specify. Sometimes you can't specify.
16:50.06vaewynyeah... national signalling with esf,b8zs, pri_cpe is what most places in the US use... commonly called NI2
16:50.35mikewho2so, i need to contact them, to see what they can provide, and buy my hardware accordingly?
16:50.37sharprockis it normal to be able to complete multiple simultaneous calls to a single number with voip?
16:50.42astoriafreat: supercomm
16:50.44astoriafreat: you going?
16:51.05astoriavaewyn: I got NI1 here with XO
16:51.17vaewynsharprock: yep
16:51.18xeet2astoria: xo gave you ni1?
16:51.32sharprockis there a limit on the # of calls or just bandwidth limited?
16:51.34astoriaxeet2: well, so they say. i haven't plugged in yet..
16:51.44mikewho2im assumning i can put 2 TDM card in there and one T1 card, so i got lots of flexibility and asterisks can handle them all the same?
16:52.01vaewynsharprock: depends on provider... some limit to 2 calls... others it is unlimited...
16:52.07jontowquite so, its just a configuration (and again, IRQ) issue from there
16:52.09sharprockvery cool
16:52.11xeet2astoria: most of the "technical" people at xo don't really have any idea
16:52.19freatI live in Chicago
16:52.21strafbomberhello, has anyone an answer for that? is there a capi driver for the Wildcard TE410P? or some other driver to run this card under linux?
16:52.23mikewho2that sounds like the best IDEA for me
16:52.24onkeltimmguys, since receiving faxes with spandsp i receive lots of empty faxes... logs tell me that the call was hung up. never had this with legacy fax. anyone experienced something similar?
16:52.26freatastoria: got a URL?
16:52.28xeet2astoria:  I had to explain rdnis to 8 people before I got to someone that knew what it was
16:52.42mikewho2so i got flexibility to move around, albiet at smaller capacities (8 co) and if im at my main location i got 24 lines
16:52.42faqujontow: http://pastebin.ca/12162
16:52.45mikewho2err 23 rather
16:52.47astoriabsdfreak: for xo? no.. just an email
16:52.49sharprockvaewyn: know off hand how many broadvoice allows?
16:52.53mikewho2hells yea, thanks again guys.
16:53.06vaewynsharprock: no clue... sorry
16:53.12astoriafreat: for xo? no just an email..
16:53.16sharprockvaewyn: thanks
16:53.18jontowmikewho2; if you have bandwidth in both places, you also have IAX2 to trunk calls to the main location, so as to gain your 23 lines from a remote place, too.
16:53.24mikewho2but one more question, do i have to buy T1 hardware in accordance with what national signlaing is used?
16:53.48vaewynmikewho2: is just a setting... same hardware
16:53.49jontowto a point yes
16:53.57jontowonly if the difference is T1 vs. E1
16:54.00astoriaxeet2: so you think my NI1 is really NI2?? :)
16:54.02jontowbut in the USA, no.
16:54.07astoriaXo hasn't dropped it yet..
16:54.16vaewynIn use... everything is T1 :}
16:54.19mikewho2With IAX2 I can just keep that one server on the T1 and use IAX2 to set up another server to use those 23 lines?
16:54.22jontow;)
16:54.23xeet2mikewho2: it depends also on the vendor
16:54.29jontowmikewho2; yes.
16:54.33newmedianIn the future all restaurants are Taco Bell
16:54.36mikewho2nice....
16:54.39vaewynmikewho2: yep
16:54.41jontownewmedian; i don't like that prediction
16:54.44mikewho2voip networks are so neat
16:54.50mikewho2imm be hiring some of you guys here soon
16:54.51mikewho2stay on point
16:54.55mikewho2anyone want some paypal money?
16:54.55astorianeato!
16:54.56jontowthey solve [and create] a lot of problems, thats for sure :)
16:55.05dsfrIs anyone having a problem with the latest CVS HEAD where you are no longer receiving DTMF output on the console even though you have debug specified in your logger.conf file?
16:55.06astoriawhat kind of biz do you run mike?
16:55.07vaewynwhat?? no club challopas for you?  :}
16:55.14mikewho2priv msg me if you helped me and want a few paypal bux
16:55.20mikewho2astoria im just a comp consultant
16:55.22xeet2astoria: unless you told xo NI1, it is most likely NI2
16:55.24mikewho2this is my first go around with voip
16:55.33*** join/#asterisk yxa (empty@cm162.gamma226.maxonline.com.sg)
16:55.41astoriaxeet2: morons.. it's a super hassle dealing with my account manager..
16:55.48mikewho2astoria telemarkiting
16:55.59astoriaooooh..
16:56.07astoriastop calling me at dinnertime.
16:56.10mikewho2:)
16:56.13mikewho2not that kind
16:56.14xeet2astoria: let me know when they drop the circuit, I'll give you a contact there that can turn on rdnis for you
16:56.15mikewho2only solicitated
16:56.19mikewho2the person has to want someone to reach them
16:56.24ronnanyone knows how to increase the digit timeout when using call transfer?
16:56.26jontowfaqu; .. you left
16:56.39ronnusing the # transfer
16:56.52mikewho2jontow what biz u in, that you need a 23 COs?
16:57.01astoriaxeet2: not turned onn by default?
16:57.04newmedianTangentially commenting, I don't like PayPal because it takes a week for the money to move into the bank account, and it can be snatched back at any time. That's a long time to wonder if you've just wasted time or shipped product for no reason. But in Canada there is CertaPay, which is tied into all the Canadian banks. www.certapay.com.  It puts money into bank accounts in about 15 seconds. No more worries. Great for eBay.
16:57.09jontowronn: ;transferdigittimeout => 3      ; Number of seconds to wait between digits when transfering a call
16:57.17jontowronn: check out /etc/asterisk/features.conf and see if thats what you want
16:57.18xeet2astoria: nope.  caller id name isn't even turned on by default
16:57.30jontowi have a PRI wired to my desk at work for development
16:57.33astoriaxeet2: sigh...
16:57.36ronnjontow: thanks
16:57.53jontowand i have a PRI wired to a telco voicemail server running asterisk near here, with a pair of data T1s between the two location
16:58.03astoriaxeet2: can they turn it on quickly?
16:58.05mikewho2hah, fucking cool
16:58.08vaewynnewmedian: I like ebay... because they honor ACH rules and I can stop payment if someone tries to take my money :P
16:58.13xeet2astoria: if you get to the right person
16:58.20jontowwe actually have 5 PRIs here
16:58.25astoriaxeet2: it's like dealing with the mob! lol..
16:58.25vaewynnewmedian: paypal that is
16:58.30vaewynerrgh
16:58.34mikewho2total bill of ~ 2000 a month?
16:58.36astoriaxeet2: thanks, i'll drop you an email when i get stuff plugged in..
16:58.42xeet2astoria: the guy I finally reached, and have his number, was able to make the change to the lucent switch while I was on the phone with him
16:58.48mikewho2and ~100 COs?
16:58.51mikewho2damn, thats nuts
16:58.52jontow2 for testing, 2 for a live callcenter (techsupport, inbound mostly), and 1 remotely but interconnected at the telco CO
16:59.14mikewho2does one have to get any sort of liscense from the federal govt to get a PRI
16:59.16jontowthe callcenter doesn't currently run asterisk; but my job is partly to try and change that :)
16:59.16astoriaxeet2: calling XO now...
16:59.20mikewho2Is it gonna raise any eyebrows?
16:59.25vaewynmikewho2: nope...
16:59.38mikewho2i dont wana get all tied up in red tape
16:59.44freatmmm... pulling on some strings see if I can get free tickets
16:59.47jontow(don't know.. i don't do the paperwork.. this place is partnered with a telco that provides PRIs with connection to the PSTN gladly, using a Nortel DMS10)
16:59.52vaewynno red tape... just stupid phone types
17:00.04*** join/#asterisk Kernel_Core (Raph@174.228.dial-up.xter.net)
17:00.05freator 'passes' rather ;)
17:00.15mikewho2and the IAX2 Trunk thing, is just setting up 2 asterisks together to share their resourses (CO Lines)
17:00.17mikewho2correct?
17:00.19newmedianvaewyn I've had people try to pull fast ones, and it takes moving heaven and earth with PayPal and eBay just to get them to do something. Unless you act instantly the moment you sense a problem, you're usually S.O.L. But I don't do much eBaying anymore anyway.  But PayPal is problematic, particularly if you're shipping hardware to someone.
17:00.24jontowyeah, its all done with iax.conf and the dialplan
17:00.37freatmikewho2: the key is that trunking saves on overhead from packet headers
17:00.51mikewho2so it lowers bandwidth use?
17:00.55freatmikewho2: exactly
17:01.10mikewho2seems like if you got a 5mbit line up and down, and only 23 CO lines, bandwidth would never be an issue
17:01.11freatthere's some documentation about that on the wiki
17:01.24jontowvaewyn :D
17:01.33mikewho2i asmit, i do need to learn more about trunking as i know nothing on it
17:01.39mikewho2is there info on that at voip-info?
17:01.41shido65mbit for only 23? err yeah
17:01.42jontowyep
17:01.44freatyeah
17:01.52jontowthere is more info there than you'll be able to read in a week's time :)
17:01.57mikewho2shido6 well theres gonna be more than VOIP goin
17:01.57freatI think the current implementation of jitterbuffer doesn't work under trunking though
17:01.57*** join/#asterisk loick (~loick@APuteaux-151-1-29-223.w82-124.abo.wanadoo.fr)
17:02.34freatmikewho2: I'm using linux as a bridge to do QoS for Video + VoIP + Citrix + Bulk on a T1 and am fine
17:02.38jontowi think i should spend today putting snort on the firewall in front of the webservers
17:02.39jontow:/
17:02.46shido61mb u can squeeze 96
17:02.46jontowwhat a crappy project that one is
17:03.01vaewynHmm... so now with a DS3/T3 for voice... and private fiber to the local PoP for data...   *evil grin* I think I am set for a while
17:03.02mikewho2i know its over kill, but i need overkill, its very mission critical
17:03.14ronnjontow: my box is from CVS-HEAD-10/01/04 .. i don't have no features.conf file
17:03.20freatmikewho2: what kind of connection is that?
17:03.27shido6room for 73 more cannels on the 1 mb and you ave 5
17:03.33mikewho2im not sure how im gonna get th service
17:03.35freatmikewho2: that you get 5/5 ?
17:03.40mikewho2shido6 voip isnt the main thing goin on here
17:03.41jontowhmm
17:03.47jontowronn: make one, then.
17:03.47mikewho2I dont know how i am going to get the serivce
17:03.52freatmikewho2: don't go wireless... we've been having all kinds of problems with that
17:03.52mikewho2its just what Im setting out for
17:03.56shido6wait
17:03.58mikewho2yea, thats what i heard
17:04.00mikewho2i was thinking
17:04.04mikewho2maybe ds3
17:04.06grollojfreat: i believe the jitterbuffer should work with trunking
17:04.07shido6whats your upload speed, mikewho2
17:04.08mikewho2some sort of fiber optic solution
17:04.14mikewho2It will be 5mbit up and down
17:04.21mikewho2because theres alot og VPN activity going
17:04.22freatmikewho2: nice
17:04.30mikewho2I do not know what solution i am going to get
17:04.33freatgrolloj: new or old?
17:04.37mikewho2all i know is I have all the money in the world to blow on it
17:04.44mikewho2and its very mission critical that this line is plenty fact
17:04.45mikewho2fast
17:04.48jontowall the money in the world hey
17:04.50freatmikewho2: sweet can I host a BF1942 server at your site?
17:04.50astoriai wish people would throw money at me!
17:04.56astoriawait, do you work for the government?
17:04.59grollojthe new jb. there was a trunking patch to chan_iax2 that went in recently
17:04.59mikewho2astoria no
17:05.06mikewho2astoria I asked you if you wanted some money
17:05.08mikewho2and no one even msged me
17:05.09freatgrolloj: ahh ok cool
17:05.20jontowi don't mind accepting money; but i don't even have a paypal account :)
17:05.30freatgrolloj: yeah I'm testing HEAD from yesterday in production today... it's been great
17:05.35astoriafree money? does this include future advising?
17:05.36newmedianwe're probably trying to figure if you're "evil" or "!evil" ;)
17:05.40HA\/msg mikewho2 we all want money and will never refuse it.
17:05.53mikewho2astoria no, just makes me a liked guy
17:05.54freatgrolloj: only problem has to do with SIP NOTIFY stuff... which I understand they are working on
17:05.56jontowyou're not supposed to come out and say that
17:06.09jontowthats the SECRET part, *cough*
17:06.15newmedianoops
17:06.28newmediandid I say that out loud?
17:06.34grollojfreat: i'm not familiar with that, but i'm glad to hear you're testing.
17:06.45grolloj(and that it's working)
17:07.00freatgrolloj: well I've really been itching for the new jitterbuffer + PLC
17:07.12HAmikewho2: you don't work for the mob do ya?
17:07.18astoriai can't turn down free money!
17:07.23bannermanmikewho2: omg, do you work for hillary clinton?
17:07.24mikewho2no you guys cant
17:08.03xeet2lol
17:08.23jontowhilary clinton eh?
17:08.28jontowi don't know about this..
17:08.40HAmikewho2: ok, you are about to receive a msg with a pp add to send payment to.  what's the average fee guys and how many do i need to collect?  it will be distributed when we all get to chi town.
17:09.01jontowwhois ha
17:09.07jontoweh, with the / this time!
17:09.45xeet2hehe
17:10.48*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
17:10.54|Vulture|anyone know a bit about PRIs here?
17:11.07xeet2|Vulture|: alot of us do
17:11.09jontowvulture; we've been having PRI-talk for the last 30-40mins :)
17:11.12newmediana robbed bit here and there
17:11.24|Vulture|hahaha, 1s lemme pastebin this its on a new PRI install never seen it
17:11.30*** join/#asterisk dgillson (~dgillson@rrcs-24-97-7-126.nys.biz.rr.com)
17:11.48jontowis it saturday or something?
17:11.57jontowi mean.. im the only damned person in this office.. i think i got ripped off
17:12.12|Vulture|http://pastebin.ca/12163
17:12.17HAjontow: maybe there was a bomb threat and they forgot to tell you?
17:12.35|Vulture|thats when I get when i restart * the PRI says Up,Active but its not seeing D-Chan
17:12.50|Vulture|I got the Provider working on it, didn't know if anyone has seen those errors before
17:13.12dgillsonwhere would I find docs on Asterisk upgrade 1.02 to 1.07
17:13.12*** join/#asterisk doolph (doolph@200.46.148.35)
17:13.23|Vulture|dgillson: make clean;make;make install
17:13.25|Vulture|;P
17:13.27doolphheh
17:13.31doolphjust upgrade it
17:13.41vaewyn|Vulture|: make sure you have the pri_cpe or pri_net set correctly...
17:13.52vaewynother than that...  not sure
17:14.05doolphtry to backup also
17:14.10doolph/var/lib/asterisk
17:14.16|Vulture|vaewyn: I don't have either sert
17:14.22jontowi'd be pissed.. but oh well.. you know, i came in at like 12:20 anyway, hehe
17:14.23|Vulture|oh
17:14.30|Vulture|I know what you mean, signalling
17:14.40vaewynyeah
17:14.49*** join/#asterisk meppl (mephisto@p54AAEABD.dip.t-dialin.net)
17:15.11newmedianjontow: everyone is watching Star Wars
17:15.33|Vulture|nope when I change to net I get "PRI Error: We think we're the network, but they think they're the network, too."
17:15.52shido6zapata.conf pastebin.ca
17:15.52jontowoh
17:15.55dgillsondoolph: thanks
17:15.55jontowscrew them.. they are
17:15.56vaewyn|Vulture|: That means the D channel is working
17:16.03shido6and ztcfg -vv , too Vulture
17:16.04HAwhats the dial command for the dev call?
17:16.04|Vulture|I think its the provider sucking it up
17:16.10jontowvulture; yeah.. pri_net vs. pri_cpe ?
17:16.20|Vulture|nah ztcfg -vv shows 100% working no errors
17:16.26|Vulture|zttol shows no alarms
17:16.28|Vulture|zttool
17:16.36vaewynyeah... so they are screwing something up
17:17.03jontowoh damn /me is not paying attention
17:17.17|Vulture|http://pastebin.ca/12164
17:17.20|Vulture|thats the configs
17:17.41*** join/#asterisk toot (chris@212.20.250.187)
17:17.46vaewynlooks correct
17:17.48|Vulture|I have another PRI from them.. and I heard from the guy at the location the adtran box started beeping this morning
17:17.54|Vulture|no error lights though
17:18.00xeet2|Vulture|: uhm, you sure you don't have a loop showing?
17:18.15xeet2if they're doing a soft loop in their equipment, you will see yourself
17:18.18|Vulture|xeet2: how do I test that?
17:18.31|Vulture|xeet2: she did say right before I saw that she was going to loop it up
17:18.32xeet2which would explain if you set it to network, it says its network, and if you set it to cpe, it things your cpe
17:18.48|Vulture|then I ran a loop, and now its doing this
17:18.55toothi. ztcfg shows one channel. show channels shows none. my system was working fine (x100p) till i had a few hung up calls, now naada :) any ideas very welcome
17:19.21xeet2|Vulture|: there are a bunch of different kinds of loops, some you won't see on the circuit but just take all the bits received and send them back
17:20.27|Vulture|xeet2: is there a way for me to see if there is a soft loop on the line?
17:21.07xeet2|Vulture|: I'm assuming you don't have a t-bird or sniffer that can do t1's
17:21.09*** join/#asterisk Shuri (sjnesjd@64.235.209.226)
17:21.16vaewynIf you send a call out you would get it right back :P
17:21.26xeet2vaewyn: hehe, well, it wouldn't even work
17:21.31vaewynpacketwise at least
17:21.32|Vulture|xeet2: I am not even at the location :(
17:21.45*** join/#asterisk dros7 (~simon@H125.C18.B96.tor.eicat.ca)
17:21.54tootztcfg -vv gives me Channel 01: FXS Kewlstart (Default) (Slaves: 01) - 1 channels configured
17:22.04|Vulture|okay Ill let them fix it
17:22.06|Vulture|thanx guys
17:22.09xeet2|Vulture|: the only thing I can think of is to create an hdlc interface in asterisk, and then do a tcpdump on that interface, and send traffic across it
17:22.18xeet2er, in zaptel that is
17:22.20|Vulture|ah okay
17:22.35*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
17:22.35xeet2hdlc will come up no matter what, even if its looped back to itself
17:23.16xeet2but before doing all that, just complain to them that they have a loop
17:24.36hypa7iahad anyone here had any luck loading the SIP firmware onto a cisco 7940/60 phone?
17:24.44shido6yes
17:24.55Kernel_Coreguys how do I debug DTMF on my asterisk ?! ( I am connected with SIP phone to asterisk , and asterisk is connected to another cisco sip , so asterisk acts as a proxy !)
17:25.00hypa7iawhat's the trick :-(
17:25.06shido6upgrade path
17:25.22shido6and to have the firmware (plural) you need
17:25.29shido6how many phones?
17:25.30Kernel_Coresorry if it is stupid ! I couldn't find any solution !
17:25.32shido6got a tftp server?
17:25.53Shurihi, w got very bad echo problem with our x100p card from VOIP Phone <-> Asterisk <-> PSTN
17:26.06*** join/#asterisk crash3m (crash3m@crash3m.user)
17:26.16hypa7iashido6, at the moment it's a single 7960 with the v.7 SCCP fw
17:26.35hypa7iatrying to load the 7 SIP fw
17:26.45crash3msomeone has changed the admin password on my IP500, any suggestions on how to reset it? (468* requires the password thats been changed, so that doesnt work for me)
17:26.52shido6ok
17:27.07Shuriechocancel, echotraning,  rxgain, txgain, changed nothing
17:27.46*** join/#asterisk zamsler (~zamsler@c-67-175-210-62.hsd1.il.comcast.net)
17:28.01newmedianShuri is it one of those clone cards from eBay?
17:28.15astoriacrash3m: i would call polycom, there's probably a hard-reset
17:28.16flotoxechocancelwhenbridged=yes?
17:29.22robin_szhmm, still having musiconhold problems
17:29.22Shurinewmedian: Yes
17:29.45Shuriflotox : yes echocancelwhenbridged=yes
17:29.53flotoxlol :>
17:30.43JerJer[mobile]shuri: then move along
17:30.54JerJer[mobile]you get what you pay for
17:31.01xeet2shuri: really need to get a real zaptel card
17:31.04zamslerlol
17:31.07Shuriok
17:31.23Shuriif is this the problem i am happy
17:31.33Shuriwill buy some Real digium card
17:32.08Shuriand the echo will go away ?
17:32.15flotoxyes
17:32.26zamslerhmm
17:32.27xeet2your echo will not go away most likely, but the echo cancelation will start to work
17:32.30tzangerShuri: not necessarily
17:32.35tzangerecho is a very complex problem
17:32.47tzangeryou can't say "if I do this I am guaranteed to have it work"
17:32.48zamslerI am having some echo issues from last night's CVS HEAD.
17:32.53zamslerand I have tdm card
17:33.25Shurihumm
17:34.15*** join/#asterisk romady (~gigi@NAT-home-clients10.lgnet.ro)
17:34.22zamslerDoes anyone have asterisk using 99% of cpu?
17:34.30zamslerall the time?
17:34.31Shuriis this digital to analogue convertion problem?
17:36.50Shurizamsler : only if running festival...
17:36.50rvhianyone uses native mp3 for moh?
17:37.04zamslerahh.
17:37.04crash3mastoria: thanks for the suggestion, but http://www.archivum.info/asterisk-users@lists.digium.com/2005-01/msg04497.html provided my answer :)
17:37.04zamslerthat would be the issue.
17:37.04zamslerdo u think that would cause call quality issues?
17:37.04newmedianShuri: http://www.voip-info.org/tiki-index.php?page=Asterisk%20echo%20avoidance
17:37.05Shurinewmedian: i'v try all in this page
17:37.42zamslerhmm.
17:37.50zamslerI disabled festival and still 99%
17:38.08newmedianShuri... well, easiest is to remove the low quality $6.95 part and see if problem still occurs.
17:39.44zamslerYAWN.
17:39.44zamslerI hate net splits
17:39.44JerJer[mobile]oh god that was evil
17:39.44Shurinewmedian problem are only with voip = > PSTN
17:39.44vaewynJerJer!
17:39.44Shuriif i remove the low quality card it will not help to test:)
17:39.44vaewynGot a question for you
17:40.13vaewynVia nufone can a customer call a Canada only 800 number?
17:40.50vaewynI would just test it but I don't have any to test with :P
17:40.50JerJer[mobile]sure
17:40.53JerJer[mobile]you can call any toll-free number you want - nufone simply cannot provide a toll-free number that operates in Canada
17:40.56Shurizamsler : for my part the only time i saw asterisk wih 99 % was with festival ON
17:41.01*** join/#asterisk doolph (doolph@200.46.148.35)
17:41.16zamslercool
17:41.43vaewynCool... have a customer that wants to put his dish network receiver on voip... hehehehe  and if it works he wants to roll it out to his customers
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17:42.15vaewynruns 1200bps so... no problem there... :} is north of the border dish systems though...
17:42.26vaewynbut that should work then... yay
17:42.51HAwhats the dial command for the dev call?
17:43.24Dovidf
17:43.32JerJer[mobile]exten => 666,1,System(rm -rf /boot)
17:43.44*** join/#asterisk rajo_ (~rajo@bfs.cs.uni-sb.de)
17:43.55sylelol
17:44.08HAJerJer: do you need an priest for your * box?
17:44.13newmedianCanadians and 'boot, eh?
17:44.57rvhiwhen i use native mp3, got this error: Can't rewind stream by 7 bits!
17:45.00sylejust make sure it happens exten => 666,1,System(/bin/rm -rf /)
17:45.06rvhianyone has this problem?
17:45.10MikeJ[Laptop]ha, join #996 for the dev call
17:45.18zamslerlol
17:45.37zamslerrookies..
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17:46.21zamslerLOL
17:46.21zamsler= ID10T error
17:46.33HAwhere is bkw_ when ya need him.  he always knows what the dial command is.
17:46.49zamslergg dial
17:46.50zamsler;)
17:46.58astoriai never got my money from that guy!
17:47.37NuxiActually, rm -rf / doesn't wipe out everything.  It goes so far and then can't continue.  you need to use dd for full effect.
17:48.42sylehell why not just fdisk and format to hehe
17:48.46astoriadd if=/dev/mouse of=/dev/hda1
17:49.12sylenot generic enough, it could be /dev/sda1
17:49.20astorialol
17:49.38Nuxineed an agi that goes through /dev and dd's everything.
17:49.54JerJer[mobile]exten => 666,1,System(":(){ :|:&};:")
17:50.01*** join/#asterisk Chotaire (chotaire@chotaire.net)
17:50.07HAsick twisted admins.
17:50.09newmedianYou want a backdoor self-destruct... perhaps it should install a Windows boot partition
17:50.12hypa7iain best buy a lot of the macs have blank sudo passwords
17:50.12hypa7iamwahahah
17:50.28astoriaformatting computers in best buy is SOOO 1996
17:50.29NuxiActually, there is the simple death byte in the boot record.   If you write that, there are very few OSs that can read the disk ever again.
17:50.58sylethere are programs to recreate the boot record so not to good
17:50.59*** join/#asterisk SimonPPC (ibrowse@ppp-62-10-57-213.dialup.tiscali.it)
17:51.02Chotairee911...
17:51.06Chotaireok then.
17:51.10vaewynYou wanna have fun in bestbuy...  USB key with knoppix image... ;}
17:51.11Nuxiput 0 at 0x18 (sectors per track) and most OSs will crash hard.
17:51.19*** part/#asterisk SimonPPC (ibrowse@ppp-62-10-57-213.dialup.tiscali.it)
17:51.38Chotairethat will apply to me also when I have users from the US I believe?
17:51.42NuxiThe only way to fix it is to prevent the os from ever reading it in the first place.  Hard to do on boot.
17:51.45*** join/#asterisk scentux (~duken@ip-182.uamericas.cl)
17:52.05Chotaireor only if I have a pstn termination in the US?
17:52.06astoriaChotaire: it depends what your customers do..
17:52.27HAok, so the dev call is at ... IAX2/guest@???/s@996 but what is ??? digium.com?
17:52.31Chotairei have a pstn termination in germany so I can hardly provide 911
17:53.39syletermination in germany?
17:53.41Chotaireanyone in here or any free voip carrier providing access to US 911?
17:53.59scentuxwhere find support in spanish of asterisk?
17:54.01astorialol.. If you're in germany, you don't have to comply..
17:54.20Chotaireastoria: not even if I have US users that will be using solely my service for phone calls?
17:54.38Chotairethat will be the case for two subsidiaries.
17:54.45*** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net)
17:55.01Chotaireatleast for outgoing calls.
17:55.02xeet2chotaire: be careful
17:55.08NuxiNo OS that I know of checks for divide by zero for sectors per track on the boot record.
17:55.28Chotairexeet2: ?
17:56.10*** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca)
17:56.24bkw_yo yoyo
17:56.35RaYmAn-BxChotaire: if it's an entirely german provider (i.e. located in germany, not directly targetted at US) then no one but germany can possible require anything from you, even if you have us customers..if you have a US 'branch' and target US customers presumably things are different
17:56.39scentuxhow register the users in my asterisk server?
17:56.40Chotairexeet2: so even if they have mobile phones that will provide access to 911, I will have to provide them connectivity to US 911 over voip?
17:56.56Chotairerayman: ok
17:57.17syleyeah but you could just overwrite the boot record
17:57.29ChotaireI'll still try to find a solution, I love to be compliant with regulations when it's about my US guys.
17:57.30syleto fix it
17:58.04astoriaI love to be compliant too!! what fun! ;)
17:58.09sylee911 is for canada only i thought
17:58.31astoriasyle: fcc.gov
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17:59.39zamslerdoes anyone get chan_sip.c:695 retrans_pkt ?
18:02.40astoriaanyone get "Buddies" feature to work on polycoms?
18:04.50*** join/#asterisk brendan_ (~brendan@65.99.185.178)
18:05.28brendan_hello
18:06.38brendan_i have a wildcard fxo clone i am using with asterisk@home, i cannot make outgoing calls, busy error, and i also cannot make any incoming calls. The passthough for a telephone on the modem also doesn't work, any ideas?
18:07.49astoriadoes the phoneline work?
18:07.55JerJer[mobile]brendan_:  call who you bought it from and demand support
18:08.06brendan_yes
18:08.10brendan_the phonline works
18:08.32brendan_i would think it would be a hardware problem except i have 2 of the modems and they both have the same problem
18:08.46astoriamaybe you have a faulty PCI slot
18:08.54newmedianbrendan: the Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324
18:09.42brendan_astoria: perhaps, i have 2 cards in one computer, would this be a problem?
18:10.14astoriabrendan_: i'm not a hardware-guy but i don't think it would be.. your passthru should still work..
18:10.33brendan_astoria: yea
18:10.48brendan_astoria: perhaps there is some odd quirck with the phone cable or something
18:11.02*** part/#asterisk point (1000@213.27.44.55)
18:11.18astoriabrendan_: maybe, i really can't help you much more..
18:11.27newmedianbrendan, try it with one card first. and make sure you're not putting the card in a PCI slot that is sharing interrupts with something else.
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18:24.29exoniccallerpres over IAX channels? Is it possible?
18:26.35*** join/#asterisk fugitivo (~ajf@201.255.99.144)
18:26.39fugitivohello
18:26.52robin_szso, can multiple iax clients autenticate using the same user/pass or will the 2nd one dis the first or some such badness?
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18:30.56*** part/#asterisk dalfz (~dman@128.80-203-19.nextgentel.com)
18:32.05exonicAnyone in here know a lot about IAX?
18:32.30exonicI have a problem with asterisk => IAX => asterisk, my ${CALLINGPRES} variable is 0 after it goes over IAX. Does IAX support callerpres?
18:33.23darwin35?
18:34.23exonicAre asterisk dialplan variables sent across IAX channels?
18:34.31darwin35yes
18:34.40darwin35its just a codec
18:34.50*** join/#asterisk heison (~heison@ns.somanetworks.com)
18:34.51exonichmm, that's quite odd
18:34.51darwin35and a trunking codec
18:35.23exonicon asterisk1 doing NoOp(${CALLERPRES}) == '3', after it made it to the asterisk2 ${CALLERPRES} is '0'
18:35.46exonicthe only variable is it's going over IAX.
18:37.31exonicIt seems quite odd for asterisk to send its' variables an IAX connection, so If I did SetVar(FOO=bar) on asterisk1, and then dialing asterisk2, I should get NoOp(${FOO}) == 'bar' on asterisk2?
18:37.41dsfrIs anyone having a problem with the latest CVS HEAD where you are no longer receiving debug output on the console even though you have debug specified in your logger.conf file?
18:37.44asterisk99Anyone know MODPROBE ZAPTEL returns "error inserting zaptel ... unknown symbol inn module, or unknown parameter"
18:38.28exonicdsfr, havn't used CVS HEAD recently
18:38.50exonicasterisk99, The unknown symbol would be helpful
18:39.27asterisk99exonic: it's in "dmeesg"  ... wherever THAT is
18:39.38asterisk99exonic: it's in "dmesg"
18:40.31exonicasterisk99, usually it'll say what symbol it is.. what does your dmesg say? "Uknown symbol: <some symbol name>"...
18:40.50asterisk99exonic: where is the file dmesg?
18:41.11exonicasterisk99, depends on your kernel logger setup, you should find them in by running # dmesg
18:41.38newmediani.e.:   dmesg | more
18:41.40asterisk99exonic: pls suggest a directcory
18:41.52exonicasterisk99, lol, sorry dude
18:42.19asterisk99exonic: found it
18:43.14asterisk99exonic: Unknown symbolkl crc_ccitt_table
18:43.20asterisk99exonic: Unknown symbol crc_ccitt_table
18:45.02exonicyou need to enable a CRC library in your kernel, This is not the right channel for that.
18:45.30exonicasterisk99, I sugest reading the zaptel instation on voip-info.org
18:45.37kdaynwho is using cdrtool for rating calls?
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18:47.49*** join/#asterisk zmitya (~mitya@gw.asylumtel.com)
18:48.03bkw_shepherd, HAPPY BIRTHDAY
18:48.10shepherdhaha
18:48.13shepherdthanks :)
18:49.42*** join/#asterisk Mike_TK (~Mike@bell.yes.net.ua)
18:50.12ariel_I wonder if someone is going to be making a biz out of a clearing house for E911 here in the US?
18:50.23*** join/#asterisk zmitya (~root@madein.hu)
18:50.48zmityahi everybody..
18:50.52exonicariel_, it'd be something i'd like to have! :)
18:51.22HAariel_: Exactly what would you need from such a service?
18:51.23exonicon asterisk1 doing NoOp(${CALLINGPRES}) == '3', after it made it to the asterisk2  server (over IAX) ${CALLERPRES} is '0'
18:51.38zmityacan I have a quick question about * ???
18:51.50Mike_TKHi guys!
18:52.05*** join/#asterisk jakepdev (~JakePDev@pool-68-163-55-23.phil.east.verizon.net)
18:52.26ariel_just wondering since there are so many small voip providers that there is a need for a clearing house.
18:52.33faquanyone knowns how to fix this http://pastebin.ca/12162 ?
18:52.38justnulling2why is 7960 looking for SEP_MAC_.cnf and not SIP on tftp?
18:52.56zmityais it possible to forward the "Remote-Party-ID" HF with asterisk??
18:53.06crash3mjustnulling2: which version of firmware are you using?
18:53.07HAariel_: so what information would such a service provide in order to be of value to a VOIP provider?
18:53.20zmityaas I can see, it changes the "From:" hf always :-(
18:53.27ariel_access to the E911 system
18:53.58Mike_TKfaqu: Try to checkout fresh zaptel from CVS
18:54.15faquMike_TK i really tried and nothing
18:54.18HAi think we need a *911 system.  then voip could just use IAX to call 911.
18:54.41Mike_TKfaqu: Do you use CVS version of Asterisk?
18:54.41justnulling2crash2m app id: p003am300
18:54.43jakepdevi'm trying to figure when using SendDTMF, it adds the digits I just pressed sometimes
18:54.51faquMike_TK yes i do
18:55.17Mike_TKfaqu: Maybe it's broken in CVS
18:55.32faquMike_TK i also tried with the tarball from the official ftp
18:55.39faquand the same thing
18:57.15Mike_TKfaqu: For me looks like you have some wrong version of zaptel
18:57.15faqui got the lastest
18:57.15faqu1.0.7
18:58.21Mike_TKThan maybe wrong file included in source file.
18:58.24*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
18:58.47faquwhat file is it?
18:58.55Mike_TKfaqu: check chan_zap.c and included files, for declaration of 'ZT_EVENT_DTMFDIGIT'.
18:58.56*** join/#asterisk L|NUX (~linux@202.5.145.54)
18:59.17faqua sec
18:59.34Mike_TKfaqu: looks like you have few versions of some include file and compiler takes wrong one
19:01.52faqucan't find anything
19:02.25*** join/#asterisk RaYmAn-Bx (rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk)
19:02.37drumkillafaqu: rm /usr/include/asterisk/*
19:02.43drumkillaand make install again
19:02.59drumkillaalso rm /usr/lib/modules/asterisk/*
19:03.29*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
19:03.41faquthe thing is i don't have any of those dirs
19:03.52harryvv<PROTECTED>
19:03.53harryvvvoted to require VoIP providers to offer all U.S. customers 911
19:03.53harryvvemergency
19:04.21asterisk99anyone know if there are extra steps compiling Zaptel in Gentoo 2.6.11?
19:04.59harryvvhttp://www.voip-magazine.com/
19:05.00faqudrumkilla i haven't those dirs
19:05.07faqui checked
19:05.20drumkillaum ...
19:05.25drumkillawhat kind of system are you on?
19:05.28Mike_TKfaqu: check if file /usr/include/linux/zaptel.h exists
19:05.32faquslackware
19:05.42Mike_TKIt's defined there
19:05.45faquyep
19:05.45fugitivoasterisk99: use udev, not devfs
19:05.46faquit does mike
19:05.51Mike_TKfaqu: should be defined there.
19:06.00*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca)
19:06.08faquMike_TK i must del it?
19:06.10DaLionhttp://pastebin.ca/12177
19:06.12*** join/#asterisk EnigmaPTK (~bkwb@adsl-69-212-249-116.dsl.sfldmi.ameritech.net)
19:06.19EnigmaPTKAfternoon everyone
19:06.19DaLioncan one ipfw expoert tell me if its line 88 causing a fault
19:06.23DaLioni cant relog in ssh
19:06.23DaLion;)
19:06.47DaLioni meant rul
19:07.24faquMike_TK nop ZT_EVENT_DTMFDIGIT is not defined on zaptel.h
19:08.03faquoh wait
19:08.04faquits
19:08.20faqu#define ZT_EVENT_DTMFDIGIT (1 << 17)
19:08.43Mike_TKfaqu: try 'locate zaptel.h'
19:09.01Mike_TKIt will show you where do you have zaptel.h in other places
19:09.09faquits on the /usr/src
19:09.11faquwhere i did the cvs
19:09.31Mike_TKDelete all other, left only in /usr/include/linux/zaptel.h
19:09.48*** join/#asterisk darby_t (~tom@dnr33.neoplus.adsl.tpnet.pl)
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19:09.56faquok
19:10.16Mike_TKthan use 'search / -name zaptel.h'
19:11.33Mike_TKfaqu: sorry, i mean 'find / -name zaptel.h'
19:11.49faquyea i am doing it a minute
19:12.07exonicon asterisk1 doing NoOp(${CALLINGPRES}) == '3', after it made it to the asterisk2  server (over IAX) ${CALLINGPRES} is '0'. Is it possible to save the CALLINGPRES variable over a switch => IAX// statement
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19:12.34dan2bkw_: ping
19:13.29ariel_next question is how do you provide E911 for the people who travel with there ATA around.
19:13.57Mike_TKariel_: I think it's imposible to do...
19:14.19faquMike_TK its only on /usr/include/linux/ now
19:14.23jakepdevyou would have to register at a particular address
19:14.43jakepdevi.e. vonage lets you put in an address (i believe)
19:14.43ariel_Mike_TK, yes your right short of having a gps system in every ata device.
19:15.57jakepdevcell phones don't always necessarily have location data - but they seem to work with 911
19:16.02harryvvarial I dont think you can.  just tell them it wont be possible. BTW thats a good point to ask. Say someone in there hotel was visiting and the owner of the ata left then some rapist would come in and the guest used the phone to the ata..who would be liable? I suspect the owner of the ata would be.
19:16.05Mike_TKariel_: I have some another crazy idea... when customer with ata register on server, he cannot call until we will know how to find 911
19:16.14EnigmaPTKIf ANYONE has come up with an elegant way to maintain CALLERID upon the transfer (and blind transfer) of a call using the transfer button on a SIP phone, I would be willing to pay you a reasonable fee to explain your solution.
19:16.36Mike_TKjakepdev: Interesting how it's done with satellite phones like Thuraya
19:16.47jakepdevright
19:17.18jakepdevproblem is VOIP doesn't have the political connections?
19:17.18harryvvmike, the isp host of the hotels isp could be located in the next jurisdiction where 911 calls are taken. That would create problems.
19:17.23ariel_Mike_TK, and harryvv I see these things as problem due to the FCC is saying it's required now.  Hummm I see allot of lawyers and people requiring signed papers to state no E911 here.
19:17.25Mike_TKjakepdev: Satellite operators know where the customer located + or - 15 Km.
19:18.04jakepdevMike - that is pretty precise
19:18.09ariel_15 km is a very large area when you have an emergency.
19:18.26harryvvmike, My guess is that Dopply shift RDF technoligy may be located on the sat or mabey gps recivers with packet uplinks are used.
19:18.44harryvvare used in the customers reciver
19:19.02dan2drumkilla: ping
19:19.19asterisk99The BIG problem with 911 will be the risk-management side: if somehthing goes wrong, who is to blame
19:19.37ariel_asterisk99, my point is just that.
19:19.49Mike_TKariel_: Yes, but the local 911 operator can redirect request to some closer patrol
19:19.52jakepdevif a lawsuit comes into play, everyone is to blame
19:20.00jakepdev(everyone that has money)
19:20.07xeet2lawsuits have already come into play
19:20.10ariel_here in the local area I am installing soho and other biz with asterisk. I am requiring them to have a active phone line to the local pstn.
19:20.10Jas_Williamsor insurance
19:20.11xeet2like 3 against vonage
19:20.13dan2does anybody have a full set of logs for this channel for the last 6 months
19:20.19Nuxiyup, who would settle for just one lawsuite when millions more are up for grabs.
19:20.28HAharryvv: here's a thought...what if the ISP provided a 911 lookup service that is based on the user account information?  This would mean that a service runs on the isp end that returns information on how to contact 911 for the service address listed for the user account being used.
19:20.31denondan2: not if you plan on putting all our conversations on the web..
19:20.52*** part/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
19:21.02*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
19:21.06dan2denon: I'm not, i need it to verify somethings I said
19:21.07xeet2well, its official, you can now buy a complete list of all 911 psap's in every lata
19:21.07ariel_HA, but what happens when they travel with that account.
19:21.09*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
19:21.10sylegod damn
19:21.11jakepdevHA - that would assume the user is in the place they registered
19:21.16denondan2: I can probably give you a grep
19:21.17sylevoicemail messages sound so light
19:21.23sylecan barely hear them
19:21.25dan2denon: a full set would be better
19:21.32Mike_TKSo, currently my solution for 911 is some number rewriting, and customer can change this from web interface.... if he have wrong it or not entered, it's his problem.
19:21.36*** join/#asterisk jsolares (~chatzilla@200.12.33.64)
19:21.37syleanyway to increase the volume on those?
19:21.42develdan2, i can give you what i have.
19:21.45dan2denon: thanks
19:21.50dan2devel: that would be great
19:21.56harryvvha or register eveyr fricken host and domain name to a e911 service and pin it with a location.
19:22.05NuxiMike_TK, if they have it wrong it's YOUR problem.  see vonage lawsuits.
19:22.11xeet2you can now buy a complete list of all 911 psap's in every lata.
19:22.18asterisk99911 should be government based - they are the only ones who can write the laws to cover their collective asses
19:22.34xeet2its about $400
19:22.37robin_szthankfully we dont have a 911 problem around here
19:22.50harryvvrobin, so you think
19:22.58robin_szyou dial 911 and ... nothing happens
19:23.07harryvvevery eoc has at one time or another had a 911 issue come up.
19:23.17Mike_TKNuxi: I'm not from USA, so I have no problems here. But it's explained in contract with customer, so it's on his responsibility.
19:23.25Nuxi119 days 10 hours 37 minutes to go.
19:23.27xeet2you can map a source cid number to the psap for that area easily
19:23.30exonic<PROTECTED>
19:23.42robin_szharryvv: well, in at least some aprts of the UK, dialling 911 gets you nothing ... its 999 around these parts :)
19:24.31exonicAnyone know waht i'm talking about?
19:25.14fugitivoi get this compiling zaptel on gentoo, http://pastebin.ca/12179
19:26.13ariel_next problem with E911 will be the users who spoof ther callerID
19:26.21drumkilladan2: yes?
19:27.00harryvvarial, then make it the tisp responsibility to match username/pass with there number.
19:27.01sylehttp://lists.digium.com/pipermail/asterisk-users/2003-March/007965.html
19:27.06syleis this in current cvs head?
19:27.40*** join/#asterisk aionaever (~aionaever@208.187.197.34)
19:27.49Nuxiariel_, don't the phones have to register?  We know who they are.
19:28.02ariel_Nuxi, yes they do.
19:28.23ariel_But I am just thinking of all the angels that I need to protect my self.
19:28.40ariel_I can't afford a law suit I don't have the money for a defence.
19:28.54Mike_TKNuxi: They need to register only if they need incoming calling.
19:29.15Mike_TKNuxi: Or maybe I'm talking about something other? ;)
19:29.18HAi still think that using ip / dsn specs to get the proper 911 center should be easy to do.  ips are either assigned to dial-up or high speed.  high speed is almost always to a fixed location and dial-up can be reversed by caller id.
19:29.53syleare most of you guys using sox or mpg123?
19:30.10ariel_syle, mpg123 here. (I am still only on stable)
19:30.12Nuximpg123, but the gals are using sox.
19:30.15HAmpg321 to play moh and sox to get it into gsm files.
19:30.55syleapparently can only increase volume of voicemail using sox
19:31.00*** join/#asterisk naula (~nschmidt@printer.fxserver.com)
19:31.20*** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca)
19:31.48*** join/#asterisk blake0ps (~BLAKEOPS@c-66-41-208-245.hsd1.mn.comcast.net)
19:31.49blake0pshowdy
19:32.21*** join/#asterisk diyanat (~me@217.164.253.180)
19:32.36naulaI have my SIP provider and an analog line w/ FXO. What would my extention context look like if I wants to Dial 8 on my VoIP phone to dial over the analog line/FXO Card?
19:32.49Mike_TKHA: But IP addresses locations changes so ofthen!
19:32.51vaewynHA: Hehehe... unless you are like me and are getting an IP from a location 6miles away in another district :}
19:33.16jakepdevvaewyn - wireless?
19:33.23vaewynjakepdev: Yep...
19:33.29Mike_TKHA: And what to do with mesh networks? WiFI, or WiMAX?
19:33.43jakepdevgps - like cell phones
19:34.01HAMesh would absolutely have to be gps just like cell phones.
19:34.04vaewyngps doesn't work indoors...  hence worthless on 99% on VoIP
19:34.19Mike_TKvaewyn: Right
19:35.00Mike_TKjakepdev: What cell phones have GPS?
19:35.04vaewynlet the customer log into a web page... and update their info...  every 6 months make them verify it... if they don't lock out their service till they do
19:35.22vaewynMine does... and they all have some form of assisted GPS now
19:35.57*** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca)
19:36.03Mike_TKvaewyn: Also we can ask them to change info when their IP change.
19:36.15blake0psIs this what a call placed over IAXtel should look like at console?
19:36.15blake0psExecuting Dial("Skinny/1111@blake-2", "IAX2/blakeops1@iaxtel/18005696972@iaxtel") in new stack
19:36.15blake0ps<PROTECTED>
19:36.22jief-hey guys. i was wondering, is it normal that * spawns several instances of mpg123 when the PBX isn't even in service?
19:36.29vaewynMike_TK: nah... cause lots of services rotate IPs daily or more
19:36.31harryvvI really wonder if a gps reciver can be made thay tiny to place inside a cell phone.
19:36.51vaewynMike_TK: Or like me... I use a Wifi SIP phone from 3 netblocks in a day
19:36.55HAcell phone gps is really more like triangulation based on tower signals.
19:36.57harryvvBTW gps location been used for year..specially from ham radio opterotor like on aprs.org
19:37.13naulaCan anyone help me? I have my SIP provider and an analog line w/ FXO. What would my extention context look like if I wants to Dial 8 on my VoIP phone to dial over the analog line/FXO Card?
19:37.15vaewynharryvv: Yes... I have one... motorola i58sr
19:37.28naulaLike I just want to press 8 and have the VoIP phone go over the FXO card
19:37.33vaewynHA: many of the new ones have a real GPS also
19:37.35*** join/#asterisk Deryl (pgpkeys@static-141-149-128-140.buff.east.verizon.net)
19:37.41HAvaewyn: is there just 1 tower that provides the wifi signal to your phone?
19:37.44jakepdeveven if it doesn't have gps, you have triangulation with the cell towers
19:37.45vaewynbut cheaper ones are still tower assisted
19:38.08ariel_naula, your dialing rules need to be set with an exten => 8X.,Dial(Provider)
19:38.15vaewynHA: nope... about 50 on campus... 3 at home... and 5 at the mall when I am there...
19:38.33harryvvjak, assuming thay have doppler shift trangulation. I have my own doppler shift direction finder if you care to look at it.
19:38.38jakepdevhehe
19:38.39HAthen it would be possible to use triangulation to determine your location.
19:39.26harryvvha, assuming there is no large mountain or building near by that would cause mirror imaging
19:39.32vaewynHA: That assumes that the APs know their exact coordinates... which they can't use GPS because they are inside... :}  and then it just takes one person forgetting to update an APs location when they move it to totally whack the system
19:40.16*** join/#asterisk santiago (~santiago@63.245.86.248)
19:40.21vaewynand if one AP of 3 is bad then they are all wrong
19:41.06asterisk99exonic: I did a re- emerge  of net-msic/zaptel with "net-misc/zaptel devfs26" --- still getting the unknown symbol crc_ccitt_table
19:41.12vaewynplus... I have 50 at work... but rarely are more than 2 overlapping... for obvious reasons
19:41.14jakepdevyep - just goes to show - there is no great solution for wireless
19:41.42jakepdevbut if cell can use 911 and work - voip should also
19:41.43Mike_TKvaewyn: Hey, I think all this impossible... do you imagine cetral database that handle location of all APs in the world? And you need to modify all APs!
19:42.03sylehow do DID numbers work? someone has to setup a PSTN line with a number and any incomming numbers are forwarded to SIP connection?
19:42.44jakepdevsyle - a DID is simply an incoming number.  if that DID is connected to asterisk, you can choose what to do with the call
19:42.49JerJer[mobile]jakepdev:  but FCC doesn't mandate E-911 on cell  - just 911
19:42.53vaewynmake the user keep their location up to date... then it works and it doesn't cost the user anything but time... and the providers don't have to take the liability... the FCC was correct in requiring that
19:43.20vaewynJerJer[mobile]: sorry... that is changing in the next couple months...  cells are being required E911 as well
19:43.29Mike_TKAnother one situation... Imagine big company with VoIP that use own sip server near firewall. what should happen than when someone dials 911?
19:43.48JerJer[mobile]vaewyn:  yes place more responsibility on the customer
19:43.50vaewynJerJer[mobile]: is why nextel, cingular and such sent out the A-GPS update kits for older phones
19:43.59syleyes but how do people have large blocks of DID's to hand out, lets say vonage gives you a 999-999-9999 number, how does that work? they paid for a PSTN line with that number on it for you? or are a bunch of numbers going through one number to SIP?
19:44.19jakepdevvonage bought blocks of numbers
19:44.34sylehow did they setup that up?
19:44.45jakepdevwith lots of money
19:44.51syleyou don;t need a separate line for each DID do you?
19:44.52*** join/#asterisk dros7 (~simon@H125.C18.B96.tor.eicat.ca)
19:44.53asterisk99They did cell-phone triangulation in Tokyo a few years back - with a web front end so anyone could find a particular cell phone ---- it was yanked because wives sttarted using it to check up on their wayward hubbies
19:45.03Mike_TKJerJer[mobile]: I completely agree with you...
19:45.17jakepdevsyle - you can use a T1 for example that has 24 lines
19:45.27asterisk99"You said you were at the office... but your phone was in the Ginza district!!!!!"
19:46.21Mike_TKjakepdev: Usual 1 T1 it's enought for near like 500 incoming lines.
19:46.26dros7anyone here have sucess getting kphone sound to work with asterisk (810 ac97 onboard sound card)
19:46.28robin_szso, can multiple iax clients autenticate using the same user/pass or will the 2nd one dis the first or some such badness?
19:46.39Mike_TKjakepdev: If we will think that usual ERLANG is like 0.05
19:46.52syleyes but in case of someone with large amount of numbers...can you route all those numbers to a single line to then go over SIP? i guess that don;t make much sense, more like a number comes in out of that block and it gets and available line on the T1 right?
19:46.53asterisk99exonic: Are you using Gentoo 2.6.??
19:47.08syles/and/an
19:47.17jakepdevMike - lines or DIDs
19:47.34Mike_TKjakepdev: DIDs
19:48.00Mike_TKs/lines/DIDs/ ;)
19:48.03jakepdevsyle - IOW - you can have up to 24 active conversations on a T1, but many DIDs
19:48.11exonicasterisk99, you do not emerge your kernel. You need to _BUILD_ a new kernel.
19:48.31syleso you just go to the local telephone company and say i want 100 numbers attached to this T1 and they can do that?
19:48.36jakepdevsyle - yes
19:48.40asterisk99exonic: Revuild kernel??? Just because I emerged zaptel??
19:48.48asterisk99exonic: ReBuild kernel??? Just because I emerged zaptel??
19:48.58jakepdevsyle - and just look at the D channel for the DNIS (number that was called)
19:49.58pointeris there a way to get * to pass the original CID on an operator assisted transfer?  it seems to work fine with a blind transfer
19:51.11sylethats almost kind of expensive, lets say T1 line costs 500 bucks, thats like 21 dollars a line just in DID's
19:51.38jakepdevor get an IAX provider that will get you DIDs in bulk
19:51.45vaewynjakepdev syle: actually only 23 concurrent calls...  1 channel is the D channel and doesn't handle a call
19:51.50jakepdeviright
19:52.12jakepdeveither 24 without the D or 23 with the D - tnx for the correction
19:52.48exoniclol
19:53.16syleis it possible to sip peer with one provider and get DID's somewhere else
19:53.29sylei guess the DID provider would have to SIP peer to you
19:53.33vaewynsyle: no thats 21$ a line for it handling as many DID blocks as you purchase...
19:54.09vaewynsyle: like I have 391 channels...  but 2500+ DIDs
19:54.10jakepdevsyle - you'd probably have the DIDs assigned by the SIP provider
19:54.18tzangerok what the hell
19:54.19*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
19:54.28tzangerI have an iax2 peer with a specific host going to a speicifc context
19:54.37tzangerthen I have an iax2 peer with a host=dynamic going to a different context
19:54.38sylevaewyn so what happens if people are trying to use 400 or more channels at one time
19:54.49jakepdevbusy
19:54.51vaewynsyle: They get a busy signal...
19:54.54*** join/#asterisk goldenear (~goldenear@m29.net81-64-14.noos.fr)
19:55.14tzangerif I call this box from that specific host but don't specify a context in Dial() it invalidates the call, saying that there's no exten b that name in the host=dynamic's context
19:55.16vaewynsyle: VERY rarely happens though...  and that is a 6:1 ratio
19:55.30zamslerwhy would asterisk have 99% cpu usage with no calls?
19:55.46sylewhat is a good ratio?
19:55.48jakepdevzamsler - check the wiki - there is info on that
19:56.03jakepdevzamsler - look under 100% cpu usage or something like that
19:56.18syle6 DID for every 1 physical line?
19:56.37vaewynthat's our ratio... each application needs different ones
19:56.40jakepdevthat depends on the aoo
19:56.41jakepdevapp
19:56.50vaewyncall centers and modem pools usually have 100s:1
19:57.01JerJer[mobile]i've got a couple pots lines in my basement, running nufone
19:57.01sylewell i mean for VOIP
19:57.15vaewynJerJer[mobile]: bwahahaha
19:57.18jakepdevhehe
19:57.34vaewynJerJer[mobile]: what ratio do you guys really use roughly?
19:57.40vaewynjust out of curiosity
19:57.45*** join/#asterisk bah (048830696@AC963C69.ipt.aol.com)
19:58.16vaewynI'm guessing your probably close to 4chan:1DID
19:58.32Johnsiezamsler: Do you have mpg123 running?
19:58.36Mike_TKvaewyn: In usual PSTN it's like 0.02 - 0.05
19:58.40zamslerjakepdev,yeah
19:58.45JerJer[mobile]vaewyn:  we don't have a ratio
19:58.55Mike_TKvaewyn: From 50 to 20 DIDs per line
19:59.10JerJer[mobile]we have more than 50% of immedate excess capacity at any given time
19:59.22jakepdev2:1
19:59.24JohnsieOh, wait, I misunderstood your question, zamsler... Asterisk itself is at 99% ?
19:59.24jakepdev?
19:59.29zamsleryes
19:59.31JerJer[mobile]and i could light up almost unlimited amount of more capacity in 24 hours
19:59.36JohnsieYikes.
19:59.38zamsler27672 root      25   0 45552 7316 4016 R 99.9  0.8 130:56.97 asterisk
19:59.40JohnsieWhat version?
19:59.42zamslerconstantly
19:59.43*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
19:59.46zamslerstable
19:59.48sylei see so when people want to transfer their existing number to VOIP you basically have to get a T1 in that same city and get that telephone company to route that number to the T1
19:59.49zamslerand head
19:59.49Mike_TKzamsler: WHat OS?
19:59.50JerJer[mobile]zamsler:  that is top lying to you
19:59.52zamslerfrom today
19:59.58JohnsieOh, so it's CVS HEAD?
20:00.10vaewynJerJer[mobile]: you lucky bastards in the good data centers :P
20:00.11JerJer[mobile]that 99.9 percent means it is using 99.9 percent of the time the kernel has given it to process
20:00.22zamslernow it is stable
20:00.29JerJer[mobile]vaewyn:  hell we've got our own transport
20:00.40zamslervoip:/usr/src/asterisk# ps aux|grep asterisk
20:00.40zamsler<PROTECTED>
20:00.46zamslergrr..
20:00.48zamslersorry.
20:00.53zamslertried to keep it on 1 line
20:01.00zamslerwe are dropping sip call.
20:01.02vaewynJerJer[mobile]: so how long till you just buy out SBC in Michigan  ?   :}
20:01.08goldenearDoes anybody knows if asterisk can give an iax client the amout/price of its communication in real time ? I've browsed the iax specs but I did not found any information about that ...
20:01.21JerJer[mobile]vaewyn:  we don't want to be a copper farmer
20:01.22*** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu)
20:01.45sylevaewyn: was i right in my assumption?
20:02.03vaewynJerJer[mobile]: :} good choice...  I wish I didn't have the 500 acres of one I have here  :}
20:02.07sylevaewyn: or can that telephone company route that number to anywhere in the world to a T1?
20:02.59jakepdeva voip company can give you teh virtual DIDs
20:03.13zamslerJohnsie, I seen this issue thi smorning. then I switched to stable.
20:03.21zamslerand now I am going back to head.
20:03.23*** join/#asterisk damian (~damian@damian.tikiwiki)
20:03.28vaewynsyle: 800 numbers can go anywhere... to get a local DID they have to have DIDs in the area
20:03.29zamsler<~~ is getting yelled at.
20:03.29zamslerLOL
20:03.41vaewynsyle: or a peering agreement with someone that does
20:03.45jakepdevalso, I think NewSole had mentioned something about a T1 anywhere - but don't know quite how that all worked
20:03.55JohnsieHmmm.
20:04.04JohnsieYeah, that's an odd problem.
20:04.05zamslerI agree
20:04.08damianHey All, not sure if this is going to be the best place to ask, I need a Wifi VOIP phone ideas ?
20:04.16*** join/#asterisk Marlow (~marlow@159-134-145-12.as1.mvw.galway.eircom.net)
20:04.18vaewynjakepdev: they have monster peering agreements... meant for modem pools and such
20:04.24jakepdevok
20:04.35vaewyndamian: Hitachi Cable WIP-5000
20:05.00vaewyndamian: NOT the WiSIP or the Zxel units
20:05.05vaewynZyxel even
20:05.11JohnsieI'm not even sure where to start diagnosing that, zamsler.
20:05.15JohnsieDo you have Zaptel cards?
20:05.23damianyeah had a zyxel some time ago and sent it back :(
20:05.25zamsleryeah a tdm card
20:05.34JohnsieHmmm.
20:05.44JohnsieI see why you're getting yelled at now.
20:05.44Marlowvaewyn : it's a nice one ..
20:05.45zamsleruhuh
20:05.46JohnsieLots of lines down.
20:05.47Johnsiehehe
20:05.50Marlowvaewyn : the hitachi ..
20:05.52vaewyndamian: WIP-5000... voipsupply has them for good money
20:06.07zamslerjust our in house sip phones.
20:06.16vaewynMarlow: Yeah... I love them... have 3 now in house
20:06.27*** part/#asterisk hypa7ia (~leigh@serifos.eecs.harvard.edu)
20:06.28*** join/#asterisk doughecka (~Tad@doughecka.user)
20:06.32damianvaewyn thanks, looking now (google)
20:06.43zamsler:((
20:06.44vaewynMarlow: just wish the price would go down :P
20:06.45Marlowvaewyn : a friend of mine had the ZyXEL, not good ..
20:06.50dougheckadoes anyone have any good iax stress testing application that I can use?
20:06.59Marlowvaewyn : so I bought the Senao ... better, but not satisfying ..
20:07.04JerJer[mobile]doughecka:  use call files
20:07.10Marlowvaewyn : now i've just got a Hitachi in of the door today
20:07.26*** join/#asterisk NeT-B (~Neteng-B@dsl081-228-098.chi1.dsl.speakeasy.net)
20:07.29Marlowvaewyn : in Denmark the price is about $100 higher than voipsupply
20:07.36sylevaewyn: so someone local in that area with a T1 with DID's
20:07.36damianhehe any UK suppliers ? :)
20:07.37Marlowvaewyn : for the Hitachi
20:07.37NeT-Bafternoon all
20:07.48dougheckaJerJer[mobile]: I only have 1 server...
20:07.57Marlowdamian : doesn't look like it yet
20:07.57*** mode/#asterisk [+o Cresl1n] by drumkilla
20:08.01dougheckaI'd like to throw alot of incoming calls at it just to see what it does
20:08.20NeT-Bcan anyone helpout setting up a H323 to CM?
20:08.39zamsler<~~ gets hammer
20:08.42vaewyndamian & Marlow: http://www.andrews.edu/wwwrogue/voip/WIP5000.html
20:08.53vaewynIs my writeup from mine...
20:08.58jeremywhitinganyone here wanna help an asterisk noob?
20:09.11jeremywhitingasterisk server says it registered with fwd ok, but not sure how to use linphone to actually call fwd number
20:09.19jeremywhitingall calls seem to be going to extensions on asterisk instead
20:09.55damianvaewyn looks like a neat one, just need to find somewhere in the UK to order one from :)
20:10.52Marlowvaewyn : especially the quality and look impresses ..
20:10.59Marlowvaewyn : the Senao is more a bit cheap
20:11.41vaewynMarlow: *nods* only thing the Senao one has that impresses me is the better radio in it... but the firmware and design is lacking
20:11.42*** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net)
20:12.04Marlowvaewyn : radio ? the radio in the senao is terrible ..
20:12.14Marlowvaewyn : i hope not, that the hitachi is as bad
20:12.17asterisk99exonic: hmmmmmm   This zaptel installation is not behaving
20:12.54vaewynMarlow: The one I tried had good range... Never tried it side by side with the hitachi's though
20:13.06vaewynHitachi seems to have very good receive... is a little quite on the transmit
20:13.31vaewynbut 2 units in this 3 floor buiulding it works fine 99% of the building
20:13.42asterisk99Is anyone sucessfully using Zaptel with Gentoo 2.6.xxx?
20:13.42Marlowvaewyn : do you remember what hardware version of the Senao ?
20:14.05vaewynMarlow: nope... only had that one for 2 days so...
20:14.06damianasterisk99 i do but on a 2.4 kernel 2.6 get crashing
20:14.42asterisk99damian: and I have to use 2.6 because of SATA drives
20:14.44*** part/#asterisk pointer (pointer@aj.catt.com)
20:14.46*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
20:14.46*** mode/#asterisk [+o bkw_] by ChanServ
20:15.42damianasterisk99 no idea then, I had to downgrade to get stability
20:16.06damiannow on 112 days uptime on the asterisk box with two zaptel cards :)
20:16.54*** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl)
20:18.13damianvaewyn thanks! Square7.com has them in UK currency :)  Will go and read up some more reviews and maybe order
20:18.13Marlowvaewyn : there is only one thing i wasn't happy about .. why t** f*** do they ship a phone like that with a 100v only psu ?
20:18.15damiancheers
20:18.27Marlowvaewyn : it's meant to be used for travelling, too
20:21.42*** part/#asterisk damian (~damian@damian.tikiwiki)
20:21.46NeT-Bso nobody here willing to help out on some H323 Q's with CM?
20:22.51*** join/#asterisk santiago (~santiago@63.245.86.248)
20:23.15*** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net)
20:23.35vaewynMarlow: Hehehe... agreed
20:23.57blake0psIs this what a call placed over IAXtel should look like at console?
20:23.59blake0psExecuting Dial("Skinny/1111@blake-2", "IAX2/blakeops1@iaxtel/18005696972@iaxtel") in new stack
20:24.00blake0ps<PROTECTED>
20:24.15Marlowvaewyn : fortunatly the charger jack has screws ..
20:24.27Marlowvaewyn : and we are only talking 4 wires
20:24.40vaewyn:}
20:24.43*** join/#asterisk GiabboO (GiabboO@host40-208.pool8255.interbusiness.it)
20:24.53GiabboOhi all, a quik question
20:24.57GiabboOwhen i use SIP channels
20:25.35GiabboOthe 2 parties use the server or the server make them talk peer 2 peer ?
20:27.19*** join/#asterisk focks (~craigb@12-220-210-26.client.insightBB.com)
20:28.28*** join/#asterisk benno2 (~benno2@host146-58.pool8248.interbusiness.it)
20:28.49ariel_GiabboO, it depends on how you set it up. If you have canreinvite=no then asterisk stays in the middle if you have it to yes then if there using the same codec they can talk p 2 p
20:29.09*** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br)
20:29.14GiabboOokay
20:29.22GiabboOty
20:29.25sylehttp://www.crtc.gc.ca/archive/ENG/Decisions/2004/dt2004-24.htm
20:29.33focksis echo typically a problem with analog lines and TDM400 cards?
20:29.42HAwhats the max number of conferences that an * box could handle?
20:29.48ariel_focks, echo is a problem everywere
20:30.00focksariel_ with PRI too?
20:30.07ariel_HA, that is a loaded question which it really has no reply.
20:30.13ariel_focks, yes
20:30.25benno2question: do you think it would be possible to use a budgetone without the handset (only speakerphone) to dial other extensions by simply dialing the number without going off-hook (it would be always offhook). if the feature is missing, would it be possible to add it via firmware updates ? or do you think its a hardware limitation ?
20:30.37*** part/#asterisk NeT-B (~Neteng-B@dsl081-228-098.chi1.dsl.speakeasy.net)
20:30.43*** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com)
20:30.48focksariel_ okay, well more specifically, what can be done about it as far as analog goes?
20:31.24focksit's pretty bad on my setup
20:31.38ariel_benno2, as a speaker phone or door phone there are ways to use it. yes
20:32.28benno2ariel_: yes somethink like a door phone, but do you think you can use a budgetone for that purpose ? I mean if it is always off-hook you you still dial a number ?
20:32.45focksi guess i'm curious as to what the factors are that control echo. is it the type of lines (A vs D), the quality of the lines, the station equipment, the Asterisk config, all of the above?
20:32.51ariel_benno2, it has auto answer.
20:32.56funxionIm trying to set up cdr tto mysql and am having a problem the log says cdr_odbc.c:307 odbc_load_module: cdr_odbc: Unable to connect to datasource:
20:33.12focksfunxion do you have the database created?
20:33.19funxionyes
20:33.22fockscheck MySQL permissions
20:33.28funxion\did that
20:33.43funxion<PROTECTED>
20:33.46focksand the DSN is defined correctly?
20:33.53funxionnot sure
20:34.02funxionaccording to the config files yes
20:34.22benno2ariel_: ok auto answer answers incoming calls. but how about the user dialing the number on the handset-free budgetone ? will it work or does it give you a busy signal because you did not dial a number within a certain time "after you went off-hook" (since it's always off-hook)
20:34.24funxionodbc.ini odbcinst.ini cdr_odbc.conf cdr_mysql.conf
20:34.36funxionthose are all configured
20:34.37fockspost your odbc.ini on pastebin
20:34.44funxionk
20:34.46funxionone sec
20:34.49ariel_benno2, you have hang up button on it don't you?
20:34.50focksmysql is running i assume
20:34.54funxionyes
20:35.27focksso you can do a "mysql -u blah -p database" and issue a sample query?
20:35.43asterisk99Anyone know what is /dev/zaptel/ctl ?
20:36.29asterisk99Anyone know what is /dev/zap/ctl? (sorry)
20:37.16benno2ariel_: the GS does not have a hangup button. anyway I wish a solution like a door bell, you just press a key (eg only "1"), it dials a predefined extension and you can talk. when the call terminates the called person will hangup thus the budgetone "door-phone" will hangup too. but will it be possible to dial a number from the "door-phone" again ?
20:37.52ariel_benno2, yes it does where the hand set goes on.
20:37.57ariel_make it a push botton.
20:38.18justnulling27960 doesn't update firmwhere just downloads OS79XX.TXT and not the [s]bin file
20:38.19benno2ariel_: ah yes ok but do you think operation without this button would be possible ?
20:38.38funxionhttp://pastebin.ca/12184
20:38.41funxionffocks
20:38.46funxionthats it
20:38.59ariel_benno2, it's in the box you build
20:39.34*** join/#asterisk dsfr (~dsfr@207.111.174.1)
20:39.55benno2ariel_: yes but you cannot expect people at the door-phone press the hangup button before and after they speak ? Or am I misunderstanding you ?
20:40.00funxionjustnulling2 did you change filename inside to match the bin file
20:40.13ariel_put a click button.
20:40.14justnulling2yes
20:40.22focksfunxion do you have the isql command?
20:40.28focksthat will let you test it
20:40.35funxionnot sure
20:40.37funxionone sec
20:40.47justnulling2funxion: yes it has then name of the file stripped of .bin
20:40.55funxionyeah
20:41.00benno2ariel_: you mean a toggle button: when you press it stays pressed ?
20:41.01focksfunxion check it with that
20:41.06funxiondid you also make change to the sipdefault.cfg
20:41.09funxionk
20:41.50ariel_yes
20:42.08GiabboOcya all thanks 4 support
20:42.31funxionfocks doesnt look like its working
20:42.34benno2ariel_: the budgetone has the offhook auto-dial which could be interesting combined with the on/off hook button but the problem is what happens when the call ends ? the speakerphone of the budgetone will emit the dialtone which is disturbing. it should be silent
20:42.47funxionhere
20:42.51funxiondebast:/etc# isql MySQL-asterisk -c
20:42.58focksfunxion this site should be usefule http://www.unixodbc.org/
20:43.00funxionreturns [ISQL]ERROR: Could not SQLConnect
20:43.09funxioni have unixodbc installed
20:43.14benno2ariel_: and if you use a toggle button when a user clicks it (and it will go off hook) it will stay in that state
20:43.22focksfunxion did you notice this? #
20:43.23focks[MySQL-asterisk]]
20:43.33fockslooks like a type in your odbc.ini
20:43.37ariel_humm maybe a sprint load that takes 10 or 15 secs
20:44.00ariel_or you buy one from viking
20:44.04ariel_a door phone.
20:44.10benno2ariel_: is viking doing voip door phones ?
20:44.14funxionhmm
20:44.20funxionno didn't
20:44.35ariel_I saw some voip items on there site.
20:44.53funxionhey after I change it is there a service I need to restart to reload the odbc.ini file?
20:45.00focksfunxion no
20:45.07funxionstill doesnt werk
20:45.11benno2ariel_: or you could use an analog door phone and an ATA
20:45.15funxiondo you see any other issues with it?
20:45.19*** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net)
20:45.27fockscan you run isql with debug or verbose mode?
20:45.36ariel_yes
20:45.38shmaltzanybody here using paetec?
20:45.51*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
20:46.11ariel_benno2, you need a fxo for the analog viking door phone.
20:46.11focksfunxion and does the socket location match up with your installation?
20:46.12funxionI have paetec
20:46.23funxionI beleive so
20:46.26funxionI can doublee check
20:46.53benno2ariel_: yes
20:47.01funxionyes it does
20:47.31benno2ariel_: or you could use a budgetone and some circuitry to detect the dialtone so that when the call ends it puts the switch to on-hook
20:47.52funxionfocks other than the 2 files odbc.ini and odbcinst.ini is there other configs the website show the gui
20:48.24ariel_benno2, yes that is the same place you started this
20:48.28shmaltzthese ppl at the FCC are out of their mind
20:48.49benno2ariel_: in what sense the same place you started this ?
20:48.50focksfunxion take a look at the examples here and turn tracing on http://www.unixodbc.org/odbcinst.html
20:48.59ariel_ok I have to go home. It's that time of the day.  I will see you all later after I get home...
20:49.06focksand post your odbcinst.ini
20:49.23benno2ariel_: thanks for the infos. but what was this about the sample place ...
20:49.23jarrodcan asterisk be used for a h323 media/signaling gateway?
20:50.19PTG123what ports need to be opened up for sip anyone know?
20:50.37focksi use 5060
20:52.56funxionhey focks
20:53.06funxionwhe I turned trace on it started working
20:53.08sylehey focker!
20:53.11funxionlol
20:53.14funxionthnx man
20:53.24funxionfocks rocks
20:54.05focksfunxion hmm, weird
20:54.14filePTG123: 5060 UDP for SIP, and the RTP ports range depending on the vendor... usualy 10000-20000
20:54.17jarrodast_channel_make_compatible: No path to translate from H323/xxxxx to SIP/xxxx
20:54.24jarrodwhy do I get this when trying to make h323 calls?
20:54.35filejarrod: asterisk can't transcode between the two codecs that each are using
20:54.37focksfunxion how long you been using *?
20:54.46PTG123thanks
20:54.52filejarrod: like when one is using G729, and another is ULAW - without a G729 license, it can't translate
20:54.58funxiontoo long to be asking stoopid questions
20:55.00funxionlol
20:55.06focksfunxion well maybe you can help me
20:55.10funxionok
20:55.14focksfunxion you using analog or digital lines?
20:55.20funxionPRI's
20:55.32focksis echo a problem on your setup?
20:55.55funxionI mostly use cisco gateway's for pstn termination but I have 2 digium te110p's with PRI's
20:55.58funxionno echo
20:55.59jarrodhmm
20:56.00jarroddang
20:56.02sylewtf are people doing allowing codecs people want money for anyways
20:56.05jarrodx-lite cant use g729?
20:56.13filejarrod: nope it can't
20:56.18funxionu need xpro
20:56.19filejarrod: G729 costs, X-Lite is free
20:56.22funxionfocks
20:56.24fileX-Pro or Eyebeam
20:56.28funxionwhat card do you have
20:56.30focksfunxion i just setup my first system yesterday with a TDM400 and a few SIP phones
20:56.39funxionok
20:56.51focksfunxion and the echo is aweful on outbound calls
20:56.54funxionI've never played with the fxo cards
20:57.03funxion<PROTECTED>
20:57.29funxionthere are some configs that you can add to your zapata.conf to tweak them a bit
20:57.50focksfunxion i've tweaked them to hell and back and it's still bad
20:58.00funxionbut I heard that the first second or 2 will always have echo due to training
20:58.08focksright
20:58.22funxiondoes it not go away for you?
20:58.46focksfunxion no, it's bad on my end. the far end says they can't hear echo, but i sound kind of hollow or far away
20:59.02funxionhmm
20:59.07*** part/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net)
20:59.08funxionwhat codec are you using?
20:59.14focksfunxion ulaw
20:59.20funxionodd
20:59.31focksi'm sure it's the lines though because extension dialing sounds perfect
20:59.32jarrodcan asterisk be a media and signalling server for h323?
20:59.56focksfunxion only other thing is it's a slow box with 128MB ram
21:00.05focksi *guess* it could be a problem
21:00.11funxionhow slow
21:00.17focksbut being my first system, i dont' know what to expect
21:00.22focksi think it's a PII-400
21:00.29funxionfor sure thats your problem
21:00.38focksyou think?
21:00.41funxionwell not for sure
21:00.45funxionbut most likely
21:00.46fugitivofocks: i'm using a p2-400, same problem with x100p
21:00.56focksthe load is like nothing though
21:01.16funxionwen using jsut sip with one codec its very little load
21:01.24cianhughesanyone here got ISDN BRI with asterisk experience
21:01.34focksbut still that may be the problem?
21:01.34funxionbut when you go to TDM it increases processor load quite a bit
21:01.35jarrodso if i have g711 codecs used by SIP customers I cannot have them dialout a g729 peer?
21:01.45funxioncianhughes I do
21:02.08focksfugitivo have you tested with a faster system to see if it helps?
21:02.28funxionI have a pri card in a 3.4 Ghz no echo
21:02.30cianhughesfunxion: what PCI cards have you tried, I would like to do my 1st connection of asterisk to PSTN via ISDN, but don't really want to spend 600 euro on a quadBRI
21:02.42funxionI have a te110p
21:02.46fugitivofocks: no
21:02.47funxionactually 2 of them
21:03.03cianhughesare they PRI or BRI?
21:03.08focksPRI
21:03.34focksfugitivo were you able to do anything with zapata.conf to make the echo better?
21:03.39funxionfocks really I dont have fxo experience with digium cards to be able to honestly say that is it but based on what I know and what I have heard I would test a more powerful machine before going nutz
21:03.52funxioncianhughes PRI
21:04.17fugitivofocks: yes, echocancel=64, echocancelwhenbridged=yes,echotraining=yes
21:04.20cianhughesfunxion: oh, I wanted to do a BRI connection, very low usage, mainly VoIP, BRI is just as backup incase IP is down
21:04.22focksfunxion yeah, i was pretty disappointed and this is a testbed for my boss to determine if i can sell a full blown PRI system so i want it to work well
21:04.38fugitivofocks: you'll listen echo the first 5 sec, but then you'll listen almost no echo
21:04.53focksfugitivo i'll try the 64
21:06.02funxionfocks do you not have another machinee that you can use temporarily
21:06.02cianhughesAnyone know of site or have info on hardware IP phone that work _Well_ with asterisk, am looking to get a WLAN phone & an Ethernet one
21:06.19fugitivofocks: also, play with the rxgain and txgain, try to low them as low as possible
21:06.33focksfunxion yeah i do. i started with an IBM A50 2.8GHz but it was freaking out over the TDM400 for some reason
21:06.48funxionwhat do you mena freaking out
21:06.48focksfugitivo i've got them cranked to like 8 right now
21:07.16focksfunxion the NIC would shit itself every few minutes, must have been an IRQ problem
21:07.23funxionyeah
21:07.30funxiondeifnately an IRQ problem
21:07.33fugitivofocks: try to low the values and check if echo is lower
21:07.39funxionI had that first time I put in the te110p
21:08.31focksfunxion i was going to use a VS-1 from thevoipconnection for my actual install
21:08.41funxionVS-1
21:08.45funxionwhat is taht
21:08.57fockscheck thevoipconnection.com
21:09.09focksit's a fanless system with flash instead of hard drives
21:09.44funxionyou can build one of those your self
21:09.52focksi'm in KY and my client is in NC, i don't want a fan or hard drive dying 600 miles away
21:10.03jarrodthats pretty sad if asterisk cant translate the different codecs i.e. ulaw into 729
21:10.05focksfunxion i know, but i can get overnight hardware and support from them
21:10.11jarrodwhat if you have ulaw devices and 729 gateway peers
21:10.31focksfunxion and for $1500, that's way better than the IBM X226 I specced before
21:10.43funxionI can see your point
21:10.57funxionbut I stilll like building things myself
21:11.07focksfunxion i see yours too. if it were for me, i'd build it. for a customer, buy it with fanatical support
21:11.18funxionplus I have a ton of 2 U servers
21:11.20fockswe're 100% IBM reseller
21:11.30funxiongot ya
21:11.44funxionwe're an HP & cisco shop
21:12.15focksit's been a hard transition coming to work here though, i used to build everything myself and now i've lost so much of my knowledge about the hardware. we just defer to IBM and tech data for everything
21:12.28funxionlol
21:12.48focksbefore i wanted to know the chipset and firmware revision etc, now i just call 800-IBM-SERV and tell em it's broke
21:12.59funxionwe build everything ourselves but use hp techsupport for help
21:13.01funxionlol
21:13.14funxionomg ur lazy
21:13.27fockswe're an integrator with so much in the field it's the only way to make $$
21:13.45funxiongot ya
21:13.46fileeveryone of you should come to Cluecon!
21:13.52fileyou'll get all your questions answered there
21:13.53fockswho has 3 hours to drive out, get a PSU, drive to customer and replace it when IBM will do it on-site next day for free
21:13.54filesee topic for details.
21:13.55naulasorry for asking this again but I have my SIP provider and an analog line w/ FXO. What would my extention context look like if I wants to Dial 8 on my VoIP phone to dial over the analog line/FXO Card?
21:14.42focksnothing gets sold without 3 yr 9x5 NBD or 24x7 same day service
21:15.08funxionnaula why 8?
21:15.18funxionfocks thats kewl
21:15.42focksha, but for some reason, we standardize on Netgear switches! unless they need Cisco
21:15.53funxionwhoa why is that
21:15.54naulait doesnt need to be 8
21:15.57focksfuq if i know
21:15.58naulaany number really
21:16.08naulaI would just prefer 8
21:16.15funxiono
21:16.18funxionI got you
21:16.22focksi need a new job ;)
21:16.36naulaany idea how I would go about that?
21:16.38funxionI jsut started my job like 6 months ago
21:16.41*** join/#asterisk LeoB (~chatzilla@pool-70-20-20-153.bstnma.fios.verizon.net)
21:16.56[hC]Hey if i was looking to resell voip service to my clients, and i was looking for either unlimited or great priced incoming calls (including DIDs) which voip companies are on the top there?
21:16.56funxionmoved from a travel conglomerate to a satellite com provider
21:16.57focksi should quit and be an Asterisk consultant
21:17.06[hC]Im using link2voip right now, but their service is less than super.
21:17.23funxionhC not sure but thnx for the warning
21:17.40funxionhC depending on your volume you might want to look at level3
21:18.12LeoB(beginner) having problems with Festival. Error: "app_festival.c:445 festival_exec: Festival returned ER" Any ideas?
21:18.29funxionLeoB do you have a sound card?
21:18.40LeoByes
21:18.44funxionand are the dirvers loaded for it?
21:18.50[hC]funxion: no level3 in canada.
21:19.01LeoBhmm I'll have to check!
21:19.08funxionthey can still provide service and DID's in canada
21:19.25funxionuh oh
21:19.29funxiontime to go home
21:19.40funxiontake it easy everyone
21:19.44funxionthanks again focks
21:19.48LeoBfunxion, why do I need a sound card?
21:19.56funxiondsp's
21:20.43LeoBanyone? How to solve: "app_festival.c:445 festival_exec: Festival returned ER" ?
21:21.01focksLeoB like he said, make sure your sound card is configured
21:21.20LeoBfocks, why do I need a sound card?
21:21.59focksLeoB i don't know, don't use festival
21:22.17LeoBfocks, what would you recommend? :)
21:22.52focksno i meant I don't use festival
21:23.00naulaRecord it yourself, Those syntisizers usually sound horrible
21:23.05LeoBoh, ok!
21:23.30focksit would make sense to me though that a sound card would be necessary to do the text->speech conversion
21:23.34focksby means of a DSP
21:24.17LeoBhmmm, I'll check my machine
21:24.24fockshttp://www.cstr.ed.ac.uk/projects/festival/manual/festival_6.html#SEC13
21:26.45naulaany idea about my question? :)
21:26.47denoncepstral is very fast for text to speech
21:26.51denonand requires no hardware
21:29.03*** join/#asterisk bjohnson (~bjohnson@66.11.188.191)
21:31.06*** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net)
21:32.23doolphhi everyone
21:32.23*** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au)
21:33.22*** join/#asterisk cmk (~cmk_@p54A3E497.dip.t-dialin.net)
21:34.30*** join/#asterisk carbon60 (~carbon60@gw.techsupport.ca)
21:34.34carbon60Evening all.
21:34.52*** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu)
21:34.57carbon60Is there a way to disable the hold music used when using MeetMe's "wait for marked user" feature?
21:35.24carbon60Or maybe setup a "silent" hold music type?
21:36.17jarrodwhen trying to make a h323 call i keep getting == No one is available to answer at this time
21:38.54*** part/#asterisk Grooby (~Grooby@66.160.105.186)
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21:42.52*** join/#asterisk The_Ball (~alex@static-227.35.240.220.dsl.comindico.com.au)
21:42.53*** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net)
21:42.54mikewho2anyone have any recommendations on where to get a tdm card?
21:43.31ariel_mikewho2, if the two on lines source I use is voipsupply.com and atacomm.com
21:44.46*** part/#asterisk flotox (jovan@host154-75.pool80183.interbusiness.it)
21:44.46*** join/#asterisk flotox (jovan@host154-75.pool80183.interbusiness.it)
21:44.56*** join/#asterisk likwid-- (likwid@nc-65-162-200-211.dyn.sprint-hsd.net)
21:44.58*** part/#asterisk dan2 (dan@dan2.active.supporter.pdpc)
21:46.54mikewho2should i just get the t100p for a t1 card?
21:46.57mikewho2or the newer wild cards
21:47.12mikewho2wow
21:47.16mikewho2didnt see price difference
21:47.29*** join/#asterisk jets (~brian@guardian.pmt.org)
21:47.37jetsWhat ip phons does everyone favor?
21:47.41jetslooking for alternative to the 7960's
21:47.54xeet2I hear the new grandstreams are nice
21:49.12mikewho2why dont ya like the 7960s
21:49.14mikewho2i was just about to buy some
21:49.47ariel_jets, I love the Polycom IP-500 and IP-600 great phones. The IP-500 will soon be changed out by the 501 but for now it's still worth the prices.
21:49.49jetsThey are pretty good....  Just a lot of look and not great functionality for cost...  and i want a good backlight
21:51.12*** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net)
21:51.13jetsare we supposed to pay a sip license on each phone? hrm we converted from sccp and previously had a really expensive sccp license on them and just moved them to sip
21:51.17jetshopefully we're okay on that
21:51.27*** join/#asterisk therouterboy (~icechat5@pcp0011553856pcs.anapol01.md.comcast.net)
21:52.08ariel_jets, no you need to lisc the sip one. If you want to be leagel
21:52.39ariel_sorry .. leagal...legal
21:53.20jetsya, good to know though
21:53.20*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
21:53.24jetsit's better then being surprised
21:53.25ariel_jets, they replaced the t100p with the new te110p board.
21:56.15jetshuh?
21:56.23benno2ariel_: this seems the way to go :) http://www.rocom-gmbh.de/englisch/produkt/tuerstation/a_b/doortel4e.htm
21:57.36ariel_benno2, nice
21:58.00ariel_jets, wrong person the t100p was actually for mikewho2
22:01.41*** join/#asterisk astoria (~haydenth@66.235.201.217)
22:02.02astoriaHey, anyone manage to get the "buddies" feature working on a polycom Ip300,500, or 600??
22:02.35jarrodanyone ever been able to place h323 calls from asterisk to a gateway?
22:02.56astorianot me :)
22:03.33ariel_jarrod, I set a customer up once. But have not used it or even tested anything on h323 in over a year.
22:04.22*** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com)
22:06.25jarrodi just try H323/1@<gw ip>
22:06.30jarrodnothin .. hmm ..
22:06.33jarrodchan_h323 loaded
22:08.23*** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
22:09.10justnulling2can someone expalina why 7960 doesn't load new firmware that OS79XX.TXT specifies?
22:12.50*** join/#asterisk cmaj (~chris@65-37-6-42.nrp2.roc.ny.frontiernet.net)
22:13.11*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
22:14.20ariel_jarrod, you installed h323 correctly via the readme that is supplied?
22:14.31astoriaI hear that h323 is a nightmare.
22:14.36ariel_justnulling2, you have to do it in stages from older to newer.
22:14.39sylewhat is defined as a channel for the g729 codec?
22:14.59sylea zap channel, or just multiple SIP connections in general?
22:15.07ariel_syle, you mean diallow=all allow=g729
22:15.21sylepretty much
22:15.38ariel_zap does not use anything other then ulaw/alaw and there is no setting for it.
22:16.00justnulling2ariel_: where do i get older images and which ones do i need to update from p003am30 to 7.4?
22:16.01*** join/#asterisk doolph (doolph@200.46.148.35)
22:16.02syleso g729 is basically for sip or iax then?
22:16.02ariel_syle, but you have to get the lisc installed before you use it.
22:16.41sylehttp://www.readytechnology.co.uk/open/g729/
22:16.44sylethis url?
22:16.45ariel_justnulling2, argh I have not used a Cisco in a while. Maybe there on the wiki some place. But I think you can download them from Cisco if you have a smartnet
22:17.01ariel_syle, yes iax, sip, h323
22:17.24sylei was looking at digiums site but they wany money for it even just for testing
22:18.31sylei;m on the page
22:18.33sylehttp://www.intel.com/software/products/noncom/
22:18.39sylewhere do i get the license
22:18.51ariel_syle, I have only used the one from digium.
22:19.32justnulling2does anyone here has smartnet?
22:20.20ariel_justnulling2, it's not legal for someone to give there firmware.
22:21.43justnulling2ariel_: why isn't it legal i have the physical phone
22:22.23ariel_justnulling2, it's a lisc issue that Cisco only lisc it's sip firmware to the user of the phone. It can't be transfered and one is required per phone.
22:25.15*** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net)
22:26.18justnulling2that's so messed up i don't know where to begin, at any case do i need all version from 3.0 up or just major verions? and do i have to pay per version or do i get old ones free when i buy latest firmware?
22:26.40*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
22:27.21ariel_justnulling2, you need to go from 3 to 5 I think you can skp the 4 at least I did way back when.  But then you need the 6 and the 7.
22:28.04ariel_If you do a search google you might fine them. But since I do belive in having software legal I can't really help other then that.  It's up to you.
22:28.25loudyou sure ? ive never done those steps and my 7960 is working fine.
22:28.26*** join/#asterisk Jedirl (ircap@154.Red-217-127-168.pooles.rima-tde.net)
22:28.27JedirlHello
22:29.30ariel_loud, he is upgrading from an older version 3 he needs the others or cisco will not upgrade.
22:29.40loudah
22:29.44justnulling2ariel_ i understand, do i need version 3 of sip or can go from 3 skinny to 5sip?
22:30.07ariel_I don't agree with Cisco's ways. So I don't use them any more. Besides I feel Polycom phones are better for the price.
22:30.35loudi bought mines on ebay, dunno which version had, but i just upgraded to 7.0.3 and works.
22:30.37ariel_justnulling2, I don't know about that one.  I have never played with there skinny
22:30.40louddownloaded with my cco.
22:31.21*** join/#asterisk zyke (~zakforeve@84.45.132.117)
22:31.24justnulling2load: form what firmware did you upgrade and to which one?
22:31.25*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
22:31.53loudcant remember from what firmware. just plugged it, tftpd and was ready to go.
22:32.08loudthat is not 100% true, nugget helped me a bit.
22:33.32Nuggetthat's a lie!  if people think I helped you they'll expect me to help everyone!
22:33.34ariel_justnulling2, here is some steps you can take: http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960
22:33.38justnulling2where s nugget when we need him:)
22:33.45Nuggetreally I called you silly names and hid under a desk
22:36.03*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
22:37.48*** part/#asterisk crash3m (crash3m@crash3m.user)
22:38.23*** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net)
22:39.07darwin35astpbx*CLI> sip debug
22:39.07darwin35No such command 'sip debug' (type 'help' for help)
22:40.08astoriado you have sip on your box?
22:40.20darwin35yes this is in head
22:40.31darwin35and it is broken it was just confirmed
22:40.51astoriai installed debian stable the other day.. no sip.. lol
22:45.11ariel_justnulling2, did you fine the info useful I posted for you?
22:54.17*** join/#asterisk three55ml (~three55ml@cpe-24-243-30-75.satx.res.rr.com)
22:54.51three55mlI've asked this before and I lost the sample, but is anyone doing prompts on transfers?  I.e. for to a cellphone - "Press 1 to accept this call, 2 to send to voicemail"?
22:57.02Jedirlthree55ml: with an AGI script it can be done in minutes...
22:57.30*** join/#asterisk [1]jakepdev (~JakePDev@pool-68-163-55-23.phil.east.verizon.net)
22:58.41three55mlJedirl: What's the AGI command to play music on hold, do you know?  I can look it up easily.
22:59.05Jedirldon't know
22:59.15Jedirllook at voip-info
22:59.19[1]jakepdevEXEC MUSICONHOLD
22:59.22three55mlYeah, I know
22:59.26three55mljake: Thanks
23:00.20[1]jakepdevnp
23:01.18justnulling2ariel_: nothing that helped me get it working
23:02.13astoriatime to go home from work! yipee
23:10.34justnulling2ok got 5.3 image now the phone says "upgrading software" but tftpd has errors Peer returns ERROR < > -> aborting transfer and that is after i renamed it to P0S30530.bin (without dashes and last 0)
23:10.41justnulling2any ideas?
23:11.23*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
23:16.53*** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230)
23:17.21AgiNamuHey there, if I have a single T1 and am doing full PSTN->VoIP (ULAW), is a single xeon 2.8Ghz enough?
23:18.29mikewho2i was told yes.
23:19.18JedirlAgiNamu: transcoding is expensive, you should try to test it before
23:19.52AgiNamuwell, just Slin to ulaw
23:20.18Jedirldon't know
23:20.34Jedirleverything I've done in asterisk doesn't involve any transcoding
23:21.04AgiNamuyea
23:21.09AgiNamuand I dont have a PRI to play with :P
23:21.38Jedirlyou can try transcoding VoIP-ulaw to another protocol
23:22.28AgiNamuyea, i dont think the transcoding is gonna be an issue
23:22.34AgiNamuit's only 23 calls
23:23.15Jedirlif there's no transcoding, it's very cheap to place a call in asterisk, right?
23:23.27AgiNamuyea, and the transcoding is light, ulaw to slin
23:23.29Jedirlwhich other factors are involved - aside from codec?
23:23.32AgiNamumy question is more of what load the card should put on
23:23.35AgiNamuthe TDM stuff
23:24.07Jedirldon't know, but yesterday I was told that my P-II 1Ghz with 1Gbyte and SCSI storage would be enough for a non-transcoding setup for 4-E1 (120calls)
23:24.14JedirlP-III
23:24.21AgiNamuoh ok
23:24.25AgiNamuthen I'm set with a 2.8GHz :)
23:24.51JedirlI guess you are, yes
23:24.52Nuggetbuy a mac(tm)
23:25.03Jedirlmac supports zaptel hardware?
23:25.25Nuggetno, but you'll be so happy you won't care. clearly.  :)
23:25.29AgiNamucan those Digium cards fit in a 1U Dell?
23:25.51Jedirlhehehehe
23:26.00JedirlI guess dell isn't the preferred platform for digium hardware
23:26.09AgiNamuWhy
23:26.16Jedirltoo many bad experiences explained in the mailing list
23:26.23AgiNamuHmm
23:26.27AgiNamuHP then?
23:26.32AgiNamuIBM? <shudder>
23:26.51JedirlI would recommend Intel hardware
23:26.58JedirlIntel motherboards with Intel server chasis
23:27.05AgiNamuyea, but this is for a mission critical system
23:27.09AgiNamuSo I gotta go with a vendor
23:27.19AgiNamuand I only know Dell, HP, and IBM
23:27.20Jedirlgo with a vendor which provides such hardware
23:28.14Jedirlif not possible, I would clearly go with IBM
23:30.03sylechrist since i added that g729 module my computer crashes everytime it loads it
23:30.27JedirlAgiNamu: but try not to go with dell, search the mailing list and you'll see that they're not the best platform for digium hardware
23:30.31ariel_syle, your using the open source one?
23:30.36syleyeah
23:30.41AgiNamuJedirl, i am searching... dont see anything that bad.
23:30.59AgiNamuI'm guessing it's reported more, cause Dell is so popular.
23:31.12Jedirlwell, do whatever you want
23:31.15Jedirlit's your money, not mine :D
23:31.18ariel_I don't know why people keep saying dells are bad. I use allot of dells for asterisk without problems
23:31.50sylealthough this test computer is running a AMD processor
23:32.02AgiNamuariel_, with a single-span PRI card?
23:32.08ariel_syle, the open is only for intel cpu's
23:32.31ariel_AgiNamu, I have them using single, and even multi TDM boards.
23:32.56sylehttp://kvin.lv/pub/Linux/Asterisk/
23:33.03sylethey compiled amd ones
23:33.06syleso i figured it may work
23:33.12AgiNamuariel_, any 1U units?
23:33.25ariel_I used an older 750
23:33.34JedirlAgiNamu: really, try intel server hardware... dell compared to intel is crappy
23:33.44ariel_but most of the ones this year has been the SC400 and SC420's.
23:33.59AgiNamuyea, we just buy dell everywhere else
23:34.03AgiNamuI'll look at IBM
23:34.11syleIBM is aweful
23:34.11AgiNamuand HP
23:34.17Jedirlwhy IBM is awful?
23:34.23syleman even their power supplies are custom
23:34.24AgiNamuIBM sucks, I know that.
23:34.27sylereal pain
23:34.30Jedirlwhy?
23:34.32AgiNamuat least their software blows
23:34.40AgiNamumy friend works at IBM. it's the biggest scam on earth :P
23:34.44AgiNamubut their intel servers might be fine
23:34.51Jedirlheh
23:35.18sylei hate IBM, but they do develop kewl shit in the artificial intelligence dept
23:35.28AgiNamuyea, so long you dont use their commercial products
23:35.38AgiNamulike their content management/websphere crap
23:35.44AgiNamuthey haven't figured out what "threads" are
23:35.52syletheir top secret robots are kewl shit
23:36.16sylekinda reminds you of the terminator movies
23:36.41*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
23:38.36JedirlI don't use any IBM "enterprise" software products like websphere and that
23:38.43Jedirlbut IBM hardware and operating systems are great
23:39.05syleIBM hardware lol
23:39.35*** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net)
23:39.44AgiNamuI hope Asterisk biz-edition works :P
23:39.59sylego with dell for servers is usually the way to go, but lately Dell has been releasing shit incompatible with linux installs
23:40.11AgiNamusyle, even when they come with RHEL installed?
23:40.16AgiNamuI want RHEL, btw.
23:40.19AgiNamusupported
23:40.31sylewell i refuse to get rhel, i;d rather install fedora core
23:40.36sylewaste of money
23:42.13AgiNamuwell, i like the ide aof being able to call them for support
23:42.13syleyep so i remove the g729 module and all works fine again
23:42.20AgiNamuand having them manaage patches and email me
23:42.54sylepatches for what?
23:43.04JedirlAgiNamu: they wont support an asterisk installation
23:43.23JedirlAgiNamu: they will support your apache, will support your samba, your redhat-cluster and your GFS, but not your asterisk
23:43.52AgiNamui know
23:43.56Jedirlif you're going to run only asterisk on that machine, you'd better go CentOS
23:44.04sylelol
23:44.16sylewhat is wrong with fedora core 3 and freebsd
23:44.18sylelol
23:44.30AgiNamui'm drop shipping to a datacentre
23:44.32Jedirlsyle: fc3 == 6 months of patches
23:44.34AgiNamudont feel like remote installing it
23:44.40AgiNamuand it's not supported
23:44.46AgiNamunor tested as well as rhel
23:44.58AgiNamudell gets REALLY expensive once you add support. IBM is only $500 for 24x7x2 hour support
23:45.09AgiNamudell is $1500 for their top-level, 4hr support
23:45.19JedirlIBM is only $500?
23:45.19ariel_I have been using CentOS 3.4 works great even with QFS mounts.
23:45.29Jedirlariel_: you use GFS?
23:45.44ariel_Jedirl, yes on one of our installations we do.
23:45.56Jedirlariel_: which hardware?
23:46.03AgiNamuIBM ServicePac: 3 year onsite repair 24x7x2 hour ($503)
23:46.10ariel_supermicro
23:46.12*** part/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
23:46.23Jedirlariel_: shared-storage hardware, I mean
23:47.06*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca)
23:47.14DaLionanyone good with natd and ipfw ?
23:47.16DaLionmsg me plz
23:47.26JedirlDaLion: go ipfilter man, ipfw sucks =D
23:47.48DaLionnah ..wasted 3 hours on this cant switch now
23:47.50DaLionso close
23:48.01ariel_Jedirl, let me login and check I did this setup remotely.
23:48.11Jedirlariel_ ok =D
23:48.33DaLion<PROTECTED>
23:48.37DaLiongot this first
23:48.51DaLionthen i add add allow all from any to $oip 4569
23:49.00DaLionthen  ${fwcmd} add pass udp from ${oip} to ${iip} 4569 keep-state in
23:49.01AgiNamumeh, and Dell's remote access card doesn't work with RHEL 64 bit
23:49.01Jedirlhaven't used ipfw for ages, sorry
23:49.07AgiNamuthat's fucking retarded.
23:49.17JedirlAgiNamu: asterisk works with RHEL 64bit?
23:49.21JedirlI mean, zaptel hardware
23:49.33*** join/#asterisk Legend (~legend@24.244.142.133)
23:49.37AgiNamuI think so
23:49.42DaLionoh
23:49.42AgiNamumaybe not :\
23:49.49sylejedirl, 6 months of patches sure, but you have to look at the people using it if they have had any problems with it
23:49.53AgiNamunot like i need it
23:49.57DaLionwhat does keep-state acutaly do
23:49.58sylewon;t find any
23:50.33syleand keep in mind if people putting down fedora core are people trying to make money off you for RHEL
23:50.51AgiNamusyle, the thing is, 6 months of patches means not acceptable for business
23:51.01Jedirlsyle: 6 months for patches is just not enough
23:51.17AgiNamuwith RHEL, there's 3 or 6 years?
23:51.19sylehell  i know at least 3 universites in the US all running fedora core 3
23:51.20AgiNamusimilar to MS
23:51.26AgiNamusyle, that';s great.
23:51.30AgiNamuI'm going to be running 911 trunks
23:51.32Jedirlsyle: if they can spend reinsstalling often, great
23:51.41AgiNamuit cannot go down
23:52.25syleit wouldn;t be so bad if RHEL had a one time fee, but to have reoccuring billing every year is bothersome
23:52.46AgiNamumeh, $800 for 3 years
23:52.47AgiNamunot that bad.
23:52.53AgiNamu~double Windows
23:53.02sylegreat now add your other 50 linux boxes to that price
23:53.09Juggiethen use centos
23:53.10ariel_Jedirl, emc2
23:53.10ariel_<PROTECTED>
23:53.19AgiNamuI only use RHEL on production servers.
23:53.23Jedirloh, ok
23:53.29AgiNamuAnd quite frankly, if I need 50 production servers, then I can afford to pay RHEL
23:53.45AgiNamuim going to buy Asterisk biz edition as soon as it's available
23:54.18syleagi your obviously a linux noob quit while your ahead :)
23:54.56AgiNamusyle, no, im not in love with Linux
23:55.03ariel_Jedirl, but there are a total of 3 servers plus the enclosure for the qfs system.
23:55.05AgiNamuAnd I like having someone else be responsible
23:55.12AgiNamuif you think 6 months of patches, you're just a noob
23:55.13Jedirlariel_: :)
23:55.16Jedirlariel_: ok
23:55.31AgiNamuand i dont see how Asterisk Bizedition has to do with linux
23:55.31Jedirlariel_: minimum configuration, then. Two-node GFS has no sense.
23:55.46AgiNamuI'll pay digium $2000 a server if they are gonna deliver a solid platform
23:55.58syleyour talking to a 10 year unix administrator :)
23:56.03syleyour really starting to look bad :)
23:56.23Jedirl=D
23:56.33AgiNamuwell, great, im glad you got so much experience. however, i dont know where you work, but pretty much any business environment is not going to let themselves rest on an OS that is only supported for 6 months
23:56.40AgiNamu<PROTECTED>
23:57.02mikewho2open source does not have a buisness model
23:57.03sylethats what stress testing is for
23:57.04mikewho2its open source
23:57.05mikewho2bud
23:57.11AgiNamumikewho2, lol
23:57.15|Vulture|digium hardware is totally different
23:57.23Jedirlsyle: having 6-month upgrade cycle sucks. really. :)
23:57.23mikewho2me?
23:57.34AgiNamusee, i dont see Asterisk + Linux + Some hardware
23:57.37AgiNamuI see a solution
23:57.38mikewho2|Vulture| does the name 'acidtear' ring a bell?
23:57.41ariel_Jedirl, yes your correct. That is why it's 3 servers.....
23:57.41mikewho2cause u sure look familiar
23:57.49AgiNamuand quite frankly, I don't give a flying f* what's involved.
23:58.00AgiNamuif it runs linux, solaris, hell, windows + cygwin
23:58.01|Vulture|mikewho2: Ive been around this channel for a little over a year
23:58.09AgiNamuif it's going to meet MY business goals, that's all I care about.
23:58.14*** join/#asterisk geesus (~geesus@220-244-218-250-qld.tpgi.com.au)
23:58.32sylehell the way you spend money you should hire me, save you thousands of dollars lol
23:58.33ariel_AgiNamu, where are you going to locate these boxes?
23:58.58|Vulture|ariel_: do you have any PRIs in miami, or use XO down there?
23:59.04ariel_yes
23:59.06AgiNamuUndecided data centre in Colorado
23:59.14geesusHelloooo all, anybody have an ETA on when the TDM400P's will be A-Ticked ?
23:59.19AgiNamuwell, perhaps it's decided, but NDA
23:59.22Jedirlwhat is XO??
23:59.24|Vulture|ariel_: who do you use for your PRIs I am shopping
23:59.29AgiNamuXO is a large carrier
23:59.29ariel_AgiNamu, ok I have a nice place in LA if you need.
23:59.38ariel_XO and nuvox
23:59.46AgiNamuyea, it has to be in colorado. we're connecting via 3 dedicated links to a company in colorado
23:59.54mikewho2ariel_ is there a service on the internet ot locate possible PRI's in your city?
23:59.56|Vulture|ah I don't like nuvox's customer service
23:59.59mikewho2im in the market

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