irclog2html for #asterisk on 20050805

00:00.02jerosip is said to be mad when using more than one level of nat
00:00.24*** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net)
00:01.16*** join/#asterisk meppl (~mephisto@84.245.165.130)
00:03.31jskcrjero, Ive double natted it before
00:04.08jeroi guess it should work with stun and symmetric communications
00:04.46RandomAndyThanks Qwell. Finally a clue. I'm trying to figure out how to read that iptables output. It doesn't saying anything about ports in it...
00:06.03Qwellthen none are open
00:06.08frenzy[02:35] frenzy: how to enable B2BUA on Asterisk?
00:09.02QwellIsn't asterisk a b2bua?
00:09.49Qwellhttp://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+not-proxy
00:12.04frenzyI want the SIP to only signal
00:12.21frenzythe rtp stream to flow direct between UAs
00:12.58*** join/#asterisk Luke-Jr (~luke-jr@CPE-24-31-252-99.kc.res.rr.com)
00:13.21*** part/#asterisk ClubBarf (~me@host-87-74-0-72.bulldogdsl.com)
00:13.29Qwellfrenzy: You want it a reinvite?
00:13.32Qwellit to*
00:14.37QwellDon't msg me
00:14.52Qwellespecially not in freaking color
00:14.55*** join/#asterisk kshumard_home (~ksh@pcp08979908pcs.huntsv01.al.comcast.net)
00:15.36RandomAndyQwell: I lied. There is a line "ACCEPT tcp -- anywhere anywhere          state NEW, RELATED, ESTABLISHED tcp dpt:4569". I haven't found any lines that match udp though.
00:16.02QwellRandomAndy: You need to add one then.  Find your firewall script
00:16.12Qwell(or make one...I suggest quicktables)
00:16.14frenzywhat doee reinvite do?
00:16.23frenzydoes**
00:17.16Luke-Jrfrenzy: IIRC, tells both parties to connect directly
00:17.31Luke-Jrany recommendations for origination service?
00:17.32RandomAndyQwell: Thanks. I'm using Suse so I'll check Yast first and make sure I didn't overlook upd there. Thanks for the suggestion.
00:17.42tuxinator_linuxMRandomAndy: You can also do it in 'setup'
00:17.44frenzyIIRC?
00:17.51Qwell~iirc
00:17.51jbotsomebody said iirc was "if I recall correctly"
00:18.23Qwellfrenzy: are you behind a NAT or anything?
00:18.29frenzyyap
00:18.37frenzyNot the Asterisk
00:18.39frenzymy phone is
00:18.42QwellThen canreinvite isn't easy
00:18.45tuxinator_linuxMjbot: You are so smart!
00:18.46jbottuxinator_linuxM: what are you talking about?
00:18.47Netgeeks8-/
00:18.47Qwellerm, reinvite
00:19.03Qwellthe option would be canreinvite=yes, but...yeah, nat sucks
00:19.43*** join/#asterisk jsaunders (jsaunders@S01060060971c5817.vs.shawcable.net)
00:19.47jsaundersWhat does "Forbidden - wrong password on authentication for INVITE to '"Robert Pogue" <sip:asterisk@70.70.36.136>;tag=as3aa408ad'
00:19.48fugitivoany iax softphone recomendation? (not firefly please)
00:19.55jsaundersmean?
00:20.11Qwellfugitivo: I like iaxcomm
00:20.41fugitivoQwell: does it have transfer and hold?
00:20.42frenzycanreinvite goes to [general]?
00:20.56frenzyor [<user>]?
00:21.20Qwellfugitivo: yeah, think so
00:21.21frenzyin the sip.conf
00:21.41fugitivoQwell: is it userfriendly? i need it for a person that doesn't know too much about computers
00:22.13Qwellfugitivo: yeah, just a few buttons
00:22.29fugitivook, thanks
00:23.35jsaundersAnyone know what "Forbidden - wrong password on authentication for INVITE" means?
00:25.18shaneQit means wrong password!
00:27.48Saaibhi all... i'm testing the following confinguration:  Client1->FW->Inet->FW->Asterisk->Client2, if Client2 calls Client1, Client1 can listen , but can't transmit
00:27.53jsaundersThe client is registering properly w/ *, proper secret.
00:27.58Saaibis there anything that needs to be configured on the client side ?
00:28.01jsaundersHappens when I attempt an outbound call.
00:28.25frenzylater folks..
00:28.57glm2kjsaunders: check your outbound context
00:29.02Cybertoyanyone know a good provider where you can send faxes through as well?
00:29.04frenzyhow do I turn off my PC?
00:29.09glm2klol
00:29.12Cybertoypreferably with pay as you go plan
00:29.12frenzyhahhaa :)
00:29.27NetgeeksFreny: What kind of computer is it?  Laptop, Desktop, Handheld?
00:29.29Saaibon the FW from Client1 i can see SCR=Client1IP DST=Client2IP
00:29.45glm2kCybertank: faxes need a lossless codec IIRC
00:29.52glm2kso...Vonage?
00:29.55Saaibof course Client1 can't connect to Client2IP directly, so, anything i need ?
00:29.55frenzyNetgeeks: What do those mean?
00:30.08frenzyI know the pc is on my desk
00:30.14NetgeeksFrenzy: do you have a gun close by?
00:30.17glm2klol
00:30.37frenzyI have to check if its loaded
00:30.43frenzywhat do you want to do with it?
00:31.10frenzybaaaahh byyyeee
00:31.15Netgeekslol, stopping here
00:31.39jsaundersglm2k:  No outbound context.  Just use a Dial(SIP/${exten}@1.2.3.4).
00:31.49jsaundersErr, ${EXTEN}
00:31.50frenzylater... Netgeeks: caffine is a killer
00:31.51QwellNetgeeks: should've gone one more step...
00:32.33Netgeeks;)
00:32.52frenzyQwell: PMs are not always so bad :P
00:34.09glm2ki believe the format should be: username:password@${EXTEN}@1.2.3.4)
00:34.29iCEBrkrhrrm.
00:34.35iCEBrkrApparently voicepulse is dead
00:34.55jsaundersThe SIP gk doesn't need a password.
00:35.34jsaundersIP based authentication
00:36.08jsaundersAnd it's to pstn, not a sip ua.
00:36.25glm2kjsaunders: what do the logs say?
00:37.36jsaundersglm2k:  Same.  "Forbidden - wrong password on authentication for INVITE"
00:39.53*** join/#asterisk mjmac (~mjmac@mjmac.active.supporter.pdpc)
00:41.16*** join/#asterisk pressure_man (~pressure_@ip-202-37-228-1.internet.co.nz)
00:41.50pressure_manhello, can someone offer some insight into have sip friends in multiple contexts?
00:42.19shaneQlaterz..
00:42.24*** part/#asterisk shaneQ (~shane@CPE000f66913259-CM001225d88588.cpe.net.cable.rogers.com)
00:42.29pressure_mani've seen in the README.extconfig in CVS HEAD, a sip peer/user having multiple contexts specified
00:42.33mjmacanyone else having trouble getting to VPC?
00:42.59mjmactheir dns seems slow, and traceroutes are wacky
00:43.02Ariel_yes I am having problems with voicepulse as well
00:44.05mjmacheh...  weird coincidence.  i was just testing something else, and my wife started complaining.  "you broke the phone again!"
00:44.31Ariel_glm2k, the dial should be exten => XNXXNXXXXXX,1,Dial,(IAX2/username:password@ipaddress/${EXTEN})
00:44.49mjmacanyone here had any success running a tdm400p in a soekris net4801?
00:45.08Ariel_mjmac, how are you going to connect the power plug?
00:45.48Ariel_VPC web site also seems down.
00:45.56mjmacAriel_: i have a separate 12V supply
00:46.08*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
00:46.10Ariel_wonder if I can port my number I have with them to another voip outfit.
00:47.32*** join/#asterisk kingtux (~susekid@pool-151-196-126-24.balt.east.verizon.net)
00:47.55glm2kAriel_: yep. thanks.
00:48.29*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
00:48.30*** mode/#asterisk [+o twisted] by ChanServ
00:50.52twistedwtf yo
00:50.54twistedtoo quiet in here
00:51.03shidoBOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOo!
00:51.08twistedwhoa
00:51.09twistedoka
00:51.33mjmaci didn't think anyone else wanted to hear my totally 80s flashback hits
00:51.34Ariel_twisted, how are you tonight?  it's not slow it's nice.
00:52.45Ariel_ahh there offering wireless internet service in my area for 15.99 by a company called thebluezone. Any one heard of them?
00:53.22glm2kjsaunders: chan_sip.c says that's possible result of a 403 Forbidden error.
00:54.21NetgeeksTwisted: I added the comment I was trying to tell you last night onto the mantis item 4832... which was shortly snuffed by some guy named nick....  take a look when you have time
00:55.21Ariel_don't you like that word... snuffed....
00:56.35hardwireAug  4 16:56:01 NOTICE[31644]: chan_sip.c:7654 handle_request: Failed to authenticate user <sip:1111@voip001.corp.tdxnet.com>;tag=ydt8zwbrt2 for SUBSCRIBE
00:56.38hardwirethis is killing me
00:56.40hardwire1.0.9
00:56.46hardwiredo IU have to spec a subscribe context?
00:57.00Netgeekswhat version of asterisk, hardwire?
00:57.04hardwire1.0.9
00:58.11hardwireit auths just fine over sip
00:58.13hardwirejust not for subscribe
00:58.28hardwireI am totally missing something.. this was working a good few days ago but I reinstalled completely :)
01:00.22*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
01:00.24blitzragehey hey hey
01:00.30Qwellblitzrage: y0
01:00.30twistedho ho ho
01:00.35blitzrageQwell: ahoi!
01:00.40Qwellblitzrage: I didn't need it afterall. :)
01:00.42*** part/#asterisk pressure_man (~pressure_@ip-202-37-228-1.internet.co.nz)
01:00.50QwellI got it taken care of this morning
01:01.31NewSoleOk I got a crazy question..... who here lives in ontario canada area
01:01.34blitzrageQwell: awesome! great to hear. So I'll be meeting you in October then?
01:01.36Ariel_hardwire, you said it auths over sip just fine what does that mean?
01:01.41Qwellblitzrage: indeed
01:01.45blitzrageNewSole: I do (Toronto)
01:01.48twistedokt0b3rf35+
01:01.50Qwellblitzrage: and I'll have to make good on that beer. ;]
01:01.51hardwireit auths the sip line jhust fine (on the phone)
01:02.01hardwireso it shouldn't be rejecting a secret based auth..
01:02.07blitzrageQwell: w00t! I already owe JerJer two :)
01:02.12hardwireprobably just giving me a generic failure to subscribe w/in the right context
01:02.24Qwellblitzrage: I owe alot of people alot of beers.  I'm hoping they'll forget. :p
01:02.40loudany news about voicepulse ?
01:02.49blitzrageNewSole: ahhh, sorry, already been offered a couple of those and had to turn them down (unless the pay was too rediculous to turn down :))
01:02.52Ariel_Qwell, go down to the corner get a case and drop it off.
01:02.53blitzrageQwell: LOL
01:03.00QwellAriel_: yeah, really...
01:03.02Ariel_loud, no still down
01:03.26NewSoleabout 5k per month before taxes
01:03.32loud:( thanks.
01:03.46QwellNewSole: shit, where at?
01:03.52blitzrageNewSole: wow, thats not too shabby :)
01:03.53QwellI could use a new career
01:03.59Ariel_I could use a job as well
01:04.24blitzrageI have no idea how much I'm making this year... I'm crazy for being self employed I think :)
01:04.35blitzrageNewSole: could you send me a job description?
01:04.40NewSoleWe need someone that can be at office and do work needed on servers
01:04.48blitzrageNewSole: where you located?
01:05.07NewSoleon your local MSN
01:05.24blitzrageNewSole: really? are you on my MSN?
01:05.44NewSoleyour msn up
01:05.46blitzrageNewSole: I'm not at home right now (I'm puttied into my PBX at home which also runs my irssi client in screen)
01:06.01blitzrageNewSole: will be home in about an hour
01:06.14NewSolek cause u are on my MSN
01:06.42blitzrageNewSole: really? crazy... whats your nick?
01:07.11NewSoleonly problem is we need someone to come to office for meetings in ottawa too
01:07.20NewSoleMike
01:08.34jeroi can come to your office if you want :)
01:08.45blitzrageNewSole: ahhhh gotcha - travels not a problem via Via train :)
01:08.57NewSole:P
01:09.36NewSoleQwell... as for career you would be verry impressed at quality we have
01:09.55QwellNewSole: anything's better then what I've got now. ;]
01:10.03blitzragebrookshire: j00 'round?
01:10.18blitzragedamn... that was harder to type than "are you around?"
01:10.41NewSolewe got unlimited North America for $26.95 per month CDN
01:10.46blitzragetwisted: octoberfest y0! :)
01:11.09NewSolefor persoanl accounts
01:11.53Netgeeks:))
01:18.40*** join/#asterisk zotz (~zotz@24.231.36.100)
01:21.55droothwhere are the asterlink guys??? nobody is around. did they all drop off the earth
01:21.59*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
01:22.06Qwelldrooth: Chicago...close enough
01:22.16loudclueing
01:22.17NewSolegot booted
01:22.23droothit's happening now???
01:24.21Hmmhesaysi'm at cluecon
01:24.27Hmmhesaysneed a ride to the strip club
01:24.38NewSolelol
01:24.46droothsomeone go over to the asterlink booth
01:24.47*** join/#asterisk Tili (Tili@202.105.67.81)
01:24.54Hmmhesaysbooth?
01:24.59droothdo they have one?
01:25.01twisteddrooth, i'm sure your problem is not as big of an emergency as you think
01:25.02droothi would have assumed?
01:25.06Hmmhesaysone is busy, one is getting drunk
01:25.14droothtwisted: thanks for the comment
01:25.21Qwellgetting drunk ~= busy
01:25.24Hmmhesaysone is guarding
01:25.36Netgeeksand one is stripping?
01:25.37Hmmhesaysbut in an hour, we'll all be at the nudie bar
01:25.42Hmmhesaysthe full nudie bar
01:25.44twistedHmmhesays, even bkw?
01:25.44HmmhesaysI hope
01:25.44QwellNetgeeks: ^
01:25.57Hmmhesayshe's got a car
01:26.10NetgeeksOnly nudie bard in Chicago worth the entrance cost is The Admiral's Club
01:26.26Netgeeksbar, even
01:26.43NetgeeksNext Cluecon needs to be in Ybor City
01:26.48hardwiregrr
01:26.49hardwiresubscribe this
01:26.57Netgeeks:))
01:27.14twistedhey Hmmhesays, what would you say attendance is at?
01:28.02hardwireNetgeeks: you think subscribe in 1.0.9 only works for Zap?
01:28.08Hmmhesaysmediocre
01:28.19twistedHmmhesays, i meant in numerical form :)
01:28.29twistedhardwire, subscribe is a sip message... how would that work for zap?
01:28.29Hmmhesays50 about
01:28.34Qwell50?!
01:28.38hardwiretwisted: err.. hints
01:28.40twistedHmmhesays, so it's more like a "summit" than a conference
01:28.45Netgeekshardwire, no.  What I think might be the case is that subscribe in 1.0.9 doesn't work period
01:28.53hardwireNetgeeks: I can show subscriptions
01:28.58jsaundersScrew that.  Cluecon @ Vancouver, BC.  Aww yeah.
01:29.00Hmmhesaysmore or less
01:29.06hardwirevoip001*CLI> sip show subscriptions
01:29.06hardwirePeer             User        Call ID                URI
01:29.06hardwire10.10.50.200     1111        3c26700d2e63-5kit1cwp
01:29.07hardwire10.10.50.198     1112        3c26700c5573-gyuoj2b3
01:29.09hardwirethey finally authenticates
01:29.11Hmmhesaysits the people that know what the hell they are doing already
01:29.15hardwireI needed to add some mojo to the Snom 360
01:29.20twistedHmmhesays, ahh.. so a big party ;)
01:29.24Hmmhesaysmore or less
01:29.34puowvipsigh.
01:29.50Netgeeksplease rephrase to "It's some of the people that know what the hell they are doing already"
01:30.19twistedNetgeeks, i know that I know what i'm doing already, i don't have to be reassured ;)
01:31.19NetgeeksI only sorta know what I'm doing, so reassurance is fine with me
01:31.44twistedhehe
01:31.54twistedokay, let me rephrase
01:32.07Netgeeksbut I do know that I have sip clustering using realtime working including NAT....
01:32.07glm2kAriel_: iax2.conf says i should use iax.fwd.net while the docs online say iax2.fwd.net, which is which?
01:32.08twistedI know MOST of what i'm doing already
01:32.30twistedNetgeeks, re-routing if it goes unreachable?
01:32.55Netgeeksyes, I basically re-route over iax to the hosted system when the net directive is positive
01:33.01twistedyea
01:33.03Netgeeksnat directive
01:33.08Hmmhesayshahahahahahahahah
01:33.14Netgeeksof course I'm doing all this in dial plan
01:33.22Hmmhesaysyou guys want to hear something completely off topic?
01:33.26twistedthat's kinda what i was getting at with mine, but I'd like * to be intelligent enough to do it itself ;)
01:33.27Netgeeksabsolutely
01:33.38twistedI may write the patch to do it, assuming anthm's goes i
01:33.40twisted*in
01:33.44glm2khmmm, there's iax2.fwdnet.net as well...
01:33.53NetgeeksI also have remote trunks working as well - same idea, find the trunk, re-route
01:33.54twistedHmmhesays, fire ;)
01:34.24Hmmhesaysok, this female friend i've been seeing, she blew me off completely, now that I'm out of town she's calling and leaving messages every day
01:34.27HmmhesaysLOL
01:34.30HmmhesaysI love it
01:34.40twistedHmmhesays, absence makes the heart grow fonder
01:34.46glm2kaye
01:34.49Netgeekscall her back and leave a message while in the strip club
01:34.51twistedeither that or they move along quicker ;)
01:35.11Hmmhesaysindeed
01:35.20*** join/#asterisk Twister (Twister@216.30.232.108)
01:35.23NetgeeksHowever, that action may not result in the response you desire
01:35.36glm2kexten => ${EXNUM},1,Playback(soundsoflovemaking)
01:36.49Hmmhesaysmaybe I shouldn't have taken that big ol' duker in her toilet on the first date
01:36.55Netgeeks"Stabbed in the back lately?  Our 5 star trauma center hosts the best doctors and equipment in the entire state of......."
01:37.27*** part/#asterisk mkrufky (~mkrufky@user-12lcl1s.cable.mindspring.com)
01:38.10Netgeeksoh-hold advertisement for e911....
01:38.23Hmmhesaysok she signed back in
01:38.28Hmmhesaysi'm going to ignore her
01:39.34Netgeeksstill hanging around, Twisted?
01:39.36Twisteranyone ever worked with Cisco 30 VIP Phones (or maybe this is a cisco thing in general) My phone displays Program Update and in the CLI it says Starting skinny session from ip (its the correct ip) Device SEP(mac) is attempting to register --Device 'florian' successfuly registered Requesting capabilities version request Requesting CapabilitiesRes
01:43.10*** join/#asterisk brc__ (~DarthClue@brc.base.supporter.pdpc)
01:48.13*** join/#asterisk brc__ (~DarthClue@brc.base.supporter.pdpc)
01:53.40*** part/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net)
02:00.09*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
02:00.17shmaltzhelo every1
02:03.05*** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo)
02:03.28*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
02:16.48DaminAnyone alive?
02:17.19*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
02:18.55hardwireok
02:18.56hardwiredamnit
02:19.03hardwirenotify for zap channels 100% okey dokey
02:19.06shmaltzDamin, I am
02:19.10hardwirenotify for hinted sip channels.. no okey dokey
02:19.12hardwirehmm hmm hmm
02:19.17Darwin35almost back up and running
02:19.42kingtuxQuestion!!!  I've setup a callback system with my server using DISA....
02:20.18kingtuxProblem I'm having is when my system calls me back DISA doesn't give me enough time to dial
02:20.34Darwin35just have it drop a call file and make it call you back
02:21.02kingtuxI have that working
02:21.04kingtuxwith DISA
02:21.33kingtuxWhen disa drops me to a dial tone it dials out the number before i'm finished dialing the number
02:22.22kingtuxhow can I set more time to dial
02:22.23kingtux??
02:22.42Darwin35once oyu put your password in disa it should not dial till you enter a nmbr\
02:23.00kingtuxfor some reason its dialing out??
02:23.03kingtuxdon't know why
02:23.04Darwin35disa is mainly a admin tool
02:23.12kingtuxadmintool??
02:23.18Darwin35then you have something majorly screwed
02:23.42Darwin35made so you can dial in from outside and use the system as if you where in the office
02:23.59*** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985538.sympatico.ca)
02:24.01hardwire~asterisk
02:24.01jbotrumour has it, asterisk is the symbol that looks like a star (shift-8 on north american English keyboards)
02:24.05hardwire~asterisk-cvs
02:24.07hardwireheh
02:24.10Darwin35you need to reread your dial plan
02:24.12hardwireGOTCHA YOU DAMN BOT!
02:24.23kingtuxWell i'm using it for cutting down on my cell phone since i have free incoming
02:24.51kingtux[disa-custom]
02:24.51kingtuxexten => s,1,Answer
02:24.51kingtuxexten => s,2,DigitTimeout,5
02:24.51kingtuxexten => s,3,ResponseTimeout,10
02:24.51kingtuxexten => s,4,Wait(5)
02:24.52kingtuxexten => s,5,Authenticate(1111)
02:24.54kingtuxexten => s,6,DISA,no-password|from-internal
02:25.01kingtuxthats my code
02:25.03Darwin35~pastebin
02:25.04jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
02:25.11kingtuxdoes that look correct
02:25.12kingtux??
02:26.56Darwin35brb
02:27.07kingtuxNO
02:29.05*** join/#asterisk SplasPood (~jwb@brooklyn.paravolve.net)
02:29.12jeffikcan anybody look at a pastebin post?
02:29.35Nuggetyes, anybody can.
02:29.37jeffikjbot: ha ha just say your pos
02:29.49jeffikNugget: ok, so would you?
02:29.54Nuggetsure  :)
02:30.15jeffikNugget: thanks: http://pastebin.ca/19351
02:31.19jeffikit's cli output. when i point my DID to an extension it's ok, when i point it to digital receptionist it goes nowhere, and digital receptionist works incoming from other trunks from different providers
02:32.11kingtuxwho's got some cheap NY dids??
02:32.18kingtux516
02:32.45*** join/#asterisk SarahEmm (~sarahemm_@2.35.220-216.q9.net)
02:33.20shmaltzI have a PRI from AT&T, with some DIDs from them, I have an IVR for one of the DIDs, someone forwarded a phone number to that this DID, and DTMF doesn't work, if they call directly the DID it works, what might be the problem?
02:33.43shmaltzI have no clue what type of phone the forwarded (could be VoIP)
02:34.23jeffikkingtux: just started using les.net got a 718 ritght away
02:34.36jeffik2.5/mo+.016.min
02:35.42kingtuxhow are thier prices
02:35.58*** join/#asterisk brc__ (~DarthClue@brc.base.supporter.pdpc)
02:36.20Twisterwhen * records a call where does it store it by default?
02:37.09shmaltzTwister, have tried the wiki?
02:37.29jeffikshmaltz: got these guys from the wiki
02:37.54shmaltzjeffik, what?
02:37.54jeffiklook for les.net
02:38.21jeffikshmaltz: sorry wrong message
02:38.43Hmmhesaysfile
02:38.50loudles = iax ?
02:38.51kingtuxDamn 16cent amin
02:38.57HmmhesaysI SUMMON THEE
02:39.14kingtuxtelasip unlimited for 14.95
02:39.22shmaltzTwister, http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Monitor
02:40.43*** join/#asterisk Sedorox (~Brandon@sedorox.staff.smartserv)
02:40.47*** join/#asterisk _solstice_ (~xhackers@iphost-64-56-144-213.cgy.wiband.net)
02:40.49shmaltzlooks like Twister doesn't have a manners, comes in here, asks a question and doesn't even let anybody help him
02:41.18_solstice_Hi, I was woundering if anyone ever setup asterisk to trunk with another IP phone system like Televantage over H323 ?
02:42.04Hmmhesaysi'm doing sip to h323 with transcoding
02:42.08_solstice_I am having a problem that i am only getting 1-way audio .. I can hear it on a sip phone connected to my asterisk box, but the user connected to the televantage can't hear me. I setup H323 dialing in the extentions ..
02:42.17_solstice_hmm transcoding .. how you set that up?
02:43.49*** join/#asterisk ComputerWarm (TestMasTer@d198-53-32-174.abhsia.telus.net)
02:43.56ComputerWarmHello all anyone here use oh323?
02:44.22glm2ki have plans for oh323...after i get my fwd incoming to work...
02:44.41_solstice_I should probly be clear... I am using a sip phone (eyebeam) on asterisk ... and us G711 to connect.. it uses G711, u-law 20 frames to connect to the Televantage, but only get one way audio.
02:44.53ComputerWarmI am getting errors when i am compiling hoping someone can help me out with them they are posted at http://pastebin.com/329682
02:46.00ComputerWarmJerJer are you around?
02:46.50ComputerWarmcould anyone plz help me with this problem
02:47.47ComputerWarmglm2k did you already install oh323?
02:48.09_solstice_ComputerWarm .. what version of asterisk are you running?
02:48.27glm2kComputerWarm: sorry. nope.
02:49.12*** join/#asterisk rene- (~rene-@201.144.60.209)
02:51.24rene-hi, im having problems detecting dtmf on a wildcard clone, i have relaxdtmf=yes and rxgain=20.0 because volume is very low (its only one extension but it has a dsl filter)
02:51.32ComputerWarm_solstice_ one second
02:51.45rene-outgoing dtmf and sip dtmf is flawless
02:51.56ComputerWarmalittle out dated i suspect CVS-HEAD-08/07/04-19:25:06
02:53.11rene-incoming dtmf as used to drive the ivr is almost never working right
02:53.44SarahEmmyou're using an X100p clone?
02:55.04Netgeeks<crickets>
02:55.04rene-thats right SarahEmm
02:55.09rene-three of them
02:55.22Qwell3?  Just get a tdm400p
02:55.26Netgeeksthree in one box?
02:55.39SarahEmmyeah, ewww.
02:55.50rene-i know, i would use anything else but customer is a cheapstake
02:56.01rene-i have disabled almost everything so i dont have irq issues
02:56.06ComputerWarmbut why would the version of asterisk have to do with the errors i am getting with compiling asterisk-oh323
02:56.08Netgeeksfirst thing I would do is remove two and them and see if that makes any difference
02:56.14SarahEmmrene-: from what i've heard, you will be unhappy with them heh
02:56.20Netgeekssee if you can get 1 working first
02:56.45_solstice_ComputerWarm, i compiled OH323 version 0.65 just fine with my version just fine .. just used to defaults to build it
02:56.46Netgeeksyou can fart around with txgain and rxgain settings, but you are just asking for problems
02:56.56*** join/#asterisk fugitivo (~ajf@168-226-244-58.mrse.com.ar)
02:57.17rene-Netgeeks, volume with setting at 0 is so low is ridiculous
02:57.21ComputerWarmok  where do i get that version from?
02:57.33NetgeeksAsterisk + analog cards (be it X100's or TDM's) = lots of client phone calls, IMHO
02:57.43rene-i know
02:57.50rene-iam testing a clipcomm 4*fxo
02:57.54rene-works well
02:58.10NetgeeksAs of about 6 months ago, if someone calls me and says they want me to do consulting work and it includes analog interfaces, I say no
02:58.27rene-thats a good policy, i just cant be that picky right now :)
02:58.32Netgeekskk
02:58.41NetgeeksI don't know what to tell you, Rene
02:58.55Netgeeksclones are inherently poor performers in a field of poor performers
02:59.20fugitivoNetgeeks: what problems are you having with TDM's?
02:59.25rene-i will install an original x100p and if that works then thats it either he switchs or i cant do anything else
02:59.37Netgeekscall progress, echo, channels getting stuck to name a few
02:59.57rene-channels get stuck all the time
03:00.15rene-but whatsya gonna do? not everyone can afford t1/e1
03:01.30fugitivoi don't have those problems
03:01.34SarahEmmrene: why can't you go with a VoIP ITSP?
03:01.42fugitivowhat TDM are you talking about?
03:01.54NetgeeksHow many analog installations do you have, fugitivo?
03:02.15*** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co)
03:02.17fugitivotwelve, including mine
03:02.44Netgeekshrm, I'd be interested in finding out how you solved the hung channels, echo, and hangup detection
03:02.58SarahEmmhangup detection should just be kewlstart, polarity reversal..
03:03.06blitzrageSarahEmm: ahoi hoi!
03:03.10fugitivoi've experienced echo problems with poor lines
03:03.18SarahEmmhiya blitzrage
03:03.36fearnortdms are so homogay
03:03.39Netgeeksmy installations are in places like Suriname, Paraguat, Kazakstan
03:03.45Netgeeksso poor lines are a given
03:03.54SarahEmmahh.
03:03.54NetgeeksParaguay even
03:04.15NetgeeksI had a system in pakastan, but it either got blown up or stolen
03:04.23fearnorterrorists attacking asterisk?
03:04.33Netgeekspakistan... I need to lay off the caffine
03:04.54rene-Sarah, im in mexico and vonage wont sell me mexico city dids since im not US based
03:04.58NetgeeksNah, more likely a power surge explosion...
03:05.13SarahEmmrene-: err, there's lots of ones other than vonage tho
03:05.27SarahEmmvonage doesn't let you use asterisk anyway so that's not a great solution to begin wtih
03:05.54fugitivoi had hung channels, while calling out using one zap channel to a loop ivr, flash, then call with another zap channel to the same ivr, hook again, and you have an infinite bridge between your zap channels
03:06.08rene-been there fugitivo:
03:06.15NetgeeksAlthough I've had a problem with hangup detection and echo using TDM cards to provide a VoIP link between two PBX systems  (company wanted to link a small office key sysytem to thier HQ PBX via VoIP over thier point to point data T1)
03:07.10NetgeeksSo given a perfect environment (I have three CO lines into my home office asterisk box) analog lines work just fine
03:07.18rene-i dont know if hangup detection works on phone systems outside the US
03:08.07fugitivoworks ok in argentina with tdm400
03:08.08NetgeeksLuckily, I'm finally in the position not to have to take on any work involving analog cards, and I'm happy that way
03:09.31NetgeeksI've got a number of clone cards around, but I'm just using them as timing sources, and nothing else
03:10.42SarahEmmworks great, but i'm not doing voice over it
03:11.22Netgeeksum, it's your main FXO but you aren't doing voice over it......  *boggle*
03:11.32Netgeeksfax?
03:12.00SarahEmmTTY/TDD
03:12.09Netgeeks!
03:12.25SarahEmm@
03:12.44blitzrage#
03:12.48Qwell$
03:12.54blitzrage*
03:13.02Qwellshit, I have to change my password now
03:13.06blitzragelol
03:13.08Netgeeks:))
03:13.13SarahEmmwhat's with the puncuation Netgeeks? ;)
03:13.31blitzragedouble chin!
03:13.33NetgeeksI didn't expect the TTY/TDD answer, thus ! = suprise
03:13.59Netgeekshey double chin face make trillian laugh like a little kid, which for some reason I find pleasant
03:14.17blitzragetrillian? ewww ;)
03:14.40Netgeekswindows = eww, but it's what I have on my right screen...
03:14.47blitzrageI like Windows
03:14.48SarahEmmNetgeeks: ahh. i'm working on lots of TTY stuff in *, fixing up the support and adding features and such
03:14.53blitzrageI hate Linux desktops
03:14.57*** join/#asterisk vuvie (~vuvie@bb219-74-44-131.singnet.com.sg)
03:15.06blitzragejust use Linux at the CLI via SSH and screen
03:15.40rene-bye all
03:15.51fugitivoi love my linux desktop
03:16.17hellopIt is not possible to search ebay for "tdm400".  It changes it to tm400.
03:16.31Qwellhellop: because it isn't called a tdm400
03:16.34Qwelltdm400p
03:16.43Qwellthe p stands for Pretty damn important
03:17.02blitzragelol
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03:20.46SarahEmmi've never seen one on ebay..
03:22.07glm2kno one's getting rid of them once they buy em hehe
03:23.26Netgeekshttp://photon.netgeeks.net/asterisk/Desk.jpg  <-- my work area
03:24.35QwellYou need a kvm like no other
03:24.49Netgeeksactually I use the center keybord and mouse and Synergy
03:25.05Qwelltip: You should hide notes from WFSF
03:25.10SarahEmmhttp://sarahemm.net/gallery/desk
03:25.12blitzrageNetgeeks: that software is great
03:25.19SarahEmm^-- my desk about 6 months ago... it looks somewhat neater now
03:25.25Netgeeksthe keyboard to the left is just there because I have to restart synergy now and then
03:25.59Netgeeksnice desk sarah
03:26.02Netgeeks;)
03:26.15SarahEmmit's... not tidy.
03:26.16SarahEmm:)
03:26.28blitzragemy desktop: http://leifmadsen.com/gallery/1406bedroom/H0030150_P
03:26.28kingtuxI'm looking for DIDs from El Salvador and Guatemala and alot of central and south america
03:26.31NetgeeksWFS is just my stack of freeby stickies from some company I must have interfaced with at some time
03:26.35kingtuxany know of some provides
03:27.06blitzragehttp://leifmadsen.com/gallery/1406bedroom/HPIM0161
03:27.43Nuggetyay beer
03:27.52blitzrageoh yah :)
03:27.57shidoooh
03:27.59shidoa lighter
03:28.01Netgeekswhat blurry city night skyline is that?
03:28.22blitzrageNetgeeks: Oakville, ON, Canada (just south of Toronto)
03:28.32Nuggethttp://slacker.com/photos/powermac/IMG_3931  <-- my desktop
03:28.52Netgeekssweet monitor
03:28.55blitzrageNugget: nice monitor
03:29.11blitzrageto bad you have a Mac though
03:29.19Nuggetfeh.  macs > *
03:29.36Nuggetunix and I can play games and run office.  best of all worlds.
03:29.38blitzragemacs < *
03:30.33Nuggethttp://slacker.com/photos/computers/SlackerNOC  <-- all the nonmac boxes are in the server room
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03:31.31blitzrageNugget: Budweiser?!  ugh
03:31.41Nuggethrm?
03:32.23Nuggetthat's ivo, not me.
03:32.41blitzrageahhhh gotcha
03:32.44blitzrageBud sucks :)
03:32.46NuggetI'm just making fun of him for drinking that swill.  :)
03:32.50blitzragecome to Canada and get some real beer
03:32.57blitzragelol
03:38.42*** join/#asterisk Craziman2 (~Craziman2@208.3.11.172)
03:39.19Craziman2Guys I am building a server to support 8-12 fxo ports, what brand/config would you reccomend?
03:39.47SarahEmmTDM400p cards are the likely option
03:39.54SarahEmmbut why aren't you using a PRI at that number?
03:40.14Craziman2Cost is the main reason
03:40.25SarahEmmwell, if you *have* to go POTS, tdm400p...
03:40.31SarahEmm3 cards stuffed with FXO modules would do 12
03:41.06*** join/#asterisk IronHelix (~irc@ool-45785cfe.dyn.optonline.net)
03:41.16Craziman2Yea... what brand PC would you recommend?  I have tried the Dell SC420's and they don't work so well.
03:41.38SarahEmmi'm not the person to ask about that, i'm running it on some old asus board
03:41.54Craziman2fun fun
03:42.45fearnorwhy dont they work?
03:42.45twistedwhoa
03:42.47twistednetsplit
03:42.53fearnordel sc420 are ghetto fabulous but they work just fine
03:43.08fearnorand you should really go with channel bank instead
03:43.11Craziman2IRQ Problems, clicking and popping on the line
03:43.19fearnorcause digium tdm cards are the homogay
03:43.27fearnorwell then, resolve your irq problems.
03:43.34fearnorproblem between the chair and keyboard.
03:43.37SarahEmmfenlander: he should really go with a PRI :P
03:43.57Craziman2can't... the system assigns the IRQ to the PCI slot and you can't change it.
03:43.59twistedblitzrage, molson is NOT real beer.
03:44.15fearnorcrazy: did you disable all unnecessary devices?
03:44.19Craziman2yes
03:44.21fearnorNEXT
03:44.25fearnor:)
03:44.30Craziman2Still popping and echo.
03:44.35fearnori'd love to see your /proc/interrupts
03:44.46fearnori'm telling you, you didn't disable all devices.
03:44.54fearnoryou don't need usb etc etc etc etc
03:44.59fearnorparport etc
03:45.02*** join/#asterisk likwid-- (~likwid@nc-69-68-77-240.dyn.sprint-hsd.net)
03:45.07Craziman2k... can I priveate message it to you?
03:45.13fearnoror post it on pastebin.ca
03:45.19Craziman2k
03:45.32Delta34doesnt ACPI take care of the interrupts
03:45.41twistedyou mean APIC?
03:45.41fearnoryeah that's another good pointer
03:45.49fearnorwell, APIC and ACPI both :)
03:45.49Delta34yeh APIC
03:46.00fearnorACPI is required to discover APIC usually ;)
03:46.47Delta34well u got lspci -v then u got lspci -bv
03:46.54Craziman2<PROTECTED>
03:46.54Craziman2<PROTECTED>
03:46.54Craziman2<PROTECTED>
03:46.54Craziman2<PROTECTED>
03:46.55Delta34which one should u go off
03:46.55Craziman2<PROTECTED>
03:46.57Craziman2<PROTECTED>
03:46.59Craziman2<PROTECTED>
03:47.01Craziman2<PROTECTED>
03:47.02*** mode/#asterisk [+q craziman2!*@*] by twisted
03:47.06twistedPASTEBIN
03:47.15*** mode/#asterisk [-q craziman2!*@*] by twisted
03:47.25fearnorthat actually looks about right.
03:47.28fearnordisable hyperthreading
03:47.31fearnorand you'll be cool.
03:47.42Delta34so thats APIC
03:48.35Delta34u dont need hyperthreading?
03:48.46fearnorthere are almost no cases when you actually need HT.
03:48.50fearnorHT is homogay
03:48.54fearnorbest to be disabled.
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03:49.22Netgeeksrest of my home office space: http://photon.netgeeks.net/desk/  (since you didn't get to see the phones I use and such)
03:50.39fearnorget some faster intarweb or scale images yo
03:50.55Netgeeksnevah!
03:50.56SarahEmmheh
03:51.19Netgeeksfear my charter cable 3m/384 business cable!
03:51.41Delta34ls
03:51.43Craziman2so you guys thing HT can cause interupt problems and thus popping?
03:52.07NetgeeksI did a test a while back for a client that wanted 4 4-port T1 cards in a single server
03:52.19fearnornetgeeks: crazy mang
03:52.22NetgeeksWe tested with HT turned on and HT turned off
03:52.42Netgeeksturned out that HT resulted in about a 10% worse performance for asterisk under those conditions
03:52.43*** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net)
03:52.52fearnorwerd.
03:53.08NetgeeksThats the only real world data I have on HT affects on asterisk
03:53.31fearnorwell, bottom line is, HT does not help except on very very few workloads
03:53.36fearnorand it may break lots of things.
03:53.43fearnorso, disable HT. ktnxbye
03:53.43NuggetMIATA REBORN!
03:53.56Netgeeksyeah, I just got my motor trend mag today
03:54.07Netgeekshaven't had a chance to read it yet
03:54.29NuggetI missed my miata so much that I talked my girlfriend into buying one.
03:54.43Netgeeksnever owned or even ridden in one  :(
03:54.44Craziman2fearnor :  Thanks I will turn on HT, I am just trying want to know if there are otherthings to try to get rid of popping.
03:54.51Nuggetthey're insanely fun
03:54.58Craziman2er I meant turn off :)
03:55.08SarahEmmlol
03:55.11SarahEmmon, off, same thing.
03:55.14fearnorwhy don't you first follow advice you are given and then come back
03:55.15Nuggetyou have voicemail, Netgeeks.  :)
03:55.25Delta34anybody know how to use sipp?
03:55.36NetgeeksI always have voicemail  :)
03:55.49SarahEmmDelta34: sipp?
03:55.55Delta34hp sipp
03:56.00Delta34to load test sip
03:56.29SarahEmmahh
03:56.52Delta34http://sipp.sourceforge.net/
03:57.24fearnorsipstoned
03:57.41Netgeeksyumm, jolly rancher, thanks for the reminder!
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04:05.08*** join/#asterisk Rowter (~SilverDra@dsl-201-129-88-148.prod-infinitum.com.mx)
04:05.20Rowterto register iax with fwd you need to generate a rsa key_
04:05.21Rowter?
04:05.24*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
04:10.40blitzrageRowter: nope - the FWD public key comes with Asterisk
04:10.49blitzrageRowter: you need to activate it on their website though
04:11.45NetgeeksSarahEmm: !!
04:11.56NetgeeksThat an HP/Apollo 7XX series?
04:12.20blitzrageApollo?  Terrible printers
04:12.32blitzrageused to sell those at a computer store I worked at in highschool
04:12.47SarahEmmNetgeeks: ??
04:12.53SarahEmmNetgeeks: what?
04:13.01SarahEmmNetgeeks: oh
04:13.04Netgeeksyou have an HP system standing up under your side desk
04:13.04SarahEmmNetgeeks: yeah, 735/125
04:13.09NetgeeksWow!
04:13.10SarahEmmwas a /99, upgraded it to a /125
04:13.18SarahEmmwas my main box at home for a long time
04:13.19SarahEmmwhy?
04:13.47NetgeeksWhen I worked at the Nuke plant, the first massive piece of software I wrote was on those
04:13.50SarahEmmwhat picture can you see that in?
04:13.59SarahEmmooh i see
04:14.03SarahEmmyeah, that's my HP :)
04:14.06Netgeeksnumber 5 of 8
04:14.08SarahEmmand an Indigo2 sitting beside it
04:14.20SarahEmmand various other boxen to the left/right of that you can't see heh
04:14.26NetgeeksI've never ever met anyone else who had one they actually used
04:14.29SarahEmmahh :P)
04:14.30SarahEmm:)
04:14.44SarahEmmyeah, that was my main machine. it's got decent amount of RAM in it, 125MHz PA-RISC, 100Mbps NIC (FDDI)
04:15.43NetgeeksI threw my two 735's away last move (from SF Bay up here to Oregon where I am now)
04:15.44MaarkenFDDI is weird stuff.
04:15.46Netgeeks:(
04:15.49SarahEmmNetgeeks: awwwww :(
04:16.18Netgeeks*nod* I have three Sun Ultrasparc II workstations now
04:16.33SarahEmmnot sure why... just never picked up any
04:16.52NetgeeksI got them as a going away present from one of the failed startups I worked for
04:17.12SarahEmmahh
04:17.23Netgeeksnot bad machines, 3GB memory each, dual 333MHz procs
04:17.41*** join/#asterisk cfrank_ (~cfrank@wsip-24-234-137-140.lv.lv.cox.net)
04:17.57Netgeeksbut they aren't doing much, one is running SER, one is running MRTG for my local net, and the other is just running up my power bill
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04:19.54Netgeeksyou running HP/OS on that thing?
04:19.54SarahEmmheh
04:19.57SarahEmmHP/UX
04:19.58SarahEmmyeah
04:20.16Netgeeks<-- still amazed
04:20.18SarahEmm10.20 iirc
04:20.25SarahEmmamazed at what?
04:20.38Netgeeksthat you are still using it
04:21.24SarahEmmahh
04:21.35SarahEmmwhy? because it's older?
04:21.39NetgeeksJust fond memories of them, but I pretty much concluded they had gone the way of the Dodo bird and I would never see one again, cept maybe in an HP musuem
04:21.43SarahEmmahh
04:21.49SarahEmmmy house is kind of a makeshift museum. :)
04:21.57SarahEmmi have a bit of older stuff.
04:22.06SarahEmmthat's not really old by my standards..
04:22.13SarahEmm'older than the kitrich' is old
04:22.32NetgeeksI was like that for a while, then I got married, and she turned my three "computer rooms" into guest bedrooms
04:22.37Netgeeks*sigh*
04:23.03SarahEmmahh
04:23.10SarahEmmy'see, my partner is a geek. :)
04:23.19SarahEmm(not that we live anywhere near eachother *sadface*)
04:23.49Netgeekscomputers got moved to the garage, and when we moved to Oregon, they got "discarded" but we moved from 3000 square foot house to 1900 sq. ft... so I didn't have anywhere to put them
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04:26.18Netgeeksoh, and nice to see the vacuum cleaner handle next to THAT desk... lol
04:26.42Netgeeksit's that kind of attention to detail that really sticks out!
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04:27.13SarahEmmlol
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04:27.29SarahEmmmy house is Much Cleaner now
04:27.38SarahEmmwhen my ex moved out i did a Lot of tidying...
04:27.46Netgeeksyou need some after pics
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04:30.23SarahEmmyeah, i have some but they're .. not public
04:30.25NetgeeksI hate working remotely on systems in central america.... I feel like I'm working on a 150 baud modem
04:30.27SarahEmmi should take some now
04:30.37Netgeeks:))  <--- not public
04:30.42SarahEmmmew?
04:32.53KaBewMmoo!
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04:33.37NetgeeksSarah is taking "pblic safe" desk pics now.....
04:33.43Netgeeks"public safe" even
04:34.24Netgeekslol
04:34.45SarahEmmlol
04:34.52NetgeeksI have no idea how to interpet her comment that she has recent pics of her cleaned up desk, that are NOT public safe
04:34.56SarahEmmthe ones taken immediately after the cleanup was done have pictures of a different ex in them...
04:35.08SarahEmmNetgeeks: lol. because they include a different ex in them
04:35.14SarahEmmhence, i need new ones without people in them
04:35.20Netgeeks*whew*
04:35.25SarahEmmlol
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04:38.08CherebrumAnyone know what the CALLING TYPEOFN and CALLING TRANSIT headers are for in an IAX2 packet?
04:40.46kshumard_homeCherebrum, http://splurge.peoples-wireless.com/iax/iax.html
04:40.57kshumard_homeCherebrum, that'll help. : )
04:41.12kshumard_home"The purpose of the CALLINGTON information element is to indicate the calling type of number of a caller, according to ITU-T Recommendation Q.931 specifications."
04:41.19Cherebrumthank you
04:41.37kshumard_home"The purpose of the CALLINGTNS information element is to indicate the calling transit network select of a call, according to ITU-T Recommendation Q.931 specifications."
04:41.39kshumard_homeno problem. : )
04:43.12CherebrumAnyone here at cluecon?
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04:49.07ComputerWarmhey anyone here that can help me with oh323 i am having nothing but problems trying to get it installed. http://pastebin.com/329728  I tried the older versions and now this one. all of them give me the same error i am unsure what i am missing
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04:51.35ComputerWarmanyone please.
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04:57.45NeonLevelgood evening
04:58.40NeonLevelI'm new to the Dundi and planning on installing it, is it hard? or maybe there's an step-by-step guide?
04:58.48NeonLevelthanks
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05:01.38Twisterin sip.conf when i do record_out=always is there a place i can change the filename or is that an *@home thing?
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05:10.38shidoreboot
05:13.37glm2kdoes no one man the 55555 fwd number?
05:13.53Qwellits just helpers, isn't it?
05:14.02glm2kbelieve so
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05:14.53glm2kbut, i mean, there are thousands of fed subscribers, surely there are a handful in each timezone available?
05:15.00glm2ks/fed/fwd
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05:16.59sudhir492Hi all
05:19.47*** join/#asterisk Katty (~Katrina@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
05:20.57Kattyhihi
05:21.07NetgeeksHello again Katty
05:21.25KattyAgain?
05:21.32NetgeeksThey didn't drag you to the strip club?
05:21.58Kattythat's a bad choice of words.
05:22.01Netgeeksyeah, I don't know why I said again
05:22.08glm2klol
05:22.11Kattylet's just say i was the one who wanted to go in the first place
05:22.17Netgeeks:))
05:22.21*** join/#asterisk MGSsancho (~sancho@ppp-67-126-131-62.dsl.irvnca.pacbell.net)
05:22.24Kattybut i'm too tired now
05:22.34Netgeeksawww
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05:23.18Kattyheh, it's not like anyone else was going to go
05:23.27Kattyi would've just been going by myself and possibly Hmmhesays
05:23.28Netgeekswhat?!
05:23.42Netgeekswhat kind of geeks are there?
05:23.57NetgeeksI thought all geeks jumped at the opportunity to go to strip joints
05:24.18MGSsanchoim would like to set up asterisk at work. and we have 3 existing phone lines. if i get the PCI card with the 3 green one (FXO i bleeive), what my boss wants is 8 people to make out going calls at once
05:24.30QwellMGSsancho: fxo is red I think
05:24.34SarahEmmNetgeeks: i don't tend to...
05:24.43SarahEmmNetgeeks: but i'm a dyke, and avoid any places with lots of people anyway.
05:24.47KattySarahEmm: mew :>
05:24.50SarahEmmkatty! MEOW!
05:24.53Netgeekslet me rephrase..  all male geeks...
05:24.53SarahEmm*pamples*
05:24.54QwellMGSsancho: You might consider getting a Digium T1 card, and a channel bank with FXO and FXS
05:25.00Netgeeksall straight male geeks....
05:25.01glm2kbingo. all male geeks
05:25.11Qwelland, 8 people calling at once, with 3 phonelines?  Thats not possible
05:25.18glm2knot possible.
05:25.24MGSsanchoQwell; we have 3 stanard analog lines
05:25.26KattyNetgeeks: it's kinda funny really... no one is talking to me but a handful of people...
05:25.30glm2kMGSsancho: you need 8 lines.
05:25.37Netgeekssomeone said you growled at them
05:25.43MGSsanchowe can do it curently with some toshebia 16dx crap
05:25.52SarahEmmuhh...
05:25.54SarahEmmno.. you can't
05:25.58SarahEmm3 POTS lines is 3 POTS lines
05:26.08Netgeeksso if thats the case, I wouldn't........ hrm, never mind, I would... growling girls are sexy
05:26.08SarahEmmyou can have 8 phones
05:26.11SarahEmmbut only 3 calls at once
05:26.11MGSsanchoi thought soo. let me talk to boss (dad)
05:26.20glm2khehe
05:26.27glm2kyou got it easy.
05:26.50KattyNetgeeks: i glared at mister bwk.
05:26.53Kattyi mean bkw
05:27.06KattyNetgeeks: he attempted to drag me into the middle of several drunken males and introduce me
05:27.13KattyNetgeeks: it was a very Bad Idea
05:27.21glm2kfor the males?
05:27.27Cherebrumwho's asterlink DID is 8003931078 ?
05:27.37MGSsancho>_>
05:27.40Cherebrumit hit my asterisk box about 3 times and then I dialed it
05:27.42Netgeeks<logs said bad idea away should I attend future events in which Katty is an attendee as well>
05:27.46Cherebrumand it was a meetme conference with 3 people in it
05:27.47MGSsanchoyeah looks like were going to use asterisk :)
05:27.57glm2knice
05:28.00QwellCherebrum: what makes you think its an asterlink did?
05:28.09CherebrumQwell: because the packet came from asterlink
05:28.19QwellCherebrum: Do you have an asterlink account?
05:28.22Cherebrumyes
05:28.24MGSsanchoi already submited 7 voip phones and a powered hub
05:28.27QwellThen thats...supposed to happen
05:28.44CherebrumI'm supposed to get calls to DIDs that aren't mine?
05:28.52KattyNetgeeks: ah well. I don't exactly fit in here anyway :)
05:28.53QwellIt was TO that DID?
05:28.57Cherebrumyes
05:28.59MGSsanchohow lomg does it take to set it up on average (i have average linux skills)
05:29.05Cherebrumit wasn't my DID
05:29.07Qwellwait
05:29.09Cherebrumbut it hit my box
05:29.12Qwella DID hit your box...
05:29.14NetgeeksKatty: Really?  why is that?
05:29.17Qwellbut when you dialed it, it didn't hit your box?
05:29.22Qwellsounds funky
05:29.22CherebrumAug  4 23:50:07 NOTICE[25013]: chan_iax2.c:6573 socket_read: Rejected connect attempt from 66.250.69.13, who was trying to reach '8003931078@'
05:29.31Cherebrumyea.. when I  dialed it I got a meetme
05:29.34Cherebrumand it wasn't my meetme
05:29.44KattyNetgeeks: It should be obvious. I'm not a developer of any sort. I'm not a contributor of any sort... I don't appear to be a geek, nor am I male....
05:29.57SarahEmmMGSsancho: depends on how complex you need stuff. * is half l33t un1x sk1llz and half telephony skills
05:30.00CherebrumI got 3 of those messaged
05:30.02Cherebrumer messages
05:30.06Cherebrumand then I dialed it
05:30.10Cherebrumand it was a meetme
05:30.11glm2kMGSsancho: depends on what you need to do
05:30.19QwellCherebrum: dunno, talk to file or bkw
05:30.38Cherebrumis file around?
05:30.56KattyNetgeeks: I don't know enough to ask an intelligent question if any of the speakers ask if there is a question. I've been nearly almost completely lost during all the talks... I don't even eat the lunch here cause i'm vegan.
05:30.57NetgeeksKatty: okay... I assumed ClueCon was more like a NANOG meeting.. guess not
05:31.10KattyNetgeeks: nanog?
05:31.28CherebrumKatty: are you at cluecon?
05:31.31SarahEmmnorth american networ operatiors groups
05:31.32SarahEmm-s
05:31.34NetgeeksKatty: North American Network Operators Group.  www.nanog.org
05:31.36MGSsanchocal waiting with music. umm standard stuff. caller id. and voice mail
05:31.37fearnorhah
05:31.41fearnornignog
05:31.42KattyCherebrum: yes
05:31.48CherebrumKatty: me too
05:31.50SarahEmmNetgeeks: beat you ;)
05:31.55KattyNetgeeks: I haven't any idea what that is :)
05:31.55Netgeeksit was the gathering meeting for Internet operators, mostly male geeks
05:32.00KattyCherebrum: (=
05:32.04Netgeeksbut then females started showing up
05:32.10Netgeeksand it got real fun then
05:32.11MGSsanchoO_O
05:32.22glm2kooohhh. females of the species...
05:32.24SarahEmmlol
05:32.40fearnornetgeeks: what the heck are you talking about
05:32.48fearnornanog has no women really
05:32.52KattyI'm not sure why people aren't talking to me, to be honest.
05:32.53Netgeeks<-- admits I only went to NANOG for the social interaction....
05:32.58KattyCherebrum: have you said hi to me yet?
05:33.06fearnorunless you are turned on by susan harris and betty burke
05:33.07Netgeeksfearnor, you just have to look around
05:33.28fearnorits just an excuse to get wasted and shit
05:33.37CherebrumKatty: I don't think so
05:33.41fearnorand listen to vijay gill
05:33.49KattyCherebrum: have you even seen me?
05:33.50QwellCherebrum: How can you not know for sure? :p
05:34.01KattyManxPower: (=
05:34.19CherebrumKatty: cause aren't you the only female here?
05:34.26KattyCherebrum: indeed
05:34.28Cherebrumhe heh
05:34.29SarahEmmawww
05:34.31SarahEmmkitriches should be there :P
05:34.35fearnorfunny thing is, #nanog has a lot of crossover with #asterisk
05:34.46SarahEmmthere's a #nanog?
05:34.49KattyCherebrum: why haven't you said hi yet?
05:34.50fearnorand same attitude
05:34.51Netgeeksthere were alot of women there actually....  <in respect to a great friend> Abha Ahuja, um, Ren Nowlin, lots of tech company sales girls who I can't remember names
05:34.52fearnorsarah, efnet
05:34.55SarahEmmahh
05:34.59CherebrumKatty: here is what I look like: http://www.flickr.com/photos/18631674@N00/
05:35.03fearnorok, respek to abha, and ren
05:35.15fearnoralthough i dont knew abha, and i dont know ren
05:35.17CherebrumKatty: I'll come say hi to you tomorrow
05:35.21KattySarahEmm: i'm basically wandering around with two people cause i feel so out of place :<
05:35.26KattyCherebrum: :>
05:35.27ComputerWarmhey anyone here that can help me with oh323 i am having nothing but problems trying to get it installed. http://pastebin.com/329728  I tried the older versions and now this one. all of them give me the same error i am unsure what i am missing
05:35.33NetgeeksAbha passed away a few years ago.... *sniff*
05:35.35SarahEmmKatty: awwww :o(
05:35.38CherebrumI've been hanging out with Kristian
05:35.44JerJerComputerWarm:  find the author of that code and complain to her
05:35.44SarahEmmKatty: can't get time off work :o( *sadkitrich*
05:35.50MGSsanchoif im using asterisk pots lines... can i have phone extentions. like extention 1234 rings the financial department?
05:35.53CherebrumHe's doing a presentation on astlinux tomorrow
05:35.57*** join/#asterisk dijungal (~ovr@206.113.106.126)
05:35.58Cherebrumer today
05:35.59KattySarahEmm: teh sukc :<
05:36.01QwellMGSsancho: sure
05:36.06dijungali'm BACK!!!.. :)
05:36.06glm2kMGSsancho: yep
05:36.14dijungalhey qwell.. :)
05:36.15SarahEmmKatty: agreed :(
05:36.24NetgeeksYeah, asterisk got presented to NANOg um, uh, like over 2 years ago
05:36.30dijungalhi netgeeks
05:36.31MGSsanchoand what if more than 3 people cal. will it keep ringing? or will they  hear a prerecorded message?
05:36.38MGSsanchohi
05:36.42QwellMGSsancho: over the pstn?
05:36.44CherebrumKatty: Did you see the little bluetooth contact message that poped up on File's laptop while he was on stage?
05:36.45QwellThey'll get a busy signal
05:36.51Cherebrumit said "hello.vcd"
05:37.00MGSsanchook thanks
05:37.03CherebrumI sent that from my phone.. heh heh
05:37.04KattyCherebrum: I haven't been around file's laptop
05:37.10KattyCherebrum: oh.
05:37.14KattyCherebrum: heh (=
05:37.18KattyCherebrum: read that wrong.
05:37.30Cherebrumwhen he was giving his presentation
05:37.30Cherebrumhe left bluetooth on
05:37.37QwellCherebrum: heh, hacker
05:37.48Cherebrumha ha
05:37.57CherebrumI sent one that said "CherebrumSaysHello.vcd"
05:37.59KattyI'm glad my boss didn't come with me. He would not be impressed
05:38.09Cherebrumbut he turn ed the projector off
05:38.15KattyPersonally, I like that it's relaxed and such....
05:38.20QwellCherebrum: Why .vcd?
05:38.23CherebrumKatty: Where do you work?
05:38.26MGSsanchoyes Qwell
05:38.33KattyCherebrum: I doubt you've been there (=
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05:38.37QwellMGSsancho: Then they'll get a busy signal
05:38.46KattyCherebrum: wouldn't you rather know what my job is?
05:39.04QwellMGSsancho: 2 outgoing calls, and 1 incoming call, would tie up all 3 lines
05:39.06MGSsanchomy dad likes it. just need to wait for ceo and people from jarcarta to come to the US and aprove my bidget
05:39.41ManxPowerYou're getting a bidet that require corporate approval?
05:39.44Cherebruma vcd is a contact card thing
05:39.44CherebrumwhatWhat is your job?
05:39.48MGSsancholol
05:39.50Cherebrumdamn wireless
05:39.55fearnorbidet++
05:39.56MGSsanchonet admin like most of us
05:39.58CherebrumI'm getting a CRAPPY signal in my room
05:40.06MGSsanchoautomotive parts distrobuter
05:40.14KattyCherebrum: basically, my company sets up an office. The furniture, copiers, faxes, printers, servers, networking, phone system, etc.
05:40.20MGSsanchocompany rapidly expanding.
05:40.26Cherebrumaha.. like officescape
05:40.35KattyCherebrum: the whole office...I take care of all the computer/networking/software/phone/ stuff
05:40.40MGSsanchothey hate their current phone system. $4000 POS
05:40.46glm2klol
05:41.15NetgeeksKatty: and you use asterisk for phone systems?
05:41.20KattyCherebrum: the ordering, the delivery/setup, installation and configuration, troubleshooting when they break and phone support too :)
05:41.22MGSsanchoonly like umm 9 people their. but other investors
05:41.24KattyNetgeeks: yes, i do
05:41.29MGSsancho:)
05:41.33KattyNetgeeks: thanks to Hmmhesays and his patience :)
05:41.49MGSsancholol
05:41.53KattyNetgeeks: i could probably set up my own box now from the documentation that i have and notes jotted down
05:41.57CherebrumI'm just the local nerd at work
05:42.05NetgeeksKatty:  *nod*
05:42.09KattyCherebrum: www.copi-rite.com/logo/angie.png
05:42.28SarahEmmi wanna use * for work :P
05:42.29NetgeeksMy head hurts from thinking too much today  :P
05:42.30KattyCherebrum: see official title (=
05:42.34QwellSarahEmm: yeah, me too
05:42.36Cherebrumheh heh
05:42.40*** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net)
05:42.49*** join/#asterisk dooder (~nateputna@h-67-102-173-136.sttnwaho.covad.net)
05:42.51MGSsancholol
05:42.58dooderi got my pbx working kinda.
05:43.02NetgeeksGood billing systems are way too complicated
05:43.03dooderi can call out. but can't recieve calls
05:43.14glm2kdooder: tell me about it
05:43.23CherebrumKatty: I work for this place: http://www.zentality.com/
05:43.26MGSsanchothanks a lot. for my questions. when i get everything approved. i can start building the new servers and set everything up. i hate waiting
05:43.28harryvvdooder, with iax or sip?
05:43.39Kattymost of this conference is going way over my head. the sangoma info about echo cancelation was informative... but i don't even use agi
05:43.39dooderharryvv : sip/broadvoice
05:43.40harryvvdooder, what are the symptoms
05:44.07dooderharryvv : get a fast bussy when calling my place
05:44.26dooderregular busy that is
05:44.28NetgeeksAGI is the devil
05:44.31CherebrumKatty: Have you used astlinux?
05:44.35shidois there a live # we can listen in on?
05:44.40CherebrumKatty: It's a project Kristian put together
05:44.40KattyCherebrum: i have not.
05:44.49Cherebrumit's an embeded asterisk distro
05:45.01MGSsanchocool
05:45.01glm2kcool
05:45.03Cherebrumit runs on a single board computer
05:45.11Cherebrumand uses flash for storage
05:45.12Cherebrumand there are no moving parts
05:45.23harryvvdooder fasty bussy as in the second you enter in the last digit it goes fast bussy or does it take a few seconds untill you hear the fast bussy?
05:45.44dooderharryvv : its a regular busy. after a couple seconds
05:45.57CherebrumKatty: http://www.soekris.com/bundles.htm
05:46.02harryvvokay, that is in your dial plan or sip.conf
05:46.10CherebrumKatty: it runs on that
05:46.26harryvvdooder, what about your fw..setup for sip?
05:46.49KattyCherebrum: we don't use t1 or e1 cards
05:47.06dooderharryvv : i've allowed all the ports.
05:47.12CherebrumI was using the sangoma cards but they were giving me fits
05:47.16Cherebrumso I took them all out
05:47.21Cherebrumand I'm using Cisco 1750 routers now
05:47.32harryvvdooder, what ports and did you forward them.
05:47.49CherebrumI've got a drawer full of the sangoma cards at work
05:47.53syle2why does SER have to be such a cryptic peice of shit to learn, why can;t it have standard config files lol
05:48.05Cherebrumand they keep trying to give them away to me here at cluecon
05:48.07Kattyi was given digium cards to use.
05:48.08NetgeeksHrm, interesting.  I've got about 80 Sangoma cards in production systems now
05:48.09dooderharryvv : I opened up 69,5060-5063 and 10000-20000 UDP
05:48.13CherebrumI just want an IAXy
05:48.23harryvvsyle, so your typical mom and pop bussiness cannot configure it and you charge them to configure it :)
05:48.24Kattyhotel is freezing :<
05:48.34CherebrumKatty: Switch me rooms!
05:48.38harryvvdooder 69 what?
05:48.54KattyCherebrum: i'm currently in brian's room, sharing bandwidth with file
05:49.14harryvvbrians room?
05:49.25Kattywe drove down together
05:49.30harryvvi see
05:49.33Netgeekswell, while you are in the Chicago area, there are two must eat at restraunts...
05:49.34Kattyalong with mister darthclue
05:49.45harryvvi see
05:49.50*** join/#asterisk _blue (~blue@adsl-68-94-15-143.dsl.rcsntx.swbell.net)
05:49.52harryvvalot of you people in that area :)
05:49.54Kattydo you, do you really?
05:49.58Kattyand with both eyes?
05:50.02_bluecan anyone help me with this little project....
05:50.03NetgeeksThe Schezwan Restraunt (I can't spell) on Michigan avenue
05:50.12CherebrumKatty: it's a bit warm in here
05:50.12dooderharryvv : according to http://www.broadvoice.com/support_faqs_network.html thats what it should be
05:50.21CherebrumWhere is that?
05:50.21CherebrumI'm in 204
05:50.21Cherebrumby the pool
05:50.27Netgeeksand "Walker Brothers" restraunt... there are several of them, but I think one pretty close to the hotel
05:50.30Kattyharryvv: indeed. you'd think with that many people around here there'd be someone to talk to :)
05:50.32harryvvdooder, can you call betweeen phones
05:50.40_bluei'm trying to think of a way to have PBX servers implemented with AP's extended throughout a building being able to replace older radios
05:50.42harryvvyea
05:50.50harryvvcluecon should come here
05:50.50harryvv:)
05:51.01harryvvone of the best places to visit.
05:51.02KattyCherebrum: brian's in 255 (=
05:51.09CherebrumI was down there I think
05:51.12Cherebrumwe moved rooms
05:51.14Netgeeksharryvv: where is that?
05:51.14KattyCherebrum: well...he's at the hospital now
05:51.15Cherebrumcause the beds sucked
05:51.18harryvvbeen meeting people who are tourist from island, sanfrancisco other countries.
05:51.19TwisterOMG!! SarahEmm *lix*
05:51.19Cherebrumhospital?
05:51.19dooderharryvv : yep. and I can call out. and I'm peered with broadvoice
05:51.21Cherebrumwhy?
05:51.24harryvvVancouver BC canada.
05:51.34harryvviceland
05:51.36Netgeeksyep, Vancouver is nice
05:51.41harryvvis what I meant to say.
05:51.49KattyCherebrum: peter got ill....infection in the intestine track i think i heard anthm say
05:51.53harryvvTourist meca on Robinson street.
05:51.54NetgeeksI flew over iceland once...
05:52.31harryvvthese tourist were from some small island between island and scottland
05:52.36CherebrumKatty: he must have eaten the lasagna
05:52.40harryvvnever even knew there were such islands.
05:52.40NetgeeksIsle of Mann
05:52.48KattyCherebrum: dunno... i can't eat here
05:52.56CherebrumKatty: why not?
05:53.02glm2kshe's a vegan
05:53.04KattyCherebrum: they don't understand what vegan means
05:53.07Cherebrumaha
05:53.17KattyCherebrum: and even though they were /supposed/ to make arrangements, it never happened
05:53.33Cherebrumthat sucks
05:53.50JerJerMeat tastes like murder and murder tastes pretty goddamn good
05:53.54Kattywhat sucks even more is that i went to spaghetti warehouse
05:54.12harryvvTwo light planes collide in midair; one crashes into empty school
05:54.16Kattygot my drink...then looked at the menu....asked if the spaghetti with tomato sauce was vegan
05:54.17CherebrumI went there today
05:54.22harryvvin renton washington
05:54.26Kattythe waiter had to get someone else..cause they didn't know.
05:54.31harryvvthats nuts
05:54.35QwellHow can you now know what vegan is?
05:54.38Kattyand then the other person said no it wasn't vegan...so i asked if anything was vegan...they said no
05:54.47Kattythen, they came by again and wanted me to spell vegan for them
05:54.47Cherebrumhmmm
05:54.47Qwellno meat, no animal junk, etc...it can't be that hard, can it?
05:54.58Kattyi ended up writing vegan down for them. heh.
05:55.05Netgeeksoh my
05:55.06Cherebrumcan you eat Subway?
05:55.08Kattyno one has a clue around here
05:55.11Cherebrumthere is one across the parking lot
05:55.21KattyCherebrum: that's where i've been eating for lunch ever since tuesday
05:55.24Cherebrumaha
05:55.33KattyCherebrum: and there is no protein there either
05:55.42KattyCherebrum: so i've gone half a week without protein basically
05:55.50Kattyheh, probably why i'm freezing my tail off
05:55.52NetgeeksChicago won't be very vegan friendly
05:55.55CherebrumI found a nice pizza place down the street.. I think I have the menu here somewhere..
05:56.01Cherebrumthey ship pizzas all over the world
05:56.03KattyNetgeeks: it will be, at the right places downtown
05:56.04Cherebrumin dry ice
05:56.09QwellCherebrum: Chicago pasta house?
05:56.14glm2kmmmm, i miss soy ...
05:56.32Cherebrumhmm... cleaning lady threw it out
05:56.37Qwellvegan enchiladas?
05:56.40Qwellinteresting
05:56.45harryvvkatty, any other woman you know of interested in asterisk?
05:56.51Qwellharryvv: SarahEmm
05:57.01NetgeeksBeen a few years since i lived in chicago tho, so I probably don't know anymore
05:57.08Kattyharryvv: SarahEmm, obviously :)
05:57.12harryvvyea
05:57.14CherebrumKatty: Lou Malnati's Pizzeria
05:57.15glm2ktry vegan bbq
05:57.17Netgeeksbut sarah bites
05:57.23*** part/#asterisk _blue (~blue@adsl-68-94-15-143.dsl.rcsntx.swbell.net)
05:57.26KattyCherebrum: that's not vegan
05:57.42KattyQwell: http://www.webcon.net/~izaah/vegan/2005/06/corn-taco-enchiladas.html
05:57.43CherebrumKatty: http://tinyurl.com/77t2l
05:57.46SarahEmmi'm interested in asterisk!
05:57.50SarahEmmNetgeeks: only if i like you.
05:57.56Cherebrumthey might have like a vege pizza
05:57.59Netgeeks*sigh*
05:58.04KattyCherebrum: egg in the crust
05:58.06KattyCherebrum: cheese on top
05:58.10Cherebrumaha
05:58.23CherebrumI see
05:58.25harryvvWhat iax providers allow the asterisk customer to set his own CID?
05:58.36SarahEmmhardwire: i'm with Voctel, they do
05:58.39JerJerforget that - just give me a cow, i'll carve off what I want and ride the rest home
05:58.42SarahEmmNetgeeks: meow? whyfor sigh?
05:58.50JerJeryeeee haw
05:58.50CherebrumI really like asterlink... they send ANI and ANI2
05:58.55Kattyi've no transportation since brian drove up, but i do have multipass for train/subway/bus (=
05:59.11syle2http://bugs.digium.com/bug_view_page.php?bug_id=4904
05:59.34NetgeeksSarah: consider it a good *sigh*
05:59.36QwellKatty: take the subway to subway?
05:59.47dooderharryvv : now it rings on the calling phone but not inside
05:59.49SarahEmmNetgeeks: ahh, okay. meow.
05:59.51KattyQwell: :<
05:59.58Qwellno good? ;/
06:00.06KattyQwell: i want protein :<
06:00.13harryvvdooder not inside what
06:00.14CherebrumKatty: If you need a ride somewhere then hit me up
06:00.16QwellI wouldn't know what protein is
06:00.24KattyCherebrum: i'm far too polite to ask (=
06:00.27Qwellmine is part of my diet...
06:00.28KattyCherebrum: but thank you anyway
06:00.32NetgeeksI'm off, it's an early morning with the Database programmer tomorrow
06:00.56Kattys/wekk/week
06:00.57MGSsanchoga'night and thanks for the help
06:01.06harryvvbr4b
06:01.38*** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be)
06:01.57CherebrumKatty: There is an Osco down the street I went to the other day.. it's a grocery store
06:02.04Cherebrummaybe you can get something there
06:02.08dooderi call my broadvoice line from my cell. its ringing on my cell but not on the broadvoice line
06:02.27KattyCherebrum: perhaps...not sure.
06:02.27CherebrumThat's because broadvoice doesn't know what they are doing
06:02.50KattyCherebrum: it's a bit difficult to find lunch in a grocery store. it generally requires cooking, etc.
06:02.51CherebrumI got some frapachino and coke
06:03.04syle2lay off the drugs
06:03.27*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
06:03.33Katty:>
06:03.38file[laptop]hey Katty
06:03.45file[laptop]I'll bbs
06:03.47file[laptop]showerrrrrr
06:03.49Kattyk
06:04.02QwellKatty: he smells, doesn't he?
06:04.11CherebrumKatty: http://tinyurl.com/7ob3l
06:04.17KattyQwell: he's not close enough to find out (=
06:04.23QwellThats a no then.
06:04.47file[laptop]my feet smell :P
06:04.50file[laptop]thus why I'm having a shower
06:04.51Kattyi see.
06:04.54file[laptop]because I'll be cleannnnn
06:07.10Kattyfile[laptop]: thanks for sharing.
06:07.10Kattyfile[laptop]: i'll put that in drawer 13 for safe keeping
06:07.41Cherebrumfile: hey.. I had some weird stuff happen with asterlink
06:07.43KattyCherebrum: fascinating
06:08.01Cherebrumfile: I was getting inbound calls for DIDs that weren't mine
06:09.27Cherebrumhmm
06:09.56*** join/#asterisk Inv_arp (junya@adsl-156-139-181.mia.bellsouth.net)
06:11.21*** join/#asterisk _ioscanner (~ioscanner@c-67-162-251-133.hsd1.tx.comcast.net)
06:11.38Kattyi'm trying to figure out why my email notification (freight) is borken
06:11.52Kattyand it's all in asp :<
06:12.02Cherebrumew
06:12.08*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
06:12.21_ioscannerany one compile zaptel drivers with config_zapata_net and config_zapata_ppp?
06:12.23Kattyunderstatement of the week.
06:12.29_ioscannerI can't get it to compile.
06:12.43_ioscannerI have to keep using an older build I did that it works
06:13.09_ioscannerMeowwwwwwwwwww
06:13.29Katty:>>>
06:13.34CherebrumWell I'm gonna goto bed
06:13.39QwellSo, how much longer for ClueCon?
06:13.39_ioscannernight
06:13.41Kattynite
06:13.48Cherebrumgoodnight
06:13.58_ioscannerSo anyone using t-1 card with the current zaptel drivers?
06:14.06QwellSomebody is making a bootleg video of it all, right?
06:14.09*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
06:14.15dijungali out for the night
06:15.02wunderkinanyone here handy with sox and lame? trying to convert a .raw file from asterisk to mp3.. i just get static
06:15.49KattyQwell: brian is making a dvd, yes
06:16.00Qwellahh
06:17.31*** join/#asterisk af_ (~af@ip-142-5.sn1.eutelia.it)
06:19.55*** join/#asterisk konfuzed (~kvirc@72.0.72.129)
06:22.23wunderkinfor some reason when i soxmix 2 raw files they are a little staticy too but you can not hear anything at all when i convert it to mp3.. now i am just trying to convert the input file to mp3 to make sure thats ok and its not ;/
06:22.41*** join/#asterisk jeffgus (~jeffgus@2002:d856:c704:0:0:0:0:1)
06:22.59konfuzedas a matter of interest, I'd like to here about what asterisk should not be used for in context of telephony vs sports
06:23.38konfuzedsome people think one thing might be able to do everything but that is rarely the case.
06:24.11*** join/#asterisk jeffgus (~jeffgus@2002:d856:c704:0:0:0:0:1)
06:24.15konfuzedis there some articles that review this perhaps. as in what not to do with asterisk?
06:30.22*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
06:32.20Kattybye bye
06:34.47MGSsancholol
06:36.32*** join/#asterisk r3d5un (~r3d5un@80.121.192.21)
06:36.36r3d5unhi all
06:38.39*** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au)
06:39.18konfuzedhhhhmmmmmmmmm
06:39.26konfuzedwhat up
06:41.46*** join/#asterisk nextime (~nextime@213-140-6-96.ip.fastwebnet.it)
06:42.19*** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net)
06:44.04*** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net)
06:44.21|Vulture|Anyone familliar with E911 over a PRI?
06:46.21*** join/#asterisk Katty (~angela@68.112.15.110)
06:46.49Kattyrehi
06:46.56|Vulture|sup Katty
06:47.04Kattyceiling
06:47.07QwellWhat happened to bed?
06:47.08|Vulture|...
06:47.20file[laptop]I wonder where brc went
06:47.23Kattyi never said i was going to bed.
06:47.29Qwelltrue
06:47.34|Vulture|is E911 on a PRI passed or is it based on the installation address?
06:47.58Netgeeksyeah, she just said bye
06:48.16Kattytv so blah :<
06:48.32*** join/#asterisk KaBewM (~kabewm@24-180-28-208.dhcp.psdn.ca.charter.com)
06:48.36Netgeeksi however am off to bed
06:48.42SarahEmmnini
06:48.42Kattynitenite
06:48.47SarahEmmi wanna go to bed *Whine*
06:48.48SarahEmm:P
06:48.50Netgeeksdon't have too much fun without me  ;0
06:48.52*** join/#asterisk postel (~jp@postel.user)
06:48.56Kattyk
06:50.15*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
07:19.12*** join/#asterisk kb1_kanobe (~jsmith@h24-207-96-50.cst.dccnet.com)
07:19.28kb1_kanobeevening all.
07:19.53r3d5unhi kb1_kanobe
07:20.16kb1_kanobeAny ideas why tonights cvs-head is segfaulting on load? during/after chan_iax2.so loads there is a 'doing lookup for...' and then <poof!>
07:20.29drumkillaooh, nice
07:20.51kb1_kanobejust trying to move up from a mid-may cvs...
07:21.14drumkillai hope it wasn't my fault
07:21.17drumkillalet me see here ...
07:21.23kb1_kanobeWas working fine before. did a fresh checkout too (into a new directory)
07:21.52kb1_kanobeAnything I can trace for you?
07:22.04drumkillawell, a backtrace would help
07:22.14kb1_kanobegot a few minutes to walk me through the process?
07:22.19drumkillasure
07:22.23drumkillado you have gdb?
07:22.38kb1_kanobeinstalling now.
07:22.42drumkillak
07:22.46kb1_kanobedone.
07:22.55drumkillamust not be using gentoo!  :)
07:23.02kb1_kanobe;-)
07:23.14*** join/#asterisk bjohnson (~bjohnson@i216-58-14-198.igs.net)
07:23.16kb1_kanobeplain debian.
07:23.55drumkillawell first, do a "make clean; make dont-optimize; make install" and make it crash running that
07:24.03drumkillathat will make sure that you get a clean backtrace
07:24.23kb1_kanobepresumably only need asterisk, not zaptel/libpri?
07:24.30drumkillacorrect
07:24.38kb1_kanobewhrrring.
07:27.20drumkilladone?
07:27.26*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
07:27.31kb1_kanobealmost.... it has a small brain.
07:27.49*** join/#asterisk shakuhashi (~shakuhash@200.219.182.202)
07:30.05kb1_kanobeodd... 'make install' is generating: make: *** No rule to make target `/usr/include/asterisk/build.h', needed by `cli.o'.  Stop.
07:30.14kb1_kanobes/is/has started
07:30.28drumkillahm ... did "make dont-optimize" do the install part?
07:30.46kb1_kanobeno, it's faceplanting on the same .o
07:31.11drumkillawell, at the end of the first time you ran it, did it do the install
07:31.48kb1_kanobeyeah, looks like the binary is newer at least.
07:32.04drumkillaok ...
07:32.18drumkillaso run asterisk and when it crashes, it should create a core file
07:32.30drumkillacore.<somenumber>
07:32.39kb1_kanobegot it.
07:33.01*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
07:33.10drumkillacd /usr/src/asterisk
07:33.21drumkillagdb ./asterisk /path/to/that/core/file
07:33.34kb1_kanobeloaded.
07:33.44drumkillapastebin the output of 'bt' and 'bt full'
07:34.42kb1_kanobehttp://pastebin.ca/19365
07:35.49drumkillahold please :)
07:36.23JerJerx080a124d in ast_get_ip (sin=0x0, value=0x0) at acl.c:230
07:36.25JerJerthat looks bad
07:36.39drumkillaindeed
07:38.49*** join/#asterisk lth (~lth@219.95.128.52)
07:39.35lthHi everybody
07:39.41r3d5unhi lth
07:39.44kb1_kanobeevening.
07:39.49*** join/#asterisk Xumpi (~SysOp@a81-84-68-51.cpe.netcabo.pt)
07:40.03lthWell - afternoon here really :)
07:41.07lthAnybody know if any asterisk developers are around?
07:41.12lthI know it's probably bad time
07:41.17SarahEmmuhh
07:41.21Xumpigood morning
07:41.22SarahEmmthere's always * developers around
07:41.24SarahEmmwhat subsystem?
07:41.28lthIAX2
07:41.34SarahEmmnot I, then :)
07:41.37SarahEmm(it's never me. :P)
07:42.00kb1_kanobeYou never know, but someone might... just toss out the question :-)
07:42.06lthHa, ha - what's your subsystem - I am sure I can come up with a question :)
07:42.18SarahEmmlth: pretty much just TTY/TDD and text processing
07:42.51lthWell - basically the good old one way audio on IAX transfer
07:42.56lthIt's been up now and then the past year
07:43.03*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
07:43.04lthBut now it's really driving me nuts
07:43.14lthI just described in a new bug report
07:43.15kb1_kanobewhat codecs?
07:43.29lthhttp://bugs.digium.com/view.php?id=4905
07:43.37lthCodecs tried: gsm/ilbc
07:43.43JerJerone-way audio only can be caused by a restrictive firewall or codec mismatch
07:43.45JerJeron iax
07:43.52lthNo, no - it's a transfer issue
07:44.17lthReceived trunked frame before first full voice frame
07:44.18JerJerwe do iax native brigding with transfer all day long
07:44.22JerJerdon't trunk
07:44.23lthRepeated again and again
07:44.29lthWell :)
07:44.41JerJeriax2 trunking is hella broken still
07:44.44JerJerimho
07:44.47lthJerJer: I tried with or without trunking - ONLY the error message is different
07:44.56lthWith trunking: Received trunked frame before first full voice frame
07:44.56JerJerthen you have version skew
07:45.07lthWithout trunking: Received mini frame before first full voice frame
07:45.13JerJerthat message simply states it doesn't know what codec to use, yet
07:45.21lthAll 3 systems up to date as of CVS HEAD 2 hours ago
07:45.28JerJersince a full voice packet has never arrived
07:45.35drumkillakb1_kanobe: i should have a fix for you in a minute here
07:45.37JerJersomething is scewed
07:45.48JerJerskewed
07:45.49*** join/#asterisk pa (~Paolo@pa.user)
07:46.05JerJeror screwed
07:46.21Inv_arpanyone have a voicepulse ip i can test ping
07:46.24lthAll 3 systems have: disallow=all allow=gsm allow=ilbc
07:46.39JerJerInv_arp:  how about nslookup voicepulse.com  ?
07:46.40lthActually let me try to disable all but GSM
07:47.03Inv_arphmm would prefer an acutally voip server
07:47.12Inv_arpactual rather
07:48.02JerJerdump their dns then
07:48.06JerJersee what hosts they have defined
07:48.12JerJerthen pick one
07:48.29ComputerWarmanyone know of any h323 soft phones?
07:48.37Inv_arpnp lemme  dig then
07:49.10*** join/#asterisk cgcorea (~cgcorea@205.240.200.105)
07:50.24drumkillakb1_kanobe: you still around?
07:50.47kb1_kanobeyep. any luck?
07:50.50drumkillayeah
07:50.56drumkillaedit channels/chan_iax2.c for me ...
07:51.10kb1_kanobeok.
07:51.28drumkillago to line 7879
07:51.31lthJerJer: tried with only mention of codec in global section of iax.conf: disallow=all allow=gsm - and nothing else on all systems
07:51.37lthStill fail occasionally
07:51.52drumkillayou'll see ... if (!ast_get_ip(&sin, tmp)) {
07:51.58kb1_kanobeyes.
07:52.01drumkillakb1_kanobe: change 'tmp
07:52.03drumkillaer
07:52.09drumkillachange 'tmp' to 'addr'
07:52.15kb1_kanobedone.
07:52.22drumkillarecompile and test again
07:52.28kb1_kanobergr. brb.
07:53.08JerJerlth:  detail your findings in a bugreport i guess
07:53.23lthJerJer: http://bugs.digium.com/view.php?id=4905
07:53.35JerJerthings seem to be normal here since we do that quite a whole lot
07:53.41kb1_kanobelth: perhaps also add the output of 'iax debug' from both sides.
07:53.50JerJerthen again i have found people like to complain in public but not to us directly, so i don't really know
07:54.19JerJerlth and perhaps a tcpdump -x udp port 4569 from both as well
07:54.19lthI only see the problem with iax transfers
07:54.30JerJerthat's how we do load balancing
07:54.45JerJerswitch-1 is just a softswitch
07:55.17JerJerthere are gateways that the soft switch knows about
07:55.18drumkillawhat are you using to determine how to distribute your load?
07:55.26JerJerrandom number
07:55.30drumkillagotcha
07:55.31JerJernothing fancy
07:55.52drumkillacould even just 'round robin' ...
07:55.55lthJerJer: I got 3 systems - one with static ip  and two with dynamic ip - problem is when calling between the systems on dynamic - who by nature can't really know about each other without going through the only system on public ip
07:56.16lthJerJer: Don't know if that could have anything to do with it
07:56.30JerJerdrumkilla:  yeah we did do that for a while, but then we got this bright idea to do the random number thing
07:56.40drumkillaha, which is totally cooler
07:57.01JerJeryeah it seems the randomness gives a slightly better spread
07:57.12drumkillacool deal
07:58.13kb1_kanobedrumkilla: bingo - that resolved it.
07:58.21kb1_kanobethanks :-)
07:58.24drumkillakb1_kanobe: yay!
07:58.26drumkillai'll commit the fix now
07:58.49JerJercvs commit -m "fix trivial little seg fault issue" acl.c
07:58.50kb1_kanobety again
07:59.09drumkillaJerJer: channels/chan_iax2.c actually :-p
07:59.11JerJeri hate those kinds of phuckups
07:59.16JerJergotcha  :)
08:00.02drumkillait was just a mixup related to variables after using them with a strsep ...
08:00.16JerJerso not quite so trivial
08:00.24drumkillanot really :)
08:01.11drumkillait wouldn't have crashed if there was a port number with that IP  :)
08:01.17drumkillastill wouldn't have worked correctly, though
08:01.31kb1_kanobeglad to be of service
08:01.42drumkilladrumkilla++
08:02.00drumkillamaybe fixing that bug will help me sleep
08:02.39*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
08:03.32JerJerand there are people out there that say open-source doesn't work
08:03.39drumkilla:D
08:03.42*** join/#asterisk nickv111 (~nick@70-33-44-221.clspco.adelphia.net)
08:04.01nickv111Hello
08:04.15drumkillaHelllllllo!
08:04.21nickv111drumkilla...
08:04.24nickv111Hehe
08:04.28nickv111How's it going, man?
08:04.35drumkillapretty good
08:04.47drumkillacan't sleep, actually
08:04.50nickv111Haven't seen you for a while
08:04.52nickv111Me either
08:05.20drumkillai do a lot of idling on IRC ...
08:05.39nickv111Well, I was wondering how to use asterisk to set up a SIP connection between three people ;)
08:05.40drumkillamostly end up talking in other channels a lot of the time, too
08:05.46nickv111I see
08:05.49*** join/#asterisk Dovid (~dovi5988@pool-151-198-8-101.mad.east.verizon.net)
08:06.08drumkillanickv111: 'show application meetme'
08:06.20nickv111drumkilla: Thanks
08:06.25*** join/#asterisk frenzy (~frenzy@193.220.82.108)
08:06.28drumkillathat's the conferencing application
08:06.29JerJerif zaptel is available
08:06.31nickv111drumkilla: I've been interested in VoIP
08:06.35JerJertiming at least
08:06.54frenzyhello all!
08:07.00JerJermooo
08:07.01drumkillagreetings frenzy !
08:07.04nighty-glop
08:07.54nickv111Heh. Is there like a "getting started" page for asterisk?
08:08.11drumkilla~docs
08:08.11jbotextra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
08:08.14frenzyvoip-info.org
08:08.21*** join/#asterisk Fabe (~spamhere@pd95b0bf7.dip0.t-ipconnect.de)
08:08.21nickv111Hehe
08:08.22nickv111Thanks
08:08.29JerJervi asterisk.c
08:08.29drumkillatry the wiki
08:08.40drumkillathat has a lot of info ... not very organized, though
08:08.44drumkilla... and you can't really trust it :)
08:09.05drumkillaJerJer: did you strart with asterisk.c?
08:09.22JerJerbasically
08:09.25drumkillaI think I started with app_voicemail ... it was just the easiest code into from the user perspective
08:09.39JerJerbut i've been around since early 2000 so ya know
08:09.42drumkilla*to get into ...
08:09.43drumkillanice
08:09.59*** join/#asterisk jeffgus (~jeffgus@2002:d856:c704:0:0:0:0:1)
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08:10.04drumkillaI still feel pretty new
08:10.07JerJerway before digium even existed...just Linux Support Services
08:10.26Inv_arphmm this provider allows g729... do i need a license to use it with HT486?
08:10.30drumkilladid you ever have to recompile because you changed your dialplan?
08:10.54JerJerhmm - don't believe so
08:11.07drumkillaheh, wasn't sure how long that existed
08:11.11JerJerbut I never really used Asterisk that hardcore until asterisk-oh323 came out
08:11.27Inv_arpso asterisk has to be put in passthru mode i assume
08:11.32lthJerJer: ok - it seems if trunking AND jitterbuffer are both disabled it's somewhat more stable
08:11.38tzafrir_laptopyou? oh323?
08:11.40nickv111One time a friend and I had an idea to just pipe /dev/dsp over TCP...
08:11.42*** join/#asterisk dooder (~nateputna@h-67-102-173-136.sttnwaho.covad.net) [NETSPLIT VICTIM]
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08:11.46frenzyWhat does REINVITE do>
08:11.49nickv111Ugh. I really hate netsplits
08:12.02JerJermakes me feel like we are on efnet
08:12.10nickv111Heh
08:12.22JerJerlth: both of which we have always had disabled
08:12.29drumkillamy client supresses the output from netsplits :)
08:12.39nickv111drumkilla: Let me guess. irssi?
08:12.43frenzyWhat does REINVITE do?
08:12.43drumkillaindeed
08:13.23JerJertzafrir_laptop: yes the instant i saw the mailing list post anouncing the release of asterisk-oh323...I really sat down and learned asterisk
08:13.30r3d5unif i know it correct, reinvite means that the Server handles the connection startup and then the two phones connect directly
08:14.07nickv111Well, I really have to go to bed
08:14.08nickv111G'night all
08:14.08JerJertzafrir_laptop: then i was disqusted by asterisk-oh323's implemenation and was told if I could write a better H.323 driver to do it - 3 days later chan_h323 was released
08:14.10frenzythats exactly what I'm trying to setup
08:14.10frenzyhow do I implement tht?
08:14.14nickv111It's 2:00 here
08:14.19JerJer4:14 here
08:14.19nickv111Good to see you again, drumkilla
08:14.29r3d5unif you dont use reinvite the whole traffic of the call will pass through the server
08:14.32JerJersleep is for the weak
08:14.35*** part/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
08:14.40drumkillanickv111: you too, good night
08:14.43nickv111;)
08:14.54frenzyHow do I use reinvite?
08:15.05drumkillafrenzy: it uses itself
08:15.06JerJerfrenzy:  it just happens, unless you disable it
08:15.28r3d5units a channel configuration parameter  "canreinvite=no" for disabling it
08:15.29frenzyI tried it.. doesnt work
08:15.38frenzymy phone is behind NAT
08:15.47r3d5unperhaps your network setup doesnt allow reinvite
08:16.05frenzyhuh
08:16.17r3d5unbecause there doesnt exist a path between the two phones?
08:16.38frenzyI tried it with PortaOne, worked..
08:17.43JerJerreigster thru the NAT to the proxy
08:17.54JerJerwell asterisk - since asterisk is not a 'sip proxy'
08:18.11r3d5uncan't asterisk work as a proxy ??
08:18.16JerJerno
08:18.27JerJerat best asterisk is a hybrid b2bua
08:18.28frenzyhmm
08:18.30drumkilla~notproxy
08:18.38frenzyso I have install SER?
08:18.38*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
08:18.42drumkilladang
08:18.54SarahEmmdrumkilla: you should createx0r it.
08:19.03drumkillaSarahEmm: oh noez1!
08:19.10DovidMorning ALl
08:19.16Dovidall*
08:19.29drumkillayes, it is morning, isn't it
08:19.34drumkillai should totally be asleep right now
08:19.45JerJertheoretically speaking its morning
08:20.32loickgood morning... Here in Paris it' 10h00 AM :-)
08:20.34SarahEmmit's morning, but i'm at work until 0800...
08:21.23*** join/#asterisk FlungDung (~wayne@rndf-146-17-180.telkomadsl.co.za)
08:21.28JerJerdamnit i just missed 4:20
08:21.43drumkillasux0rz
08:21.57drumkillaJerJer: you can set your clock back
08:22.02JerJernaw - took me a minute to find the hooka
08:22.34drumkillai'll smoke with you!
08:22.38drumkillacheers
08:23.06*** join/#asterisk hugo-v6 (~hugo@ipv6.keepnick.info)
08:23.16JerJerahh i wish 4:20 came around more often
08:23.36*** join/#asterisk grimse (~grimse@p5481F39E.dip.t-dialin.net)
08:23.42hugo-v6hmz. always this regging at nickserv *sigh* have to script this :/
08:23.47hugo-v6remoins
08:24.01drumkillamost clients let you automate pretty easily ...
08:24.08drumkillaautomate it*
08:24.14hugo-v6drumkilla: of course, irssi and perl :)
08:24.46drumkillaew, perl!
08:24.53hugo-v6pfff ;)
08:25.13SarahEmmmirc and vb is the way to go.
08:25.20hugo-v6vb? ROTFL
08:25.21drumkilla/kick SarahEmm
08:25.22hugo-v6go away
08:25.24SarahEmmlol
08:25.35drumkillahm... didn't go through!
08:25.40SarahEmmnooo kicking the kitrich..
08:25.47*** join/#asterisk MuppetMaster (~MuppetMas@27.Red-213-97-53.pooles.rima-tde.net)
08:25.54MuppetMasterHello.
08:25.57drumkillahave no fear, MuppetMaster is here!
08:26.00MuppetMasterWhat does this message mean?  chan_sip.c:5617 check_auth: stale nonce received
08:26.09*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
08:26.17MuppetMasterI am getting this regularly from an Avaya 4602SW
08:27.42hugo-v6well... gotta go now... some work to do :) l8r
08:28.26Inv_arpso if a provider only allows g729.. I have to purchase a license from digium to use this provider with * correct?
08:28.30*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
08:28.43JerJerInv_arp:  depends
08:28.46MuppetMasterInv_arp:  Only if you want to manipulate the stream.
08:28.58MuppetMasterIf you want to use pass through then you do not need a license for G729
08:29.20drumkillaMuppetMaster: is this causing you problems?
08:30.11drumkillawhat it means is, when we're checking the auth from the avaya, they didn't send the same nonce in their auth response that we put in the auth request
08:30.11MuppetMasterdrumkilla:  Not sure, as my 4602SW has also stopped working, have not had time to go through a SIP debug to see what is happening.
08:30.15*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
08:30.17*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
08:30.23Inv_arpMuppetMaster/JerJer:  just want to setup menu's etc.. its all incoming
08:30.30MuppetMasterCurious as to what the message means, as only started seeing it fairly recently.
08:30.41MuppetMasterI am on the latest CVS HEAD (as of 2 days ago I believe).
08:30.43drumkillathey try to call -> we request auth -> they send auth response
08:30.58drumkillawhen we send that requset, we give a random number that they should use
08:31.05drumkillaand then they didn't use it
08:31.10drumkillaso that's what the message is about
08:31.11MuppetMasterInv_arp:  If you want to be able to have an IVR using G729 then you will need to have G729 licenses.
08:31.14JerJerInv_arp:  then your asterisk box is going to have to know how to deal with G.729
08:31.19JerJerwhich means licenses
08:31.21Inv_arpgot it
08:31.22drumkillanot sure about the language in the RFC about it
08:31.54MuppetMasterdrumkilla:  I see, so the Avaya 4602SW is not playing nice.
08:31.56MuppetMasterFigures...
08:32.15drumkillayep
08:32.27drumkillait's possible they're breaking the RFC, but ...
08:32.35drumkillait's too late to go looking :)
08:32.45*** join/#asterisk formen|LSK (~formen@christiang.whitebird.no)
08:33.23MuppetMasterAlthough I do notice I am receiving it on a Sipura 3K as well, and never had that before.
08:33.41MuppetMasterHere is what I am using:  -- Saved useragent "Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26" for peer 3006
08:35.51JerJerlooks pretty nasty already
08:36.01JerJerthat's a lot of version numbering
08:36.26MuppetMasterJerJer:  Yes it is.  And the process to convert it from H.323 (factory set) to SIP (downloaded from the Avaya website) was a nightmare.
08:36.33JerJerlol
08:36.42MuppetMasterI had to have a TFTP and HTTP server (H.323 firmware uses TFTP and the SIP firmware uses SIP).,
08:36.45MuppetMasterWhat a joke.
08:36.45JerJersucks to be you
08:36.58MuppetMasterAnd the web UI that it uses could have been written better by a 10 year old.
08:37.01ComputerWarmJerJer are you the one that created the oh323?
08:37.04*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
08:37.09JerJerchan_h323
08:37.14MuppetMasterAmazing to see such crap from Avaya, as they have such a good switch.  Expenseive of course.
08:37.26ComputerWarmJerJer where can i get chan_h323 from?
08:37.34JerJerits native to asterisk
08:37.47ComputerWarmso its already there?
08:37.47JerJerunlike others I gladly disclaim my code
08:38.08JerJeryes asterisk/channels/h323/README
08:38.18ComputerWarmoh ok thanks
08:38.38*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
08:39.11JerJerexpect your computer to blow up if you use it
08:39.15*** join/#asterisk pbxbart (~pbxbart@ip-80-226-229-206.vodafone-net.de)
08:39.22*** part/#asterisk pbxbart (~pbxbart@ip-80-226-229-206.vodafone-net.de)
08:39.35JerJerthen if you are motivated enough to make it work, then you can be happy
08:39.48JerJerotherwise use chan_woomera
08:39.56ComputerWarmlol all i want is something i can test  a h323 server with
08:40.23JerJerh.323 is not that simpile
08:40.25JerJersimple
08:40.42JerJerH.323 has rampant interoperability problems
08:40.44*** join/#asterisk clive- (~pirch@rndf-146-2-14.telkomadsl.co.za)
08:41.51ComputerWarmi am using gnugk on one of my servers. and I need to make some tests i can`t find any good soft phones that will handle connecting to the server. so i was thinking my only other option is to get asterisk working to support the convertion from sip to h233
08:41.52ComputerWarmh323
08:42.38JerJeri am sorry
08:43.53loickComputerWarm: which softphone are you using?
08:43.58frenzyhow do I create a mailbox for a user in sip.conf?
08:44.03ComputerWarmi tried SJPhone
08:44.16ComputerWarmand advance phone i think its called.
08:44.16loickand didn't work?
08:44.23JerJerfrenzy:  you don't
08:44.26ComputerWarmbut it won`t allow me to connect to the gnugk server
08:44.28JerJeryou create the mailbox in voicemail.conf
08:44.29glm2khow about gnomemeeting and netmeeting?
08:44.39JerJerand point to it in sip.conf with mailbox=666@devil
08:44.52ComputerWarmhave tried gnomemeeting i am on windows and netmeeting on here is ducked up
08:45.17loickComputerWarm: it's strange it didn't work !!
08:45.57frenzy1234 => 5678
08:46.05frenzyso the pw is 5678?
08:46.16ComputerWarmwell this gnugk setup its by prefix and ip address. it doesn`t need username and passwords
08:46.26glm2kaye
08:46.30glm2ki used it last year
08:46.44ComputerWarmso when i tried doing it with out a password. it screams security error or something like that
08:47.52ComputerWarmall this to get a phone to test with so i can test my other server
08:47.55ComputerWarmwhat a pain lol
08:48.15glm2ktell me about it. heh
08:48.42loickComputerWarm: what kind of profile did you create on SJphone?
08:48.56loickand which version of SJphone did you installed?
08:49.03JerJervoicemail.conf.sample is your friend
08:49.05ComputerWarmvery simple one ip address and the prefix.
08:49.10ComputerWarmloick the newest
08:49.15frenzyhow to start asterisk?
08:49.18loick1.6?
08:49.23frenzy<PROTECTED>
08:49.29ComputerWarmfrenzy safe_asterisk
08:49.31drumkillayes
08:49.47ComputerWarmloick ya
08:49.51frenzyhuh?
08:50.00ComputerWarmfrenzy type safe_asterisk
08:50.05ComputerWarmit starts it :-)
08:50.12frenzynop
08:50.12drumkillaJerJer: I know what we can do for fun!  rewrite the app_voicemail config parser!
08:50.17frenzydoesnt..
08:50.20ComputerWarmwhat os
08:50.27frenzyRHEL
08:50.39loickComputerWarm: did you try the "Calls through H.323 Gatekeeper" profile type?
08:50.44ComputerWarmsame here.. and it starts it everytime... please go read how to install asterisk
08:50.52ComputerWarmloick ya
08:51.11frenzyAsterisk ended with exit status 1
08:51.12frenzyAsterisk died with code 1.
08:51.12frenzyAutomatically restarting Asterisk.
08:51.12frenzyAsterisk ended with exit status 1
08:51.12frenzyAsterisk died with code 1.
08:51.12frenzyAutomatically restarting Asterisk.
08:51.14frenzyAsterisk ended with exit status 1
08:51.16frenzyAsterisk died with code 1.
08:51.18frenzyAutomatically restarting Asterisk.
08:51.20SarahEmmwe get it
08:51.20frenzyAsterisk ended with exit status 1
08:51.21ComputerWarmok stop
08:51.23frenzyAsterisk died with code 1.
08:51.24frenzyAutomatically restarting Asterisk.
08:51.25JerJerdon't run safe_asterisk until you know asterisk will run
08:51.26frenzyAsterisk ended with exit status 1
08:51.28frenzyAsterisk died with code 1.
08:51.29SarahEmm...
08:51.30frenzyAutomatically restarting Asterisk.
08:51.32frenzyAsterisk ended with exit status 1
08:51.34frenzyAsterisk died with code 1.
08:51.35glm2k:)
08:51.36drumkillafor the love of
08:51.37frenzyAutomatically restarting Asterisk.
08:51.37Martohtarwheee
08:51.38JerJeri feel a kick coming on
08:51.38frenzyAsterisk ended with exit status 1
08:51.40frenzyAsterisk died with code 1.
08:51.41loickComputerWarm: and even if you enter you login and password it doesn't work
08:51.42*** kick/#asterisk [frenzy!~russell@drumkilla.developer.and.stable.maintainer.asterisk] by drumkilla (stop it)
08:51.57ComputerWarmloick ya it keeps saying firewall blocked it
08:52.07loickOuuupps !!!
08:52.34loickit reached my basic knowlegde of H323 gatekeeper sorry
08:52.39*** join/#asterisk frenzy (~frenzy@193.220.82.108)
08:52.47frenzyhow do I stop that?
08:52.51ComputerWarmnp thanks for the try...
08:52.56frenzyAsterisk died with code 1.
08:52.57frenzyAutomatically restarting Asterisk.
08:53.03ComputerWarmfrenzy killall -9 safe_asterisk
08:53.04frenzykeep on getting it
08:53.09JerJerfrenzy:  don't run safe_asterisk until you are sure asterisk is going to run
08:53.11loickfrenzy you have been kicked
08:53.22loickbe less verbose :')
08:53.25ComputerWarmloick do you have a h323 server setup?
08:53.35loicknop not anymore
08:53.52ComputerWarmshoot to bad. anyone got h323 setup and wanna help me do some testing?
08:54.00JerJerno
08:54.13glm2kComputerWarm: a server? or endpoint?
08:54.42ComputerWarmserver i think it would be classified as i need someone to make some calls to cuba for me
08:54.54glm2ker, cuba?
08:55.08ComputerWarmdon`t ask
08:55.11glm2khehe
08:55.18*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
08:55.23frenzyhow come 8500 gives me engageD?
08:55.24*** part/#asterisk MuppetMaster (~MuppetMas@27.Red-213-97-53.pooles.rima-tde.net)
08:55.24*** join/#asterisk MuppetMaster (~MuppetMas@27.Red-213-97-53.pooles.rima-tde.net)
08:55.32frenzyrather: 11:54:58 Error returned: Not Acceptable Here
08:55.34*** part/#asterisk MuppetMaster (~MuppetMas@27.Red-213-97-53.pooles.rima-tde.net)
08:55.37glm2kas long as i don't have to buy anything from there i guess. hehe
08:55.45ComputerWarmlol no buying needed
08:56.00ComputerWarmglm2k mind if i message you
08:56.05glm2knope
08:56.37drumkillaso has everyone seen the new web site?
08:56.44glm2kwhat new website?
08:56.45SarahEmmyep!
08:56.47JerJeryep
08:56.49ComputerWarmfrenzy start asterisk like asterisk -gcvvvvv find out what your errors are
08:56.55syle2think i came up with a unique VOIP idea
08:56.59drumkillaSarahEmm: w00t!
08:57.00glm2ko_0
08:57.08SarahEmmsyle2: oh?
08:57.16syle2blah thinking outloud
08:57.37drumkillasyle2: did you figure out power over wireless ethernet?
08:57.42drumkillabecause that would be pretty sweet
08:57.49glm2kdrumkilla: lol
08:58.14syle2drumkilla lol
08:58.19ComputerWarmno figure it out for over cell wave first :-)
08:58.27syle2i don;t think i can get any voltage out of the air unless using solar power
08:58.44drumkillasyle2: just get some big tesla coils ....
08:59.21frenzy08:58:23 WARNING[7031]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 1123232049-338
08:59.25syle2well i read recently a guy from saskatchewan was selling uranium at 100 dollars a pound
08:59.29drumkillawarning ... this product increases your risk of cancer by 99%
08:59.37glm2klol
08:59.38*** join/#asterisk nitram (foo@superblob.com)
09:00.20syle2i think nuclear power is going to replace solar power though just wait 10 years
09:00.34glm2knot in the continental us
09:00.42ComputerWarmglm2k let me know when you are ready to help test
09:01.06glm2kComputerWarm: sec, i'm trying to get my old gnugk setup running :)
09:01.21syle2depends how much is available i guess
09:01.24glm2kall it did was crash my lappie. heh
09:01.26ComputerWarmoh ok
09:02.43syle2hell i can;t wait till these engineers figure out a way to make this source of power stable, i;m going to call the hydro company and tell them i no longer need their service after i;ve hooked up the electrical panel to it lol
09:03.45syle2anyways talking out loud again, ever since i been doing this telecom shit, i;ve been learning alot how about electrical as well, hell i could rewire my whole house like an electrician already lol
09:04.24drumkillatelecom ... home electrician .. same thing
09:04.28syle2but i decided it would be a real pain in the ass
09:04.33syle2unless it was a newly built house
09:04.41syle2then running electrical would be fun
09:05.21syle2hell i;'d run 30 40 and whatever amp lines to every room in the house just for fun
09:05.49syle2took a look at bill gates house specs
09:05.58drumkillaand a gig eth network
09:05.58syle2fucker has fiber optics running everywhere
09:06.19syle2damn must have cost some good coin for that wire
09:07.19drumkillai'm going to try to sleep now
09:07.20drumkillag'night
09:08.12syle2i don;lt know if you would call electrical and telecom the same thing
09:08.32syle2telecom is alot more the application layer , electrical is more the physical layer
09:11.02r3d5unhow can i setup the dialplan that after 30 seconds of no answer the caller is sent to the Voicemail. I didnt find it on the wiki they always assume that the User directly calls the voicemail number
09:11.03*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
09:13.05SarahEmmread the wiki..
09:13.09SarahEmmthe 't' extension is 'timeout
09:13.16r3d5unok thanks
09:13.30r3d5unthat was the word i've beenlooking for
09:13.51frenzyimum retries exceeded on call 1123232054-3380-ALI@192.168.1.110 for seqno 814 (Non-critical Response)
09:13.51frenzy<PROTECTED>
09:13.51frenzyI keep getting these 09:11:02 WARNING[8922]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 1123232054-3380@192.168.1.110 for seqno 814 (Non-critical Response)
09:15.02frenzy"09:11:02 WARNING[8922]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 1123232054-3380@192.168.1.110 for seqno 814 (Non-critical Response)"
09:20.28frenzy?
09:21.10*** join/#asterisk fanguin (~user@p548F2937.dip0.t-ipconnect.de)
09:22.32*** join/#asterisk lth (~lth@218.208.244.148)
09:32.41*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
09:33.02kb1_kanobedumb, offtopic question - what's the formatting for an 'immediate if' construct in c? ie. it goes something like "value ? true condition : false condition"
09:34.30kb1_kanobescrub that - answered my own question :-)
09:36.12r3d5unx = ( condition ) ? statement for true : statement for false;
09:36.30kb1_kanobegood, thanks. I was on the right track. :-)
09:36.45r3d5unyep :) but i didnt read your second line fast enough
09:36.49frenzyHow do I use remote config?
09:44.16*** join/#asterisk |cleric| (~dacleric@p548293A4.dip0.t-ipconnect.de)
09:45.18ltersSgtVimes, howdy :)
09:46.08formen|LSKcan someone name me a prefered sofphone?
09:46.26ltersfrenzy, what are you trying to accomplish? with remote config?
09:46.38ltersformen|LSK, firefly
09:47.11formen|LSKexpensive?
09:47.26ltersyeah, you have to download it. free otherwise
09:48.26ltersanyone get to test the a104d ?
09:49.10ltersformen|LSK, I guess you understood, it is free :)
09:53.23*** join/#asterisk Assid (~assid@203.115.64.59)
09:56.03frenzyI'm looking for a billing solution that is seamless with Asterisk & ser
09:57.10lthfrenzy: lots of hacks available - but seamless - doubt it very much.
09:57.51lthI guess it depends on the scale of what you'll be doing
09:57.58frenzylth: I need something that will allow easy operability of Asterisk & SER as a softwswitch
09:58.08*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
09:58.10frenzyand allows pre-paid (voucher based)
09:59.19lthWell - the thing with open source - it usually require some work to get it running - but the great thing is that it's possible.
10:00.02lthAsterisk is almost ideal to develop pre-paid solutions in since it stays in the signal path and can tear down the calls.
10:00.28lthSo it's definitely doable - but I seriously doubt you'll find a solution that just plugs in seamlessly without any work.
10:01.00frenzyI know bits requires customization...
10:01.06frenzywho does that?
10:01.19*** join/#asterisk FlungDung (~wayne@rndf-146-1-101.telkomadsl.co.za)
10:02.20lthWell - I guess it depends on the scale of the project.  I used to work for a billing system provider (did pre-paid for China Unicom among others - 65 mio customers) - but that's large scale.
10:02.59lthI assume you're not quite that size
10:04.34FlungDungHi all, my server shows no channels. Do the channels only come online once the PSTN line is attached to them or should they be there regardless?
10:06.17schlitzer_edhow can i setup a fail-over with asterisk?
10:06.23SarahEmmFlungDung: what card(s)?
10:06.25ltersshow channels only show active calls.
10:07.21schlitzer_edand how can i set up a load balanced asterisk (with two or more servers) they shoul handle the same extensions
10:07.24FlungDungWildCard TDM400P
10:07.26schlitzer_edany suggestions?
10:07.42ltersschlitzer_ed, sip? zap?
10:07.43SarahEmmFlungDung: Does it show up under show zap <something>?
10:07.49SarahEmm(i don't remember the something....
10:07.59ltersshould be zap show channels
10:08.07JerJerzip
10:08.12JerJer:)
10:08.13schlitzer_edsip / zap for load-balancing?
10:08.31SarahEmmlters: hrm.. there's some way to show them without any active calls, no?
10:08.42ltersschlitzer_ed, no, what kind of phones are u going to use
10:08.45JerJerschlitzer_ed:  you need to hire a consultant (not me) if you have to ask that question
10:08.45FlungDungmy channel is pseudo in the show zap channels
10:08.53ltersSarahEmm, yes, is "zap show channels"
10:09.18SarahEmmFlungDung: did you configure /etc/zaptel.conf, /etc/asterisk (or wherever) /zapata.conf, and run ztcfg before you started *?
10:09.24schlitzer_edokay, i want to use sip
10:09.30JerJerAsterisk will absolutely let you slit your own throat
10:09.36ltersschlitzer_ed, that is bad.
10:09.38FlungDungYup, /etc/zaptel.conf looks good.
10:09.47SarahEmmFlungDung: what does ztcfg -vvvvvvvv show from cmdline?
10:09.54JerJerthen laugh at you
10:09.56SarahEmm(reminder: don't paste long things here)
10:10.13JerJer~pastebin
10:10.13jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
10:10.37JerJereek - i think i see the sun
10:10.38schlitzer_edsip for the ip-phones, the servers should be connectet using iax
10:10.46ltersschlitzer_ed, sip sets up sessions. that makes 2 ip boxes complex to make work together.
10:11.29ltersif you could have your sip phones connect to ser, than use ser to figure out which * it might work ok.
10:11.43FlungDungRight, done heres the url http://pastebin.ca/19370
10:11.57ltersJerJer, the sun? its dark here :)
10:13.02schlitzer_edhmm, but the system should work even if one of the two+n servers fail
10:13.10SarahEmmlooks good FlungDung
10:13.14SarahEmmFlungDung: you set up zapata.conf too?
10:14.31FlungDungSara. It is using ostly defaults
10:14.50SarahEmmerr
10:14.51ltersschlitzer_ed, that would be the trick for ser to figure out. I have never used ser
10:14.51SarahEmmwell that won't work
10:14.59SarahEmmyou need to set it up for your system
10:16.14FlungDungHeres what it looks like at the moment http://pastebin.ca/19371
10:16.40FlungDungThe card has 4 fxo ports
10:16.51SarahEmmyeah.. that's not right
10:16.57FlungDung:(
10:17.01SarahEmmumm, have you read the wiki on this?
10:18.11FlungDungNo, I'll do that now. Every piece of documentation that I've read tells me to put in channels => 1 but when I do asterisk falls over untill I remove it
10:18.28FlungDung*channel => 1*
10:18.41SarahEmmdefine 'falls over' what error?
10:19.09FlungDungAsterisk died with code 1
10:20.44JerJerdo not run safe_asterisk until you know asterisk will run
10:20.54JerJerstart asterisk with:  asterisk -vvvgc
10:21.13JerJeronly after you are done with all configuration should you run safe_asterisk
10:23.08FlungDungRight, doing that now.
10:23.51DovidMorning All:
10:23.58Dovidanyone here use a gui for asterisk ?
10:24.15Dovid(for thier clients)
10:24.40ltersno
10:24.47lterserr not me :)
10:24.57JerJerwhy?
10:25.10FlungDungHmmm, the error seems to be "Ouch ... error while writing audio data: : Broken pipe"
10:25.19JerJerno that's mpg123 crap
10:25.21JerJerlook harder
10:25.36JerJeror 321 i forget
10:25.53FlungDung"== Unregistered channel type 'Zap'"
10:25.59r3d5unmpg321 is crap mpg123 is good, but mpg321 sets the link for mpg123
10:26.01JerJerkeep going
10:26.15SarahEmmuhh
10:26.18SarahEmmthat's an issue, no?
10:27.06JerJerthere is a reason why zap got unregistered
10:27.30JerJerperhaps his logger.conf is not setup properly
10:27.34JerJerHINT
10:27.56SarahEmmor the module just isn't loaded, no?
10:28.46Assidis there such a thing as randomly choosing an extension?
10:29.07FlungDungmodules are loaded "zaptel                178560   0 [wcusb wcfxs]"
10:29.09Assidlike they have for sales teams and call centers..
10:29.29JerJerAssid:  run a queue with whatever appropriate technology you wish
10:29.30JerJerround robin
10:29.34JerJerringall
10:29.35JerJeretc
10:29.40SarahEmmFlungDung: err, i meant Asterisk modules actually
10:29.49Assidotherwise it will always follow the same first this one.. then this one..
10:30.03JerJeri would bet the  wcusb crap is broken
10:30.17JerJerbut that's my opinon since i don't have any detail
10:31.34FlungDunglol, ok I found an error in the asterisks logs.
10:31.45AssidJerJer: queue is better than SIP/ext&SIP/ext2.... ?
10:31.54FlungDungAug  5 06:27:14 ERROR[3419]: Signalling requested is FXO Kewlstart but line is in FXS Kewlstart signalling
10:32.30JerJerAssid: depends
10:33.30Assidim worried about one thing tho.. if it is on ringall. what happens if another person calls in.. and goes to that queue.. the interfaces would return busy
10:33.54JerJerdoent work like that
10:34.08JerJeronly one call at a time is processed in app_queue
10:34.26FlungDungcool, I think I got it :) thanks guys
10:34.44Assidlike if i do Dial(SIP/ext1&SIP/ext2)  .. then if i call it rings both.. but if i try calling the same extension again..  and if nobody has answered my first call.. it just gives me busy
10:34.50JerJerThank you, drive thru
10:35.04Assidyou have to leave your first call ringing to try this
10:35.10*** join/#asterisk the_devil_dont_s (~Administr@62.77.178.121)
10:35.24JerJerthen dont do that
10:36.04the_devil_dont_shi there
10:36.07Assidanyway to do it like ringall.. but if one more call comes into the queue.. then the ring all is shared by the other call as well ?
10:36.24JerJer(06:34:05) JerJer: only one call at a time is processed in app_queue
10:36.51Assidwhat happens to the next call?
10:36.53the_devil_dont_si have a question about setting up an intel 536dep pci fax/modem/data in my asterisk system
10:37.06JerJerit gets processed after the previous one is dealt with
10:37.15the_devil_dont_sbefore any one starts on me about not using the digium wildcards this is just til i can see it working
10:37.36JerJerie the agent answers or the caller hangs up
10:37.57the_devil_dont_si have removed the r13 reistor but on my card there is no r19 that i can see, any ideas
10:37.59Assidwhat happens in the meantime?
10:38.07Assiddoes he keep listening to MoH?
10:38.15JerJeryep
10:38.19Assidexcellent
10:38.24Assidqueuing it is then
10:38.49Assidi wish i had a place to test it tho
10:38.54Assid:(
10:39.05Assidnot enough extensions
10:48.38the_devil_dont_sany ideas anyone
10:59.30*** join/#asterisk Newbie___ (me@211.24.146.12)
11:00.02Newbie___hi, can any one recommend a decent billing for asterisk ?
11:03.25*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
11:04.12Assiddoes the agent id have to be the same as extension?
11:04.12jontowAssid; hell no :)
11:04.12jontowput it this way
11:04.12jontowi have an agent id of 112 .. for my cellphone
11:04.18jontowi use AgentCallBackLogin ...
11:05.21jontowdial in; enter agent number, then password, then when it says new extension.. 918005551212
11:05.21jontowwhere 18005551212 is replaced with my number
11:05.21jontowjust gotta be REALLY careful about letting untrusted users do that..
11:06.02Assidnah.. they will be from a closed extension
11:06.02jontowtheoretically it can land you badass longg distance bills if someone really wants to make a long distance call and knows there are the correct conditions (only 1 agent, or ringall strategy, etc)
11:06.02jontowjust giving a tested example :)
11:06.05Assidhrmm
11:06.09*** join/#asterisk ramtha (~kuepper@td9091815.pool.terralink.de)
11:06.11ramthahi
11:06.12jontowmy cellphone # definitely doesn't match the agent # :D
11:07.06ramthahow can i aktivate clip/clir over the phone? my phone has a feature, dialing 1# meens, clip bit asterisk didnt recognize this
11:07.13ramthawhere is the magic?
11:10.36AssidAug  5 07:10:21 WARNING[28718]: chan_sip.c:2300 sip_write: Asked to transmit frame type 64, while native formats is 1024 (read/write = 1024/1024)
11:10.40Assidheres something weird
11:10.58ramthasounds like wrong codec?
11:12.21Assidgsm?
11:12.38Assidi really do need to have a bettewr testing that just calling the same machine i am on
11:13.55Assidwhen i put MoH.. for jumping between lines.. that error comes up
11:20.21*** join/#asterisk asteriskmonkey (~phil@69.158.154.80)
11:21.08*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
11:21.19asteriskmonkeydoes anyone have an example of ques.conf ?
11:26.03Assidi wish i could grab the archive of voip-info
11:27.58*** join/#asterisk startled (startled@d220-238-92-14.dsl.vic.optusnet.com.au)
11:29.36Assidanyonw wanna help me on this agent/queue system?
11:29.37asteriskmonkeyi have an asterisk card giving irq missies becuase its sharing an irq with a few things is there a way of chaning its irq on the cli in linux?
11:29.50asteriskmonkey11:   38688150          XT-PIC  usb-uhci, wcte11xp, eth0
11:40.39*** join/#asterisk djin (~djin@213-132-172-4.multikabel.nl)
11:41.36Assidshouldnt a agents number ring ? instead of just start up..
11:42.31*** join/#asterisk grimse (~grimse@p5481F39E.dip.t-dialin.net)
11:42.34*** join/#asterisk sparky0001 (~mark@mark.keele.netcentral.co.uk)
11:43.17asteriskmonkeyassid: ?
11:43.55Assidlike i logged in as an agent..
11:44.00Assidthe call came in directly to me
11:44.03Assidbut..
11:44.11Assidi want to be able to "pickup" the call
11:44.25Assidhaha.. this is funnny   "Aug  5 07:43:25 WARNING[28718]: app_queue.c:2037 try_calling: Agent on Agent/1001 hungup on the customer.  They're going to be pissed."
11:44.28Assidlol
11:44.37formen|LSKlol
11:44.46Assidi wanna pickup the call instead of it just coming into my line
11:44.56*** join/#asterisk zeitgeist_y2k (~ceicke@mail.inmedias.it)
11:44.59Assidlike i press # or something.. to answer the call
11:45.02sparky0001Can someone please help, I am replacing our PBX and have used Asterisk with a Digium E1 card, when calls come in the CallerID showing on the phones is missing the leading zero (which we expect to see in the UK) and I cannot seem to find a way to correct this. Our old PBX does seem to present a zero so I am unsure if this is BT or the card / asterisk that's not presenting the zero?
11:45.30Assidsparky0001: you can always prepend the 0's
11:45.45Assidset(callerid)=00${CALLERID}
11:45.53zeitgeist_y2kcan anyone help me with my CAPI channel problems?
11:46.49Assidasteriskmonkey: any clue on this?
11:47.47sparky0001Assid: The biggest problem with that is that the calls are going direct to the DDI numbers and I cannot seem to put a global overide in place so it means a re-write of the system to put the code on each DDI number I was hoping it could be done another way?
11:48.21asteriskmonkeyits all in how your extensions.conf is setup
11:48.40Assidsparky0001: exactly what monkey said
11:48.58Assidasteriskmonkey: i meant regarding "answering an agent line"
11:49.03Assidi dont want it to be thrown
11:53.08asteriskmonkeyok
11:53.22asteriskmonkeypaste your extensions.conf on pastebin.ca so we can all have a look
11:53.45sparky0001I was using AMP and have modified this significantly but when I have done all the work I was using an external ISDN sip box which was always presenting the CallerID with a leading zero (in fact all the devices I have been using seem to do this) it's just when I got my Digium cards I found out that actually the zero is not present at all which now causes me a real headache as I can't use Digium cards and external hardware?
11:55.40Assidokay im outta here.. catch you guys later
11:56.05startledsparky0001: You sure they're dropping the 0 and it's not that the others have been adding it on at some point? I've not got my hands dirty with asterisk yet, but every switch I've used leaves it off
11:56.34sparky0001startled: are you using UK switches?
11:56.56asteriskmonkeyah.. amp bad
11:57.27asteriskmonkeyi started out wit amp cause i used asterisk@home to begin wish i didnt it stunted my learning curve bad :P
11:58.22startledsparky0001: Nah, I'm in Oz but I'm pretty sure our hardware is the same.
11:58.23Assidamp??
11:58.36*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
11:58.37asteriskmonkeyasteisk management portal
11:58.51asteriskmonkeynewbie friendly webpage based asterisk configuerer
11:59.09startledanyways... I'm not going to be offer you any specific advice on how to fix it anyways (yet)
11:59.16asteriskmonkeybut you really shouldnt use it if you want to do good things with your asterisk box
11:59.21sparky0001I started without it and then added it as I was lazy and it gave me a whole load of functions but mainly the web interface to allow office workers to use it
11:59.33Assidhrmm
11:59.42sparky0001It has been modded quite a bit though
12:00.11Assidalrite.. catch you guys in a bit
12:00.17Assidgotta go lose some weight
12:01.03sparky0001That's my big problem really as the other units always presented a leading zero so I 'Assumed' this was correct and have written everything for this
12:01.16*** join/#asterisk mayank (~mayank@210-210-81-203.lan.sify.net)
12:03.47mayankhello ... I want to develop a new application for Asterisk. Can anybody guide me through how to go about this?
12:03.47mayankTo be more specific, say I write a small application ISPY, and include it as the first thing in the dial-plan. So how do I register it ?
12:04.27*** part/#asterisk sparky0001 (~mark@mark.keele.netcentral.co.uk)
12:10.28*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
12:12.44*** join/#asterisk tecONE (~tecONE@port-83-236-180-106.static.qsc.de)
12:13.41*** join/#asterisk grimse (~grimse@p5481F39E.dip.t-dialin.net)
12:16.19*** join/#asterisk sparky0001 (~mark@mark.keele.netcentral.co.uk)
12:18.29sparky0001I was just here asking about Asterisk and Digium Hardware stripping the digits from callerID and it was suggested that out telco (BT) prbably do not send a leading zero for the dialing code.
12:19.07sparky0001I have just checked BT's service specification sheets and they say they actually send the number in the format of 01234-3435345
12:21.04startledsparky0001: I'd think it would be more likely that within the hardware or asterisk the calling NUMBER is treated as a number, and as such loses the leading zero. As I said, a side effect I've seen on every switch I've dealt with personally
12:21.04sparky0001They then say that any hardware supplier should only strip the non numeric characters from the stream and this allows for call return to happen, anoyone know why that Digium hardware (TE11XP E1) cards seem to strip this
12:21.34*** join/#asterisk zotz (~zotz@24.231.36.100)
12:21.55sparky0001startled: Ah that makes some sense as you canno store numbers beiginning with a zero in a database either
12:23.24tecONEhello I have a little question...., I have a problem with asterisk 1.0.7 and an EICON ISDN card. If I call from my sip phone to my mobile, I can't hear anything on both sides just silence.... From my mobile to the sip phone it works well. has this anything to do with alsa.conf or oss.conf? thanks!
12:25.20sparky0001Does anyone know how to actually get Digium to answer calls??? I have tried several times and just get a leave message, the problem I have is that non-digium hardware (external line ATA's) all present asterisk with a leading zero but internal cards to the system do not so really in the zapata.conf we need an option to add a zero for UK to make it uniform?? thoughts please?
12:25.32*** join/#asterisk zoo (nobody@ip-46-16.travedsl.de)
12:25.41*** join/#asterisk frenzy (~frenzy@193.220.82.108)
12:25.56asteriskmonkeyah you have non digium hardware
12:26.44asteriskmonkeyits all configurable in asterisk
12:27.00asteriskmonkeyif you post your configs on pastebin.ca some people might be able to see whatss going on :D
12:27.10zoohello
12:29.36zooI am using asterisk to hook up many SIP-Accounts to my SIPURA SPA-1001 phone. All incoming calls are using Dial(SIP/user@myrouter.dnydns.org), so that all calls are reaching me. Like this all calls are routed through my asterisk. Is there a way to use something like "302 REDIRECTED" to redirect directly to my phone??
12:29.40tecONEasteriskmonkey: sorry, do you mean me?
12:29.49zooor like a HTTP Location:-Header?
12:32.35*** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
12:33.11blitzragezoo: if you want to use 302 Moved Temporarily, then you need to use the Transfer() application (only in HEAD)
12:33.31zooblitzrage: okay, great, thanks
12:33.41blitzragezoo: the phones aren't behind NAT are they?
12:34.08blitzragezoo: the other thing is that just the signalling goes through Asterisk, and the media goes between end points using canreinvite=yes
12:34.16zooblitzrage: yes they are, but don't worry, I do manage to connect to them :)
12:34.56zooblitzrage: but the phones are not registered at asterisk
12:35.01blitzragezoo: I just don't think the Transfer() app will work for devices behind NAT as the router isn't going to know how to send the media to the end points...
12:35.25blitzragezoo: either way, Transfer() is what you want for 302 Moved Temporarily :)
12:35.32zooblitzrage: the phones are reachable from the network through siproxd and portforwarding
12:35.46blitzragezoo: coo
12:36.01zooblitzrage: thoe only thing that asterisk knows about my phones is the line in extentions.conf: Dial(SIP/user@myrouter.dnydns.org)
12:36.18*** part/#asterisk sparky0001 (~mark@mark.keele.netcentral.co.uk)
12:36.23zooblitzrage: "coo"?
12:36.32blitzragecoo == cool
12:36.36zoom'kay
12:37.09zooactually, it is only siproxd
12:38.34formen|LSKcan someone recommend a good sip soft phone?
12:38.48formen|LSKfree
12:39.01tecONEformen|LSK, xlite?
12:39.06blitzrageformen|LSK: X-Lite
12:39.18*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
12:39.19formen|LSKthx
12:39.19blitzrageformen|LSK: yes, it works in Linux (under Wine)
12:40.42formen|LSKok
12:41.03tecONEblitzrage, v2 of xlite is now available for os x an linux. you don't need wine anymore :)
12:42.11tzangeris v2 any better than v1?  and have they given me the option of a nonskinned UI?
12:42.27frenzylooking for someone to do an Asterisk setup and impliment a simple billing solution.. pls PM
12:43.26blitzragetzanger: totally agreed
12:43.26tecONEdon't really know if v2 is better... but I hope so ;)
12:43.31blitzragetecONE: oh yah? cool, didn't know that :)
12:44.55tecONEtzanger, no, it seams that there is the same skin than in v1, and no possibility to remove it.... :(
12:46.16*** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com)
12:46.59frenzyI'm looking for someone to do an Asterisk setup and impliment a simple billing solution..
12:47.46*** join/#asterisk eldu (~damajor@tuxmania.org)
12:47.49elduhi hi
12:48.08blitzragecan't believe I woke up at 8am today
12:48.23blitzragethats good, I can work until 5, then start drinking :)
12:48.33eldu:)
12:48.52elduone thing to fix and i can leave when i want
12:49.07blitzrageeldu: doh! :)
12:49.08formen|LSK[14:48] <blitzrage> can't believe I woke up at 8am today
12:49.26blitzrageformen|LSK: yep, thats what I said :)
12:49.53blitzrageformen|LSK: normally I wake up around 11am (due to working until 3 or 4am though)
12:50.24*** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no)
12:51.27formen|LSKjeez, go find a new job ;)
12:52.30blitzrageformen|LSK: I work from home (self-employed) :)
12:52.44blitzragealthough no idea why I do, I probably work more hours for less money
12:52.52blitzragebut I love it
12:52.53tzangerblitzrage: that's usually how it goes :-)
12:53.11blitzragehehehe
12:53.35formen|LSKu need to put up some moneymaking products that dont need that much work
12:53.36tzangerI have the kernel followed directly my the romfs.  I have romfs and even cramfs enabled and a 1M ramdisk (no initrd) but it can't find the root device
12:54.13tzanger"VFS: Cannot open root device "<NULL>" or unknown block device (0,0) -- well DUH it's a standalone image, you're supposed to look past the end of the kernel for the romfs
12:54.24tzangerI wonder if I rdev it and tell it it's on a floppy
12:56.15blitzrageformen|LSK: I suppose, but I'm more into writing and training. I do peice work for a company in Florida to help pay the bills.
12:56.50SuPrSluGhi all
12:57.27*** join/#asterisk mut (~animenodv@65.111.201.79)
12:58.15SuPrSluGis there a way to dial based on time. I want to use my free minutes, then go to my voip provider.
12:58.40blitzrageSuPrSluG: GotoIfTime()
13:00.02SuPrSluGthanx i'll read up on that. i thought it was a open-close type app only.
13:03.41*** join/#asterisk T-Squared (~ted@hidden.serreyn.com)
13:09.53elduBefore the dial, how can I set the output telco number on a SIP trunk of 10 telco numbers ?
13:10.10blitzrageeldu: huh?
13:10.28eldublitzrage: i got a trunk with 10 telco num in
13:10.31*** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
13:11.14IPDanyone complie h323 yet on debian?
13:11.29eldui thought (need to verify) that my trunk provider support callerid and how i can set the output number
13:11.47IPDIm looking for openssl-devel-0.9.6b - can't find the source for debian 3.1 anyone got an idea?
13:12.06blitzrageeldu: ${CALLERID}
13:12.23EthonWhat's the correct way to make anonymous calls using a bri channel with * ?
13:12.25eldublitzrage: i tried that ... without success
13:12.48elduEthon: SetCallerID(anonymous)
13:13.05EthonI'll try
13:13.49eldublitzrage: the prob is that i dunno the telco number that goes out
13:14.45blitzrageeldu: what wasn't successfull about it? I think I'm confused as to what your problem actually is
13:15.13eldublitzrage: well i got 10 telco nums entering in a SIP trunk
13:15.22elduthat works very well
13:15.59elduand my provider send me callerids from others
13:16.38elduoutput dials work fine too, BUT i dont know which telco num is used to go outside
13:16.57eldumoreover I dont have callerid on external phones
13:17.14*** join/#asterisk coppice (~chatzilla@223.203.17.210.dyn.pacific.net.hk)
13:17.19eldunot "anonymous" msg but "no callerid received"
13:17.37blitzrageeldu: not really sure how to fix that... seems like a far end problem, not Asterisk
13:17.37eldunow i think u should be able to locate my prob ;)
13:18.23*** join/#asterisk Stephnie (st@203.215.180.254)
13:18.28elduthe thing that distrub me is that i never set the output telco number, so i dunno which one is used
13:20.21blitzrageeldu: it doesn't matter.. its just a channel. The far end is just using a channel too. The number really only matters when the call is incoming (so it can determine where to route)
13:20.47eldui see
13:21.06elduso just doing a properly "SetCallerID" should work
13:21.23blitzrageyep, assuming the far end is accepting it and passing it on
13:21.31eldusure
13:21.58eldui will certainly have a good WE :)
13:22.01elduthx blitzrage
13:22.08blitzragenp
13:22.48*** join/#asterisk sangee (~rkuru@207.188.77.86)
13:23.57*** join/#asterisk Godsey (~lanny@Godsey.sustaining.supporter.pdpc)
13:26.01Stephniehi...
13:26.37gordonjcphello
13:27.03StephnieI have 4 peers to dial out  ......[outgoing1], [outgoing2], [outgoing3], [outgoing4] . . I need to dialout from available peer..but I am unable to know which one is busy..
13:27.16Stephnieany help please?
13:27.18blitzrageStephnie: ChanIsAvail()
13:27.32elduperhaps SER could do the job too ?
13:28.17sangeei got this error when i start asterisk " chan_zap.so: load_module failed, returning -1"
13:28.23sangeewhat could be the issue?
13:28.49Stephnieblitzrage: yes I tried that..but the problem is ....these 4 peers are SIP based means provide multi calls from 1 peer....so I dont get CONGESTION or UNAVAILABLE return . .
13:29.17h3xStephnie: The answer is, don't use a crappy provider :P
13:29.37Stephnie:D ... most of them provide multi calls from 1 account...
13:30.24*** join/#asterisk likwid-- (~likwid@nc-69-68-66-219.dyn.sprint-hsd.net)
13:30.24Stephnieand I want to use 1 account for 1 call..... that is why CHANISAVAIL() is almost useless...
13:31.12Stephniehow do I know that Asterisk PEER is busy other than Provider is busy or congestion..
13:32.45jpmcallisterStephnie: what about the setgroup function
13:33.30StephnieI am not familiar with that function....should I check wiki if thats the solution of my prob?
13:34.37blitzrageStephnie: yes, if you want to only allow a set number of calls to one peer, you want SetGroup() and CheckGroup()
13:34.47blitzrageshow application setgroup
13:35.09Stephnieokey...thanks....going to check..
13:35.40blitzrageStephnie: if running HEAD, look at the ${GROUP()} and ${CHECK_GROUP()} functions (show functions - from the CLI)
13:36.02blitzragesorry... not ${CHECK_GROUP()}, its ${GROUP_COUNT()}
13:36.14StephnieI am running CVS
13:36.17Stephnie:(
13:36.24h3xthat is head
13:36.29blitzragewell, CVS can either be the 1.0 or HEAD branch :)
13:36.31Stephnieoh :D
13:36.52Stephnielet me read about that function first...
13:37.02blitzrageStephnie: do a 'show version', you should get something like: Asterisk CVS-HEAD built by root@pbx.leifmadsen.com
13:37.02h3xshe would have known that if she set the 1.0 revision
13:37.02h3xheh
13:37.15blitzrageh3x: not necessarily - I've seen stranger things :)
13:37.50StephnieAsterisk CVS-HEAD-06/02/05-08:36:40 built by root@MYLINUX on a i686 running Linux
13:37.52RoyKoverriding callerid for SIP-SIP calls doesn't work :(
13:38.21mjmacwoo...  net4801 with a tdm400p works just fine
13:39.21tzangerwhy wouldn't it?
13:39.27blitzrageStephnie: yep, then you want to start learning how to use dialplan functions as many applications will go bye bye at some point
13:40.21mjmactzanger: just heard nay-saying and doom and gloom about it...  but i've got the net4801 and the tdm400p both running off of the same 12V supply and it's fine.
13:41.31tzanger??  that's odd
13:41.40blitzrageaye
13:41.45blitzragebreakfast time!
13:41.58tzangerblitzrage: damn I want breakfast
13:42.09tzangerbacon and eggs and saussage and grits and coffee and oj ...  mmmmmmmmmmmmmmm
13:42.33coppicecroissants, there must be croissants
13:42.39tzangerno
13:42.46tzangercroissants I was never partial to
13:42.46*** join/#asterisk funxion (~nunya@mtnuser.icgws.com)
13:42.49mut..
13:42.51tzangergimme texas toast
13:43.03mutbacon and eggs
13:43.07mutand some toast
13:43.10mutthats all ya ever need
13:43.16NuggetWhat is texas toast?
13:43.18coppiceis that like french toast, but with oil instead of egg?
13:43.29*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
13:43.37mutit's just thick toast..
13:43.45tzangerNugget: just thick toast slices
13:44.00mutwow
13:44.03mutkeep saying toast
13:44.05tzangerfor me it has to be either well buttered or sugarless peanut butter
13:44.06mutit sounds really weird
13:44.16tzangerthe kraft blue-lid peanut butter is my fave
13:44.17muti hate when that happens
13:44.21h3xyou just gave me an idea
13:44.21zeitgeist_y2ktzanger, sounds all pretty unhealthy to me...
13:44.22tzangerI can't stand regular peanut butter anymore
13:44.24sangeeanyone know this issue " chan_zap.so: load_module failed, returning -1"?
13:44.25tzangerzeitgeist_y2k: nonsense
13:44.27h3xthe world's first voip phonesex services
13:44.34tzangersangee: have you run ztcfg?
13:44.41tzangerzeitgeist_y2k: gotta lube up the arteries
13:44.50sangeeyes
13:44.51zeitgeist_y2ktzanger, b?h
13:44.56mutprobably a vegan eh zeitgeist_y2k
13:45.10zeitgeist_y2kmut, no way, but for breakfast I cannot stand grease
13:45.28zeitgeist_y2knothing like cornflakes with yoghurt :-)
13:45.38*** join/#asterisk darkskiez (~mhb@host-84-9-79-99.bulldogdsl.com)
13:45.44*** join/#asterisk frenzy (~frenzy@193.220.82.108)
13:45.45mutheh
13:45.50muti never even eat breakfast
13:45.59mutless i stop at mcdonalds for a bacon egg & cheese bagel
13:46.06*** join/#asterisk glLoadIdentity (~tuyan@81.214.255.57)
13:46.11frenzyhow do I impliment canreinvite (am behind NAT)
13:46.59glLoadIdentityhi all.. what is the number of maximum concurrent calls with g729 passthru with dual xeon ?
13:47.17mutlike
13:47.23mut500
13:47.49mutif you're not encoding/decoding anything
13:47.54*** join/#asterisk mkrufky (~mk@68.160.103.77)
13:47.59mutit takes like no cpu
13:48.13frenzy?
13:50.24*** join/#asterisk ellvis (~ellvis@adsl-data-148.84-47-83.telecom.sk)
13:50.29nighty-Any idea why linphone would not auth properly against asterisk ?
13:50.32ellvishi people
13:50.36nighty-hi
13:51.10ellvisdoes anyone know how/if it's possible to set up CallerID on outgoing ZAP channels?
13:51.22fearnorellvis what kinda zap
13:51.30fearnoras in, pstn or pri?
13:51.38glLoadIdentityit should be possible with pri
13:51.41ellvisfearnor: BRI with DDI numbering
13:51.47*** join/#asterisk slak- (slak@undroppable.co.uk)
13:51.56fearnorell: you *might* be able to. depends on your telco. technically, possible.
13:52.00fearnortry and see.
13:52.27ellvisi know that with classic BRI it's probably not possible, but i believe it should work with DDI numbers
13:52.33mutmy provider blocks my cid
13:52.38fearnorhm, why wouldn't it be not possible?
13:52.39*** join/#asterisk azrishahril (~azrishahr@60.50.207.209)
13:52.42fearnorits same q.931
13:52.53fearnorsame signaling to place call
13:52.55fearnoras pri
13:53.29RoyKeldu: it's possible. SetCallerID etc
13:53.42ellvisfearnor: i was checking wiki for a correct syntax. is 'exten => _X./204,1,SetCIDNum(59103704,a)' the right way?
13:53.44slak-hey guys im trying to set up a conference app MeetMe for my boss and what i did was add "conf => 200,9966" to meetme.conf and the following exten lines in extensions.conf: http://pastebin.ca/19386 but then when i call in it asks me to authenticate and then i see this in logs: Aug  5 09:49:56 WARNING[11141]: pbx.c:1291 pbx_extension_helper: No application 'MeetMe,' for extension (toll-access, 200, 4)
13:54.06fearnorlooks about right. but see the logs etc
13:54.19ellvisfearnor: ok, thanks!
13:54.19file[laptop]slak-: do you have zaptel timing? did you add it after you compiled asterisk? if so, none of the zaptel stuff was compiled (like meetme)
13:54.33slak-file i have a digium card
13:54.41frenzyAug  5 13:53:15 WARNING[14198]: rtp.c:1460 ast_rtp_bridge: codec0 = 14 is not codec1 = 1024, cannot native bridge.
13:54.42*** join/#asterisk santiago (~santiago@63.245.86.172)
13:54.52slak-and from the looks of it MeetMe is compiled. but my configuration is flawed
13:55.02slak-could u please take a look at that pastebin link its only 5lines
13:55.09file[laptop]oh you have an , where there shouldn't be on
13:55.22file[laptop]er one
13:55.31slak-im sorry where?
13:55.36*** join/#asterisk clinthome (~clinthome@mail.dasaonline.com)
13:55.40slak-og
13:55.41slak-oh
13:55.42file[laptop]exten => 200,4,MeetMe,(200|p) should be exten => 200,4,MeetMe(200|p)
13:55.43slak-line 4?
13:55.44slak-k
13:56.22ellvisfearnor: on BRI i have doubts as it's not working, the DDI will check later today
13:56.24slak-okay
13:56.31slak-why did it ask me for the password twice
13:56.42slak-is it cause i have a pass in meetme.conf and Authenticate?
13:56.49file[laptop]yes
13:57.13*** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985765.sympatico.ca)
13:58.45slak-Aug  5 09:54:44 WARNING[11154]: app_meetme.c:962 conf_run: Unable to write frame to channel: No such file or directory
13:58.50slak-not sure what that means
13:59.03file[laptop]someone hung up? ;)
13:59.07slak-think it happened when i hung up\
13:59.38file[laptop]ooh I'm psychic
14:00.02slak-hey this is odd
14:00.12*** join/#asterisk fanguin (~user@p548F2937.dip0.t-ipconnect.de)
14:00.12slak-this works when i had my coworker join in
14:00.20slak-but when i tried to dial in from the outside it failed
14:00.22slak-heres the output
14:00.35slak-Aug  5 09:55:44 WARNING[11161]: pbx.c:1935 ast_pbx_run: Invalid extension '0', but no rule 'i' in context 'incoming'
14:01.05slak-this is when i tried to use my cell to dial in and then it answered and i entered 200 for the meetme conf
14:01.16slak-200 works when i call from my office phone
14:02.23slak-is it wrong to have an Answer in that meetme exten
14:03.05mutbumble bumble
14:03.32ManxPowerslak-: It looks like you have relaxdtmf=yes on your Zap channels
14:04.03ManxPowerslak-: That can cause Asterisk to incorrectly think you were dialing "0" when you were actually dialing something else.
14:08.19*** join/#asterisk astoria (~cluecon@67.107.50.40.ptr.us.xo.net)
14:08.23astoriaGood mroning.
14:08.39ManxPowermornings are never good
14:08.42mutlook at the sun
14:08.53astoriai need some coffee
14:09.09*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
14:09.43Wonkaaargh. i hate that movie.
14:09.57mutanyone live in a better job market than michigan? i need to expand to another state or states
14:10.12astoriaI love michigan.
14:10.23astoriaTo me, it feels like the best job market in the country.
14:10.23ManxPowerI like Michigan....in the summer.
14:10.39mutheh
14:10.44astoriaI have no problem getting a job in MI, but lots of problems in cool cities, like Chicago or NYC.
14:10.51astoriaPlus, I love Detroit.
14:11.05mutwell i've been applying for stuff for a year and a half now
14:11.17mutand the only reply i got was when i applied in alaska one tim
14:11.25mutrest are just FOAD's
14:11.35slak-hey question for you guys can i take an existing conference and place a call to someone and connect them
14:11.41*** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com)
14:11.44astoriaanyone here at cluecon understand the coffee situation at this hotel.. i don't want to pay 5 bucks for coffee
14:11.49slak-that someone can only accept incoming calls and needs to be in the conference
14:12.02slak-not necesarily actually joined in just able to hear/speak
14:12.14mishehuugh.
14:12.29file[laptop]astoria: I can like, oh lemme see
14:12.30slak-like 3way a conf to another number
14:12.36mishehuastoria: they charge for coffee there?  or you talking about the coffee in the rooms?
14:12.51slak-love detroit...god whats wrong with u ;)
14:12.57astoriamishehu: coffee downstairs
14:13.07mishehuastoria: they charge for that coffee???
14:13.15mutcharge for coffee? wtf
14:13.20astoriamishehu: you have to pay for the continental breakfast!
14:13.21mutit costs like 10 cents for 20lb ofit
14:13.40mutyou poor souls
14:13.45file[laptop]astoria: Tony might go to Starbucks again and get good coffee :)
14:13.49mishehuastoria: you can't just say "but all I want is caffeine to the main vein" ?
14:13.57ManxPowerPeople that charge for coffee go to hell.
14:14.07astoriafile: i'll throw in money for coffee.
14:14.22astoriai don't care if it's crappy coffee
14:14.28mishehufile[laptop]: even though I'm not a coffee drinker, I do know that starbucks != good quality coffee.
14:14.38file[laptop]it was good yesterday
14:14.42file[laptop]according to everyone
14:14.49file[laptop]and very much appreciated
14:14.49astoriayeah, it was great.
14:14.50file[laptop]:P
14:14.59astoriastarbucks coffee is top notch.
14:15.06Cherebrummorning
14:15.06Ariel_coffee... hummm I drink more what starbucks calls Latte.
14:15.06ManxPowerfile[laptop]: Um, from what little I know about eastern canada.....y'all don't know Good Coffee.
14:15.11file[laptop]ANYWAY, I'm gonna run away - brb
14:15.14file[laptop]ManxPower: I don't drink coffee
14:15.25file[laptop]I'm going on what people said
14:15.39Ariel_file[laptop], you should try some espresso
14:15.44ManxPowerfile[laptop]: You don't drink coffee?  You're weirder than I imagined. *tease*
14:16.00mutyay
14:16.01mishehuthey should serve "bots" (mud, turkish coffee) at starbucks, just so I could go there and find out how bad they make it.
14:16.06muti just got another FOAD email!
14:16.18mutactaully 2 of them at the same time.. just wanted to rub it in i guess
14:16.28Ariel_there is nothing like a good Cuban cafe'
14:16.44mishehufile[laptop]: run away?  you can't run away at cluecon
14:18.10*** join/#asterisk pa (~Paolo@pa.user)
14:18.20mishehuis the asterlinux presentation good?  I'm missing it, couldn't get out of bed.
14:18.23*** join/#asterisk FaUl (~immo@hobbynuttenverzeichniss.de)
14:18.29FaUlhi
14:19.53astoriamishehu: it's good.
14:20.30file[laptop]Katty is ubertired :<
14:20.49mishehuastoria: I imagine it's mostly about how it was put together to fit on 32mb
14:21.12mishehufile[laptop]: why is everybody seeming to be obsessed with what Katty is doing during a presentation?
14:21.15mishehuheh
14:21.29*** join/#asterisk clinthome (~clinthome@24.75.94.8)
14:21.30file[laptop]mishehu: uhm, i'm katty (using file's laptop)
14:21.37astoriamishehu: he's talking about things you can do with it.
14:21.38ManxPowermishehu: I'm sure this is the first time many of them have been this close to a female.
14:21.48mishehufile[laptop]: THIEF!
14:21.51mutburn
14:21.54IPDhas anyone here comlied h323 with debian?
14:21.57file[laptop]mishehu: he's not using it (=
14:22.11astoriaI'm the one who wants the coffee..
14:22.13Cherebrumkatty: hi
14:22.18mishehufile[laptop]: look for his sekkrit documents while he's not looking
14:22.18file[laptop]ManxPower: don't drink coffee, but thanks anyway (=
14:22.28FaUli've some problems dialing out with zaphcf from latest bristuff-0.2.0-rc8j
14:22.34file[laptop]Cherebrum: allo (=
14:22.47*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
14:23.04Cherebrumkatty: I'm on the far right 3rd row back
14:23.24file[laptop]Cherebrum: don't forget to say hi sometime today (=
14:23.27Beirdo_poor girl.  being geek-mobbed :)
14:23.30*** join/#asterisk PhreeStyle (~PhreeStyl@cpe-24-221-52-165.az.sprintbbd.net)
14:23.32Cherebrumkatty: ok
14:23.34Cherebrum:)
14:23.39mishehuManxPower: well, if it was one of my LUG installfests instead of cluecon, she'd be swarmed.
14:23.43file[laptop]Beirdo_: quite the opposite, actually
14:23.46mutanyone here a really good resume writer? i need mine tweaked i think..
14:23.48mut;P
14:23.52FaUlit sayes everytime that "No one is avilable to answer at this time"
14:23.52Beirdo_good to hear, file :)
14:23.55file[laptop]Beirdo_: absolutely no one is talking to me here...
14:23.55astoriathe geek girls i know would LOVE the attention.. ha ha
14:23.55Cherebrumkatty: pay attention to Kristian presentation.. his project is really awsome
14:24.04FaUlwhich can't be true, any hints?
14:24.08mishehuthe guys at my LUG will hit up on anything with boobs.
14:24.11file[laptop]Cherebrum: i am (=
14:24.14*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
14:24.14*** mode/#asterisk [+o bkw_] by ChanServ
14:24.49Beirdofile[laptop]: you having fun though?
14:25.18mishehufile[laptop]: do you know who I am now?  the other day you said you didn't know me.  surely you've seen me typing stuff here before...
14:25.24file[laptop]Beirdo: sure
14:25.52file[laptop]mishehu: have you said hi to me yet?
14:26.15mishehufile[laptop]: on wednesday at least.
14:26.40file[laptop]mishehu: then i've probably seen you, even if i can't match your /nick to a face
14:26.43mishehufile[laptop]: of course, brian couldn't connect faces with names with nicknames...
14:27.30file[laptop]OKAY FOLKS
14:27.35file[laptop]20-30 minutes until coffee arrival
14:27.42astoriaTHANK YOU FILE
14:27.44astoriaYOU RULE
14:27.49file[laptop]same as yesterday.
14:27.50*** join/#asterisk pif (ldm@zenon.apartia.fr)
14:28.05mishehucoffee schmoffee
14:28.23ellvisfearnor: it's working
14:28.28Beirdofile[laptop]: cool. :)
14:28.29mishehuI woke up with a hangover, which is odd, because I didn't drink anything last night.
14:28.30ellvisbye all, have a nice weekend or so:)
14:28.33mutomg
14:28.45Beirdo20-30 min for coffee?
14:28.48file[laptop]mishehu: the combined alcohol intake of everyone has affected ... everyone
14:28.58Beirdoyou guys need to go back to bed for 15-25min
14:28.58file[laptop]Beirdo: that's what Starbucks said
14:29.00blitzragewhat? people drink at conferences?
14:29.15mishehufile[laptop]: no wonder I was getting my ass kicked at kwak3 last night.
14:29.25Beirdommmm, drink
14:29.39mishehuBeirdo: you're not here at the conference are you?
14:29.49Beirdono
14:29.50Beirdo:)
14:29.55Beirdoonly in spirit
14:29.57file[laptop]too bad
14:30.08astoriawow, this astlinux is really impressive.
14:30.25Beirdoyeah, saving up my vacation-like time for trips to the caribbean
14:30.26mishehuastoria: I'll need to watch the video later
14:30.45*** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-34.w193-253.abo.wanadoo.fr)
14:31.11mishehuand I need to grab my stuff and get going to the hotel or else I'll miss the next presentation too
14:31.12jhiverhi all
14:31.40astoriais there a way to create dynamic contexts using realtime, as well as dynamic extensions?
14:31.49jhiveris it possible to run more than 1 instance on asterisk on a given box (i.e. say, using one IP address for each instance)?
14:32.03file[laptop]astoria: the way the pbx core works, not really
14:32.19astoriafile: thats what i suspected. thanks, just wanted confirmation on that.
14:32.21file[laptop]the context needs to exist in extensions.conf so that the switch for realtime can be used, switches can't be used... elsewhere...
14:33.01Ariel_ManxPower, are you around? I have a quick question for you.
14:33.06*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfl1p.dialup.mindspring.com)
14:34.33astoriathanks file
14:34.38file[laptop]yw
14:34.45*** join/#asterisk funxion (~nunya@mtnuser.icgws.com)
14:35.02Ariel_ManxPower, did you post the settings you had to put into the t1 connection for the pbx connection you were working about 6 months ago?
14:35.13mutholy crap
14:35.15astoriadynamic IVR creation would work better in AGI, than realtime, then.
14:35.29mutmonster.com charges $170 for someone to create a resume
14:35.46mutit better be plated in gold
14:35.49file[laptop]astoria: meh I'm hesitant to use AGIs now a days because they don't scale
14:35.51mutand get me any job i ever apply for
14:36.25file[laptop]astoria: but it would depend on what you mean by 'dynamic IVR creation'
14:36.26Ariel_does anyone one have the asterisk side for a TE110p connection to an T1 E&M wink Nortel MICS system?
14:36.32astoriafile: how would you go about doing dynamic database driven IVR's then? dynamic .conf rewriting?
14:36.45astoriafile: actually, thats probably the best idea..
14:36.53file[laptop]:)
14:37.11funxionAriel_ what kind of trunk? DID?
14:37.22*** join/#asterisk Xumpi (~SysOp@a81-84-68-51.cpe.netcabo.pt)
14:37.24Ariel_funxion, yes
14:37.28Xumpiyellow
14:38.01funxionlet me see what I havee
14:38.02*** part/#asterisk T-Squared (~ted@hidden.serreyn.com)
14:38.09Ariel_we can send calls to there system fine. But we don't seem to be getting all there digits.  And 1/2 the time we only get the first digit.
14:38.36file[laptop]bkw_: did tony leave?
14:38.49Xumpiis there any freebsd port which i can use as an iax client ?
14:39.00*** join/#asterisk fanguin (~user@p548F2937.dip0.t-ipconnect.de)
14:39.04Ariel_funxion, thank you
14:39.16file[laptop]wikibob to bkw_
14:39.51funxionAriel_ check out http://pastebin.ca/19388 see if that helps
14:40.04file[laptop]peter, poke brian
14:40.38Ariel_funxion, there not using the pri board but the older t1 board.
14:40.42*** join/#asterisk MattH (MattH@63.174.244.175)
14:41.07mishehualright, time to go
14:41.11MattHHi.. when doing something like exten => s/number  should I be able to do => s/570111XXXX to match anything coming from 570111? like 5701112222 and also 5701113333 ?
14:41.40*** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
14:41.40*** mode/#asterisk [+o anthm] by ChanServ
14:42.26*** join/#asterisk [Outcast] (~bill@c-24-218-94-11.hsd1.ma.comcast.net)
14:42.50slak-hey guys i set up conference but the volume of the callers is too low
14:42.53*** join/#asterisk [Outcast] (~bill@c-24-218-94-11.hsd1.ma.comcast.net)
14:42.56slak-which should i adjust tx or rx
14:43.40Ariel_rx is the recieve tx is transmit
14:44.09slak-ok so both need to be increased?
14:44.46pawallsAriel_, Did you end up having any other suggestions for me from yesterday regarding "#" transfers not working?
14:44.47sangeehow do i check D channel is up or not?
14:44.53*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
14:44.56EthonIs there a way to debug MOH problems? It seems to me that the timing is wrong, moh plays much to fast and sounds like white noise
14:45.12Ariel_pawalls, not unless I can see your complete setup.
14:45.35Ariel_sangee, asterisk will let you know if it's down
14:45.35pifwhen you wake up with a sore ass?
14:45.59*** join/#asterisk astoria (~cluecon@67.107.50.40.ptr.us.xo.net)
14:46.27*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
14:46.31sangeeis there anyway i can check Dchannel status on CLI?
14:46.45astoriasangee: you could do a zap debug
14:47.10sangeeok
14:47.10pawallsAriel_, I'm using the stdexten macro from the sample configuration, apart from adding "tTm" options to the Dial() in there. I turned off loading of "app_parking.so" because I'm using "app_valetparking.so" instead. Then in my dialplan I have _XXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
14:47.51[Outcast]i just had an idea i am must likely not first one with it, How about a sip ATA/ Phone with vpn client built in?
14:47.54pawallsAriel_, The call goes through and operates without a problem, but when I press "#" (or any other DTMF), I get "Urgent handler -- Attempting to do native bridge between SIP/XXX and SIP/YYY"
14:48.17pawallsAriel_, Where XXX and YYY are the caller/callee
14:48.35sangeehow do i do the zap debug (i don't find zap debug in CLI)
14:48.42Ariel_pawalls, sounds like you are using canreinvite=yes
14:48.51pawallsAriel_, I'm not supposed to have that on?
14:48.54Ariel_zap channel debug
14:49.08sangeethx
14:49.15Ariel_not if you want asterisk to handle transfer
14:51.16*** join/#asterisk loick (~loick@APuteaux-151-1-29-202.w82-124.abo.wanadoo.fr)
14:51.31ManxPowerAug  5 09:50:28 WARNING[13965]: channel.c:609 ast_channel_free: PBX may not have been terminated properly on 'SIP/0004f201fb0e-a-e2d4'
14:51.37ManxPowerThere's a message I've never seen before.
14:52.06Ariel_ManxPower, that is a new one on me as well.
14:53.04Ariel_ManxPower, did you post your setting for the t1 to pbx you were working some time ago. I am trying to demo asterisk for a possible clinet and I could not get there t1 with e&m wink to send me dtmf correctly.
14:53.27FaUlwhat does 'No one is avilable to answer at this time' exactly mean? that there is no line connected?
14:53.50jontowFaUl; general protection fault, what else?
14:53.53FaUlor that asterisk was not able to ring via this line?
14:54.00Ariel_FaUl, it means allot of things but that is the first one to think about.
14:54.10ManxPowerAriel_: I don't know if I ever posted those configs.  The E&M system is a very complex T-1 with data, FXO, and E&M channels
14:55.04FaUlAriel_: i've got the problem that the asterisk is connected via hfc-s card to my isdn-line, incomming calls seem to work perfectly, but if i try to call outbound it says the message above
14:55.19Ariel_ManxPower, ok just wondering. I hate the older pbx's  This Meridian is suppose to be a MICS which spec's says has a PRI but it actually has a t1 e&m connection only board.
14:55.44FaUlwhich is quite strange IMHO
14:56.19ManxPowerAriel_: nortel's requires a software upgrade to enable the PRI support.  Also, I don't know if Nortel makes special PRI cards or not.
14:57.06*** join/#asterisk PakiPenguin (uppal@202.147.163.80)
14:57.10PakiPenguinhello everyone
14:57.22Ariel_this is just a demo to see if we can connect to it. So I really can't invest money into it.
14:57.42FaUlif i try to call internal (another hfc-s in NT mode) that works without any problem, also calling from outside to inside works as I said before
14:58.45Ariel_FaUl, I don't use hfc cards I would check to see if your using the correct dial device like capi or something like that.
14:59.04RienzillaI use HFC cards, since 2 days :0
14:59.19FaUlAriel_: should use zaptel via zaphfc
14:59.37rikstaanyone know off the top of their head what  "requested format = 8" is? before i go hunting in the source
14:59.42pawallsAriel_, Same thing.
14:59.51pawallsAriel_, I enabled it for both devices specifically in sip.conf
15:00.02pawallsAriel_, By setting "canreinvite=no"
15:00.08Ariel_pawalls, and you reloaded asterisk
15:00.19pawallsCompletely restarted it.
15:02.04EthonI have serious problems with anonymouses calls at the end of a native bridge..
15:02.20*** part/#asterisk frenzy (~frenzy@193.220.82.108)
15:02.23EthonSetCallerID works on the wrong zap channel...
15:02.29*** join/#asterisk frenzy (~frenzy@193.220.82.108)
15:02.48ManxPowerriksta: show codecs
15:02.52pawallsAriel_, Do I need to have app_parking loaded in order to do blind transfers?
15:02.54rikstadoh, cheers
15:04.46pawalls-- Executing Macro("SIP/116-5037", "stdexten|101|SIP/101") in new stack
15:04.46pawalls-- Executing Dial("SIP/116-5037", "SIP/101|20|mTt") in new stack
15:04.53Ariel_pawalls, I have never not loaded app_parking
15:05.00pawallsAlright, I'll try loading it..
15:06.49*** join/#asterisk RoyK (~roy@142.80-203-78.nextgentel.com)
15:07.04*** join/#asterisk clinthome (~clinthome@snap.helixsystems.com)
15:07.15*** join/#asterisk Maxxed (~max@cpe-70-114-238-9.houston.res.rr.com)
15:07.16pawallsI'm completely out of ideas..
15:07.54ManxPowerpawalls: what is the problem?
15:07.55pawallsI've tried every feasible combination of DTMF methods, I've tried with and without app_parking.
15:07.59Maxxedhey can i tftp xml services onto the phones?
15:08.06pawallsManxPower, When I try to do a call transfer with "#" key, it does not work.
15:08.15ManxPowerpawalls: what phone?
15:08.17Maxxedlike, a company phone list, with out http
15:08.22pawallsManxPower, Example.. I establish a call and press "#801" and it doesn't do anything.
15:08.29pawallsManxPower, one xtenlite and one budgettone
15:08.43ManxPowerpawalls: what codec?
15:08.44pawallsManxPower, I'm trying to get app_valetparking working
15:08.46pawallsManxPower, ulaw
15:09.08pawallsManxPower, I've tried inband, rfc2833, and info DTMF methods.
15:09.09frenzywhere the best place to get a dedicated server from ? (in Europe)
15:09.24ManxPowerpawalls: you want rfc2833 set BOTH on the phone and on Asterisk
15:09.32pawallsManxPower, It is so right now.
15:10.15ManxPowerpawalls: If you do a "sip debug" then everytime the phone sends DTMF to Asteirsk you should see it if the PHONE is correctly sending rfc2833 DTMF
15:10.39pawallsI'll try that.
15:11.33MaarkenI wish xten had a less nasty GUI.
15:11.46ManxPowerYay!  TODAY IS FRIDAY!
15:12.49*** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo)
15:13.02sangeeHow do i findout that T1 card i put it there working?
15:13.55Ariel_sangee, what do you mean what t1 card? to work with what?
15:14.21*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
15:14.39*** join/#asterisk VonBraun (~VonBraun@adsl-065-005-205-049.sip.gnv.bellsouth.net)
15:14.52sangeei install a t1 card and i also put cross over cable with cisco which has t1
15:14.56*** join/#asterisk asteriskmonkey (~phil@Quebec-HSE-ppp3620248.sympatico.ca)
15:15.03sangeebut i don't see D channels
15:15.16asteriskmonkeybrimstone.. any other digium guys here?
15:15.37VonBraunCan anybody point me to documentation on how to setup two TDM cards on the same machine (Debian linux 2.4)
15:15.39sangeehow do i check the T1 card is working?
15:15.49asteriskmonkeyzttool sangee
15:16.08asteriskmonkeyalso if its got a green light at the back .. do a pri debug span 1
15:16.12Ariel_VonBraun, 2 cards just make sure you are not sharing irq's
15:16.18asteriskmonkeyand call yourself youll see data comming accross it
15:16.28sangeejust type zttool?
15:16.32asteriskmonkeyyep
15:16.58asteriskmonkeyzttool on linux cli not asterisk cli
15:17.38sangeei got OK on span 4, what is that mean
15:17.39slak-VonBraun  i have one tdm card on debian
15:17.40slak-2.6
15:17.45asteriskmonkeyanyone know why something wont dial one local number yet allot of others?
15:18.24asteriskmonkeyah span 4 is where your card is allocated so in asterisk do pri intense debug span 4
15:18.39brimstoneasteriskmonkey: you highlighted?
15:18.45asteriskmonkeyand try call yourself if ill is set up youll see crap come up on the screen (you have to do this in the asterisk terminal)
15:18.50asteriskmonkeyyes brimstone
15:19.10asteriskmonkeyi have a weird issue with asterisk that no one from digium has got back to me on.
15:19.15syle2asteriskmonkey: because you don;t know that order is important in your dialplan?
15:19.19brimstonedid you send us an email?
15:19.29asteriskmonkeyyes
15:19.38asteriskmonkeyi set up dial plan for any 10 digit local
15:19.41sangeei got this "asterisk1*CLI> pri intense debug span 4
15:19.41sangeeNo PRI running on span 4
15:20.03syle2your not following its ok
15:20.15ManxPowersangee: then you don't have PRI configured on span 4
15:20.25slak-zhopa
15:20.30brimstoneasteriskmonkey: what was the subject of your email?
15:20.32asteriskmonkeybrimstone,style2 : why the heck is it not working one local number and works on the rest?
15:20.38sangeeyou think it's the protocol setting problem?
15:20.40syle2i just told you
15:20.48asteriskmonkeyah crap dont remember but it will be from admin@massivecomputers.com
15:20.53ManxPowersangee: or you don't know what span does what.
15:21.16sangeewhat is that mean?
15:21.30brimstoneok, let me look
15:21.30ManxPowersangee: Well you have 4 spans.
15:21.35sangeeyes
15:21.43asteriskmonkeysangee: what card do you have and what does your conf file look like in /etc
15:21.46ManxPowermaybe your PRI is on span 1, 2, or 3
15:22.25Ariel_asteriskmonkey, do you see on the CLI the number being sent to the zap channel?
15:22.35*** join/#asterisk pa (~Paolo@pa.user)
15:22.36asteriskmonkeyyep
15:22.47sangeei put this on zaptel.conf
15:22.56sangeespan=1,0,0,esf,b8zs
15:22.56sangeespan=2,0,0,esf,b8zs
15:22.56sangeespan=3,0,0,esf,b8zs
15:22.56sangeespan=4,0,0,esf,b8zs
15:22.56sangeebchan=1-23
15:22.56sangeedchan=24
15:22.58Ariel_asteriskmonkey, I had a problem with some numbers that XO was blocking about 1 year ago
15:22.58sangeebchan=25-47
15:23.00sangeedchan=48
15:23.02sangeebchan=49-71
15:23.04sangeedchan=72
15:23.06sangeebchan=73-95
15:23.08sangeedchan=96
15:23.08ManxPowersangee: DO NOT FLOOD THE CHANNEL!
15:23.13ManxPoweruse pastebin.ca instead.
15:23.14brimstoneasteriskmonkey: did you get a "thanks for emailing us, here's your ticket number" email? if so, /msg me the ticket number
15:23.36ManxPowersangee: So Span 4 isn't even being used.
15:23.52ManxPowersorry, yes it is.
15:24.00sangeei put span 4?
15:24.00ManxPowernow what is the signaling set to?
15:24.01*** join/#asterisk pbxbart (user@p54B02B2E.dip0.t-ipconnect.de)
15:24.14sangeecan you tell me what should i put it in zaptel.conf?
15:24.15*** part/#asterisk pbxbart (user@p54B02B2E.dip0.t-ipconnect.de)
15:24.17VonBraunAriel_ I only see one listing in my /proc/interrupts  for wctdm
15:24.17ManxPowerIt should be pri_cep for most people
15:24.31ManxPower..er... pri_cpe for most people
15:24.35pawallsManxPower, It started working.. and I didn't even change anything.
15:24.43RoyKpri_net only for the cool ones
15:24.47*** join/#asterisk file[laptop] (~file[lapt@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
15:24.51pawallsUgh..
15:24.54Ariel_VonBraun, what server is it?
15:25.03RoyKpri_not
15:25.14VonBraunDebian 2.4
15:25.27Ariel_VonBraun, box???
15:25.35pawallsDoes anyone know if ValetParkCall jumps to a certain priority when there is already a call parked in that slot?
15:25.41RoyKAriel_: it's a gray one
15:25.44*** part/#asterisk the_devil_dont_s (~Administr@62.77.178.121)
15:25.58Ariel_ROFL
15:26.04VonBraunI have no idea what you are asking
15:26.11VonBraunthe hardware?
15:26.46Ariel_VonBraun, yes you put to cards into the system and only one is showing up. Make sure your bios is able to set irq's for different pci bus
15:26.58Ariel_to/two
15:27.21file[laptop]astoria: coffee is gooood
15:27.23Ariel_VonBraun, also make sure it's in a normal pci bus 2.2 complient
15:27.36Ariel_file[laptop], but you don't drink coffee.
15:27.44file[laptop]I know but it's waking people up
15:27.44*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
15:27.44file[laptop]which is good
15:27.59droothfile: SIP is good
15:28.07MaarkenVonBraun: you might also try a different slot for one of the cards, most machines assign IRQ based on PCI slot.
15:28.17*** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
15:28.17*** mode/#asterisk [+o anthm] by ChanServ
15:28.32QwellSIPing coffee is good
15:28.44droothquality is much better on your system than IAX.
15:29.11file[laptop]IAX is evil
15:29.18droothI don't think so
15:29.21file[laptop]it has problems... can't trace 'em down
15:29.27VonBraunAriel, Maarken - Thanks for the info
15:29.36droothI had very few IAX problems in the past
15:29.40Ariel_ok someone from digium knows any good settings for a Nortel Meridian MICS T1 not pri to a digium TE110p card?
15:29.52droothand I had IAX working on other providers.
15:30.09droothfor some reason on your network SIP works better, not surew hy
15:30.12file[laptop]yay for SIP
15:30.21droothIAX is a leaner protocol
15:30.36droothfile: did you get my email
15:30.58h3xasterisk still does better with sip :P
15:31.14file[laptop]drooth: I may have, I'm filtering through email and stuff and trying to wake up
15:31.18droothAfter swtiching to SIP, DTMF is not receiving correctly anymore.  It will recognize 2 digits when you only press one.  Etc.  We tried RFC2833, info, and inband.  The latter two didn't work at all.  So we should use RFC2833, still doesn't work.
15:31.25pawallsIs there a variable in the dialplan that shows the return status of an application?
15:31.30file[laptop]lemme log in and check
15:31.36pawallsLike how ${DIALSTATUS} works for Dial() ?
15:31.47*** join/#asterisk funxion (~nunya@mtnuser.icgws.com)
15:33.23*** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
15:33.37droothit will take SOME digits but not all.
15:33.39anthmno,
15:33.53file[laptop]drooth: yeah DTMF is set for rfc2833 on our end as well
15:34.13droothso how do I solve this issue?
15:34.28file[laptop]I'd need to sit down and look at the rtp debug to see if the timestamps on the DTMF is going funky
15:34.39anthmapps return 0 when nothing happened, and -1 when they want to die, and a rare few return a dtmf digit to queue to the channel
15:34.46file[laptop]which is hard right now because I'm in the symposium listening to Peter Nixon
15:34.56glLoadIdentityehe
15:34.57droothokay
15:35.25droothif you could put it on your priority list, would be great.  IVR is almost unusable in current DTMF state.
15:35.45BentleyDoes anyone know if the subscribe/notify support in * v1-0 is complete enuf to work with Snom phones?
15:35.48h3xi want to shoot the morons that came up with 3 different ways to pass dtmf
15:36.08h3xBentley: It definately dosent work in stable
15:36.15h3xi havent tried head yet
15:36.38Bentleythere is an uncommittted patch in bugs.digium.com that is supposed to make it work in head
15:36.38pawallsWhat does asterisk do in the dialplan if a registered application returns "-1" ?
15:36.41droothfile: one more q, how do I change my failto number for my DIDs?  It should be in the web interface, shouldn't it?
15:36.48anthmcall over
15:36.58anthm< 0 = hangup
15:36.59coppiceh3x: what about the morons that came up with three different signalling protocols?
15:37.06Bentleyh3x: thx tho
15:37.21h3xcoppice: Or you could testify, the crazy fax machine implementations
15:37.55h3xBentley: v1-0 is soon to be ditched in favor of making whats now head, stable
15:38.17coppiceh3x: well that follows the crazy fax spec - you can't actually implement a fax machine from the fax specs. you need to know some folklore too :-)
15:39.21file[laptop]drooth: it will be!
15:39.21mishehuugh.  headache.  :-/
15:39.42*** join/#asterisk anti (russ@anti.developer.gentoo)
15:41.03*** join/#asterisk brookshire (~matt@207.111.174.1)
15:42.36file[laptop]everyone here at Cluecon still alive?
15:43.34astoriame!
15:45.02*** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
15:45.05mishehufile[laptop]: not sure, with this headache.
15:45.33mishehufile[laptop]: though it's still funny that I came from much further and still got here before you guys did with the coffee
15:45.50file[laptop]yup
15:45.56file[laptop]took awhile for them to make it
15:46.51h3xhahahaaah/
15:47.28*** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net)
15:47.30DerkommissarHello
15:47.31file[laptop]the person I was looking forward to talking to has disappeared
15:47.47droothfile: meanwhile how do I change failto?
15:47.54mishehuI wonder how many of the folks here at cluecon have driven by lucent's former hq in naperville
15:47.56file[laptop]you can't yet, we're working on it!
15:47.59droothfile: will you contact me re: DTMF asap?
15:48.09file[laptop]Asterlink is going to be improved heavily
15:48.13droothfile: can't change failto??  hmmm
15:48.13file[laptop]with cool new stuff
15:48.19droothfile: that's not what I was told
15:48.19mishehuto see the all-glass building with the dish that beams information back to the mothership
15:48.23ManxPowerdrooth: file[laptop] is your personal Asterisk Support Bitch?
15:48.26DerkommissarQUESTION, wit cdr_odbc can i make the asterisk write the cdr's to 2 fifferent servers ?
15:48.31astoriai hear naperville is a nice place.
15:48.36mishehufile[laptop]: who did you want to talk to?
15:48.42file[laptop]mishehu: forget his name :(
15:48.53mishehuastoria: nice, crowded, overly expensive, and boring.
15:49.02mishehufile[laptop]: you're too young to be senile
15:49.19file[laptop]pfft
15:49.42mishehuastoria: put it to you this way, when a town has a law against driving around the block 3 times, you know something is up...
15:50.02*** join/#asterisk jeffgus (~jeffgus@2002:d856:c704:0:0:0:0:1)
15:50.13ManxPowerDerkommissar: ForkCDR?
15:50.23anthmthis place has a law against having a beer in both hands
15:50.42mishehuanthm: unless you have voice-command on yoru computer
15:50.58Derkommissarno
15:51.03mishehuor the rms arrangement - an army of interns to type for you
15:52.13mishehufile[laptop]: any word on your luggage?
15:52.14Derkommissarforkcdr is an aplication to write 2 cdrs,,, but still to the same database rigth ?
15:52.37Derkommissari though fork cdr was discontinued on the cvs head :-/ i dunno where i heard about it
15:52.53Derkommissari want the same cdr's to be writen in 2 different databases
15:52.54ManxPowerDerkommissar: what does "show application forkcdr" tell you? What does the Wiki say about it.
15:52.55Derkommissar:-/
15:53.01anthmuse cdr_shell
15:53.17h3xuse ODBC
15:53.33ManxPowerh3x: he is using ODBC
15:53.36Derkommissari use ODBC
15:53.38h3xwell
15:53.47h3xdosent bkw's shiznit let you put in multiple database targets
15:54.05Derkommissari been reading the wiki
15:54.07mishehuDerkommissar: I still see ForkCDR in HEAD, at least from a month ago
15:54.11Derkommissarthere is nothing there about it
15:54.33frenzyhow do I exit CLI without shutting down asteirs :S
15:54.41frenzyasterisk**
15:54.42mishehuI'm running Asterisk CVS-HEAD built by root@rakdanit on a x86_64 running Linux on 2005-07-15 06:41:36 UTC
15:54.45Qwellfrenzy: How did you start asterisk?
15:54.45DerkommissarCauses the Call Data Record to fork an additional cdr record starting from the time of the fork call.
15:54.45Dovid~gui
15:54.46jbot[gui] (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html.  Of course Real Programmers use the command line interface.  See cli
15:54.56anthmI wrote forkcdr what do you need to know?
15:55.08crash3m~cli
15:55.08jbot[cli] a Command Line Interface, the best form of interface around, of course  Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction  Common Language Infrastructure (See mono or .net)
15:55.09frenzyum... asterisk
15:55.11ManxPowerfrenzy: If you start Asteirsk with the -c option you cannot exit asterisk without killing it.
15:55.20ManxPowerIf you don't use -c then you should be able to.
15:55.38frenzyI started simply by entering asterisk
15:55.41Qwellif you connected to a running process with -r, just type exit
15:56.01frenzyoki dokie
15:56.01Derkommissaranthm, forkcdr is not what i need :-/.....
15:56.02frenzythanks
15:56.11anthmyes that is obvious
15:56.11ManxPowerfrenzy: use safe_asterisk and then connect to it using asterisk -r
15:56.25Derkommissarhttp://voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc   says nothing about writing to multiple databases
15:56.28frenzythou shant use safe_asterisk
15:56.47frenzyit gave me erros
15:56.51frenzyerrors**
15:56.56ManxPowerpersonally I use service asterisk start
15:57.03ManxPowerfrenzy: then you need to resolve those errors
15:57.16Derkommissari was thinking about using cdr_odbc and cdr_tds :-) each to a different database.... but the wiki says not to do it
15:57.37DerkommissarSUP Ariel_
15:57.38frenzyAsterisk ended with exit status 1
15:57.38frenzy<PROTECTED>
15:57.43frenzywhat does that mean?
15:57.45Ariel_Derkommissar, how are you doing?
15:57.52frenzyI get that with safe_asterisk
15:57.52ManxPowerfrenzy: that means Asterisk failed to start.
15:57.53Derkommissaryou forked asterisk
15:58.03*** join/#asterisk newmember (~newmember@dsl-lkbn-66-18-211-34-cgy.nucleus.com)
15:58.06ManxPowerlook at the safe_asterisk script to see what might be causing it.
15:58.07Derkommissarran asterisk when it was problably running :-)
15:58.15Derkommissarran safe_asterisk
15:58.27DerkommissarAriel_, getting ready to go to ecuador rigth now.
15:58.37Ariel_Derkommissar, have a safe flight.
15:58.47Derkommissarthanks :-)
15:59.01frenzy<PROTECTED>
15:59.18*** join/#asterisk clinthome (~clinthome@snap.helixsystems.com)
15:59.46anthmif you want to cheat
16:00.21anthmclone the cdr_odbc, and rename the name it registers with and the config file string and call it odbc2
16:00.22frenzyhow does load balancing work?
16:00.27file[laptop]yay working cellphone!!!
16:00.29anthmif you dont want to cheat
16:00.31frenzyjust equally distributes laod?
16:00.34file[laptop]mishehu: I have to go in and fill out forms and stuff
16:00.37*** join/#asterisk fugitivo (~ajf@168-226-244-58.mrse.com.ar)
16:00.39fugitivohello
16:00.49anthmthen you need to start code
16:00.51anthming
16:01.11tzangeranthm: shut up what do you know anyway I bet you've never writtena lick of code in your life.  :-)
16:01.17fugitivotdm400 isn't detecting fax, it detects fax when the call is picked up, not at the first ring
16:01.42frenzyIs this setup possible: a server in US second server in UK when a user connects the two figure out which is the shortest route ?
16:01.47tzangerfugitivo: how the hell is it supposed to know if it's a fax call if it can't hear anything
16:02.14anthmawwww
16:02.15fugitivotzanger: i mean, when the extension picks the call :)
16:02.27coppicetzaanger: temporal defocussing
16:02.29tzangerfugitivo: eh?
16:02.30file[laptop]anthm: did I miss anything fancy?
16:02.41ManxPowerfugitivo: You have to answer the call for Asterisk to hear the fax tone on an analog port
16:02.45frenzyso ?
16:02.45Ariel_fugitivo, wait(3)
16:02.53anthmyah
16:02.57tzangercoppice: ahh you've got res_temporalshit working?  I heard you were ironing out the last few bugs
16:03.04file[laptop]bah, customers come first!
16:03.06frenzybecuase I'm connecting to Asterisk over a satellite connection
16:03.51Ariel_frenzy, dundi
16:03.52frenzyand some satellites connect to the net via Europe... other via US/CA
16:04.10fugitivoAriel_: s,1,Wait(3) and then fax,1,Dial() ?
16:04.44frenzyAriel_: I got to dundi's site... but I didnt get the gist of what it does
16:04.53frenzy< few hrs back >
16:04.54Ariel_fugitivo, s,1,answer s,2,wait(3) s,3,what's next.   in the same context you then have fax,1,Blah
16:05.03fugitivoAriel_: thanks
16:05.34Ariel_fugitivo, use the power of a macro
16:06.17Ariel_frenzy, dundi lets you connect enterprise servers so that you can publish routes on them for numbers from each other then asterisk sends the call directly to that server.
16:08.08*** join/#asterisk jpcarvalho (Jeff@201.30.193.135)
16:09.31jpcarvalhoHello all , i'm in troubles with G729 (open/readytechnlogies) running on FreeBSD 5.4. It works great on Linux ... but in FreeBSD i have no audio. The interesting is that in 'show translation' it's ok.
16:09.35jpcarvalhoAny ideas?
16:10.12ManxPowerjpcarvalho: how many G729 licenses did you purchase?
16:10.17jpcarvalhonone
16:10.31jpcarvalhoi'm using the open g729 as i use in my linux box
16:10.35ManxPowerjpcarvalho: I can't help you with illegal software.
16:10.49jpcarvalhook.
16:11.04ManxPowerand neither will any one else that I know of.
16:11.22*** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com)
16:11.25jpcarvalhook , no problem
16:11.32mutem
16:11.33jpcarvalhoI'll find a way to solve this.
16:11.36muthow is that illegal?
16:11.48mutand jpcarvalho.. why not just buy a license?
16:11.51mutthey's cheap
16:12.25jpcarvalhoDigium's License Application runs only in Linux systems
16:12.32jpcarvalhoI'm planning to use BSD
16:12.51jpcarvalhoSo , i don't have time to wait digium ... release a version for BSD
16:13.01jpcarvalhoI need this working , now.
16:13.08mutheh
16:13.15mutsuch conviction
16:13.40jpcarvalhoAnd if is illegal , let's justice discuss this later.
16:13.57ManxPowerSo ftp://pub/telephony/asterisk/g729/unsupported/freebsd-5.2.1 doesn't work?
16:14.19jpcarvalhoWorks , when i run show translation... is perfect
16:14.38jpcarvalhobut when i client requests the codec .. says that invalid codec
16:15.21*** part/#asterisk FaUl (~immo@hobbynuttenverzeichniss.de)
16:15.27jpcarvalhoAug  5 13:15:10 NOTICE[622]: chan_sip.c:2795 process_sdp: No compatible codecs!
16:15.48jpcarvalhobut on show translation is there with a 3ms time
16:16.04jpcarvalhoZaptel/ Libpri is working perfetc
16:16.06jpcarvalhoperfect
16:16.17jpcarvalhomy only problem is G729 and G723
16:16.52jpcarvalhoI use Linux a couple of months running Asterisk .. but i'm sure that FreeBSD is better
16:16.59jpcarvalhoI'm feeling this
16:17.10jpcarvalhoI was afraid about LibPri/Zaptel
16:17.11mishehuactually, DOS 3.11 is better.
16:17.18Beirdoif you need it working NOW, use Linux
16:17.20jpcarvalhobut is woring
16:17.37blitzrageyer feeling that FreeBSD is better for Asterisk even though its developed for Linux? That seems backwords to me.
16:18.01blitzrageRun it on *BSD if you want, but certain things are just not going to work
16:18.06drumkillablitzrage: t3h g1bs0n runs FreeBSD!
16:18.09coppiceno. it seems like the thoughts of a fanatic
16:18.28blitzragedrumkilla: what does t3h g1bs0n know? :)
16:18.34drumkillablitzrage: that won't make sense if you haven't seen Hackers :)
16:18.45blitzragedrumkilla: I have - I choose to ignore that fact :)
16:19.00anthmat least, get it to work completely on linux first =D
16:19.02jpcarvalhoI just need to know ! - Someone is running G729/G723 on BSD?
16:19.11jpcarvalhoIs running on Linux..
16:19.14jpcarvalhoperfectly
16:19.20mishehuI'm running it on a ti-99/4a.
16:19.24jpcarvalhoI have 3 servers running on Linux
16:19.25BeirdoThen why break it?
16:19.30blitzrageI want to run it on my SL-5500 :)
16:19.31jpcarvalhobut my experience is on BSD
16:19.38jpcarvalhoI love BSD
16:19.39*** join/#asterisk Abbas (Abbas@203.81.223.69)
16:19.42*** join/#asterisk cp5 (~samy@dsl093-032-201.snd1.dsl.speakeasy.net)
16:19.45Abbashi all
16:19.48jpcarvalhoand i need running on BSD
16:19.50cp5has anyone ever had polycom phones "lock up" on them?
16:19.55*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
16:19.56jpcarvalhois not so far for me Guys
16:19.59Beirdowell, good luck, jpcarvalho
16:20.02Ariel_don't you love this but I want it to work on this os not that one that works.
16:20.05jpcarvalhojust two codecsd.
16:20.06anthmnah, I don't me get your perception of "working" i mean working as in nothing left to do but decorate it
16:20.25jpcarvalhoThe most difficult was done..
16:20.28Abbasanthm:   can we listen the SIP calls?
16:20.29blitzragedrumkilla: holy crap - you patched that QUICK :)
16:20.30mishehucp5: there is a known issue with certain firmwares.  some people schedule automatic reboots
16:20.34jpcarvalhoI have two TE-405P working
16:20.37jpcarvalhoon FreeBSD
16:20.54anthmabbas , whatcha mean?
16:21.02jpcarvalhostop this .. only because i'm having a little problem with codecs!
16:21.03mishehuanthm: hmm...  and how do you decorate your systems?
16:21.07drumkillablitzrage: ha
16:21.09cp5mishehu, i can't find any pages on this, do you know where i can find moer info?
16:21.09azrishahrilanyone with oh323 experience, care to share experience ?
16:21.31mishehucp5: you looking on voip-info?  I think that's where I saw the information about it.
16:21.36cp5k
16:21.37jarrodhey if my cisco is dialing <digits>@<asterisk server ip> and there is a sip user configured for the sourcing cisco machine should that route those digits to the proper context?
16:21.44Beirdojpcarvalho: I wish you luck.
16:21.48jpcarvalhoThanks
16:21.58jpcarvalhoI promisse give a feedback here
16:22.05Abbasanthm: i mean    can we barge the SIP calls same like we do the calls going through ZAP
16:22.07anthmhmm the ppl with questions seem to outnumber the ppl with answers today
16:22.08jpcarvalhoAnd who knows BSD ..
16:22.20jpcarvalhoIs easy to see why i need it !
16:22.21jpcarvalho:)
16:22.27anthmabbas chanspy you mean ?
16:22.27blitzrageWhy is the sky blue?
16:22.33Abbasyah
16:22.37mishehujpcarvalho: my OS is gooder than your OS!
16:22.42anthmyah you can
16:22.58Abbasactually we want to listen the active SIP  or  IAX  calls
16:23.06Abbascan this work on    SIP and IAX  both ?
16:23.13ManxPowerblitzrage: http://www.sciencemadesimple.com/sky_blue.html
16:23.18blitzrageManxPower: LOL
16:23.24anthmyou guys want some answers too bad you cant be here listening to nix he's got a whole bunch
16:23.51mutmishehu: it's gooderer
16:24.00mishehumut: no no, more better!
16:24.19mutbetterer
16:24.30mutthe more er's ya add the more betterer it is
16:24.41wunderkinthe besterest!
16:24.48file[laptop]Nixon's doing great
16:24.48wunderkinthe most besterest!
16:24.52mishehuanthm: he says a lot of good stuff, but I don't agree with everything he says.
16:25.07mishehufile[laptop]: it's illegal to dance here
16:25.13Abbasanthm:   can we use chanspy  for both  SIP and IAX?
16:25.14file[laptop]mishehu: I make the rules :P
16:25.18anthmthat's the point =D he's only presenting what he does not telling you to do it too =D
16:25.19*** join/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
16:25.28mishehufile[laptop]: be sure to tell it to the judge ;-)
16:25.35file[laptop]pfft
16:25.40jarrodblargh.. anyone using an external cisco machine to forward inbound calls from a PRI to an asterisk machine?
16:25.41QbYwhat would make calls between extensions be horrible, but to the outside world they are fine..?
16:25.42wunderkintell that to judge mathis
16:25.49wunderkinwhile you are over there
16:26.25anthmthe ppl to pay the most attn to are those who don't care if you agree with them or not. ;)
16:26.34mishehuanthm: well when he was talking db engines, he was more or less telling ppl what to do
16:27.00anthmya think?
16:27.03mutthats the best way to do it, geeks don't listen to other geeks
16:27.07anthmi didnt get that impression
16:27.36anthmI told him 2 nigts ago how i do cdr that was day and night and he thought it was cool
16:27.49mishehuanthm: about the only thing that I caught that mysql actually does lack is the subtransactions (at least when you take mysql 5.0.x branch into consideration).  however, mysql maxdb does support them.  and you can cluster mysql servers too...
16:27.51*** join/#asterisk riksta (~rick@84.93.243.170.broadband.plus.dyn.plus.net)
16:27.54file[laptop]I luv our system
16:28.00file[laptop]but it requires patience
16:28.24anthmi use mysql but dont trust it
16:28.30mishehuSo most telcos use radius but not OSP for billing?
16:28.39mishehuanthm: what ver?
16:28.39anthmso every morning i erase the whole db and make a new one
16:28.53blitzrageI like pgsql
16:28.57anthmthen i use the "kinda" cluster to keep it synced
16:29.11anthmthen every morn i crush them and start over
16:29.27Abbasanthm:   can we use chanspy  for both  SIP and IAX?
16:29.31mishehuanthm: donno, pointer and I use a lot of mysql...
16:29.32anthmyes ?
16:29.35blitzrageI'm not a DB admin, I like it because its easy to user.
16:29.39blitzrageuse*
16:29.57anthmboth mysql and pg have done things to piss me off
16:30.02droothfile: any update on dtmf?
16:30.05anthmbut it was mostly in late 90's
16:30.12anthmand i hold grudges
16:30.18tzangermysql sucks goat dick
16:30.26mishehuanthm: computers have done many things to piss me off.  95% of that is from microsoft actually
16:30.27mishehuheh
16:30.29blitzragelucky goat
16:30.32tzangerhaha
16:30.35anthm*grin*
16:30.38tzangernah, mysql uses its teeth when it does it
16:30.41astoriaI don't have any problems with mysql, but i can't figure out their licensing module...
16:30.45file[laptop]drooth: I'm not a miracle worker.
16:30.46astoriamodule->model
16:30.52file[laptop]I can't be in two places at once
16:30.53Qwellastoria: its easy
16:30.54droothfile ?
16:31.02droothfile: okay, just let me know
16:31.09tzangerastoria: use it as a real DB and you'll quickly find issues with it
16:31.13astoriaQwell: it is? i don't know when it's okay to use for free and when i have to pay..
16:31.14Qwellastoria: If you use it for personal use, its free.  If you use it for an open source product, its free.  if you use it in a corp, its not free.
16:31.20file[laptop]would be cool if I could be...
16:31.22astoriatzanger: i use it as a real DB all the time with no problems.
16:31.24mishehuah look at what I started
16:31.28blitzragetzanger: do you know if you can transfer cell numbers in Ontario yet?
16:31.34tzangermysql works as a place to store shit and get it back reasonably well.  but it's a pitiful failure as a RDBMS.
16:31.35mishehunow it's "my rdbms is gooder than yours"
16:31.44tzangerblitzrage: nope, not until 2006 at the earliest
16:31.46astoriaQwell: so if i'm installing this with * to my clients, do i have to pay??
16:31.48anthmtypically it's for sure a decent embedded db for a single pj
16:31.55blitzragetzanger: shitty - guess I gotta keep my cell phone until then :)
16:31.57*** part/#asterisk zoo (nobody@ip-46-16.travedsl.de)
16:31.58anthmlike a mantis type thing
16:32.00Qwellastoria: Only if you pay for asterisk, I'd think
16:32.01tzangerif you're using it as an embedded db, use sqlite
16:32.04blitzragetzanger: can't lose my wicked cell number :)
16:32.05anthmwhen you dont want to pay
16:32.16anthmthat is a far as i trust it tho
16:32.19tzangerblitzrage: what's your wicked cell #
16:32.35blitzragetzanger: 519 591 5119
16:32.37anthmasterisk nees sqlite but nobody will agree
16:32.38astoriaI use mysql for my healthcare database software with no problems at all...
16:32.39tzangerhahahaha
16:32.42Qwellhmm, that is pretty wicked
16:32.44anthmi have it sitting there for a year now
16:32.50tzangerthat's like the license plate pair I've seen... MWMWMWM and WMWMWMW
16:32.51anthmshrug
16:33.00*** join/#asterisk bprice20 (~bprice20@Unassigned-216.120.255.29.hrwebservices.net)
16:33.03blitzrageMMW! (Medeski, Martin, and Wood)
16:33.05QwellI had gotten 800-4-LA-Times from Asterlink, but somebody stole it away from them. :(
16:33.19tzangerdamn
16:33.49Qwelltzanger: my wifes friend bought a car in CA with the plate "abcdefg".  The lady who sold him the car said he couldn't keep it though
16:34.05tzangerhahaha
16:34.23tzangerI want a license plate for a VW Beetle to say "FEATURE"
16:34.38mishehuif I was a cop, I'd pull over anybody with a place of 'abcdefg' whenever I could.
16:34.52mishehutzanger: ichs bug.
16:34.57Qwellmishehu: This guy would have given you plenty of reason :p
16:35.26tzangermishehu: "it's not a bug, it's a feature"
16:35.44tzangerI also wanted "31337" for a plate but they won't let me get it in Ontario (too many consecutive #s)
16:35.48droothfile: i was getting a busy signal into my IVR all morning, now it's gone.  our system doesn't produce a busy signal, so it must be something on your end.  we didnt make any config changes.
16:35.57Ariel_I am off to a customer site to pull some cards out of some stupid meridian systems. See you all later.
16:36.01*** part/#asterisk frenzy (~frenzy@193.220.82.108)
16:36.04droothfile: can you look into that as well?
16:36.31mishehutzanger: heh, kinda slow this morning still.  headache is just starting to go away
16:36.34blitzragetzanger: that's w34k
16:36.40droothfile: also I am on a call just now and it dropped.  can we troubleshoot?
16:37.39jarrodanyone using cisco voice card for PRI service then forwarding to * via sip?
16:37.44tzangerdrooth: hard to say.  what's the CLI say
16:37.54droothtzanger: checking
16:37.54tzangerblitzrage: 347 M3
16:38.02droothwow...lost another call
16:38.14*** join/#asterisk santiago (~santiago@63.245.86.172)
16:38.50blitzragetzanger: I don't get it :)
16:39.03blitzragetzanger: nevermind - just slow
16:39.15ManxPowerdrooth: Do you have busydetect=yes or callprogress=yes?
16:39.30droothno
16:39.31mishehutzanger: if you had a plate like that, I'd pull you over even more often than the person with a plate of 'abcdefg'
16:39.46tzangerhahha
16:39.50tzangermishehu: good thing you're not a cop then
16:39.52file[laptop]ManxPower: he's on VoIP
16:39.52drooththat applies when you have a regular phone line connected.
16:40.22droothyeah just lost 2 incoming calls.
16:40.23droothsigh.
16:40.30droothi dont think SIP is the answer
16:40.33tzangerwhat kind of connection are you talking about then drooth
16:40.52mishehutzanger: maybe I should become one so that I stop getting pulled over for bullshit...  like being accused of 'driving agressively' because I switched a lane AND used my turn signal AND checked my blind spot!
16:41.11*** join/#asterisk glm2k (~GLM@rrcs-24-199-11-45.west.biz.rr.com)
16:41.15tzangermishehu: clearly that's agressive
16:41.20tzangerjust switch without signalling or checking
16:41.27tzangerchrist do kids not know how to drive these days??
16:41.33astoriai never use my blinkers.. thats giving away your strategy.
16:41.44mishehutzanger: it's not just the kids.  it's 90% of the people out here.
16:41.47tzanger:-)
16:41.48Qwellnever use a blinker if you're going to make an illegal turn
16:41.54Qwellif you do, it becomes premeditated
16:42.04tzangerpremeditated illegal turn?  wtf?
16:42.06mishehuI can count the number of people who even use their signals prior to lane change on one hand...  and that's within a week.
16:42.27QwellIf you use a blinker, you can't say it was an accident...you lose alot of options
16:42.36mishehuastoria: remind me to shoot you after nix's presentation
16:42.41blitzrageanyone know where I can get a cheap power meter to monitor how much power I'm using from an outlet?
16:42.52Qwellblitzrage: I saw one on thinkgeek
16:42.54QwellI think
16:42.56droothfile: if you are busy are there any other techs who can help me?
16:43.15blitzrageQwell: thanks, will go look - need to monitor the amount of power I'm going to be using in my new lba
16:43.18blitzragelab*
16:43.39tzangerQwell: what the hell kinds of turns do you take that you need an excuse like that for
16:43.41Qwellblitzrage: http://www.thinkgeek.com/gadgets/electronic/7657/
16:43.48Qwelltzanger: don't ask
16:43.56tzangerblitzrage: rat shack had them for a while
16:44.05blitzragetzanger: hrmmm, will check their site too
16:44.07tzangerI have some decent equipment but it's not exactly cheap
16:44.55*** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net)
16:45.14tzangermind you that's 3 phase and can monitor unattended for 12 months but still.  :-)
16:45.20blitzragetzanger: yah... this is just for home so I can show my room mates I'm tracking my power usage in the lab :)
16:45.22docelm0Say does anyone the rough amount of minutes you can shove thru a PRI in a month?
16:45.32tzangerdocelm0: well look at it this way
16:45.35tzangerT1 PRI is 23 channels
16:45.40docelm0yes
16:45.46blitzragesimple math could figure it out....
16:45.48Qwellabout 993,000
16:46.03blitzragemins per month * 23
16:46.04docelm0ok good..
16:46.19tzangerthere are there are 1440 minutes per day
16:46.22docelm0I need to order 2 for Toll free origination
16:46.46tzanger43830 minutes per month
16:46.50tzangerper channel
16:47.02tzangerso just a shade over a million
16:47.17blitzrage1026720 @ 31 days
16:47.51docelm0ok was just curous..
16:47.56jpcarvalhoSomeone needs VERY GOOD termination in BRAZIL ?
16:48.20docelm0What kinda rate are you looking for?   Whats the code?
16:48.21tzangerblitzrage: I did 1440 * 365.25 / 12
16:49.09docelm0I am building a A-Z termination/origination provider now..  Should be only maybe this weekend..
16:49.35*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
16:49.53blitzragetzanger: I did 1440 * 31 * 23, but your formula is definately more accurate :)
16:50.14blitzragetzanger: well, more of an average per month, but whatever :)
16:50.27docelm0I just need a rough number..
16:50.28docelm0:)
16:50.31tzangerit's all good.  rule of thumb.  T1 PRI = mil min/mo
16:50.44file[laptop]drooth: someone restarted the switch
16:50.47blitzrageQwell: that thing is pretty cool on thinkgeek - only $30 too
16:50.51Qwellyep
16:50.54file[laptop]anthm: was it... you!
16:51.03anthmno?
16:51.29blitzragesounds suspicious
16:53.55mishehuthese pics that nix is putting up on the display make me want to shout "AZIZ!   LIGHT!"
16:54.08mishehuthat and "where's the camels?"
16:55.00mishehuI'd not go to Iraq unless both my team and I were permitted to carry assault rifles and enough ammo.
16:55.22*** join/#asterisk bprice20 (~bprice20@Unassigned-216.120.255.29.hrwebservices.net)
16:55.27*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
16:55.55astoriaHe's got balls going to Iraq.
16:56.50file[laptop]bad agents messaging me trying to get me to do their job
16:57.12glm2ki smell a rat on my router.
16:57.17mishehuastoria: yeah, for not being military trained
16:57.18twisted[asteria]hah
16:57.37mishehuI am military trained and I know better than go to Iraq ;-)
16:57.37glm2ki switched to iax2 on fwd and i call me still fails.
16:57.44drumkillahm, I thought this was #asterisk
16:57.57Qwelldrumkilla: you were mistaken
16:58.04mishehu(also doesn't help if you're enemy #1...  even above americans on that list)
16:58.25blitzrageI'm glad I'm not in the army
16:58.33drumkillai'm pretty sure I said I thought this was #asterisk
16:58.33mishehudrumkilla: we are talking about *, as in "everything"
16:58.48blitzragest00pud!
16:58.50*** join/#asterisk chendy (~alex@218.1.209.242)
16:58.55chendyhi there
16:59.03drooth<file[laptop]> drooth: someone restarted the switch --is that why it was busy?
16:59.04twisted[asteria]c'mon guys, let's move to more related topics
16:59.18file[laptop]possibly, I'm trying to track down who did it
16:59.33file[laptop]we only do that if something drastic has happened, and it was all working fine
16:59.58droothwell, i have 3 issues so far to deal with
17:00.05blitzrageguh.. I hate buying stuff from the US... costs so much to ship it (plus taxes and duty across the border)
17:00.11droothdtmf, busy, dropped calls
17:00.13file[laptop]yes they're on a list in Word on my Powerbook
17:00.18blitzragea $30 peice of equipment will probably cost me $70
17:00.20Luke-JrIs there a simple way to say "for the first 400 minutes, use this macro; after that, use another one"?
17:00.27Qwellblitzrage: find the make/model, see if you can find it local?
17:00.59blitzrageQwell: yah, not sure where I'm going to find that power meter you sent me a link to
17:01.24blitzrageQwell: ahhh... made by P3... getting closer ;)
17:01.28ManxPower"There Are No Stupid Questions, But There Are A Lot Of Inquisitive Idiots."
17:03.59mishehuand make something foolproof, and there will be a new type of idiot that appears
17:04.57*** join/#asterisk coppice (~chatzilla@82.199.17.210.dyn.pacific.net.hk)
17:06.27astoriathis is the kind of debate I came to cluecon for! :)
17:06.58file[laptop]astoria: hehe
17:07.22mishehuoh wait, you're not participating in the debate
17:07.40blitzrageQwell: cool, found a place in Canada to get it at least ($60 CAD... but no duty at least)
17:07.46astoriai like to hear people debate this kind of stuff.. where's jerjer when we need him ;)
17:07.53Qwellblitzrage: little better
17:08.05blitzrageQwell: yah... shipping from Toronto at least
17:08.17mishehuastoria: boycotting us because cluecon is a bunch of shameless self-pluggers?
17:08.28astoriaha ha
17:08.29mishehuor something to that effect
17:09.24file[laptop]I'm hungry myself
17:09.36mishehuyeah, it's about lunchtime no?
17:09.46astoriawhat ever happened to digiums DS3 card? Are they still doing that?
17:09.46mishehuwe must be running behind a bit.
17:09.53file[laptop]it was lunch about... 40 minutes ago
17:10.11mishehufile[laptop]: well, if you like to stop the debate...
17:10.25file[laptop]meh
17:10.43mishehubrian's behind me, I could tell him you said to call out the lunch wagons
17:11.53jarrodhey on a fax with t38 enabled on both endpoints.. the sip proxy/sbc doesnt have to be t38 compliant doe sit?
17:12.15coppicejarrod: yes it does
17:12.27mutHAHA
17:12.31muti've been caught!
17:13.00jarrodi hear that ser can pass t.38 calls tho
17:13.15coppiceso it is t.38 compliant
17:13.17JerJerser doesn't care
17:13.23JerJerits just messages
17:13.44jarrodit 'proxies' the rtp tho, correct?
17:14.01jarrodbecause my sip 'users' are behind nat
17:14.16mishehualright, lunchtime!
17:14.51Luke-JrIs there a simple way to say "for the first 400 minutes, use this macro; after that, use another one"?
17:15.03coppiceser needs to understand the SDP to handle t.38
17:15.25JerJerser doesn't care
17:15.31JerJerit just processes messages
17:15.38JerJerit does NOTHING with the media stream
17:15.52JerJeror does it get involved with the processing in any way
17:16.01jarrodi need to load a newer IOS then to be able to send sip uri messages instead of just digits@<sip server>
17:16.07JerJerit simply processes the udp port 5060 traffic as configured
17:16.15coppiceJerJer: doesn't it even check the SDP?
17:16.32JerJeryou can check it for NAT traversal
17:18.18coppicewell, lots of SIP things will throw out SDP they don't understand, and those would need to recognise T.38. Maybe SER just passes any old crap through
17:18.32jarrodi believe that is the case
17:18.39coppicedealing with t.38 in the SDP is pretty trivial, though
17:18.59*** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com)
17:19.27DrWho17Is there anyway to get MWI to work with realtime SIP and realtime Voicemail?
17:21.02*** join/#asterisk jeffgus (~jeffgus@2002:d856:c704:0:0:0:0:1)
17:22.05Qwellrealtime voicemail or odbc voicemail?
17:23.01Qwellerm, odbc storage (which is just a fancier name for voicemail)
17:23.38bprice20DrWho17 I have it working w/ realtime voicemail using mysql
17:23.44*** join/#asterisk used (~used@c-24-21-40-68.hsd1.or.comcast.net)
17:23.50*** join/#asterisk Darwin35 (~kvirc@ip70-179-215-116.dl.dl.cox.net)
17:23.50bprice20so it DOES work
17:24.06QwellWith odbc storage, the patch on bug 4403 has been extended to work with MWI
17:24.11bprice20you may need to add rtcachefriends=yes to your sip.conf
17:24.18Darwin35ok this bites
17:24.40*** part/#asterisk bprice20 (~bprice20@Unassigned-216.120.255.29.hrwebservices.net)
17:24.43Inv_arpQwell: you setup asterisk for your job?  (u mentioned ure heading to astricon)
17:24.49QwellInv_arp: nope
17:24.59*** join/#asterisk bprice20 (~bprice20@Unassigned-216.120.255.29.hrwebservices.net)
17:25.53DrWho17bprice20: nice
17:26.00DrWho17bprice20: I have that set
17:26.30bprice20DrWho17 is it working for you now?
17:26.32DrWho17I don't get a stutter tone, when using the realtime voicemail backend, I do get it when using the flat file
17:26.43QwellDrWho17: see what I said :p
17:26.56QwellWith odbc storage, the patch on bug 4403 has been extended to work with MWI
17:28.23DrWho17yea, not using odbc, but this may help
17:28.38Qwelloh, the flat config file...
17:28.53DrWho17no, mysql
17:29.00DrWho17mysql realtime
17:29.33DrWho17notification works fine with the mailboxes in a flat file, and sip users realtime
17:29.50QwellI doubt that patch will help you then
17:30.03mutsay cheese
17:30.04muthttp://www.nasa.gov/images/content/124415main_image_feature_380a_ys_full.jpg
17:30.59DrWho17right, I already have the context and mailbox in my schema
17:31.11DrWho17for sip users
17:33.16bprice20Damn vonage is getting peering agreements w/ telcos all over the world
17:33.34bprice20oops wrong window but it applies here as well
17:33.57fanguini use the command "SAY NUMBER 2 *" in an agi script and the digit "two" is said correctly. But after the digit is said i always get this two warnings: "ast_openstream: File  does not exist in any format" and "ast_streamfile: Unable to open  (format ulaw): No such file or directory"
17:35.01*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
17:35.03fanguindoes anybody know what asterisk trys to say there?
17:35.29fanguinor which file asterisk expects?
17:35.30*** join/#asterisk Broom (Broom@jescobar.ayustar.net)
17:35.55*** join/#asterisk ai-a (~gandalfii@81.168.0.204)
17:36.26Broomhello all, i have a question, i have an asterisk pbx up and running but my boss (he said he ordered correctly) ordered a tdm40B digium card which is 4 port fxs, is there a way i can receive incoming calls from my telephone company in that card? like a converter or something
17:36.50ai-acan someone point me to a qucik tutorial of adding a single user to asterisk so i can connect over iax?
17:39.33MaarkenBroom: only thing I've seen is this, and I have no idea if it works or not.
17:39.34Maarkenhttp://worldcall.brinkster.net/pcphoneline/fxsfxo.htm
17:40.35Broomok, thanks a lot
17:41.28*** join/#asterisk pasifico (Santon@adsl-200-136.tricom.net)
17:43.12pasificohello I am new in asterisk and I do not write ingle well qui can help me (hola soy nuevo en asterisk y no escribo bien el ingle qui me puede ayudar) ?
17:43.57*** join/#asterisk santiago (~santiago@63.245.86.172)
17:44.23Broompasifico: dime
17:45.20pasificoBroom: hola yo quiero conectar via voip mi casa con mi trabajo como lo aria
17:46.04Broompasifico: bueno, tienes que instalar asterisk primero que nada, luego tendrias que crear extensiones SIP para poder utilizar IP Phones ya sea en tu casa como en tu trabajo
17:46.12tzangeryo quiero taco bell!
17:46.26mutquesedilla!
17:46.29tzangerhahhaa
17:46.44tzangerI had a gf that prounced that workd "kwizzy-deelia"
17:46.50pasificoBroom: bueno yo tengo asterisk@home
17:46.57muthaha
17:47.29Broompasifico: eso tiene un webinterface
17:47.53pasificoBroom: ya instalado y q tarjeta tendria q comprar
17:48.04pasificosip
17:48.11Broompasifico: ninguna
17:48.21Broompasifico: las llamadas serian por VoIP
17:48.34blitzrageSIP question: anyone know if SIP is capable of sending a 302 Moved Temporarily message to an end point which contains multiple forwards for the same number?
17:49.15pasificoBroom: ok pero me abian hablado de que tendria q comprar una fxo fxs para mi coneccion
17:49.21pasificoh
17:50.03Broompasifico: la fxo es para conectar una linea que venga de la calle para hacer llamadas desde el PBX o recibir a través de es enúmero
17:50.19Broompasifico: fxs es para conectar telefonos analogos (regulares) al cuadro asterisk
17:50.59Broompasifico: ya que tu vas a conectar tu trabajo a tu casa puedes hacerlo por VoIP
17:51.07muti don't like where this conversation is headed
17:51.23pasificoBroom: ok y como lo aria
17:51.41Broompasifico: en realidad debes leerte el manual: pasifico: puedes tratar este link: http://asteriskathome.sourceforge.net/handbook/index.html
17:51.50Broomlo explica muy bien
17:52.12glm2kmut: why not? Broom is doing a good job.
17:52.23muthow am i supposed to know that
17:52.23muti
17:52.31glm2kmut: oh. i see.
17:54.09Broomhe was asking how he could connect his office to his house utilizing VoIP, since someone told him that he needed to put an fxo/fxs card for it to work.
17:54.12fanguini guess it is important that my problem only occures with german digits.
17:54.17glm2kmut: basically, pasifico wanted to know how he could connect his house and his office. and as a newbie, broom, just gave him a fxo/fxs rundown
17:54.26pasificoBroom: ok pero y con las tarjeta fxs fxo q tengo ya yo podria conectar via mi telefono analogo y resibir las llamada por esa via por menos costo
17:54.40Broompasifico: si
17:55.05glm2kand he just agreed that it would cost less :)
17:55.26Broompasifico: si el costo de internet es muy alto, puedes recibir la llamada a través de la fxo al pbx y conectar un tel analogo a la fxs
17:55.28mutthx
17:55.53pasificoBroom: tan solo necesitaria eso para la coneccion  fxs al telefono
17:55.57syle2hey pasifico you got some hot women in columbia land
17:56.13glm2kmut: hehe, we're all "cheap", with a lot of time to spend configuring and not a lot of money to speed things up >:)
17:57.51pasificoBroom: pero dime una cosa cuando la conecte tendria q tener una pc de un lado y otra del otro como la vpn
17:58.07*** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
17:58.19Broompasifico: si lo haces con fxo/fxs no
17:58.21*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
17:58.53Broompasifico: porq en tu trabajo utilizarias el telefono regular para llamar al num del fxo y lo recibirias en el tel de tu casa a través del fxs
17:59.23JerJerwhoa did i just like transported to asterisk-espanol ?
17:59.24Broompasifico: a menos que configures una extensión SIP, que puedes utilizar un IP Phone en tu trabajo (ya sea fisico o xlite) y puedes marcar la extensión de tu casa
17:59.31glm2kJerJer: hehe
18:00.04pasificoBroom: pero en mi trabajo tiene telefono analogo y en mi casa
18:00.57Broompasifico: no creo que estes entendiendo, la manera mas rápida de entender esto es si te lees el manual, por lo menos la parte de introducción y eso para que entiendas como trabaja asterisk
18:01.50pasificoBroom: pues entonc yo conectaria la linea telefonica por el puerto de la fxo y el telefono por la fxs el telefono no es como lo aria
18:02.09*** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net)
18:02.14Broompasifico: si, en terminos generales
18:02.18focksanyone using 7914's with 7960s?
18:03.08pasificoBroom: pues la verdad es que si y escusame que te moleste pero es q necesito la coneccion
18:04.06Broompasifico: no te preocupes
18:04.06*** join/#asterisk pabelanger (~a@67.71.252.98)
18:04.11pabelangerg'day all
18:04.40pasificoBroom: pero luego de esa coneccion la conetaria de una sucursal a otra y mi jefe me dejaria migral todo a linux
18:05.15pasificopues tansolo me a dejado migral los server y solo una pequeno %
18:05.24pasificoBroom: pues tansolo me a dejado migral los server y solo una pequeno %
18:05.38*** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET)
18:05.45Broompasifico: aqui estan tratando asterisk para ver como trabaja y migrarlo completo
18:07.01brookshireis http://beta.digium.com slow for anyone else?
18:07.16drumkillabrookshire: your mom is slow
18:07.23Qwellbrookshire: how about not resolving?
18:07.24brookshirehush.. i'm serious
18:07.28brookshirehmm
18:07.30pasificoBroom: eso le propuc a mi jefe y medio muy buena vista
18:07.30JerJerhmm i get a dns error
18:07.37focksanyone been able to program a Cisco 7914 with line appearances?
18:07.41drumkillabeta.asterisk.org
18:07.43brookshireoh yeah
18:07.43focksthe sidecar
18:07.45drumkillanot digium.com
18:07.48brookshirehttp://beta.asterisk.org
18:07.49brookshireLOL
18:07.52brookshireMY BAD
18:08.38*** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net)
18:08.47JerJerthe initial load took about 30 seconds, then it was very fast to load the other pages (about, features, support, etc)
18:08.50harryvvArial, you on?
18:08.59brookshireyeah..
18:09.07hardwiremy boobies hurt
18:09.09brookshirewell the images would be preloaded then
18:09.18hardwireanybody heard anything on the ipvolution cards?
18:09.25JerJerperhaps a database connection it not getting cached?
18:09.39brookshireno.. it's a problem with the images
18:09.44JerJeroh ok
18:09.47twisted[asteria]brookshire, FIX IT!
18:09.52brookshirei'm working on it
18:09.53drumkillabrookshire: NOW
18:09.53harryvvNorth Island NAS is making news on cnn..I was there in 1999
18:10.02brookshirei think bandwidth.com has bandwidth issues :)
18:10.07twisted[asteria]your mom has bandwidth issues
18:10.09harryvvhi brookshire
18:10.41brookshireyou still can ;)
18:11.28drumkillaanyone have any experience installing a web cvs viewer?
18:11.34drumkillai'm trying to decide which one to pursue
18:12.20*** join/#asterisk Dovid (~dovi5988@pool-151-198-8-101.mad.east.verizon.net)
18:13.11pasificoBroom: bueno pero me imagino que yo configuraria mi pc  con 2 tarjeta una fxo y otra fxs una para el co y la otra para el telefono analogo
18:13.26*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
18:14.40Qwelldrumkilla: you guys should use svn :p
18:14.53drumkillathat's another issue, silly head.
18:15.07citatsdrumkilla: viewcvs is about as simple as it gets
18:15.08QwellThen you could use a good websvn viewer
18:15.17pasificoBroom: gracias por todo cuando pueda me repondes
18:15.30drumkillacitats: that's what I figured
18:15.34drumkillai'd rather use the perl one than the python one i saw :)
18:15.55drumkillaor is viewcvs the python one ...
18:16.15citatshmm, dont recall.  lemme check
18:16.54drumkillayep, it is in python
18:17.04citatsyep python
18:17.06citatsheh
18:17.08*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
18:17.11drumkillahowever, it supports cvs and svn
18:17.22drumkillaso it would be an easy change if we needed to make it
18:17.32pabelangerAnybody happen to know what __zt_exception: Exception on 11 is?  Specificly the 11.
18:19.31syle2http://www.ubergeek.tv/article.php?pid=54&swfSize=1
18:19.32drumkilla11 == 3
18:19.51mutwhat dimension did i just warp into..
18:20.03Qwellmut: the Nth dimension
18:21.13brookshireugh.. python
18:21.27brookshireno python please
18:21.37brookshirehi harryvv
18:22.44Broompasifico: si, esa es la configuración
18:23.06*** join/#asterisk Stealthmethod (~123@adsl-070-148-141-013.sip.mia.bellsouth.net)
18:23.24*** join/#asterisk NeonLevel (~NeonLevel@dsl-201-128-242-219.prod-infinitum.com.mx)
18:24.04brookshirewe could get nicholson to put it up.. i don't want to deal with it is in python though
18:24.09pasificoBroom: bueno gracias por tu ayuda yo estoy en la fundacion codiga abiertos
18:24.17NeonLevelgood day, i'm having problems with callerid issues and this messages appears, anyone has fixed this? ""callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-85)"" thanks.
18:24.28pasificoBroom: dominicana
18:25.01*** join/#asterisk ayano (~erik@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
18:25.24ai-adoes asterisk@home gives a http server front end to asterisk ?
18:25.38NeonLevelgood day, i'm having problems with callerid issues and this messages appears""callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-85)""  anyone has fixed this, i need calllerid to route incoming calls please help!
18:25.53PakiPenguinai-a, yes
18:26.17ai-aPakiPenguin: i have asterisk installed already, can i get a hold of this html code to stick on my intranet?
18:26.34ai-await, ive just seen webmin interface, i will try that.
18:26.40PakiPenguinai-a, nope , it does lots of stuff , it uses amp and very cutomized dialplan
18:26.42pasificoBroom: pues si puedes ayudarme+ somo una comunida de linuxzero en republica dominicana
18:27.02NeonLevelplease help me with my callerid problems.
18:27.18ai-aPakiPenguin: is it worth having a dedicated server then? i wanted this box to be used for other things aswell,,
18:27.33PakiPenguinai-a, asterisk@home installs centos
18:27.47PakiPenguinyou get a standard linux box
18:27.50Broompasifico: yo soy de puerto rico
18:27.54PakiPenguinyou can use it for any purpose you want to
18:28.07ai-aPakiPenguin: but i have fedora installed with asterisk,, is there a add on that i can use for this?
18:28.19pasificoBroom: interesado en cualquiera ayuda q puede aportar
18:28.28pasificointeresado en cualquiera ayuda q puede aportar
18:28.30PakiPenguinai-a, i doubt that , you'll have to change the scripts to use rpms for fedora
18:28.37pasificointeresado en cualquiera ayuda q puede aportar
18:29.10pasificoBroom: esquesame mi teclado estab astascado
18:29.23ai-aPakiPenguin: how do i add a user to asterisk so i can connect via my iax client?
18:29.31pasificoBroom: esquesame mi teclado estab atascado
18:29.51PakiPenguinai-a, read the wiki
18:30.41pasificoBroom: oh q bien eres de puerto rico
18:31.13*** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net)
18:31.18pasificoBroom: Q tan avanzado estan con asterisk
18:31.55fockshow is conversation recording usually handled? by transferring the current call to a MeetMe and recording that?
18:32.21ai-adoes asterisk come in a rpm that i can automate with yum?
18:32.28docelm0no
18:32.42docelm0It comes in CVS form you can automate
18:32.47Qwellfocks: look at features.conf, automon
18:32.54Broompasifico: no conozco mucha gente que lo utilize, yo lo estoy probando en donde trabajo a ver si migran todo
18:32.54Qwellfocks: You can record a call with *1
18:33.07ai-aive written a update.sh, make.sh and install.sh, i could automate that.
18:33.19docelm0there ya go
18:33.37ai-ajust wondering if yum could do this nightly for me :), i will make it compile every few weeks.
18:33.58*** join/#asterisk stkn (~stkn@stkn-active-pdpc.developer.gentoo)
18:33.59*** join/#asterisk JunK-Y (~foobar@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
18:34.00docelm0Nope..  No distro like that as far as I know..
18:34.06docelm0You could be the first to make it..  :)
18:34.11*** join/#asterisk mrgoby (~mrgoby@aa.linuxbox.com)
18:34.15Qwellautomate the rpm build process
18:34.20focksQwell, seems like * combinations are ignored on my system. does Asterisk listen for DTMF during a SIP call?
18:34.28Qwellfocks: should
18:34.49pasificoBroom: ok pues nosotro esta muy interesado de parte de mi fundacion en realidad por otro la do mi jefe solo quiere beneficio en el trabajo pero eso
18:34.57mrgobyfocks: depends on your signalling method
18:35.16focksQwell, hmm, during a call i do *1, but i don't see anything happening in Asterisk's console logging
18:35.17mrgobyif you specify inband in your dtmfmode for your user, then yes
18:35.27Qwellfocks: it has to be fast, and automon has to be set
18:35.31focksmrgoby, that's in sip.conf right?
18:35.34*** join/#asterisk the_devil_dont_s (~Adam@195.26.12.229)
18:35.37focksQwell, automon is set
18:35.38mrgobycorrect
18:35.44pasificoBroom: y donde trabajo son muy gwindowsiano
18:35.45Qwellit has to be really fast, like < 500ms
18:36.08focksQwell, it's fast man ;)
18:36.36focksi do see attemping native bridge in there when i *1
18:36.41*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
18:36.54the_devil_dont_shey can i ask a question
18:37.01mrgobyfocks: that sort of dtmf signalling isnt realiable for codecs other than ulaw/alaw
18:37.12focksi'm usiong ulaw
18:37.21the_devil_dont_sim setting up a test asterisk box to sow my boss that it would be better than the f**kin pbx were using
18:37.38focksmrgoby, any thoughts?
18:37.40pasificoBroom: pero te invito a q entres a la pagina abiertos.org yo soy elvyn encargado de el aria de redes
18:37.49mrgobyon waht ?
18:37.57focksmrgoby, why it seems to be ignoring *1
18:38.01harryvvdevil, what does your current pbx do and whats with the nasty words?
18:38.04PakiPenguinhey
18:38.23PakiPenguinanyone has some nice call forwarding process/dialplan logic?
18:38.29the_devil_dont_si put in an intel 536dep fax/data/voice modem and these are suppose to work if you make a change to the wcfxo.c file before compling or by removing the r13 resistor and r19 resistor
18:38.54*** part/#asterisk fanguin (~user@p548F2937.dip0.t-ipconnect.de)
18:38.57the_devil_dont_sas the software change i made didnt work, i tried to remove the resistor but r19 isn't there
18:38.58mrgobyhard to say.  what does your dialplan look like?
18:38.59the_devil_dont_sany ideas
18:39.05*** join/#asterisk dasuberdavid (~David@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
18:39.47focksmrgoby, want me to pastbin it?
18:39.48*** join/#asterisk brettnem (~Brett@207.90.232.34)
18:39.49PakiPenguinmrgoby, just want call forwarding for users , like they dialin some number and their call's forwarded to that
18:39.54PakiPenguinoops
18:39.56PakiPenguinsorry
18:40.08ai-ahow do i completly remove asterisk libpri and zaptel ?
18:40.10brettnemhello all
18:40.21brettnemsucky day in texas
18:40.25mrgobysure
18:40.35Qwellbrettnem: most days in Texas are, I'd imagine.
18:40.44brettnemQwell: oh come on, it's not that bad
18:40.47Qwell:p
18:40.53brettnemjust hot or raining
18:41.07JerJerstay in the air conditioning
18:41.21brettnemright.. if it works
18:41.28JerJerwith the windows shut and the music blasting
18:41.34JerJerthat's what i do
18:41.37mrgobythey have sweet tea in texas, no ?
18:41.45focksQwell, do i need anything in the diaplan in addition to fetaures.conf for recording?
18:41.48JerJerthey call it Koolaid
18:41.59JerJertea flavored koolaid
18:42.02mrgobyyikes
18:42.03brettnemit's like what douglas adams said, "some days it's so awful outside you just want to open a window and stick your head in"
18:42.37brettnemno sweet tea in texas.
18:43.09mrgoby:(
18:44.01Inv_arpcontext "A"  is my incoming IVR  "B" is internal sipphones and "C" is outside pstn line ;  A includes => B  and B includes => C,  how can i prevent context "A" from accessing pstn "C"?
18:44.36JerJermake a better dialplan
18:44.41brettnemheh
18:44.57hardwireblah
18:45.00brettnemdon't like the house?? throw away the key
18:45.01JerJeruse what i call a landing context
18:45.06JerJer[staff]
18:45.10JerJerinclude => local
18:45.14hardwireblah and 1/2
18:45.14JerJerincldue => longdistance
18:45.18JerJerinclude => stations
18:45.23JerJer[public_phone]
18:45.26PakiPenguinJerJer, any call forwarding example please or followme type of thing
18:45.26JerJerinclude => local
18:45.30JerJerinclude => stations
18:45.31hardwirewhats the most stable version of asterisk cvs?
18:45.37hardwireversion/revision
18:45.47pabelangerhardwire: v1.0
18:45.50hardwireI am not terrible familiar with how CVS revisions work
18:45.52JerJerdefine most stable
18:45.53Inv_arpJerJer: ahh k
18:45.55mishehubah.
18:46.04hardwireJerJer: more stable than the 1.0.9 source release
18:46.07hardwirewith fun options
18:46.17mishehuwhat are "fun options" ?
18:46.27JerJerfun options == shit that breaks
18:46.28brettnemI think fun=bugs
18:46.31brettnemhaha
18:46.46twisted[asteria]hardwire wants his cake and eat it too
18:46.46brettnemyeah what he said
18:46.51brettnemmm cake
18:46.54brettnemI want cake too
18:47.07hardwireyeh
18:47.09hardwireI kinda doo
18:47.15brettnemsam's club makes an awesome cake
18:47.23mishehutoo much cake makes you fat.
18:47.27twisted[asteria]hardwire, wait for 1.2
18:47.28mishehuand diabetic
18:47.36brettnemthank you oopaloompa
18:47.56mishehubrettnem: I'm way taller than an oompaloompa
18:48.06hardwiretwisted: yeh
18:48.09hardwireI figured as much
18:48.09mishehuat least from the original movie version of willy wonka
18:48.12hardwireseptember ish?
18:48.15mishehuwith gene wilder
18:48.15twisted[asteria]OR
18:48.16brettnemI'm underweight.. so I can have the cake.. :)
18:48.18twisted[asteria]you can check out cvs-head
18:48.23brettnemthe original was a whole lot better..
18:48.27twisted[asteria]it runs pretty damn well IMHO
18:48.27hardwiretwisted: wanting atleast limited sip presense support
18:48.29Wonkaargh
18:48.30brettnemI didn't like the new oopaloompas..
18:48.39twisted[asteria]hardwire, oh.  haha. good one ;)
18:48.46hardwiretwisted[asteria]: heh..
18:48.51hardwirewhats the holdup on something like that?
18:48.58twisted[asteria]someone writing it?
18:49.21mishehuWonka: a fish?  a chocolate fish maybe?
18:49.24brettnemmy 4 year old daughter was scared of the new willy wonka movie.. come on.. burning and melting carnival atractions are kinda spooky
18:49.37Wonkamishehu: no!
18:49.48*** join/#asterisk salmandr (~salmandr@mdsnwinas01pool2-a21.mdsnwi.tds.net)
18:49.54Wonkamishehu: i've got nothing in common with that movie or that novel...
18:50.01mishehubrettnem: how abotu the boat scene from the gene wilder version?
18:50.05Wonkaand everywhere i'm hilighted :/
18:50.13hardwiretwisted[asteria]: if you had to guess.. how many companies are funding developers to work on asterisk features?
18:50.20mishehuWonka: do you have an assistant named Umpa ?
18:50.22brettnemmishehu: yeah, that's pretty freaky
18:50.27Wonkanope
18:50.50brettnemhardwire: 2374
18:50.56brettnemoops.. 2375
18:51.19blitzrageHEY!
18:51.24blitzrageAll of you - get back to work!
18:51.29brookshireshh!
18:51.31*** join/#asterisk gaffney (~gaffney@70.88.90.25)
18:51.31blitzragetwisted[asteria]: especially j00!
18:51.35mishehuhow can I work?  I'm listening to moc talk
18:51.55twisted[asteria]blitzrage, I *AM* working
18:52.12blitzragetwisted[asteria]: no you're not, you're chatting in here at me
18:52.24brettnemoh here comes that multithreading crap
18:52.26twisted[asteria]it's called extreme multiasking
18:52.30mishehuand he has a french(?) accent, so I need to pay attention so I don't misunderstand
18:52.36blitzragetwisted[asteria]: :D
18:52.40twisted[asteria]and lunch
18:52.40twisted[asteria]bbl
18:52.43brettnemI believe that's threading, not tasking, right? ;)
18:52.49blitzragemishehu: yes, french Canadian
18:53.02gaffneyI'm having a weird problem with asterisk when I reboot the system it fails to start but starting it immediately after it works just fine
18:53.12brettnem"it works"
18:53.34gaffneymeaning it that it comes up and is usable
18:53.36brettnemwhat did you do to make asterisk start when you reboot? and what distro are you running?
18:53.37mishehublitzrage: ah, worse then no?  ;-)
18:53.45mishehuquebecois
18:53.50*** join/#asterisk ayano (~erik@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
18:53.50*** join/#asterisk astoria (~cluecon@67.107.50.40.ptr.us.xo.net)
18:53.55shidogaffney - u have a .so that is missing something
18:53.59gaffneyIm running gentoo and it's being started in the default run level
18:54.02astoriaAnyone do anything with SMPP and SMS?
18:54.11mishehubut he's giving a pretty good presentation thus far
18:54.15brettnemyeah probably a messed up library..
18:54.26shidocheck ld.so.conf AFTER you find out whats missing
18:54.28mishehuastoria: smpp or xmpp ?
18:54.28gaffneyshido: I did just remove speex and bri / pri support
18:54.28shidothen add it there
18:54.40coppiceastoria: yes, but not with *
18:54.44astoriamishehu: smpp.. for sms.
18:54.58*** part/#asterisk the_devil_dont_s (~Adam@195.26.12.229)
18:55.06astoriacoppice: i'm looking for an smpp provider that doesn't require minimums (so i can learn things and play around)
18:55.30mishehuastoria: ah, donno much about that new fangled sms stuff heh
18:55.38mishehuwhich isn't really new or fangled
18:55.39file[laptop]astoria: you're back!
18:55.52brettnemheh, only if your an american.. :P
18:55.53astoriai just went to lunch, then to my room for a while to do some work.
18:55.53mishehufile[laptop]: you been able to figure out who I am yet?
18:56.02coppiceastoria: they don't usually entertain people making SMPP connections unless they process substantial traffic
18:56.03file[laptop]mishehu: didn't care that much :P
18:56.03gaffneyThanks for the help,I'll try rebuilding it
18:56.12file[laptop]but I'm right behind you
18:56.19brettnemgaffney: look in your logs to determine what is broken..
18:56.22astoriacoppice: oh. how does one learn about SMPP or test things, then?
18:56.22file[laptop]and no I'm not Peter
18:56.34coppiceastoria: what SMPP software do you intend to use?
18:56.40astoriacoppice: kannel
18:56.40mishehufile[laptop]: oooooh I'm insulted!  and heck, I was standing right near you at the entrance to the symposium before I went out to talk with craig, brian, and tony.
18:56.52coppicekannel is crap.
18:56.57astoriacoppice: and I was thinking about playing around with perl::SMPP
18:57.02file[laptop]mishehu: hi
18:57.08mishehufile[laptop]: no fair, you can see over my shoulder
18:57.14file[laptop]yup
18:57.17file[laptop]I cheated
18:57.30astoriacoppice; it's hard for me to learn what is crap and what isn't, when I can't find someone to terminate the stuff for me..
18:57.47gaffneyHmmm... apparently app_voicemail is missing
18:57.48hardwireman this is just insane
18:57.50coppiceI didn't known there even was a perl::SMPP
18:57.58hardwireI have so many sip phones to auto-configure / put in sip.conf
18:58.03hardwirebut they all need different configs
18:58.04hardwirewell
18:58.06astoriacoppice: yeah, there isa n SMPP module for perl. :)
18:58.06hardwire10% of them do
18:58.17hardwireif there was only a way to do all this so very easily :)
18:58.36astoriacoppice: what SMPP client do you prefer?
18:58.47coppiceastoria: Logica were promoting a free Java SMPP thingy at one point. I never tried it, but you might look at that
18:58.48astoriahardwire: realtime is overrated...
18:58.49Qwelluhh...ok, so...
18:58.59hardwireastoria: maybe
18:59.06QwellI've got a 7960, and a ethernet-wireless bridge.  Do I want a straight-thru, or crossover cable?
18:59.06hardwireastoria: why is twisted in you?
18:59.21hardwireah.. nm
18:59.24astoriahardwire: what are you talking about?
18:59.27hardwirenothing
18:59.31hardwirejust read something wrong
18:59.42*** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net)
18:59.55hardwireWi_Fi: are you on wirelessly?
18:59.56Qwellnm, I'll bring both
18:59.57astoriacoppice: kannel is a pain to use, but once you get it setup, it looks pretty easy and stable.
19:00.10coppiceastoria: I wrote a complete SMPP platform of my own, because things like kannel turned out to be a PITA. I can't make the whole thing free, though. I half adapted the bits I could make free into a package at one time, but never finished
19:00.19Netgeekshrm, realtime is overrated because people treat it as the use for everything, or because it's a bad idea?
19:00.37astoriacoppice: may i ask who you terminated your SMPP traffic with?
19:00.44hardwireas well as a snom auto config tool
19:00.50hardwireand oh.. the extensions as well
19:00.56hardwirejust using different views for each
19:01.04coppicekannel is useless. it fails to do anything but the most elementary things, and I don't think it has changed a lot since I used it (since the main developers went bankrupt)
19:01.24astoriacoppice: it doesn't look like they've updated anything since nov 2004..
19:01.28coppiceI terminated traffic with one of the operators in HK (I am in HK)
19:01.39*** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
19:01.39*** mode/#asterisk [+o anthm] by ChanServ
19:02.02astoriacoppice: oh. I'm in the US - trying to get in the market while it's still pretty early.. I have some business ideas but just want to test thigns, but these providers make it difficult.
19:02.08JerJeryo find us some cheap parts there in HK   :)
19:02.33coppicecheaps parts for what?
19:03.01JerJerresistors and micro controllers
19:03.23JerJerbut it was my lame attempt to make a joke i guess  :)
19:03.59coppiceyou want parts for microcontrollers? they aren't usually considered repairable :-)
19:04.12coppicemost things are cheaper in the US these days
19:04.33JerJerno like many thousands of microcontroler pieces
19:04.37coppicealthough a lot of US products are crippled compared to the version sold elsewhere
19:04.49JerJerchips
19:04.50astoriacoppice: http://search.cpan.org/~sampo/Net-SMPP-1.03/SMPP.pm
19:06.31*** part/#asterisk mrgoby (~mrgoby@aa.linuxbox.com)
19:06.41*** join/#asterisk tim27 (~tim27@97-70.dr.cgocable.ca)
19:06.55coppiceastoria: it looks like that just does the bit banging for the PDUs. you really need something persistent to do SMPP properly, which takes some real work.
19:07.19*** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
19:08.14astoriacoppice: hmmm...
19:08.44coppiceI used a MySQL database, and passed everything through that.
19:09.32astoriacoppice: Oh, so if I already have some kind of storage/frontend, I could pass SMS thru Perl::SMPP or kannel, pretty effectively?
19:09.40QwellSo, somebody wanna give me a sanity check here?
19:09.55Maarkenyou're crazy.  get help.
19:10.01*** join/#asterisk eYeLes (eYeLess@jnv-bsr181.alshamil.net.ae)
19:10.11eYeLesHi all.
19:10.14QwellI'm taking a 7960 with me, I'm bringing a wireless bridge.  Setup the bridge, put the 7960 on the bridge, tftp to my router, forward tftp to the tftp server
19:10.15mishehuQwell: sure.  and we all fail.
19:10.17QwellDone, and done?
19:10.43Qwelland should it be a crossover or straight-thru from the 7960 on the bridge?
19:10.55Maarkenstraight
19:11.07Maarkenand yeah, that should work
19:11.09gaffneyMy issue with asterisk crashing on boot is being cause by it starting before mysql... causing the voicemail module to die
19:11.27coppiceYou could probably cook up a reasonable SMSC like setup in perl. avoid kannel though. if you are serious about SMPP you'll just waste time, get frustrated, and throw it away in the end
19:11.28QwellMaarken: ok, thanks.  Didn't want to forget anything :p
19:11.34eYeLesHi, I have 6 BRI Lines, 5 callers will be on hold and one will be transfered to a phone, is Asterisk what I need for such a task ?
19:11.49astoriacoppice: thanks, you probably just saved me the rest of my afternoon :)
19:11.54*** join/#asterisk [hC] (~hardcore@8.10.2.5)
19:12.27*** join/#asterisk sedwards50 (~chatzilla@adsl-67-125-150-70.dsl.irvnca.pacbell.net)
19:12.34astoriacoppice: i was wondering what happened to kannel's development. The kannel-users list is still pretty active, but not in a good way. Mostly confused people and me trying to find a provider.
19:12.34*** join/#asterisk cjk (~cjk@80.92.64.103)
19:12.57file[laptop]silly bugs...
19:13.06mishehutricks are for rabbits?
19:13.08mishehutrix
19:13.16file[laptop]or dingos
19:13.19twisted[asteria]tricks are for hos
19:13.20sedwards50How can I keep queue members from hearing each other's DTMF?
19:13.42file[laptop]hey twisted
19:13.52twisted[asteria]like file
19:14.02coppicekannel was started by a swedis startup making a WAP platform. WAP bombed and so did that. for a couple of years nothing happened to kannel. they some minor half-hearted stuff began to happen. I think that's about the state of it
19:14.04twisted[asteria]:)
19:14.26JunK-Yfile sucks :P
19:14.46brookshirereally?
19:14.46[hC]The cable that connects from a smartjack to the PRI cards port, is that a standard rj45 ethernet cable?
19:14.55pabelangerhC: yes
19:15.05[hC]pabelanger:  hmm okay.
19:15.16[hC]I wonder if my sangoma A104U is supposed to li ght up green when i connect it up then
19:15.17eYeLesHi, I have 6 BRI Lines, 5 callers will be on hold and one will be transfered to a phone, is Asterisk what I need for such a task ?
19:15.22astoriacoppice: thats the story from too many projects.. What do you think of SMPP perl module? I'm a decent perl programmer, but still a newb at SMPP.
19:15.23brookshirepri is usually a t1 cabel
19:15.39JunK-Ybrookshire: why do u think he's not responding, he's busy...
19:15.39JunK-Yhehe
19:15.50pabelangerhC: if you not getting sync you may need a T1 crossover
19:16.16mtgh~google smpp
19:16.18*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
19:16.30*** join/#asterisk fanguin (~user@p548F2937.dip0.t-ipconnect.de)
19:16.55sedwards50Any clues on queue() and turning off hearing DTMF?
19:17.14*** join/#asterisk allanon (allanon@c-24-18-189-146.hsd1.wa.comcast.net)
19:17.31eYeLesdoes anyone have a guide on what the wiring should look like? i mean you have an incoming line (modem) but where do the internal phones connect to?? sorry im very new to this
19:17.51*** join/#asterisk pointer (pointer@aj.catt.com)
19:18.30pabelangereYeLes: what type of phones are we talking about?
19:18.51coppiceastoria: I just had a quick look at the opensmpp library I started making about 3 years ago. it looks like I did get all the proprietary stuff out, do I can't just release what is there for someone else to start from
19:19.09eYeLespabelanger normal phones at home, pots
19:19.33sedwards50SOMEBODY on this list has got to have a clue on queue() and turning off hearing DTMF?
19:19.45coppices/did/didn't
19:19.46eYeLesi read the guide, i couldnt find anything on the physical setup
19:19.53twisted[asteria]sedwards50, this is not a list.
19:19.56twisted[asteria]sedwards50, this is IRC.
19:20.13*** part/#asterisk pointer (pointer@aj.catt.com)
19:20.14sedwards50OK, my mistake, s/list/irc/g
19:20.20pabelangerThe CO (your telco) will plug into an FXO modules, and your internal phones will plug into FXS.
19:20.33*** join/#asterisk Romik (~romik_@1.fix.netvision.net.il)
19:20.35pabelangerhttp://www.digium.com/index.php?menu=fxsvfxo
19:20.40Romikhello
19:20.59twisted[asteria]and yeah.  we'll just stop them from hearing swear words while we're at it
19:21.01eYeLesaha! i need to read on that :) thanks
19:21.16Romiki need few tollfree numbers in canada (sip or iax) which DID provider support it?
19:21.22pabelangereYeLes: np
19:21.37astoriacoppice: that's okay. I wasn't asking you to give me anything that you've done or release anything. Thanks for looking though!
19:21.42sedwards50twisted, am I asking this question in the wrong place?
19:21.55*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
19:22.12pabelangerRomik: I too have been looking, but I can't find anything under $.07 /min
19:22.35Romik<PROTECTED>
19:22.51coppiceastoria: well I did want to get an open implementation of SMPP going when I started that library, mostly inspired by the fact there is nothing really solid out there that is free.
19:23.02Romik<PROTECTED>
19:23.53shido?
19:23.57shidowhat Romik ?
19:24.10shidosell what?
19:24.24Romikshido: canadian tollfree
19:24.26mishehuhe's looking for ca dids
19:24.29mishehufor tollfree
19:24.37shidohow many do you need?
19:24.59*** join/#asterisk alerios (~alerios@63.245.87.180)
19:25.04astoriacoppice: have you seen some of the other smpp libraries out there such as: opensmpp.logica.com ?
19:25.10Romikshido: i need bulk of 5 ;) and i need 5 more US tollfree dids...unable to order them from website
19:25.25tim27shido i want 800 DID canada too
19:25.33eYeLespabelanger I have 6 BRI Lines, 5 callers will be on hold and one will be transfered to a phone, is Asterisk what I need for such a task ?
19:25.46astoriacoppice: perhaps this is the java one you were talking about.
19:25.57brettnembarf.. java
19:26.17brettnemoops pardon me
19:26.35pabelangereYeLes: depends on which BRI modem you have:  check out http://www.voip-info.org/wiki-BRI
19:26.44coppiceastoria: that was released after I built my own platfom. I haven't looked recently, but what they released at the beginning was rather limited. It would give you a basic SMSC simulation to interact with, though
19:27.09*** join/#asterisk clinthome (~clinthome@snap.helixsystems.com)
19:27.20pabelangereYeLes: once you get your driver setup, you should be able to do what you want.
19:27.30eYeLespabelanger i didnt get the modems, im just making sure if asterisk is what i need, so i start researching in that direction
19:27.50astoriacoppice: hmm insteresting. I don't know any java though. But I'm just looking to dump outbound SMS through another SMSC, then the perl module should be able to handle that, I presume.
19:27.52[hC]so for a sangoma t1 card, most likely unless the telco did something funky with the smartjack, a standard ethernet cable should work for a patch?
19:27.58mishehumy eyes aren't so great, kinda hard to see the font on the screen
19:28.53coppiceastoria: simplistic dumping soon turns into a more sophisticated solution, if the traffic has any importance at all.
19:29.00pabelangerhC: possible, let me see if I can find a wiring diagram
19:29.21eYeLespabelanger you see the cards at diginum have the fxo and fxs on the same card, now fxo in my case is the bri modems, what is fxs?
19:29.34coppiceastoria: it looks like opensmpp hasn't been updated since the initial release at the end of 2001
19:30.25astoriacoppice: oh dear. there's not a whole lot out there. What about smpp makes it difficult to "dump out" messages or causes reliability problems?
19:30.37astoriacoppice: thanks, by the way, i appreciate your advise on this..
19:30.47sivanaheison: ping
19:30.50ManxPower~fxofxs
19:30.50jbot[fxofxs] An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
19:31.05heisonsivana: talk to tim27 and romik
19:31.39pabelangereYeLes: you will not use FXO and FXS modules if you plan to use a BRI's.  You need digital cards, FXO and FXS are analog
19:31.49*** join/#asterisk gnarf37 (~corey@west-rock.rockriver.net)
19:31.56gnarf37Hey.
19:32.13eYeLesoww idiot!
19:32.19eYeLessorry i didnt know that
19:32.56gnarf37Have a question, I just wrote a patch for cdr_pgsql.c to enable a "spool file" if for some reason the connection to the SQL server is lost.  How do I go about submitting it correctly, cant seem to find the right page with instructions ;)
19:33.22coppiceastoria: SMSCs seem to go down. I've had this experience with several SMSCs. If you have a persistent system you buffer and automatically pass on the messages when you can. do you care about delivery receipts? you probably think no, but later it will turn out you do. now you need persistence to tie up the messages and their eventual result. it just goes on, leading from an initially simple...
19:33.24coppice...notion to a full scale SMSC :-)
19:34.26coppiceaccounting may be necessary too
19:34.55ManxPowerI'd like to the the IP or phone number of my SMSC
19:34.56astoriacoppice: you're right. I was thinking about that - how'd I'd have to send out the SMS, then wait for the delivery receipt back from the SMSC, and match the two together in a db, such as mysql or postgresql. I guess a little bit of binding would be required.
19:35.03pabelangerhC: This should work for you:
19:35.03pabelangerSideA = 1 2 3 4 5 6 7 8
19:35.04pabelangerSideB = 4 5 3 1 2 6 7 8
19:35.09pabelangerwiring diagram
19:35.35*** join/#asterisk squinky86 (~jon@border0hsv.asterisksgi.com)
19:35.57[hC]pabelanger: if the other side is looped, will i get a link?
19:36.04[hC]pabelanger: when I got here, there was a loop plug in the smartjack
19:36.40squinky86How difficult would it be to detect if a dialed number is a fax machine or a real person?
19:37.16twisted[asteria]about as difficult as bkw dating a girl
19:37.32coppiceI just saw the latest prediction for cellphones in China in 2008 - 500M. That will require some nice beefy SMSCs. They like SMS in China :-)
19:37.47twisted[asteria]hey coppice, know any way to do what squinky86 was asking?  I'm kinda curious too ;)
19:37.52astoriacoppice: and i'd have to tie it in with some kind of billing facility as well.  Wow, Kannel sucks - it can't do any of this.
19:38.03*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
19:38.11aleriossquinky86, you can detect incoming fax with exte=fax
19:38.18alerios*exten
19:38.21squinky86yeah, I know I can do that.
19:38.28astoriacoppice: fortunately, I'm here int the US, where we won't get those beefy SMS until 2020. :)
19:38.28pabelanger[hC]: I think so...
19:38.28coppicekannel can't even handle comms errors properly, and its slow too
19:38.40ManxPoweryou mean like NVFaxDetect (see the Wiki)
19:38.52pabelangereYeLes: np
19:39.05squinky86I don't want to detect if the incoming call is a fax, I want to detect to see if we connect to a fax machine.
19:39.11*** part/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu)
19:39.15astoriacoppice: well, I think I"m going to give the perl module a shot. I can program perl and mysql and all that good stuff, so perhaps I can bind everything together...
19:39.57coppicetry the SMSC simulator from logica. it might be quite helpful. then again, it might not :-)
19:39.59mutanyone have a link to where i can get a phone line extender/repeater?
19:40.02squinky86(please use the duct tape on twisted)
19:40.06mishehuwhich is almost always mispronounced as "duck tape"
19:40.35mishehusquinky86: do you have weird, twisted fantasies?
19:41.01squinky86mishehu, no, I'm not 15, so he won't do anything to me ;)
19:41.38tim27shido: are yout here
19:41.41coppicesquinky86: when you call into a FAX machine it might send you a burst of answer tone. snag is, these days most fax machines don't. The calling machine almost always goes beep-beep-beep. You can detect that, and know with confidence the caller is a fax machine. knowing what the answerer is may be tough
19:41.45astoriacoppice: everything is always a hundred times more complex than you think it is :)
19:42.14coppiceastoria: not really. I expect everything to escalate massively
19:42.45squinky86coppice, Thank you, so in other words, we can't gaurantee 100% success rate on detecting connection to fax machines, only <50%?
19:42.57astoriacoppice: well maybe I still have to learn some lessons the hard way ;)
19:44.01coppicesquinky86: I guess a proper VAD might tell you with high confidence that a voice was there. without that you could guess its a modem.
19:44.38squinky86hmm, and if I guess it's a modem and send the "beep beep beep" to it, will there be a response?
19:44.58coppiceanyone know what happened to the mailing list?
19:45.22mishehuI don't remember if it's the sending or receiving side that initiates the handshake.
19:45.39mishehuI know the sending side usually sends a tone every few seconds
19:46.02jontowbee-boop, BEEEP boop BEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEeeeskhhhhhhhhhhhhhhhhhhhhhhhhhh
19:46.19brookshireoh no you didn't
19:46.31jontow[+++ATH0]
19:46.39squinky86hmm, I guess I'll just have to go experiment.  Thank you mishehu and coppice!  <jontow: System is going down for system halt NOW!!!>
19:47.05jontowhooray for accurate ascii representation of modem handshake noise
19:47.28*** join/#asterisk mrFr33Ze (mrFreeZe@cust.93.4.adsl.cistron.nl)
19:47.53coppicesquinky86: its messy. the basic rule in fax is nobody complies with the spec :-). The answering machine should send the first message, and the calling machine will respond. However, lots of machines try to identify if the caller is voice or fax. this will conflict with any attempt to identify the answerer
19:48.47*** join/#asterisk Romik (~romik_@1.fix.netvision.net.il)
19:49.01astoriacoppice: the mailing list is back up now. I think they were having issues with postfix.
19:49.17tzangerin soviet russia, postfix have problems WITH YOU!!
19:49.28squinky86coppice, so the best method would be to wait for an answer, then wait for a voice.  If the voice keeps talking and talking, it's an answering machine.  If the voice stops, it's an answer. If there is no voice, send a "bee-boop" and a "BEEEEEEeeskhhhh"?
19:49.37*** join/#asterisk cgcorea (~cgcorea@63.245.14.194)
19:49.42coppicei started getting mail today, but it appears some people got it yesterday, and maybe the day before
19:49.53mrFr33ZeHi, mayb someone can help me out. I am on Debian, in the Netherlands, with asterisk on a normal ISDN line, everything is working fine. But @ my office, we have a Detewe PABX, with a 8xS0 card in it. When connecting asterisk to that, i can't get it working. Nothing responds. Strange thing is though, that a regular ISDN phone is working on the port...
19:49.53Beirdosquinky86: or it might be my sister
19:49.55Beirdohehe
19:50.04Beirdoshe just never shuts up
19:50.12syle2what does CLEC stand for again
19:50.16syle2completely forgot
19:50.28squinky86Beirdo, That's what hidden cameras and blackmail are for.
19:50.33coppicesquinky86: trying to differentiate human from answering machine just doesn't work - been there, failed to do that :-)
19:50.33astoriasyle2: competitive local exchange carrier
19:50.39syle2ty
19:50.55astoriasyle2: as compared to an incumbent local exchange carrier ILEC
19:51.08astoriasyle2: don't you just love FCC ackronyms? :)
19:51.42syle2yeah wonderful
19:51.43coppicesqinky86: A company telephonist answers the phone is a way that is very similar to an answering machine (except for the beep)
19:52.06Darwin35bsd users of asterisk  with issues can join #asterisk-bsd so we can document issues
19:52.42Beirdomuhahaha
19:52.47Beirdosorry
19:53.29mrFr33Zesomeone can give me a little help with my setup ?? :)
19:54.08shidowhats up mrFr33Ze ?
19:54.26twisted[asteria]WOOHOO
19:54.26mrFr33ZeHi, mayb someone can help me out. I am on Debian, in the Netherlands, with asterisk on a normal ISDN line, everything is working fine. But @ my office, we have a Detewe PABX, with a 8xS0 card in it. When connecting asterisk to that, i can't get it working. Nothing responds. Strange thing is though, that a regular ISDN phone is working on the port...
19:54.30twisted[asteria]i get my own (sorta) office!
19:54.50shidosorta office?
19:55.00squinky86mishehu, alright, get the duct tape back out
19:55.07astoriamrFr33Ze: what kind of card are you using in your * box?
19:55.19mrFr33Zeastoria: an AVM Fritz for the moment
19:55.36mrFr33Zejust want to use it for voice and call deflecting
19:56.54astoriacoppice: there are even SMPP PHP classes :)
19:57.22coppicea whole SMSC in PHP. wonderful
19:57.49astoriaha ha.
19:59.40[hC]anyone here using a sangoma t1 card?
20:01.30shidono, we tend not to slap digium in the face
20:01.37mutfsckin sipura
20:01.49*** part/#asterisk alerios (~alerios@63.245.87.180)
20:01.50coppiceastoria: I wonder what people use these classes for?
20:01.52mutthey don't do any kind of support over the phone
20:02.03muti need to know how long of a run this box will power!
20:02.03shidowhats wrong with the sipura
20:02.05harryvvinteresting that asterisk is the forth largest chat chanell on freenode
20:02.05*** join/#asterisk jaxxan (~Snak@202.70.125.109)
20:02.07*** join/#asterisk gtigene (~chatzilla@70.89.216.41)
20:02.19shidoits getting bigger
20:02.19shido:)
20:02.24mutand where i can get range extenders if needbe
20:02.24astoriacoppice: nothing probably. Looks like some guy made it for the sake of making it.. the pear module is really new though and looks promising.
20:02.26*** part/#asterisk brettnem (~Brett@207.90.232.34)
20:02.43coppiceharryvv: 4th by users or by waffle?
20:02.44brookshireintresting that we are the largest chat channel that actually manages to stay on topic most of the time ;)
20:02.45jaxxanwhere do you get the latest zapata-1.0.9
20:02.53gtigene?
20:03.02astoriajaxxan: digium.com
20:03.03jaxxanit's not on the main page. i need to rebuild my asterisk box, i was running 1.0.1 before
20:03.05brookshirezpata-1.0.9.1
20:03.15brookshirehttp://beta.asterisk.org
20:03.16drumkillazaptel, not zapata
20:03.18drumkilladifferent things :)
20:03.20brookshireoh.. nm
20:03.20brookshirehaha
20:03.37pabelangerjaxxan: http://www.asterisk.org/html/downloads/zaptel-1.0.9.1.tar.gz
20:03.56jaxxanso zapata was combined with zaptel ?
20:04.25*** join/#asterisk gtigene (~chatzilla@70.89.216.41)
20:04.44gtigene?
20:05.20sivanashido: lol... you're funny
20:05.25jaxxani have zapata-1.0.1.tar.gz and zaptel-1.0.1.tar.gz
20:05.44mutall i see is this
20:05.45muthttp://www.twacomm.com/catalog/model_LLA-1.htm?sid=E76D6E2E66BD9D33962B071A1DA3C90C
20:05.51jaxxanbut i'm going to 1.0.9 now
20:05.56gtigeneMy log says ""Failed to authenticate user 1150 <sip:1150@192.168.1.9>... for SUBSCRIBE." Could this be related to my missed calls list not working?
20:05.57muti dunno if thats what i want tho
20:06.00pabelangerjaxxan: my mistake, I tought you where looking for Zaptel latest
20:06.06jaxxanno
20:06.30jaxxanany idea where i can get zapata-1.0.9? it's not on the asterisk.org download page
20:06.40*** join/#asterisk Damin (~damin@nucleus.nacs.net)
20:08.00*** join/#asterisk brettnem (~brettnem@72.29.102.158)
20:08.15drumkillawhy do you need it?
20:08.20*** join/#asterisk mrgoby (~mrgoby@aa.linuxbox.com)
20:08.37gtigeneMy log says ""Failed to authenticate user 1150 <sip:1150@192.168.1.9>... for SUBSCRIBE." The phone works but the missed calls list doesn't. What could be the problem?
20:08.43jaxxancause my asterisk box is having some serious hardware issues, so i'm making a whole new box now
20:08.51*** part/#asterisk brettnem (~brettnem@72.29.102.158)
20:08.54eYeLespabelanger is this the only piece of hardware i need?? http://www.avm.de/en/Produkte/Server-Produkte/C4/index.html
20:09.09jaxxani'm using tormenta2 T400P cards
20:09.34brookshirejust download zaptel
20:10.14jaxxanso it's the same thing then ?
20:10.54squinky86jaxxan, no, but for your purposes, pretend like it is.
20:10.55jaxxancause in 1.0.1, zaptel contains my t400p card settings, and zapata.conf contains my channels and stuff
20:11.23jaxxanso i have a T400P with a PRI going to a DMS100
20:12.15jaxxani always needed zapata before but, ok
20:12.49pabelangereYeLes: Looks like the right hardware, but I don't know if it is support under asterisk.
20:14.19eYeLespabelanger it is
20:14.24eYeLesi just checked :)
20:14.29drumkillahttp://cvsweb.digium.com
20:14.29drumkillayay
20:14.35*** join/#asterisk zotz (~zotz@24.231.36.100)
20:15.11pabelangereYeLes: Aight, it looks like you found your board.
20:15.28pabelangerdrumkilla: looks good
20:15.31*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
20:15.42*** join/#asterisk crash3m (crash3m@crash3m.user)
20:16.01*** join/#asterisk jpmcallister (~jpmcallis@kapla.escelsa.com.br)
20:16.05*** join/#asterisk meppl (~mephisto@87.193.5.179)
20:16.41eYeLespabelanger i still dont understand something, lets say 3 bri are connected to this board for incoming, where do i connect the phone im forwarding to? the last bri jack??
20:17.27jpmcallisterhello everyone. I'm testing asterisk using xlite and some others softphones. I'm having problem with low volume sounds when I call another extension
20:17.33jaxxanyeah, the readme file in zaptel-1.0.9.1 states that some of the testing programs still require the zapata library, the zttool (which i use) requires libnewt
20:17.37*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
20:17.56jaxxando i need an updated version of zapata? or can i just use the 1.0.1 version that i already have ?
20:17.58muthow many REN is a normal telco line?
20:18.03shidojpmcallister,  if its a softphone an u have low volume then pump up the mic volume
20:18.05jpmcallisterthe weird thing is that if al call someone in the siphone network, our fwd the volume is normal
20:18.13shidoor turn on the "boost" most soun card programs have
20:18.16*** join/#asterisk brettnem (~brettnem@72.29.102.158)
20:18.21shidobut be sure you have a noise cancelling mic
20:18.39shidoare you calling another softphone, jpmcallister  ?
20:18.55pabelangereYeLes: No, you would need another interface either an analog FXS or SIP.  There is no such thing as a BRI phone, that I know of.
20:18.57jpmcallisteryes
20:19.09shidothen turn up the volume :)
20:19.47brettnemsure there are BRI phones..
20:20.12[hC]Hmm. zaptel doesnt want to compile cleanly on 2.4.27..
20:20.17jpmcallistershido: I've tried that. The problem seens to be only betewen extensions in asterisk, If I call my wife that is connected to sipphone using xlite the sound is normal
20:21.07jpmcallisteror any call made to fwdnet, etc
20:21.14jpmcallisterthe sound is away ok.
20:21.32*** join/#asterisk xlizard (~lizard@p508F6F0A.dip.t-dialin.net)
20:22.43mrFr33Zesomebody got some experience with asterisk behind another PABX ? (detewe in my case?)
20:22.47pabelanger[hC]: Linux asterisk 2.4.26 worked for me if you want to downgrade.
20:22.54mrFr33Zethis is so frustrating :}
20:23.31pabelangermrFr33Ze: how are you connected to the PBX?
20:23.55mrFr33Zepabelanger: to an S0 card
20:24.10mrFr33Zeasterisk(AVM Fritz)<-->Detewe(S0-1)
20:24.39*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
20:24.39mrFr33Zeon a regular BRI asterisk is working fine
20:24.50mrFr33Zeon the detewe, it don't pick up
20:24.56mrFr33Zeno logs, no nothing,
20:25.03thalmrFreeze: even with set verbose 4?
20:25.13mrFr33Zebut when i hook-up an ISDN-phone i can call, and be-called
20:25.19mrFr33Zeyes, asterisk -cvvvv
20:25.36thalmrFreeze: what version of chan_capi are you using?
20:25.47mrFr33Zethal: let me check
20:26.04shidojpmcallister, show me your extensions.conf and sip.conf
20:26.19mrFr33Zethal: 0.5.4
20:27.14mrFr33Zethal: does that make sense? :)
20:27.33mutwho else makes ata's bside spura?
20:27.34mutsipura
20:27.40mutsomeone i could call tech support for
20:28.03thalmrfreeze: yes
20:28.10thalmrfreeze: this is the sourceforge version?
20:28.20mrFr33Zethal: yes ?
20:28.47eYeLeswith queues you need to have as many lines as queues?
20:28.52thalmrfreeze: well. looks like a cabling problem, i assume
20:29.38mrFr33Zethal yes, thats the strange thing of it, when i hookup a normal ISDN-phone to the same cable, it works flawless
20:30.04mrFr33Zeand when i hookup my asterisk (with the same cable) to a regular line, it works also fine
20:30.09thalmrfreeze: maybe this detewe thing is assuming a system phone on that port?
20:31.05mrFr33Zethal: could be, but then the ISDN-phone should't work..., and its a dedicated S0 card in the Detewe (8xS0)
20:31.12coppicemut: tech support for ATAs is virtually non-existant
20:31.14*** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218)
20:31.25mrFr33Zeso basicly i can't but UPN phones on that ports :)
20:31.32mutit's all email
20:31.57coppiceand the e-mails get answered after 2 months and several reminders
20:32.00mrFr33Zeanyone else mayb got a clue ??
20:32.11mutyea
20:32.21muti don't have the resources to test this tho
20:32.23mutO_O
20:32.30brookshiredigium's support is pretty quick :)
20:32.33drumkillacoppice: you can get quick support for an iaxy :)
20:32.59coppicedoes an iaxy do t.38?
20:33.01mutdrumkilla: i need to know how far of a run the iaxy can do over the rj11
20:33.10mutthey probly have no idea..
20:33.29jpmcallisterhttp://pastebin.ca/19418
20:33.29drumkillamut: support could probably give you an answer ...
20:33.41Beirdoworks OK, but it's a toasty little bugger
20:33.43drumkillaBeirdo: the new ones don't get so hot :)
20:33.49brookshireit's only tested for 6 feet i think
20:34.02brookshireat least that is what the certification says
20:34.06brookshirebut i know it can do more
20:34.07Beirdodrumkilla: good :)
20:34.12jontowmut; try it ;)
20:34.26muttry what
20:34.31jontowa long run
20:34.43jontowgot an RJ11 crimper and a piece of cat3?
20:34.47Beirdomy iaxy's taking a one-way trip at the end of the month
20:34.48muti jus said i don't have to resources to tes it
20:34.54jontowah, missed that line, sorry :/
20:35.10muto_O
20:35.31BeirdoI will have an extension far far away.  will save me (and her) tons more money as long as our internet links stay up
20:35.48brookshirethat's hot
20:35.50Beirdoand I didn't want to mess with SIP remotely :)
20:35.52Beirdoheh
20:36.03BeirdoIAXy to the rescue :)
20:36.09brookshireBeirdo: did you see gtkiaxyprov?
20:36.16Beirdonah
20:36.22BeirdoI don't use X
20:36.26Netgeeksjpmcallister you really should wipe out your passwords before posting conf files to pastebin
20:36.28Beirdothat wouldn't help me much anyways :)
20:36.37brookshireahh.. there is one for windows too
20:36.38brookshirelol
20:36.42mutmeh
20:36.43pabelangerDigium should make an FXO + FXS IAXy device
20:36.45pabelanger:)
20:36.47BeirdoI use it from a commandline
20:36.50*** join/#asterisk dsfr (~dsfr@dsfr.digium.sponsor.pdpc)
20:36.57drumkillapabelanger: you should give me a million dollars
20:37.06Beirdoheh
20:37.11Beirdomake it 2 million
20:37.24mutholy crap
20:37.27muttoo many menu's
20:37.46mutpaid support
20:37.48mutpah!
20:37.53BeirdoI'd like a "normal" processor in there, but once again...  it takes big money to develop devices that work well
20:37.57brookshiremut: support@digium.com
20:37.58pabelangerdrumkilla: Hmm... no?
20:37.59Beirdoand DSPs are cheap, but still
20:38.02brookshireor sales@digium.com
20:38.05muthaha, i use the same hold music
20:38.22brookshirehuh?
20:38.26brookshiredsps are not cheap
20:38.30Beirdoif someone wants to fund me, I'd happily work on DSP-based solutions
20:38.32Beirdosure they are
20:38.36Beirdo$5-6 in quantity
20:38.40brookshirethey are if you manufacture them
20:38.41mut"i don't think theres a phone cord restriction"
20:38.44drumkillathat's expensive :)
20:38.45muthe has no clue
20:38.57Beirdoheh
20:39.16Beirdodrumkilla: I know where you're coming from, I did design work for years. :)
20:39.21Beirdobut still :)
20:39.24xlizardi have a problem: anytime my sip-phone tryes to connect to asterisk, asterisk is killed.  i got "asterisk[4418]: segfault at 00000000ad272731 rip 00002aaaab3f3154 rsp 00002aaaad26ea08 error 4" in dmesg ... anybody, any idea?
20:39.36*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
20:39.36pabelangerwhat does digium sell the Dialogic asterisk drivers for?  I have a ton of Dialogic boards I could play with ;)
20:39.48coppiceBeirdo: a DSP capable of doing VoIP isn't $5-6
20:39.52Beirdosend email to sales@digium.com, dude
20:39.54coppiceits much less :-)
20:40.01mutno more than 20 or 30 feet
20:40.02mutheh
20:40.07muthe had no clue at all
20:40.08muto well
20:40.10Beirdocoppice: ones capable of doing the ulaw that the IAXy does?  sure
20:40.23brookshireiaxy does echo can too
20:40.25brookshire:)
20:40.34Beirdoif you want to do G729, etc, well, yeah, it would be tougher
20:40.52*** join/#asterisk meppl (~mephisto@87.193.5.179)
20:40.57Ariel_pabelanger, good luck with the dialogic board hope you have the time and hair left to work the problems out.
20:40.58coppicesilicon to do the IAXy job + proper echo can and g.729 would be <$5
20:41.00*** join/#asterisk rover (~ruse@c-24-126-77-233.hsd1.ca.comcast.net)
20:41.06Beirdothe current DSPs are a lot cheaper than people do expect in the end
20:41.26BeirdoI should finish my CHU receiver project some decade
20:41.29Beirdoheh
20:41.37Beirdoused an Atmel AVR do do DSP work
20:41.42Beirdoand it worked
20:41.50Beirdoand the Atmel rep was drooling
20:41.58Beirdothen my boss cancelled the project
20:42.17Beirdobut the code was mine from before I worked there, and is GPL.  hehe
20:42.18Beirdosucker
20:42.47Beirdobut now I don't have someone to pay for the board manufacture, etc, so the project kinda stopped
20:42.51Beirdo:(
20:42.56jpmcallisterNetgeeks: ye. big mistake
20:43.26coppiceA $5 DSP would be a complete VoIP setup on a chip - ethernet, analogue converters and so on.
20:43.37mutAudioCodes MediaPack™ MP104 4-Port FXS Analog Gateway - SIP Based
20:43.40mutanyone ever used one of those/
20:43.46mutor recommend another 4 line ata?
20:44.42Beirdocoppice: yeah, about that
20:45.02Beirdowith only about a $1M-2M startup cost
20:45.37Beirdothe general purpose DSPs would be more like $8 for one that can do it, but likely needs more stuff off-chip
20:45.44coppiceI wonder what makers pay for the PA168?
20:45.55brookshireso that's $8 / channel?
20:45.58Beirdodunno
20:46.04coppicethere are very few general purpose DSPs over $8
20:46.08Beirdoyou could price it out, I'm sure
20:46.43BeirdoI've seen some still up to $50 in quantity
20:46.48Beirdobut they are bitchin DSPs
20:47.07Beirdolike you might be able to do G729 on a whole T1 or something
20:47.23*** join/#asterisk astoria (~cluecon@67.107.50.40.ptr.us.xo.net)
20:47.26Beirdounfrigging believable what's available if you have money
20:47.36coppice30 x G.729 would be <$10
20:47.36Beirdoof course, then you need to license the codecs too
20:47.43astoriaanyone here @ cluecon?
20:48.04*** join/#asterisk zx225 (~me@65.183.42.3)
20:48.32Beirdocoppice: the coding of the DSP is a lot of work though unless you buy algorithms
20:48.40Beirdoeven then, a fair amount of work
20:48.52*** part/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net)
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20:49.32coppiceactually, a $10 DSP would do more like 60 x G.729 + 60 x echo can and all the rest
20:49.56Beirdowhich ones?  the ADI ones?
20:50.01coppicebut you would need to cost in external RAM
20:50.05Beirdoyeah
20:50.07Beirdotrue
20:50.20coppicethe make doesn't really matter
20:50.43Beirdoyeah, I guess not, all three of the big ones have good prices now
20:51.41Beirdoand at that point it becomes a religious war which manufacturer to use
20:52.35coppicea sane choice would be based on what software modules can be bought, and for how much
20:53.48Beirdoaye
20:54.02Beirdoand which tools are cheaply/readily/etc available
20:54.16Beirdoand also what extras are integrated into the chips
20:54.58Beirdoit was fun doing FIRs in an Atmel AVR though
20:54.59Beirdoheh
20:55.33coppicemost small MCUs are used for a variety of DSP jobs these days
20:55.46Beirdoin realtime off an 8kHz sampling rate, filtering the tones in Bell 103
20:55.51Beirdoand also 1kHz
20:56.19Beirdohad to use hand-assembly for part of it in the end, but it was fun
20:56.23coppicethat's pretty trivial. more common is doing something rather more complex at very low rates
20:56.28Beirdoumm
20:56.31Beirdonot trivial
20:56.39Beirdoit used 90% of the CPU bandwidth
20:56.54BeirdoI don't think you quite comprehend how slow the AVRs are :)
20:57.00Beirdoit's like using a PIC
20:57.11coppiceI know exactly how fast they are
20:57.12Beirdobut with better instruction set
20:57.14BeirdoOK
20:57.23tzangerthe AVRs are a little better than PIC since there's no branch prefetch hit
20:57.26coppiceits certainly a step up from a PIC
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20:57.33BeirdoI think I was using 5th order FIRs
20:57.34tzangerand I think they've ramped the clock up on those buggers significantly too
20:57.38Beirdowith power calculations
20:57.50tzangerI've got code that does FFT and FIR filters on PIC17C42s I think
20:57.59Beirdo32bit fixed point math in a 8 bit processor
20:58.11coppiceFIRs are a lousy way to do that job
20:58.14Beirdoit could keep up, but not by much
20:58.24tzangerI cheated with my power calculations
20:58.32Beirdoum, not really, not when you need accurate timing
20:58.46tzangerlookup tables and Taylor series for square roots and divide by 3s
20:58.55Beirdothis was to make an NTP source from off-air shortwave radio signals
20:59.27coppicetzanger: why would you need to do a sqrt by taylor?
20:59.31Beirdoand a very noisy one if you don't happen to live in Ottawa
20:59.43tzangercoppice: I didn't do sqrt with taylor.  sqrt was lookup table IIRC
20:59.52tzangerthe divide by 3 was taylor
21:00.07BeirdoI have test recordings from Florida that I could distinguish by ear, but took a while to get filtered well enough to be useful
21:00.15coppicestill,  greater + 3/8th lesser is usually good enough
21:00.23Beirdothe PPS 1kHz tone was easy enough
21:00.34Beirdothe 300baud signal every minute was a PITA
21:01.01tzangergreater + 3/8 lesser?
21:01.04Beirdoespecially as you need to turn on the 1kHz detection immediately after
21:01.47shidowhat has carbonated water, guarana flavor, sugar, citric acid, soium benzoate, color, an licensed by blue spike beverages inc. ?
21:02.01Beirdodunno, but it sounds good
21:02.10coppicetzanger: modulus = greater re & im + 3/8 of the other
21:02.11Beirdoand I doubt the guarana's for flavor :)
21:02.13shidoit comes in a Blue bottle..
21:02.18shidoribbed
21:02.18Beirdoit's there for the caffeine
21:02.19Beirdo:)
21:02.26tzangercoppice: interesting
21:02.40shido300 ml per bottle
21:02.50shidotastes like cream soda
21:02.59opus_hmmm.
21:03.02Beirdoanyways, coppice...  was 3 years ago I worked on this last, I can't remember all the details anymore
21:03.03shidobeen aroun forever...
21:03.06opus_is there a way for asterisk to receive a SMS message?
21:03.12opus_for cheap, free, etc
21:03.23shidoBAWLS
21:03.30coppiceBeirdo: you shouldn't have used up all your CPU power doing FIRs. the modem would have been easier then :-)
21:03.37Beirdono no no
21:03.44Beirdothe idea was to put it all in one chip
21:03.50tzangercoppice: :-)
21:03.51BeirdoCOTS chip
21:04.19coppicetzanger: I don't think he catches on very quickly :-)
21:04.25Beirdoand I had it as one chip, plus a programmable clock generator
21:04.37Beirdoplus some LRC circuitry.
21:04.48Beirdothat was an entire shortwave receiver
21:06.14Beirdocoppice: how would you suggest to extract Bell 103 tones out of really noisy shortwave reception?
21:06.16opus_are you guys talking about SR?
21:06.34coppiceno. modems
21:07.10Beirdowhere the signal at times is like 1.5dB above the substantial noice floor
21:07.18coppiceBeirdo: well if its really noisy, you don't want to start with FIRs
21:07.33tzanger1.5dB?  Heaven!  Back in my day we had to strain just to hear the NOISE!  and we liked it that way!
21:07.36coppiceyou should be able to detect bell 103 well below the noise
21:08.50Beirdooh, and this is off shortwave (amplitude modulated)..  and sometimes the tones will be recieved off-frequency
21:09.25Beirdowithout using nasty algorithms, how do you propose doing that... and with nearly no delay?
21:09.51Beirdoa correlation is as computationally expensive if not more so than FIR
21:10.03coppiceBeirdo: maybe you should study a little signal processing :-)
21:10.30Beirdobah
21:10.40BeirdoI have.
21:10.50coppiceOK, a little more
21:10.52Beirdoand so had several coworkers that were helping.
21:11.08Beirdoincluding one who did noise reduction for hearing aids
21:11.41Beirdobut whatever :)
21:12.11opus_noise reduction for hearing aids?
21:12.24coppiceyeah, turn them down :-)
21:12.54Beirdoyeah, like removing background noises so voices are easier to hear
21:13.05Beirdoincluding such things as waterfalls
21:13.06*** join/#asterisk rkvarala (rkvarala@202.65.130.81)
21:13.28*** join/#asterisk mtrifiro (mtrifiro@dhcp-159-145.tis.utexas.edu)
21:13.35Beirdoanyways, all fun stuff
21:14.00coppicebasically you adaptively detect repetitive patterns in the sound, and remove them. Many of the more annoying background noises are rhythmic
21:14.11Beirdoyup
21:14.25Beirdolike the jackhammer outside.
21:14.30Beirdoshut UP, bastards!
21:14.59rkvaraladoes we have any application in asterisk where we can listen to two users conversation and we have facility to talk to only one person in that conversation
21:15.13Beirdogah, someone's on topic
21:15.17BeirdoI should go home now :)
21:15.45coppiceyou're on topic, actually. that same denoising is now in cellphones, and some IP phones
21:15.53[hC]if asterisk -vvvvvc halts after the zapata.conf line, does that mean theres a problem with the config file?
21:15.55[hC]like here:
21:15.55[hC]<PROTECTED>
21:15.55[hC]<PROTECTED>
21:15.55[hC]<PROTECTED>
21:15.56[hC]<PROTECTED>
21:16.00[hC]<PROTECTED>
21:16.05[hC]*dies*
21:17.35Beirdothat's true
21:17.36*** join/#asterisk tsume (~tsume@tsume.user)
21:17.40Beirdosome of that is in use now
21:17.58*** join/#asterisk YoYo_ (YoYo@pool-70-110-6-88.roa.east.verizon.net)
21:18.02coppicebut more watts are available to do it :-)
21:18.24Beirdoheh, yeah, considerably more FLOPs too
21:18.34opus_have you tried implementing follow me before?
21:19.23xhelioxDoes anyone know what the Astericon hotel rates are?
21:19.30rkvaraladoes we have any application in asterisk where we can listen to two users conversation and we have facility to talk to only one person in that conversation
21:20.01rkvaralai.e third person can talk to only first guy
21:20.03opus_whisper?
21:20.05shido[hC], yes
21:20.17shidorkvarala, yes
21:20.20[hC]Hrm... looks fine to me.
21:20.33rkvaralashido : what is name of application
21:20.48shido[hC], pastebin your zaptel.conf and zapata.conf but what does ztcfg -vv say first?
21:20.50rkvaralaso that i can check it in voip-info.org
21:21.09shidoit not on voip-info
21:21.15rkvaralaok
21:21.23rkvaralaso where can we find that
21:21.25shidoyou can use the monitor app to make your own
21:21.25*** join/#asterisk IRCMonkey0815 (~asterisk-@dsl-084-056-130-051.arcor-ip.net)
21:21.28shidochanspy, too
21:21.36shidoand the manager interface
21:21.37rkvaralachan spy is to listen
21:21.44shidoyes you're right
21:21.50shido1+2 = 3
21:21.55rkvaralabut  not to talk to another person
21:22.01rkvaralawith that is it now
21:22.01shidoyes
21:22.55shidoto build the forensics app ( as I call it ) you need a manager user monitor app , schanspy and a phone connected to your * box
21:23.19JerJer1 + 1 = 3
21:23.56Beirdo1 + 1  = 11
21:24.05drumkillathe limit of 1 + 1 = 3 as 1 approaches 1.5
21:24.24IRCMonkey0815hi, i want to call meetme but DTMFs are recognized double (dialing '1234#) (Recognizing: '11223344#')
21:24.29Beirdoanyways, I'm gonna go home.  You all have a good weekend
21:24.57shido1 + 1 = 10
21:25.11IRCMonkey0815I'm calling from external via zaptel device
21:25.34JerJerthere are 10 types of people. Those that understand binary and those that do not.
21:25.51blitzragebut thats only 2!
21:26.20*** join/#asterisk jpcarvalho (Jeff@201.30.193.135)
21:26.57jpcarvalhoHello All , someone could me donate a script to rate my cdr's based on postgresql ?
21:27.35tsumeheh
21:27.46tsumewhy do people think everybody's time comes cheap?
21:27.55gnarf37jpcarvalho you mean to store them in postgresql?
21:28.17tsumejpcarvalho: you want something, you pay for it, or do it yourself.
21:28.37jpcarvalhoNo , they're stored but i need rate each call and send a report with summary
21:28.48tsumejpcarvalho: or you can go to rentacoder and get a child in college to do it for you.
21:28.55jpcarvalho:)
21:28.58gnarf37speaking of postgres, anyone actually using it to store their cdr information?
21:29.09jpcarvalhoI'm using it !
21:29.24[hC]shido: http://pastebin.ca/19422
21:29.29tsumegnarf37: well since postgresql is better than mysql ;)
21:29.29Netgeeks<-- uses rate_engine from trollphone with mysql
21:29.32jpcarvalhoi have > 59000 registers
21:29.40gnarf37i'm lookin for a few people to test out a patch i wrote to spool CDR inserts to a file if the postgres server goes down
21:30.04*** join/#asterisk A-Tuin (~a-tuin@gw.ip.v4.me.uk)
21:30.11jpcarvalhomy server is  in production mode :(
21:30.28jpcarvalhotsume : I don't use mysql
21:30.50shido[hC], and what o u get at the cli when u run asterisk - vvvvgcd
21:30.53jpcarvalhoI just use 3 prices , is not hard do rate it
21:31.11[hC]Last 3 lines:
21:31.12[hC]<PROTECTED>
21:31.12[hC]<PROTECTED>
21:31.12[hC]<PROTECTED>
21:31.12[hC]vg1:/etc/asterisk#
21:31.20shidoooooh
21:31.21gnarf37anyone know a good replication solution for postgres?
21:31.26shidook
21:31.36shidowell are you running the rivers you JUST compiled?
21:31.48shidothere's a d hiding there somewhere
21:32.00[hC]??
21:32.00[hC]lol
21:32.13[hC]I just compiled the sangoma drivers yes. Maybe i need to recompile the zaptel ones again
21:33.03*** part/#asterisk mtrifiro (mtrifiro@dhcp-159-145.tis.utexas.edu)
21:33.40shidook
21:33.41shidoBING
21:33.43shidosangoma?!
21:33.52shidonothing I can do for you :(
21:33.59[hC]:(
21:34.08[hC]Im sorry, I didnt choose. :/
21:34.09shidowhy' u buy sangoma and not digium?
21:34.19[hC]It arrived that way
21:34.21shidoreading my mind here :)
21:34.22Netgeeksdo you have the lastest sangoma drivers?
21:34.29loudits common sense to spend your $ in digium.
21:34.33Netgeeksyou should check to make sure
21:34.34[hC]I agree.
21:34.45Netgeeksyou may need to do a firmware upgrade to the sangoma
21:34.53shidoDOA send it back get a refund and buy digium.
21:35.07JerJeryes cuz you won't find much sympathy for Sangoma supporters around here
21:35.13xlizardmmm ... my asterisk exits with a SIGSEGV when it tryes to setsockopt(11, SOL_IP, IP_TOS... while i am trying to connect with an sip-phone ... ;(
21:35.26loudbesides that, it gave you asterisk.
21:35.54IRCMonkey0815hi ,2nd try, can anyone help me with my DTMF Problem ?
21:36.01heisongnarf37: replication?
21:36.39[hC]So, were this a digium card, what would cause * to fail out there?
21:36.55NetgeekshC: with sangoma, you have to do the following
21:37.09[hC]I see the problem.
21:37.11NetgeekshC:1. Make absultely sure you have the latest driver tarball
21:37.16[hC]Aug  5 14:36:45 ERROR[24982]: chan_zap.c:10118 setup_zap: Unknown signalling method 'pri_cpe'
21:37.16[hC]Aug  5 14:36:45 ERROR[24982]: chan_zap.c:9749 setup_zap: Signalling must be specified before any channels are.
21:37.23Netgeeks...
21:37.45[hC]looks like it didnt link libpri maybe?
21:38.03[hC]Sorry I didnt mean to interrupt.
21:38.06Netgeeksonce you run the Sangoma Setup program, you have to recompile Zaptel
21:38.08shidoso set the signalling before the channel
21:38.25NetgeeksSangoma patches zaptel during it's setup install
21:39.17Netgeeksthe latest tarball will also come with a firmware updater and latest firmware, you should update the card's firmware if it's not current
21:39.44[hC]nod.. I thought i had read something about recompiling zaptel after sangoma install.
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21:44.26NetgeeksSangomas are a REAL pain in the arse to configure (as compared to Digium cards), however they perform significantly better in a high load environment
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21:44.56*** mode/#asterisk [+o twisted[asteria]] by ChanServ
21:44.59twisted[asteria]whee
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21:45.17gnarf37heison: server clustering / backup servers, etc
21:45.34brookshireNetgeeks: not compared to 2nd gen digium cards
21:45.55wunderkin2nd gen would be the new ones with echo cancel?
21:45.59brookshireno..
21:46.05wunderkinor referring to firmware version
21:46.24brookshireall 205, 210, 405, 406, 410, 411
21:46.33brookshireout now
21:46.48Netgeeksmy tests compared sangoma A104u with Te410P
21:46.51funxionwhen are the DS3 cards coming out
21:46.51brookshireyes.. second generation firmware
21:47.09brookshirereal soon now.. we have prototypes ;)
21:47.15funxiontru
21:47.27coppiceeveryone who uses the sangoma card seems to find them more tolerant of shared interrupts, different motherboards and so on. they also see lower loading. not sure why, though. do they use a lower interrupt rate?
21:47.28gtigeneMy Polycom phones don't register missed calls correctly. No matter if you miss the call or you talk for 10 minutes, it still shows up as a missed call. What could be el problema?
21:47.34brookshireNetgeeks: how long ago, second generation firmware has not be out long
21:47.40wunderkinhow much does it cost to upgrade the firmware? are the digiums self updateable yet?
21:48.00brookshirewunderkin: sales@digium.com
21:48.02NetgeeksI ran the tests in January
21:48.15*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
21:48.15*** mode/#asterisk [+o bkw_] by ChanServ
21:48.27twisted[asteria]bkw_
21:48.30twisted[asteria]whassabi
21:48.31funxionbrookshire how much machine do the DS3 cards neeed?
21:48.34twisted[asteria]is cluecon over?
21:48.56brookshirefunxion: more than one, right now
21:49.02funxionDOH
21:49.13funxionhow many what processor?
21:49.24drumkillaa single 200 khz processor
21:49.40coppicewith enormous IPC
21:49.41twisted[asteria]drumkilla, woot
21:49.57drumkillathere will also be a DS3 iaxy
21:50.08drumkilla*** THE PREVIOUS STATEMENT WAS FALSE ***
21:50.08brookshireHAHA
21:50.20brookshireoh.. way to throw them for a loop
21:50.43brookshirealthough.. it was true statement, lol
21:52.09*** join/#asterisk bjohnson (~bjohnson@i216-58-65-113.igs.net)
21:52.46gtigeneMy SIP phones don't register missed calls correctly. No matter if you miss the call or you talk for 10 minutes, it still shows up as a missed call. What could be el problema?
21:53.08coppicehas anyone heard recently if atacomm is still working on his E1/T1 cards?
21:53.18gtigeneI have tried all kinds of manuevers to fix it, no joy.
21:53.25Netgeekshaven't heard anything from him, Coppice
21:53.36Netgeeksstill shows the ipVolution card on his site tho
21:53.41gtigeneI read the 80 page manual
21:54.05gnarf37gtigene: maybe your calling the phone twice? :)
21:54.37coppiceNetgeeks: ipvolution.com, tdm120.com, and tdm60.com have lapsed
21:54.37gtigenegnarf37: That is an interesting idea... Like, an error in the dial plan?
21:55.10Netgeekscoppice: read second paragraph on this page: http://www.sangoma.com/company/news_releases/octasic.htm
21:55.12gnarf37gtigene: potentially, check your Dial strings... it was kinda a joke, but it could possibly be right
21:55.20*** join/#asterisk mtrifiro (mtrifiro@dhcp-159-145.tis.utexas.edu)
21:55.42gtigenegnarf37: THanks :)
21:55.46*** join/#asterisk BingoPajama (~zcw100@68-233-16-90.stcgpa.adelphia.net)
21:56.34coppiceNetgeeks: what of it?
21:57.00coppicethe world is really running out of company names when someone calls their company octasic
21:57.00NetgeeksThat offers most of the same functionality of the ipVolution card
21:58.00coppicei see no possibility the ipvolution card could do what they say. I suspect they are still trying to get it to market in some form, though.
21:58.22*** part/#asterisk mtrifiro (mtrifiro@dhcp-159-145.tis.utexas.edu)
21:58.35NetgeeksCoppice: I agree with your statement at the price they were offering
21:59.27brookshireocatasic is just an echo canceller
21:59.33coppiceIts not the price. its the strategy. they bit off more than they could chew
22:00.23brookshireoh.. and omg.. digium cards do that today
22:00.43*** join/#asterisk tholo (~tholo@nat.sigmasoft.com)
22:00.57coppicebrookshire: not just echo cancellation, but the rest is pretty lightweight, and is no bother on the main CPU. It doesn't do the codecs
22:01.57*** join/#asterisk nwhit (~chatzilla@adsl-68-76-249-238.dsl.chcgil.ameritech.net)
22:02.07*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
22:02.13brookshirecoppice: octasic makes echo cancellation chips..
22:02.28brookshireoctasic will not make a sangoma card do g729
22:02.42coppicebrookshire: duh! I think that's what I said
22:02.54coppiceThey will make it do 8 E1s, though
22:03.41Netgeeksy depth into octasic, they make echo cancellation chips, Vocoder chips, and Several different packetization chips
22:03.54brookshireyeah.. and you can do 12 t1s with the current digium cards on dual 3ghz intel xeon
22:04.25brookshireand g729
22:04.38brookshireno wait.. g729 with only one card
22:04.43Netgeeksvocoder chips supports 255 full duplex streams at 16,24,32,40, and 64kbps rates
22:04.46coppiceI mean 8 E1s on one small card. Dunno where they fit the connectors, though
22:04.49brookshirebut 12 t1s ulaw
22:05.11Netgeeksbrookshire: those are the new 411's?
22:05.18brookshirehttp://www.digium.com/index.php?menu=press/pr_2gen_firm
22:05.45Netgeeksdigium needs a spell checker
22:05.55Netgeekss/duel/dual/
22:05.59*** join/#asterisk xkev (~kevin@orbit.xmission.com)
22:06.00brookshirei saw that
22:06.05xkev# asterisk -c
22:06.06xkevIllegal instruction
22:06.16drumkillathat's hot
22:06.19xkev# strace asterisk -c
22:06.23xkevsched_setscheduler(0, SCHED_OTHER, { 0 }) = 0
22:06.23xkev--- SIGILL (Illegal instruction) @ 0 (0) ---
22:06.33xkevwtf.  1.0.9
22:06.34coppiceNetgeeks: don't you know mark sees this as warefare? :-)
22:06.51Netgeekssees what as warefare?
22:06.51coppiceseems I need a spell checker too
22:07.04Netgeeksoh, warfare
22:07.10Netgeeksyou mean the other card makers?
22:07.17coppiceyep
22:07.21drumkillano, that's not true
22:07.31drumkillaare digium cards the only thing listed on asterisk.org ?  no.
22:07.49NetgeeksGreat, I'm glad he does, it means that I will have a much better selection of hardware now
22:09.12coppicedrumkilla: I dunno what he's feeding you guys, but too many people around here are starting to sound life apologists for microsoft. this is a bad things. a very bad thing
22:09.17NetgeeksNext he needs to build the ds-3 card with an on-board echo canceller and a vocoder as well, IMHO
22:09.30nwhitnetgeeks: mark needs to settle down with sangoma
22:09.48nwhitcompetition is good
22:10.10coppicesangoma have backed off from their DS3 work, as they found so little demand. seems like mark may have done so too
22:10.15fugitivocoppice: microsoft sux
22:10.38nwhitcoppice; there clear channel ds-3 works really well though
22:10.41xkevhttp://lists.digium.com/pipermail/asterisk-users/2005-June/111557.html <- my problem apparently
22:10.49nwhitive been running it for awhile now
22:11.03NetgeeksI'll gladly fork out the money for a ds-3 card that actually works (i.e. can do echo can and vocoding), whereas I have zero interest in a ds-3 interface that dumps the load on the hosting system
22:11.05nwhitand they'll get their channelized soon enough
22:11.26coppicenwhit: so I understand, and for data I understand it sells well. they were going to rush through a channelised version, but found little pull from customers
22:11.49tsumeomfg
22:11.51nwhitcoppice, not many people have 672 voice channels coming in
22:11.59nwhitcoppice, and running asterisk
22:12.06tsumegastman is the worst pos I've ever seen. I hope the guy who wrote this didn't get paid
22:12.17coppicenwhit: that rather depends who your target audience is
22:12.18NetgeeksI've got a client with 80 sangoma a104u cards all running on asterisk...
22:12.23tsumehe doesn't even check checksum, so when it screws, it screws
22:12.31nwhitcoppice, from a data standpoint (which i would use it for) i am really interested in channelized
22:12.31brookshireNetgeeks: won't happen until asterisk gets more efficient
22:12.43nwhitNetgeeks, running asterisk?
22:12.54Netgeeksyes, obviously on more than 1 machine  :)
22:13.03coppicenwhit: what for? split sources of data?
22:13.07nwhitNetgeeks, id hope so
22:13.16nwhitcoppice, ptp t-1s to customer premise locations
22:13.32Netgeeksnope, TDM gateways for Multi-tennant buildings
22:13.59nwhiti don't want to buy another m13 mux next time i order a ds3 and a bunch of 4port 1-1 cards
22:14.07nwhitits a much more gaceful solution
22:14.18Netgeekswhat nwhit said..
22:14.31nwhiti don't use cisco or juniper or etc....
22:14.39nwhitbeen using sangoma cards for 6 yrs
22:15.01coppiceits a lot of channels in one box, though. dual dual core machines will be pretty cheap in a few months, but its still a lot of channels to do any work on
22:15.10Cresl1nyeah
22:15.16nwhitcoppice, no cpu overhear
22:15.18nwhithead
22:15.20Cresl1nthat's more along the lines of what I think
22:15.24nwhitits all handled by the card
22:15.32NetgeeksBut I could care less who makes the card, I'll recommend the best card when asked by a client
22:15.38nwhitlinux can easily route 45mb of data
22:15.39Cresl1nnwhit: you don't seem to understand how this stuff works
22:15.42nwhitreal easy
22:15.46coppicenwhit: there doesn't appear to be EC on this card
22:16.14Netgeekson the sangoma ds-3 card?
22:16.19nwhitCresl1n, why do you say that
22:16.25coppicenwhit: its completely different from data. its lots of little streams, and the overheads from that are large
22:16.33Cresl1nnwhit: There's a lot more involved then just the card can do
22:16.38fugitivodoes sangoma has any card similar to the tdm400p?
22:16.39nwhitcoppice, i am just talking pure data
22:16.51nwhitno voice
22:16.56coppicethere doesn't appear to be EC on either DS3 card
22:16.59nwhitor... pri voice i should say
22:17.01Cresl1nnwhit: like what are you going to do with the channels once you terminate them?
22:17.45coppicenwhit: if you are handling PRI voice you have 672 little streams
22:17.47nwhitjust what i do right now with them... run data over them... ptp t1s to customer locations for data
22:18.13nwhiti don't want to do pris... thats inefficient
22:18.38coppiceyou just said "or... pri voice i should say"
22:18.58nwhitif i am going to do voip... leave it that way... use gsm or g729 to compress so that i can get 1.5mb of data and some channels of voice that don't take alot of bandwidth
22:19.15nwhitno voice ... or... pri voice i should say
22:19.39coppiceOK
22:20.15Netgeeks[15:18] nwhit: i don't want to do pris... thats inefficient <----  ??
22:20.34nwhitNetgeeks, out to customer primise locations
22:21.10Netgeeksso you own point to point ds-3s (or fiber), and run your IP network over that and put VoIP onto it?
22:21.13nwhitid much rather put a little linux box out there running asterisk that handles the voice and for the transport just use data... then i can get compression
22:21.39brookshirebut how are you going to get your channels into the asterisk box?
22:22.05nwhiti have a channelized ds3 from the telco that they are muxing t1s on out to my customers
22:22.21nwhitbrookshire, ptp data t-1
22:22.22brookshireso you'll be pri at some point?
22:22.59coppicefugitivo: the only people I know making analogue cards that will work with * are digium and voicetronix, and neither seems to have entirely happy customers
22:23.04nwhitbrookshire, if they have a existing pbx... then at the remote end i have asterisk running with a pri going into their pbx
22:24.24nwhitanyways... my only point is that if i am going out to customer locations, it makes more sense to be able to compress the voice than running a pri out there... they can get about 2-to-1 ratio of channels doing it that way
22:24.57brookshireheh
22:25.08brookshirewhatever works for you..
22:25.10nwhitand, at least in my case, don't think that i would be ordering from the telco 672 channels of voice on one single line
22:25.20Netgeeksso you get a DS-3 centralized PSTN interface, and run VoIP everywhere else?  thats what I assume you are looking for
22:25.48brookshireNetgeeks: i don't think he knows what he is looking for truely
22:26.11nwhitbrookshire, i am doing what i am looking for
22:26.14brookshireor understands what is involved in compression of that many channels
22:26.24brookshireand echo cancellation
22:26.45nwhiti am not explaining myself very well then
22:26.49nwhitsorry and never mind
22:27.24*** part/#asterisk xlizard (~lizard@p508F6F0A.dip.t-dialin.net)
22:27.28Netgeekssounds to me like he knows, just not getting it across clearly, but I understand, or think I do
22:27.34coppicethese E1/T1 cards should have had an H.100 bus, so connecting them to a compression card would work out nicely. its an excellent way to break up a E3/DS3, too
22:27.59nwhitNetgeeks, thanks
22:28.00NetgeeksHe wants a single large interface method (read centralized connection) to his PSTN provider
22:28.12Netgeeksand wants to run the rest of his Telephony network as pure IP
22:28.27shidooh god
22:28.34shidoyou remind me of brooktrout
22:29.02NetgeeksIn that scenario, a DS-3 card with 672 channel echo cancellation and a 672 channel vocoder would be perfect
22:29.06coppicethat was unkind and uncalled for :-)
22:29.13nwhitactually, i am using a large voice term provider for everything that isn't local
22:29.32shidoI used to work for a mass fax broadcasting company before Nufone and they loved those damned brooktrout cards
22:29.35nwhitand most of my traffice right now isn't local
22:29.36shidoexpensive little bastards
22:29.42shidowell not little...
22:29.46shidothe cards were full length
22:29.57shido$5k per card for 24 ports
22:30.30coppiceshido: the competitors were even more expensive. brooktrout grew by undercutting dialogic
22:31.25coppicethey used to charge about 8K for a 12 channel board
22:31.31shidolord
22:32.29coppiceand they really really sucked too
22:37.30Romiksomebody know what is means: Aug  5 18:55:24 WARNING[1969]: app_dial.c:362 wait_for_answer: Unable to forward frame
22:37.56Romik<PROTECTED>
22:37.57Romik<PROTECTED>
22:37.57RomikAug  5 18:55:24 WARNING[1969]: app_dial.c:362 wait_for_answer: Unable to forward frame
22:37.57Romik<PROTECTED>
22:41.27*** part/#asterisk thal (~thalunil@walledcity.de)
22:47.19*** join/#asterisk coolhp (~crap@mtl149-99-190-66.dedicated.sprintdsl.ca)
22:47.49coolhpGood day all`!
22:49.37rabelaiswith sip, it is conceivable that when a call is placed from a client to a server, the rtp stream becomes offloaded to a more local client so that a more direct conection is established (better voice quality) but the sip control commands still be transmitted between the caller and the gateway it registers with?
22:50.20coolhpIn theory yes, the signaling path is independant from the media path.
22:50.36coolhp(As far as I know ;-P)
22:50.47rabelaiscoolhp: so it is possible that my service provider could have implemented something like that?
22:50.58rabelaisbecause one of the representatives claims that's what happens
22:51.07Netgeeksyes, quite likely
22:51.09coppicethat's normal
22:51.16coolhpvery likely.
22:51.46rabelaisis there a way for me to verify that this happens?
22:51.54coolhpYeah...
22:51.56Netgeekspacket sniff
22:51.58rabelaisbecuase sip debug doesn't show me anything aside from the two ips
22:51.59coolhpyup
22:52.01rabelaisah
22:52.10rabelaisI see
22:52.11coolhpLOL.. netgeek was faster on it than me :-)
22:52.36rabelaisso even in the sip debug, I won't see the media path redirection?
22:52.48coolhpYeah you will
22:52.50rabelaisoh
22:52.52rabelaisI will?
22:52.55rabelaishmm  : (
22:52.56coolhpOf course.
22:52.59Netgeeksyeah, you should see the redirect for media
22:53.06rabelaisso I should see another ip that it goes to?
22:53.14coolhpA media path change is operated at the signaling level...
22:53.17Netgeeksyeah, I forget what the format looks like
22:53.23Netgeeksbut it has to be intere
22:53.47rabelaishmm
22:53.47rabelaisok
22:54.08coolhpCheckout RFC 3261 for the message types...
22:54.49coolhpOut of curiosity, is it possible to get ringing both inband and out of band on a PRI ? I've got a strange issue where when placing a call out through a PRI, ringing sounds distorted... it really sounds like I've got 2 ringings happenning at out and slightly out of phase.
22:56.02coppiceringback is only in-band. ringing of a phone is only out of band
22:57.35Netgeekscoolhp: you dial out via the PRI and you hear double ringing?
22:57.42coolhpyEAH
22:58.20Netgeeksif dialing out through a PRI, your asterisk server generates the ring tone.  The PRI just sends a message to Asterisk telling it that the end user is being rung, asterisk generates the ring for you locally
22:58.42rabelaisif I say "sip show channels"
22:58.56rabelaisis the peer's ip that's listed the one that the media path is going to?
22:59.12Netgeeksif you say "sip show channels" most people in listening vicinity won't know what you are saying....
22:59.31coolhpThis is the weird part... I'm pretty sure I've got ringing happening on the D-Channel and the B-Channel.
22:59.35rabelais...
22:59.50Netgeeksno, thats the ip address of the signalling peer or the ip address of the device that has registered
22:59.56rabelaisah ok
23:00.14coolhprebalais : Try using a tool called iptraf
23:00.37rabelaiscoolhp: I did
23:00.37coolhpIt may be a little easier to see than good old tcpdump :-)
23:00.49Netgeekscoolhp: that doesn't sound right
23:02.11shidowhen I make calls i always see ringing on the d channel....
23:02.16shidoNooooooo..
23:04.21coolhpHmmm
23:04.56coolhpDebugging at the PRI level, here's what I get : a CALL PROCEEDING message... immediatly followed by an ALERTING message
23:05.18coolhpThe ALERTING seems to trigger the audio channels to come up
23:05.49Hmmhesayshey folks
23:05.59coolhpUnless I'm mistaken, the CALL PROCEEDING should trigger the out-of-band ringing at the SIP level.
23:06.16coolhpBut then the ALERTING on the PRI opens up the Audio channel at the same time.
23:06.40*** part/#asterisk tholo (~tholo@nat.sigmasoft.com)
23:06.53coolhpI must admit I'm no expert in Q931 but something just doesnt look right...
23:08.14twisted[asteria]hey Hmmhesays is cluecon over?
23:10.03coolhpHow man of you use ael for extensions ?
23:10.36coolhpOr actually , is it worth it to migrate to ael ? is it the way of the future ? :-)
23:11.07nwhittwisted, yes
23:11.24*** join/#asterisk grimse (~grimse@p5481C7DB.dip.t-dialin.net)
23:12.01*** join/#asterisk tuxinator_linuxM (~tuxinator@216.132.53.3)
23:15.43Hmmhesaystwisted[asteria] yeah
23:16.10Hmmhesaysi'm now sitting in the hotel room a  bit bored
23:16.37NetgeeksI haven't tried AEL myself
23:17.40NetgeeksI have to re-write a dialplan thats about 5000 lines, maybe I'll try ael on it
23:17.59*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
23:18.22*** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net)
23:22.29twisted[asteria]Hmmhesays, did bkw, katty, and darth leave already?
23:26.34tsumeits so great to know people love to write unportable software
23:29.59tsumeWhy can't people have a good gui design? :) It must take a genious to follow one of the HID docs ou thtere.
23:31.33Netgeeksuh oh
23:31.47NetgeeksI'm writing a GUI, and not following an HID doc
23:35.24tsumeNetgeeks: well all you crappy devels suck :P
23:36.05*** join/#asterisk philm (~a@adsl-067-034-185-229.sip.mco.bellsouth.net)
23:39.02tsumebesides the fact which windows business are switching to linux. .net is a waste of effort until novell finishes the port of their .net to *nix.
23:39.41Cresl1ncoolhp:
23:39.44tsumebtw, its HIG, not HID :P  and heres a really good paper to read so apps don't look like complete crap. The document is portable across OSes to make apps look decent. http://developer.gnome.org/projects/gup/hig/2.0/
23:39.52Cresl1ncoolhp: you still there?
23:40.00Hmmhesaystwisted[asteria] they are here
23:40.04Hmmhesaysstaying one more night
23:41.06ManxPower*blink* *blink* *blink*  You mean there's a method to keep GUI application interfaces from being crap?
23:41.38tsumeManxPower: I guess Mark never followed it when creating gastman ;)
23:41.47Netgeekswell, my interface is developed in Zope, so I don't have to worry about the .NET problem
23:41.56tsumeNetgeeks: then you are awesome ;)
23:42.17tsumeNetgeeks: you still must follow web based HIGs to make it usable :P
23:42.24Netgeeksroger that
23:42.34tsumeone thing I hate are unportable apps.
23:42.49tsumeI want to run apps on *BSD and Linux most the time, _not_ windows
23:43.00Netgeeksfirst release will be ugly *sigh* but functional.  I'll have to work on pretty later
23:43.03*** join/#asterisk FryGuy- (fryguy@c-67-182-162-175.hsd1.ca.comcast.net)
23:43.40tsumehehe, well I need to finish the TL astrisk cmanager, my friend is paying me :)
23:43.44*** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com)
23:47.07jontowSipura SPA-841 opinion: i like.. :)  just talked to my dad on it for 20mins.. very clear and GOOD volume
23:47.10NetgeeksHow's it going, Craziman?
23:47.28NetgeeksYep, for the price, I like the 841
23:47.39jontowslight 'ocean' effect on the PSTN side with ilbc as the codec though
23:47.52*** join/#asterisk dooder (~nateputna@h-67-102-173-136.sttnwaho.covad.net)
23:47.56jontowim not sure what that might be; though i've got a fan spinning at good speed 8ft away from the mic
23:48.19hardwireyay
23:48.43hardwireconfiguring parts of sip.conf extensions.conf and voicemail.conf from one flatfile and a python script
23:49.12Netgeekspython good
23:49.17hardwireyes
23:49.24hardwirethe directory structure it sets up is neat
23:49.27rikstacan someone please tell me how to change the frequency of sipgate being registered and spamming up the console? thanks
23:49.29hardwirelet me pastebin it
23:53.39dooderwhat things should I check if I can make calls with my broadvoice account but can't recieve
23:53.42hardwireif pastebin will ever load
23:54.31hardwireNetgeeks: here.. http://pastebin.com/330350
23:54.43hardwirethe first file there phones.csv describes the system
23:54.56*** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com)
23:55.10hardwirethen the python script below that (two of them) create the layout for the spool dir (listed at top)
23:55.20hardwireeach of which have a voicemail,sip,extension file in them
23:55.31hardwireand then in spool/includes/extension it lists those include files
23:55.35hardwirefor jsut extension
23:55.45hardwireits all there in the pastebin
23:55.47hardwirenice and easy
23:55.53hardwireand doesn't take over the entier sip.conf
23:56.09hardwire#
23:56.09hardwire==> ./spool/includes/extension <==
23:56.09hardwire#
23:56.09hardwire#include "autoconfig/spool/snptours_anc/extension"
23:56.09hardwire#
23:56.12hardwire#include "autoconfig/spool/pbs_anc/extension"
23:56.12hardwire#
23:56.16hardwire#include "autoconfig/spool/tdxcorp_anc/extension"
23:56.18hardwire#
23:56.20hardwire#include "autoconfig/spool/tgsp_anc/extension"
23:56.26Netgeeksnice work
23:56.50*** join/#asterisk PBXtech-mobile (~upirc@66.209.31.29)
23:56.55NetgeeksI love the readability of python
23:57.20PBXtech-mobileis there a script for cisco phones that can disposition a call?
23:57.34hardwireNetgeeks: you must joke.
23:57.48hardwirenone of my code is readaable so you are joking
23:57.49dooderincomming calls aren't even hitting my asterisk box it looks like
23:58.20Netgeeksno joke, python just for me seems inherently more redable than php or perl
23:58.30hardwireNetgeeks: this is going to use that flatfile to configure the snoms as well from the metadata part of the phones.csv
23:58.48hardwireNetgeeks: I tried to use php after 2 years and failed
23:58.54hardwireperl I haven't touched in 4
23:59.00hardwireall I know is python
23:59.45hardwire#include "autoconfig/spool/includes/extension"
23:59.51hardwireto extensions.conf and is happy

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