IRC log for #asterisk on 20080327

00:04.28lesouvageI just listen to the recording of the called party (before mixing) and there seem to be no recording of the announcement at all (just far away in the background) just silence. Is this as it suposed to be?
00:06.57riddleboxif I can issue sip show subscriptions, and core show hints, and get this response, http://pastebin.ca/958875 I should be able to have blf's right?
00:10.29*** join/#asterisk MACscr (n=Mark@adsl-75-23-64-188.dsl.peoril.sbcglobal.net)
00:10.55*** join/#asterisk bluemerlin (n=chatzill@82-33-65-91.cable.ubr01.trow.blueyonder.co.uk)
00:11.01galera1Just for the record, i can get incoming calls working (to an * behind nat) changing bandwith detection on eyebeam's softphone
00:11.04MACscris there a rpm of asterisk for centos? I know there is, just wanted to know if anyone had a link. The main ones I had found were taken down
00:11.22*** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net)
00:12.45*** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net)
00:14.08NatRHMACscr: http://djflux.net/rpms/enterprise/   I make no promises, just found on voip-info.org
00:14.45MACscrNatRH: thanks!
00:15.00NatRHeasily installed using source however
00:15.24MACscrI prefer point and click =P
00:15.31*** part/#asterisk galera1 (n=galeras@190.156.212.43)
00:17.11bluemerlinif there is a outdated file in the iax sources how would I update it?
00:18.25russellbuse a text editor?
00:19.47eric2use vi
00:25.07*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
00:29.32*** part/#asterisk efaistos (n=efaistos@ns24847.ovh.net)
00:29.45*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
00:34.26*** join/#asterisk [hC] (n=hardcore@216.251.157.146)
00:40.04shasta~grandstream
00:40.05jbotmethinks grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
00:40.20mwallingheh
00:40.24mwalling~rob0
00:41.19*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-76df5e87cc03ad87)
00:44.11bluemerlinmmmm grandstream I think Yugo is a compliment to them
00:47.01*** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net)
00:50.15ManxPowerYou won't get much support for RPM installed Asterisk around here.
00:50.16drmessanoAre there any good Asterisk related Jabber conferences?
00:53.12*** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net)
01:00.38*** part/#asterisk antdedyet (i=ady@gonzo.fearandloathing.org)
01:05.42boblutzI developed the whole IVR on a TDM11B ... Only 1 line and never had this problem
01:06.44boblutzToday, my boss and I tested out a TDM402B card.  Our existing phone system is kinda weird (to me at least).  We have 4 lines over 2 physical lines.  For each physical line, it was plugged into the Asterisk box.  To test out the IVR, we called the box from the other number on each physical line.   However, we noticed Asterisk would pick up calls - even after we hungup.  Is this normal/am I missing something crucial ?
01:12.09*** join/#asterisk klin3d (i=ircN@pc-32-231-86-200.cm.vtr.net)
01:14.33Juggiethe lines are obviously different?
01:16.25*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
01:18.30*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
01:21.49*** part/#asterisk bluemerlin (n=chatzill@82-33-65-91.cable.ubr01.trow.blueyonder.co.uk)
01:22.41*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
01:26.48boblutzJuggie, What do you mean the lines are obviously different?
01:30.24*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
01:31.51*** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose)
01:33.18Juggieboblutz, the analog lines did you test/develope on a different line then your production lines
01:34.46*** join/#asterisk Katty (n=The@hera.copi-rite.com)
01:34.49Kattyjbot: 5060
01:35.14Kattyjbot: rtp?
01:35.14jboti heard rtp is The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets.
01:35.23Kattyjbot: rtp ports
01:35.28Kattyjbot: ports?
01:35.29jboti guess ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm
01:35.35Kattybleh.
01:35.55Kattywhat rtp ports go with 5060 in teh firewall?
01:36.13Juggiecheck your rtp.conf
01:38.31KattyJuggie: oh ah
01:44.07lmadsenah oh
01:44.21Kattysip:110@copi-rite.com
01:44.23Kattysomeone call
01:44.27Kattyam testing
01:44.39Kattyfile: call me
01:44.41lmadsenwonder how long that'll take me to dial on my polycom, lol
01:44.56Kattylmadsen: file has me on speeeeeeeddial
01:45.06Kattyno one's calling me ;<
01:45.08Katty:<<<
01:45.09drmessanosmells phonesex
01:45.22NivexKatty: stby
01:45.40Nivexlooks up syntax of "originate" at console
01:46.01fileKatty: O.O
01:46.07Kattysomeone called me
01:46.10lmadsenthat's me!
01:46.17lmadsenalthough no audio / ringing on my side
01:46.24lmadsenblames nat
01:46.32Kattyhmm, yes
01:46.40Kattyit's getting past firewall
01:46.45Nivex's PBX made groaning noises like the Millenium Falcon trying to go into hyperspace
01:46.49Nivex"It's not my fault!"
01:46.50lmadsenI normally go through my asterisk box, so I don't have the NAT stuff setup on my polycom
01:46.53drmessanoHA!
01:47.01Kattyhmmmmm
01:47.23Katty[Mar 26 20:45:56] NOTICE[28526]: chan_sip.c:13859 handle_request_invite: Call from '' to extension '110' rejected because extension not found
01:47.50Katty110 is there, but...the context is missing, mayhaps
01:48.01Kattydigs through sip.conf
01:49.10boblutzJuggie, same lines
01:49.32boblutzsorry, i had to run to the store to buy cigs for my mom
01:50.37drmessanoO.o
01:52.03Kattyk, call me again
01:52.25Kattysip:110@www.copi-rite.com
01:52.34*** join/#asterisk Cle0 (n=cleo@41.250.136.123)
01:53.13lmadsenyou mean me? or did someone else try?
01:53.22Kattyohoh
01:53.26lmadsenat least this test will let you know if it matches right :)
01:53.27Kattyi see ringing
01:53.29lmadseneven if we get no audio
01:53.34Kattywho's calling me?
01:53.34lmadsen(or at least me get no audio)
01:53.35lmadsenme
01:53.44Kattyyou're not getting voicemail?
01:53.48lmadsendid it match right this time?
01:53.58lmadsenoh I probably am... just no audio because my polycom isn't setup to work behind nat
01:54.01Kattyyes
01:54.04Kattyyou're going to the right place
01:54.05lmadsensince it normally just talks to my asterisk
01:54.06Kattyoh okay
01:54.10lmadsenok cool, you're good to go then
01:54.10Kattyi've got RTP ports open
01:54.15Kattythanks
01:54.19drmessanolmadsen is having NAT issues????!!!!!?????
01:54.23drmessanoZOMG THE BOOK IS A SHAM
01:54.25Kattynow to test gizmo (=
01:54.30lmadsenoh... maybe if you set nat=yes in your general section?
01:54.42drmessano**ONLY JOKING PEOPLE**
01:54.51lmadsenkicks drmessano in the pants
01:55.01boblutzlol
01:55.01lmadsenI'm not having NAT issues... my polycom is :D
01:55.05drmessanolol
01:55.15lmadsenbut it makes sense since I'm not rewriting the packets like I probably should be :)
01:55.35lmadsenbut if Katty set nat=yes in general, that might help
01:56.07lmadsenif things were working really well, then I guess I should get audio even though I'm behind nat :)
01:57.01Kattygizmo won't dial me :<
01:57.15drmessanogizmo can be a pain
01:57.39drmessanomake sure you set an insecure=very or port,invite
01:57.45lmadsengizmo did what to the who now?
01:57.46*** join/#asterisk jmesquita (n=jmesquit@200.170.114.149)
01:57.50Kattyhmm, ok
01:58.08drmessanofor the Gizmo peer, I mean
01:58.16drmessanoOtherwise, it worky worky bad bad
01:58.37drmessanoSetting up Gizmo was a week of my life i'll never get back
02:00.18drmessanoUgh
02:00.35boblutz:-/
02:00.53*** join/#asterisk emist (n=emist@unaffiliated/emist)
02:01.05boblutzdrmessano, I respect you
02:01.18Kattyasterisk doesn't even show me i'm trying to call myself
02:01.32Kattypunts gizmo
02:01.32frogonwheelsRepeat qq: Scenario -   Nokia SIP via Wlan to  Asterisk on router  to PAP2T (SIP) on lan  - Connects but audio only from Nokia to PAP2t
02:01.45frogonwheels<PROTECTED>
02:01.47drmessanoI am a Reese's ANYTHING fiend.. If it's a Reese's product, I have tried it.. frequently.. I get this hollow chocolate rabbit for easter, and my wife suggests I put peanut butter on it.. in jest.  Well, I tried it anyway.. and now I wanna... puke.
02:02.13lmadsen"well hello there little buddy! come on in! whats you got there with ya? Is that yer dog? Guess there's no reason they can't have dogs up there on the moon; that's a pretty good lookin' pooch. I guess you've come back for some of Dr. Deathtubes delicious elixer"
02:02.15frogonwheelsBUT - if I ring an external number (via iinet) from the mobile - then it works fine!
02:02.45*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
02:02.51drmessanoSerious. Chocolate. Peanut butter. Overload. Where's. My. Insulin. ZOMG. Toothpicks.
02:03.41frogonwheelsiinet being a sip client over NAT
02:03.49frogonwheelsiinet being a sip server over NAT
02:04.59Kattyso have they made a sip phone for a blackberry yet?
02:05.05Kattybesides this gizmo insanity
02:05.13drmessanoNo
02:05.17drmessanoThe Gizmo thing sucks
02:05.21Kattyyeah
02:05.22lmadsenindeed
02:05.26Kattydamn
02:05.27lmadsenI tried it for like 10 mins on my nokia
02:05.30lmadseninstalled immediately
02:05.35Kattyindeed
02:05.35lmadsenuninstalled even
02:05.42Kattyi still kinda like it's IM bits
02:05.46drmessanoGizmo as a free ITSP, ala FWD... kinda cool.. Gizmo on the Phone/PDA.. suxor
02:05.56lmadsenso I just use the SIP stack that comes with my nokia
02:06.01drmessanoId rather use GTalk on the BB
02:06.08Kattymy blackberry doesn't have ny sip support with it :<
02:06.12Kattydumps bb in toilet.
02:06.14lmadsenattaches to my asterisk, and all is happy
02:06.14Kattyflushes
02:06.22lmadsendoes the BB have wifi?
02:06.27Kattyno
02:06.28frogonwheelslmadsen: nope
02:06.31lmadsengross :)
02:06.35drmessanoI want a BB sip phone so I can use my data plan for making calls to my PBX at home
02:06.35lmadsenhugs his Nokia E61i
02:06.36Kattyyeah
02:06.40Kattyi need an AT&T tilt
02:06.42Kattyel pronto
02:06.48Kattytoo bad it's el expensiveo
02:06.48frogonwheelslmadsen: so you got a Nokia connected to asterisk.
02:06.54lmadsendata in canada is prohibitively expensive
02:07.10frogonwheelslmadsen:  you have problems calling from Nokia to local extension?
02:07.14mihinomenestKatty: ever consider buying a sip <-> gsm gateway?
02:07.17drmessanoI'm gonna get a windows softphone and wait for Microsoft to develop an open source SIP client that works with Asterisk
02:07.17lmadsenfrogonwheels: yeppers --works pretty good too -- I've used the speakerphone with it on the desk and talked to clients, and they say it sounds good
02:07.17Kattythey need to make Pidgin for blackberry too
02:07.22drmessanosmartphone*
02:07.24Kattymihinomenest: bleh, no
02:07.27Kattymihinomenest: and no thanks
02:07.38lmadsenfrogonwheels: don't think so... seems to do everything I want, even when I'm remote at my parents house
02:07.41frogonwheelslmadsen: not to external - I mean to internal extensions (behind f/w)
02:07.43Kattymihinomenest: i wash my hands of that insanity (=
02:07.56drmessanoDoesnt the gateway defeat the whole purpose?
02:07.57lmadsenI don't know... there's only me at home, so I don't call myself too often :)
02:08.02frogonwheelsdid you see the problem I had?
02:08.02lmadsenmaybe I'll try now
02:08.19frogonwheelsyou just got phones on an ATA?
02:08.41*** join/#asterisk tengulre (n=tengulre@124.42.50.9)
02:08.52lmadsenfrogonwheels: I just called my polycom at x100 and it worked fine
02:08.56frogonwheelslmadsen: the problem I have is when I call  nokia to ata (via asterisk)  I don't get audio from ATA to nokia - only nokia to ATA
02:09.09lmadsensounds like an ata config issue...
02:09.12frogonwheelspolycom is sip, right?
02:09.17lmadsenyou should not get one-way audio on a lan
02:09.22lmadsenyes, all sip here
02:09.27lmadsenmy landline is independent of asterisk
02:09.28frogonwheelspossibly.  If I call  ATA to Nokia it's fine!
02:09.37frogonwheelsmine's not.
02:09.40lmadsenstrange
02:09.45drmessanoOk so
02:09.48frogonwheelshave no 'landline' only sip phone
02:09.49lmadsenso ok
02:10.14drmessanoI go to Trixbox.org and there's a pic of this rocker chick on the home page with the caption "Meet our box".  Isn't that giving off the wrong image?
02:10.22lmadsenKatty: can you try sip:ivr@leifmadsen.com and see if that works for you?
02:10.26lmadsenhasn't tried it in a very long time
02:10.30*** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.179.104)
02:10.38boblutzlmadsen, I called that yesterday at work
02:10.39lmadsendrmessano: no, it's giving off the right image
02:10.42lmadsenboblutz: oh?
02:10.44drmessanolol
02:10.45lmadsendid it work? :)
02:10.46Kattylmadsen: on gizmo?
02:10.51drmessanotouche`
02:10.52lmadsenKatty: just regular sip
02:10.57Kattylmadsen: i'm not at work
02:11.00lmadsenah ok
02:11.05Kattyi'm sitting on the couch
02:11.09Kattywatching UFOs
02:11.09lmadsennicely played
02:11.10drmessanonaked?
02:11.11Kattyon history channel
02:11.19drmessanooh
02:11.21lmadsenrolls his eyes
02:11.25Kattydrmessano: no, but i'm sitting next to my man.
02:11.25lmadsendrmessano: I'm naked if that helps
02:11.33drmessanolol
02:11.35Kattydrmessano: and my 4 kids.
02:11.35lmadsenworks from home
02:11.41drmessanokatty: that's hot
02:11.46drmessanokatty: tell me more
02:11.46Kattydrmessano: and by kids i mean ferrets.
02:11.56lmadsenneeds more beer
02:11.57Kattykids are little demon spawn come to life.
02:12.05Nuggetagrees
02:12.12lmadsenthank god I have no kids
02:12.14boblutzsips a coors light
02:12.16Kattyhi nuggety!
02:12.22Nuggethuggles Katty
02:12.23Kattyi had a banana daquari earlier
02:12.24lmadsenboblutz: so did that sip:ivr@leifmadsen.com work yesterday?
02:12.26Kattywith mexican foods.
02:12.31Kattyand then a raspberry daquari.
02:12.31Nuggetjust poured a Spaten Optimator
02:12.47KattyNugget: jeezy chreezy what's that?
02:12.50lmadsenI had raspberry vinagrette on my salad tonight
02:12.54Nuggeta frou-frou fancy beer
02:12.59drmessanoI'm trying to find that one XKCD about why women don't chat on the internet.. because guys are pigs.. and they send a girl with an EMP cannon to this dudes house
02:12.59boblutzlmadsen, I actually had my sound turned off, but linphone said it connected ... I called it a couple weeks ago and it worked
02:13.05lmadsenhawtness
02:13.06KattyNugget: meh, beer. no thanks.
02:13.08lmadsenthx for the update
02:13.12lmadsenused it for testing?
02:13.13Juggielmadsen, can you do a svn update on trunk, then a make dist-clean, then make menuconfig and see what it does.
02:13.14KattyNugget: i'll have a fluffy bunny drink, thxs.
02:13.20lmadsenJuggie: sure
02:13.29boblutzmore curiosity ... calling sip:10.69.1.120 is kinda lame
02:13.37lmadsenhaha, fair enough
02:13.38KattyNugget: i bought a floor steamer earlier today!
02:13.44lmadsenI'm glad I haven't broken that sip uri then
02:13.45JuggieQwell, commited a fix, but looks like he broke it
02:13.53drmessanohttp://xkcd.com/322/ <--- CLASSIC
02:13.59Nuggetfun!
02:14.05lmadsenJuggie: sure, updating now
02:14.11Juggielmadsen. basically i want to know if menuselect compiles or not
02:14.20KattyNugget: i'm going to chase the ferrets around with steam tomorrow.
02:14.23lmadsenfair enough, I'll let you know in a couple mins
02:14.25Juggiebasically, you should do make dist-clean then make menuselect, no ./configure
02:14.29KattyNugget: it'll be grand.
02:14.34filehere it goes, here it goes, here it goes again
02:14.40Kattyfile: hai file
02:14.46lmadsenmake distclean seems to have forced me to run ./configure
02:14.48fileKatty: ohai2u, orange juice?
02:14.51drmessano"Dude, she's hot.  Is she single?" "Joanna, Fire."
02:14.59Kattyfile: no
02:15.02Kattyfile: no orange juices
02:15.05Nuggetheh
02:15.07Juggielmadsen, can you pastebin the output
02:15.08Kattyfile: i has some lettuces in the fridge tho
02:15.09fileKatty: fruit punch?
02:15.17Kattyfile: orange vodka
02:15.18lmadsenJuggie: still running ./configure at this time
02:15.22Kattyfile: and some vanilla vodka
02:15.28Kattyfile: half a bottle of rum...some diet coke
02:15.28lmadsengoes to grab his bowl
02:15.34Kattyfile: and some milk
02:15.35fileKatty: oic
02:15.36lmadsenfile: did you get your vino?
02:15.45filelmadsen: what?
02:15.55filedoesn't keep wine around
02:15.55lmadsenyour facebook said you had to obtain wine
02:15.58Juggielmadsen, make dist-clean runs configure but then cleans it again
02:16.08fileno! I was forced to drink wine at a fancy cafe over in the french part of town
02:16.27lmadsenya.. said I had to run configure first... so I ran it... then ran distclean, now running make menuselect
02:16.30lmadsenlol
02:16.33lmadsenf: command not found
02:16.38Juggieya ok so Qwell broke it
02:16.43Qwellhuh?
02:16.48KattyHI QWELL
02:16.50fileQwell doesn't break anything. ever.
02:16.51lmadsen/bin/sh: f: command not found
02:16.51lmadsenUnable to open 'build_tools/menuselect-deps' for reading!  Did you run ./configure ?
02:16.51lmadsenmenuselect changes NOT saved!
02:16.59Qwelldid you run configure? :p
02:17.00JuggieQwell, heh.. you said you commited 'my' patch for the makefile
02:17.04Juggiebut you did sometihng else
02:17.05Qwellwhich one?
02:17.07Juggiewhich does not work
02:17.12lmadsennot after my make distclean :)
02:17.14Juggiehttp://svn.digium.com/view/asterisk/trunk/Makefile?r1=106250&r2=108799&pathrev=110911
02:17.17lmadsenrunning it again though
02:17.24lmadsenI'll see if it works at all, heh
02:17.30Juggielmadsen, if you run configure now, and then run make menuselect every option will be selected
02:17.31Kattyit will asplode.
02:17.54lmadsenimplodes
02:17.58JuggieQwell, way to break my patch!
02:18.01Qwellpoints to russell
02:18.06lmadsenI breaka yo face!
02:18.17lmadsenwoh.... fancy menuselect now
02:18.24JuggieQwell, you commited it, i blame you :)
02:18.32lmadsenand yes... everything is selected now
02:18.37Juggieyes its very hosed.
02:18.38Qwellno I didn't
02:18.40drmessanoUnstable crap
02:18.51Qwellyou're reading it wrong :p
02:19.02Juggieah, so i am
02:19.02lmadsenso I noticed tonight that ABE C.1.4 didn't install .g729 files for me, :)
02:19.05Juggieruss commited
02:19.07Juggiecan i fix it
02:19.15Kattywe're commiting russel?
02:19.17Kattywhat?
02:19.22lmadsenwe did that long ago
02:19.27lmadsencommitted him to a bush
02:19.44Kattywell...
02:19.47Kattywe have to cut him some slack
02:19.49Kattyhe works with Qwell
02:19.57Kattythat's enough to drive anyone mad.
02:20.09Qwellthere's a firewall between our offices!
02:20.19Qwellwell, a fire wall, anyways
02:20.21jm|laptophi guys
02:20.27Kattyhai
02:20.28filea literal wall of fire.
02:20.44Kattyfile: made with paper and crayon and scissors?
02:20.56JuggieQwell, this works for me http://www.pastebin.ca/959037
02:21.05Kattyi need to get some construction paper and write Katty Busy Today on it, and tape it to my door at work
02:21.19Kattywith maybe pls go away, kthx below it
02:21.23drmessanoIn crayon?
02:21.26lmadsen"I can only please one person per day. Today is not your day. Tomorrow does not look good either."
02:21.26Kattyyes
02:21.28Kattyin crayon
02:21.41Kattyit adds the katty professionalism to it
02:22.43jm|laptop<()  CRAYOLA   )
02:22.53drmessanoHA
02:23.02jm|laptopascii_art
02:23.05drmessanoReminds me of my sign I had on my office door at my old job
02:23.10lmadsen<[   CRAYOLA   ]
02:23.29drmessano8 1/2 x 11 copy paper.. with "DO NOT PRETURB - (GOOGLE IT)" written on it in sharpie
02:23.49lmadsenyou mean perturb?
02:23.56drmessanoYeah, type
02:23.57drmessanoYeah, typo
02:23.59lmadsenheh
02:23.59drmessanoCRAP
02:24.01lmadsenlol
02:24.03Kattydrmessano: we have a wide format kyocera
02:24.03lmadsensuxor
02:24.10Kattydrmessano: i could just print out a 36" by 8ft banner
02:24.15lmadsenok, off to watch some futurama on the couch
02:24.20Kattyand tape it on the FLOOR
02:24.21Kattyin the hallway
02:24.25jm|laptopused to have a Texjet2
02:24.28drmessanolol
02:24.52drmessanoThats way too lame
02:25.05drmessanoMake a banner on a dot matrix
02:25.07drmessanoGo old school
02:25.09Kattyeww
02:25.11Kattythat's noisy
02:25.19Kattyno noise allowed in my bubble
02:25.26jm|laptopand a Grenadier
02:25.33drmessanoGet a copy of Print Shop for DOS and go with it
02:26.06drmessanoI bet I could start an online business making dot matrix banners
02:26.07jm|laptoppsconvert untitl~1.ps
02:26.13boblutzdos?
02:26.38jm|laptopmumbles something about GraphixAdvantage
02:26.43droopsits how old folks used to compute
02:26.44drmessanoDoes any modern app support making banners across dot matrix pages in that old school style?
02:27.32drmessanoDOS + Vmware + Okidata Microline 320
02:27.34jm|laptopdrmessano: there's that macro art website where it makes separate pages from an image with a multiplier
02:27.39jm|laptopyou could aalib it then dot matrix print
02:27.59jm|laptopOkidata?! Luxury! Citizen 120D!
02:28.03drmessanoThats lame.. I want it native
02:28.08*** join/#asterisk InHisName (n=Administ@c-71-225-221-149.hsd1.pa.comcast.net)
02:28.19jm|laptophad a ZX Printer
02:28.27jm|laptopthermal transfer, baby.
02:28.31jm|laptop120x86
02:28.34drmessanoI want an app that runs in conventional memory in DOS
02:28.49drmessanoPushing 512kb
02:28.51jm|laptopbeats drmessano with emm386
02:28.59jm|laptophymen.sys
02:29.17drmessanoupper memory is for GUI lamers
02:29.19drmessanolol
02:29.29drmessano640kb is enough
02:29.29jm|laptop(:
02:29.46boblutzlol
02:30.33jm|laptoplh oakcdrom.sys
02:31.02*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
02:31.15jm|laptopmscdex?
02:31.19jm|laptopMassey Cedex
02:31.22drmessanomouse.com <---- the app that ended it all
02:31.23jm|laptopyou decide.
02:31.50jm|laptopwin /3
02:31.59boblutzmouse.com is probably worth lots of $$$$$$$$$$$$$$$$$$$$$$$$$$$$$
02:32.05jm|laptop*BUILD !BOOT
02:32.17drmessanoboblutz: The application, not the domain
02:32.31jm|laptopbuffers=26
02:32.33boblutzo, i went to mouse.com and i was like wtf this site sucks
02:32.43drmessanofiles=99
02:32.53jm|laptop:O
02:32.55jm|laptopcavalier.
02:33.47jm|laptopc:\progra~1\drsolo~1\drsolo~1.exe -auto
02:33.47drmessanoSTACKS=0,0
02:34.06drmessanoand my fav
02:34.18jm|laptopxms > emm386
02:34.23drmessanoLASTDRIVE=Z
02:35.14drmessanoI'm getting flashbacks of my last real DOS experience.. Netware lite
02:35.22jm|laptopholy crap
02:35.25*** part/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net)
02:35.38jm|laptopGood $greetingtime
02:35.42drmessanoNetware Lite is enough to piss off the pope
02:35.51*** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au)
02:36.05jm|laptopused to admin netware :/
02:36.09jm|laptopover Win95
02:36.27drmessano<-- used to admin netware on 2000/XP
02:36.37jm|laptopit was /fractionally/ better than when I did Citrix over WinNT 3.51
02:36.53drmessanoshudders
02:36.54jm|laptopset mode /install
02:37.02drmessanoThank god Citrix has gotten better
02:37.10drmessanoIt NEEDED it
02:37.11jm|laptopmumbles
02:37.20drmessanoI said "better"
02:37.21drmessanolol
02:37.42jm|laptopmany people dropped Citrix with W2K
02:37.51jm|laptopas much of the code was merged
02:47.56*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
02:49.45*** join/#asterisk Tired_ (n=chatzill@d66-183-177-28.bchsia.telus.net)
02:51.57Tired_Hi.  I've been researching some stuff about VoIP, and I came across Asterisk.  What I want to do is to connect my home phone line (POTS) to my computer, so I can access it over the internet (ie: to make a call over my home phone line, using my laptop at a wifi hotspot).  Is this the kind of thing Asterisk can do?
02:52.32Tired_I found lots of stuff on how to connect POTS equipment to a VoIP line, but not the reverse.
02:52.47droopsyes tired, this is something it can do
02:52.55*** join/#asterisk hohum (n=dcorbe@68.26.66.211)
02:53.06Tired_:)  Is it very expensive to set up?
02:53.18Tired_hardware-wise
02:53.31droopsif you already have a linux box, you only need a way to connect the asterisk box to the pstn
02:53.41droopson the voip side, you can use a softphone on your laptop
02:53.46droopslet me get you a link
02:54.14JTyou need an
02:54.20droopsi use one of these
02:54.20droopshttp://www.sipura.com/products/spa3000.htm
02:54.21Tired_That's what I was wondering about...I saw some PCI adapters on the asterisk website that cost a lot
02:54.23JTFXO card or an ATA with an FXO port
02:54.49drmessanoStay away from the X100P.. unless you enjoy arguing on IRC a lot
02:55.04Tired_My least favorite thing to do.
02:55.24drmessanoThen my work here is done
02:55.35*** join/#asterisk RoyK (n=roy@ip-108-27-149-91.dialup.ice.no)
02:56.21Tired_Is this the kind of project a reasonably proficient user could set up by himself in a reasonable time frame?  I noticed just about everything on the asterisk site was geared towards enterprise.
02:56.44drmessanoIt's not hard
02:56.45JTsure
02:56.49Tired_Just wondering if there's a more appropriate program
02:56.49JT~thebook
02:56.50jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
02:56.51JTwill help
02:56.51Nasradroops I thought that one was replaced by another one or one can still use this one....
02:56.59drmessanoSPA-3102
02:57.00Tired_:)  Thank you.
02:57.03drmessanoThe SPA-3000 is old
02:57.06droopsi still use it, and its on their site
02:57.09droopsand yes it is old
02:57.23droopsbut i like to speak from experience
02:57.39Nasrajust wondering....because I am just about to get started wanna to know....
02:58.22droopsTired_, take a look at this book http://tfot.leifmadsen.com/
02:59.06Tired_that's a book
02:59.13droopsit will answer most of your questions, if you can install from source in linux and edit a few config files, you will have no trouble installing and configuring asterisk
02:59.16Tired_i'm probably not going to read it all right this minute
02:59.28droopsits a reference book
02:59.38Tired_i'm comfortable with linux, this is just novel to me
02:59.42Tired_oh, i see
02:59.47Tired_sweet, thank you
03:00.05droopsyou can buy a printed copy from oreilly
03:00.26Tired_does the project benefit from the sale?
03:00.32Tired_or just o'reilly
03:00.35droopsi dont know
03:00.50droopsthe guys who wrote it benifit
03:01.00droopsand they do give the thing away to the community
03:01.06Tired_and us, i suppose, for reading it
03:01.16droopsyes
03:01.29droopsi have a printed copy on my desk most of the time
03:01.34*** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au)
03:01.47droopsok bbl
03:02.05Tired_well, it sounds like i am on the right track, then...i'm going to go off and read a bit.  thanks for the pointers
03:02.19Nasradroops...lemme ask you a question..since I am a newbie on all these....
03:02.30droopsNasra, ok
03:02.46Nasrathe box for asteriks has to be just asterisk?
03:02.50droopsno
03:03.00Nasrabesides linux
03:03.21Nasrayou know what problem....
03:03.22*** join/#asterisk PepOSX (n=angeldav@190.72.130.192)
03:03.28droopsat my house, i have a p3 that is my asterisk box, a testbed webserver and our file server
03:03.38Nasraok
03:03.51droopsfor my work stuff, they are mostly asterisk boxes, as hardware is cheap
03:04.08drmessanoAsterisk needs a box that isn't bogged down though
03:04.13droopsyes
03:04.15drmessanoSo sharing is fine
03:04.16drmessanoBut
03:04.19drmessanoNot a high use box
03:04.26drmessanoor.. one that's too loaded for it's speed
03:04.36NasraI just bought a flea market special for $30 today and is a p3, 128 ram , 10.0 HD
03:04.43NasraWill this be okay ?
03:04.51drmessanoI'd up the RAM
03:05.00Nasra256
03:05.03droopsfor your house, up the ram, but other than that yes
03:05.07drmessanoand use some unbloated distro.. no X
03:05.23Nasrawhich distro though
03:05.28NasraI am using Ubuntu
03:05.41droopsdebian, centos
03:05.51Nasraoh by the way.....
03:06.01drmessanoLemme guess
03:06.05drmessanoYou're on dialup
03:06.08NasraI ordered some cd's from Disc.com
03:06.11drmessanooh
03:06.34droopsok im out
03:06.49Nasraand they sent me 5 Cd's with Kde .....and when I tried to install it it does not run at all....what is going on?
03:07.09Corydon76-dig'night, droops
03:07.17boblutzNasra, You need to boot from the cd perhaps?
03:07.28drmessanoTry #someotherchannel
03:07.37boblutzlol
03:07.38Nasrathey are Debian boblutz...
03:07.49boblutzo ok, you dont need to boot from cd then
03:07.55Nasralol
03:08.12drmessanoDebian doesn't even need to be inserted
03:08.25drmessanoYou can install Debian while it's sitting on a shelf across the room
03:08.27drmessanoIt's that good
03:08.51drmessanoand you can install debian on ANYTHING
03:08.52Nasradrmessano....wanna to install it to run Asterisk
03:09.09drmessanoI currently have debian running on a Cheez-It
03:09.12Nasra64 bits processor
03:09.21boblutzNasra, http://goodbye-microsoft.com/
03:11.32boblutzdrmessano, Can you PM me your wiki ?
03:11.37boblutzurl that is
03:12.32Nasraboblutz...what  is the procedure once I have the linux box ready to go?
03:12.52Nasrafor setting up the Asterisk I guess
03:13.03boblutzi could be a dick and say ..
03:13.05boblutz~thebook
03:13.06jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
03:13.26*** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net)
03:13.45boblutzbut really, all you need to do is make sure you have all the dependencies, then get the zaptel, libpri, and asterisk source, and then compile libpri, then zaptel, then asterisk
03:13.56boblutzif you are going pure voip, you no need libpri or zaptel
03:14.14boblutzdrmessano, thanks
03:14.26NasraI will tell you what I have in  just in moment
03:15.03boblutzNasra, hold up just a second
03:15.18boblutzyour getting ahead of yourself, you implied you couldnt install linux a minute ago?
03:15.57Nasrawhat I said was...that I had Debian....5 cd's but it did not run at all....but then....
03:16.13NasraI have Ubuntu running properly....
03:16.51Nasrainstead of Debian....but is running beautifully.....now what I want is install the Asteriks so I can start testing with it....
03:17.35drmessanoHaving X on there is going to be slow
03:17.37NasraI am not that good on Linux yet
03:17.58Nasrajust reading alot at linuxbasics.org
03:18.58drmessanoI think I am gonna implement the "Circus Advisory Protocol" for now on..
03:19.12Nasragetting to know all these commands....but once in a while I ask a few questions....regarding Asteriks
03:19.27*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4ab13ca345cdb641)
03:19.29drmessanoI'll comment once on something, then sit back and eat my popcorn
03:19.47drmessanoIt's Asterisk
03:20.07drmessanoSecondly, how fast is that P3?
03:20.11drmessano550?  600?
03:21.00NasraI think it's 550...no sure just got it
03:21.21drmessanoYou really need to not start off on a system with X running on it..
03:21.24Nasra1000
03:21.36boblutz`cat /proc/cpuinfo` ?
03:21.37Nasrap111 1000
03:22.03drmessanoP3 1GHZ with 128MB Ram?
03:22.09drmessanoThat's... unbalanced
03:22.14Nasraok
03:22.27jameswf-home[]D[][]V[][]D
03:22.42boblutzrandom but lol
03:23.40Nasramy Asterisk box will be just simple .....just to pickup some messages and use the Ivr....and music on hold....
03:23.49jameswf-homeso my deadlock solution for someone who couldnt upgrade exec($(killall -9 asterisk; asterisk;))
03:24.25Nasramaximum 3 Exts
03:25.25drmessanoLOL
03:25.25jameswf-homewants a (free) macbook air
03:25.26drmessanoOk
03:25.27drmessanoSo
03:25.47drmessano3 extensions (check), IVR (check), MOH (FAIL)
03:25.55drmessanoYou need to not run X
03:25.58boblutzwants a pony
03:26.16jameswf-homeI saw a donkey in a show once
03:26.35drmessanoI'm totally screwing my Circus Advisory Protocol
03:26.42Nasradrmessano....what you exactly mean by need not to run X?
03:26.43drmessanomunches on his popcorn
03:26.57drmessanoOh nothing.. mmmm butter
03:27.07jameswf-homeAsterisk + X = ummm dude wtf *SLAP*
03:27.20drmessanoNo
03:27.21jameswf-home~X
03:27.23drmessanoFAIL
03:27.49boblutzNasra, X11 is like.....
03:27.52drmessanoAsterisk + X = u mm  D de  TF *SL P*
03:28.06jameswf-homeheh
03:28.09boblutzits like falling into old habits
03:28.40jameswf-homeif you really must surf porn on your * box use lynx
03:28.53boblutz8===D ftw
03:28.58boblutzuh
03:29.02boblutz8===D () ftw
03:29.03boblutzi mean
03:29.12jameswf-homeonly === more like =====//====//===//====
03:29.15jameswf-home:)
03:29.39jameswf-home8=////==D ewwww
03:29.40boblutzNasra, X11 is basically the software that lets you run KDE
03:29.47friedrich|can someone hand me libbfd-1.18.so for i586? :)
03:29.47Nasraok
03:30.29jameswf-homeE: Couldn't find package libftw
03:30.57drmessanoA ter k on X r ns gr at i yo 're  ot i to c ll cl  ity
03:31.05Nasraboblutz...what I want to do is use my copper line wiht voip on Asterisk just something basic
03:31.28drmessano~cheap
03:31.28jbotextra, extra, read all about it, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
03:31.46boblutzis tdm410 a cheap card?
03:31.48jameswf-homecarefull copper theaves everywhere
03:32.02jameswf-home~cheap
03:32.02jbotextra, extra, read all about it, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
03:32.04drmessanoZOMG STOLE MY F O CARD
03:32.27jameswf-homewow XWindows and clone cardz..... oh my
03:32.59drmessanoDarth Vader used X100P's on the Death Star..
03:33.12boblutzx100p is a fxo module ?
03:33.31jameswf-home~x100p
03:33.32jbotx100p is, like, an obsolete card.  You don't want to bother trying to make it (or any of the "digium compatible" clones) work.  Get a TDM01B, and you will save your sanity, your hair, and countless other things.
03:33.35drmessanoNo, it's a glorified modem that happens to work as a FXO device
03:34.10Nasrawhich is the best sip / voip provider that I can use ?
03:34.20boblutzsee thats f'd up
03:34.29jameswf-home~itsplist-us
03:34.29jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net
03:34.51plikbasic cheap hardware for home PLUS Best voip ITSP = bad thinking
03:35.08drmessano~nowwhat
03:35.09jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
03:35.11jameswf-homeNasra:  is batting 100000
03:35.23drmessanoNasra is just like a mini-mall
03:35.45Nasraoh lol
03:36.01plikmini-mal... unlike his X install
03:36.24drmessanoNasra: Cheap is not good.. you don't need $1000000 to set up Asterisk, but no one will support you if you're a cheap bastard that doesn't take the advice that's given
03:36.32jameswf-homeheh my wife officialy lothes sammy stevens
03:36.49drmessanoheh.. mine was just asking me how many times I watch it a day
03:37.05drmessanoI told her 6 on the weekdays.. and if I miss one, I make up for it the next day
03:37.05jameswf-homeOk I have the mp3....
03:37.06Nasrano, don't get me wrong man.....I am allright....wanna have the best ...
03:37.21drmessanoYou're already batting 1000 Nasra
03:37.22jameswf-homeits on my blackberry :))
03:37.25boblutzDoes the tiger jet chipset make one a cheap bastard?
03:37.39jameswf-home~tigerjet
03:37.41drmessanoboblutz: No, your RoMex watch makes you a cheap bastard
03:37.46boblutzlol
03:37.52jameswf-homemembers only jacket
03:38.18Nasraplease treat me nicely guys.....you deserve all the best respect.....I just learning ......
03:38.26jameswf-home~newb
03:38.26jbotmethinks newb is Don't bother telling us you're a "newb", "n00b", or "nub". We can tell.
03:38.30jameswf-home~nice
03:38.30jbotwell, nice is prime example of SuperJuan, or a good term for GNOME a derogatory term meaning bland, boring, feeble, or just crap. Example: That's a nice haircut. a city in france, or a program that will run a program with a modified scheduling priority (from -20 to 19, where 19 is the lowest).
03:38.32jblackHow do I know whether I need a 3.3 volt digium card, or a 5 volt digium card?
03:38.46drmessanoDVM?
03:38.55boblutzjblack, That depends on your pci bus
03:38.58jameswf-homemulti meter?
03:39.01boblutzon the mobo
03:39.11jameswf-home~sarcasm
03:39.12jbotOh a sarcasm detector, that's a *really* useful invention!
03:39.24jblackI don't have probes small enough to get into a pci slot. :P
03:39.37drmessanojblack: That's not what I heard
03:39.44drmessanoDamn.. That was bad
03:40.09jblackYeah. That one cost you some coolness points in my book.
03:40.24drmessanoOh no it didn't
03:40.35drmessanoYou loved it.. you're too sick not to have
03:40.52jblackheh. Anyways.
03:41.06*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
03:41.22drmessanojblack: How many PCI slots do you have?
03:41.28jblackSeriously. I honestly don't know how to decide which I'd need.
03:42.04jblackOffhand, I don't know how many slots in the system. It's slated for a core 2 duo, 2ghz range, gig of ram.
03:42.16boblutzweak!
03:42.40boblutzu can buy a laptop from walmart that trumps harder
03:43.23drmessanoWho the hell buys a Quad Core with 4GB RAM for a home system with 3 extensions.. Themselves, Dad, and Mom?
03:43.26drmessanoOh. boblutz
03:43.37drmessano;)
03:43.38boblutzdrmessano, get it right
03:43.41boblutzits just me and mom
03:43.50jblackDon't worry. It's just an interface box for a much more powerful system.
03:43.54drmessanoSo you have 2 extensions then...
03:44.04drmessano101 - Kitchen and 102 - Basement
03:44.13boblutz"mom is my hot pocket done?"
03:44.25drmessano"Bring it down please"
03:44.30boblutzhaha
03:44.35jblackThe real servers are dual quad cores with 16 gigs of ram, running * wihin xen. This box is just to take in the PRI w/ the card, and shuffle it over iax to the real machines
03:45.00drmessanoWeak.. TDMoE is better for internal
03:45.26boblutzi am yet to use asterisk in the home.  i told my boss that we look like ass holes deploying a tdm402b on a xeon + 2 gigs
03:45.26Nasraguys good nite thanks for chat
03:45.36boblutzNasra, dont give up
03:45.42Nasranever and thanks
03:45.43jblackSo, back to the original question... how do I know whether I need a te210p, or a te207p?
03:46.07boblutzjblack, unless someone wants to prove me wrong, it depends on the type of pci bus you have
03:46.10jblackAre 5 volt systems for x86, and 3.3 for amd64, or the inverse, or something else?
03:46.50jblackSo for any given desktop system out there these days, if you point at one, there's a 50/50 shot at it being 5 volt
03:47.22plikjblack: google knows
03:47.29boblutzhttp://www94.web.cern.ch/hsi/s-link/devices/s32pci64/slottypes.html
03:47.37plikrather, google knows who knows
03:47.56jblackboblutz: Thank you!
03:48.15plik"You can tell by observing the keying patterns in the PCI connector"
03:48.21drmessanoA DVM would fit in there
03:49.04drmessanoI want an AGP TDM card
03:49.11drmessanoForget PCI
03:49.21UnixDognever going to happen
03:49.35boblutzagp makin a comeback
03:49.36UnixDogthere are usb units in the works thou
03:49.37Nuggetbah, AGP is bollocks.  You want sources
03:49.43Nuggeter, VESA-VLB
03:49.47drmessanoYES
03:50.10drmessanoA 3 1/2 foot long VESA TDM card would rock
03:50.21boblutzby the way, i push 4 gigs on a 32bit OS
03:50.21Nuggetbtw, does anyone have the EISE Configuration Utlitity driver disk for an X100EISA?  :)
03:50.26Nuggeter EISA
03:50.45UnixDogfirewire
03:50.49drmessanoI want a card that fits the guide on the back of the front fan carrier, damnit
03:51.01drmessanoFront to back, bitches
03:51.09UnixDogMR
03:51.41jameswf-home~drmessano
03:51.41jbot[drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway
03:52.03boblutzgross!
03:52.11drmessanoI need a nice monochrome LCD monitor
03:52.31boblutzso....a char is really just a small number
03:52.33boblutzthats deep
03:52.33drmessanoI have a hercules video card for it
03:52.54*** join/#asterisk klin3d (i=ircN@pc-32-231-86-200.cm.vtr.net)
03:53.26drmessanoI miss the old days of... "DONT TAKE OUT MY MONOCHROME VIDEO CARD OR I CAN'T PRINT"
03:53.29drmessano:(
03:54.33NuggetOB/GYN == Oh Boy!  Got Ya Nekkid
03:54.44boblutzLOL
03:54.47drmessanoHA
04:01.20jameswf-homesomeone wants to disable oslec in trixbox :))
04:02.52drmessanolol
04:03.34*** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290)
04:04.56*** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net)
04:09.11MACscrok, I downloaded the asterisk files from the digium site, but I do not see a folder with the libpri files or even zaptel
04:10.57jameswf-home~nowwhat
04:10.58jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
04:11.53MACscrjameswf-home: thanks, but my issue isn't even with configuring it, that's not an issue at all. Its with installing all the base files
04:12.30drmessanoYou need to read the install guides on the site
04:13.10MACscrim guessing the readmes aren't good enough?
04:13.10jameswf-home~libpri
04:13.58jameswf-homeMACscr: check out elastix
04:14.54jblackI'm a littel confused by the te122 description. It's a 24 channel card, but from the way I read it, it can only handle 12 voice channels. Is this correct?
04:15.22MACscrjameswf-home: thanks, I appreciate it, but I have no interest in those distros. I just want asterisk installed, nothing else at the moment
04:15.42jameswf-homethen
04:15.46jameswf-home~book
04:15.47jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
04:15.55drmessanojblack
04:16.08drmessanoNo, it appears to handle voice and/or data
04:16.12*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b7753c253f95af3c)
04:16.31drmessano"For Example"
04:16.56jblackYeah. I can't tell if it's an optional feature "Yeah, you can even do 12 & 12", or a requirement.
04:17.50drmessanoWell
04:17.52drmessanothe TE122 may be combined with Digium's VPMADT032 echo cancellation module to deliver 128ms of echo cancellation across 30 channels in E1 mode or 24 channels in T1 mode
04:18.02drmessanoObviously you wouldnt need an echo canceller on a data channel
04:18.58jblackyeah. I figured it can do 12 channels of voice. I need to cover 24 channels of voice.
04:19.30jameswf-homezaptel wont start EC on a d chan
04:19.54jameswf-homeRCB24FXX
04:20.04drmessanoThats why I said
04:20.06jameswf-home:)
04:20.16drmessanoIt's incredibly obvious it will do 24 voice channels
04:20.29jameswf-homeoh snap R1t1
04:20.32jameswf-home:)
04:20.42jameswf-homewhere did newb go
04:21.25drmessanoHe's off installing AsteriskWin32, I think
04:21.29jblackDear world: I'm completely incompetent when it comes to T1 hardware. I admit this openly mostly because it's blatently obvious.
04:21.50jblackI'm very keen on any suggestions of things that I should go rtfm for a bit.
04:22.31jameswf-homeT1's are easy signallinfg dchan bchan BAM
04:23.30jblackI'm naive enough that I think it possible that a T1 card that claims "12 & 12" might be limited to 12 & 12 because of DSP limitations, or io bus limitations.
04:23.48jameswf-homeI have thousands of hours on asterisk so its all second nature I have no idea how difficult it is
04:24.12drmessanoAlthough the same card has a provision for an echo canceller that works on 24 channels on that card?
04:24.27drmessanoand the 12 and 12 is cited as an "example"
04:24.47jameswf-homeour DSP do 32 without thought I imagind digium is using the same stuff so it should be ok
04:24.47jblackGreat. So 24 voice channels is an option with the t122. That rocks.
04:24.57jblackWhat do you sell?
04:25.07jameswf-homeworks @ Rhino
04:25.14drmessano[00:17] <drmessano> the TE122 may be combined with Digium's VPMADT032 echo cancellation module to deliver 128ms of echo cancellation across 30 channels in E1 mode or 24 channels in T1 mode
04:25.17drmessanoI said that earlier
04:25.32drmessanoYou wouldnt apply echo cancelling to a Data channel
04:25.50jblackSure... But can the TE122 do 24 channels without the VPMADT032
04:25.57jameswf-homedrmessano: is ocd he repeats stuff :)
04:25.57jblackI think you've answered my question.
04:26.04drmessanoThe VPMADT032 is an echo canceller
04:27.05jblackYah, I caught that. But if it worked as a coprocessor to augment the te122 (thus enabling it to do 24 voice channels rather than 12 by offloading the echo cancellation)... See where I might be worried that I'd get the wrong thing?
04:27.32drmessanothe TE122 may be combined with Digium's VPMADT032 echo cancellation module
04:30.16jblackDescriptions that I found too ambigious to base a purchase on don't become any clearer just because they're pasted in irc.
04:30.20jblackBut thanks for trying. =)
04:31.11jblackI'll call them up tomorrow and ask them for a definitive answer... "Can the te122 do 24 voice channelswithout the vpmadt032"
04:31.34jameswf-homeR1t1 does 24 definately....
04:31.50jblackHow much is a r1t1?
04:32.02jblacknever mind. googled it
04:32.04jameswf-homewith EC or no EC
04:33.04jblackguesses that EC stands for error correction.
04:33.11drmessanoUh
04:33.13jblackI don't know whether I need that or not
04:33.14jameswf-homeEcho Cancellation
04:33.16drmessanoOr Echo Cancellation
04:33.20jblackoh christ.
04:33.48jblackObviously, I'm having a bad day.
04:35.03jameswf-homew/o Echo Cancellation $499 w/ Echo Cancellation $699
04:35.07*** join/#asterisk Sparc__ (n=chatzill@c-76-126-162-83.hsd1.ca.comcast.net)
04:35.18jblackWhat do you generally recommend with polycoms?
04:36.04jameswf-homeI say no EC generaly not needed.
04:36.39Sparc__I'm using AMI's redirect command to redirect in-progress calls to a context that plays a recorded message. However when I do so, the end user never hears the first 3 seconds of the audio file being played. Can anyone point me to a solution to this problem?
04:36.51JTi say you should always buy a card with HWEC if interfacing the card to the PSTN :P
04:37.18drmessanoThe phones you're using are irrelevant..
04:37.27*** join/#asterisk HaMYaI (n=LAMER@ppp-58-8-10-181.revip2.asianet.co.th)
04:37.27MACscrwhich is more important to asterisk, cpu or ram?
04:37.34drmessanoEC is a PSTN issue and needs to be addressed
04:37.44drmessanoMACscr: All of the above
04:37.50MACscrfor a sip only system
04:37.53drmessanoMACscr: All of the above
04:37.57JTindeed
04:38.12JTecho is induced mainly by poor far end analogue pstn lines
04:38.17MACscrjust curious as im about to set it up on a p3 733 with 512mb ram
04:38.20JThaving a polycom won't help
04:38.34JTyou need hwec especially over half a dozen channels
04:38.47drmessanoJT: That's not entirely true
04:38.55MACscronly planning on 1-3 simultaneous sip calls though
04:39.00drmessanoIf you don't have EC, you can get by with some Grandstreams ;)
04:39.14JTheh
04:39.15jblackLooks like the r1t1 does voltage detection/
04:39.40JTimho they should never have released any cards without hwec
04:39.43JTat least not now
04:39.46JTbut it's a cash cow
04:40.19drmessanoLook Ma, I found a 600ohm purely resistive telco line.. and the holy grail.. and a cure for cancer
04:40.22JTif you buy a sip gateway like a cisco AS5400 or quintum, etc, hwec is NOT an option
04:40.24jblackOk, so a r1t1 with ec.
04:40.36JTit is standard in all decent sip gateways
04:40.37jblackWe're talkign all of two hundred bucks of difference.
04:41.19*** join/#asterisk [hC] (n=turnerd@216.251.157.146)
04:42.47[hC]I thought there was a way in queues to stop sending calls to an agent if they havent answered a call in a while, but I cant find it.
04:42.52*** join/#asterisk BugKhaM (n=LAMER@ppp-58-8-10-181.revip2.asianet.co.th)
04:43.26BugKhaManyone knows if Nokia E Series work with *?
04:45.38*** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net)
04:46.01[hC]PermitRootLogin no
04:46.01[hC]PasswordAuthentication no
04:46.01[hC]PermitEmptyPasswords no
04:46.07[hC]Woops, sorry!
04:48.14jblackOk. The r1t1 is a 3.3v card. ;)
04:51.05JTwhy not buy a universal voltage card? :D
04:52.49jblackI _think_ it's a 3.3 card. It also looks similiar to dual voltage cards.
05:00.15jblack~pcihelp
05:01.02jblackjbot: pcihelp is at www.digi.com/pdf/prd_msc_pcitech.pdf
05:01.02jbotjblack: okay
05:03.56jblackOk. The r1t1 is a 32bit, that can do both 3.3v and 5.0v. That is my final answer.
05:04.27jblackIn simpler terms, you can plug the r1t1 into anything made post... oh perhaps 1995
05:08.14*** part/#asterisk keith4 (n=kbe2@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com)
05:11.27jameswf-homeheh http://wylfwt.com/home/daily-articles/worst-motivational-posters/slap
05:12.46jblacklol
05:13.16jblackI should check into sending a large print of that to my ex-wife
05:14.09jameswf-homeheh http://img339.imageshack.us/my.php?image=cosgw7.jpg
05:14.28jblackjameswf-home: btw, thanks for the help earlier
05:14.32jameswf-homenp
05:14.48jameswf-hometell my boss I sent you maybe he will get me something nice :)
05:15.16jblackI'll tell him nicer than that.
05:18.40jblacktries to decide what to do next; ldap, or realtime *
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06:42.18MACscrwhat's the best way to troubleshoot why a sip trunk isn't registering?
06:42.37MACscrjust try making a call with that trunk and then watching the cli?
06:42.40*** part/#asterisk kamanashisroy (n=kamanash@202.56.7.133)
06:42.48MACscror is there a way to initiate the registration from the cli
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06:52.35*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)
06:53.19joelsolankiHi room
06:53.27joelsolankimy pbx was working perfect until yesterday.
06:53.34joelsolankibut today suddently it has stop accepting g729 calls
06:53.41joelsolankii have 6 g729 licensed installed
06:53.49joelsolankiwhen i make call from xlite with g729 codec it tells me 499 not acceptable here
06:53.59joelsolankiIn trunks g729 is allowed, and in extension all is allowed.
06:54.06joelsolankiwhat could be the reason for this issue ?
06:54.35joelsolankiit clearly looks like it is something with codec. but this was working alright from last 6 months.
06:54.47joelsolankinot able to figure out. any clues plz
06:56.11*** join/#asterisk shinao1 (n=shinao1@41.222.65.165)
06:56.13alrsjoelsolanki: if you're running 1.4 you can get an idea if g729 is working with "core show translation" at the asterisk console prompt
06:59.04joelsolankii m using 1.2 series
06:59.34joelsolankii m doing show translation
06:59.39*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
07:00.17joelsolankihttp://www.pastebin.ca/959207
07:00.24joelsolankido u think anything is wrong ?
07:05.19alrslooks OK there.
07:05.37alrstry allowing only g729 and disallowing everything elese
07:06.06joelsolankihmm.
07:06.14joelsolankitrying..
07:06.16alrsthere's always the chance that Asterisk is doing fine, and that X-Lite "updated" itself to a broken state
07:06.24*** join/#asterisk vicram (n=vr@bb219-74-56-253.singnet.com.sg)
07:06.57joelsolankihmm
07:07.11*** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net)
07:07.24joelsolankilet me try from cisco ata. so i can know if xlite is creating problem
07:11.32joelsolankiset g729 in trunks and extensions but same 499 not acceptable here
07:11.32joelsolankiconfiguring ata and trying
07:15.31*** join/#asterisk Dextorion (n=dex@fingerbottom.tekproj.bth.se)
07:16.21Dextorionasterisk have a telnet open for remote admin. Which works great. But, is there a ssh connection aswell? Would be nice to do the same over ssh.
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07:17.51Nuggetssh to the machine and run asterisk.
07:18.45Dextorioni cant. and thats not the same as the telnet session, which asterisk runs on its own.
07:18.58Nuggetperhaps I misunderstand what you're talking about, then.
07:19.25Nuggetis that something new in 1.6?
07:19.26Dextorionah. :)  Asterisk is running a own telnet connection. Where you can connect and remote control the asterisk server.
07:19.56Nuggettelnet like on port 23 and all that?  or are you just using telnet to mean something else?
07:20.05Dextorionhm. dont think so.   Anyways, im connecting to that telnet session from an application.  I would love to do the same thing, but over ssh.
07:20.16alrsAMI
07:20.18Nuggetdo you mean the manager api?
07:20.30Dextorionright! :) thats the one i guess.
07:20.39Nuggetyou could wrap it with stunnel I suppose.
07:21.56Nuggetor with ssh, for that matter.  :)
07:33.21joelsolankii tried with ata.
07:33.30Dextorionthanks Nugget.  It looks like 1.6 have ssh connection aswell..hm
07:33.30joelsolankicall went but blank voice.
07:33.41joelsolankiand when i checked sip show channels it went with ulaw
07:33.51joelsolankithough my ata is configured with g729
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09:53.26loompekumm.. asterisk-addons-current
09:53.55loompekhttp://rula.net/407
09:54.15loompeki'm using mysql binary mysql-5.0.45-linux-i686-glibc23/
09:54.24loompekany ideas what to do?
09:54.49*** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com)
10:04.11mvanbaakloompek: do you have asterisk-current installed ?
10:04.35mvanbaaklooks like version mismatch between asterisk and -addons
10:05.26loompeki've got latest ...
10:05.27loompekumm
10:05.42loompekAsterisk 1.6.0-beta6
10:06.02mvanbaakthen I have no idea
10:10.51*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
10:20.56hi365whats the "proper" way to forece a polycom to reread the cfg/directory files?
10:23.22*** join/#asterisk Hyphenex (n=Hyphenex@60-241-72-151.tpgi.com.au)
10:23.46Hyphenexhey gang.  Just wondering how many ways there are to connect my Asterisk server to a PSTN phone system?  and what would be the cheapest?
10:24.34tzafrirThe cheapest would be an X100P card, I guess
10:25.12Hyphenexhow would one use this X100P card?  (would it work on an old system like a P3 733Mhz 128MB ram sort of thing)
10:26.54tzafrirsure
10:27.01tzafrirBTW: do you happen to have ISDN?
10:27.37Hyphenextzafrir: Nope, no ISDN.  This thing is just going to be used for a conference with my cousin overseas and my aunt though
10:27.59HyphenexI'll only need one outbound line on my PSTN phone :)
10:31.49tzafrirHyphenex, another option is for you to call through VoIP
10:32.18tzafrirAsterisk is a bit of an overkill for your needs (if this is all you need)
10:32.27Hyphenextzafrir: the overseas line of VoIP (my VoIP provider only gives me one outbound, and my bandwidth is kind of limited)
10:33.00Hyphenexand I've already got my Asterisk server up and running (yeah, for one phone :P) I just want to know how to expand it! :)
10:33.08tzafrirSomething interesting Asterisk could allow you is for your cousin to connect to conference rooms on your server :-)
10:33.40Hyphenextzafrir: I was thinking of dialing up my cousin, then transfering her to a conference channel :)
10:33.50Hyphenexthen calling up my aunt and transfering her
10:34.32tzafrirHyphenex, then you should research voip, I guess
10:37.14Hyphenextzafrir: I've got an asterisk server I've already set up :P
10:37.29*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
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10:40.27ice_crofthi all
10:41.05ice_croftby which algorithm aster generatin channel-id ?
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10:48.47*** part/#asterisk dominic1 (n=dob@213.221.82.242)
10:55.12HyphenexAre there any devices that sit on a network and bridge the SIP/PSTN networks?
10:56.22*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
10:56.40EmleyMoorIs there a simple way to tell if a channel has already been Answer()ed?
10:56.44*** join/#asterisk steliosk (n=Stelios@79.131.104.95)
11:01.24tzafrirHyphenex, those are generally called ATAs
11:01.26tzafrir~ata
11:01.27jbotit has been said that ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
11:01.51HyphenexCool.  are they expensive?
11:02.11tzafrirDepends
11:02.40Hyphenexoahh, looking at the wiki, I don't think that's what I want
11:02.52tzafrirThough for obvoius reasons I'd prefer that others would answer related questions
11:03.03*** join/#asterisk NoamRotter (n=noam@mail.browarnik.com)
11:03.22Hyphenexlike, I want to be able to pick up my VoIP phone, have the dialplan pick up _9. then forward that to my PSTN network somehow
11:05.35EmleyMoorHyphenex: Are you saying you want to be able to add a PSTN line to your Asterisk box?
11:12.50Hyphenexyeah, That's what I want to do :)
11:13.51EmleyMoorHyphenex: In that case, you need to add an FXO port - there are several ways - I use a card but there are ATAs that can do it too
11:14.47Hyphenexwhat's the cheapest/best?
11:16.15EmleyMoorFor ease of setup, a TDM410 or 800 card with an FXO module would do it (I have a 400). It might be easier depending on your wiring to use an ATA but I don't actually know much about them.
11:17.00*** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller)
11:17.20Hyphenexahhk, I'm only connecting one external line though (like, I just want to hook it into my PSTN phone port)
11:18.31*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
11:19.29*** part/#asterisk ice_croft (n=nolan@85.172.5.106)
11:19.32EmleyMoorHyphenex: You can get those cards with just a single module. There are or were some single-port FXO cards but I am not sure how readily available they are now.
11:19.46EmleyMoor(X100P clones?)
11:20.06HyphenexI don't know where I'd find any
11:20.12HyphenexE-Bay is not looking promising
11:21.41*** join/#asterisk Treytor (i=PJIRCWeb@ip68-4-124-32.oc.oc.cox.net)
11:21.49Treytoralright, hi guys
11:21.54Treytorbrand new to asterisk here
11:21.58Treytorgot a trixbox installation
11:22.14TreytorI think I have it working, but I don't have a SIP phone to test it with
11:22.20Treytorhow do I dial out?
11:22.28Treytorstupid question, I know :(
11:22.51HyphenexEmleyMoor: just a thought... why can't I use a dial up modem?
11:22.52Treytorpretty sure I got the trunk and route set up correctly
11:23.50Treytoranyone?
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11:27.46EmleyMoorHyphenex: Dial up modems are simply not capable of it, unless they are X100P compatible
11:29.10HyphenexEmleyMoor: but with some dial up modems, you can still use the computer as a softphone
11:29.45Treytor...
11:32.17*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
11:32.48HyphenexOoo, this might be what I want http://voip-warehouse.com.au/digium-asterisk-tdm400p-p-4531.html
11:33.03EmleyMoorHyphenex: Unless they've provided a way to use it as an FXO with Asterisk, you are a little stuck
11:33.40EmleyMoorHyphenex: That is a base board with no modules - you need an FXO module to go with it
11:34.09EmleyMoorThe X100M module shown does that
11:35.00Hyphenexso I need to buy two things?
11:35.06HyphenexI can see this is not going to be cheap :(
11:35.44EmleyMoorHyphenex: X100P clones may be relatively cheap but how good they will be for you is another matter
11:36.47Hyphenexyeah, I don't know why there so expensive.  there just like a sound card with something to detect the 45V arn't they?
11:37.35tzafrirHyphenex, they originally cost ~10$ or even less
11:37.53tzafrirThey are basically a modem
11:38.07Hyphenextzafrir: I've heard bad things about the cheap ones and echoing though
11:38.38tzafrirfor a single channel you can use software echo canceller (e.g: OSLEC)
11:38.51Hyphenexon my 733Mhz machine?
11:38.54tzafrirsure
11:39.05Hyphenexthat might be something worth looking into then
11:39.12Hyphenexwhere might I find one of these ~$10 things then?
11:39.44tzafrirThis may not be worth saving on a professional installation. But for a home setup - sure
11:39.55tzafrirnot sure
11:40.10tzafrirI bought mine on ebay
11:40.15Treytorguys, can I please get some help here?
11:40.23tzafrir~ask
11:40.24jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
11:40.54Hyphenexhmm, ok
11:40.55Treytoryes, thank you.  Is it against the rules to copy / paste what I've already asked?
11:41.16HyphenexI might look into it tomorrow.  Thanks tzafrir
11:41.18tzafrirah, haven't noticed that
11:42.01*** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca)
11:42.04tzafrirTreytor, dial out to where?
11:42.08Treytoranything
11:42.10Treytormy cell phone
11:42.15Treytorsomething to see if it works
11:42.25tzafrirbut this is basically a trixbox / /freepbx question
11:42.31TreytorI just want a box to take web requests and dial out
11:42.43Treytoroh, so asterisk wouldn't know?
11:44.00tzafrirYou want to originate the call from the SIP phone (simpler) or from a web interface (much less trivial)
11:44.25TreytorI'm sorry if I'm confused, I'm barely going on 5 hours here using any sort of asterisk / trixbox / freepbx
11:44.30Treytorhell, any linux distro
11:44.46TreytorI feel I am way over my head, but I'm learning
11:45.34EmleyMoorTreytor: Do you have any other computer on which you could run a softphone?
11:45.52Treytorsure, I'm using one now.  I don't know how to do that, though :(
11:45.56Treytoralso, this is running vista
11:47.54EmleyMoorwww.voip-info.org - look up the softphone and see if there's any Asterisk configuration instructions for it
11:48.07tzafrirTreytor, I forgot the '?' in the end of my last sentense
11:48.45Treytorit's okay, I forgive you just this once.
11:49.46Treytorwhich soft phone software package would you recommend?
11:49.56Treytorminisip.org?
11:50.27Treytorno...
11:50.36EmleyMoorX-Lite is popular and not bad
11:53.36Treytorokay, now do I have to set this up in freepbx?
11:54.21EmleyMoorTreytor: Yes - it needs a SIP account on the freepbx and a dialplan
11:54.48*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
11:55.51bluregardhello all
12:06.36Treytorhmm
12:06.36TreytorI can't get the softphone to connect
12:06.36Treytorto my asterisk server
12:06.36hi365core show hints shows the status of all extensions as idle even when there in use. is there a setting for this somewhere?
12:08.11*** join/#asterisk lirakis (i=lirakis@66.252.24.133)
12:09.12*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
12:14.01*** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq)
12:26.23*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:29.15*** join/#asterisk stony (n=oloch@p57B39B9C.dip0.t-ipconnect.de)
12:30.30stonyi'm looking for a script that tells me if the callerid is international, national or even just a local call
12:30.42stonyhas anyone ever seen such a script ?
12:39.50*** join/#asterisk sbrobou (n=sbrobou@201.48.5.105)
12:40.05*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
12:40.17sbroboudoes asterisk work with megaco?
12:40.39[TK]D-Fendersbrobou: * can work with MGCP phones, but not servers
12:41.00[TK]D-Fenderstony: Nope.  This looks like a simple few # of Gotoif's on your part.
12:41.24*** part/#asterisk airjump (n=zielonka@62.159.95.82)
12:43.15sbroboui never worked with megaco/h248. Do you know about its complexity? It is as hard as h323? or easier like SIP?
12:44.01sbroboui can bet that it is harder than h323. ITU-T looks like a hell.
12:44.39[TK]D-Fendersbrobou: Avoid if you know whats good for you (as far as * is concerned)
12:47.37stony[TK]D-Fender: it's not that easy ...
12:48.00stony[TK]D-Fender: 'cause you don't know if this is a internatinal or local number: 49320394854
12:48.08[TK]D-Fenderstony:well get ready to make a database then.
12:48.30[TK]D-Fenderstony: And how WOULD you know then?  You jsut said that # can't be determined.
12:48.52[TK]D-Fenderstony: Don't tell us something is impossible and then ask us how to do it.
12:49.06sbrobou:) i will die.... rrrrrrrrrrrrrrrrrrrrr
12:49.51*** join/#asterisk shido6 (n=shido6@204.126.120.132)
12:50.42padoes anyone have an unofficial ubuntu package for zaptel-source?
12:50.45stony[TK]D-Fender: i was hoping someone else did the trick :D
12:50.48[TK]D-Fendersbrobou: Please stand on that plastic sheet over there then so you don't make a mess...
12:51.02pai saw the ubuntu version does not compile with kernels >=2.6.23
12:51.05[TK]D-Fenderstony: If YOU don't even know how to tell, why should we?
12:51.14sbrobouehehhehehe :)
12:51.22[TK]D-Fenderstony: And of course you didn't even tell us where you were located.  Go ask someone local.
12:51.23stony[TK]D-Fender: 'cause you are the gurus aren't you ?
12:51.36stony:D
12:52.07shido6if you provide the hat we might be able to produce the rabbit but without the hat... you're going to see finger puppets and a Kmart light
12:52.33stonylol
12:52.44*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
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13:06.19bluregardWay OT and I'll probably die a horrible death for asking, but has anyone eval'd Win2k8 Server?
13:06.51[TK]D-Fenderbluregard: I'm looking at converting our ancient Novell5 server to it shortly...
13:07.28*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:07.28*** mode/#asterisk [+o lmadsen] by ChanServ
13:08.31bluregard[TK]D-Fender, because it's honestly an improvement over 2k3 or because MS is going to be forcing it to market
13:08.41*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
13:08.46Kattylmadsen: you didn't leave me a voicemail :<
13:08.58Kattyshakes fist
13:09.11[TK]D-Fenderbluregard: Because its current and Novell5 is ANCIENT
13:09.26lmadsenKatty: I wasn't getting any audio
13:09.27bluregard[TK]D-Fender, fair enough
13:09.33Kattylmadsen: oh right.
13:09.34[TK]D-Fenderbluregard: Or I could live with 2K3 w/ upgrade covered in the cost
13:09.36Kattylmadsen: i forgot
13:09.42*** join/#asterisk airjump (n=zielonka@62.159.95.82)
13:09.49[TK]D-Fenderbluregard: Basically I want to consolidate my file-serving.
13:09.51lmadsenKatty: did you happen to try adding nat=yes to your general section? I could try again to see if I get audio after that
13:11.41*** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb)
13:12.58*** join/#asterisk bsaxon (n=bsaxon@66.0.66.4)
13:14.05Kattylmadsen: nat=yes has been there.
13:14.13lmadsenoh ok... stupid polycom then :)
13:14.14Kattylmadsen: you can call again if you want tho
13:14.15bluregardsome of the improvements over 2k3 do _sound_ nice
13:14.28lmadsenoh ringing this time!
13:14.42bluregardNAP for one
13:15.11lmadsenw00t audio
13:15.21Kattyhorays, audios
13:15.25Kattysorry i cut it short.
13:15.33Kattypeople were all lolzrmathiskthxbi
13:15.37*** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp)
13:16.22lmadsenno that's fine :)
13:16.32lmadsenI figured it was going to be a very quick call if it did work, heh
13:16.33*** join/#asterisk [intra]lanman (n=lanman@va-76-6-209-179.dhcp.embarqhsd.net)
13:16.40Kattyinv_arp: GET OUT
13:16.41Kattyoh
13:16.44Katty[intra]lanman: GET OUT
13:16.51[intra]lanmanwtf? i just got here
13:16.55Kattyhehehehe
13:16.57Kattyhugs [intra]lanman
13:17.01[intra]lanmanawwww
13:17.05[intra]lanmanhugs Katty
13:17.07Kattyvaults already made me hyper.
13:17.22[intra]lanmanvaults? you robbed a bank already?
13:18.09Kattywww.drinkvault.com
13:18.27Kattyi've had Red Bull (1), Vault Zero (1), and have a Monster Energy Drink (1) still to drink.
13:18.37[intra]lanmanoooohhh, those vaults
13:18.45bluregardis it me or does Vault taste like the old Surge?
13:18.51Kattybluregard: it so does.
13:18.59bluregardI thought so
13:19.03Kattybluregard: it also reminds me of GreenWorks by Colorox
13:19.10[intra]lanmanvault is good with vodka
13:19.16[intra]lanmani call it a safecracker
13:19.17Kattyredbull is good with vodka
13:19.22Kattykeeps me from fallign asleep when drinking
13:19.27lmadsenKatty: I just use good ol' fashioned green tea :)
13:19.36Kattylmadsen: to clean your floors?!
13:19.45lmadsenpffffffft... I was doing redbull+vodka before it was hip
13:19.54Kattylmadsen: go back to bed you insanity mumbling sleepyhead!
13:20.02lmadsenI just drink rye and gingers now
13:20.07Kattyrye?
13:20.13lmadsenrye whiskey
13:20.14Kattyyou must be canadian.
13:20.16lmadsensorry... canadian thing
13:20.17lmadsen:)
13:20.23Kattys'ok.
13:20.24[intra]lanmanhahah
13:20.27Kattyi know plenty of canadians.
13:20.27[intra]lanmanKatty: good call
13:20.30lmadsenrye and gingers are liquid candy
13:20.35Kattyfile..
13:20.36Kattyjunky
13:20.42lmadsenjsmith
13:20.45Kattyjunky's kinda hard to understand sometimes
13:20.45lmadsen:)
13:20.57Kattyme and him have had good times tho
13:21.01lmadsenhe has the thick french canadian accent :)
13:21.21lmadsen[TK]D-Fender is also canadian
13:21.29Kattyyeah
13:21.32Kattybut he doesn't count
13:21.34*** join/#asterisk jsmith (n=user@72.21.36.138)
13:21.34*** mode/#asterisk [+o jsmith] by ChanServ
13:21.36Kattyhe acts french.
13:21.42Kattyjsmith: hai
13:21.42[TK]D-Fenderran out of fingers & toes....
13:21.45*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:21.47jsmithhears rumours that he's Canadian now
13:21.51Kattyhugs _ShrikE
13:21.56jsmithAll hail the president of Canadia!
13:21.57*** join/#asterisk d3wayne (n=deeewayn@76.29.245.9)
13:21.57*** mode/#asterisk [+o d3wayne] by ChanServ
13:21.58Katty[TK]D-Fender: are you less grumpy today?
13:22.07jsmith~lart lmadsen
13:22.07jbotsends a legion of lawyers after lmadsen's head
13:22.11*** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net)
13:22.18lmadsenjsmith: lol... you mean Prime Minister! :)
13:22.30jsmithlmadsen: Naw, I mean Queen.
13:22.36lmadsenknows that jsmith knows that he know that jsmith knows he's just joking to get lmadsen all riled up :)
13:22.40jsmithlmadsen: After all, she's still on your money and stamps.
13:22.40[TK]D-FenderKatty: You really read that wrong yesterday BTW... wasn't "grumpy", but that would mean you were mistaken and you're quite simply nto ready to accept such a silly notion :p
13:22.45lmadsenjsmith: heck ya
13:22.55jsmithlmadsen: Canadia... isn't that the 51st state... up there north of Minnesota?
13:22.56lmadsenhugs the queen
13:23.04lmadsenjsmith: ya, the one with all the oil
13:23.13Katty[TK]D-Fender: okay. so perhaps grumpy is the wrong word.
13:23.14jsmithlmadsen: And all the whiny singers?
13:23.19Katty[TK]D-Fender: but you had a very snippy attitude!!!
13:23.21lmadsennah, that's Las Vegas
13:23.30lmadsenKatty: oh that's normal
13:23.35jsmithlmadsen: You can have her back... in fact, *please* take her back
13:23.39lmadsenno way
13:23.42Kattylmadsen: i know, but usally i'm not at the recieving end of said snip.
13:23.44jsmithlmadsen: I refuse to visit Vegas until she goes back to Canada
13:23.48lmadsenwe got rid of her and give-backs
13:24.01[TK]D-FenderKatty: Just jealous because I run at about 100% any time I'm conscious, heh
13:24.03lmadsenjsmith: meh... LV isn't all it's cracked up to be
13:24.11*** join/#asterisk anonymouz666 (n=anonymou@201.19.122.138)
13:24.16*** join/#asterisk siya (n=djerk@194.60.207.239)
13:24.18siyare
13:24.23lmadsener
13:24.31jsmithlmadsen: Amen, brother!
13:24.32Katty[TK]D-Fender: your brains gonna sizzle.
13:24.48Katty[TK]D-Fender: SIZZLE AND SPLODE
13:24.50jsmith[TK]D-Fender: Don't worry... I've got a case of scrambled brains today too
13:25.00[TK]D-FenderKatty: Lightly fried, with bacon, sausage, has browns...
13:25.03[TK]D-Fendermmmmmmm breakfast
13:25.07Kattyhas browns?
13:25.08siyaanyone who can tell me why a compile of *-1.4 fails with ilbc, where it used to compile fine a while back?
13:25.12Kattyhasbeen browns?
13:25.14[TK]D-Fender"hash browns"
13:25.15siyadebian etch
13:25.16Kattythose aren't nummy.
13:25.22Kattyhash browns are nomable tho.
13:25.28Kattyif not too greasy
13:25.32Kattyand made with EVOO
13:25.33lmadsenbreakfast sounds like a great idea
13:25.40x86indeed
13:25.41lmadsenI think I might even shower and put on some pants
13:25.43[TK]D-Fendersiya: PASTEBIN is your friend.
13:25.46x86lmadsen: what are you buying for us? :)
13:25.50Kattylmadsen: horay for shower!
13:25.57Kattylmadsen: i had one of those at 6:30am (=
13:25.58lmadsenKatty: indeed :)
13:26.05lmadsenKatty: there's a 6:30am now?
13:26.13lmadsenI saw one once... but only when I stayed up really late
13:26.13Kattylmadsen: oh yes. it's all the rage this season.
13:26.17siya[TK]D-Fender: ok, will do in a min
13:26.26lmadsen'rage' being the key word :)
13:26.35Kattylmadsen: i am required to wake up that early so i don't snap at everyone at work at 8am.
13:26.42lmadsenheh
13:26.52Kattylmadsen: and freshly awoken katty is a dangerious creature indeed.
13:27.24lmadsenKatty: I like girls like that, always a challenge
13:27.36Kattyyou're mad.
13:27.59lmadsenya, you're not the first to say that
13:28.06lmadsenfeisty girls are my specialty
13:28.11lmadsenjust ask jsmith
13:28.35jsmithKatty: Stay away from lmadsen... he has a habit of having his heart broken by feisty girls
13:28.41lmadsenit's true
13:28.52Kattyjsmith: oh, don't worry.
13:28.57Kattyjsmith: i have shiny already (=
13:28.57siya[TK]D-Fender: I don't need to have some libilbc installed right?
13:29.14siyaI noticed that debian had removed it from the repository due to licensing
13:29.15[TK]D-Fendersiya:  Quite possibly.
13:29.25siyathat would explain it then
13:29.27jsmithsiya: If you're loading the very latest version of Asterisk from SVN, then yes you do
13:29.39jsmithsiya: There's a script in the contrib/ directory to download it and install it
13:29.53lmadsensiya: ya, and Asterisk will be removing ilbc from the main distro in the next release because of that too
13:30.11lmadsen(and from latest SVN as per jsmith)
13:30.27x86ilbc sucks anyway though
13:30.49lmadsenheh
13:30.53lmadsenya, I've never bothered to use it
13:31.05Kattyhey how's everyone backing up their server?
13:31.07Kattyexternal hd?
13:31.15jsmithKatty: rdiff-backup
13:31.28Kattyjsmith: what are you dumping to?
13:31.34jsmithKatty: Another box
13:31.35lmadsenwhat's a backup?
13:31.38Kattyah, right. k
13:31.52*** join/#asterisk ManxPower (n=manxpowe@5.sub-70-223-0.myvzw.com)
13:32.00Kattylmadsen: that thing people in walmart do, in their little motorized carts. BEEP BEEP BEEP
13:32.02bluregardKatty, DDS4 and LTO2
13:32.02jsmithlmadsen: You subscribe the Linus Torvalds method?  You know... make all your stuff freely available and have the whole world mirror your work for you?
13:32.06Kattyhugs ManxPower
13:32.33ManxPowerhands Katty a mouse.
13:32.46Kattygosh.
13:33.07Kattygives said mouse cheese and a cage full of fluffy bedding
13:33.30*** join/#asterisk bminish (n=bminish@78.152.253.119)
13:33.37Kattyi don't think my ferrets are gonna like the mouse :<
13:35.08lmadsenI think they'll like him just fine :)
13:35.50x86Katty: HP MSL2024 robotic tape library w/ (1) LTO4 tape drive
13:36.02*** join/#asterisk quigon (n=matias@200.61.187.185)
13:36.06Kattygasps
13:36.06x86Katty: my tape library can hold 38.4TB worth of tapes
13:36.16Kattytapes? :<
13:36.18Kattytapes hate me.
13:36.23Kattythey EAT MY SOUL
13:36.24Kattyand my data
13:36.35x86I use backup exec, so no biggie
13:36.52*** join/#asterisk SwK (n=SwK@70-46-62-90.atl.fdn.com)
13:37.05bluregardx86, are you using Veritas' version or the new Symantec one
13:37.08Kattyokay this is weird.
13:37.16Kattymy ftp MAC.xml directory thingy
13:37.22Kattyonly let's me put 46 entries
13:37.22x86I've got a windows NAS (HP AiO1200 9TB), and I put a disk image with an xfs file system on it, and I mount that disk image remotely and rsync to it, the whole image gets backed up in full every night
13:37.29x86bluregard: symantec
13:37.33Kattyis 46 entries the max?
13:37.36*** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83)
13:37.41x86bluregard: the newest one too... we just upgraded to 12
13:37.42ManxPowerMany people backup data, not nearly as many are able to RESTORE data.
13:38.02x86ManxPower: I test our backups once a quarter
13:38.13ManxPowerx86: A good idea.
13:38.18Katty[TK]D-Fender: do you know if 46 entires is the max number in MAC-directory.xml?
13:38.22x86indoubedebly :P
13:38.58[TK]D-FenderKatty: No, it isn't.  I think the limit was 100 or 200
13:39.04Katty[TK]D-Fender: dangit.
13:39.11Katty[TK]D-Fender: it's not showing my 47th entry :<<
13:39.12[TK]D-FenderKatty: Check the admin guide
13:39.22[TK]D-FenderKatty: Showing where?
13:39.47Kattyin the Speed Dial Info on my IP501
13:39.51ManxPowerKatty: even a very minor typoe can mess things up.  Polycoms can be SUCH a diva
13:40.13[TK]D-FenderKatty: Well Speed-dial is differnt that jsut the directory...
13:41.06bluregardx86, are you guys using any kind of deduplication on that NAS?
13:41.12Katty[TK]D-Fender: are you telling me polycoms have a limitation of entries on the Speed Dial section?
13:41.16KattyManxPower: yeah, but i copied and pasted.
13:41.20KattyManxPower: three times
13:41.21KattyManxPower: :<
13:41.45ManxPowerKatty: The Admin Guide is your friend
13:42.03Kattyi've never needed an admin guide for the last 46 entries :P
13:42.21Kattyokay okay.
13:42.21Kattyfine.
13:42.26Kattylooks for admin guide
13:42.32ManxPowerKatty: then you are lucky
13:43.47*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
13:45.34*** join/#asterisk ZPertee (n=ZPertee@dhcp171-093.wireless.uakron.edu)
13:47.21*** part/#asterisk mocker (n=kyle@mocker.org)
13:47.23bluregardKatty, what SIP firmware are you using?
13:47.26*** join/#asterisk mocker (n=kyle@mocker.org)
13:47.39*** part/#asterisk jsmith (n=user@72.21.36.138)
13:47.39x86bluregard: what do you mean by deduplication? finding duplicate files?
13:48.12bluregardx86, yeah, not only duplicate files but duplicate bytes or blocks
13:48.13*** join/#asterisk ddunavant (n=David@75.145.240.14)
13:48.28x86bluregard: nope
13:48.30ZPerteehow can I have different features in different contexts?  For example I have 5 lines.  For call parking I want each line to go the same parking position.  Line 1 = 701, Line 2 = 702, and so on.  I want to keep the same key sequence to put the call on hold and the same one to pickup the call.  I use all pots lines
13:49.00x86bluregard: we've only got about 3TB to backup, and our dailys go to disk (Hitachi SANs)
13:49.22x86bluregard: weekly and monthly backups go to tape which are taken off site
13:49.30*** join/#asterisk ice_croft (n=nolan@85.172.5.106)
13:49.37ice_croftyo people
13:49.40x86bluregard: weekly backups are retained for 8 weeks, monthly backups retained for 1 year
13:49.45bluregardx86, just curious.  Some of the vendors are boasting 20:1 - 50:1 data reduction
13:49.51bluregardusing dedup that is
13:50.01ice_crofti've finnaly found a way to bill and acct * to radius!
13:50.27bluregardI imagine realistically though that it would be more like 10-15:1
13:50.35ice_croftsorry, auth and acct :)
13:50.35siyaright "Asterisk will be removing ilbc from the main distro in the next release"
13:50.57siyathen I'd better remove it from the list
13:51.07x86bluregard: yeah, we have no real need for that... we don't have enough data :P
13:51.42x86bluregard: our available storage capacity is about 40 times the amount we actually need
13:51.58bluregardx86, a luxury I'm not familiar with
13:52.02x86heh
13:52.41bluregardemail pack rats and all
13:52.44x86when I got in here, I ripped out all of the funky mess the previous guy had left them with, and I convinced them to let me rebuild it the Right Way(tm)
13:53.39bluregard(tm) MS Corp
13:54.03*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
13:55.01*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
13:55.02x86so I've got a pair of Hitachi SANs, an HP NAS, and 38.4TB worth of tape library...
13:55.21x86SANs are 12TB, NAS is 9TB
13:55.23filesiya: the source code won't be included for iLBC, but you can still download and use it yourself
13:55.44x86so we're just under 60TB of available storage, and we have about 3TB of data
13:56.02x86ok, so I said 40 times, but it's really more like 20 times ;)
13:56.43bluregardI had a 2TB CX200 and a 24 tape LTO2 library at about 89% capacity
13:56.44x86siya: you dont want ilbc anyway... use GSM :)
13:57.05x86LTO2 is like 200MB/400MB right?
13:57.12bluregardyeah
13:57.26x86our LTO4 tapes are 800GB/1.6TB ;)
13:57.53bluregardyou getting good reliability out of your tapes
13:58.12x86so far... of course, we've only had the system about 9 months now
13:58.32Kattybluregard: bootrom?
13:58.37siyax86: exactly
13:58.53*** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it)
13:58.54siyaI can always add it if I need it later due to some obscurity
13:58.57bluregardI must have gotten a bad lot of tapes, I had 12 LTO2 tapes up and refuse to write data one day
13:59.00*** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net)
13:59.28boblutzWhat is the reason you can only specify contexts and not extensions nor priorities in sip.conf ?
13:59.45bluregardKatty, yeah, nevermind.  I have a printed admin guide, but it's way out of date
13:59.54x86boblutz: because it would have no idea what context that extension was in?
13:59.56x86;)
14:00.30boblutzx86: True, I meant [context] or [context,extension] or [context,extension,priority]
14:00.36boblutzshrugs
14:00.39boblutzworks for me though!
14:00.57bluregardKatty, it's says <sd> </sd> takes a value between 0-40 but like I said, way out of date
14:01.17Kattybluregard: i changes some stuff in sip.cfg about dir.local stuff
14:01.28Kattybluregard: about the ammount of flash memory the dir can handle. i'm about to test again
14:01.45Kattybluregard: the admin guide says something about a limit of 99
14:01.51Kattybluregard: 47 is a long way from 99 (=
14:02.13bluregardKatty, slightly
14:03.02x86Katty: I always end up making a web directory for directories that uses the asterisk manager interface to dial your phone after you click on an entry
14:03.50x86Katty: i find it's much easier to point and click on my 24" widescreen monitor than to scroll through the list on the little screen of the Polycom IP501
14:04.19Kattyx86: that does sound a bit easier (=
14:04.40Kattyx86: i don't suppose you'd care to share it?
14:06.58*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
14:06.58*** mode/#asterisk [+o anthm] by ChanServ
14:07.23*** join/#asterisk ZPertee (n=ZPertee@dhcp171-093.wireless.uakron.edu)
14:10.49x86Katty: well it's part of a much bigger system... it's hella easy to write tho
14:11.03x86Katty: gimme ssh access and I'll write it for you on your system
14:12.54*** join/#asterisk skirmisha (i=skirmish@87-126-225-188.btc-net.bg)
14:13.04boblutzlol
14:13.12skirmishaguys why asterisk load part of modules even autoload is yes
14:13.19boblutzskirmisha: ?
14:13.30skirmishain modules.conf autoload=yes
14:13.37skirmishaand asterisk does not load all modules
14:13.45boblutzyea, that will load all modules you have compiled in /usr/lib/asterisk/modules
14:13.46skirmishai don;t see any error in log file
14:13.51boblutz(default location ^)
14:13.51*** join/#asterisk wmaulik (n=wmaulik@158.59.192.218)
14:13.57skirmishayes i see 140 modules there
14:14.02skirmishabut only 38 are loaded
14:14.03Kattyx86: :<
14:14.05ManxPowerskirmisha: Autoload does NOT mean "load all modules", it means "load the modules you need and any modules those modules need"
14:14.07*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
14:14.10Kattyx86: but i wanna know how to do it
14:14.25Kattyx86: blog it! (=
14:14.33skirmishaManxPower means?
14:14.47skirmishait does not even load chan_sip
14:14.55skirmishaonly iax
14:15.00boblutzskirmisha: Do you have a sip.conf ?
14:15.02x86Katty: you know Perl?
14:15.10skirmishaif i set chan_sip to be loaded in modules.conf then it is loaded
14:15.28skirmishaalso restart command from cli is not loaded, don;t know in which module it is
14:15.36skirmishayes i do sip.conf
14:15.43ManxPowerskirmisha: You installed from a package, didn't you?
14:15.51skirmishai tried to recompile asterisk and it is still same
14:16.00skirmishanot package
14:16.01skirmishafrom source
14:16.09ManxPowerskirmisha: you must be reading 1.2 docs and using 1.4 Asterisk
14:16.19skirmishait is 1.4.13
14:16.25skirmishanow i am upgrading to 1.4.18
14:16.25ManxPowerin 1.4 it's "core restart now"
14:16.35ManxPowernot "restart now"
14:16.35skirmishanope
14:16.52boblutz`core restart now` doesnt work for me
14:16.57boblutzusing 1.4.17
14:17.16ManxPowerboblutz: I thought ALL core commands started with "core"
14:17.22ManxPowerskirmisha: what does "help" give you?
14:17.26boblutzcore <tab> only yields "clear" "set" "show" for me
14:17.50ManxPowerskirmisha: what ERROR do you get when you try to do a restart
14:18.08ManxPowerboblutz: none of my servers use 1.4
14:18.27skirmishai don't have that option
14:18.33skirmishaonly reload
14:18.39skirmishaprobably module is not loaded
14:18.56boblutzi would imagine that is a core module that gets loaded regardless of modules.conf
14:19.17boblutz`restart [now|gracefully|when]` works in 1.4.17
14:19.22ManxPowerskirmisha: I won't ask a 3rd time.  What error message do you get?
14:19.23skirmishawhy it is loading modules partialy
14:19.33skirmishai see loader.c sees 140 modules
14:19.47skirmishaand then on cli i see only 48 loaded with show modules
14:20.03skirmishano error msg
14:20.07*** part/#asterisk wastrel (n=wastrel@nylug/member/wastrel)
14:20.08skirmishajust modules are not loaded
14:20.16skirmishacan it be file permisions
14:20.28ManxPowerpbx-1*CLI> freddiemac
14:20.28ManxPowerNo such command 'freddiemac' (type 'help' for help)
14:20.55ManxPowerTHAT is what you should see when you type an invalid command.  Since you are not getting an error message when you issue a "restart now", you have serious problems.
14:21.21ManxPowerskirmisha: put your /etc/asterisk/modules.conf on pastebin.ca
14:21.39boblutzthere better not be a noload in there
14:21.47lmadsenif you run restart now and it doens't restart, something is blocking
14:21.58lmadsenI've seen that happen, and sometimes it takes a little bit for it to go
14:22.12ManxPowerlmadsen: I call that a "soft crash" 8-)
14:22.13boblutz`restart now` disconnects me from *CLI>
14:22.17lmadsenheh
14:22.21ManxPowerboblutz: that would be expected
14:22.35lmadsenyes, unless you started asterisk in the foreground
14:24.16boblutzYea, just stating
14:24.52boblutzIs there a difference between "channel => 1-2" and "channel => 1 channel => 2" in zapata.conf
14:24.56boblutz?
14:25.08*** join/#asterisk Blackthorn (i=blacktho@76.77.160.10)
14:25.12boblutz(imagine the second quotation is on 2 lines
14:25.51boblutzhmm ... perhaps "channel => 2" will over write "channel => 1" ?
14:25.53ManxPowerbob yes and know, it depend son your config
14:26.03ManxPower..er... "yes and no"
14:26.07boblutzha
14:26.36ManxPowerif they are right after each other with nothing inbetween, then they should be the same
14:26.44boblutzok
14:26.57BlackthornWhen starting asterisk -vvvv I get an error "chan_zap: unkown signaling methond "pri_cpe". Was told that most likely libpri wasn't installed/correct. So I started new removing the source directories and recomplied the libpri and then asterisk but still does the same.
14:27.45ManxPowerBlackthorn: It sucks to be you.
14:27.55tzafrirBlackthorn, ldd /usr/lib/asterisk/chan_zap.so | grep libpri
14:28.05tzafriryou'll see nothing -> no libpri support
14:28.21*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:28.32tzafriranyway, did you re-run ./configure in the asterisk source directory after installing libpri?
14:28.41Blackthornok it says no such file or directory...
14:28.48Blackthorntza: yes
14:29.07tzafriroops: /usr/lib/asterisk/modules/chan_zap.so
14:29.28skirmishastill same after upgrade of asterisk
14:29.28*** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)
14:30.04Blackthornwhen i run the modfied caommdn it shows "p libpri"
14:30.08Blackthorncommand
14:31.00Blackthorni do show the chan_zap.so in the directory /usr/lib/astersik/modules
14:31.00skirmishawhat is main module in asterisk
14:31.12Nasra~bood
14:31.16Nasra!book
14:31.22ManxPower~book
14:31.23jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
14:31.28Nasrathanks
14:33.35Blackthornwhen I do the menuselect and go into modules it shows that chan_zap is selected and is available
14:34.04*** join/#asterisk m4sk4r4 (n=m4sk4r4@20150237020.user.veloxzone.com.br)
14:34.35Blackthorndo i need to re-comple zaptel in order to get this to work?
14:35.54*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
14:38.12boblutzIs Tiger Jet crap hardware ?
14:38.14ManxPowerBlackthorn: this system did not originally have a packaged Asterisk, did it?
14:38.21ManxPowerboblutz: YES! YES! YES!
14:38.29boblutz@#$!!!!!
14:38.47ManxPowerboblutz: It seems to be the cause of most compat issues with Digium cards (modern digium cards use a different chipset)
14:38.55boblutzwtf
14:39.32boblutzmy boss has been buying cards from a reseller - is it possible they are putting the cheap 10 dollar fxo modules on the card ?
14:39.35ZPerteein features.conf I am trying to setup call parking?  I use all pots lines and I want to use the same key sequence to put a call on hold for all of the lines.  however if I call is put on hold (by pushing *25) on line one then I want to be able to pickup the line  (*50).  If the call is put on hold on line one I don't want line two picking up the call and vice versa.  how does that work?
14:39.39*** join/#asterisk daniev (n=ganbarim@201.245.223.90)
14:39.42*** join/#asterisk javar (n=javar@69.79.134.24)
14:40.12ManxPowerboblutz: is it a Digium card?
14:40.21ManxPowerand the tigerget is the PCI controller on the base card
14:40.31boblutzit says digium
14:40.34ManxPowerI believe the TDM400P still uses Tigerjet
14:40.45boblutzManxPower: Yea, lspci showed the tdm400p as tigerjet
14:41.01ManxPowerboblutz: That would be expected.
14:41.32*** join/#asterisk RoyK (n=roy@ip-108-27-149-91.dialup.ice.no)
14:41.38boblutzx100p is the cheap pci card....x100m is a module placed on a tdm card for fxo functionality ? [confirm/deny]
14:42.09EmleyMoorconfirm
14:42.24Kattybluregard: i found out the probably with my directory.
14:42.39bluregardKatty, what was it
14:42.45Kattybluregard: the directory button on the IP501 takes you to the directory...and everything is there
14:42.57Kattybluregard: the up button, which goes to speed dial, only shows the top 46 of the directory
14:43.05bluregardaah
14:43.36bluregardKatty, are you using 4.0 bootrom
14:43.50Kattybluregard: no
14:43.54Kattybluregard: 3 something
14:43.54Blackthornmanx: nope, it was a clean install. I pulled al the files from the digium mirrors. In fact I even wrote a guide how exacly i went about it located at http://www.voip-info.org/wiki/view/Asterisk%3A+Installing+Asterisk+on+Ubuntu+7.10+Server+Edition
14:44.22ZPerteedoes nobody no the answer to my call parking question?
14:44.42bluregardKatty, I'm still on 3.2.2 SIP 2.1, I'm debating whether or not to update to 4.0 and SIP 2.2 or 3.0 if I can get my hands on it
14:44.49ZPerteeI mean "know the answer"
14:45.37ManxPowerZPertee: the answer is "it doesn't work"
14:46.23ZPerteeManxPower, what do you mean by it doesn't work?  seems pretty elementary put a call on hold pickup the call later????
14:46.26ManxPowerZPertee: You would want to transfer the call to a parking slot, then pick up the call from another phone.
14:46.27*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:47.03ManxPowerZPertee: hold/pickup is a key system feature and not a feature of Asterisk.  You MIGHT be able to emulate that functionality using SLA, but don't ask me how.
14:47.27*** join/#asterisk GBLRA77 (n=GBLRA77@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
14:48.10ZPerteeManxPower, in features.conf I can configure a key sequence to dial to put a call on hold right?
14:48.19ManxPowerZPertee: no.
14:48.44ManxPowerThere is really no reason for "hold", as "holding" is always part of another feature.
14:48.51*** join/#asterisk GBLRA77 (n=GBLRA77@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
14:48.53ManxPower"hold" is ALSO a key system feature.
14:49.04ManxPowerYou can do what you want, you can't do it the WAY you want.
14:49.21boblutzWhat is it called when you use 1 RJ-11 cable to carry 2 lines over both sets of twisted pairs ?
14:49.22ManxPowerI already told you the recommended way of hold/pickup
14:49.30ManxPowerboblutz: RJ-11
14:49.48Blackthornhehhe
14:49.58boblutz2 logical lines over 1 physical line ?
14:49.58ZPerteeManxPower, when?
14:50.13ManxPowerManxPower: ZPertee: You would want to transfer the call to a parking slot, then pick up the call from another phone.
14:50.18boblutzlol
14:50.31ManxPowerboblutz: no, you have two physical lines over two pair of wires.
14:50.58Kattyanyone know of a SIP phone for a blackberry?
14:51.00Kattyor iax
14:51.04Kattyanything
14:51.07Kattygizmo doesn't count
14:51.09x86boblutz: two physical lines over 2 physical pairs with one phyisical connector... logic is not involved here ;)
14:51.48x86boblutz: you can actually do 3 lines over 3 pairs with one physical RJ11 connector too
14:52.10x86boblutz: RJ11 plugs come in 2, 4, or 6 conductor/pin configurations
14:52.12ManxPowerZPertee: you can either fight Asterisk and it's oddities and live a miserable pointless life, or you can accept Asterisk's "oddities" and live a full and good life.  You pick.
14:52.12*** join/#asterisk NoamRotter (n=noam@mail.browarnik.com)
14:52.20tzafrirx86, hmm.... what would you stand on?
14:52.36x86tzafrir: eh?
14:52.39tzafrirnm
14:52.51boblutzx86: ManxPower: ok thanks for the clarification, I thought rj-11 was always a 4 pin connection, like rj-45 is 8
14:53.26x86boblutz: rj-45 is always 8 conductor/pin configuration, yes...
14:53.37boblutzok, but rj-11 can be 2,4 or 6
14:54.01x86yeah, although I think we're splitting hairs with RJ10/RJ12 at that point
14:54.53x86http://en.wikipedia.org/wiki/Registered_jack
14:55.22boblutzdanke - wasnt sure what it was called
14:55.42x86err
14:55.52boblutz?
14:56.10x86RJ11 is standard single line / 2 pin, RJ14 is dual line / 4 pin, RJ25 is triple line / 6 pin
14:56.20boblutzok there we go
14:56.23x86so RJ11 is always 2 pin
14:56.29x86I was slightly wrong :)
14:56.36bluregardweird, I just got real bad echo on my 501 on an internal call between an it and an IAXy
14:56.38boblutzhaha, its cool, i just needed some fodder to google
14:56.40bluregardthat's a first
14:56.45ManxPowerHUH?
14:57.12*** join/#asterisk theron (n=theron@65.198.151.150)
14:57.18ManxPowerNo, RJ-13 is always 1 pair, RJ-11 can have 1, 2, or 3 pairs (2, 4, 6 wires)
14:57.27x86ManxPower: wikipedia is your friend
14:57.40x86RJ13C/RJ13W: 6P4C, for one telephone line behind the key system
14:57.47x86RJ13 is 4 conductor
14:57.51mihinomenestwhy don't you just call AT&T and have them sort it out?
14:58.01boblutzThe RJ11 standard dictates a 2-wire connection, while RJ14 uses a 4-wire configuration, and RJ25 uses all six wires.
14:58.09boblutztaken from x86 link ^^
14:58.09x86right
14:58.12ManxPowerboblutz: Ah!
14:58.17boblutz?
14:58.35boblutzmihinomenest: that would be too easy thats why
14:59.05*** join/#asterisk Skarmeth (n=Skarmeth@200.253.26.142)
14:59.16mihinomenestright, I forgot, it's against the open source ideals.
15:01.51x86http://www.arcelect.com/RJ_Jack_Glossary.htm
15:02.28boblutzThis was all prompted by strangeness when using a splitter on my rj-14 line to plug into 2 separate fxo modules on the same tdm410 card
15:03.28ManxPowerboblutz: I don't see a problem.
15:04.09boblutzManxPower: I wouldnt either, Asterisk would start even when no one was calling
15:04.21boblutzIm told the phone system in this office is jerry-rigged though
15:04.27boblutzhence the rj-14
15:04.32x86boblutz: asteirsk wasn't already running?
15:04.45Dextorionwhat linux dist are asterisknow based on?
15:04.45boblutzx86: No, haha, I said it wrong
15:04.47x86but somehow started when it saw voltage on the line?
15:04.48x86:P
15:04.56x86say what you mean and mean what you say :P
15:05.06tzafrirDextorion, rpath
15:05.11ManxPowerboblutz: Plug 2 analog phones into the adapter and see if it works.  Oh, I'm SURE you tried that already, right?
15:05.22boblutzManxPower: No, we only have rj-14
15:05.34boblutzwait i misread that
15:05.35Dextoriontzafrir :) wow.. never used that one..
15:05.37ManxPowerboblutz: the how the hell did you plug the line into Asterisk?
15:06.15*** join/#asterisk twitchnln (n=raleigha@cpe-orncorp.dktc.atl.oneringnetworks.net)
15:06.23boblutzManxPower: Plugged the lines into the fxo modules, called the number from another line in the office
15:06.25twitchnlnmorning
15:06.26ManxPowerI'll bet you have a PBX line, not an analog line.  If that is the case, you may have already blown the port
15:06.30boblutz4 lines in the office, 2 are plugged into asterisk
15:06.49*** part/#asterisk airjump (n=zielonka@62.159.95.82)
15:07.15ManxPowerboblutz: the the next step is to plug two $9 phones from walmart into the line splitter and see if you can make/receive calls.
15:07.25*** join/#asterisk skirmisha (i=skirmish@87-126-225-188.btc-net.bg)
15:07.25twitchnlni'm experiencing a strange issue with a rhino analog card, is there a zaptel option that disables *67?
15:07.27skirmishaguys
15:07.29boblutzManxPower: No fxs modules though, is that possible?
15:07.32skirmishaproblem was in iax
15:07.48tzafrirtwisted, there is no built-in zaptel feature for *67
15:07.50ManxPowerboblutz: You are NOT listening!
15:07.50skirmishait could not find iax provision file
15:07.59x86boblutz: plug the phones into the lines, not asterisk
15:08.10skirmishaand that prevent asterisk to load other modules
15:08.11tzafrirtwisted, sorry, that was for twitchnln
15:08.15skirmishawhy is that?
15:08.19x86boblutz: test each line individually to make sure they work correctly
15:08.26ManxPowerx86: you either have a blown module or a bad line.  I cannot help you further as you are not listening to me.
15:08.43twitchnlntzafrir: then why when i use a sip phone does it work, but when i use an analog phone off the rhino card it say that feature is not available on this line
15:08.47x86ManxPower: I'm not the one needing help anyway, so I guess it's good you can't help me ;)
15:08.47boblutzOh, my bad
15:08.51ManxPowersorry, that was for boblutz not x86
15:08.55x86;)
15:09.25tzafrirtwitchnln, please provide a more complete trace of the "doesn't work" scenario
15:09.39ManxPowerboblutz: and if you try to use one the the PBX phones to test this I'll smack you so hard you'll timewarp into next week.
15:10.02*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
15:10.14skirmisha???
15:10.16ManxPowertwitchnln: you are using a GUI, aren't you?
15:12.51x86ManxPower: I love how you're so friendly and eager to help people
15:12.53x86grins evilly
15:13.11twitchnlnManxPower: no
15:13.34ManxPowerx86: I'm happy to help people, I am not happy when users do something totally different from what I request."
15:13.46boblutzManxPower: Ok, so i just reformatted asterisk, now what?
15:13.49boblutzjust kidding!
15:14.04ManxPowertwitchnln: The SIP phone has it's own *67 feature
15:14.15x86ManxPower: you're always a meanie though
15:14.21ManxPowerx86: not always
15:14.40twitchnlnManxPower: it worked with 1.2 when I upgraded to 1.4 it broke it
15:14.45ManxPowerMuch like [TK]D-Fender I have little tolerance for fools, non-fools get lots of help
15:14.53Blackthornx86: better not upset him more.. we won't get anything answered for the rest of the day :P
15:14.58ManxPowertwitchnln: I cannot help you further.
15:15.12twitchnlnManxP: thanks for trying
15:17.06[TK]D-Fendertwitchnln: When you say "i use an analog phone off the rhino card it say that feature is not available on this line", WHAT exactly is saying this message, and how?
15:17.07ManxPowerBlackthorn: One thing that irritates me is when people ask questions answered in upgrade.txt
15:17.21tzafrirtwitchnln, with the available data indeed I can't help. What you need to do is to provide more relevant data. Who "says that the feature is not available on that line"?
15:17.27ManxPower[TK]D-Fender: he's a lost cause.
15:17.30tzafrirDo you see that as an error message?
15:17.45tzafrir:-)
15:17.46twitchnlngimme a sec, i'm posting debug
15:17.50[TK]D-FenderManxPower: Maybe, I'll give him a few smal chances to adapt.
15:18.02boblutzDoes a punch block denote a pbx line?
15:18.22*** join/#asterisk jpeeler (n=jpeeler@216.207.245.1)
15:18.26ManxPowerboblutz: it denotes nothing
15:18.59ManxPowerboblutz: You're just going to continue to fail to make progress until you do what I suggested.
15:19.33ManxPowerboblutz: I believe what YOU think is 2 analog lines on the cable is really a digital PBX line.  That line could easily blow the port on a Digium cards if it's plugged into a Digium card.
15:19.51boblutzWould a blown port not work?
15:19.59ManxPowera blown port could do ANYTHING
15:20.09Blackthornwhat irritates me manix is that thers no docs that say were all these "hidden little doc files are" :P you can go look up on the wiki's and digium etc etc. no were that i can see it easly says "go read the text files for install and upgrade info"
15:20.14ManxPowerBut even if the port is not blown, you still could have a digital pbx line.
15:20.22Blackthornmost of the online thigns seem to be outdated and incorrect
15:20.45ManxPowerHopefully the Digium card does not have the whatever pins the PBX uses for voltage to power the phones connected.
15:20.58ManxPowerBut I have wasted enough time with you, boblutz
15:21.19boblutzshrieks
15:21.46ManxPowerask [TK]D-Fender he has more patience than me today.
15:21.56tzafrirBlackthorn, I suggest you do one of two things: (1) file bug reports with specific points for better documentation, (2) improve voip-info.org  :-)
15:22.30boblutzManxPower: Thanks
15:22.56boblutzwasnt tryin to be "that guy"
15:23.01[TK]D-Fenderboblutz: Not even knowing what you have is a dead-end state.  When you know what you have, and what you want maybe then we can help you.
15:23.40ManxPowerboblutz: you'll stop being "that guy" the moment you try plugging in cheap analog phones into the line adapter.
15:24.19boblutzMy original question was about the 2 lines over 1 rj-11 connector, which i found out is really rj-14
15:25.37boblutzManxPower: It just clicked in my brain
15:25.57skirmishaguys why iax is giving me problems
15:26.13skirmishawhen it loads after that no modules is loaded
15:26.16[TK]D-Fenderskirmisha: Why isn't my old car running?
15:26.24[TK]D-Fenderskirmisha: Ah, details!
15:26.42[TK]D-Fenderskirmisha: perhaps you're getting caught in a DNS lookup failure or something.... go look at the hosts it refers to
15:26.45skirmishameans if i have iax.conf file etc... and restart asterisk
15:27.09[TK]D-Fenderskirmisha: if you "noload" it in modules.conf, does everything else load clean and fast like normal?
15:27.18skirmishaasterisk load iax module and after that module no other modules are loaded
15:28.44skirmishaif i remove iax.conf and all related iax conf files, then all is loaded perfect
15:29.04skirmishaif i do manual load of file it is loaded correctly also
15:30.18skirmishasorry manual load of module
15:30.18ManxPower[TK]D-Fender: I suspect he is using the sample config files.
15:30.35[TK]D-FenderManxPower: I've never heard of any issue using the sample.....
15:30.47[TK]D-Fenderskirmisha: pastebin your iax.conf masking only passwords.
15:30.48[TK]D-Fender~pb
15:30.49jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
15:30.51[TK]D-Fender^^^^^^^^^
15:31.07skirmishai have several boxes with asterisk with same setup and don;t hav such problem
15:31.28skirmishaonce asterisk loads iax module, no other modules are loaded after
15:31.47yangI am getting this error when i want to send outgoing FAX - ast_rtp_read: Unknown RTP codec 102 received from  I do connect over IAX to my asterisk from there on it goes via SIP
15:32.11boblutzskirmisha: add a "noload" to modules.conf, let everything else load, and then load the module manually from *CLI>
15:32.23boblutzperhaps?
15:32.52skirmishathat will do the job, but why it is giving me such error, can it be something in iax config
15:33.09boblutzif the conf is set up wrong, yea
15:34.36skirmishadon't see anything wrong
15:35.16boblutz(somebody else might see something wrong if you pb it)
15:35.21*** join/#asterisk Eckrall (n=kevin@eckrall.co.uk)
15:35.56[TK]D-Fenderboblutz: You mean like I already asked him to?
15:36.11[TK]D-Fenderskirmisha: and of course you don't see anything wrong otherwise you'd be asking about it.
15:36.30[TK]D-Fenderskirmisha: Or would have fixed it already.  So let us see if there is something suspicious in there.
15:37.48*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
15:37.57skirmishaguys X in dial plan is for 1-9
15:38.04skirmishaand Z is for 0-9
15:38.39Blackthornwell guys.. i just don't know what to do now.. i'm tried of banging my head against the computer screen... I've untared, reloaded, recompled, checked all my config files overa nd over. and I still get the message unkown signling pri_cpe.. and the chan_zap.so is in the moduel directory
15:39.31twitchnlnBlackthorn: did you install libpri?
15:40.01twitchnlnBlackthorn: do you get the same error with pri_net?
15:41.30ManxPowerskirmisha: wrong again.  X = any digit, N = 2-9,  Z= 1 - 9
15:42.23Qwellany numeric digit
15:43.00[TK]D-FenderQwell: "A" as well?  Don't want those people working in hex to feel excluded, do we? ;)
15:43.10caio1982i deeply wish digium could sell some of these, in orange color please ;) http://designzen.wordpress.com/2008/03/23/asterisco-usb-hub-by-joel-escalona/
15:43.12Qwellis A numeric? :p
15:43.22boblutzA char is really just a small number
15:43.23Qwellcaio1982: we're trying to get the author to make us an orange one :P
15:43.36[TK]D-FenderQwell: Depending on a certain point of view, yes :)
15:43.40*** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net)
15:44.27siyaanyone familiar with SPA3102 and it lagging regularly?
15:44.32caio1982Qwell: you kidding?
15:44.40Blackthorntwitch: yes the libpri is isntalled and the lib_pri.so is in the module director. I do not know what you mean by pri_net?
15:44.47Qwellcaio1982: well, unofficially.  we aren't going to sell them or anything
15:44.49shasta[TK]D-Fender, and 2+2=5 for extremely large values of 2 ;-)
15:44.56Qwella few of us were talking about that this morning though
15:45.13twitchnlnBlackthorn: for signalling, as opposed to pri_cpe
15:45.28caio1982Qwell: save a few for contributors :D
15:45.40twitchnlnBlackthorn: flip it around so it thinks its the telco, see if you see the same error
15:46.03*** join/#asterisk coppice (n=chatzill@218.0.193.160)
15:49.26*** join/#asterisk Corydon76-vcch (i=green@pdpc/supporter/bronze/Corydon76-home)
15:49.26*** mode/#asterisk [+o Corydon76-vcch] by ChanServ
15:50.58Blackthornahh. i see what you mean. no it's defently pri_cpe. I have an older asterisk server running at the moment.. trying to get thes newer server working. basicly I had an much older single port pri card and now i'm moving to a newer digium 4 port card
15:54.44ManxPowerBlackthorn: since it's libpri.so not lib_pri.so.......
15:55.05Blackthornwhen i remove the zapata.conf file zap then shows up in asterisk
15:56.49*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)
15:57.01boblutzthe hampster is running on its wheel
15:57.24Blackthornthe hampster is spinning in it's own wheel :P
15:57.50ManxPowerBlackthorn: listen more, talk less
15:59.52ManxPowertwitchnln: Blackthorn: flip it around so it thinks its the telco, see if you see the same error
16:00.24Blackthornwere should libpri.so exist?
16:01.00boblutzBlackthorn: I got mine in /usr/lib/
16:01.15ManxPower<PROTECTED>
16:02.06Blackthornok it's there. libpri.so, libpri.so.1, libpri.so.1.0, libpri.a
16:02.38ManxPowerthat would be expected
16:05.23ManxPowerwhat version of libpri, zaptel, and asterisk did you install?
16:06.09Blackthornasterisk 1.4.18.1, libpri 1.2.7, and zaptel 1.4.9.2
16:07.54Blackthornwrong version of libpri i bet
16:08.03ManxPowerThere you go.
16:08.12ManxPowerThere's several hours of my life I'll never get back.
16:08.24Blackthornlol, and about two days of mine
16:08.26[TK]D-FenderSMRT
16:08.28ManxPowerWhat in the WORLD make you think 1.2 would work with 1.4
16:08.41ManxPowerDidn't we tell you to get the latest?
16:08.52Kobazthe latest isn't always the greatest
16:08.59boblutzau-contraire!
16:09.05ManxPowerKobaz: no, but in this case it's required.
16:09.34Kobazyeah hmm that is an old libpri
16:10.39*** join/#asterisk latebind (i=latebind@wbs-41-208-219-82.wbs.co.za)
16:10.47latebindhello
16:12.06latebindPlease help me out, I'm looking for a java framework to do sip unit testing. I've tried sipunit but that didnt support dtmf and I found a really good one in php, but I dont want to convert it unless theres nothing else... thanks
16:12.11johndoWhat exactly does "insecure=" do in sip.conf?
16:13.25mvanbaakit makes it insecure
16:13.54*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:14.40*** join/#asterisk fedya (n=fedya@75.112.143.226)
16:15.28johndoI assume it has a purpose, does it make forwarding or reinvites work?
16:15.35*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
16:16.27ManxPowerjohndo: it's for bad, evil, stupid providers
16:20.32johndoSo I should probably turn it off
16:20.41johndodoes turning it off disable any features?
16:21.25ManxPowerjohndo: you won't be able to accept calls from bad, evil,. stupid providers.
16:22.01johndoin that case, I should turn it on.
16:22.47ManxPowerjohndo: never enable unless you need it.
16:23.30johndoseveral examples have it turned on for hardphone "friends"
16:23.35johndoI need to understand what it does
16:23.55johndonevermind
16:23.58johndoI missread them
16:24.19johndoI have it turned on for a bad, evil, and stupid provider.
16:24.32johndo(;
16:24.49joeif you have a switch for a context on an other system via and iax2 trunk is there a way to test if the switch takes place otherwise play a message other than invalid number w/o having an external script to do the testing? ie something w/in asterisk already that detects such failures (seems something that might be needed by many...)
16:25.12*** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
16:25.22methodscan i have extensions and agents use the same number ?
16:25.48johndoI thought the point was to have them not use the same number
16:26.04methodsis that the point ?
16:26.31[TK]D-Fendermethods: Do whatever you feel like.  Its your system.
16:26.41methodsbut my phone isn't registering anymore
16:27.10bkruseManxPower: That is not true
16:27.13[TK]D-Fendermethods: then you've messed up your sip.conf or your phone
16:27.27[TK]D-Fendermethods: These have nothing to do withe xtensions or agents.
16:27.39ManxPowerbkruse: you suggest enabling insecure= by default?
16:27.49bkruseManxPower: Sorry, I had some major scrollback, nevermind :]
16:27.52*** join/#asterisk jmesquita (n=jmesquit@201.7.117.114)
16:29.28*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
16:29.43*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
16:29.47ManxPowerjoe: you did not read "show application dial" or channelvariables.txt
16:29.55BlackthornI just goofed and pulled the wrong one.. thats all.. i know better :P
16:31.22*** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com)
16:31.25joeManxPower: nope, will do. Thanks
16:33.23*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193)
16:34.07*** part/#asterisk kamanashisroy (n=kamanash@202.56.7.193)
16:34.09ZPerteecan someone tell me if if this http://pastebin.ca/959665 entry is valid or not?
16:34.38*** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org)
16:34.58ManxPowerZPertee: you can't use channel variables in features.conf
16:35.22[TK]D-FenderZPertee: How about you just TRY it.
16:35.27ManxPowerZPertee: go ahead and keep flapping your arms.  It won't do you any good, but it's funny to watch.
16:36.20ManxPower[TK]D-Fender: he wants to have the hold/pickup feature of Key Systems.
16:36.29ManxPowerHe's trying to do it via features.conf.
16:37.05ZPerteeManxPower, the REASON that I switched from MS is that I liked to do things MY WAY and customized to my customers needs.  and now YOU ARE trying to tell me that LINUX and OPEN SOURCE SOFTWARE SUCH AS ASTERISK is the same CRAP as microsoft!
16:37.48ManxPowerZPertee: no, I'm telling you that Asterisk is a PBX, not a key system.  You CAN do what you want, in theory, if you have the right phones and use sla.conf, but really, it's not worth the week or so you will need to make it work well.
16:38.14*** join/#asterisk solar_ant (n=John@122.164.229.143)
16:38.24ManxPowerZPertee: Asterisk has a method for call park and pickup.  Use it.
16:38.39[TK]D-FenderZPertee: Not "the same crap".  Get your head out of your ass about thinking that you can just free-form invent syntax off some perverted extrapolation you come up with in your head and go read up on how * actually works.
16:39.06[TK]D-FenderZPertee: And sure, * is open source... if you don't like, get coding.
16:40.42yangI am getting this error when i want to send outgoing FAX - ast_rtp_read: Unknown RTP codec 102 received from  I do connect over IAX to my asterisk from there on it goes via SIP
16:41.31ManxPoweryang: http://www.asteriskguru.com/tutorials/unknown_codec_received.html
16:42.04ManxPoweryang: look likw 102 is iLBC
16:42.15ManxPoweryou don't have iLBC installed.
16:42.21Qwelleww, fax over iLBC?
16:42.26[TK]D-FenderAnd per earlier discussions * has just removed ilbc support from the base tarball
16:42.32ManxPowerQwell: At least it will remove the error.
16:42.40ManxPower[TK]D-Fender: Why?
16:42.46[TK]D-FenderManxPower: Licensing.
16:43.08ManxPower[TK]D-Fender: Ah.  It's SpeeX that I was thinking of
16:43.09[TK]D-FenderManxPower: From what I caught of that discussion.
16:43.11filethe iLBC source is removed, codec_ilbc itself is still there - you just need to download/install the stuff yourself to make it build
16:43.38[TK]D-Fenderfile: thanks for the clarification.  This is newsworthy.
16:43.41ManxPoweryang: you will not have reliable FaxOverVoiceOverIP.
16:43.55ManxPowerif you want it to maybe work at all, you MUST use the ulaw or alaw codec.
16:43.57coppicewhat's the issue with iLBC source code?
16:44.02tzafrirerr... how legal is it to distribute Asterisk with iLBC to your client?
16:44.04yangManxPower: i am using that alaw
16:44.12yangManxPower: the incoming works ok !
16:44.24ManxPoweryang: SOMETHING is still trying to use iLBC
16:44.58ManxPowerI guess I should go outside and do yard work.
16:45.03tzafrircoppice, strange license
16:45.12yangManxPower: i cannot find anything related to ilbc in debian ...
16:45.49*** join/#asterisk frieze (n=frieze@cpe-66-65-1-44.nyc.res.rr.com)
16:46.14coppiceI know it has a wacky licence, and the copyrighting of the source code is ill defined. is there a specific issue that got it removed, or is it just the vagueness?
16:46.33friezeMaybe this is a stupid question but is it possible to set up an asterisk server so I can plug in my cell phone (an iPhone) when I get home and have incoming calls feed into my PBX as another line?
16:46.37ManxPoweryang: perhaps a disallow=all allow=alaw in sip.conf would be good
16:46.43yangdo you think its a missing codec of my VOIP provider?
16:46.58yangManxPower: its made that way
16:47.00ManxPoweryang: no.
16:47.07boblutzinteresting ...
16:47.10ManxPoweryang: well it's not using that information for outgoing calls
16:47.16twitchnlnfinally got the debug for my *67 problem, see http://pastebin.ca/959687
16:47.21ManxPowerI guess you need to paste JUST the Dial line you are using
16:47.32tzafriryang, Debian has removed ilbc long ago
16:48.01ManxPowertwitchnln: YOU ARE USING GUI CONFIG FILES!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
16:48.05ManxPowerYou lied.
16:48.33ManxPowerDid you really think we would not notice that spaghetti dialplan?
16:48.33yangI have a hylafax<->IAXMODEM(fax machine)<->IAX2<->ASTERISK<->SIP<->DID VOIP PROVIDER
16:48.40coppicetzafrir: debian removes anything that's even been in the same room as something tainted
16:49.06twitchnlnManxP: config files, yes, but freepbx breaks the rhino when you click apply, so i've been working them manually
16:49.19Qwellof course it breaks it
16:49.21Qwellthat's what it does
16:49.32ManxPowertwitchnln: we can't help you with GUI config files.
16:49.46ManxPower~freepbx
16:49.47jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
16:49.48ManxPower~amp
16:49.49jbotit has been said that amp is NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
16:50.29boblutz~zombie
16:50.30jbotLibrary and server for developing networked apps/games.. URL: http://www.infa.abo.fi/~chakie/zombie/
16:50.39tzafrirtwitchnln, actually, not every GUI. But freepbx's dialplan is not fun to watch
16:50.43ManxPowertwitchnln: look at the brightside: You now have the info needed for people to help you on the correct channel
16:51.31yangManxPower: at work we got fax machine which works over ATA + SIP it simply works, i can send and receive, why my hylafax doesnt send?
16:51.49ManxPoweryang: it would take me hours to tell you the answer to that.
16:51.56tzafrirtwitchnln, try something like:   originate Zap/CHAN_NUM/NUM_TO_DIAL application Dial SIP/peername
16:52.10boblutzI called my line....sounds good, I hang up.  exactly 28 seconds later, *CLI> shows "Starting simple switch on 'Zap/1-1'" and the call times out (via my dialplan).  The same thing repeats 28 seconds later
16:52.12tzafrirthis does not use your dialplan in any shape or form
16:52.19boblutzIs this a zombie call ?
16:52.22Kattywibbles.
16:52.32yangManxPower: i understand "something" about incompatibility for t.38 codec
16:52.48iCEBrkr. o O ( wibbles? )
16:52.57EmleyMoorboblutz: Is Zap/1 your FXO?
16:53.01boblutzYes
16:53.13boblutzZap/1 and Zap/2 = fxo
16:53.18EmleyMoorI get the odd "call that may not ever have been" on my FXO
16:53.34boblutzEmleyMoor: What card ?
16:53.39yangManxPower: but i don't see the reason, as i can accepp well with hylafax, why sending makes a problem...? maybe wrong hylafax dialstring?
16:53.47EmleyMoorTDM400P, 3 FXS 1 FXO
16:54.32EmleyMoorAre there any good dialplan analysis tools available to help me find redundancy in my dialplan?
16:54.54boblutzhm...  I built the entire system using a TDM400P with 1 fxs and 1 fxo, but this new card, TDM410P ( the 2 fxo's make it a TDM402B) is causing trouble
16:55.03boblutzEmleyMoor: IVR?
16:55.18EmleyMoorboblutz: I do have an IVR
16:55.21*** join/#asterisk orn (n=orn@85.197.193.24)
16:55.25znoGhey all, with Asterisk 1.6.x series .. what version of Zaptel can I use with it? any?
16:55.35QwellznoG: 1.4
16:55.50znoGthanks
16:55.54EmleyMoorI would also prefer not to depend on implicit answering, but avoid unnecessary lines as well
16:56.20ornIf I press * on an incoming call the call gets immediately terminated. These are my features shown in the console. http://www.pastebin.org/25697 -- Any ideas why?
16:57.23tzafrirznoG, try recent zaptel 1.4
16:57.37[TK]D-Fendertwitchnln: Let me put it this way.  you have a DIALPLAN issue.  Your dialplan is created by FreePBX, and it owns your ass.  Learn how better to administer it, live with it, or get rid of it.  Your choice.
16:58.00[TK]D-FenderEmleyMoor: Nope
16:58.05znoGtzafrir: i'm using 1.4.9.2, but it's giving me a bit of trouble with incoming calls (detecting and clearing alarms when a call comes in)
16:58.48EmleyMoor[TK]D-Fender: Fair enough - my dialplan is mostly right anyway
16:59.26tzafrirright. Someone just asked about this on the -users mailing list
17:00.19tzafrirThis should be fixed in the SVN 1.4 branch, IIRC. Qwell, do you recall more correctly?
17:00.39*** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net)
17:00.59znoGthat someone was me :) SVN 1.4 branch of Zaptel?
17:01.07tzafrirOf Zaptel
17:01.22znoGcool, i'll give it a shot anyway... willing to try anything :)
17:03.53Blackthornls
17:05.49boblutzhttp://www.pastebin.ca/959717 <--- I think my box is calling itself?
17:08.46*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
17:08.49[TK]D-Fenderboblutz: and why do you think that?
17:08.57*** part/#asterisk twitchnln (n=raleigha@cpe-orncorp.dktc.atl.oneringnetworks.net)
17:09.32boblutzIm watching *CLI> start a switch  and timeout, hangup, and repeat it again exactly 28 seconds later
17:10.08boblutzI modified my dialplan to see what number that is that keeps calling, and it shows up as null, as seen in the pb
17:10.09[TK]D-Fenderboblutz: And what is Zap/1 plugged into?
17:10.33[TK]D-Fenderboblutz: Just because it doesn't show a callerid, doesn't mean there isn't one, and does say anything about who is calling.
17:10.48[TK]D-Fenderboblutz: You have to tell * to use callerid.
17:11.02[TK]D-Fenderboblutz: this is the point where you show us your zapata.conf so we can see what you set up.
17:11.23boblutz[TK]D-Fender: ok, I am going to write a little paragraph as well
17:11.32[TK]D-Fenderboblutz: Save it for after
17:13.04boblutzhttp://www.pastebin.ca/959727
17:13.17x86Katty: http://bryceporter.info/wiki/index.php/HOWTO_Asterisk_Web_Directory
17:13.35boblutzWorks fine on the TDM400p, with only 1 fxo module
17:13.38x86Katty: should do ya...
17:14.11x86Katty: just make sure the user you run Apache as has access to write to /var/spool/asterisk/outgoing
17:15.04boblutz[TK]D-Fender: I have a mistake, http://www.pastebin.ca/959731
17:15.07*** join/#asterisk a-s (n=user@89.38.174.194)
17:15.24a-shello
17:15.35a-swhat data contains a linear frame of silence ?
17:15.43Qwella-s: what?
17:16.24a-sif I want to send silence on the network, what data shell I put into a frame?
17:16.40*** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
17:17.48*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
17:17.56[TK]D-Fenderboblutz: You need "callerid=asreceived
17:18.36boblutz[TK]D-Fender: May I PM you for a quick minute?
17:18.38[TK]D-FenderKatty: Better still : a polycom XML browser app.
17:19.00ZPerteewhat's wrong with this features.conf entry "send_to_voicemail => 16*1,VoiceMail(16)" ?
17:19.02[TK]D-Fenderboblutz: PM is reserved for security restricted stuff, and consulting.
17:19.12[TK]D-Fenderboblutz: so PM only if applicable
17:20.40boblutz[TK]D-Fender: My fxo module is plugged into a RJ-14 jack in the wall.  I first called the number and everything works fine.  I hang up, and *CLI> keeps showing a new switch starting 28 seconds after it hangs up.  My boss just came back from lunch and checked the voicemail, and it has seemed to stop "calling itself"
17:20.47[TK]D-FenderZPertee: First thing, dynamic features don't transfer the call you're on.
17:21.26[TK]D-Fenderboblutz: you card will only use the inside pair.....
17:21.32[TK]D-Fenderboblutz: And your card can't call itself.
17:22.14boblutz[TK]D-Fender: the fact that it would start a switch exactly 28 seconds after hangup isnt strange?
17:22.35[TK]D-FenderZPertee: Second your formatting is wrong even if your idea was valid.  http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf
17:23.07[TK]D-Fenderboblutz: Did you plug an analog phone in parallele to hear whats going on with the line?
17:23.34*** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
17:23.42methodshow do i make a certain dial pattern go out a certain trunk ?
17:23.58[TK]D-Fendermethods: Its your dialplan.... change where the dial goes!
17:24.02Qwell~book
17:24.03jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
17:24.25boblutz[TK]D-Fender: No analog phone to test with, bout to go buy one.  I would plug the analog phone into ... ?
17:24.41[TK]D-Fenderboblutz: Geez... do I really have to spell this one out?
17:24.46*** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
17:25.33boblutz[TK]D-Fender: Unplug the current phone I am using and plug the analog phone in
17:25.54[TK]D-Fenderboblutz: Put a splitter on the wall jack, plug * & the phone into it.
17:26.02ManxPower560244
17:26.04[TK]D-Fenderboblutz: Listen for it to ring, and then answer the phone.
17:26.26[TK]D-FenderManxPower: 8675309?
17:26.42ManxPowerboblutz: you are not using a current phone, you are using Astersk
17:26.49bluregardx86, did you write that perl code
17:27.22boblutzManxPower: I was referring to the phone I am using to call Asterisk
17:27.45ManxPowerboblutz: NO!
17:27.55ManxPowerThe plug the new analog phone into the WIRE GOING TO ASTERISK
17:28.07ManxPowerwell, into the splitter that has the wires going to Asterisk.
17:28.13ManxPowerWe are trying to test for dialtone on the line.
17:28.18ManxPowerat least.
17:28.18ZPertee[TK]D-Fender, ok this vmfeature => 16*1,caller,VoiceMail,16 is in the correct syntax.  now whats the deal about dynamic features? how do I fix that?  Ultimately I want to be able to dial EXT*1 to send a call to voicemail
17:28.50boblutzManxPower: yea that is understood, "current phone" refers to the statement I made to [TK]D-Fender before I knew to use the splitter with an analog phone
17:28.53*** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com)
17:29.09[TK]D-FenderZPertee: I'll venture a guess to say that you can't USE that app there at all.  Go test it with something you see in the sample to see if feautres.conf is even being properly implemented.
17:29.30[TK]D-FenderZPertee: And like I told you already it will not transfer your caller to voicemail.
17:30.07friezeI asked this before but may have phrased it wrong. Is there any kind of gizmo, bluetooth or usb based that will allow me to make my iphone connect to an asterix box like through an FXO so I can have incoming calls run through the pbx when it's home/plugged in?
17:30.10ManxPowerZPertee: you want to set up a featuremap for a blind transfer
17:30.24ManxPowerthe extension you transfer to should match EXTEN*1
17:30.37ManxPowerthat will then run voicemail with the options you specify.
17:30.47ManxPowerYou enjoy making this as hard as possible, don't you?
17:30.56ZPerteeManxPower, ok I
17:31.11Qwellfrieze: asterisk has a bluetooth channel driver
17:31.13*** join/#asterisk `paul (n=aldee@203.192.188.226)
17:31.16ManxPowerThis isn't rocket science, millions of people do this every day
17:31.32Qwellit's not the best, but it works pretty well
17:31.34Blackthornwoot! finally got everything working again...now off to a new project :))
17:31.40friezealmost a billion speak chinese. but no matter how hard I try...
17:31.56bluregardfrieze, you want to connect an IP phone to an FXO?
17:32.10ZPerteeManxPower, ok I'll go your route.  sorry I am being such a jerk about the whole thing.  I just get kinda stubborn once I get an idea in my head.
17:32.19friezeno. a cell phone. made by apple. confusingly called an iphone
17:32.25ManxPowerZPertee: then you might as well give up now.
17:32.40Qwellfrieze: 1.6, chan_mobile
17:32.41`paulhelp pls... my problem is monitor cmd wont mix in and out files in parallel i mean it would play "in" first then the "out" audio not at the same time... help pls
17:32.46ZPerteeManxPower, sorry, not my nature
17:32.54bluregardfrieze, is it an unlocked/jailbroken iphone you can run software on?
17:33.03tzafrircan chan_mobile now connect more than one device per BT adapter?
17:33.14[TK]D-Fenderfrieze: Nobody here's had any mention of iphone specific hardware.  Get googling.  If you can get it to FXS/FXO then you can do something with it.
17:33.27Qwelltzafrir: it used to be able to.  it was removed because it's not feasible to do with bluez
17:33.40friezebluregard: not right now...may go whole hog and run a fully hacked 2.0
17:33.44ManxPowerfrieze: There are several ways, you can use Asterisk's experimental Bluetooth support, or you could buy an analog adaper for the phone and then plug that device into an Asterisk analog port.
17:34.01*** join/#asterisk KillerQueen_86 (n=alan@201-212-100-188.cab.prima.net.ar)
17:34.08ManxPowerI doubt any analog adapters support the Apple iPhone
17:34.11johndoI am getting this message:
17:34.12johndo[Mar 27 12:32:10] WARNING[32339]: res_config_mysql.c:144 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info.
17:34.19johndoi did set core debug 10
17:34.26johndobut I don't get any useful information
17:34.36Qwelljohndo: tell logger.conf to log debug to console
17:34.43friezeManxPower: yeah, I'm assuming that their default bluetooth implementation leaves out some of the access you would need as well
17:34.44bluregardfrieze, I know there have been successful SIP calls made with an ipod touch
17:34.54`paulmy problem is monitor cmd wont mix in and out files in parallel i mean it would play "in" first then the "out" audio not at the same time... help pls
17:34.56Qwellfrieze: it just emulates a headset
17:35.02Qwellsurely the iPhone supports headsets?
17:35.09KillerQueen_86hello everyone... is there a way to disable jitterbuffer from CLI? the commands "iax2 set debug off" or "iax2 set debug jb off" didn't work
17:35.21friezeQwell:  yeah. I guess I'll poke around some more
17:35.30ManxPowerKillerQueen_86: that sets the debug
17:35.32KillerQueen_86and i hate to have this things on CLI:  JB STATS:IAX2/asterisk1-6 ping=3 ljitterms=-1 ljbdelayms=0 ltotlost=-1 lrecentlosspct=-1 ldropped=0 looo=-1 lrecvd=-1 rjitterms=0 rjbdelayms=40 rtotlost=0 rrecentlosspct=0 rdropped=0 rooo=0 rrecvd=1
17:35.57ZPerteeManxPower, for my own use I'm not near as particular.  My customers in this job are people who sell supplies to farmers...nice people but not tech savvy at all.  they are used to key systems and i'm trying to use asterisk as close to what they are used to as possible.  the closer I get the better they like me...too different and they'll never speak to me again.
17:35.58johndoQwell: same message, nothing useful
17:35.58ManxPowerKillerQueen_86: do you want to disable the jutterbuffer or the log messages?
17:36.14QwellKillerQueen_86: upgrade, those messages were moved
17:36.33ManxPowerZPertee: no offence intended, but you are not even close to knowing enough to install an asterisk system for a customer
17:36.56KillerQueen_86the log messages! i think jb is a useful thing (i don't know how it works, but if it's doing something, it's doing it well)
17:37.21Kobazis there a way to bump up the internal jitter buffers in grandstream phones?
17:37.30QwellKobaz: replace them with polycoms
17:37.30KillerQueen_86I am at 1.6beta3... should i update to beta4?
17:37.39KobazQwell: besids that
17:37.49ManxPowerKobaz: I know it's totally illogical, but that would be a function of the phone, not Asterisk.  You should check the docs for the phone.
17:37.51*** part/#asterisk mfedyk (n=mfedyk@adsl-71-134-153-204.dsl.irvnca.pacbell.net)
17:38.12KobazManxPower: which is why i was asking specifically about grandstream phones
17:38.41ManxPowerKobaz: best of luck.
17:38.47ZPerteeManxPower, they understand that I am fairly new to this.  the idea is that they are letting me put the system in so that I can get experience and I don't charge them as much.  besides I am a friend of the family who owns the business and they would rather have me help them then someone that they don't know
17:38.47Kobazheh
17:38.48johndoQwell: That does put a lot in info on the console now
17:39.33ManxPowerI really wish Pidgin had a "perm ignore" option
17:41.10plik`paul: either use MixMonitor to record, or bodge it with sox
17:42.56*** join/#asterisk pithen (n=pithen@mail.graphlogic.com)
17:43.08ZPerteeManxPower, besides for still being a senior in high school and only taking a few college classes I am doing ok.  Keep playing with asterisk and by the time I graduate from college and gain some more experience I'll be doing alright, hopefully :-)
17:43.44KillerQueen_86ManxPower: how may I disable the weird JB log messages?
17:43.54QwellKillerQueen_86: I already told you how
17:44.31ManxPowerKillerQueen_86: Qwell already told you how.  Did you not like his answer?
17:44.53KillerQueen_86haha... just "upgrade" to beta4?
17:46.02pithenhi all..just got myself a chan bank..it seems like the /etc/zaptel.conf is auto generated, but it makes all the FXS ports ls instead of ks.. how can i override this?
17:46.28KillerQueen_86~Clinton
17:46.29jbotsomebody said clinton was the best president we've had since george bush, dammit! or the only president we've had since george bush, dammit! or the Pants Dropper in Chief or the guy blowing away small countries or the president whose staff continues to classify cryptographic software as munitions or a shitty human being or stupid or in control of nuclear weapons or a psychopath, or clit-ton
17:46.56Kattyx86: cheers (=
17:47.08KillerQueen_86:D
17:48.21plikpithen: sed ? :)
17:49.07*** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net)
17:49.12pithenplik,heh..i was hoping for something more permanent :)  whats the dif b/t ls and ks anyway?
17:49.47plikmore permanent that writing the config file... <shrug>
17:50.07*** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
17:50.14methodsagents are supposed to just hear a beep right ?
17:50.14pithenplik, when the file is generated at boot time thats not very permanent
17:50.46plikthe book explains the difference between loop & kewl start
17:50.55plikoverwrite it with your own then
17:53.11x86Katty: works? :)
17:54.35bluregard~book
17:54.36jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
17:55.42*** part/#asterisk KillerQueen_86 (n=alan@201-212-100-188.cab.prima.net.ar)
17:56.22JayTee52~Bush
17:56.23jbotooooh Bush me no likey bring on the tile baby... wanna borrow my razor....
17:56.33JayTee52lol
17:58.06znoGtzafrir: i compiled and installed the latest zaptel from SVN (1.4 branch). Still no luck (throwing alarms when calls come in). Any other ideas?
17:58.36Kattyx86: working on xen stuff right now (=
17:58.50Kattyin /etc/resolv.conf the search MYDOMAINNAME bit at the top... what does that actually /mean/
17:58.57Kattysearch mydomainname for dns servers?
17:59.13znoGtzafrir: do I need to recompile Asterisk as well?
17:59.44*** part/#asterisk theron (n=theron@65.198.151.150)
18:00.38x86Katty: no, you can do host resolution without the domain, and the domain is appended... for example "ping foo" gets translated into "ping foo.bar.baz" automatically
18:01.20x86Katty: you can specify multiple search domains so if "foo" didn't exist in the "bar.baz" domain, it would search "bling.zoo" second, for example
18:01.44Kattyx86: so if i don't want this thing updating itself, i need to take that search bit out.
18:02.01x86Katty: err, I didn't say anything about any updates ;)
18:02.06Kattyoh good.
18:02.18x86Katty: it's searching for an FQDN, given just a hostname
18:02.22x86so it can resolve the host
18:03.02x86Katty: for example, if you put "search yahoo.com" in resolv.conf, and then went to on to do "ping www", it would ping www.yahoo.com
18:03.55Kattynods
18:04.15tzafrirznoG, no need to rebuild asterisk
18:06.37znoGtzafrir: d'oh .. then I'm still having the same issue
18:08.29tzafrirznoG, ah, so you are the one who asked on the list....
18:20.00pithentzafrir, i just got my astribank today and im trying to set it up...all the fxo ports (4th span) have the lights on, and all the other ports are off.. does that indicate something?
18:20.55tzafrirpithen, likely it has not been initialized yet.
18:21.35BlackthornIs there a differnt channel for asking about how to add some type of web based application that would allow non-technical people add/remove sip phones and caller extentions?
18:22.03pithenthat explains why it worked initially when i boot of the included cd, but not after i reset the power... the manual only lists to modprobe xpp_usb, zt_registration on, xpp_sync auto, and then ztcfg. am I missing something?
18:22.11*** join/#asterisk klin3d (i=ircN@pc-32-231-86-200.cm.vtr.net)
18:22.36*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:22.41x86Katty: try that script i wanna know how it works for you :)
18:22.47tzafrirpithen, also, what's the output of zaptel_hardware -v?
18:22.50tzafrirhttp://zaptel.tzafrir.org.il/README.Astribank.html#_lsusb_test
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18:23.48*** part/#asterisk flambaz (n=flambers@unaffiliated/flambers)
18:24.00pithenill have to tell you in a min..i just boot off the cd again to see if it casme back up (it did)
18:24.10tzafrirBlackthorn, there are a number of such applications. Each of them provides a rather different view of Asterisk, and hence has its own IRC channel, usually
18:24.12*** join/#asterisk solar_ant (n=John@122.164.229.143)
18:25.20*** join/#asterisk kannan (n=kann@123.201.60.110)
18:25.40kannangreetings to all
18:26.14pithentzafrir, usb:005/005          xpp_usb-     e4e4:1151 Astribank-multi USB-firmware
18:27.00*** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
18:27.09Blackthorntza: do you know some of those channel names please?
18:27.24x86tzafrir: odd, when I do zaptel_hardware -v, it says "Usage: /usr/sbin/zaptel_hardware"
18:27.25tzafrirpithen, fpga firmware not loaded. fpga_load missing?
18:27.39*** join/#asterisk daniev (n=ganbarim@201.245.223.90)
18:27.43tzafrirx86, you have an older version of it, right
18:28.02*** join/#asterisk real0ne (n=Ismail@41.251.74.200)
18:28.06tzafrirthat older version always shows the verbose display
18:28.20x86tzafrir: 1.4.4
18:28.54pithentzafrir, the command exists :) not sure how i should be calling it though
18:31.03tzafrirpithen, what distro is that?
18:32.11pithendeb etch
18:33.01danievhey ppl
18:33.30danievwhich distro has better integration with asterisk between debian and centOS ?
18:34.50tzafrirpithen, good. Debian has a separate user.log
18:35.14tzafrirdo you see any related messages in /var/log/user.log ?
18:35.40pithenhmm.. that i do, last 6 lines. shall im pm them to you?
18:35.48tzafriryes, please
18:35.49*** join/#asterisk Deeewayne (n=dwayne@216.207.245.1)
18:35.49*** mode/#asterisk [+o Deeewayne] by ChanServ
18:36.09znoGtzafrir: yep, thats me
18:36.45boblutzdaniev: whatever one is easier for you to use
18:37.11*** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net)
18:38.08*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
18:38.08tzafrirpithen, can you try running: /usr/share/zaptel/xpp_fxloader reset
18:38.11hescooperating on a Debian install, haviong just done the update/upgrade thing, my extensions.conf is only partailly reloading.  What might the issue be?  What should I be looking for?
18:38.34*** join/#asterisk MattJ (n=MattJ@88-110-69-232.dynamic.dsl.as9105.com)
18:38.34pithentzafrir, i just unplugged the usb and plugged back in..got the same thing on 006 this time
18:38.37danievboblutz : for me it's easier and better debian
18:38.59tzafrirpithen, not exactly the same, but has a useful effect
18:39.19danievi have more experience working with debian
18:40.04pitheni initially installed the zaptel drivers from digium, then afterwards installed yours...could there be a conflict?
18:44.37tzafrirpithen, no. It appears fpga_load wasn't built originally
18:45.32tzafrirWhat exactly do you mean by "from Digium"? any recent enough version of Zaptel from Digium should work well
18:45.46*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
18:46.11tzafrir(and if not - I should get flogged, as I have commit access there as well)
18:46.21*** join/#asterisk quigon (n=matias@64.116.224.17)
18:47.05*** join/#asterisk Cle0 (n=Ismail@41.251.98.101)
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18:47.21*** join/#asterisk erwinpogz (n=niwre@67.159.178.21)
18:48.08erwinpogzhello, how can i gave an account to a user with only one channel?
18:48.27*** part/#asterisk Blackthorn (i=blacktho@76.77.160.10)
18:48.31[TK]D-Fenderhesco: PASTEBIN is your friend....
18:48.34[TK]D-Fender~pb
18:48.35jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
18:50.59pithentzafrir, know what i did.... initially i had the 1.4.9.2 from dig installed..but when i plugged in and got it working i kept getting a wierd error on asterisk (turned out to be a rogue setting in users.conf (im not used to 1.4 yet).. so I installed your driver... still had trouble and couldnt find zttool, so I apt-get'd zaptel from deb
18:51.16pithenwhich is why it was out of sync..i thought it was only a couple of binary tools that wernt included with the driver
18:52.09tzafrirpithen, if you want to apt-get, use my backport: deb http://updates.xorcom.com/rapid etch main
18:52.19pithenno amd64 :(
18:52.30*** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU)
18:52.34tzafriroops. So is my desktop. But it is Lenny
18:53.08pithenheh..i almost got a free embedded asterisk from you guys
18:53.16*** join/#asterisk quigon (n=matias@64.116.224.17)
18:53.48pitheni ordered a XR0009 with TCO, and I got a XR1009 with TCO cable but without the option
18:55.04djs26Hey guys.  I know this is likely a common question, but could I have recommendations on the best value voip phone for sip, or at least, the really bad ones to avoid?
18:55.56erwinpogzdjs26 for value voip phone, i use budgetone 101
18:56.53x86DONT DO THAT
18:56.54x86EVAR
18:57.02x86~grandstream
18:57.03jbot[grandstream] the Yugo of VoIP hardware.  Run.  Run away now.
18:57.11*** join/#asterisk NirS (n=NirS@77.126.252.169)
18:57.13ZPerteehates users.conf
18:57.20ZPertee~users.conf
18:57.21jbotusers.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system.
18:57.25bluregardthe 1KHz timer for ztdummy is configured in the kernel under CONFIG_HZ_1000 right?
18:57.46bkruseLOL
18:57.47pithenheh..i wouldnt mind it if it were at least documented
18:57.57bkruseIf you cannot use users.conf, you are a nub
18:58.25bkruseI agree with that statement, except for the fact that most of it is parsed just like a context in the respective config files.
18:58.28*** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
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18:58.43djs26quickly closes google on grandstream
18:58.53pithenthe idea is simple..but there are some options you just have to guess the names of
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18:59.20bkruseZPertee: What are you getting owned by in users.conf?
18:59.29*** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
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18:59.35x86djs26: what do you need? a cheap IP phone?
18:59.45x86djs26: Polycom IP320 is your friend
18:59.55djs26x86: Yes, just something to tide me over for the minute.
19:00.08tldSiemens Gigaset IP products.
19:00.09djs26I will go with much better later, and I am broke.
19:00.14tldCheap, good enough SIP support.
19:00.16tzafrirbluregard, it's in the Zaptel README: http://zaptel.tzafrir.org.il/#_kernel_configuration
19:00.21*** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
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19:00.28x86djs26: get the IP320... about $100
19:00.33erwinpogzis it possible to add a SIP user with only one channel? how?
19:00.59ZPerteebkruse, not sure if it is just a random problem...rebooting my machine.  I have voicemail configured in my users.conf file but when I try this VoiceMailMain(15) it doesn't work for some reason.
19:01.05x86http://bryceporter.info/wiki/index.php/HOWTO_Asterisk_Web_Directory
19:01.08x86gah
19:01.13x86http://www.voipsupply.com/index.php?cPath=95_112
19:01.16x86there we go :)
19:01.19bkruseZPertee: is the extension 15?
19:01.28bkruseZPertee: Do a VoiceMailMain(15@default)
19:01.42djs26x86: Too much for now.  I need something $50 or less
19:01.44RobHCan I use pattern matching in a Gotoif statement to parse callerid number and if its in my internal extension range (6XX) send it to a labeled priority?  I am trying it here, and I am doing something wrong.  http://rafb.net/p/vA09HA95.html
19:01.51bkruseZPertee: You can see this by the 'voicemail show users' command.
19:01.59djs26I need something for home use better than a crappy soft phone.
19:02.08boblutz[TK]D-Fender: Do you like echotraining?
19:02.14djs26Software that is.
19:02.25[TK]D-Fenderboblutz: Never needed it.  Thats what HWEC is for :p
19:02.53*** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
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19:02.57boblutzgoogles
19:03.00x86djs26: use a softphone until you can afford a real IP phone... otherwise you're wasting your money
19:03.07[TK]D-FenderRobH: "core show function REGEX"
19:03.11x86djs26: you do NOT want grandstream EVAR
19:03.20djs26Ok, will avoid them
19:03.31ZPerteebkruse, nm.  something stupid as extra parenthesis in extensions.conf
19:03.37*** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
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19:03.48[TK]D-Fenderx86: Voipsupply= overpriced
19:03.50bkruseZPertee: heh, not users.conf's fault :D
19:03.54djs26x86: But I need something independent of my computer.
19:04.32[TK]D-Fenderdjs26: Get a Linksys ATA then
19:04.50tlddjs26: I'd seriously recommend you consider the Gigaset ones.  Wireless is nice, and they're not expensive.
19:04.50RobH[TK]D-Fender: Thanks, just a bit much expressions to call in the manner I was doing eh?
19:05.04*** join/#asterisk aksyn (n=aksyn@78-86-127-220.zone2.bethere.co.uk)
19:05.17x86[TK]D-Fender: sure, for people buying things one at a time... I buy stuff in bulk, have a dedicated account rep, and they do me very good on pricing
19:05.25[TK]D-FenderRobH: You could do 2 tests with an "and" in there.  1 for length =3, other for starts with "6"
19:05.33pithendjs26, could just get an ata
19:05.36djs26tld: Looking now.
19:05.57*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
19:06.05[TK]D-Fenderx86: Meanwhile we're talking about a guy who's fussing over 50$.  Right tool for the right job I say :)
19:06.10pithendjs26, just make sure you get an unlocked one
19:06.10x86djs26: well you can waste $50 on a grandstream.... but to be honest, I'd rather wipe my ass with a $50 bill than buy a grandstream phone... at least that would feel better
19:06.19x86[TK]D-Fender: lol, true
19:06.33tlddjs26: If you need more than one, the S675IP offers wideband as well, so you get insane call quaity compared to ISDN.  And jabber support is nice. :)
19:07.52x86tld: dude the C470 is DECT even... DECT rocks
19:08.59tldx86: all of them are dect
19:09.33*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
19:09.45x86tld: who sells them?
19:09.52x86I found a philips that looks cool
19:10.03tldx86: gigaset is king
19:10.23x86where to buy?
19:10.33tldnot sure. :)
19:10.34tlderr
19:10.36tldnot sure. :(
19:10.42tldI used a norwegian shop.
19:10.44tldhmmm
19:11.25x86tld: the E series is not even IP though
19:11.31x86tld: which ones are IP? just C series?
19:12.07tldx86: The ones with IP in the name.
19:12.07tldLike S675IP, CE460IP R etc
19:12.07tldS675IP is the only one with wideband
19:12.14tldback and forth.  send me a privmsg if there is anything
19:14.25ZPerteeI am using * with an avaya pbx and a Linksys ATA.  pots -> Asterisk -> ATA -> Avaya PBX .  My question is I want to do a ChannelRedirect but I'm not sure for the channel part of the ChannelRedirect cmd if I should use the SIP port or the zap port.
19:14.36boblutzA phantom ring is when a line rings, but no one is really on the other end. Phantom rings can be due to "voicemail waiting" being sent from the telco. ( taken from http://kb.digium.com/entry/72/ ) <-- Is there a remedy for this?
19:16.06erwinpogzis it possible to add a SIP user with only one channel? i dont want other softphone line 2 enabled
19:17.02*** join/#asterisk bsaxon (n=bsaxon@66.0.66.4)
19:17.32*** join/#asterisk Treytor (i=PJIRCWeb@ip68-4-124-32.oc.oc.cox.net)
19:17.50TreytorI'm in desperate need of some help setting up trixbox
19:18.01TreytorI'm so close, but know I am overseeing one small thing, can anyone help?
19:18.12sysreq`~trixbox
19:18.13jbot[~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
19:18.13TreytorI'm trying to get it setup with voipstunt - http://www.voip-info.org/wiki/view/VoipStunt
19:18.28znoG~ask
19:18.29jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
19:18.29Treytor:(
19:18.45Treytorit's mostly with freepbx, I think
19:18.52sysreq`~freepbx
19:18.53jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
19:18.57lirakisTreytor: go to #trixbox already jeeze
19:19.01TreytorI am
19:19.09Treytoralready in there
19:19.10lirakisTreytor: this is not #trixbox .. so ask there
19:19.47boblutzscoffs
19:19.53Treytorthat channel is dead, and I know my problem is something very simple I'm overlooking
19:20.20*** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com)
19:20.27lirakisTreytor: it wont get answered here ..
19:20.28hescoMy incompletly loading extensions.conf file looks like this, on its tail, in the console on a reload.  http://pastebin.com/d1b389aa5
19:20.45lirakisTreytor: maybe #trixbox is dead for a reason ;)
19:21.13x86tld: can't find anyone selling the S675IP...
19:21.19Treytorwell, what would you suggest for someone with virtually no linux / asterisk experience?
19:21.27RobHoh i know, pick me, pick me... oh wait, that was one of those rhetorical questions huh lirakis
19:21.29alrsTreytor: if no one is in #freepbx try #voipcoop
19:22.00boblutzhesco: Try and PB your actual extensions.conf
19:22.00*** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
19:22.13RobHTreytor: asterisk @home, cuz then atleast you get a bit more support in that channel.
19:22.22jameswfping tzafrir tzafrir_home
19:22.26RobHor is it asterisknow, my bad
19:22.27Treytorhmm
19:22.40Treytorwell, let me ask THIS, then
19:22.41lirakisRobH: asterisk@home is trixbox
19:22.50RobHwhich one is the digium release?
19:22.58Treytorwhat's the best way to interface a voip with the web?  I want to make calls out from my box, which are instigated from a website
19:22.58ZPerteeasterisknow
19:23.01lirakis... head + wall
19:23.01RobHasterisk.org hates me today.
19:23.03jameswfor is trixbox asterisk@home we may never know
19:23.49RobHTreytor: http://www.asterisknow.org/
19:23.54lirakisTreytor: read
19:23.56lirakis~thebook
19:23.57jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
19:24.02RobHthebook knows all.
19:24.07*** part/#asterisk MACscr (n=Mark@adsl-75-23-64-188.dsl.peoril.sbcglobal.net)
19:24.15lirakisTreytor: and spend some time on it... trixbox.. etc. are a crutch
19:24.16jameswf~asterisknow
19:24.16jbotit has been said that asterisknow is based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards.  Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk
19:24.51RobHOk, I hate regular expressions >_<
19:24.58x86regexes are fun
19:25.02x86in perl?
19:25.07jameswf#regex
19:25.29jameswfdo noe cast your perl before swine
19:25.29lirakisregex is really powerful.. im reading sed & awk now.. great stuff..
19:25.35hescoAnd this is the tail of my extensions.conf file: http://pastebin.com/d1f261f0a
19:25.50ZPerteeTreytor, google "asterisk click to call" or "asterisk click to dial"
19:26.04Treytorokay I'll do that, thanks
19:26.26boblutzhesco: you in cleveland?
19:26.26Treytorcan you at least tell me where I'm supposed to enter this? - Dial(SIP/00{EXTEN}@voipstunt)
19:26.29lirakisTreytor: seriously! read the book .. its a free download!!! even if you use asterisknow or whatever .. it will help you learn
19:26.30jameswfis trying to motivate self to learn python...
19:26.44TreytorI've been reading a book for the past 6 hours
19:26.51boblutzTreytor: ${EXTEN}
19:26.54Treytora trixbox book, however
19:26.54lirakisTreytor: and it will tell you the answer! ...
19:26.56jameswfa book or "the book"
19:26.58boblutzread harder
19:26.59x86jameswf: both python and ruby are hot right now
19:27.03lirakisTreytor: THE BOOK .. not a lame trixbox book
19:27.07lirakisTreytor
19:27.09lirakis~thebook
19:27.09jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
19:27.18Treytoryeah, getting that now
19:27.59jameswfyeah I like php people slam it but seriously php is like a gun its good if you know it bu sadly so simple a moron can easily shoot himself or someone within a short distance
19:28.14ZPerteeGO PYTHON!
19:28.36jblackHas anyone done realtime asterisk with postgres lately on ubuntu, say hardy heron or so?
19:28.39RobHx86: I am trying to use them to figure out if the callerid number of a caller is in my internal extensions, if it is, when it makes an outbound call, it should set the callerid to the office's caller id.  If it is not an internal extension, and is someone hopping through our system (due to them calling an extension with followme set) I want it to send their original callerid.  http://rafb.net/p/MCF6BH17.html
19:28.40lirakisx86: OT .. but i think python is kind of a ... hobby language.. i mean its hot for a small group of people .. ruby is kind of blowing up... that and alot of adobe related stuff to like Flex is BIG in "web 2.0" stuff
19:28.55jblackThe book's examples are referring to a library that seemingly doesn't exist yet.
19:28.59boblutzflex SUCKS!!!!!!
19:29.17x86RobH: that's easy, use a perl AGI :)
19:29.37lirakisboblutz: im glad you can so discretely communicate your ideas
19:29.41RobHthat sounds like using a car to travel 10 feet...
19:29.44lirakisboblutz: .. but it does suck. ;)
19:29.48boblutzlol
19:30.02x86actually it sounds like using a truck to transport goods ;)
19:30.06jameswfmost things can be done in a 4 line shell script
19:30.10x86aka, using something for what it was made to do
19:30.14*** join/#asterisk VaNNi (n=VaNNi@lgb-static-216.70.165.200.mpowercom.net)
19:30.26x86RobH: all of your extensions are in SQL right?
19:30.36RobHI just want to compare the callerid number, if its 6XX, set callerid to one thing, if its not 6XX, go to another thing
19:30.45RobHnope, all my extensions are in the dialplan.
19:30.49jameswfmost people can be replaced with a 4 line shell script
19:30.55lirakisRobH: you can do that with a macro.. you probably dont need agi involved .. look at gotoif
19:31.06RobHthats what I was using...gotoif.
19:31.09RobHits in my paste.
19:31.21friezecan anyone point me to a good guide for choosing hardware for a soho asterisk box? I'm trying to figure out if I can simpy add a card and run it on my current poweredge server I'm using for a firewall + media server
19:31.28boblutzfxotune compatiable with tdm410?
19:31.44lirakisRobH: right.. but you have it matching a pattern of _1NXXNXXXXXX
19:31.47jameswf~cheap
19:31.48jbotwell, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
19:31.57lirakisRobH: so how can it ever match an internal extension?
19:32.15jameswflirakis: CONTEXT
19:32.16jameswflirakis: CONTEXT
19:32.17jameswflirakis: CONTEXT
19:32.19jameswf<PROTECTED>
19:32.22RobHlirakis: correct.  its matching the internal extension elsewhere in the dialplan.  then it hits followme.  followme uses the context i pasted to make outgoing calls
19:32.25friezewhy am I not surprised when the robot tells me to spend a lot of money on computer hardware
19:32.43lirakisjameswf:  I KNOW   I KNOW   I KNOW   I KNOW   I KNOW   I KNOW   I KNOW  ..
19:32.43jameswf~botsnack
19:32.43jbotjameswf: :)
19:32.46lirakiscalm down
19:32.59RobHi just want it to read the callerid nmber, which it does, and then compare it to a pattern match
19:33.00jameswf~developers
19:33.01jbotit has been said that developers is http://www.youtube.com/watch?v=KMU0tzLwhbE
19:33.02lirakisjameswf: RobH didnt give a complete paste
19:33.16jameswfRobH: No Doughnut
19:33.18RobHi dont need a complete paste, i just need help with the callerid number comparison.
19:33.28RobHi get to my context just fine....
19:33.35RobHits complete in regards to this issue.
19:33.56RobHno doughnut, but its cuz im diabetic ;]
19:34.31jameswfbacon flavored doughnuts it ould be like fat people crack
19:34.48jameswfis a fat people BTW
19:34.57jblackAnyways, the book is referring to Driver = /usr/lib/odbc/libodbcpsql.so, which is seemingly no longer available on ubuntu (however, libodbcpsqlS.so is)
19:34.59alrsthe redfone ethmf driver works without a redfone, I'm happy to say
19:35.00lirakisRobH: 6[0-5]{2,}
19:35.14lirakisRobH: oh sorry nope
19:35.25RobHIf the callerid(num) is in the 600 to 699 range, its my internal extensions, and in that case, I want it to go to one label priority, if its not in that range, go to antother
19:35.26RobHno?
19:35.34RobHdarn, was hoping you just solved it ;]
19:35.47RobHif i am explaining this poorly, my apologies
19:35.50lirakisRobH: so for 600-699   RobH: 6[0-9]{2,}
19:36.14jameswfexten => _699,1,GOTO()
19:36.20jameswfexten => _6XX,1,GOTO()
19:36.33lirakisjameswf: he isnt matching the exten
19:36.35RobHjameswf: they are callerid numbers, not extensions.
19:36.38lirakisjameswf: he is matching the origin
19:37.20RobHlirakis:  exten => _1NXXNXXXXXX,n,SET(ORIGIN=${REGEX(6[0-9]{2,} ${CALLERID(num)})})   ?
19:38.10*** join/#asterisk ZachMen (n=Zach@pool-71-186-7-232.chi01.dsl-w.verizon.net)
19:38.56lirakisRobH: im not sure how the asterisk regex function works.. but that expression should match 6 .. then any digit 0-9, twice
19:39.19RobHwas what i had technically workable?  (Just curious)
19:39.35RobH[600-699]?
19:39.45RobHor is it on a base ten basis?
19:39.47*** join/#asterisk Skarmeth (n=Skarmeth@201009024030.user.veloxzone.com.br)
19:40.24znoGquestion: I'm doing an Answer() when a call comes in, for some reason Asterisk only really answers the call 3 or 4 lines down the context (i can see in the log it does an Answer, a Wait, a Background() then another Background(). When I dial in, i only hear starting from the 2nd Background being played
19:40.27hescook, after backing up my original extensions.conf, I trimmed out everything but a few contextes which I am actually using.  It appears to have completely loaded.  However, when I run my test (drop a .call file into the outbound spool) I see the following two errors on the console:
19:40.29znoGany ideas?
19:40.31hescopbx_spool.c:346 scan_service: Unable to open /var/spool/asterisk/outgoing/17707551543.call: Permission denied, deleting
19:40.43hescopbx_spool.c:388 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/17707551543.call'
19:40.57hescoWhat does this mean and what can I do about it, please?
19:41.00lirakisRobH: regex acts on single characters... so .. its not really a "base 10"  .. its just that many regex engines understand [0-9] to be a special series that means any of the digits between
19:41.02*** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com)
19:41.07lirakisRobH: just like [a-z]
19:41.07jameswf6[0-9]{,2}
19:41.26RobHznoG: try a wait (0.5) after answer to let the audio settle and that may be it.  (imma newb, i could be wrong)
19:41.38RobHsorry Wait(0.5)   no spaces
19:42.20ZPerteetry changing the permissions on 17707551543.call.  I personally just chmod 777 17707551543.call.  not sure what's required but thats how I do it
19:42.27jameswfhesco: who owns the .call
19:43.14lirakisRobH: actually it is 6[0-9]{2} with no comma.. that will match exactly 2 of the previous [0-9] sets
19:44.47znoGRobH: i had a Wait(1) after the Answer()
19:45.03znoGRobH: wasn't enough .. i just added a Wait(5) and now i hear the audio from the beginning, but it doesn't sound right
19:46.08jameswfexten => 911,1,playback(tt-weasles) exten => 911,n,goto(1)
19:46.13RobHznoG: so the first problem is now fixed, you hear all the audio, now its a quality issue.
19:46.13RobHcorrect?
19:46.54RobHjameswf: hahaha, weasels make me laugh, which makes this bullet wound to the chest hurt more, silly weasels killing my 911 connection ;]
19:47.28znoGRobH: no, i just think that having to add a Wait(5) doesn't sound right
19:48.21RobHznoG: well, you had 1 second and it cut off audio due to the audio connections not being in place is my understanding.  5 seconds fixes it, but is an extemely long pause.
19:48.36jameswfwonders if you can do s,1,answer() s,n,goto(3) s,n,goto(2) h,1,goto(2) h,2,goto(1) and send asterisk in to a loop of death
19:48.39*** join/#asterisk Blackthorn (i=blacktho@76.77.160.10)
19:48.49RobHThis sounds like a connection to the server issue, try shortening that wait, and if the phone is not local to the system, check the latency
19:49.37BlackthornHi again, I asked over in the astericknow channel over an hour ago but no one seems to be talking over there today. Can you add asteriskgui to an already installed asterisk system?
19:49.50jameswf~astersiknow
19:49.57jameswf~asterisknow
19:49.58jbotasterisknow is probably based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards.  Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk
19:50.07lirakisRobH: so here is a "Final" regex for your problem
19:50.11lirakisRobH: ^6[0-9]{2}$
19:50.15boblutzwow...
19:50.40RobHlirakis: this worked:  exten => _1NXXNXXXXXX,n,SET(ORIGIN=${REGEX("6[0-9]{2}" ${CALLERID(num)})})
19:50.51lirakisRobH: the previous one would match anything that contained the 600-699 set within even a subset .. like 165323523 will match
19:51.06lirakisRobH: this ensures that it only matches 3 digits
19:51.24RobHwhat I am using now will match anything within the subset huh?
19:51.48RobHwould be odd, but a pain, heh
19:51.59RobHfixed it
19:52.02lirakisRobH: yup.  if you set origin to 165323523 it would set your clid wrong
19:52.08RobHlirakis: thank you very, very much!
19:52.12lirakisRobH: ;)
19:52.26RobHeven without wanting to, i learned some regular expression stuff.
19:52.54hescoWhat does this mean and what can I do about it, please?
19:52.56RobHall just so the folks who set followme can know who it is without recording the name
19:52.57hescopbx_spool.c:346 scan_service: Unable to open /var/spool/asterisk/outgoing/17707551543.call: Permission denied, deleting
19:53.00hescopbx_spool.c:388 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/17707551543.call'
19:53.04lirakisRobH: they really are powerful.. and useful in .. some times arcane.. but .. always time saving ways (time saving after you learn them )
19:53.28RobHI have some shell scripts i have to write which I am sure will use them, so I guess I have no choice
19:53.31RobHlearn or die =P
19:53.31lirakishesco: it means the whatever process is trying to delete the call file doesnt have permissions
19:53.38*** join/#asterisk Isaiah (n=Spam@74-46-51-104.dr01.hnvr.mi.frontiernet.net)
19:53.42[TK]D-Fenderhesco: jameswf>hesco: who owns the .call
19:53.49boblutzhesco: are YOU from cleveland?
19:53.53[TK]D-Fenderhesco: jameswf asked you a question.  Answer it.
19:54.14hescoI'm looking that up now.
19:54.29[TK]D-Fenderhesco: What user wrote the .call?  What perms?  What user is * running as?
19:54.38lirakishesco:  ls -alh /var/spool/asterisk/outgoing/17707551543.call | awk '{print $3}'
19:54.44[TK]D-Fenderhesco: The error is extremely blatant.
19:55.07hescoThe .call file is not in that spool directory.  In fact it is empty.
19:55.07*** join/#asterisk vladtser1 (n=Vladimir@adsl-065-015-017-142.sip.mia.bellsouth.net)
19:55.27hescoIt was placed in that spool by a perl script run as sudo root
19:55.34lirakishesco: long term solution is to do setgid of asterisk on the directory
19:56.16lirakishesco: or chmod the files ... or .. make it so your perl script doesnt run as root
19:56.38vladtser1the Master.csv file logs the time of a call about 6 hours ahead of my local time. Since I'm in the EST and we're on DST shouldn't it be only 4 hours...or does asterisk use something beside GMT to log calls?
19:56.44hescoI just an hour ago upgraded my debian version of asterisk.
19:56.58lirakishesco: ... okay .. thats .. totally irrelevant
19:57.03boblutzLOL
19:57.05boblutzapt-get ftl
19:57.30hescoI'm guessing the debian version just changed the presumptive user for asterisk, which I've tended to run from a bash cli as root, in a way which permits me to monitor its console.
19:57.41lirakishesco: does your perl script need root priv?
19:58.02hescoBefore the upgrade it did to drop the file into the spool.
19:58.09hescobut for nothing else.
19:58.22lirakishesco: then i guess youve got your solution ;)
19:58.53lirakisbut for now.. chown -R asterisk:asterisk /var/spool/asterisk/outgoing
19:59.58lirakisvladtser1: it uses localtime
20:00.51lirakisvladtser1: you could set /etc/zoneinfo on your system to GMT so that your localtime is GMT.. or whatever you want it to be.  check our system time with date
20:00.55vladtser1lirakis: that's a neat trick...my local time is 16:00:00, but if I call my asterisk server it will log the call in Master.csv as coming in at 20:00:00
20:01.21lirakisvladtser1: `date` returns?
20:01.30vladtser1excuse me it will log the call as coming in at 22:00:00
20:01.43vladtser1date returns 16:01:32
20:02.00hescoIts already there: drwx------ 2 asterisk asterisk 48 2008-03-27 13:05 outgoing
20:02.32lirakisgtg .. conf call
20:03.06[TK]D-Fenderhesco: Show us the script placing the call and where you set perms, and where you moved the file to the outgoing folder
20:04.11*** join/#asterisk JoseBravo (n=Your@190.156.225.15)
20:04.49vladtser1oops, it must be using GMT...I just called the * box at 16:02:30 and it logged the call at 20:02:30, which is GMT...thanks for the pointers!
20:06.49*** join/#asterisk dhill (i=dhill@fog.mindcry.org)
20:07.08dhillwhat is the difference between Dial(SIP/user) and Dial(SIP/user/${EXTEN})
20:07.12dhillis the latter a trunk?
20:08.45vladtser1lirakis: I'm running FreeBSD not Linux...can you elaborate on this zoneinfo file...I have a /usr/share/zoneinfo folder but I don't see a config file in there
20:09.37*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
20:09.59vladtser1nevermind...I think I found it...I'll use TZ instead...
20:09.59*** join/#asterisk tomcontr3 (n=gcontrer@165-150-222-201.adsl.terra.cl)
20:10.09tomcontr3Hi,  I have an critical problem...
20:10.18tomcontr3I have just upgrated from 1.2 to 1.4
20:10.28tomcontr3and now when I try to register my phones I get:
20:10.38tomcontr3[Mar 27 16:12:42] NOTICE[6635]: chan_sip.c:15075 handle_request_register: Registration from '214 <sip:214@192.168.2.200:5060>' failed for '192.168.2.214' - No matching peer found
20:10.53real0neany one here have debian or openbsd
20:10.54tomcontr3please.. any help will be really apreciated...
20:12.28hesco[TK]D-Fender: preparing a pastebin of that now.
20:12.44tomcontr3anyone please??
20:12.54[TK]D-Fenderdhill: Both will dial the peer you specified.  The latter will target an exten you pass, the other in the case of a phone will pass back the return contact of a phone that reg'd typically.
20:13.16Blackthorntom: i think that is telling you that the user that your trying to register with does not exist in the sip.conf file
20:13.19dhillok, thanks
20:13.30[TK]D-Fendertomcontr3: * can't ID the caller.  pastebint he SIP debug of a failed attempt
20:13.55[TK]D-Fenderdhill: You would not use the latter for phone-type devices normally.
20:13.58dhilli have a customer that needs 5 phone lines..  but their software needs 5 sip connections.  It can't handle Dial(SIP/customer/${EXTEN})
20:14.40*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
20:15.25dhillor so they say
20:15.56[TK]D-Fenderdhill: "Their software" being?
20:17.28vladtser1can anyone tell me where the 2nd edition of *-TFOT is located?
20:17.39dhill3CX
20:18.07[TK]D-Fenderdhill: Well if thats what you're forced to do, so be it...
20:18.11[TK]D-Fender~book
20:18.12jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
20:18.14[TK]D-Fendervladtser1: ^^^^^^
20:18.17dhill[TK]D-Fender: Does Dial(SIP/user/${EXTEN}) set the extension in the To: ?
20:18.34[TK]D-Fenderdhill: Yup.
20:18.50[TK]D-Fenderdhill: (IIRC)
20:19.10dhilloh good lord.. i am reading the pdf for their software.. it does support it
20:19.12dhill*sigh*
20:19.26dhilli will have to show them :P
20:20.30dhillyes, it does put it in the To:
20:20.34dhilli just tested with debug
20:20.44dhillthanks
20:25.07[TK]D-Fenderheading out, BBL
20:25.09Kattywho was it that told me about g4u?
20:25.15Katty[TK]D-Fender: was it you?
20:25.16[TK]D-FenderKatty: I did
20:25.23Katty[TK]D-Fender: i'm cloning a dummy xen server.
20:25.31Katty[TK]D-Fender: i'm excited to see the results.
20:25.50[TK]D-FenderKatty: Its a raw partition dump.  Data is data beyond that...
20:26.03[TK]D-FenderKatty: Should be just fine normally.
20:26.08[TK]D-Fenderok, I"m off, BBL
20:26.23Kattyanyone know what happens to the 'extra space' leftoever from a g4u clone?
20:26.24*** join/#asterisk anonymouz666 (n=anonymou@201.19.122.138)
20:26.35Kattysay if the original is 40GB and the destination is 80GB
20:26.43Kattyand the linux machine is only using...uhh 10gb
20:26.49Kattyand the partitions equal 40gb
20:26.59Kattywhat does g4u do with the extra 40gb?
20:27.29*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
20:27.29*** mode/#asterisk [+o russellb] by ChanServ
20:29.14Kattyrussellb: do you know what g4u does with the extra space if it clones a 40gb to an 80gb?
20:31.24hesco[TK]D-Fender: my perl script whose operation is now showing permission issues in my newly upgraded Debian 1.2.13 Asterisk: http://pastebin.com/d6f4efa56
20:31.58x86Katty: did it work? :)
20:32.03Kattyx86: mew?
20:32.07Kattyx86: oh, your perl thing?
20:32.13x86Katty: yeah! try it out!
20:32.15russellbnew?
20:32.15Kattyx86: i'm still tinkering with xen lol
20:32.16Kattyokay
20:32.24x86has been sitting here patiently all day :P
20:32.28Kattysorry >.<
20:32.33x86hehehehehe
20:33.18x86muh mediawiki skills are lacking a bit, but you should be able to get the idea even with the jacked up formatting
20:34.52Kattyyeah, i think so
20:35.01Kattyi'm going to pretend i have all the prereqs
20:37.12*** join/#asterisk yassine (n=yassine@unaffiliated/yassine)
20:37.17Kattyx86: so now i just open up dialer.pl in a browser?
20:38.21*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:41.59Kattyah
20:42.07Kattyi think i'd need to write a form around it, based on looking at it lol
20:42.10ZachMenhas anyone install asterisk on FC8?
20:42.36*** join/#asterisk axisys (n=axisys@155.70.141.45)
20:42.51x86Katty: yeah I just made it quick and dirty... best thing to do is make it output CSS ;)
20:43.07Kattyx86: naturally
20:43.12Kattyx86: at first i was confused about it
20:43.17Kattyx86: so i ran it from terminal
20:43.21x86Katty: err make it output HTML with CSS tags in it, and use a CSS stylesheet to make it purdy
20:43.27Kattyx86: and then when DUH when it said zomg where's $number
20:43.47x86Katty: also, it's setup for SIP extension 200, you might want to change that :)
20:44.10znoGdamnittttt... these damn zaptel "Detected alarm: No Alarm" is driving me nuts
20:44.14znoGcan't receive or make zap calls
20:44.27Kattyx86: yeah i'll do somethin with that
20:44.37Kattyx86: g4u's done, so i gotta attend to that now (=
20:44.39QwellznoG: upgade
20:44.41Qwellupgrade too
20:44.42x86Katty: make sure the user that apache runs as has access to write to /var/spool/asterisk/outgoing too
20:44.51x86g4u?
20:44.52znoGQwell: to?
20:45.04Qwellthe latest versions of zaptel/asterisk?
20:45.05x86znoG: what version ye be runnin?
20:45.08znoGQwell: i've tried Zaptel 1.4.9.2 and Zaptel from SVN (1.4 branch)
20:45.22x86znoG: what version of asterisk?
20:45.29znoGand now I'm trying Asterisk 1.4 (latest) instead of 1.6 just incase it was an Asterisk beta thing
20:45.35znoGx86: 1.6.0-beta4 until half an hour ago
20:45.53znoGx86: now Asterisk 1.4.18.1
20:46.38x86bad hardware?
20:46.46x86what hardware is it?
20:46.46znoGstill no go... asterisk detects alarms called "No Alarm" on incoming and outgoing calls
20:46.55znoGx86: it was working ok until today .. nothing changed really
20:47.01znoGx86: its an openvox A400P card
20:47.21x86znoG: ok, so what's you're point? you don't think a card can just go bad? :)
20:47.25erwinpogzanyone knows a good FOSS predictive dialer for asterisk?
20:47.28yassinehi is the asterisk config engine set per default to odbc ?
20:47.30x86(especially openvox heh)
20:48.30znoGyeah, i'm sure it can go bad.. but it's not THAT likely .. specially considering that if I use zaptel 1.4.7.1 i can receive calls, and making calls simply has a horrendous noise during the call but.. it works
20:48.32x86heh, openvox.com has a default apache test page
20:48.36x86that's awesome ;)
20:48.53x86znoG: ah
20:49.00x86znoG: call openvox for support
20:49.17znoGyou probably want www.openvox.com.cn
20:49.36znoGis that the easy solution for this? go call someone who cares? ;)
20:49.44lirakisis away (leaving..."the internets" are safe ... for now)
20:49.54tomcontr3Im having a problem with zaptel now: ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
20:50.02tomcontr3any idea?
20:50.04x86you're having a hardware problem... usually you call the hardware maker to troubleshoot that
20:50.19x86tomcontr3: incorrect zaptel.conf configuration
20:50.26x86tomcontr3: pastebin your zaptel.conf
20:50.35tomcontr3I also get: Zaptel Version: Unknown
20:51.03tomcontr3http://pastebin.ca/960026
20:51.32tomcontr3I have just migrated to asterisl 1.4    zaptel was working perfectly with version 1.2
20:51.41tomcontr3ofcourse I also upgrated the zaptel version
20:52.20*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
20:52.27x86yeah so your configs basically need a complete re-write ;)
20:52.39x86tomcontr3: which upgrade guide did you follow?
20:52.50EmleyMoorwill no doubt have "fun" when the next Debian release happens
20:53.02tomcontr3non
20:53.07*** join/#asterisk seme (n=seme@rrcs-208-105-67-178.nyc.biz.rr.com)
20:53.15jameswfis allready on Asterisk 1.8 it is web 3.5 compatible....
20:53.23Qwell3.5, pfft
20:53.24*** join/#asterisk riddlebox (n=cts@75-128-170-26.static.stls.mo.charter.com)
20:53.24semehi guys
20:53.27Qwellmight as well be running Etch
20:53.43ZachMenanyone know of free trunks for outbound to 800 numbers
20:53.49riddleboxcan anyone explain to me how to stop a snom 300 phone from inserting a 9 when it dials out?
20:53.57EmleyMoorZachMen: North American?
20:53.59jameswf~snom
20:54.00jbotwell, snom is like all German products. High quality, but wacky engineering. :)
20:54.10kannanlol
20:54.21kannan~polycom
20:54.22jbot[polycom] the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html
20:54.30jameswf~grandstream
20:54.31jbotwell, grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
20:54.31kannankewl
20:54.33tomcontr3anyidea?
20:54.35x86tomcontr3: you have no channel configurations in zaptel.conf
20:54.45kannan~Yugo
20:54.46jbotmethinks yugo is a politically incorrect reference to a nation that no longer exists in that form, producing cars of questionable quality
20:54.46EmleyMoorI know you can call North American 800 numbers usinf FreeWorldDialup and also using sipbroker
20:54.53kannanaah
20:54.56x86tomcontr3: wait... hold one
20:55.01EmleyMoorUK, DE, NO and NL also over FWD
20:55.10tomcontr3dono..  thats the smae file I used with version 1.2  and it worked ok.
20:55.23EmleyMooris still looking for one that can call numeros verts francais
20:55.26tomcontr3Channel 01: FXS Kewlstart (Default) (Slaves: 01)
20:55.27tomcontr3Channel 04: FXS Kewlstart (Default) (Slaves: 04)
20:55.33jameswf~pb
20:55.34jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
20:55.36ZachMenEmleyMoor yes NA...
20:55.52EmleyMoorZachMen: Do you have any VoIP accounts?
20:56.19jameswfthinks jbot can handle most of this room on his own
20:56.22x86tomcontr3: http://pastebin.ca/960037
20:56.29x86tomcontr3: try this config
20:56.30ZachMenEmleyMoor yes i am using siptel
20:56.42x86tomcontr3: pastebin your zapata.conf too
20:56.51ZachMenEmleyMoor do you know other good stable CHEAP trunks?
20:57.21tomcontr3same thing: ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
20:57.35x86lets see zapata.conf
20:58.01tomcontr3http://pastebin.ca/960041
20:58.19x86EmleyMoor: there is a service in Michigan that allows for toll-free termination that actually pays YOU to use it
20:58.27x86voipmich i think it's called
20:59.12x86tomcontr3: take out group=0
20:59.47x86tomcontr3: also in your first channel definition do channel => 1,4, then take out the explicit channel 4 config
21:00.05jameswfBestest Dialplan Ever http://pastebin.ca/960044
21:00.34x86jameswf: hahahaha
21:01.16jameswfthinks it will make asterisk lock up tighter then a well you get it
21:02.06x86tomcontr3: try this: http://pastebin.ca/960048
21:02.15EmleyMoorZachmen: I know FWD can be unreliable - sipbroker seems good and accessible from just about any
21:02.26*** join/#asterisk joebob777as7 (n=joe@71-210-4-23.eugn.qwest.net)
21:02.28EmleyMoorI will see about voipmich
21:02.32x86tomcontr3: also, show us your asterisk logs... put logger.conf on full
21:03.04*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
21:03.26joebob777as7hey I have a problem with touch tones when I call a system that asks me to put in numbers like verify a social or put in a bank account or such. It doesn't seem to recognize properly...
21:03.27tomcontr3ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
21:03.29tomcontr3same thing
21:03.39tomcontr3there seems to be a problem with zaptel
21:03.39x86tomcontr3: yeah let me see your logs
21:04.00tomcontr3take a look at this: http://pastebin.ca/960051
21:04.24*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.133)
21:04.51x86tomcontr3: odd... dunno how you managed that one ;)
21:05.00*** part/#asterisk kamanashisroy (n=kamanash@202.56.7.133)
21:05.01tomcontr3ir really wird
21:05.04x86tomcontr3: completely un-install zaptel and re-compile it
21:05.24tomcontr3how do I unisntall?
21:05.37Kattymake clean
21:05.37joebob777as7i'm using g729 codec no zaptel lines
21:05.47x86Katty: that wont un-install silly
21:05.54semecan anyone tellme if it is possible to change how asterisk spells out the users name when someone searches the directory... right now, if I dial bob, it says "B. O. B. if this is the extension you were looking for..." but I just want it to play a recorded name or something more sane
21:05.56x86Katty: that just cleans the binaries out of the source tree ;)
21:05.58tomcontr3I nevers uninstalled version 1.2 of zaptel
21:06.04Kattyx86: apt-get remove!
21:06.06x86tomcontr3: that's probably the problem
21:06.09Kattyx86: is this a multiple choice question?
21:06.14tomcontr3how can I uninstall that?
21:06.16RobHseme: record their name in the voicemail application
21:06.52semerobh how do I do that?
21:06.55RobHseme: dial into VoiceMailMain, select mailbox and password, then in the mailbox, set options for name, busy greeting, and unavail greeting.  (You specificially want the name, not the greetings)
21:06.57semeyou mean in the actual voice mail?
21:07.05x86tomcontr3: rm /lib/modules/`uname -r`/misc/*
21:07.12semethanks
21:07.15semelet me try that
21:07.18RobHseme: core show application VoiceMailMain
21:07.22x86tomcontr3: be careful if you have some other misc modules besides zaptel
21:07.25RobHYep, it pulls it from the voicemailbox.
21:07.40Kattyponders a snack
21:07.44tomcontr3I removed all the old modules
21:07.48tomcontr3manually
21:07.57RobHyou can of course be complicated and record them and convert them and move them, but its just easier to use voicemailmain
21:07.57Kattyhow do i view my partitions?
21:08.00tomcontr3and then asterisk installed the new onces
21:08.10RobHKatty: in linux?
21:08.14*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:08.18KattyRobH: yes'r
21:08.20Katty[TK]D-Fender: hewwoes.
21:08.23RobHfdisk -l
21:08.33x86tomcontr3: asterisk doesn't install zaptel modules
21:08.39tomcontr3I know
21:08.44RobHif you are using software raid, it will look funnkkkyy
21:08.47tomcontr3but...  where are zaptel modules installed?
21:08.52RobHbut if yer just using regular old partitions, it looks fine.
21:08.59tomcontr3same folther where asterik install its modules?
21:09.06hesco[TK]D-Fender: Welcome back.  Before you left you had asked to see this: my perl script whose operation is now showing permission issues in my newly upgraded Debian 1.2.13 Asterisk: http://pastebin.com/d6f4efa56
21:09.35ZachMenEmleyMoor can i pm you
21:09.41hescoI started invoking my perl script as sudo -u asterisk and its now working again.
21:09.55KattyRobH: this is hard to read sizes on
21:10.30KattyRobH: in fact, i can't really tell what they are at all.
21:10.34RobHKatty:  if you wanna see what you have mounted and used and such, df does it.
21:10.41RobHdf -h
21:10.57hescoOnly now my new sip trunk is not working, though my old iax2 trunk does get the message delivered.  However, the funds I have to run the next job with have already been deposited to the sip provider.  So that is my next step.
21:11.00RobHthat may be more what you need, but remember if its not mounted, df doesnt care.
21:11.02*** part/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net)
21:11.04tomcontr3?
21:11.20*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134)
21:11.27*** part/#asterisk kamanashisroy (n=kamanash@202.56.7.134)
21:11.44Kattyokay
21:11.54Kattyin terms of linux file systems, what does 'extended' refer to?
21:12.01Kattyis it like swap?
21:12.08tomcontr3I have just re-installed zaptel
21:12.11tomcontr3but I get the same error
21:12.32tomcontr3ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
21:12.57hescoto examine the disk partitions themselves, try fdisk or cfdisk.  BE VERY CAREFUL not to write any changes unless that is what you intend to do.  I prefer cfdisk, because it gives me a cheat sheet of its supported commands at the bottom of the screen.  I think both have to be invoked as root.  Also cfdisk offers a command line option which will simply priont out the partition table and quit.
21:13.04RobHKatty: in what context do you mean, when you run df you see 'extended'?  Usually with hard disk partitions, you have up to 4 primary paritions, anything over 4 means 3 primary, and a fourth 'extended' parition that holds the rest.
21:13.36RobHor use cfdisk, which I honestly had no idea existed, but damn, its kinda neat
21:13.39RobHhesco: thanks!
21:14.10hescoKatty: ext2 and ext3 are two filesystem types.  I think ext3 includes journaling of changes
21:14.34tomcontr3x86, r u still there?
21:14.34RobHoh, extended as in that... yea, what hesco says is correct, ext3 is journaled, ext2 is not.
21:15.30hescojournaling helps the file system recover its state in the case of a power failure or other incident which shuts down the machine in a haphazard way.
21:15.59x86tomcontr3: if you don't read what I say, I can't help you
21:17.59semehow do I dial the voice mail system directly?
21:18.57KattyRobH: when i do fdisk -l, i have a 10gb section where the filesystem is listed as extended
21:19.00KattyRobH: that's all it says.
21:20.05RobHseme: make an extension, like exten => 9,1,VoiceMailMain()
21:20.21semeaah
21:20.22semethanks
21:20.24RobHKatty: I am having trouble understinading, can you cut and paste it to a pastbin?
21:20.35RobHunderstanding even, durn typos
21:20.45tomcontr3sorry fi I mist a part...
21:20.52tomcontr3please tell me again.
21:22.31tomcontr3I found this :http://lists.digium.com/pipermail/asterisk-dev/2004-December/008231.html
21:22.44*** join/#asterisk Isaiah (n=Isaiah@74-46-51-104.dr01.hnvr.mi.frontiernet.net)
21:24.38tomcontr3x86?
21:25.21tomcontr3please... if I dont fix this Im in serious trouble.
21:27.51*** join/#asterisk mike-ekim (n=mike@204.13.1.94)
21:28.01mike-ekimis there a way in asterisk that I can set the caller ID to a european number?
21:28.29[TK]D-Fendertomcontr3, pastebin "cat /proc/zaptel/*"
21:28.48[TK]D-Fendermike-ekim, You can set it to FRED if you feel like it.
21:29.14tomcontr3http://pastebin.ca/960079
21:29.40ManxPowermike-ekim: It is up to your carrier as to if they will accept it or not.
21:29.47mike-ekimright
21:29.50mike-ekimthats what I was assuming
21:30.03joebob777as7hey I have a problem with touch tones when I call a system that asks me to put in numbers like verify a social or put in a bank account or such. It doesn't seem to recognize properly...
21:30.26mike-ekimI can have it show up as 0441XXXXXXX ?
21:30.43ManxPowermike-ekim: you can set it to ANYTHING YOU WANT.
21:30.49mike-ekimsweet
21:30.56mike-ekimknow much about voice pulse?
21:31.11ManxPowerYou will have to ask your carrier if they accept it.
21:31.14tomcontr3anyidea?
21:31.20Rico29somebody knows when asterisk 1.6 stable will be released ???
21:31.38[TK]D-FenderRico29, Ask Satan if he's bought that snow-blower yet....
21:31.41Rico29not the date, but if it will be un 15days or 3 months
21:32.12Rico29mmh
21:32.18Rico29i'll do it
21:32.20Rico29:]
21:32.39joebob777as7is it an issue with using the g729 codec i'm running into?
21:32.44[TK]D-Fendertomcontr3, And now "cat /etc/zaptel.conf" & "ztcfg -vvvv"
21:33.47yassinehi i have exactly the same configuration as shown here : http://kb.digium.com/entry/40/ but when i try to call my pstn analog line i asterisk does not reconize the call comming in even that ztmonitor and zap show status boths confirms that my hardware are reconized, any suggestion on this ?
21:34.29tomcontr3zaptel: http://pastebin.ca/960088
21:35.05tomcontr3ztcfg: http://pastebin.ca/960091
21:36.01[TK]D-Fendertomcontr3, Try using just port 1, then 4. See if its jsut 1 module that has a problem.
21:36.24[TK]D-Fenderyassine, pastebin yoru zaptel.conf, zapata.conf, and your inbound dialplan contexdt
21:36.57yassine[TK]D-Fender: okay one sec
21:38.04x86http://img386.imageshack.us/img386/5357/theyseemerollinhatinfratz9.jpg <-- ManxPower let someone take his picture again?
21:38.39ManxPowerI'm not even going to click on that
21:39.14ManxPoweradds x86 to the "people to exile to aussieland when I become dictator of the world" list.
21:39.23mike-ekimlol
21:39.35x86hahaha
21:39.45x86it's totally SFW
21:40.36*** join/#asterisk tomcontr3 (n=gcontrer@165-150-222-201.adsl.terra.cl)
21:40.41yassine[TK]D-Fender: http://rafb.net/p/DJ0EWA63.html
21:40.49ManxPowerI think the last pic of me on the web is from Astricon 2006
21:41.28*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:42.03*** join/#asterisk tomcontr3 (n=gcontrer@182-142-222-201.adsl.terra.cl)
21:42.07[TK]D-Fenderyassine, So you call your line and * doesn't seem to react at all?
21:42.07tomcontr3ok. Im back
21:42.24tomcontr3[TK]D-Fender... where you hable to find out anything
21:42.25yassine[TK]D-Fender: exactly
21:42.25tomcontr3?=€
21:42.39[TK]D-Fendertomcontr3, I asked you to go test something.  Go do it.
21:42.51mike-ekimwasnt astricon 2006 in miami?
21:42.58mike-ekimor was that the bootcamp
21:43.13tomcontr3sorry  Probably I didnt get the mssange becuase I got disconected
21:43.55mike-ekimtomcontr3: I bet your problem is a type
21:44.00mike-ekimtypo ** :P
21:44.17tomcontr3?
21:44.59yassine[TK]D-Fender: when i monitor the hardware using ztmonitor i can see that the signal is changing when i dial my number
21:45.15tomcontr3[TK]D-Fender: can you resend the last mesagges?=
21:45.36hescoShow codecs responds: Disclaimer: this command is for informational purposes only.
21:45.50hescoHow do I know which codecs are installed and supported by my installation?
21:45.59[TK]D-Fender<[TK]D-Fender> tomcontr3, Try using just port 1, then 4. See if its jsut 1 module that has a problem.
21:46.10tomcontr3I did try that
21:46.15tomcontr3but I got the same error
21:46.18[TK]D-Fenderhesco, "show translation"
21:46.28[TK]D-Fendertomcontr3, show us.
21:46.33tomcontr3line 21: Unable to read Zaptel version information.
21:46.34hescothank you.
21:46.39RobHhesco: show transla...
21:46.44RobHdamn tk beat me
21:46.55*** join/#asterisk digime (n=digime@99.145.104.206)
21:46.57RobHi acutlaly have to pull up my cli and confirm this stuff!  ;]
21:47.02Rico29I have a preoblem when compiling asterisk :
21:47.04Rico29checking how to run the C++ preprocessor... /lib/cpp
21:47.04Rico29configure: error: C++ preprocessor "/lib/cpp" fails sanity check
21:47.15Rico29can somebody help me please
21:47.24IsaiahWhat do people here normally use to manager their asterisk installs? Trixbox? Asterisknow? Plain install + Freebpx? Nothing?
21:47.26tomcontr3http://pastebin.ca/960106
21:47.31QwellIsaiah: nothing
21:47.36QwellGUIs are evil
21:47.40Qwell(mostly)
21:47.43[TK]D-Fendertomcontr3, "ztcfg -vvvv" please
21:47.49RobHIsaiah: gui's are a crutch!
21:47.51mike-ekimCLI !
21:47.54hescoso, [TK]D-Fender: does a hyphen in the translations table indicate that the codec is not supported?
21:47.55IsaiahQwell: sometimes ;)
21:47.59RobHmanage it with the cli and vi
21:48.03[TK]D-Fendertomcontr3, Why no version info?
21:48.04mike-ekimamen <3
21:48.07Rico29can somebody help me please ?
21:48.10[TK]D-Fenderhesco, believe so
21:48.19IsaiahRobH: They also make some things easier, but I kind of agree with you
21:48.26RobHwell, true
21:48.27tomcontr3dono... thats the problem
21:48.31IsaiahJust wondering what people thought
21:48.41mike-ekimRico29: what distro
21:48.48Rico29debian 4
21:48.48RobHbut with asterisk, you may find that many of the GUIs either fall short of full functionality, or just plain do not work and do what you need.
21:48.59mike-ekimdid you check google
21:49.11frogonwheelsRobH:  Somtimes crutches can be useful.
21:49.15[TK]D-Fendertomcontr3, Go install a SUPPORTED version of *
21:49.27tomcontr3I installed the latest
21:49.32RobHYep, I have no experience with the frontends for *
21:49.36Rico29mike-ekim > i was
21:50.03tomcontr3I dont know why it says that its an Unknown version-...
21:50.07*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131)
21:50.15RobHeverything I read before getting into * was very negative about the different GUIs out there and more often than not recommended taking the learning curve and committing to a plain install.
21:50.19frogonwheelsRobH:  Somtimes crutches can be useful.
21:50.39*** part/#asterisk kamanashisroy (n=kamanash@202.56.7.131)
21:50.47tomcontr3where are zaptel modules installed?
21:50.50yassine[TK]D-Fender: any idea how to get more info on my issue described above?
21:50.53RobHfrogonwheels: yep, just in this case I am of the understanding those same crutches tend to only let you walk down certain hallways ;]
21:51.08mike-ekimRico29: did you get this error isntalling asterisk?
21:51.26Rico29yes
21:51.34Rico29i'm trying to install it
21:51.42frogonwheelsRobH: For eg- I've been using the xmail web front end to manage my users and stuff -but still have to edit the filter configs by hand.
21:52.14[TK]D-Fendertomcontr3, Go show us what you think "latest" is, and how you did it.
21:52.14hescoWithout sacrificing a know working installation of * 1.2.13, is it possible to put a parrallel version (1.4, perhaps even 1.6) on the same server?
21:52.30[TK]D-Fendertomcontr3, Because we have to keep in mind that your ASTERISK is ancient.
21:52.45mike-ekimdo apt-get install autoconf build-essential libikseme1-dev libikseme1-utils libikseme13 libsnmp-dev python python-dev python-setuptools pciutils
21:52.49[TK]D-Fenderyassine, no clue.
21:52.58tomcontr3zaptel-1.4.9.2
21:52.59mike-ekimalso make sure you have donr apt-get instsall linux-headers-`uname -r`
21:53.05tomcontr3asterisk-1.4.18.1
21:53.08[TK]D-Fendertomcontr3, You can't use 1.4 zaptel with Asterisk 1.2!
21:53.26tomcontr3Im using Asterisk 1.4
21:53.29[TK]D-Fendertomcontr3, thats like using 1959 Ford Mustang parts on a 2005 chevy Lumina
21:53.36tzafrir[TK]D-Fender, you can
21:54.07tomcontr3I never said I was using 1.2
21:54.28[TK]D-Fendertomcontr3, I think I've just mixed PB's
21:54.36[TK]D-Fenderdammit, I'm too sub-divided right now.
21:54.36Rico29mike-ekim >
21:54.38Rico29rico@r11443:~/asterisk-1.4.18.1$ dpkg -l | grep cpp
21:54.38Rico29ii  cpp                               4.1.1-15                             The GNU C preprocessor (cpp)
21:54.38Rico29ii  cpp-4.1                           4.1.1-21                             The GNU C preprocessor
21:55.13tomcontr3but what should I do I I would like to re-install everything... how can I uninstall all modules
21:55.25tomcontr3I need to fix this some how
21:55.32jameswfping tzafrir tzafrir_home
21:55.38tzafrirjameswf, pong
21:55.59tomcontr3[root@keiserver zaptel-1.4.9.2]# modprobe wcfxo
21:55.59tomcontr3FATAL: Error inserting wcfxo (/lib/modules/2.6.18-1.2798.fc6/misc/wcfxo.ko): Unknown symbol in module, or unknown parameter (see dmesg)
21:55.59tomcontr3FATAL: Error running install command for wcfxo
21:56.05jameswftzafrir does zaptel.init parse $MODULES in order...
21:56.35tzafririt modprobes the modules in there in that order, yes
21:56.57mike-ekimdid you run that apt-get
21:57.00jameswfis the default sysconfig/zaptel yours?
21:57.06Rico29mike-ekim > yes
21:57.19mike-ekimretried installation?
21:57.22tzafrirtomcontr3, so look at the last lines in /var/log/messages
21:57.40Rico29i'm doing it
21:57.43tomcontr3Mar 27 17:54:35 keiserver kernel: wcfxo: disagrees about version of symbol zt_transmit
21:58.01tomcontr3it seems that the modules that I have are still the modules from zaptel 1.2
21:58.07tzafrirtomcontr3, the module zaptel is probably still from an older version
21:58.12tomcontr3but how can I delete them and then install the new ones
21:58.26tzafrirtry: rmmod zaptel; modprobe wcfxo
21:58.30tomcontr3I have re-installed zaptel 1.4 like 3 times now
21:58.47tomcontr3ERROR: Module zaptel is in use by zttranscode,wctdm
21:58.51tzafriryou have reinstalled the files. But you have not unloaded modules?
21:59.03tomcontr3ohhh... mybe thats what I need
21:59.07tomcontr3how can I do that?
21:59.15tzafrirrmmod zttranscode wctdm #better stop asterisk first
21:59.26tomcontr3ok Asterisk stped
21:59.51tzafriror simpler: /etc/init.d/zapetl stop
22:00.17tomcontr3ok done
22:00.24tzafrirthis will just unload zaptel and everything that depends on it
22:00.25joebob777as7hey I have a problem with touch tones when I call a system that asks me to put in numbers like verify a social or put in a bank account or such. It doesn't seem to recognize properly...
22:00.37tomcontr3now I should go an re-install 1.4?
22:00.38tzafrirtomcontr3, and now try the modprobe
22:00.46tzafrirno need to reinstall
22:01.05tomcontr3[root@keiserver asterisk]# modprobe wcfxo
22:01.05tomcontr3ZT_CHANCONFIG failed on channel 1: No such device or address (6)
22:01.06jameswftzafrir can you make my sed life a little easier and add two comments to sysconfig zaptel... #Start Digital Modules & #Start Analog modules...
22:01.35tzafrirjameswf, xpp_usb is both
22:01.57jameswfbut the init accounts for xpp_* right?
22:02.30tzafrirno. xpp_usb is modprobed from the list of modules like the rest
22:03.04tomcontr3now I get: ZT_CHANCONFIG failed on channel 1: No such device or address (6)
22:03.12tzafrirjameswf, frankly I don't really like that list of modules
22:03.43tzafrirjameswf, you want to put something after all the digital ones?
22:03.51tomcontr3fixed
22:04.11yassinegood night everyone
22:04.19*** part/#asterisk yassine (n=yassine@unaffiliated/yassine)
22:04.48tomcontr3mmm
22:04.49jameswftzafrir I am not a fan either buut to keep digital before analog I need to rewrite my post script to put our digital with digital and analog below but I cant sed to a module as I cant depend on it being there... can it be split zaptel.digital zaptel.analog zaptel.zzz
22:04.58jameswfthen the init calls in order
22:05.00Rico29mike-ekim > it works, g++ package was missing
22:05.04tomcontr3but I still cant make outside calls}
22:05.09tomcontr3Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/223-08d32358", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
22:05.21tzafrirwell, actually the location of xpp_usb is irrelevant, as its spans will always be last
22:05.51tzafrirjameswf, and anyway, this is only good for new installations. So you still have to handle existing ones
22:06.02mike-ekimsweet. good to hear
22:06.33jameswfpeople really like to be bleading edge so 6 months tops heck the rhino adds to genzap have allready spread
22:06.48jameswfhas to start somewhere
22:07.16tzafrirjameswf, why should your post script edit a user's configration file?
22:07.54tzafriror to rephrase: how can you tell if it is an original stanza or a user-edited configuration file?
22:08.29jameswfwe append to that file commented  except analog as users will call and say your card isnt working.... when in fact they never loaded it
22:09.16tomcontr3Ok... Im almost there...
22:09.27jameswftzafrir well this is why i suggest a split then appending to a zaptel.analog I dont have todepend on a file status
22:09.37*** join/#asterisk craigk (n=craigk@58.174.150.119)
22:09.41tomcontr3I can make outside calls using channel 1  but when I try to use channel 2  I cant
22:10.27jameswfI can append fully commented and tell people in instructions to simply uncomment xyz at the end of the file
22:11.12jameswfnote: this is RPM only source builders dont get any of this...
22:11.15tzafrirSeparate files are a gross overkill
22:11.50tzafrirIt creates too many oportunities for strange user errors
22:11.52jameswfI wouldnt say so as that is how apache and quite a few others do their configs...
22:13.19tzafririsn't it silly that we work so hard to configure the obvious?
22:13.31Rico29is it really useful to check "alaw" & "ulaw" sounds in menuselect when installing asterisk ?
22:13.42Rico29same thing for "music on hold"
22:13.43tzafrirThe hardware is already there.  The information is already there.
22:13.45Rico29?
22:14.18jameswftzafrir if all users were technical people we wouldnt need much of anything
22:14.46tzafrirRico29, it helps to add two more further download speed tests
22:15.09Rico29ok
22:15.12Rico29thanks
22:15.13jameswfwell then can you just ad our studd in the appropriate place... the RPM detects the lines and wont double em....
22:15.14tzafrirjameswf, for starters: kernel/xpp/utils/zaptel_drivers
22:15.26*** join/#asterisk PepOSX (n=angeldav@190.72.130.192)
22:15.31tzafrirThough the ordering might need some improvements
22:15.42jameswfbah cant type
22:15.57Rico29can an admin add in the topic that there is a #asetrisk-fr chan for people who speak french ?
22:16.01jblackThe odbc connector for postgres is broken. If I use res_pgsql instead, will I still use the app_xxxxx_odbc modules?
22:16.03Rico29it would be nice
22:16.38hescoI've got this working with an iax2 device configured with voipstreet, but when I try a sip device on the same account, I see this:
22:16.41hescoExecuting Answer("SIP/rcr08-voipstreet-sip-gsm-08153e30", "") in new stack
22:16.50hescofollowed immediately by this:
22:17.09hescoNOTICE[16123]: pbx_spool.c:279 attempt_thread: Call completed to SIP/rcr08-voipstreet-sip-gsm/17707551543
22:17.39hescowell, actually it first says: Executing WaitForSilence("SIP/rcr08-voipstreet-sip-gsm-08153e30", "2300|""|38") in new stack
22:17.57hescoWaiting 1 time(s) for 2300 ms silence
22:18.37hescoas soon as this extension answers it terminates the call.  Why might that be?
22:19.03tzafrirjameswf, with such a schema you won't need the editing
22:28.00tomcontr3for some reason my zaptel o asterisk is not taking channel 2 to dial out
22:30.22jblackoh, for crying out loud
22:33.18*** join/#asterisk weazahl (n=jeremy@adsl-76-230-116-17.dsl.ksc2mo.sbcglobal.net)
22:36.24*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
22:39.43Rico29problem when installing asterisk addons : checking for mysql_config... no. And I need it for installing realtime
22:40.46Rico29i'v installed mysql_client, as it's said in the make menuselect for the addons
22:40.46*** join/#asterisk rhombus (n=sfbosch@dsl-vlan435-66-18-218-36.nucleus.com)
22:40.47*** join/#asterisk henrique (n=henrique@unaffiliated/henrique)
22:41.20rhombusWith iaxmodem, incoming faxes are squashed
22:42.02Rico29a packet was missing
22:42.05Rico29(again)
22:49.35*** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca)
22:49.43VOiCiHi, where can i setup some AGI scripts on Freepbx webgui
22:55.16jameswfneat http://pressbulletinboard.nokia.com/2008/03/26/smart-choice-for-the-home-office-the-new-nokia-6300i-with-mobile-voip/
22:57.10*** part/#asterisk seme (n=seme@rrcs-208-105-67-178.nyc.biz.rr.com)
22:58.40Rico29jameswf > I want it !
23:01.18lesouvageIs there any specific reason that an announcement for the called party isn't recorded when monitoring the call and there is just silence on the recording during the announcement.
23:02.28jameswfsure
23:03.35jameswfthere is a  specific reason for everything the question is what is that  specific reason
23:08.51lesouvagejameswf: Do you have any clue?
23:12.13lesouvagejameswf: or any pointer that might lead to the specific reason and a way to fix the problem?
23:17.11*** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com)
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23:23.49*** part/#asterisk km- (n=pgrace@aeneas.fierymoon.com)
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23:53.31*** join/#asterisk cleone (n=cleo@adsl196-131-185-206-196.adsl196-6.iam.net.ma)
23:55.20cleonedoes Asterisk worck fine on openbsd
23:55.44*** join/#asterisk eurowerke (n=bmrjunk@paul.shepnet.org)
23:55.47eurowerkehey all
23:56.02eurowerkeis AsteriskNOW based on any particular distro?
23:56.11*** join/#asterisk jmhunter (n=jacob@72.14.86.68)
23:56.11*** mode/#asterisk [+o jmhunter] by ChanServ
23:56.16eurowerkecan't seem to find out by JFGI
23:58.56*** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net)
23:59.44riddleboxeurowerke, try #asterisknow, or #asterisk-gui

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