irclog2html for #asterisk on 20050312

00:03.32_570RM_yo, what is the Dial command to initiate a call out of a isdn4linux card.i can already receive calls, but cant make them
00:06.08*** part/#asterisk santiago (~santiago@63.245.86.95)
00:07.38*** join/#asterisk MavvieSJH (~root@barnetworks-link.syd.comindico.com.au)
00:08.55MavvieSJHonly, after analyzing the HTTP packet, I wonder why they actually bother:
00:09.12MavvieSJHGET / HTTP/1.1..Host: 202.83.176.33:80
00:09.32MavvieSJHwhose idea was it to put an IP address for the hostname there?
00:09.57srtand including the port is funny, too ;)
00:10.07firestrmwhat is wrong with this dial command? exten =>1010,dial(IAX2/10@200.106.63.85,30)
00:10.19MavvieSJHbut it made me understand why they needed four computers instead of one to roll out the callmanager system.
00:10.34MavvieSJHsrt: hadn't even spotted that one yet.
00:10.34firestrmthas should work souldnt it?
00:10.48MavvieSJHfirestrm: no priority
00:11.18Juggieexten => _401X,1,Dial(IAX2/asterisk2:asterisk@192.168.0.52/${EXTEN}@iax,${DIALTIME},r)
00:11.38Juggiecompare :)
00:11.56puppetjuggie: hia
00:12.09puppetjuggie: i got basic webinterface up now ;D
00:12.30puppetjuggie: all incoming call canbe redirected to queue,and two upphones
00:12.32Juggiepatch any of the c stuff?
00:12.34puppetipphones
00:12.37*** join/#asterisk obelisque (~samifruit@70.48.19.123)
00:12.46obelisquehello!
00:12.52obelisqueHello guys!
00:12.56obelisqueand GIRLS!
00:12.57puppetjuggie: nah havent had to use that yet the apifunctions that are done are
00:13.18YoYo~seen bkw
00:13.20jbotbkw <~bkw@u201.udal.afb.lu.se> was last seen on IRC in channel #debian, 20d 12m 23s ago, saying: 'uhm, the latter one or both of them?'.
00:13.20YoYo~seen bkw_
00:13.21jbotbkw_ <~brian@bkw.developer.and.friend.of.asterisk> was last seen on IRC in channel #asterisk, 15h 12m 2s ago, saying: 'Delmar,where?'.
00:13.21puppetget_channels,get_channel,redirect,start_monitor,stop_monitor
00:13.25Juggiepuppet, spent some time talking to someone today, asterisk needs a whole new cli system
00:13.42puppetjuggie: iiik, does that mean that i have to recode?
00:13.55Juggieone where the command which determines the output isnt responsible for formatting it
00:14.06obelisqueAny good FREE iax2 client VIDEO for windows?
00:14.55Juggiepuppet, write a list of commands that would be good to have for the web gui, and i'll look at patching them to support concise
00:14.58srtMavvieSJH: what firmware are you running on your cisco?
00:15.00Juggiei need the same stuff...
00:15.27puppetjuggie: its really all commands :s
00:15.45puppetjuggie: we want to get dialplan in better format we want sip peers we want.. yeah :/
00:15.48Juggiepuppet, you better start learning C then because i aint patching every command ;)
00:15.58puppetjuggie: yeah i have to ;p
00:16.10Juggiei'll do what i consider important
00:16.26MavvieSJHsrt: I think it's TERM70.6-0-2SR1-0s or 7970_64060118
00:16.44puppetdialplan aint that important relative easy to just put the lines with Context at start in array on that way we get all arrays we can redirect people to
00:16.48puppetcontexts*
00:17.05puppetjuggie: wanne see my progress?
00:17.06MavvieSJHsrt: at this moment I suspect a bad configuration in the call manager, but I can't login to it since I don't know which username for the webinterface (I know the password :-)
00:17.19srt*gg*
00:17.38*** join/#asterisk Damin_Mobile (~pocketirc@79.sub-166-155-81.myvzw.com)
00:17.42srtmy 7960 with at least doesn't put the port there
00:17.57Damin_MobileBack in Cleveland...
00:18.07Damin_MobileJust landed
00:18.20Damin_MobileIt is CcOLD!
00:18.45*** join/#asterisk n3tar (~geno@201.254.93.202)
00:18.47n3tarhi
00:19.41Damin_Mobile#asterisk-bugs
00:20.04brc_hi Damin_Mobile
00:20.19srtMavvieSJH: hmm at least not if i dont explicitly include it in the url ;)
00:20.20firestrmdial(IAX2/10@200.106.63.85,30) still doesnt work.. i have iax.conf sent up on the remote machine, any idea where its going wrong?
00:20.42Damin_Mobilebrc: You up for doing some more bug hunting tonight?
00:20.43firestrmcomes back busy, and nothing appears on the remote machine
00:20.50brc_Damin_Mobile, sure
00:21.08MavvieSJHsrj: I suspect it's a configuration issue on the callmanager, because the guys who installed it refused to use DNS ("it only slows everything down")
00:21.09brc_Damin_Mobile, be a minute...on hold with united to reserve my cluecon ticket
00:21.20obelisqueDo you guys know any good FREE iax2 VIDEO client for windows?
00:21.20Damin_Mobilebrc: You have to go to Von in Boston
00:21.28brc_I do?
00:21.29brc_when
00:21.30brc_where
00:21.34brc_who's going
00:21.40Damin_Mobilevon.com...
00:21.45brc_why do people never tell me these things
00:21.49brc_who's going?
00:22.30Damin_Mobilebrc; I am still on the plane, so it will be much later tonight. byee for now...
00:22.37brc_Damin_Mobile, hangon a sec
00:22.49brc_Damin_Mobile, I've gotta book my ticket by 6pm
00:23.20Juggieinternet on the plane? :)
00:23.21Juggiegeek
00:23.33Juggietho i bought a train ticket to toronto in april in first class
00:23.41Juggieso i could have internet on the train
00:24.12pauldyman this is a mess does anyone have any tips for broadvoice to asterisk to softphone
00:24.21pauldyI can dial extensions form the softphone no problem
00:24.30Juggiecan you call out?
00:24.34pauldyfull audio everything
00:24.40pauldyI can dial to the softphone form my cell
00:24.42pauldyfull audio
00:24.52pauldybut when I dial from the softphone to my cell I can't hear anything
00:24.58Juggiedoes it connect?
00:24.59pauldyand the call timer on the softphone never starts up
00:25.22Juggiedoes your cell ring when u dial?
00:25.22pauldyso when I dial out from the softphone it rings my cell
00:25.32pauldybut I cannot hear anything when I pick up
00:25.56Juggiedoes the machine your sip phone is on have more then one ip?
00:26.17pauldylocalhost and the dhcp address
00:26.24pauldythats it
00:26.45Juggiewhat does the cli say?
00:26.48*** join/#asterisk stustu (~stustu@fluffy.fatburen.org)
00:27.08pauldycrom the console
00:27.16pauldyit says pretty much what I would expect it to
00:27.17Juggie?
00:27.21pauldycrom = from
00:27.28Juggiedid you try enabeling sip debug
00:27.30wildcard0hey.  WAY offtopic.  if anyone knows stuff about stacked LNB satellite stuff, please msg me :)
00:27.30wildcard0thanx
00:27.31stustuWhat's the differnece between "Transmitting" and "Transmitting reliably" in the SIP debug log?
00:27.36Juggieto see if there was anything odd
00:27.38pauldyyea the call looks fine
00:27.58Juggiehave u tried calling anything besides your cell?
00:28.06Juggienot that it should matter
00:29.38pauldyone thing wierd when I hang up i get Got SIP response 481 "Call/Transaction Does Not Exist" back
00:30.35Juggiecheck the wiki for broadvoice sounds like something is misconfigured
00:30.37Juggiebut i am not sure what
00:30.53opus___make sure your context is correct and your extensions.conf
00:30.56pauldyme either and the configs are all over the place most of them won't even allow incoming calls
00:31.03dougheckawhen passing caller id out
00:31.07dougheckado I include a 1?
00:31.28dougheckaeh, guess so
00:31.29doughecka:)
00:31.38MavvieSJHdoughecka : ,1,SetCallerId(1${CALLERIDNUM})
00:31.54dougheckawell, I think the proper way is without the 1
00:32.06pauldyyea right now the dial context is default for the softphone
00:32.07MavvieSJHoh, read it wrong.
00:32.10dougheckacause calling 18005558355 rejected it while removing the 1 took it :)
00:32.17dougheckahaha
00:32.21Matt-E-why, when i do an outside dial it rings once then nothing...?
00:33.06stustuAny experts on SIP authentication here?
00:34.07pauldyshould I have to worry with anything like TRUNK=
00:34.34*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
00:34.34*** mode/#asterisk [+o anthm] by ChanServ
00:35.40pauldyfound the problem
00:35.56firestrmAgiNamu, still here?
00:35.56pauldyone o fthe tuts said to set pedantic=yes
00:36.06pauldyI disabled that and now everything is working
00:36.06ManxPower~docs
00:36.07jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
00:38.33*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
00:42.23*** join/#asterisk jeffik (~jeffik@m2b7c36d0.tmodns.net)
00:45.04*** join/#asterisk miguellinux (~miguellin@200.47.223.190)
00:45.11*** join/#asterisk lindi- (~lindi@kulho150.adsl.netsonic.fi)
00:45.24Matt-E-why is the console answering my outgoing calls ?
00:45.34miguellinuxHi, I have problems with IAXtel
00:45.36YoYobecause you mucked up extensions.conf
00:45.50YoYothen talk to iaxtel ppl =D
00:45.51Matt-E-for example?
00:46.11miguellinuxit doesnt suppor ulaw?
00:46.49jeffikAnybody have any exprience wiht livevoip?
00:46.49miguellinuxit forces to other very crancky codecs
00:53.59*** join/#asterisk lqdengr (~fweston@69.172.205.68.cfl.res.rr.com)
00:56.32Darwin35Freeze Drop the mouse
00:56.47Darwin35hands off the keyboards
00:57.07Darwin35this is a mandatory 10 min break from typing.
00:57.24Darwin35step away from the pc and stretch and forage for food and fluids
00:58.20Eightthe #include command doesn't accept wildcards does it?
01:00.21Darwin35#include or include =>
01:00.26Eight#include
01:00.38Eightif it did, that'd be handy =)
01:02.49Darwin35<=== brain just went fart
01:03.04Darwin35man all this work in extensions.conf
01:03.12Darwin35everything should work now
01:04.21*** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3881624.sympatico.ca)
01:04.25DaLionanyone got radio working on with .. rawplayer ???.. loooking for a decent broadcast?
01:05.03*** join/#asterisk ACiDV (~acidvicio@122-68-181.dr.cgocable.ca)
01:06.48*** join/#asterisk tuxinator_linuxM (~tuxinator@m410e36d0.tmodns.net)
01:07.02ACiDVHi =) I try to dial multiple channel like: Dial(SIP/1000&SIP/1001&Local/819477XXXX@outgoing). The problem is that the Local/ call a remote IAX server and when it call, it reply with Call accepted and other devices dont ring. Does exist a way to bypass this ?
01:07.26KalD|Workok this is a long shot - but anyone here have an old nightowl subscription?
01:12.18*** join/#asterisk jeffik (~jeffik@m6d9836d0.tmodns.net)
01:15.31*** join/#asterisk Newbie___ (some@60.48.48.175)
01:15.46Newbie___hi
01:15.57Newbie___anyone heard of telextreme.com ?
01:19.05ariel_Newbie___, have you signed up with them?
01:19.11*** join/#asterisk Gator (~krp@ip24-250-198-83.ga.at.cox.net)
01:21.38*** join/#asterisk zapa (~zant@201.135.137.236)
01:22.05*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
01:23.07*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
01:23.42Newbie___ariel_: no, i tried calling my experience with them is scary
01:23.43zapahi all, anybody has expirence with echo cancel options in  MARK2 MARK3 STEVE or STEVE2 , i canīt resolve my echo troubles with my e1 any clue
01:23.58Newbie___i dont think that comapny even exists
01:24.08ariel_The rates at Broadvoice are better.
01:24.19Newbie___i am going for unlimted cell
01:24.25Newbie___ariel_: u signed up ?
01:25.13Newbie___ariel_: brodvoice do not accept non USA credit card so i have problem there
01:25.24denonzapa: its worth searching the wiki on echo problems .. tons of articles and good info
01:25.32ariel_no I have been thinking of BV but keep reading all the problems people keep having. So I am staying for now with my did from VoicePulse and ld from voipjet,nufone and livevoip.
01:26.31Newbie___tried ping to broadvoice server, i am getting like over 200-300 ms
01:26.48zapathanks denon, really i try a lot of configuration and only one side have echo troubles, thank again
01:27.14QwellNewbie___: I believe nufone accepts paypal
01:27.54Qwellnot sure if they have an unlimited plan though...
01:28.04Newbie___cheking out nufone now
01:28.52Newbie___it doesnt say anything about unlimted calls, so i assume there is none
01:29.16Qwellwell, how many minutes do you use a month?
01:29.32Newbie___about 5000+ world wide
01:29.42QwellThats a bit, yeah
01:29.49Qwellnot sure an unlimited plan would really cover you there, heh
01:30.23Newbie___broadvoice does, but initial ping result does not look good and the crdit card
01:30.27Newbie___problem
01:30.43Qwell"unlimited" doesn't mean quite what it sounds like it means
01:31.20EightQwell: at what point does BV start shouting "unreasonable usage"?
01:31.21*** join/#asterisk santiago (~santiago@63.245.86.95)
01:31.26Qwellgot me
01:31.31*** part/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
01:31.32Qwellsurely before 5000 minutes though
01:31.34|Vulture|~5k
01:31.45Newbie___Qwell: someone told me that, and i wrote to BV and they affirm me that unlimited is unlimted. no hidden charge
01:31.56QwellNewbie___: Get it in writing.
01:32.00|Vulture|Ive heard if your consistantly getting around 5k for 2 months your gone
01:32.18|Vulture|Newbie___: no hidden charge.. but they will dump the account
01:32.38Qwellmaybe dan2 will speak up
01:32.49Newbie___|Vulture|: yeah, BV said they have the right to terminate the account if they get 'suspicious'
01:33.16Qwellhonestly though, if you need 5000 minutes, an unlimited plan isn't right for you
01:33.34Newbie___Qwell: any recommendation ?
01:33.43QwellI'd still say nufone
01:34.09moonwickI'd say that if he needs 5000 minutes that an unlimited plan is perfect.  :P
01:34.22Qwellnah, any unlim plan will boot him for 5k
01:34.51Newbie___finding provider is one thing and voice clarity is another
01:34.57Newbie___wish i am in the USA
01:35.08QwellNewbie___: another great thing about non unlim providers, is that you can test them out first
01:35.17Qwelldrop $2-5, test it for a bit...see how it works out
01:35.42Newbie___Qwell: i agrre
01:35.45Newbie___agree
01:35.50Qwelland for the record, if you're going over an ocean(I assume you are), anything will be a little higher
01:36.22Newbie___how will the voice sound like if is 200-300 ms ?
01:36.31Qwellprobably as expected
01:36.54*** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com)
01:37.36Newbie___life is a bitch
01:37.47Qwellwhere are you hailing from?
01:38.09Newbie___Malaysia
01:39.25EightAnd you're trying to get cheap calls in the US?
01:39.29ACiDVDoes it's possible to dial (ring) multiple extensions that are connected on different servers ? like : Dial(IAX2/server1/ext1&IAX2/server1/ext2&IAX2/server2/ext1&...) ?
01:39.56Newbie___Eight: no, USA + other part of the world
01:40.03*** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se)
01:40.12EightNewbie___: I wouldn't worry about the BV ping, for calls to the US (and other part of the world)
01:40.13puppetthere :)
01:40.32EightNewbie___: it's just going to take a LONG time for the info to make the trip, wether it spends most of it on the 'net or most of it on some long haul carrier's phone network.
01:40.38puppetAnyone that live in scandianvia and wanne get rid of rackboxes? ;p
01:41.43Newbie___i was given from some provider * forgot the name* the clarity is good provided the voice reach me. speak from 1-10, 20% are missing
01:42.11Newbie___i mean free 0.25 credit to try out
01:43.14*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
01:46.29*** join/#asterisk xachen (justin@toto.citelnetworks.com)
01:46.56xachenI'm trying to develop a Dial Command for a extension so I can dial out onto the phone system
01:47.06xachenhow would I make it send a call if its prefixed with 9
01:47.16xachenI just need it to strip out the 9 and send the ${exten}
01:47.17Qwell${EXTEN:1}
01:47.21xachenthanks :)
01:49.35Eightalright, i'm going to settle once and for all how ast' needs to be configured for BV.
01:50.07Eightfromuser and authname aren't even valid values, and they're in the only config that worked for me =p
01:50.52WhiteWlfI'
01:51.14WhiteWlfIn my dialplan, how could I prompt for an extension then direct them to the approiate voicemail box?
01:51.46EightWhiteWlf: check out the macro in the default extensions.conf sample file.
01:51.54EightWhiteWlf: I don't know how it works, but I think that does what you want.
01:52.00Eightor similar.
01:52.03*** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211)
01:52.16WhiteWlfEight: the uhh... default dial thingy one that I can't remember the name of?
01:52.19Newbie___hmmmm net2hone is using SIP now
01:52.31SPoon_TSXHi out there, Just wondering how is your experience on buying VoIP Equitment off voipsupply.com?
01:52.56Newbie___i bought mine from voxilla
01:52.59WhiteWlfEight: macro-stdexten?
01:53.16SPoon_TSXvoxilla? Is it a Canadian or US company?
01:53.18QwellSPoon_TSX: The only time I tried, I had an issue...turned me off to them.
01:53.35EightWhiteWlf: that's what I had in mind. I've never looked at it though, so you're on your own from here =)
01:53.55SPoon_TSXWhere I can get the VoIP equitment in Canada?
01:54.16Newbie___SPoon_TSX: is a US company
01:54.23QwellSPoon_TSX: I'm convinced that many of these will ship to canada
01:54.34WhiteWlfEight: It's called with ${EXTEN}, and sends them there using that variable... but I thought that contained the exten you were calling from... no?
01:54.46SPoon_TSXQwell: Do have their website?
01:54.51EightSPoon_TSX: gah, y'know... there was one store that was listing prices in CAD and I almost didn't delete it from my list thinking "Maybe someone will ask for the link... naaah"
01:54.51Newbie___SPoon_TSX: they sent to me half way around the world
01:55.02QwellSPoon_TSX: dunno, voxilla.com, I'd imagine
01:55.02EightSPoon_TSX: that was last night... There *is* one though.
01:55.23SPoon_TSXEight: May i have the web address?
01:55.31SPoon_TSXQwell: Thanks.
01:55.33EightSPoon_TSX: I was just saying, I deleted it last night.
01:55.34Newbie___http://www.voxilla.com/
01:55.46SPoon_TSXEight: O, is okay.
01:59.01EightIt's amazing to me there are more, and better, hardware SIP phones than software.
01:59.12WhiteWlfEight: Same here
01:59.13EightSeems so backwards from the norm.
01:59.23QwellI'd think there would be more hardware ones
01:59.45WhiteWlfIt's easier to make a piece of software than hardware.
02:00.03WhiteWlfMore so, a piece of hardware that runs software.
02:00.03Qwellsure, but who's gonna buy a softphone?
02:00.15EightWho said anything about buy?
02:00.17NuggetI own two eyebeam licenses.
02:00.17WhiteWlfwho's going to buy Asterisk?
02:00.27QwellNugget: I didn't quite mean it like that. :P
02:00.43QwellI mean...is the general public more likely to buy a softphone, or hardphone?
02:00.44WhiteWlfGood software doesn't mean it costs...
02:01.11WhiteWlfmmm... hardware
02:01.13QwellI'm thinking more from a "my company only exists to make money" point of view
02:03.07Eightanyway to kick Ast' into re-registering?
02:03.41*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
02:04.38*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
02:06.22WhiteWlfI think reloading re-registers
02:07.17NukemizerI want to link 2 Asterisk boxes for calling between systems. To help me seearch for my answers can anyone tell me what the acronym to describe * networking ? in my PBX world it is called QSIG
02:07.57Nukemizeror is it just that simple "asterisk networking" ?
02:09.16*** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3884408.sympatico.ca)
02:09.25DaLion<PROTECTED>
02:09.25DaLion<PROTECTED>
02:10.21shido6show lag?
02:10.33shido6sip show peers if you have a qualify statement set in sip.conf
02:10.33DaLionwell i want to see channel quiality
02:10.51DaLion<PROTECTED>
02:11.14DaLionhmm
02:11.20DaLionthat not the call quality
02:11.31DaLiontring to see if QOS working
02:12.02*** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com)
02:13.52Eighter, non-root.
02:14.25Eightoooh, I kinda like the new 2.6 device system... /dev/ is actually readable!
02:15.54godsmokeI'm getting this strange behavior when trying upgrade my cisco 7960 -- since it's not asterisk related, if someone could help, pm me
02:16.09*** join/#asterisk jeffik (~jeffik@m919f36d0.tmodns.net)
02:17.08Eightand I broke BV again.
02:22.25EssobiSomewhat readable.
02:23.11*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
02:23.42Newbie___hey Essobi
02:26.43*** join/#asterisk bjohnson (~bjohnson@ip137-172.dsl.istop.com)
02:32.42Newbie___hi bjohnson
02:33.18Newbie___bjohnson: u once told me to get SPA 2000, and is been working great
02:33.50*** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net)
02:34.02blaisen1anyone know where to get toronto DIDs?
02:34.38EssobiUmm.
02:34.51EssobiBell Canada?
02:35.29blaisen1well i meant someone doing SIP or IAX origination
02:35.35blaisen1but yeah i guess they'd work
02:35.39*** join/#asterisk DaLion (DaLion@Quebec-HSE-ppp224769.qc.sympatico.ca)
02:35.50DaLionhey
02:35.50DaLionQOS on sveasoft works great !
02:36.18nine76x100p's here,rebooting/installing. bye
02:36.28blaisen1hey, anyone know why i have two mpg123 processes for my music on hold (coming from a radio station's mp3 stream)?
02:36.37DaLionyes
02:36.43DaLionmpg123 has a multi thread bug
02:36.47DaLionuse something else
02:36.53blaisen1like?
02:36.57DaLionlet me check
02:37.03DaLionwhat station u using btw ?
02:37.12blaisen1LIFE 100.3 FM Barrie, Ont
02:37.29DaLiondefault => custom:/var/lib/asterisk/mohmp3_raw,/usr/bin/rawplayer
02:37.35DaLionblaisne whats the link ?
02:37.51DaLioncheck wiki for compiling.. thinks its...
02:37.58DaLion~rawplayer
02:38.06blaisen1www.fm100.net
02:38.09blaisen1rawplayer?
02:38.12DaLionyeah
02:38.38*** join/#asterisk Damin_Mobile (~pocketirc@242.sub-166-155-102.myvzw.com)
02:38.39godsmokefor some reason -- my cisco 7960 tries to load a file called P0S3-07-.bin
02:38.43godsmokeanyone seen this before?
02:38.53DaLionhy.. no.. no cisco here ;(
02:38.55PatrickDKya, that is normal
02:39.10DaLionit tftpinh
02:39.10godsmokePatrickDK: it can't be -- that file shouldn't exist, and doesn't
02:39.14dan2Qwell: if your burning 5000 minutes a month you'll be terminated if you are running a residential plan
02:39.16DaLiontftping the bin..
02:39.22*** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
02:39.28godsmokeDaLion: heh -- forget it -- you don't understand
02:39.48DaLionah.. lol
02:39.52Nuggetyeah, that filename doesn't look right
02:39.56PatrickDKhmm, maybe it forgot it's mac?
02:39.58blaisen1dalion: rawplayer will play shoutcast/mp3 streams?
02:39.58godsmokeof course not
02:40.05godsmokeno, this has nothing to do with its mac address
02:40.11godsmokeit does this before trying to load the .cnf
02:40.14Damin_Mobilegodsmoke: Look at your tftp config files.
02:40.24PatrickDKheh, it's been awhile since I loaded the 7960
02:40.37DaLion~rtfm
02:40.38jbotrtfm is, like, read the f*cking manual... try asking me about "FAQ"
02:40.47DaLion~faq
02:40.48firestrmanyone know what mightMar 12 10:51:28 NOTICE[5023]: chan_iax2.c:5444 socket_read: Rejected connect attempt from 24.68.44.53
02:40.48firestrmTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REJECT
02:40.48firestrm<PROTECTED>
02:40.48firestrm<PROTECTED>
02:40.58DaLion~tiki
02:41.00blaisen1dalion: I can't find where to download rawpalyer..?
02:41.08DaLionim trying to find tikiwiki addy
02:41.10DaLionlol
02:41.14DaLion~wiki
02:41.36DaLion~asterisk
02:41.37jbothmm... asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org
02:41.52Damin_Mobilefirestrm; rejected connect attempt due to wrong credentials
02:42.13*** join/#asterisk bjohnson_ (~bjohnson@ip137-172.dsl.istop.com)
02:42.23DaLionok hold on
02:42.29blaisen1hmmm... looks like rawplayer won't decode mp3
02:42.41DaLionhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20faking%20it
02:42.42blaisen1so it will probably not work in my application (receiving an mp3 stream from a radio station)
02:42.54Qwelldan2: Thanks.
02:43.11Qwelldan2: How much lower is the "limit"?
02:43.17firestrmim having problems connecting 2 * boxes together.. anyone know what might be happining here? http://pastebin.ca/7282
02:43.35firestrmsomthing is breaking in iax, but i dont know what
02:44.38DaLionah true
02:45.51blaisen1well it looks like it is only downloading 1 stream, so the second instance may be harmless
02:46.12blaisen1i was worried it would be using twice the bandwidth maintaining two connections to the radio station
02:46.19bjohnson_ManxPower: earlier you asked my DID provider for SW Ontario .. sixtel
02:46.32blaisen1sixtel is sloooooooooooooooow at doing anything
02:46.42Damin_Mobilefirestrm: http://lists.digium.com/pipermail/asterisk-users/2004-August/057681.html
02:46.45shido6wow
02:46.51blaisen1like adding new ontario dids
02:46.53shido6SW ontario, eh?
02:46.58blaisen1or cancelling stuff they say they've already ancelled
02:47.13shido6how many dids and where do u need them?
02:47.41blaisen1toronto/london/kitchener/ottawa and if possible windsor/hamilton/st catherines
02:47.59blaisen1it took me over 3 weeks to get a london ont did with sixtel, and even then it didn't work
02:48.10blaisen1it looks like its going to some cell phone calling card reseller.. wtf...?
02:48.27Essobinice
02:48.34blaisen1i would ixnay on the sixtelay
02:48.47Essobiixstelay
02:48.50blaisen1it is such a pita to find a good canadian DID/800 provider
02:49.03EssobiHmm.
02:49.06blaisen1i have service with livevoip which is good but the fact that they change their website and offerings on an hourly basis makes me nervous
02:49.11EssobiI'll ask at work monday.
02:49.20EssobiHow many DIDs you interested in?
02:49.31Eightgrrr, anyone know what file permissions I might be overlooking that would prevent a remote connection to asterisk from the shell?
02:49.33blaisen1when i signed up i was under the impression canadian 800 originatoin was 1.29 cents/min now i'm told its 5 cents/min USD! i can get allstream 800 orgin to pstn for 4.25 cents
02:49.36blaisen1and sprint for 3.5
02:49.38bjohnson_shido6: you have yours yet?
02:49.43EightI have already looked through this: http://www.voip-info.org/wiki-Asterisk+non-root
02:49.46Essobi5 cents USD?
02:49.50EssobiYou're getting raped.
02:49.53blaisen1essobi: no doubt
02:50.09*** join/#asterisk BoboTWF (~joshuas-a@rrcs-66-27-57-228.west.biz.rr.com)
02:50.28EssobiI guess it might depends on how many DIDs you want too thou.
02:50.32blaisen1ok so where in the heck do i get ontario DIDs for a reasonable price and canadian 800 origination for sane prices that SHOULD be below what the telcos will offer
02:50.46EssobiWe got US50 at work.
02:50.46bjohnson_Essobi: that is cheap for CDN 800 voip number
02:50.55blaisen1well we want to resell service to our ADSL customers but right now I'm concerned about all this crtc reg bs
02:50.56EssobiI dunno if there is anything in the works for canada.
02:50.59EightGrrrr, it's still putting its PID file in /var/run/ not /var/run/asterisk
02:51.02EssobiI know we cover some of mexico too.
02:51.05bjohnson_Essobi: exactly
02:51.24Essobi5 cents USD for a canadian 800?
02:51.28bjohnson_Essobi: everything that covers canada is more expensive for voip
02:51.30bjohnson_yes
02:51.37EssobiRock. I'm moving to canada.
02:51.39bjohnson_shido6: London and Kitchener
02:51.52blaisen1so how can sprint canada/callnet do 3.5 cents/min canadian orig?
02:52.02blaisen1and thats canadian $$ not US
02:52.03bjohnson_gotta go .. be back in 15 minutes
02:52.18shido6how many? shoot requirements over to shido6@gmail.com
02:52.28shido6cdn dids and 8xx
02:52.55firestrmok here is my iax config on the two machines http://pastebin.ca/7284 any one know why its not accepting credentials?
02:53.00EssobiSo what company do you exactly hustle for Shido?
02:53.33*** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com)
02:54.43blaisen1i've seen shido6@gmail.com on calltermination.com
02:55.18blaisen1i have no volume right now because i have no reliable carrier to work some packages around for my potential customers
02:55.36blaisen1so i'm just looking for reasonable rates with potential for discounts based on volume
02:56.05blaisen1i found unlimitel which seemed like a good deal.. until you realize they bill by the minute
02:56.06_570RM_hmmm
02:56.18_570RM_does anyone see why this line is not working:
02:56.21_570RM_exten => _0.,1,Dial(Modem/g1:${EXTEN:1})
02:56.40_570RM_group 1 is a BRI passive i4l card
02:57.20*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
02:57.41_570RM_it can receive calls alright, but i cant get it to dial out
02:58.33firestrmdo i have to set a username /password or will static ip do?
02:58.38firestrmin IAX
03:00.32modulus_yeah i got your IAX right here baby
03:00.58firestrmmodulus_, thanks :~
03:01.06Eight@#$# @#$# @##$
03:01.20firestrmEight, i know how you feel
03:01.41Eightit keeps dropping the pid file in /var/run/, not /var/run/asterisk/!
03:01.46Eightor, rather, failing to.
03:01.49firestrmEight, i think swearing is allowed here, it part of asterisk experince :)
03:02.46firestrmmodulus_, any ideas where my config is going wrong?
03:03.12firestrmim allmost understanding IAX config, but im missing somthing..
03:05.29bjohnson_shido6: I signed up for 1 London and 1 Kitchener at sixtel on Feb 17.  The London one I got about a week later.  I'm still waiting for Kitchener with them
03:05.44Eightasterisk.h:#define AST_PID              ASTVARRUNDIR "/asterisk.pid"
03:06.03EightMakefile:ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk
03:06.15bjohnson_shido6: both would be low usage .. likely about 30 minutes per month to start .. maybe up to 200 each per month after a year
03:06.19Eight[blake@star run]$ ls /var/run/asterisk.pid
03:06.27Eight<PROTECTED>
03:06.40Eightwhat the hell?
03:06.59jeffikbjohnson_ may i ask what DID provider you are using?
03:07.21bjohnson_jeffik: sixtel
03:07.31bjohnson_jeffik: sorry .. iax.cc
03:07.44bjohnson_jeffik: sorry .. iax.cc = sixtel
03:08.14jeffikI'm looking for a provide for Toronoto DID s
03:08.28bjohnson_there's a bunch that do Toronto
03:08.35bjohnson_even some that ONLY do Toronto
03:08.59jeffiki saw but could only get a response from unlimitel
03:09.13bjohnson_I don't remember there names because I was looking for other areas
03:09.19jeffiki's for my own asterisk@home
03:09.24*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
03:10.26bjohnson_one guy at comcast told me they could hook up a SIP connection for me but another guy said no .. I decided I wasn't interested since they only had major cities
03:11.04firestrmarrrgh IAX blows!
03:11.05DaLionNo one do Quebec Province..
03:11.52jeffikok, well i gotta get someone other than livevoip my number has not worked rihght in weeks and they blame it on their canadian provider
03:12.09bjohnson_I think someone in here about 2 months ago found 405 DIDs
03:12.24SyncrosDaLion ?
03:12.27firestrmwhy cant they give a )($#*)# proper example of how to use iax between two machines anywhere. F*&K this pisses me off
03:12.39bjohnson_firestrm: it's no big deal
03:12.52bjohnson_firestrm: when you get it working make sure you document it then
03:13.07firestrmbjohnson_, is is for me.. ive been trying to get it working for 3 hours
03:13.36bjohnson_firestrm: set up a friend on each side .. use the same username, secret, and section name on both machines
03:13.47jontowwhat about the 'asterisk dual servers' entry on the wiki? does that have one?
03:14.38bjohnson_firestrm: if one has a dynamic IP .. it has to have a register command.  If one has a static IP .. it doesn't need a register but the oppisite machine needs a host= line in iax.conf
03:14.39*** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg)
03:15.01firestrmbjohnson_ http://pastebin.ca/7284, im using static ip's so no pass/user
03:15.29jontowthat doesn't make much sense
03:16.05firestrmbjohnson_, i have them set up as peer though
03:16.18jeffikbjohnson_: sorry, are you using sixtel now?
03:18.14bjohnson_firestrm: start with friend .. peer is for sending only
03:18.25firestrmbjohnson_, i keep getting no athority found..
03:18.39bjohnson_jeffik: yes .. for one DID .. been waiting since Feb 17 for the second one
03:18.54firestrmbjohnson_, changed to friend, still same result
03:19.08jeffikok, cause i see they offer 416 numbers
03:19.53jontowyeah.. set them up as friend, put a user:pass in there
03:20.37EightThis is insane... There is something very fragile about BV.
03:20.39jontowi have many machines working IAX to IAX
03:20.48EightI've been fiddling with this config file for hours now and I still haven't identified it.
03:21.01EightI went back to my 'known working', and now it doesn't work either.
03:21.30bjohnson_firestrm: did you read this? http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication
03:21.54bjohnson_firestrm: I have it working but am using the same username, secret, and section names on both servers
03:22.42opus___eight - yeah
03:22.55bjohnson_firestrm: adding username and secret info will not add or detract from security .. I just know it works
03:23.11opus___eight - curious, what do you notice right now thats strage?
03:23.24opus___strange even
03:23.31Eightopus___: outgoing calls seem to work when expected...
03:23.37Eightopus___: but incoming are flakey as all hell.
03:23.45firestrmbjohnson_, yes but im barely grasping the concepts as it is, usr/pass confuses me even more..
03:24.00bjohnson_firestrm: then just do it
03:24.35opus___eight - shit, maybe we should all go in on a T1
03:24.42*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
03:24.46bjohnson_firestrm: you can do what works and figure it out later .. or try to figure out why the one you have doesn't work (and doesn;t match the config of 2 people with working systems)
03:24.55Eightopus___: I thought we had, and it was called Broadvoice =p
03:25.20jontowi have a freebsd machine with 4-7 IAX connections to various other machines (with differing OS'/distributions of linux)
03:25.32opus___So basically they just went around to each state and colocated a T1 box.. hmmm
03:25.46Eightopus___: yu'up.
03:25.47jontowi can put the config for a pair of them online if you want.
03:26.08heison<PROTECTED>
03:26.31opus___at $600 per t1 card, $1600 1U ibm machine, $200 per T1 = $2400 costs plus $400 /month for rackspace and T1 ...
03:26.37MikeJ[Jayden]heison, you just copied and pasted that from 5 hrs ago.
03:26.40heisonturns out cisco3 does not register with asterisk upon reboots
03:26.54jontowheison; you need to check hardware settings..
03:26.55Eightopus___: you apparently know the numbers better than I do.
03:26.56MikeJ[Jayden]there you go
03:26.57heisonMikeJ[Jayden]: yes, almost
03:27.03MikeJ[Jayden]hehe
03:27.05Eightopus___: but ya, if you want to go ahead and do that, start in Minneapolis =)
03:27.12opus___hehe
03:27.17Eightopus___: I've been thinking about it...
03:27.17heisonhow do i force the 7960 to register?
03:27.19*** join/#asterisk warmfeet (~c@213.78.240.109)
03:27.31heisonjontow: can u be more specific?
03:27.34opus___Eight - it sucks that we'd have to colocate in each city with low capital.
03:27.41jontowheison; SIPmacaddr.cnf is where you'd have to set all that
03:27.44firestrmbjohnson_, what does the dial format look like for that. i want to dial remote extension 10 username vince pass blah
03:28.00opus___i wish you could like VPN to other state's T1 lines..
03:28.04jontowmake user and [name] the same..
03:28.07jontowie.
03:28.07Eightopus___: well, you don't HAVE to collocate in each city... but you pay connection fees otherwise.
03:28.09jontow[iaxlink]
03:28.13jontowuser=iaxlink
03:28.15jontowsecret=blah
03:28.18jontowtype=friend
03:28.23warmfeetIs it possible for asterisk to create the voicemail INBOX and paths when first voicemail comes, rather than precreating them
03:28.27opus___i don't fully understand the telcom industry
03:28.27jontowhost=XXX.YYY.ZZZ.QQQ
03:28.35jontow...
03:28.47jontowthen in the dialplan, set IAXLINK=IAX2/iaxlink@iaxlink
03:29.07Eightopus___: it's old, bureaucratic, heavily regulated, and just kinda funky.
03:29.11heisonjontow: and what param forces the phone to register if it doesn't do it by itself
03:29.13firestrmjontow on which end..
03:29.15jontowexten => n,1,Dial(${IAXLINK}/${EXTEN})
03:29.22jontowboth ends
03:29.25warmfeetand does neone has exp with MWI,it seems that u need to hv individual users in sip.conf...which I dont since I have them in SER
03:29.27*** part/#asterisk DaLion (DaLion@Quebec-HSE-ppp224769.qc.sympatico.ca)
03:29.40jontowif you want to make calls from either end, and receive them on the other, that is..
03:30.17bjohnson_firestrm: I just use dial(IAX2/remoteserver/${exten})
03:30.31jontowand heison, check the firmware revisions.. what are the phones running?
03:30.32bjohnson_where remoteserver matches the section name in iax.conf
03:30.45heisonjontow: they are all SIP 7.3
03:30.49jontowhmm
03:31.06jontowheison; if they can't contact the tftp server they don't register
03:31.16jontowbut they WILL be able to make calls, as is the nature of SIP (doesn't need a proxy, per se.)
03:31.22EightAnd now BV is working again, but I didn't change anything...
03:31.34jontowthats why i said.. check the hardware (network) settings.
03:31.35firestrmjontow, dont you need a host= somwhere in there? otherwhise how can it find the other server?
03:31.39heisonthey do contact the tftp server, but doesn't register with *
03:31.53*** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net)
03:31.57jontowfirestrm; re-read my 'paste' .. :)
03:32.05heisonanyone here using SIP 7.x?
03:32.12jontowthose aren't complete entries.. let me just put mine up, it'll make this hours less of a discussion :D
03:32.14heisonon 7960s?
03:32.26jontowheison; i am.. i had 13 of them running 7.3 with no problems except slow bootup time.
03:32.34alexnsi would be if i could get sip 7.x
03:32.45heisonjontow: do you see a time stamp on your phone?
03:32.49jontowyes
03:32.52jontowthats another network feature
03:32.58heisoni don't have those anymore...
03:32.59firestrmjontow, oops there it was a few lines down :)
03:33.07jontowmake sure you've got the netmask and gateway setup correctly.. your problems are ALL in your network settings
03:33.19heisonjontow: during the swapping of hardware, i lost my SIPDefault.cnf
03:33.21jontowit needs proper DNS and gateway information
03:33.33jontowwant a sample, while i'm pulling config files?
03:33.37heisonjonton: can you show me your SIPDefault.cnf?
03:33.45heisonjontow: yes please
03:33.51jontow;)  yep, gimme a bit
03:33.55BaconHowdy.
03:33.55heisonthanks
03:34.08BaconI'm trying to setup Asterisk with BroadVoice.
03:34.20BaconI have inbound working, but not outbound...
03:34.39BaconMar 11 23:28:25 VERBOSE[1487]:     -- Got SIP response 404 "Not Found" back from 147.135.8.128
03:35.00BaconI don't know enough about asterisk to start debugging...
03:35.04BaconAny suggestions?
03:36.08opus___bacon - what is your outgoing configuration
03:36.13opus___brb reboot
03:36.57firestrmjontow, bjohnson_, that worked!! once again im in your debt.. i sure hope we meet some time so i can repay you guys in the canadian beverage of choice (beer)
03:37.34firestrmwhere i was going wring was i was trying to send the extension where the username is supposed to go..
03:37.37bjohnson_firestrm: I suspect you could do it your way and the answer lies somwhere in http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication
03:37.42*** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net)
03:37.47bjohnson_but that's not how I did it
03:38.03bjohnson_(ie the way you were unsuccessfully trying)
03:38.05Dandanheh, anyone had any success with MPG321?
03:38.10firestrmbjohnson_, at this point im just glad it worked..
03:38.12bjohnson_firestrm: now edit the wiki page
03:38.19Dandaninstead of security crippled mpg123 ?
03:38.19firestrmbjohnson_, i WILL!!
03:38.27bjohnson_Dandan: no
03:38.49jontowfirestrm; http://mno.bsd.st/~jontow/2005-03-11/iax.conf.txt
03:39.02jontowsorry its a little late
03:39.09jontowbut thats my example.. :)
03:39.29Dandanbjohnson_: maybe asterisk community should look into incorporating mpg321
03:39.37Dandanwhich is still under development...
03:39.38Baconopus___: You rebooted yet?
03:39.40Dandanand maintained
03:39.45opus___yup
03:40.06jeffikbjohnson_: using asterisk@home and wonder how difficul to get DID from sixel running
03:40.07opus___Bacon - just to let you know I haven't got mine to work either, but I'm working on it now
03:40.22firestrmjontow, where i was going wrong was i was passing the extension where the username was .. ie dial(iax/10@iaxconnect,30) rather than dial(iax/iaxconnect/10).
03:40.26Baconopus___: What do you have working?
03:40.35Baconopus___: Like I said, I have inbound working...
03:40.46firestrmit was a dumbass mistake that was invisible die to it being right infront of me..
03:41.43firestrmbut im going to go throw up a wiki page especially on the subject now.. mind if i use your example as one of the ways to do it?
03:41.57jontowand heison; http://mno.bsd.st/~jontow/2005-03-11/SIPDefault.cnf
03:42.09jontowfor the place to specify the server, look for "192.168.2.1" as a string
03:42.40jontowkeep in mind that config file sets 'telnet_level' to 2 (privileged mode)
03:42.55jontowyou need to get into the settings on the phone itself before you may get it all working
03:43.19bjohnson_jeffik: easy if you edit the config files .. I don;t know the gui
03:43.31jontowgo for it, don't even care if you mention me :)
03:43.41firestrmjontow, mind if i use you example for the wiki?
03:44.03bjohnson_firestrm: add to the dual server page .. don't start a new one
03:44.27opus___bacon - i just have inbound working
03:44.31firestrmbjohnson_ i'll try to find that one..
03:44.31opus___lemme post my config
03:44.37opus___for outgoing, ...
03:44.55heisonjontow: i found a couple problem already...
03:44.58channanhi, anyone's familiar with dialing to Toluca, Mexico? I've tried to call an old friend but automated operator announced busy tone all the time
03:45.08heisonproxy_register, sntp_server
03:45.11channanthe number is: 011-52-72 xxxxx
03:45.42channanI looked at from the web and it seemd to change to 011-52-722, but still did not work
03:45.46jontowheison ;)
03:45.51jeffikbjohnosn_: well you can edit the files direcly or use the gui. all i need is the settngs, i'm asking as i had a less than possitive experience wiht livevoip.com
03:46.20heisonit's taking forever to reboot... but will see that's my only problem, thanks.
03:46.34bjohnson_they give you the setting when you get a DID
03:46.55firestrmbjohnson_, i think i found it, do you mean this page ? http://www.voip-info.org/tiki-index.php?page=Asterisk%20-%20dual%20servers#comments
03:46.57heisonjontow: the time is back, and i no longer have a "x" besides the extension
03:47.04jontow;)
03:47.04opus___bacon http://pastebin.ca/7287
03:47.10bjohnson_they even substitute your username, password, and DID into the config examples (they're dynamically created)
03:47.23bjohnson_firestrm: yes
03:47.31heisonand Asterisk now sees the phone as registered!
03:47.34heisonthanks man
03:47.36firestrmok.. i will add my bit there..
03:47.37jontownp
03:48.02heisoni guess i took it for granted without looking thru the .cnf file before.
03:48.05jeffikbjohnson_: thanks, seems like they are worth a try
03:48.22opus___bacon - when you run asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvgc you can see the output when it tries to call, I updated that url with my error
03:48.40opus___<PROTECTED>
03:48.43modulus_how about asterisk -vvvvvvvvvvvvvvvvvvvvvvvvgc?
03:48.43jontowheison; btw, those are hacked up examples based on the ones that are available at cisco.com (if you dig REAAAALLLLY deep and have an account)
03:48.43opus___was what it said
03:48.44mikegrb...
03:48.59*** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-18-150.w81-248.abo.wanadoo.fr)
03:49.05mikegrbI run asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgc so I can be cool like opus___
03:49.05opus___modulus is there a better way to do it
03:49.13Baconopus___: You should have host=proxy.lax.broadvoice.com
03:49.21heisonjontow: i got my account about 3 hrs ago, and yes i have seen something similiar on their site
03:49.35jontow:)
03:49.45opus___bacon - that one seems to go down, but let me try it again
03:50.33Baconopus___: Are you behind a nat?
03:50.44opus___no, but my phone is
03:51.01Baconopus___: Is your phone working with your asterisk box ok?
03:51.03*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
03:51.06opus___yes
03:51.31opus___proxy.lax won't let me register, again...
03:52.15Baconopus___: Have you been here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Broadvoice
03:52.28opus___yes
03:53.17jeffikopus___: i am using x-lite begind a wifi router, can i access my asteisk without opening ports on the router using nat?
03:53.33opus___I am able to call in while dialing out, but dialing out gives me the  'nobody picked up in 30000 ms'
03:54.05opus___jeffik - I think so
03:54.55jeffikopus___: what do i need to set?  and do i need cooresponding settings on asterisk?
03:55.43*** join/#asterisk JmanA9 (~josh@pa-murraysville2b-141.pit.adelphia.net)
03:55.51JmanA9hello
03:56.04JmanA9when i'm running asterisk for the first time, i get this error message in the console and it doesn't start up:
03:56.16JmanA9[pbx_loopback.so]Mar 11 22:55:20 WARNING[21616]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead
03:56.16JmanA9Mar 11 22:55:20 WARNING[21616]: loader.c:440 load_modules: Loading module pbx_loopback.so failed!
03:56.24JmanA9google's turning up nothing, anyone know what to do?
03:56.45jontowjmana9; how did you install? compile from source? binary? where'd you get the tarball? cvs?
03:57.07JmanA9i got the tarball from the cvs
03:57.24Eightuh, what?
03:57.26JmanA9just did make clean and make samples
03:57.44BaconMar 11 23:51:03 DEBUG[1487]: Stopping retransmission on '20596ebe59e1e1a8538812d3222448f0@sip.broadvoice.com' of Request 102: Found
03:57.46jontowwhat about 'make install' ?
03:57.56jontowand 'make' for that matter?
03:58.03BaconMar 11 23:51:03 VERBOSE[1487]:     -- Got SIP response 480 "Temporarily Not Available" back from 147.135.12.128
03:58.16opus___jeffik - here is an example config http://pastebin.ca/7290
03:58.27JmanA9i think i may have forgotten to do a make install....
03:58.30JmanA9i'm such an idiot
03:58.35jontow;)
03:58.36JmanA9lol
03:58.56*** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com)
03:58.57opus___jeffik - the key is for sip.conf/[general] to have nat=yes externip=yadayadayad and for your sip device [xlite]/nat=yes/qualify=yes
03:59.54*** join/#asterisk jets (~jetsn@xyharley.dsl.xmission.com)
03:59.55opus___jeffik - i've never used xlite, there might be additional special lines needed in its sip entry. I posted how i got sjphone to work
04:01.04opus___bacon, any luck?
04:01.41*** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net)
04:02.37*** part/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com)
04:03.16jeffikopus___: thnanks, this is a good start i will try it
04:03.59*** join/#asterisk jsolares (~jsolares@200.12.33.64)
04:04.38opus___hmmm, eyebeam
04:06.13jeffikopus___: you like sjphone?
04:07.05opus___jeffik - yes
04:07.35jeffikopus___: are you in Canada?
04:07.37opus___simple, works, easy to setup. not much debugging info/or I haven't really looked
04:07.44opus___jeffik - usa oregon
04:07.55warmfeetanyone good with perl regex here....I need to match sip.domain.com in vmail.cgi as opposed to having a context with no '.'
04:08.17warmfeetwS I guess
04:08.34warmfeeti mean ~\wS$
04:08.40jeffikopus___: Toronto/Chicago
04:09.28opus___jeff cool.
04:09.39*** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
04:10.00JmanA9well, i recomplied everything twice, i'm still getting that error
04:10.14opus___jeffik - i'm looking at https://sip-communicator.dev.java.net/download.html right now, apparently it does video with asterisk
04:10.33JmanA9i obtained everything from the cvs, did a make clean ; make install for everything, also did a make samples for asterisk :/
04:11.17jeffikopus___: would be nice but got to get my asteisk@home running with Toroto/Chicago DIDs and outgoing
04:11.32opus___jman - try removing /var/lib/asterisk
04:11.35opus___or whatever
04:12.12*** join/#asterisk Jabreity (~jfkdlsjk@12-222-3-81.client.insightBB.com)
04:12.20JmanA9ok, i'll try that
04:12.36JmanA9still no good :(
04:16.31Eighthttp://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice
04:16.39EightI've added the 'second example' on that page.
04:16.49EightI think it clarifies some things somewhat.
04:19.22*** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
04:21.10Jabreityhowdy all
04:21.38*** join/#asterisk afrosheen (~afro@c-67-166-172-141.client.comcast.net)
04:21.41afrosheenhey gang
04:24.05afrosheenlively bunch tonight eh
04:24.12Groobyaye
04:24.13afrosheenthat must mean everything is working right
04:24.13jsolaresvery
04:24.25afrosheenanyone tackle VOIP payphones before?
04:24.58jsolaresnope
04:25.07afrosheenI may have a huge project on my hands if I can figure out a nice way of implementing it
04:25.13JmanA9any known issues with fc3 and this error i'm getting?
04:25.54afrosheenyes
04:26.03jsolaresafrosheen: have you looked into analogue payphones but connected to an iaxy?
04:26.04JmanA9have anything to do with udev?
04:26.11Baconopus___: I was afk for a bit.
04:26.15Baconopus___: Any luck?
04:26.16Jabreityhey, i have what i think is a valid question
04:26.27afrosheenjsolares: yeah and I did some thinking on it, it seems like a good idea for a number of reasons
04:26.33BaconThere are no invalid questions, only invalid people.
04:26.40Jabreity:)
04:26.43jsolareshehe
04:26.45_6Flamez_jman: v1 or HEAD?
04:27.00JmanA9i have no clue, whatever comes with fc3 :/
04:27.02afrosheenjsolares: reason one: keep the analog lines from the telco closet to the phones = no rewiring
04:27.16Jabreityok, its gotta be a simple config err somewhere, but i have a handfull of extensions setup
04:27.21_6Flamez_what * version?
04:27.29afrosheenjsolares: reason two: with the iaxy's in the telco closet, they can easily plug into the UPS for failsafe-ness
04:27.32jsolaresafrosheen: i have no experience whatsoever with payphones be it analog or ip. but how does the analog pay phone determine the time it has
04:27.43JmanA91.0.3
04:27.48afrosheenjsolares: that's handled on the * server
04:28.07opus___bacon -- no :(
04:28.11afrosheenjsolares: the phone must pass some kind of tones via the iaxy to * I guess :~
04:28.18jsolaresah
04:28.19opus___I think I have mine setup exactly like everyone elses as well
04:28.23Baconopus___: What errors are you getting?
04:28.32jsolaresgood thing the iaxy uses ulaw then :)
04:28.43Baconopus___: Mar 12 00:23:38 VERBOSE[1487]:     -- Got SIP response 404 "Not Found" back from 147.135.0.128
04:28.47BaconThats my pain.
04:28.49afrosheenoh
04:28.53afrosheenregistration issues
04:29.59afrosheenjsolares: yeah but it'll end up g729 before it hits the t3
04:30.28Jabreityok, i have voicemail configed proper, i can send vm between extensions by dialing into the vm system, however i dont get rollover after x rings, or when busy when i dial an extension
04:30.36jsolaresbut the * is between that? eg. iaxy to asterisk in "local" net and then onto a voip provider?
04:30.52jsolaresi havent seen many voip payphones
04:30.54Baconafrosheen: I have registration issues, or someone else does?
04:31.07afrosheenjsolares: yeah iaxy is physically 2 feet from the switch, 2 feet from the * box
04:31.14EssobiHow much does a 5300 with 4 PRI cards cost?
04:31.18_6Flamez_Bacon: you do
04:31.22jsolaresthen it *should* be doable hehehe
04:31.23afrosheenBacon: you do
04:31.41BaconOdd, I'm getting inbould calls.
04:31.44afrosheenjsolares: you got me thinking about how * will handle the time from each phone
04:32.17afrosheenjsolares: with a normal analog payphone, each coin generates a tone that's picked up by a switch somewhere I believe
04:32.25Jabreityyup
04:32.38jsolareshmmm
04:32.43afrosheenif * can listen for those tones it can calculate how long the call should last
04:32.47BaconAny hints as to where to start looking for my problems? I'm pretty new to Asterisk.
04:32.52Essobijsolares It's a 4 wire POTS with a special switch on the other end.. atleast.. that's how bell does it.
04:33.02Jabreityin some fortresses
04:33.04EssobiAll the rest COCOTS are not like that.
04:33.24Essobi2 wire, and everything is completely physically driven inside the phone.
04:33.31Jabreitycorrect
04:33.34EssobiNo change, no bring bring.
04:33.34afrosheenthe iaxy *should* pass those coin tones to * right?
04:33.39jsolaresget 2 wire payphones afrosheen!
04:33.40jsolareshehe
04:33.49Essobi2 wires don't drop remote tones.
04:33.56Essobionly the 4 wires.
04:34.02afrosheenI imagine they'll be 4 wire phones
04:34.03Essobi2 wires work like a normal phone
04:34.13afrosheenit'll be a retrofit in a big place, like an airport
04:34.20Essobiand the other 2 are odd pass or no pass signaling for the $$
04:34.25EssobiUmm.
04:34.26Jabreityin theroy you could lay cc number on the line, and after approval permit carrier on line
04:34.35EssobiI doubt you'd get a BIG airport.
04:34.42afrosheenEssobi: why's that
04:34.45EssobiAs those want dataports on their payphones.
04:34.48Essobifor the laptops.
04:34.51JabreityBIG airports = BIG money
04:34.55EssobiAs those want dataports on their payphones.
04:34.59afrosheenthere is big money at stake here
04:35.15EssobiUmm.. Yea.
04:35.16afrosheenI'm in with a guy that owns phones in 70-something airports nationwide
04:35.29EssobiSo?
04:35.35afrosheenI suggested voip termination and a light went on above his head
04:35.51afrosheenso now it's my baby to see if it can be done
04:35.54jsolareshehe
04:35.56EssobiHire me to engineer them, and I'll drop, wire and write an LCD for him.
04:36.04Essobi:)
04:36.16EssobiUm... anything can be done.
04:36.18opus___dude, i want to red box a iaxy
04:36.20EssobiAnything.
04:36.21Jabreity... just give him lsd.  wont know the diff
04:36.22EssobiHaha.
04:36.23opus___tell me which airport :)
04:36.31afrosheen'all of them'
04:36.37jsolareshehehe
04:36.44Essobiumm.. he "Owns" them?
04:36.46JabreityAll your base...
04:36.59EssobiRBELONGINZORSTOUZES
04:36.59afrosheenyeah he owns the phones, has the right to rip them all out and replace them
04:37.03EssobiAhh.
04:37.11afrosheenlike I said, big money
04:37.14EssobiI thought you meant... he owned the airports..
04:37.15Essobihah
04:37.23Jabreitymmmmm, smells like someon stepped in some fresh capitalism
04:37.27afrosheenhe said bell earns $25 per phone on average
04:37.40afrosheenand if he could cut them out of the picture = more moeny
04:37.41afrosheen:)
04:37.44Jabreityper day/month/yr?
04:37.45EssobiWhat's he going to do about faxing and data?
04:37.49afrosheendaily I believe
04:37.56Jabreitynice
04:37.59afrosheendude these are just payphones
04:38.04EssobiHe could put some 1FBs in.
04:38.06EssobiDude.
04:38.08afrosheenwe're not starting a kinko's at each damn phone
04:38.17EssobiMy local airport.. has data jacks on the payphones.
04:38.19Jabreityooooh, add a wap
04:38.25Essobi:)
04:38.36opus___yeah, they got data jacks. but there's also free wireless too..
04:38.39afrosheenI considered some kinda funky mesh network..but maybe too much hassle and fcc/faa issues
04:38.41Dandananyone using BV can show me his register string?
04:38.47Dandan<PROTECTED>
04:38.48*** join/#asterisk Weezey (~Weezey@206.210.109.226)
04:38.51Dandanthat's what i get
04:38.54jeffikopus___/bjohonson_: thanks for the help
04:38.59afrosheenDandan: did you patch asterisk yet
04:39.08Weezeyis there a device out there that acts as a SIP peer and a MGCP client?
04:39.12Dandanasterisk from CVS...
04:39.14afrosheenI thought I saw a posting from broadvoice recently mentioning some patch
04:39.17Dandanread that you need no patch
04:39.33opus___dandan http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice
04:39.41Dandanopus___: that's what i did
04:39.49Dandanand I am getting 404
04:39.53*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
04:39.59Dandanso i wanted a real life example
04:40.12*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
04:40.17BaconDandan: Are you behind a nat?
04:40.22Dandanno
04:40.24opus___register => 555121212@sip.broadvoice.com:mypassword:mynumberagain@sip.broadvoice.com
04:40.25Dandanexternal ip
04:40.45opus___thats my real life fake example
04:40.56opus___i don't live in 555 yet:)
04:40.59BaconYup. That is what I have.
04:41.05Dandanshhh
04:41.07WeezeyI call 1212121 locally.
04:41.09Dandanlemmie look :)
04:41.31*** join/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net)
04:41.38Dandanhmmm
04:41.43Dandani built asterisk today
04:41.47Weezeyyay!
04:41.48Dandanit says i need no patch
04:41.53Dandananyone can verify?
04:42.00opus___yes
04:42.10opus___with cvs head from friday
04:42.28Dandanyes: do i need it?
04:42.34Dandanor yes: i am correct?
04:42.42BaconMy config: http://pastebin.ca/7299
04:42.56jjgwere there any enhancements available via CVS due to VON?
04:43.19DandanBacon: tx
04:43.28Dandanwill work with what you pasted
04:43.36BaconDandan: I can get inbould calls, but I can't dial out.
04:43.44_6Flamez_host=proxy.lax.broadvoice.com is wrong i believe should be host=sip.broadvoice.com
04:43.46DandanBacon: y?
04:43.52BaconSome of the folks here say I have registration problems.
04:44.04Bacon_6Flamez_: I'll try that.
04:44.17*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
04:44.42SedoroxDoes anyone know if mpg123 has a problem with SMP FreeBSD systems?
04:45.11Jabreityok can anyone tell me why my voicemail isint picking up after 6 rings?
04:45.24Bacon_6Flamez_: kickass.
04:45.27BaconIt works.
04:45.39_6Flamez_cool
04:45.45BaconSo much for broadvoice's instructions.
04:45.51DandanBacon: the password is whatever you use to get to their website
04:45.52opus___bacon - you can call out?
04:45.55SedoroxJabreity: do you have it where the extention will timeout? e.g. exten => 1000,Dial(SIP,user,25,rt), where 25 is about 4 rings...
04:46.02Sedorox25 secs...
04:46.09Jabreityoh
04:46.14Jabreitycraptastic
04:46.15SedoroxBacon: BV's instructions are messed
04:46.15Baconopus___: Yup.
04:46.25opus___bacon - hmmm, is it any different from my config?
04:46.25Jabreityim a crackhead
04:46.45Baconopus___: Dunno, click the link and check it out.
04:46.48SedoroxJabreity: gotta have it will it'll time out and goto the next priority :-p
04:46.59Sedoroxwill=where
04:47.01opus___http://pastebin.ca/7288
04:48.48Jabreitysedrox -
04:48.49Jabreity101=> 101, 101,jasonbreitwieser@hotmail.com
04:48.49*** join/#asterisk JMcA (~jmcadams@67.141.1.51)
04:49.14DandanMar 11 23:48:31 NOTICE[13759]: chan_sip.c:4309 sip_reg_timeout:    -- Registration for '860XXXXXXX@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again
04:49.18Jabreityim missing something for sure
04:49.20Dandan<PROTECTED>
04:49.26Dandananyone can help me with that?
04:49.42SedoroxJabreity: thats in voicemail.conf, right?
04:49.47Jabreityyes
04:50.02SedoroxJabreity: no.. the example I gave is extentions.conf
04:50.08Jabreity?
04:50.11Jabreityduh
04:50.20Jabreitysorry, long evening
04:50.29Sedorox'tis fine
04:50.29Sedorox:-p
04:50.31SedoroxI've been there
04:50.51Sedoroxyea.. you want to have a timeout on the extention so after 25 seconds.. if no pickup.. it goes to the next priority.. which should be voicemail
04:50.56Sedoroxor dialing another phone.. etc..
04:51.11Baconopus___: Here is mine: http://pastebin.ca/7299
04:51.18Jabreityok, lemmie post my config online, i think i got it right
04:51.23Sedoroxok
04:51.31BaconDandan: http://pastebin.ca/7299
04:51.32tuxinator_linuxMAnybody still in San Jose, or is it just me?
04:51.53Sedoroxwell the extention should just look like exten => 1000,1,Dial(SIP,user,25,rt) exten => 1000,2,Voicemail(u1000) or something like that
04:51.57Sedoroxdepending on your setup...
04:52.15SedoroxI use 25.. beause its about 4-5 rings.. the average for answering machines.. altho some people's are higher
04:53.18_6Flamez_after 25 secs if will jump to 101 not 2
04:53.37Sedoroxmine always jumps to 2... hmmm
04:53.46Sedoroxif busy.. goes to 102, right?
04:53.48DandanBacon: that's what i have
04:53.59tuxinator_linuxMjust 101, not to 101
04:54.03tuxinator_linuxMjump
04:54.06JabreitySedorox: http://www.theextremeoutfitters.com/extensions.txt
04:54.19DandanBacon: how did you build your *?
04:54.28Dandandid you cvs? ftp? patched?
04:54.55SedoroxJabreity: ok.. the extentions will timeout after 60 seconds.. or 30.. depending on which one your using...
04:55.50Sedoroxexcept somewhere in there you have to have exten => 100,101,Voicemail(u100) (where 101 apparently is where it goes after 60 secs.. and the u100 is user 100 in voicemail.conf)
04:56.06BaconDandan: I cheated, I install asterisk@home
04:56.08jjghas anyone built an iax client?
04:56.09Sedoroxif you want.. I can put up some examples of mine...
04:56.09JabreitySedorox: bearing in mind im emotionally fragile at this point, this is my first time... trying to learn
04:56.27SedoroxJabreity: thats perfectly fine... should I start to curse you out? :-p j/k!!!
04:56.38SedoroxI was there a few weeks ago learning the dialplan
04:56.56Jabreity:) go for it.  i prolly deserve it from the blasting i gave my boxen
04:57.05Sedoroxahah
04:57.07_6Flamez_heh
04:57.16Sedoroxhere.. let me paste a few examples outta my config
04:57.21Jabreityokies
04:57.46opus___bacon - it just doesn't ring.. hmm
04:58.34DandanBacon: LOL :)
05:00.13SedoroxJabreity: http://pastebin.ca/7300
05:00.17*** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.rr.com)
05:00.20JabreitySedorox: did i make a noob mistake or what :)
05:00.26Sedoroxdunno
05:00.26Sedoroxhehe
05:00.32Sedoroxprobably.. don't think your including everything
05:01.00lqdengrhi guys, i'm new to *, i've got my inbound VoIP service working with an IVR menu where I can dial my extension and ring my x-lite softphone, but I can't seem to figure out how i would go about configuring * to let me dial out of the system from a x-lite, it says call not permitted
05:01.39_6Flamez_huh
05:01.57Sedoroxlqdengr: you just need to configure a extention to dial out on... or a matching plan I think.. like if you dial 928005551212.. where 9 is used to say "hey.. I'm dialing out"
05:02.04Sedoroxthere's some good stuff in the wiki about it...
05:02.26lqdengrsedorox: so basically i need to set up some extension or "code" to tell * that i need an outside line?
05:02.31Sedoroxalso.... you can look at the example configs that are posted there.. they help if you learn that way (like me.. bu seeing and implamenting)
05:02.37Dandanok, quick hack, how to make an extensions which would say date/time
05:02.38Dandan?
05:02.45Dandanas 612 in *@home
05:02.55*** join/#asterisk ikey (~kirankuma@202.54.37.186)
05:03.08lqdengrsedorox: is http://www.voip-info.org/wiki-Outbound+call+handling what i need to be looking at?
05:03.15Sedoroxwell now.. you just need to have it match.. you can have it where you can just dial the number... or you can have it where you need to press 9 first.. or any other number (most poeple do standard and pick 9 to dial out of)
05:03.22*** join/#asterisk JmanA9 (~josh@pa-murraysville2b-141.pit.adelphia.net)
05:03.26Juggie8 is cooler :P
05:03.29lqdengrlol
05:03.30ikeyhi can any one help me in configuring two sip channels in asterisk
05:03.32_6Flamez_Dandan: exten => 999,2,DateTime
05:03.41_6Flamez_be sure to Answer first!
05:03.44Dandan_6Flamez_ thx a lot
05:03.44Dandan:)
05:03.46Sedoroxlqdengr: looks a little complicated.. but yes
05:04.02lqdengrsedorox: ok ill putz around with that for a while, thx for your help!
05:04.39Sedoroxyup... like I said.. the best way would probably be under the example configs.. people have it where they can dial out a FXO.. or dial out a SIP.. its normally the same.. if you can't get it.. let me know.. I'll help ya
05:05.23*** join/#asterisk sudhir492 (~sudhir@4.7.58.171)
05:05.34afrosheenikey: just start asking, don't ask to ask
05:05.52sudhir492anyone using PAP2-NA or Sipura SPA-2000 here?
05:06.02*** join/#asterisk locovox (~locovox@218-153-89-200.fibertel.com.ar)
05:06.27sudhir492I cannot use both phones at the same time with PAP2-NA
05:06.34locovoxhi just need a little help configuring my first asterisk with a TDM21B
05:07.06afrosheenlocovox: what's in a tdm21b
05:07.41Sedoroxafrosheen: I think its a FXO and FXS....
05:07.50afrosheenlet the man answer :)
05:07.57Sedoroxhehe
05:07.59locovoxi'm trying to make it work, however i think this time i got dialton!!!
05:08.29Sedoroxoh wow.. I was wrong...
05:08.43locovoxgimme five mins
05:08.46opus___bacon --  -- Executing Dial("SIP/175-faee", "SIP/18005551212@sip.broadvoice.com") in new stack
05:08.47opus___<PROTECTED>
05:08.50opus___does your line say that?
05:09.07opus___when dialing?
05:09.57Dandanopus___: u using BV?
05:11.06opus___yes
05:11.08opus___when dialing out
05:11.36Goshenhowdy...funny you two are talking about that...I just switched over here to get help with my outbound broadvoice
05:11.46GoshenI can receive calls, but not dial out
05:11.57opus___goshen - well, I can give you my config.  For some reason it works fine with other ppl:)
05:12.06Goshenok, lets have it
05:12.14Goshenpatebin.ca or query window
05:12.15_6Flamez_sure a lot of BV related probs...
05:12.19Dandanopus___: can i have it also?
05:12.21opus___goshen - http://pastebin.ca/7288
05:12.27Goshenhttp://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup didn't work for me
05:12.29opus___Hey it works for everyone but me! :)
05:12.33Dandan_6Flamez_: their website is confusing at best
05:12.40Goshen6Flamez: thats because its crappy SIP! I hate sip
05:12.43*** join/#asterisk _santiago_ (~santiago@63.245.86.95)
05:12.58_6Flamez_what link... one is corrent and one isnt
05:12.59Goshenbut they have LMP..so they get my ring
05:13.03_6Flamez_correct*
05:13.17opus___LMP?
05:13.22_6Flamez_ya i dropped them due to sip issues
05:13.35afrosheensip is good people
05:13.54afrosheenuntil polycom and others start making iax phones
05:13.59Goshensip and its huge range of natdefying ports stinks
05:14.04Dandanopus___: and your register string?
05:14.12Goshenmy register string works
05:14.14opus___polycom probably won't make iax phones
05:14.20afrosheenyeah I know
05:14.25Goshenpolycom has IAX phones...
05:14.27afrosheenwe're lucky they're making good sip phones
05:14.32opus___register => 555121212@sip.broadvoice.com:mypassword:mynumberagain@sip.broadvoice.com
05:14.40opus___goshen oh?
05:14.42Dandanhm :/
05:14.50Goshenmake sure you use the sip registration password
05:14.56GoshenNOT the website password
05:15.12BaconYeah, that tricked me.
05:15.13afrosheensince when does polycom make iax hardware?
05:15.16opus___goshen -- incoming works fine, outgoing has problems.. that wouldn't be it would it?
05:15.19Goshenyou get your sip password from the account page
05:15.38Goshenif incoming works fine it isn't your registration string
05:15.44_6Flamez_Goshen: what do u have as your host= line?
05:18.09Goshenin my nonworking outbound sip.conf?
05:18.18_6Flamez_ya
05:18.25Goshenhost=sip.broadvoice.com
05:18.43_6Flamez_ok.. usually that's the problem, but yours is correct
05:18.50opus___exten=_9NXXNXXXXXX, 1, Dial(SIP/${EXTEN:1}@sip.broadvoice.com) still won't dial out, I just get dead air and a timeout
05:18.54Goshenthe other nonworking one that I tried is
05:18.56Goshenhost=proxy.broadvoice.com
05:19.01opus___audio works full duplex on incoming just fine
05:19.10_6Flamez_has to be sip.broadvoice.com
05:19.30opus___goshen - proxy=sip.broadvoce.com works too
05:19.32_6Flamez_if you want to use a different proxy, u'd need to add a /etc/hosts entry etc..
05:19.54opus___that will work but is not required
05:20.09opus___proxy=proxy.***.broadvoice.com works
05:20.14opus___on some servers...
05:20.23_6Flamez_that changed?
05:20.24*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
05:20.30opus___It really all depeneds on random numbers it seems:)
05:20.40Eight_6Flamez_: I tried the /etc/hosts trick, no joy.
05:20.46Goshenbah! same crap...
05:20.47EightIt seems some accounts just aren't on some proxies.
05:20.47GoshenExecuting Dial("SIP/21-b012", "SIP/8017123381@sip.broadvoice.com|30") in new stack
05:20.47Goshen<PROTECTED>
05:20.47Goshen<PROTECTED>
05:21.09Eighthttp://www.voip-info.org/wiki-Asterisk+settings+Broadvoice
05:21.09opus___gosh hmm
05:21.14EightHave a look at the second example.
05:21.15GoshenI am using the cvs stable from today
05:21.17EightThat one works for me.
05:21.41GoshenI am using the 1.0.7RC
05:21.55opus___thats todays cvs right?
05:22.06Goshenyup
05:22.13Goshenfrom this afternoon MST
05:22.20Goshenso it should include the patch
05:22.29Eight1.0.6 works without patching
05:23.01Dandan*CLI> Mar 12 00:22:42 NOTICE[13979]: chan_sip.c:8776 sip_poke_noanswer: Peer 'sip.broadvoice.com' is now UNREACHABLE!  Last qualify: 0
05:23.07Dandanhuh?
05:23.08Dandanwhich means?
05:23.34Nuggetit means pretty much exactly what it says.
05:23.36EightDandan: your 'net connection is flaking, and you can't ping sip.broadvoice.com
05:23.39Nuggetwhich part is confusing?
05:23.45EightDandan: did you try /etc/hosts
05:23.48opus___Uhh, is this correct:
05:23.53DandanEight: no, should i?
05:23.59opus___To get the current stable release, issue the following command:
05:23.59Eightactually...
05:24.02DandanMar 12 00:23:44 NOTICE[13979]: chan_sip.c:4309 sip_reg_timeout:    -- Registration for '8607772005@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again
05:24.03opus___# cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
05:24.09opus___won't that download 1.0.7RC?
05:24.21EightActually, I *just* had BV go unreachable for a moment.
05:24.24Nuggetthat will download cvs stable.
05:24.31EightI think their network flaked out for a moment.
05:24.31opus___oh i'm dum
05:24.38DandanEight: oh :)
05:24.41Dandancool
05:24.54Dandan8)
05:25.02GoshenITS A MIRACLE!!!
05:25.10GoshenThe second example on the wiki works!!!
05:25.12Dandandoesn't make it any less confusing for a newbie...
05:25.16GoshenTHANKS!
05:25.16EightGoshen: Your welcome.
05:25.22EightThat's my config.
05:25.27Sedoroxlol
05:25.32Eighterr, "you're", rather.
05:25.32Goshennow we just need to delete all of the other crap...
05:25.42opus___nugget -- wait, is stable = 1.0.7rc in cvs?
05:25.43Goshenand send broadvoice your config
05:25.53Nuggetstable is stable.
05:25.58Goshenand tell them to send that out as the example
05:26.00opus___http://bugs.digium.com/bug_view_page.php?bug_id=0003746  saids  We are ready to release 1.0.7 but need some people to test and verify that the code is without any major bugs.
05:26.08lqdengrsedorox: i looked around at some sample configs for outbound calls, and i still can't seem to get it working
05:26.10GoshenAsterisk CVS-v1-0-03/11/05-12:59:27 built by root@localhost on a i686 running Linux
05:26.24opus___"Please grab the latest code from stable CVS (cvs co -r v1-0 asterisk)."
05:26.30Sedoroxlqdengr: how do you want to dial out? PSTN, SIP, IAX, CAPI?
05:26.44lqdengrIAX
05:26.48Goshenok, everyone have their broadvoice working now?
05:26.50Sedoroxwith who?
05:26.55GoshenI have incoming and outgoing working now
05:27.07opus___goshen -- glad I could help. DOesn't work for me though:) hehe
05:27.07lqdengri have a line in my extensions.conf that looks like exten => _1NXXNXXXXXX,1,Dial(IAX2/foo:bar@gwiaxt01.voicepulse.com/${EXTEN})
05:27.16Goshenoh..and if you have pulver communicator and upnp on your nat...you are going to have problems
05:27.26Goshenbecause pulver communicator takes sip ports
05:27.38Sedoroxlqdengr: then you should be able to dial 1 and the numbers
05:27.40Goshenopus: what problems are you having?
05:27.41Sedoroxnumber*
05:27.52Sedoroxso like...
05:27.58opus___goshen -- can't dial out, I'm going to try today's cvs "stable" first
05:27.59lqdengrsedorox: i think i must have it under the wrong contect or something, because it still isn't working when i try that
05:27.59Sedorox1-123-555-1212
05:28.04DandanEight: pls pastebin it :)
05:28.06Goshenahh ok
05:28.09Sedoroxwhat context do you have it under?
05:28.16Goshenyou may not have the latest patches for BV
05:28.26lqdengr[outbound]
05:28.35lqdengrthats what the sample file from voicepulse had it listed under
05:28.46Sedoroxdo you have include => outbound, under either local or default?
05:28.53lqdengrno, let me try that
05:28.56Sedoroxdo that
05:29.03Sedoroxthen you should have access to that extnetion
05:29.07Sedoroxwhich will allow you to dial out
05:29.07Sedorox:)
05:29.27EightDandan: do the who what?
05:29.38Dandanhttp://pastebin.ca - your configs
05:30.11*** join/#asterisk spackle (~spackle@209.234.83.19)
05:30.14EightDandan: I already put mine into the wiki.
05:30.18lqdengrhmm weird, i think i have my exensions.conf file all screwed up or something.  i put it under default, but still no go
05:30.28EightDandan: you don't want the rest of my configs, doesn't have anything to do with BV.
05:30.35DandanEight: ok
05:30.45Sedoroxpastebin you extentions.conf.. just change your passwords
05:31.02Dandanor don't change them :D
05:31.10EightI have only two lines in my extensions.conf that are relevant to bv
05:31.17Eight[from-broadvoice]
05:31.19Eightand
05:31.27Eightexten => s,1,goto(default,s,1)
05:31.34*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
05:31.50EightOr are you talking about dialing out?
05:33.08Eightexten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.broadvoice.com)
05:33.14EightThat's a 10 digit US number.
05:34.09opus___later..
05:34.27EightSo... does everyone here have BV functioning?
05:34.47GoshenHey Eight: there is only one thing you are missing in your register line...that is the extension to put the incoming call in to
05:34.50Sedoroxthen again.. I don't have BV :-p
05:35.03EightGoshen: if you leave off the /extension, it just goes to s
05:35.07Goshen@sip.broadvoice.com/s   on the end will put them to s
05:35.12SedoroxI actually just got a toll free number with link2voip... so gonna set that up now...
05:35.18EightGoshen: you don't even need the /s
05:35.33Goshenso you really don't need from-broadvoice if you put them to to your default incoming context
05:35.48Goshenunless of course you just want to handle them differently
05:36.15Goshensomething that I feel needs to be made clear to new users is the fact that you must 10 digit dial without the 1 always now
05:36.18Goshenlike a cellphone
05:36.27DandanEight: i don't
05:37.17Dandanheh would like sbdy to help me step by step
05:37.18Dandan:/
05:37.25Dandanit is waaaay too confusing for me...
05:37.26Dandan:/
05:37.35EightGoshen: really? 1 works fine for me.
05:37.39EightGoshen: or did last night.
05:37.40GoshenDandan: download Asterisk@Home
05:37.55DandanGoshen: i have it on my desktop
05:38.05Dandanproblem is i like to know it from the roots
05:38.09EightAsterisk@home will DELETE YOUR HARD DRIVE>
05:38.11Dandanand that's how I learn
05:38.15GoshenEight: that dial string you posted is without a 1
05:38.25EightOnly install it on a machine you want to dedicate to asterisk
05:38.28SedoroxEight: eh?
05:38.29lqdengrsedorox: sorry, took me a while to get it copied: http://pastebin.ca/7302
05:38.29EightGoshen: I have another with the 1 =)
05:38.36Sedoroxlqdengr: thats ok
05:39.05EightI actually have it setup to give an 'outside dialtone' for Broadvoice, in a different context.
05:39.11Sedoroxwhat context do you have your phones connected to?
05:39.17Dandanso that's i would like to help me with BV priv if possible
05:39.35Goshen~docs
05:39.36jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
05:39.56GoshenDandan: spend a week or so reading :)
05:39.57lqdengrsee, i'm not sure, in sip.conf i set it to outbound
05:40.00EightSedorox: well, SIP phones come into [sippers] which includes [default], and calls from broadvoice come into [from-broadvoice] which right now just kicks right to [default]
05:40.05Dandani bought the yellow book too
05:40.43SedoroxEight: hmm.. even tho I was referring to lqdengr  :-p not sure what your thing was
05:41.00Sedoroxlqdengr: can you dial the other extentions listed there?
05:41.09GoshenDandan: this page helped me quite a bit when I was getting started.... http://iheavy.com/modules.php?op=modload&name=News&file=article&sid=35&mode=thread&order=0&thold=0
05:41.10EightSedorox: oh, silly me... I was thinking you had 'eight' on the beginning of your comment =p
05:41.15Goshenit has a nice simple working config
05:41.24jjgcan anyone tell me how to terminate a call through nufone from a laptop?
05:41.26SedoroxEight: tis fine :-p what are you trying to do? lol
05:41.30lqdengrsedorox: nope, i sure cant
05:41.31*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
05:41.34Goshenand was the exact configuration I have...one FXO card, and a grandstream sip phone
05:41.40Sedoroxok...
05:41.44EightSedorox: I think I have everything I've tried so far working fine, actually =)
05:41.45Sedoroxis that your entire extentions.conf?
05:41.48lqdengryes
05:41.53SedoroxEight: cool
05:41.56Sedoroxlqdengr: weird....
05:42.17Sedoroxok
05:42.18Sedoroxdo this
05:42.35Sedoroxlet me see if I can update it on the pastebin to make it easier
05:42.35lqdengrsedorox: this is my first * experiment, and i've been messing with that file for the past day or two getting inbound working correctly, so i must have screwed something up
05:42.40lqdengrok
05:43.04*** join/#asterisk santiago (~santiago@63.245.86.95)
05:43.08Goshenopus____: you still compiling?
05:43.19Sedoroxlqdengr: everyone's gotta learn somehow.. like I said.. was in your posistion a few weeks ago...
05:43.50lqdengrsedorox: heh.  well i appreciate it '-)
05:43.55lqdengrerr.  ;-) too
05:44.43Dandanok till tomorrow :)
05:44.44Dandan[d]
05:45.30bonez39Goshen: I have a linksys rt31p2 2 port phone adapter/router...I have flashed it once to get the newest firmware...wouldn't the fact that I can flash it suggest that the hardware is open..i.e., could be used with asterisk?
05:46.39Goshenbonez39: hey :) its possible...can you get in and change the server it connects to?
05:46.44Sedoroxlqdengr: I see a few things... so you might just wanna copy this over your exsisting one..
05:46.48Sedoroxbut save a compy
05:46.50lqdengrroger
05:46.50Sedoroxcopy*
05:46.52Sedoroxto compare
05:47.01Sedoroxand you have it as outgoing.. not outbound.. hence why it didn't work :-p
05:47.04Sedoroxbut anyway
05:47.08lqdengrlol!
05:48.28lqdengrso, did you update the copy on pb?
05:48.37Sedoroxnot yet... still checking it over
05:48.43Sedoroxtrying to do several things here.. so bear with me...
05:48.56*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
05:49.08lqdengrn/p
05:49.38Sedoroxlets see if pastebin will ne nice
05:49.39Sedoroxbe*
05:51.10Goshenbonez39: do this...take screenshots of all of your current config, and print them
05:51.18Goshenso you can switch back easy
05:51.46Sedoroxlqdengr:
05:51.48Sedoroxhttp://pastebin.com/253643
05:52.04Goshenbonez39: see this page http://www.voip-info.org/tiki-index.php?page=Linksys
05:52.07Sedoroxand either remove context=, from your sip.conf
05:52.11Sedoroxor set it to context=default
05:53.00lqdengri'm guessing there's not supposed to be a php tag on the first line there
05:53.05Sedoroxnooo
05:53.10Sedoroxpastebin adds that
05:53.13lqdengroh
05:53.16Sedoroxyou'll se the end tag at the end too
05:53.24Sedoroxthe ?>
05:53.25Sedoroxis php ending
05:53.26lqdengractually, i dont see one
05:53.28lqdengrohh
05:54.40jjgcan someone please help me make a call through nufone with a laptop?
05:55.19lqdengrhmm, nope i'm still not having any luck with it
05:55.29Sedoroxdid you do a reload on the console?
05:55.35EssobiHmm.
05:55.39lqdengri wonder if i could have some sort of config problem with x-lite.  do you have to tweak any of the default settings?
05:55.41rvhianyone knows how to run two instances of * on one server?
05:55.42lqdengryes, i reloaded
05:55.57Sedoroxrvhi: just have it bind to different addresses/ports?
05:56.02EssobiWhat kind of neet shit would you guys want from the management interface on a web control interface?
05:56.06Sedoroxummm.. for xlite....
05:56.52SedoroxEssobi: adding/removing extentions, possibly uploading files for moh or background/playback, voicemail control (add/remove users, reset passes, etc)
05:57.40lqdengrwhen i try to place a call, should i see some activity on the console?
05:57.54Sedoroxlqdengr: yes
05:57.56Sedoroxif you have verbose on
05:58.00lqdengrstrange
05:58.03Sedoroxlike I always start mine with -vvvc
05:58.08EssobiEhh, MOH is easy.
05:58.08Sedoroxin a screen session.. but anyway
05:58.09lqdengrok, thats what i did
05:58.15*** join/#asterisk ikey (~ikey@202.54.37.183)
05:58.29lqdengrand i turned on sip debug to see if that showed anything useful
05:58.41Sedoroxand does it show anything?
05:58.49lqdengrbut i don't see anything at all when i try to place a call, however * can call my softphone with no problems
05:58.56lqdengrso the sip connection is good...
05:59.11Sedoroxummmm
05:59.12Sedoroxnot all the time
05:59.15ikeyhi can any one help me in configuring sip channels and to use it with grandstream ip phones
05:59.15*** join/#asterisk Tarox (someone@pD9E7B0C0.dip.t-dialin.net)
05:59.25Sedoroxdoes 'sip show peers' show your phone?
05:59.30Sedoroxand IP and all that
05:59.36lqdengrsip show users does
05:59.39Sedoroxikey: its the same as anything else
05:59.50lqdengryeah peers does as well
05:59.58Sedoroxthere is actually a example in there for a grandstream
06:00.05Sedoroxlqdengr: hmmmm
06:00.26*** join/#asterisk sleepy_one (~chatzilla@dhcp16632045.neo.rr.com)
06:00.38Sedoroxdouble check your setting in xlite.. I think something is messed up where it isn't going to the server for outgoing calls... because your console should be floodded with stuff when you do anything on the sip channel. woith debug on
06:00.42lqdengrikey, look in sip.conf, there's a good example you can build from
06:00.53lqdengrsedorox: k
06:01.33sleepy_onehello everyone
06:01.33lqdengrsedorox: k, i dont have any idea WHY, but it magically started working
06:02.12Sedoroxahah
06:02.14Sedoroxdunno
06:02.23Sedoroxcomputers are weird like that
06:02.34sleepy_onedoes anyone know how to get mpg123 working on FC3x86_64 ?
06:02.36lqdengri clicked on the menu button in x-lite, then the console started exploding, so maybe it was just "stuck"
06:02.49Sedoroxlol
06:04.29Essobisleepy_one Umm. compile it?
06:04.30Essobi:)
06:04.42sudhir492Does asterisk run on FC3, kernel 2.6.9?
06:04.58lqdengrit should
06:05.01sleepy_oneAbsolutely! runs very well thank you!
06:05.26Goshenasterisk works well on my 2.6.10 kernel
06:05.44sudhir492Ok. when I try to compile asterisk, I get the following error: /usr/bin/ld: cannot find -lidn
06:05.44sudhir492what is libidn?
06:06.08sudhir492thanks
06:07.00sudhir492Goshen: what distro are you running?
06:07.09GoshenMandrake 10.0
06:07.22Goshenwith custom compiled kernel 2.6.10
06:07.39Nuggetlinux is poo.
06:07.43tuxinator_linuxMmake sure Makefile is looking for the right source/build files
06:07.47EssobiPEE!
06:08.00sudhir492I just installed FC3. trying to compile asterisk but got stumped by lidn
06:09.05sudhir492tuxinator_linuxM: Anything specific do you have in mind?
06:09.18tuxinator_linuxMNugget: I don't think this is the right channel to bash on linux
06:09.29EssobiWhy not?
06:09.31Essobi:)
06:09.44tuxinator_linuxMHave you read wiki pages on 2.6
06:09.56Nuggetdoesn't change the fact that linux is poo.  :)
06:10.10Goshensudhir492: do you have everything listed on this page? http://www.asterisk.org/index.php?menu=download
06:11.03nine76Hey all,I would appreciate anyone looking over my configs and giving me some clues as to why * says "No channel type registered for 'Zap'" . Thanks:-/  http://pastebin.ca/7303
06:12.02sudhir492I do, except bison devel, which I thought may not be necessary.
06:13.06lqdengrIs there any way to set outbound caller id info with voicepulse?  My calls show up as some New York phone # instead of my actual #.
06:16.51Sedoroxhow many calls can a voice T1 support?
06:16.58lqdengr23 for pri
06:17.00lqdengr24 for t1
06:17.05Sedoroxah thats right...
06:17.21Sedoroxwhat do you think it is if it was d data T1 with VoIP
06:17.24Sedoroxaround 10?
06:17.34lqdengrno, a lot more than that i'd say
06:17.40lqdengri think they say about 50kbps/call
06:17.47lqdengrso 1544 / 50
06:17.53Sedoroxthats 25 in and out...
06:18.00Sedoroxso yea...
06:18.03lqdengrabout 31, minus tcp overhead, etc.
06:18.09*** join/#asterisk ikey (~ikey@202.54.37.183)
06:18.10Goshennine76 did you modprobe your modules?
06:18.24nine76Goshen: Yes,no errors. x100p card
06:18.26ikeyhi can any one help in configuring sip channels in asterisk
06:18.35Sedoroxikey: what do you need?
06:18.50Goshennine76: wcfxo?
06:19.01nine76yes,I am doing it again to be completely sure:)
06:19.13nine76I did zaptel,wcfxs and wcxfo
06:19.25Goshenwhy wcfxs?
06:19.32Goshenthat isn't right for a x100p
06:19.43nine76Was following directions on "getting started with asterisk" guide
06:19.57Goshenwhere?
06:20.10nine76http://www.automated.it/guidetoasterisk.htm#_Toc49248763
06:20.28Goshenwrong card...that is for a TDM400P Installation
06:20.31*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
06:20.35nine76I also tried others configs, as I noticed they varied. Gettign started guide had "channel=1" while an onlamp article had channel => 1
06:20.58Goshenget rid of the wcfxs module
06:21.16Goshenx100p only has fxo
06:21.44Goshennine76: http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf
06:22.20Goshenalso, did you run ztcfg?
06:22.47lqdengranyone know if you can do ivr from within your voicemail greeting?  i'd like to give people the option of pressing 9 to have * connect them to my cell
06:23.06sudhir492There was a symbolic link for libidn.so in /usr/lib directory. After creating the link, asterisk builds fine
06:23.21sudhir492I mean sym link was missing
06:23.39Goshenlqdengr: I think there is an option to dial 0 for the operator
06:23.44nine76I removed wcfxs, restarted *, still no chan type.. ztcfg --v output is at bottom of pastebin. exited ok saying 1 chan configured succesfully
06:24.10nine76dmesg also reports wilcard there
06:24.20lqdengrgoshen: ok.  thanks
06:24.29sudhir492lqdengr: do the ivr first before running voicemail
06:24.47xkevzaptel pri gurus in here tonight?
06:25.05xkevproblem with facility message failing to handle caller-name
06:26.57lqdengrsudhir493: duh, thanks for pointing out the obvious to me.  :-)  /me rethinks getting another beer
06:27.19EssobiMmm.
06:27.34EssobiHow is * going to interact with a T1 card with DSPs on it?
06:27.50Goshenany suggestions for a gsm player for windows?
06:28.10EssobiThat'd be a massive code change ehh?  DTMF detection, silence detection, tone playback, IVR playback..
06:28.14EightGoshen: quicktime
06:28.16lqdengrQuicktime
06:28.23*** join/#asterisk jeffik (~jeffik@m9f7236d0.tmodns.net)
06:29.30xkev"Do not handle argument of type 0x80"
06:30.32Eightxkev: http://www.voip-info.org/wiki-PRI
06:31.09xkevthis is 0x80
06:31.15xkevnot 0x84
06:31.20Eightah, I see that.
06:31.21Eightmy bad.
06:31.24xkevheh
06:31.49xkevbut I just restowed some older cvs, and I bet it's my fault for something I did in zapata.conf, since it didn't fix it (twas working)
06:33.46xkevoh wait, I just rolled asterisk back not zaptel
06:34.23shido6boink
06:34.35JmanA9well, i got asterisk working :)
06:34.43JmanA9i just deleted every folder that had the word asterisk in it and reinstalled :)
06:34.57shido6hehehe
06:35.39Sedoroxfind / -name *asterisk* | rm -rf
06:35.40Sedorox:-p
06:35.53JmanA9it worked :)
06:36.10Sedoroxfind / -name * | rm -rf
06:36.13SedoroxO:-)
06:36.32EightI knew *someone* was going to have to give the command to accomplish that =p
06:37.02Sedoroxahahah
06:37.15Sedoroxeh.. I prefer dd if=/dev/null of=/dev/hda
06:37.15ikeyhi can any one help in configuring sip channels with grandstream phones
06:37.56EightSedorox: /dev/random =)
06:38.02Sedoroxhehe
06:38.19Goshenikey: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
06:38.56EightWe need jbot to send a welcome /msg to people as they join.
06:39.01Eightlisting stuff like ~docs
06:39.17Eightjbot welcome?
06:39.20SedoroxGoshen: I think he has the channel on mute or something.. he's asked three times.. and each gotten a answer
06:39.42Goshenhmm
06:40.59EightAnyone have an SRV record for their server?
06:41.17Nuggetat least one person does, yes.  :)
06:41.42EightNugget: =p
06:41.45Eightlol... whoops.
06:41.55xkeveight I do yeah
06:41.58Eightthis song on Philadelphonic starts with a phone ring.
06:42.08EightI was like "WTF?! Why is my computer ringing?! I didn't do anything!"
06:42.22Goshenlol
06:42.35Nuggetthere's a song by the cure that has a noise in it that sounds exactly like my alarm clock.  it makes me grumpy whenever I hear it
06:42.35xkevyou've been sucked into the void for too long
06:42.58Eightxkev: Mind if I verify my Asterisk install is properly doing SRV lookups for SIP addresses?
06:43.28xkevsure, try kevin@pbx.xmission.com, you'll just get my voicemail though
06:44.02Eightuh.
06:44.13Eightyou only have A and NS records...
06:44.24Nuggetslacker.com has a srv record for sip.
06:44.33Nuggettry nugget@slacker.com if you want
06:44.40xkev_sip._udp.pbx.xmission.com. 3600 IN     SRV     1 0 5060 pbx.xmission.com.
06:44.42xkevI do?
06:44.53Eighthmmm, been a while since I've used dig =)
06:44.56xkevpbx.xmission.com.       3600    IN      NAPTR   2 0 "s" "SIP+D2U" "" _sip._udp.pbx.xmission.com.
06:44.57Nuggetheh
06:45.04Goshena fellow utahn :)
06:45.20xkevhola!
06:45.34xkevyou in utopia territory?
06:45.41Goshenxkev: unfortunately NO!
06:45.45GoshenI want fiber to my home!
06:45.48xkevme either
06:46.01xkevwe're connecting up in a few months (XM is)
06:46.14GoshenXM?
06:46.23xkevxmission
06:46.43xkevdig SRV _sip._udp.foo.bar @ns
06:47.26xkevgoshen, just IP data at first, but we'll do voip and video on demand eventually
06:47.38Goshenvery nice
06:47.45Eightxkev: thanks... I thought I could get it without being so specific =)
06:48.01xkevthat's what naptr does
06:48.18xkevsuch a kludge :)
06:48.51EightI've got a friend who's into HAM radio and stuff...
06:49.03EightPondering how we might assemble some neat toys.
06:49.13NuggetI don't understand how people can be interested in ham radio now that the internet exists.
06:49.25EightNugget: the internet isn't everywhere =)
06:49.50NuggetIt's everywhere I want to be.  :)
06:49.57xkevthe internet exists on ham too
06:49.58EightNugget: It's not everywhere I want to be.
06:50.00xkevpacket radio
06:51.11*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
06:53.47jeffikpacket radio like GPRS
06:54.18xkevslow and shitty, exactly. :)
06:55.06Eightxkev: kevin@pbx.xmission.com is supposed to go to voicemail?
06:55.15xkevit'll ring a few times first
06:55.18jeffikyes slow and shitty but available where no wifi or dsl/cable is availablre
06:55.34Eightjeffik: packet radio can work where GPRS isn't available.
06:55.39Eightyou can MAKE it available where YOU want it.
06:55.45Eightas opposed to where the telco feels like it.
06:56.01xkeveight, but maybe there's some proxy auth getting in the way
06:57.49bonez39Goshen: have any idea how soon you could order fiber? or the cost?
06:58.08xkeverm, seems to work for me unauth
06:58.39*** join/#asterisk criptos (~criptos@201.135.97.238)
06:58.55xkevres_search OK (len=242)
06:58.55xkevNAPTR: _sip._udp.pbx.xmission.com
06:58.55xkevres_search OK (len=197)
06:58.55xkevSRV: 1,0,5060
06:58.55xkevSRV: pbx.xmission.com
06:58.55xkevSipClient: Sending to 'pbx.xmission.com:5060'
06:59.09criptosCan I use a fxo port and a sintax like Zap/1/1234 as an agent for a queue?
06:59.14Eightya, I'm not sure I've got my end configured very well atm.
06:59.25xkev(that's kphone output, btw)
07:00.10lqdengranyone know how to make x-lite tell * to play hold music when you put someone on hold?
07:00.25slePPhas anyone done load testing on asterisk/PRI to see how many calls/second you can drive through it?
07:00.29xkev* plays moh when it gets a sendonly
07:00.30Eight<PROTECTED>
07:00.30Eight<PROTECTED>
07:00.30Eight<PROTECTED>
07:00.30Eight<PROTECTED>
07:00.30Eight<PROTECTED>
07:00.31Eight<PROTECTED>
07:00.33EightThere's mine =p
07:00.41EightBut I don't hear a voicemail announcement.
07:00.47xkevdo you hear ringing?
07:01.01Eightya, I got the ringing, and the ringing stopped when it picked up.
07:01.05Eightthen nothing.
07:01.08Newbie___slePP: i dont think x-lite is capable of doing music on hold
07:01.09xkevodd
07:01.20xkevtry music@pbx.xmission.com
07:01.20EightI think it's a reinvite problem.
07:01.26EightI get a click.
07:01.35slePPNewbie___: i didn't ask :>
07:01.56Eightah, i'm an idiot.
07:02.01Newbie___slePP: sorry wrong person
07:02.04lqdengrlol
07:02.04Newbie___hehe
07:02.04Eightat some point I commented out my disabling of reinvite.
07:02.09Eightforgot to put it back =p
07:02.17slePPokay, who here hasn't entered the pastebin draw yet?
07:02.18xkevI had to reinvite=no for the polycoms to behave
07:02.22slePPhttp://pastebin.ca/draw.php
07:02.23slePPgo do so :>
07:02.25xkev..only had a problem on manager redirect though
07:02.43*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
07:02.49Newbie___hmmm let me see who asked that question earlier
07:03.04slePPand no, X-Lite doesn't provide MoH, the server would do that for you instead
07:03.31Eightxkev: heh, where is this music feed from?
07:03.31Newbie___i use firefly, mp3 music on hold
07:03.47xkeveight, we have like 50 hours of hold music
07:04.13xkevfrom our collections of downtempo and ambient.  there's a dash of sesame street here and ther etoo
07:04.36xkevI haven't counted :)
07:04.40slePPxkev: i use Elmo singing the alphabet for the support queue
07:04.51Eightwell, asterisk is using SRV properly, and I didn't even have to screw with it for 37 hours =)
07:05.05xkevyou need the drunken russian sailors singing 'my heart will go on'
07:05.20slePPxkev: that's in my personal collection, i bet ;>
07:05.25xkeveight, now I need someone with e164org to test with
07:05.40slePPe164.org!
07:05.42slePPthassme
07:05.47slePPwell, i'm part of it :>
07:05.48xkevword2u
07:05.54Eightxkev: what did you need to configure to get *incoming* SRV to work like it just did with me? other than the DNS record of course.
07:06.01xkevdns record
07:06.06slePPthough i haven't seen evilbunny in a while
07:06.31Eightxkev: doesn't there need to be a 'guest' user for random schmucks off the 'net?
07:06.37xkevthe dns just tells the calling UA where to send its packets
07:06.38slePPxkev: 8829900003305
07:06.41xkevoh yeah, that
07:07.04slePPof course, there's a chance those numbers are pointing at the wrong spot atm
07:07.25xkevI have context=anonymous in sip.conf [general]
07:07.27slePPyup, they are...
07:07.31slePPthey're pointed at the old server
07:07.51xkevand I created an [anonymous] with stuff I let people call, including some realtime derived from staff email addresses, etc
07:08.24nine76whome was it that came in and mentioned "I got it to work by deleting every folder found for a search on "asterisk" and recompiled,then it worked". Thanks to that guy:)
07:08.37Eightnine76: hah.
07:08.44nine76didnt wanna have to redo so many configs,but it works,so hapiness
07:09.12slePPokay, i'm done. time for beer'n'TV
07:09.13xkevI've got like 12 CVS versions under /usr/local/stow, and not a single problem with version skew
07:09.13Eightxkev: so there's nothing special to set, I'm already accepting anonymous connections?
07:09.15lqdengranybody have an opinion on switchvox?
07:09.33xkeveight, the default context in sip.conf handles that yeah
07:09.41Eightxkev: ah.
07:09.57xkev..I think :)
07:10.03xkevmine's all over the map
07:10.07Eighthah.
07:10.37xkevyeah, I bet that's what does it
07:11.00xkevsince the type=user entry for my SER proxy (where you really send port 5060 on this box) is typoed :)
07:12.13xkev(btw, all ser does is send presence subscribes to the right place, pretty much)
07:14.54jeffikanybody using asterisk@home?
07:15.08xkevI have a box warming the basement, yeah
07:15.34xkevcase ROSE_NAME_PRESENTATION_ALLOWED_SIMPLE:
07:15.35xkevmemcpy(call->callername, comp->data, comp->len);
07:15.45xkevdefault:
07:15.51xkevpri_message("Do not handle argument of type 0x%X\n", comp->type);
07:16.08xkevlibpri/pri_facility.h:#define ROSE_NAME_PRESENTATION_ALLOWED_SIMPLE     0x80
07:16.16xkev"Do not handle argument of type 0x80
07:16.18EightSomeone feel like testing something for me?
07:16.24xkevWTF?!?!?
07:16.58Eightxkev: hah.
07:17.34Eightxkev: I had a similar paradox earlier with /var/run/asterisk.pid
07:17.44EightI'm damned sure I know where I'm telling it to put it.
07:17.46Eightand it damned sure isn't.
07:19.07xkevI can test somehin for ya
07:19.18Eightcall weasels@3.141592.net
07:19.29xkevthat's a hell of a domain name
07:19.32Eight=)
07:19.38ikeyGoshen: saw the documents but these are for the advanced sip phone configurations
07:19.50EightThere's no SRV, that should use the A rec.
07:19.50xkevres_search: NO result !
07:20.24sleepy_onegnite y'all
07:20.36*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
07:20.39ikeyGoshen: i need to check with basic configurations
07:20.53PTG123Does ANYONE know how rto reset the admin password on a polycom?
07:21.03xkeveight, it figured it out, but it was slower not having the srv
07:21.04Eightxkev: you heard the weasels message?
07:21.06ikeycan any one help me in configuring sip channels?
07:21.30xkevbut kphone being a total piece of crap is now deadlocked :)
07:21.40Sedoroxikey: there is a TON of examples between the sip.conf and the wiki
07:21.41Eightxkev: cool, I just wanted to make sure I knew I was 'catching' the calls properly.
07:21.54Eightxkev: I'll poke the guy who runs my DNS records about it.
07:22.13PTG123Someone mus t know how to use the polycom phones :)
07:22.50xkevptg I haven't even changed my admin pass
07:23.09PTG123xkev: damn :(
07:23.24xkevI figure, if they want to get in and screw up their phone, then they can have no phone :)
07:23.32*** join/#asterisk TauReX (~james@colossus.trustmatta.com)
07:23.41PTG123xkev: can you set the admin password from the tftp server?
07:23.44PTG123when it does its update?
07:24.00xkevnot that I've found
07:24.02xkevI wish I could
07:24.20PTG123does the tftp stuff specify all the sip settings etc though?
07:24.27xkevyeah
07:24.39xkevbut I don't tftp, I use ftp (for the polycoms anyway)
07:24.39PTG123can you give me a copy of your setting files?  Maybe i can just set it all up that way
07:24.55PTG123any way to know which its set up to use
07:25.00PTG123what does the polycom specify for the login and password
07:25.01PTG123when you ftp?
07:25.28xkevPlcmSpIp for both, is default
07:25.41xkevand it'll get the server from dhcp op 66 (tftp server name)
07:25.46PTG123whats the default for the web iface?
07:25.55xkevPolycom and 456
07:26.02PTG123yah i think i tried that
07:26.08PTG123well so if i make a dummy ftp server
07:26.11PTG123and let it get the files
07:26.15PTG123it should fix my problem?
07:26.15xkevPolycom and your admin pass :)
07:26.21PTG123can you specify the tftp server to use in the feature?
07:26.29xkev'the feature'?
07:26.54rvhianyone knows how to have 2 instances of * on one server?
07:27.15xkevrvhi, if you ever figure it out, let me know :)
07:27.17Sedoroxrvhi: just have another bind to different IPs/ports?
07:27.36xkevyou'll have to keep all sorts of things from colliding
07:27.53rvhiquite a few things use sock
07:27.59*** join/#asterisk jmhunter (~jmhunter@64.77.200.148)
07:27.59*** mode/#asterisk [+o jmhunter] by ChanServ
07:28.09PTG123http://eknowledge.polycom.com/SRVS/CGI-BIN/WEBCGI.EXE/,/?St=14,E=0000000000001066275,K=1150,Sxi=3,Case=obj(34787)
07:28.15PTG123do those instructions look like they would work xkev?
07:28.15rvhithey won't work well with 2 instances
07:28.23xkevmake an /etc/asterisk2 and copy everything, change things to suit, and use asterisk -C /etc/asterisk2/asterisk.conf
07:28.54rvhidoes voicemail support vacation greating? something similar to the email vacation setting.
07:29.07rvhigreeting...
07:29.38xkevmain nasty collides would be /var/run/asterisk/asterisk.ctl and /var/lib/asterisk/astdb
07:30.14xkevbut if you do some fancy symlinking, you should be able to juust specify alt dirs in asterisk.conf
07:30.33rvhican you specify the name in two different *.conf?
07:30.36xkevrvhi, CVS has a temporary greeting option
07:30.55rvhithen load * with different .conf?
07:30.55ikeyhi can anyone help us out in configuring sip channels
07:31.12xkevrvhi, asterisk -C /etc/asterisk2/asterisk.conf
07:31.25xkevI haven't done this yet, but I will
07:31.32rvhicvs? i tried to avoid it as much as i can
07:31.55rvhican i just copy the app_voicemail.c to 1.0.7
07:31.58rvhiwould that work?
07:32.16xkevI need some public voip<->pstn and I don't want to wedge that simple setup into my 1500 line office pbx dialplan
07:32.26jjgis there no way to terminate through nufone via laptop?????????
07:32.28xkevrvhi, no
07:32.42xkevyou can look for the patch on mantis or asterisk-cvs list and try to backport it
07:33.16xkevstable has many many many major differences (such as flags and the channel struct)
07:33.17*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
07:33.50rvhiwhy wouldn't anyone backport these new features? realtime and voicemail?
07:33.56xkeve.g. chan->callerid in stable == chan->cid.num and chan->cid.name in cvs
07:34.06PTG123anyone know polycom phones
07:34.19xkevbecause cvs doesn't suck, and it has lots of things that make you hate stable :)
07:34.29xkevor rather, things you can't do in stable
07:34.38xkevptg123 lookin at your url
07:35.04jjgPTG123 : lots of luck...i thoght termination through nufone via laptop wold be an easy question
07:35.18PTG123jjg: whats you prob with that? :)
07:35.30jjgprob is , how to do it
07:35.54jjgdoes nufone support SIP?  not that i'm aware of
07:35.57PTG123thats why it sbetter to use companies with good support who would help with those things :)
07:35.57xkevptg123, on your link: try it
07:36.08PTG123xkev: i don't see those menus
07:36.11PTG123xkev: are there?
07:36.17rvhiwhen is cvs becoming the new stable? is it on the horizon? or i have no hope now?
07:36.37xkevyou're looking at a doc for VSX, not SoundPoint IP
07:36.44xkevI don't have a phone at home today
07:37.08jjgPTG123 : what is your point in asking the problem without offering someing like a solution?
07:37.12xkevbut look for a menu item that is reset factory defaults, I think I recall one being there
07:38.00xkevrvhi, I wish I knew.  I'm going live with my CVS after exhaustive testing and minor bug fixes (which incidentally, this latest bug exists in stable too, heh)
07:38.01PTG123jjg: b/c i am not gonna take you step by step how to do something the only reason you can't get it doen is b/c you were too cheap to spend an extra .5c :)
07:38.15jjgdoesn't it kinda mean that nufone sucks if there isnt a softphone?
07:38.17PTG123xkev: can't find it, sucks..
07:38.31PTG123jjg: well for a softphone use xlit
07:38.31jjgPTF123 : don't follow, sorry
07:38.38jjgthrough nufone?
07:38.39PTG123i use xpro + bluetooth headset
07:38.40xkev..but I must have cvs for some features
07:38.44PTG123but nufone doesn't support sip directly
07:38.54jjgwell like i said
07:38.54PTG123use firefly
07:39.01jjgi didn't say nufone + asterisk
07:39.02PTG123but like i said
07:39.03xkev..for a simple pstn gateway with some voicemail and a feature or two, I'd use stable
07:39.11PTG123i don't think nufone is the best thing to use for that
07:39.23PTG123find a good cvs build, and stick with it
07:39.26PTG123thats what i did
07:39.28jjg...s/nufone/laptop
07:39.57xkevptg, I keep patching so I have to keep updating to get my features committed :)
07:40.01jjgso there is no fucking way to terminate through nufone without an asterisk box?  jesus
07:40.27PTG123xkev: their doesnt seem much interested in submitting my features.. so i don't wanna waste the time even committing them :)
07:40.39PTG123jjg: i gave you the answer
07:40.46PTG123jjg: but this is #asterisk not #nufone :)
07:40.49jjguh, and the answer was?
07:40.52PTG123so why would anyone want to help you
07:40.53jjgi'm on #nufone
07:40.58jjgthey have shit to say too
07:41.06PTG123PTG123: but nufone doesn't support sip directly
07:41.06PTG123PTG123: use firefly
07:41.17PTG123use teliax, i know that guy would help you out :)
07:41.36jjgand firefly is a protocol?...how does firefly relate to sip?
07:41.43PTG123~firefly
07:41.44jbotsomebody said firefly was http://virbiage.com/firefly/download/firefly-thirdparty.exe
07:41.46Eight<PTG123> find a good cvs build, and stick with it <-- That's what releases are supposed to be! =)
07:41.47PTG123since nufone doesn't support sip
07:41.51PTG123has nothing to do with it
07:41.57xkeveight, we need a 1.1 yes
07:42.04PTG123~firefly
07:42.06jbotfrom memory, firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe
07:42.23xkevI'd like to see a freeze in the timeline
07:42.29Eightjbot jbot?
07:42.30jbotjbot is probably the shipboard computer, but you may call me eddie if it helps you relax
07:42.41Eighteddie eddie?
07:42.42Eight=p
07:42.45shido6whats up?
07:42.51`SauronEight, do you never sleep?
07:42.54shido6watching the Detroit / Boston Game
07:43.03Eight`Sauron: =)
07:43.26ikeyjjg: can u help me out in configuring sip channels for outbound
07:43.59jjgwuss the issus?
07:45.08*** join/#asterisk file (~file@251.134.218.209.transedge.com)
07:45.13jjgbesides, ikey
07:45.35jjgyou told me earlier today that you had 120 simultaneous calls going on a single CPU
07:45.41jjgso what's the problem?
07:46.07jjgforgot how you got all those calls rolling?
07:46.55shido6ok back , timeout
07:48.11rvhiis there a way to define the extension to reach after user presses 0 in voicemail?
07:48.29rvhie.g. ceo's voicemail, press 0 goes to secretary
07:48.41rvhiother people, press 0 goes to receptionist
07:49.23rvhithe special extension 'o' is reached after 0 is pressed
07:49.41rvhihow to i know if the original caller is in which email?
07:50.30*** join/#asterisk porkchop (~porkchop@porkchop.nat.cccp.porkchop.net)
07:51.12porkchopI'm having a problem with callerid. No matter what I do, inbound calls appear on my screen as "porkchop", the username for the phone. Any ideas? I'm not using fromuser= (anymore) in sip.conf ....
07:51.16JmanA9can anyone recommend a good asterisk gui?
07:51.39PTG123can you buy 48v adapters in stores?
07:51.41JmanA9one for changing config files and stuff, not a softphone ;)
07:52.22ikeyjjg: yeah i think i told u that on PSTN e1 Lines
07:52.33shido6double overtime
07:52.44*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
07:52.56ikeyjjg: so was trying to work on SIP too for outbound calls
07:53.08jjgok
07:53.15PTG123JmanA9: there really isn't a good one publically available
07:53.17jjgit's easier than outboudn pstn
07:53.20jjgwhat's the prob?
07:53.33JmanA9ok, editing config files it is :)
07:53.58EightJmanA9: it's just as well... it means you can SSH in and do everything as normal =)
07:54.01ikeyjjg: yeah i tried with grandstream and configured sip.conf and extensions.conf
07:54.04*** join/#asterisk jeffik (~jeffik@me97b36d0.tmodns.net)
07:54.48ikeyjjg: asterisk can be configured as sip client as well as sip server ...i think there was some confusion when i configured
07:55.00jjgikey : i'm sorry for the sarcasm, i'm totally exhausted ... pastebin yrou stuff and i'll chekc it in a coupla hours...private message me
07:55.20ikeyjjg : ok great
07:55.25ikeywill do that now it self
07:58.06shido6hehe
08:01.04PTG123a sip client is a sip server, really no difference :) to be one you need to be the other
08:01.09rvhixdev, i looked the cvs app_voicemail.c code
08:01.20rvhican't find anything about vacation greeting
08:03.42*** join/#asterisk eric_ (Ap0ll0@modemcable081.176-201-24.mc.videotron.ca)
08:03.57eric_anyone awake on this friday night?
08:04.18Eighteric_: nope.
08:04.33jeffikeric_: only those who need help
08:05.13tuxinator_linuxMI am
08:05.20eric_haha
08:05.25eric_nice to see some other insomniacs out there
08:05.44porkchopinsomnia? Its only 314am
08:05.50porkchop:)
08:05.52jeffikanyone work with *@home?
08:05.53eric_:P
08:05.53tuxinator_linuxMWhat does the sun look like?
08:05.54Eightmmm... pi.
08:05.58eric_yeah, so basically, I compiled asterisk, and am using default config, but when i start it up, it segfaults
08:06.10Eighteric_: lol.
08:06.10tuxinator_linuxMjeffik: I hear it works well
08:06.11Eighteric_: nice =)
08:06.13porkchoptuxinator_linuxM: you mean...the day star?
08:06.14eric_does anyone have any pointers where to start? google hasnt found me anything useful :/
08:06.17eric_thanks eight :P
08:06.32tuxinator_linuxMeric_: hardware?
08:06.36tuxinator_linuxMWhat is your
08:06.47tuxinator_linuxMOS?
08:06.49Eighteric_: It sounds like you have problems that aren't terribly related to asterisk... is my first (uneducated) guess.
08:07.03eric_well, its all emulated hardware, its RHEL running under UML
08:07.06drumkillatuxinator_linuxM: how was training
08:07.12Eighteric_: well there ya go =)
08:07.16eric_someone else is doing the same thing and its working for him though
08:07.24Eightok, I'll just shut upnow.
08:07.28tuxinator_linuxMdrumkilla: well, good and bad...
08:07.30Zeeekeric_  you have zaptel hardware?
08:07.30eric_haha dont shut up
08:07.49EightAlright, to heck with extensions.conf
08:07.53eric_no hardware being used -- its at a remote location... im just going to use it to route
08:07.55tuxinator_linuxMdrumkilla: Good cuz I reallize I know * pretty well...
08:08.00drumkilla:)
08:08.10Eightexten => _.,1,agi,everything.py ; =p
08:08.11Zeeekeric_ when you start do yiou get anything before the fault? any messages?
08:08.15tuxinator_linuxMdrumkilla: Badd cuz I wasted $200 to learn that I already know
08:08.17Newbie___can anyone please give me a 1800 number to try out the sound quality, i am not from the US
08:08.17eric_no messages at all
08:08.32drumkillatuxinator_linuxM: well it sounded like it was intended for people that didn't know anything about it
08:08.33jeffik*@home uses amp gui
08:08.35eric_but i can do asterisk ---nothing, and it tells me "error, no option"
08:08.39Zeeeksounds like it may be mobo related - fcertain chipsets exhibit this behavior
08:08.56eric_Zeeek: me? all my hardware is basically emulated
08:09.02tuxinator_linuxMeric_: use -vvv
08:09.12eric_thats what im doin :P
08:09.17tuxinator_linuxMk
08:09.21eric_hehe
08:09.23tuxinator_linuxMnoting in log fiel?
08:09.28eric_nothing at all
08:09.31eric_core dump actually :P
08:09.42Newbie___any 800, 888 number i can test dial to the US ?
08:09.56eric_8009993355 -- its dell ;)
08:10.09Newbie___tks
08:10.26eric_i'm stumped... i have no idea where to look first
08:10.36jeffikany canadians on tonight
08:11.01eric_anyone else have any delightful insight?
08:11.16PTG123anyone awake now know anything about polycoms? :)
08:11.24PTG123drumkilla: you don't by chance?
08:11.29tuxinator_linuxMPTG123: their good
08:11.47Newbie___trying out simpleconnect.com with firefly IAX, couldnt hear a thing calling 800 999 3355
08:11.51PTG123tuxinator_linuxM: little more info then that :)
08:11.55jeffikno canadians?
08:12.00shido6whats wrong with the polycom?
08:12.09shido6[02:52] <shido6> big project
08:12.09shido6[02:52] <shido6> heading to bed
08:12.09shido6[02:52] <shido6> greg@nufone.net
08:12.09shido6[02:52] <shido6> IM: shido6@msn.com
08:12.11PTG123shido6: i can't find out how to reset admin password
08:12.12shido6crap
08:12.18shido6then
08:12.19shido6err
08:12.24shido6put next server in the dhcp server
08:12.24PTG123shido6: any idea?
08:12.27shido6as your tftp box
08:12.29tuxinator_linuxMPTG123: I havevn't played with them yet
08:12.41PTG123shido6: what do you mean?
08:12.43shido6and reboot that summu mumma snitch
08:13.02shido6err
08:13.04shido6http://lists.digium.com/pipermail/asterisk-users/2004-October/069585.html
08:13.06shido6try that
08:14.08PTG123in the cfg files can you specify admin pwrd?
08:16.08shido6stdby
08:17.45porkchopI'm having a problem with callerid. No matter what I do, inbound calls appear on my screen as "porkchop", the username for the phone. Any ideas? I'm not using fromuser= (anymore) in sip.conf ....
08:18.10tuxinator_linuxMporkchop: I think it is a bug
08:18.26tuxinator_linuxMcheck bugs.digium.com
08:23.08shido6ok
08:23.09shido6To reset a forgotten Admin Password:
08:23.09shido61. Get the system’s serial number from the system or from the System
08:23.09shido6Information screen.
08:23.09shido62. Go to System >Diagnostics > Reset System.
08:23.09shido63. Enter the system’s serial number and select Delete System Settings.
08:23.10shido64. Select Reset System.
08:24.16PTG123shido6: i don't have a system diagnostics screen
08:24.55porkchoptuxinator_linuxM: can't seem to find a specific report that applies to me. I'll keep working on it I suppose.
08:25.11shido6sorry
08:25.12shido6im an idiot
08:25.14eric_hrmm, it seems to sefgault after parsing an almost-empty extconfig.conf
08:25.14shido6wrong box
08:25.52shido6thats for the viewstation
08:25.53shido6sorry
08:26.05PTG123damn :(
08:26.08PTG123any other ideas?
08:27.31*** join/#asterisk santiago (~santiago@63.245.86.95)
08:27.43shido6still lookin
08:29.12*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)
08:29.25*** join/#asterisk Zgarbi (~my@212.58.125.68)
08:30.32*** join/#asterisk tessier (~treed@210.245.96.123)
08:31.34*** join/#asterisk herag (herag@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
08:32.28heragwhat does it mean when I get an SIP 400, bad request when trying to take an incoming call and dial forward it out to another number?
08:33.46shido6I dont see anything PT
08:33.51shido6u have physicall access to the phone?
08:33.54heragI hadn't changed any configs, but suddenly I'm getting these weird sip 400 errors...but it only happens when a call tries to be forwarded out, I can dial straight out just fine, and calls can go into my device, but if I try to take a call coming in from broadvoice, and then forward it out to another number, it won't work anymore
08:33.56PTG123shido6: yah me either, not sure what to do
08:34.04shido6you could do the "short the damn thing" trick
08:34.17shido6unplug , spark , plug , spark, plug
08:34.26shido6but you risk turning the phone into a paperweight
08:34.36shido6hopefully it will kickstart to factory defaults
08:34.45shido6u know what I mean by spark?
08:34.53shido6plug the power in just enough to spark it
08:34.56shido6and unplug
08:34.59shido6then plug it back in
08:35.03shido6then unplug it and spark again
08:35.12PTG123that works?
08:35.14PTG123its poe though
08:35.19shido6oh lord
08:35.43shido6Im trying my damndest to not say, "fou're yucked."
08:36.03PTG123haha
08:36.07PTG123well they have shit on their site
08:36.12PTG123on how to reset everything but this phone
08:36.54*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
08:37.26modulus_jbot stable?
08:37.27jboti guess stable is the status of a Debian release when no packages will be added or changed unless a security fix is needed, or sta-ble adj; uptime in excess of 365days, or where the horses live  The current stable version of Debian is woody (3.0).
08:37.31modulus_wtf
08:37.35modulus_jbot stable asterisk?
08:37.54modulus_jbot you whore of a bot, tell me the latest stable version you bitch machine
08:38.13drumkilla1.0.6 - 1.0.7 will be very soon
08:38.40modulus_jbot should divulge that info
08:38.49drumkillait's in the topic ...
08:39.03modulus_this irc session is screen'd
08:39.13drumkillafeel free to go test it for me :)
08:40.27footnoteWhere the Horses Live Linux(TM)
08:43.10rvhitry to compile voicemail with mysql support
08:43.14rvhiget this app_voicemail.c:371:31: mysql-vm-routines.h: No such file or directory
08:43.24rvhiany suggestion?
08:43.51xkevast--                       switch (comp->type & PRI_DEBUG_APDU) {
08:43.51xkev+                       switch (comp->type) {
08:43.58xkevs/ast--/-/
08:44.12xkevthat typo was breaking caller name facility message.
08:44.31xkevEight, ^^
08:46.23godsmokewohoo
08:46.25godsmokeFWD on my cisco
08:46.27godsmokeit all works
08:47.09ta[i]ntedwhat's up with the crazy
08:48.05drumkillarvhi: it's in asterisk-addons
08:50.55rvhii found it in asterisk-addons 1.0.6
08:51.00rvhii was using 1.0.0
08:51.12rvhiguess was added after 1.0.0
08:51.19drumkillanah
08:51.24drumkillait was always in addons ...
08:51.39rvhimaybe i didn't get the right addons for 1.0.0
08:51.46rvhihard to find it now... :)
08:51.50drumkillano big deal
08:51.57rvhianyone i copy the file from 1.0.6
08:51.58drumkillayou want to be running the new stuff anyway
08:52.05rvhipraying it is going to work
08:52.12xkevlousy mysql licensing
08:52.12drumkillarvhi: you should run the code from stable cvs at the moment, actually
08:52.28drumkillathen, make a note on bug 3746 that you're using it without problems
08:53.11rvhii made some code change to 1.0.0
08:53.32rvhigoing to take some time to merge my change to 1.0.6/stable
08:53.45drumkillaok ...
08:53.54*** join/#asterisk Tommmo (~tps@203.62.181.52)
08:54.36Tommmoanyone know if it's possible to configure the call progress tones on cisco 7940/60 ?
08:55.55godsmokewhat are call progress tones?
08:56.33*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
08:57.13Tommmoe.g. the tone you hear when you call someone, and you are waiting for them to pickup
08:57.18Tommmobusy tones, etc
08:57.22godsmokeah
09:02.02*** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194)
09:12.00modulus_i think someone needs to port asterisk-addons to the freebsd ports tree
09:15.24ikeycan anyone explain how to configure asterisk as sip client and server
09:16.28Zeeekis it possible to specify callerid in a .call file?
09:17.09Zeeeknever mind
09:17.11Zeeekgot it
09:17.21srtZeeek: sure look at sample.call
09:17.25Zeeekgot it
09:19.03PTG123anyone know how to make the polycom webserver answer? :)
09:19.07PTG123i got into the phone to config it
09:19.11PTG123but now webserver won't respond
09:19.45Zeeekhow did this problem come about? I'm considering buying a Polycom
09:19.59Zeeekdid you just forget the password?
09:21.10PTG123i got it from ebay :)
09:21.14PTG123buy the cisco
09:21.17PTG123i bought both
09:21.19PTG123cisco is much better
09:21.20PTG123:)
09:21.40Zeeekthe poly I'm considering can be had new for $220 or so
09:21.46Sedoroxhmm
09:21.48Zeeekthe cisco is more I think
09:21.52*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
09:22.00PTG123umj
09:22.05PTG123i got one new for $137 :)
09:22.06Sedoroxthe channel I think, over all likes polycom more
09:22.08PTG123for a 7960
09:22.16PTG123and my cisco is a 6 line
09:22.25ZeeekI don't do ebay though
09:22.36Zeeekso I'm talking new from a distributor
09:23.21*** join/#asterisk brimston3 (me@146.229.178.20)
09:24.09brimston3i'm new to asterisk, is there a quickstart guide to getting a TDM11B running on a debian box with 2.6.8 ?
09:24.23ZeeekStarter tutorial:
09:24.23Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
09:24.23Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
09:24.23Zeeekhttp://www.automated.it/guidetoasterisk.htm
09:24.23ZeeekTHE reference of the moment:
09:24.24Zeeekhttp://www.asteriskdocs.org
09:24.31Zeeektry number three above
09:24.42brimston3thanks Zeeek
09:24.49Zeeekthen search for 2.6 kernel woes
09:25.05Zeeekon the wiki
09:26.55brimston3number three as in automated.it right?
09:26.59brimston3or asteriskdocs ?
09:27.15Zeeekautomated has a complete guide
09:27.20Zeeekbut not 2.6
09:27.24brimston3ah
09:27.34Zeeekalso the latest asteriskdocs.org has a complete setup guide
09:27.47Zeeekalso the wiki has all the stuff but not easy to find
09:28.04Zeeekhttp://www.voip-info.org/wiki-Asterisk
09:28.26Zeeekhttp://www.voip-info.org/wiki-Asterisk+installation+tips
09:28.46ZeeekLook for 2.6 HERE : http://www.voip-info.org/wiki-Asterisk+OS+Platforms
09:28.57brimston3thanks
09:32.43*** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.rr.com)
09:32.50*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
09:34.47*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
09:35.33PTG123anyone know why i can dial numbers with my polycom but it shows unregistered?
09:37.14Zeeekbeacuse there's no password in the peer/user/friend entry?
09:37.27PTG123call wouldnt work if their wasn't
09:37.31Zeeekbecause default has all possible extensions?
09:37.32GMsoftis there a command to send MWI messages for SIP ? I've got many asterisk and a centralized voicemail here and I can't find such command on voip-info
09:37.58PTG123Zeek: no its working right with my sip account
09:38.02PTG123Zeek: just not registering
09:38.06ZeeekPTG is it unregistered or unreachable?
09:38.13PTG123unregistered
09:38.18PTG123think it just never sent the register package
09:38.21PTG123its not registering
09:38.25Zeeekas in not in the show peers list?
09:38.59PTG123don';t believe so
09:39.14Zeeekbelief has nothing to do with voIP
09:39.25Zeeekit is or is not in the list
09:40.57tuxinator_linuxMMan, I'm tired
09:41.02Zeeeksleep
09:41.08Zeeekyour eyes are closing
09:41.12Zeeeksowly...
09:41.17tuxinator_linuxMyah they are
09:41.19Zeeekslowly even...
09:41.33Zeeekthink of J'Lo
09:41.40tuxinator_linuxMyou should of seen file fall asleep with his hands on the keyboard
09:41.55Zeeekdid it look like this?
09:42.03Zeeekfffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff
09:42.20modulus_beep!
09:42.26Zeeekmeep
09:42.31tuxinator_linuxMwe got a picture, I think it is on drumkillers camera
09:42.38modulus_zeeek
09:42.47Zeeekmeeep
09:43.01Eightergh... wtf.
09:43.09EightSo... I'm playing with AGI.
09:43.15ZeeekEight wft are you doing up at this hour?
09:43.23EightI SAY NUMBER twice, and then "HANGUP".
09:43.30Eightand my SIP client hears both numbers...
09:43.34Zeeekno one does good AGI at 4AM
09:43.44EightZeeek: depends on when they woke up =)
09:43.54Zeeekno, it's an absolute
09:44.08Zeeek4AM is the hour where most people die at hospitals
09:44.13EightAnyways... HANGUP seems to send 403 forbidden.
09:44.21Eight*even though* the call was already in progress.
09:44.24Eightit's fucked up.
09:44.34Zeeekwrite a module
09:44.41Zeeekyou'll have more control
09:44.44EightZeeek: god no. I want to use python not c.
09:44.47*** join/#asterisk brimstone (~brimstone@146.229.178.20)
09:44.50Zeeekmuhahaha
09:45.08EightI was thinking last night "hey, maybe I'll actually try using C for something other than a CompSci course..."
09:45.20Eightafter 2 hours bouncing around the asterisk source... to hell with that.
09:45.27Zeeekhttp://wiki.e164.org/moin.cgi/WordIndex
09:46.13Zeeekc is like the molecule that all unix is made from
09:46.27EightZeeek: I know.
09:46.27Zeeekit's the one thing they have in common
09:46.38Zeeekand it's so easy and powerful
09:46.50EightZeeek: hahaha.
09:47.01Zeeeksomeone might make up some handy macros for use with asterisk though
09:47.06EightZeeek: spend some time with a decent language some time, you'll NEVER go back to C.
09:47.08Zeeekthat would help a lot
09:47.16Zeeekname a decent language?
09:47.25Zeeekbesides py
09:47.28EightPython, Java, Ruby...
09:47.37EightLisp.
09:47.44EightSmalltalk
09:47.50brimstoneperl?
09:47.51Zeeekah you want objects... use c++
09:48.02Eightbrimstone: perl's ok, if you can code responsibly enough.
09:48.15Eightbrimstone: but it's really conducive to writing difficult to read code.
09:48.24EightZeeek: C++ is an abomination.
09:48.29EightObjective C is much, MUCH better.
09:48.35Zeeekanyway, great thngs have been done with all languages, so it's infantile to argue for one or the other. Only your own limitations are being argued. (someone write that down)
09:48.48Eightesp' in the NeXTSTEP/GNUStep/Mac OSX environment
09:49.18EightZeeek: Are you arguing that Asterisk might as well have been written in COBOL? or Assembler?
09:49.32EightZeeek: There's a right tool for the job, and there are new developments in tools.
09:49.33Zeeekagain, useless conjecture
09:49.44ZeeekEight yes, but you are arguing your opinion
09:49.52EightZeeek: I have nothing else to argue =)
09:50.03ZeeekI didn't mention COBOL which is made to do certain things as you note
09:50.04EightZeeek: Seriously, do you have much experience with languages other than C?
09:50.09Zeeekare you over 20 ?
09:50.12Eightyes.
09:50.19Zeeekhow much over
09:50.41Zeeekor, if you prefer what is the year you made your first program?
09:50.49Eightoh geezus.
09:50.50EightI have no idea.
09:50.52Zeeek(and in what lmanbgua)
09:50.56Zeeeklanguage?
09:50.57EightIt was in C.
09:51.04Zeeekwhat decade?
09:51.27EightI'd have to say... early 90s.
09:51.44Zeeekthat's a decent exposure
09:51.50EightZeeek: oh gee, thanks.
09:51.53Zeeekso I'm surprised you want to make these statements
09:52.14EightZeeek: What experiences make your opinions better than mine? =p
09:52.17Zeeekbut I do think you should use whatever language you want
09:52.19Zeeeknone
09:52.29Zeeekthat('s myy whole point - you are full of sheisse
09:52.43Zeeekno one is capable of maintaing these kinds of args
09:52.56Zeeekso you're just talking at 4AM
09:53.08Zeeekand not getting your AGI to work
09:53.13Zeeekin py
09:53.21EightDude, the python code is like 6 lines long.
09:53.27EightIt's the HANGUP command that's being weird.
09:53.31Zeeekeasy to debug then
09:54.06Zeeektalk to people on dev about this, maybe they can "fix" it
09:54.09EightZeeek: let me know when you fall off your high horse and I'll start speaking to you again =)
09:54.17Zeeekyou misunderstand
09:54.32ZeeekI don't care, I just think these blanket statements are silly
09:54.46ZeeekI do not defend or recommend any distro or language
09:55.20EightAh, now you've said something concrete.
09:55.26ZeeekI said it before
09:55.36ZeeekI was just curious about your experience - and now I know
09:55.47Zeeekhave you slept much ?
09:56.08EightZeeek: yes, I've only been awake 8 hours maybe (don't recall exactly).
09:56.21Eightok, my turn to quiz.
09:56.27EightWhat languages have you had serious exposure to?
09:56.28Zeeekok, I think you read some kind of animosity in to what I was trying to say
09:56.40ZeeekEight "serious" I don't know
09:56.53EightZeeek: you were wholly discounting my opinions prima fascia... that's generally a hostile thing to do =)
09:56.53ZeeekI'm a lousy programmer - let's get that out of the way
09:57.11Zeeekno I discount ALL opinions that say "sucks" etc
09:57.14Zeeekbut let me answer
09:57.39ZeeekI began with Z80, 6800 and 6502 machine, then assebly for same
09:58.09EightZeeek: You may be a lousy programmer, but perhaps it's just because you've not found a language that lets you spend enough time expressing yourself, and less time controlling bits (or in C's case, pointers).
09:58.11Zeeekthen did Fortran (ans basic obviously) then used PDP11 under two OS, then VAX11 assembly
09:58.22Zeeekno I am a lousy programmer
09:58.40Zeeekthen worked ona system that had terminals that spoke Forth
09:58.55Zeeekand installed forth on my little Radio Shack with 32k
09:58.57Eightalright, it's not AGI... for some reason my client is getting 403 all the time when I use hangup right now.
09:59.24Zeeekagain, concretely, talk to the guys on dev (or wait until way later int he day)
09:59.34Zeeekthere's always the mailing list too
09:59.47Eightexten => t,1,hangup
09:59.48Eightno error
09:59.52Eightexten => 2,1,agi,default.py
09:59.52Eightexten => 2,2,hangup
09:59.53Eighterror
10:00.01EightIt's not really an error, exactly...
10:00.04Zeeekstack problems?
10:00.13Eightit's just the SIP client gets a 403 instead of exiting cleanly when the connection drops.
10:00.47Zeeekwhat's the mast line of the AGI?
10:00.50Zeeeklast line
10:01.13Eightexten => 2,1,agi,default.py
10:01.13Eightexten => 2,2,saynumber(9)
10:01.13Eightexten => 2,3,hangup
10:01.17Eightstill 403.
10:01.27Eightand I hear '9' before it drops me.
10:01.32Zeeekwhat's the last line of the python AGI?
10:01.39Zeeekor last few
10:01.52Eightactually, I lied, the script is 3 lines long =p
10:02.13Eightsys.exit(0)
10:02.25EightI put it in explicitly, jus tin case.
10:02.32Zeeekok then the first two
10:03.11Eightdude, all it does is send "SAY NUMBER 1 """ and "SAY NUMBER 2 """ to stdout and flush
10:03.25EightI really, really don't think it's my code =p
10:04.36RaYmAn-Bxhave you checked it actually runs if you run it manually?
10:04.37*** join/#asterisk coppice (~chatzilla@111.196.17.210.dyn.pacific.net.hk)
10:04.40Eightexten => 2,1,saynumber(8);agi,default.py
10:04.40Eightexten => 2,2,saynumber(9)
10:04.40Eightexten => 2,3,hangup
10:04.47EightI still get 403 =p
10:04.55Eightand my code doesn't even run.
10:05.01Eightit's an extensions.conf issue.
10:05.33Eightstop talking about my code =)
10:05.33Zeeeksignificant debug element there
10:05.33djorangehi hi
10:06.14*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
10:06.32RaYmAn-BxEight: not sure if it's needed, but have you considering doing an Answer first?
10:06.38RaYmAn-Bxconsidered*
10:07.12djorangeokay i got two trunks coming in and i need to get one trunk to hit auto attendent1 and then transfer to group 200.
10:07.21EightRaYmAn-Bx: aaah, good call.
10:07.43EightRaYmAn-Bx: damn =/
10:07.49EightI really thought that might have been it =
10:08.57Eightoh, it was.
10:09.01EightForgot to reload that time =p
10:09.15EightI guess the client just thinks the call fails unless it gets 'answered' properly.
10:09.29Eightand 'saynumber' doesn't "answer" properly.
10:10.08Eightif I use playback, I don't get 403.
10:11.10EightRaYmAn-Bx: Thanks man =)
10:11.21EightZeeek: next time when i say it's not my code... it ain't my friggin' code =)
10:11.29Zeeek"New versions of Asterisk have added "Answer" capabilities to several functions like Playback(), which means that those functions will answer themselves if necessary. "
10:11.42EightYa... I think that's actually a BAD idea, to be honest.
10:11.51Eightit promotes problems like I just had =p
10:11.54ZeeekI never said it was - again - why the hostility? - I was curious to see what your py looked like
10:11.56Eight(and I thought it was a bad idea before this!)
10:12.26RaYmAn-BxEight: regardless of how correct your code is it's always a good idea to run it manually and make sure it doesn't have any stupid errors :P
10:12.33EightZeeek: I wasn't being hostile, just trying to make a joke =)
10:12.51EightRaYmAn-Bx: Ya, I'm actually watching from the Asterisk console. if Python errors it displays there.
10:13.02*** join/#asterisk Inv_arp (junya@adsl-8-232-168.mia.bellsouth.net)
10:13.13ZeeekEight I have no doubt that anyone here is stonger in programming that I am and wouldn't question their stuff
10:13.15EightRaYmAn-Bx: and I *know* it was all working, because the only code in the file results in an audible playback on the channel, which I was hearing.
10:13.46ZeeekEight however I have lived lionger than anyone here and sometimes questiion the wisdom of some statements
10:14.32Zeeeks/wisdom/"Absolute Veracity"/
10:14.37Eight=)
10:14.45*** join/#asterisk ikey (~ikey@202.54.37.183)
10:15.16EightI was going to say, "I do mistake age for wisdom, nor experience for expertise."
10:15.20Eightbut you defused it =p
10:15.26Eighterr, 'do not'.
10:15.38ZeeekEight if I may drag this on one more time, a lot of people speak of things they know little about, for example I have never used H323 and wouldn't say it sucks
10:16.01Zeeekyet I see lots of people talking about countries they have never visited
10:16.11Zeeekor people they've never met
10:16.16EightZeeek: I have used C, and in my opinion it does suck for all but very specific tasks (building operating systems is a specific task).
10:16.24ZeeekI know and understand
10:16.45Eightactually, it quite excels at building operating systems.
10:17.04ZeeekMy own prejudice wrt to asterisk devel is that I don't like talking to stdin/out
10:17.14Eightyou mean w/ AGI?
10:17.14Zeeekit seems flaky to me
10:17.18Zeeekyeah
10:17.25Zeeekdoesn't feel right
10:17.32ManxPower~docs
10:17.33jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
10:17.37Zeeekbut that's just my own experience with API
10:17.39EightIt's *really* common to do things that way in Unix.
10:17.48ZeeekGo back to bed Manx !
10:18.44ManxPowerZeeek, I just slept 8 hours.
10:18.51Zeeekok then you're cool
10:19.04Zeeeksomeone was having big Polycom problems a while ago
10:19.10RaYmAn-Bx8 hours just isn't enough :/
10:19.25ManxPowerBut as a cat I know there is no such thing as "too much sleep"
10:19.27Zeeek~seen PTG123
10:19.29jbotptg123 is currently on #asterisk (2h 58m 53s).  Has said a total of 83 messages.  Is idling for 40m 30s
10:19.39Zeeekhehe
10:19.59ZeeekManx I'm watching the Polycom stuff carefully as I want to buy one
10:20.18Eight~seen
10:20.22Eight~seen eight
10:20.23jboteight <~blake@12-205-155-39.client.mchsi.com> was last seen on IRC in channel #asterisk, 1s ago, saying: '~seen eight'.
10:20.39Eightwhat... no messages?
10:20.54Zeeek~seen Mel Gibson's latest movie
10:20.55jboti haven't seen 'mel gibson's latest movie', Zeeek
10:21.12Eighthow do you add info to jbot?
10:21.13Zeeek~got milk
10:21.14jbotACTION chugs down a carton
10:21.22Zeeeklazy bot
10:21.34Eightjbot jbot?
10:21.35jbotmethinks jbot is the shipboard computer, but you may call me eddie if it helps you relax
10:21.53Zeeek~eddie
10:21.54jbotRobust, clustering, load balancing, high availability, web server tool.. URL: http://www.eddieware.org/
10:22.07Eight~agi
10:22.08jbot[agi] the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages
10:22.16*** join/#asterisk wasim (~wasim@203.81.216.2)
10:22.24EightZeeek: there, go get pyst. It does the API for you =p
10:22.33ZeeekRappers are even better
10:22.52Zeeek"I don't need no AGI... I got it all in mah CLI!"
10:23.23Zeeekok it must be late for humor
10:24.10Zeeekgo for it!
10:24.24Zeeek(sorry, forgot the duuuuude)
10:24.45ManxPowerCommand Line Interface Terminanal?
10:25.23ZeeekTiming With Asychronous Tones
10:26.04ZeeekPush Under Symmetrical Signalling Yeti
10:26.32EightManxPower: was 'terminAnal' intentional? =/
10:26.58EightI didn't even see it for a while =p
10:27.15Zeeekyou are supposed to be implementing
10:27.21EightI can type and type at the same time.
10:27.38Zeeekmultiple windows? When did that get invented?
10:27.51ZeeekWhat editor do you use, Eight?
10:27.57Eightxemacs
10:28.04Eightor, vi, depending on what I'm doing.
10:28.09Eightor TextEdit.app.
10:28.10Zeeekok, you're not a vi nazi
10:28.25Eightvi is for config files, xemacs is for code, textEdit is for everything else.
10:28.41Zeeek"the right tool..."
10:29.20ZeeekI must be a wimp for using pico/nano - though I have used and liked emacs
10:29.43EightI used to use pico for a while, until I got around to learning vi.
10:29.46ZeeekIn fact I did use "mouseemacs" years ago
10:29.48EightI still use pico in pine.
10:29.58Zeeekpine! whoa
10:31.07ta[i]ntedpine was made at my old university
10:31.33modulus_how do i ignore exten => i,1,??
10:31.37coppiceyou went to university in a lumber forest? :-\
10:31.51Inv_arpbah paco...nano :)
10:31.55Eightmodulus_: what do you mean, ignore?
10:32.06Eightexten => i,1,noop
10:32.09modulus_ignore invalidly pressed extensions
10:32.16modulus_NoOp does not ignore
10:32.23modulus_it takes over the context and extension
10:33.07Zeeekcoppice SawMill U.
10:33.34modulus_zeeek, help me out here?
10:33.47Zeeekwhy do you want to ignore ignore?
10:34.00ta[i]ntedpretty close
10:34.12ta[i]ntedthere are a lot of trees in my state
10:34.14Inv_arpmodulus_:  exten => i,1,Goto(s,1) ; go back to first step?
10:34.18modulus_i want my menu to continue even if invalid exten
10:34.21ta[i]ntedcoppice university of washington
10:34.26modulus_inv_arp, i don't want to restart the menu
10:34.32Zeeekmodulus_ put a goto to the entry of the extension
10:34.52modulus_i want the menu to keep going and ignore if extension pressed is invalid
10:34.54Zeeekeither before any prompts or after
10:35.12Inv_arpmodulus_: use Playback ()  instead of Background()
10:35.13modulus_b/c ppl press extensions before i can read(VAR)
10:35.19Zeeeki,1,goto(3)
10:35.23modulus_inv_arp, good idea i should've thought of that
10:36.00[ro]nic3tryHOW do i stop asterisk to ask for autentification on incoming calls ?
10:36.10modulus_Authenticate()
10:36.54[ro]nic3tryno i have an sip_proxy_out in sip.conf
10:37.04[ro]nic3tryi have an sip_proxy_out in sip.conf
10:37.48Eightwait for digit returns the decimal ascii code of the digit dialed.
10:37.55[ro]nic3tryif sip_proxy_out is set as user it doesnt ask for sipProxy autentification
10:37.59EightA string, or a number... nah, can't have that =p
10:38.33Eight~agi
10:38.34jbotagi is probably the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages
10:38.34[ro]nic3tryif sip_proxy_out is set as peer or friend it asks for sipProxy autentification
10:38.36*** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com)
10:39.37ZeeekEight isn't everything strings in asterisk ?
10:40.56*** join/#asterisk mbranca (~matteo@80.152.73.227)
10:41.27Eight'200 result=51\n'
10:41.34Eightthat's the string I get back for 3.
10:41.42EightI think pyst will do the conversion for me, though.
10:42.13ManxPowerDoes anyone else think that Kevin Flemming is very smart and very crazy?
10:42.28*** join/#asterisk brimstone (~brimstone@146.229.178.20)
10:43.06Zeeekhim? http://www.integritypersonnel.com/kflemming.shtml
10:43.08brimstonethanks again Zeeek, i got wctdm to compile under debian 2.6.8
10:43.15Zeeekgreat!
10:43.22Zeeekwhere do you find the info?
10:43.46brimstonerequired me to rm all of 2.6.* and then load the kernel again from a knoppix chroot, but it's working
10:43.49ManxPowernomad_, the one on the mailing lists.
10:43.56brimstoneno place special, just kind of kicked stuff around
10:44.09ManxPowerAt first I thought he was just plain crazy, but now I think he's very smart too.
10:44.10ZeeekManx users list or other?
10:44.12modulus_if ($bal < 1.05){
10:44.12modulus_<PROTECTED>
10:44.12modulus_<PROTECTED>
10:44.12modulus_<PROTECTED>
10:44.12modulus_}
10:44.17ManxPowerThis is either a very bad thing or a very good thing.
10:44.20modulus_for some reason that $bal comparison tickles me
10:44.46modulus_guess what i'm doing wheeee!
10:44.51ManxPowerkpfleming@starnetworks.us
10:46.37*** join/#asterisk lidl (~little@213-140-6-96.fastres.net)
10:47.26ZeeekYes, sometimes what happens is that you are accidentally using
10:47.26Zeeekundocumented behavior, and when that gets "fixed" your system breaks.
10:47.35ZeeekThat's a good one!
10:47.36modulus_$AGI->debug() is a real function right?
10:47.40modulus_i read it once somewhere
10:48.02*** join/#asterisk ikey (~ikey@202.54.37.183)
10:49.01ikeycan any one help me on configuring sip channel with iptel.org
10:49.10ta[i]ntedmodulus_ your code sucks
10:50.07ta[i]ntedwhat exactly do u want to debug
10:50.39*** join/#asterisk afe ([KFVNEtAOw@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
10:50.51afeMorning, folks!
10:51.13Eightoh my god.
10:51.20EightI've been spending waaaay too much time in extensions.conf
10:51.25EightI just blanked on how to do a goto in python =p
10:51.29EightOooh ya =p
10:52.04afeYou know, it's just too much fun pokin around in extensions.conf all the time :D
10:54.32felipexany info about bluetooth ?
10:54.58ikeycan any one help me on configuring sip channel with iptel.org
10:57.08[ro]nic3tryHELP
10:57.29[ro]nic3tryi canot pe called from a ser server
10:57.41[ro]nic3tryeven i have been registered
10:57.54[ro]nic3tryand i can call on tha server
10:58.46ManxPowerDoes anyone know of a deaktop pager for Win32 like the ones in KDE, GNOME, FVWM, etc?
10:58.51ManxPowerdesktop too.
10:59.34ZeeekManx you mean a pop window?
10:59.50Zeeekfor callerid and such?
11:00.29ZeeekWe use Yac
11:01.48*** join/#asterisk afe ([p7Gv1IiBf@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
11:02.02ManxPowerZeeek, No, like the multiple virtual desktop application that basically every linux GUI has.
11:02.31Zeeekya xp has it
11:02.51AshManxPower: XP PowerToys has what you want
11:03.19ManxPowerAsh, Does it work in Win3k?
11:03.22ManxPower..er....
11:03.24ManxPowerWin2k
11:03.30AshManxPower: No, it's only for XP.
11:03.36AshThere are other things for 2k, though.
11:03.39Zeeekbut there is free ware for 2k
11:03.40Ash(I don't know what they are)
11:03.46Ashhttp://download.microsoft.com/download/whistler/Install/2/WXP/EN-US/DeskmanPowertoySetup.exe
11:03.50ManxPowerI don't even know the correct terms to search for.
11:03.52Ashis the XP one
11:04.01Zeeekvirtual desktop
11:04.04AshManxPower: "virtual desktop win32"
11:04.09Ashor win2k or what have you
11:04.27ManxPowerYeah.  I get all sorts of stuff that's pretty useless with that search.
11:04.45afe~google nstall/2/WXP/EN-US/DeskmanPowertoySetup.exe
11:04.45ManxPowerIt will be less of an issue once the new motherboard for my linux box arrives.
11:05.00Zeeekok, manx I see I need to stop what I'm doing to help you
11:05.08Zeeekwait a second adn I 'll find it for you
11:05.13afe~google windows 2000 virtual desktop
11:05.25ManxPowerZeeek, I'm looking for sometthing that works very much like the linux version.
11:05.48*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
11:05.59afeI think the first link points you to what you want
11:06.07ZeeekThese are little icos you hit to change the display to a diffenent destop
11:06.26ManxPowerThat's much closer to what I'm looking for.
11:06.27ZeeekI find this kind of thing on freeware Usenet groups
11:06.47Zeeekbut the message "virtual desktop" isn't there, only the header :(
11:07.00Zeeektry googling the freeware goup
11:07.07Zeeekgroup
11:07.20Zeeekthis is a question that comes up often there
11:07.23afetry http://virt-dimension.sourceforge.net/
11:07.47ManxPowerafe, Yes, that one looks closer to what I'm looking for.
11:08.06ManxPowerIt still is a seperate window, rather than living on the taskbar, of course.
11:08.30*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
11:09.01Zeeeklook here: http://www.ams.as.ro/index.htm
11:09.39*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
11:09.51Zeeekno, don't
11:10.07ZeeekI had one of these for Win2K but I can't find it - sorry
11:11.14ManxPowerZeeek, "looking there" doesn't seem to work for me.
11:11.37Zeeekyou want freeware, right?
11:11.49ManxPowerZeeek, If it's good enough I'll pay a small amount.
11:12.04Zeeekhttp://www.tucows.com/virtualdesk95_default.html
11:12.15Zeeektucows is a good source for this kind of stuff
11:12.26*** join/#asterisk djin (~djin@62.58.40.196)
11:12.28Zeeekoops all 4 are shareware
11:12.40ZeeekI often buy good programs for up to $30 though
11:13.36ZeeekI know there is at least opne free one but I can't find it. They aren't maintained because most people have moved to XP which has a free one in XP Toys
11:14.43ZeeekManxpower, here ya go, this looks promising: http://www.free-soft.ro/desktop/desktop.html
11:15.05ManxPowerI'm running Windows.  That means I should be utterly terrified of installing software.
11:15.16Zeeekyes be afraid. Be very afraid
11:15.21*** join/#asterisk afe ([oZrbWPFNw@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
11:15.50Zeeeksoftware "firewall" like Kerio is an absolute must (to signal callouts)
11:15.57coppiceanything that looks like work gets me pretty scared
11:16.05ManxPoweri.e. I don't want to randomly install these apps and hope one of them doesn't fuck up my system.
11:16.09brimstonein a call file, how do i make it wait till someone picks up the line on the other end?
11:16.22ManxPowerbrimstone, you stop using analog ports.
11:16.26Zeeekbrimstone put a value like 9999999
11:16.45Zeeekor none at all in dial
11:17.06*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
11:17.18brimstonein the Channel: line?
11:17.23ZeeekManx, still sourceforge
11:17.24Zeeekhttp://sourceforge.net/projects/virtual-desktop/
11:17.28brimstoneno value?
11:17.55ZeeekDial with no timeout value will wait IIRC
11:17.57afeI have a X100P clone that I'd like to put in a computer that only accepts half height pci cards - think it would be possible to just cut the metal plate to make it fit?
11:18.44*** join/#asterisk zoa (~zoa@dD577788C.access.telenet.be)
11:18.46zoahello htere
11:19.28ManxPowerIf you are using analog ports Asterisk will consider the call answered when it finishes dialing.
11:19.39brimstone:/
11:20.00brimstonei only got one analog channel to play with
11:20.04Zeeekbrimstone what are you trying to do?
11:20.14brimstonejust playing with asterisk
11:20.37brimstonei'd like for the one fxo to just ring the one fxs like a simple pass through
11:20.40brimstonethat's the current goal
11:21.09Zeeekand you can't call the fxo?
11:21.21brimstonei can now, just got that working
11:21.33*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
11:22.00ManxPowerI can't even figure out how to use these damn things.
11:22.07Zeeekwhich one Manx?
11:22.25ManxPowerZeeek, The two that I downloaded.
11:22.28ManxPower8-)
11:22.43ManxPowerk http://sourceforge.net/projects/virtual-desktop/ is the one I'm trying to look at now.
11:22.44Zeeekyou mean make clean; make ; make install didn't work?
11:22.57ManxPowerran the installer.
11:23.00Zeeekand?
11:23.03Zeeeknothing?
11:23.07ManxPowerno virtual desktops, no icon, no nothing.
11:23.17Zeeekthe trojan is installed and talking on port n?
11:23.24ManxPowerIt's prolly uploading my harddrive to a server in russia. 8-)
11:23.26Zeeekthis is Windows - you have to reboot
11:23.57Zeeekthe sourceforge one has the sources obviuosly
11:24.10ManxPowerInstalling Linux would be easier.
11:24.16Zeeekthat's an option
11:24.26Zeeekthen use wine :)
11:24.34ManxPowerZeeek, not really.  I need windows for pcANYWHERE at least.
11:24.51Zeeekthat's some strong medecine
11:24.55ZeeekpcA
11:25.11Zeeekseriously the sourceforge thing looks good
11:25.20ManxPowerOn my desktop I can at least run vmware.
11:25.37brimstonewow, that wasn't hard to do at all...
11:25.54ManxPowerZeeek, Install it, let me know how to use it 8-)
11:26.32ZeeekDon't need, went XP everywhere
11:26.38Zeeekit's worth it
11:26.46ManxPowerWill XP run well on a 600Mhz machine?
11:27.02ZeeekI'v only seen a single blue screen once in 400+ days of 4 PC
11:27.11afehow much memory do you have in it?
11:27.18ZeeekI'd say if Win2k runs, XP might
11:27.28ManxPowerI've not seen any BSODs since I got the laptop 6 months ago.
11:27.36Zeeekyou have at least 256M ?
11:27.47ManxPowerCan you get rid of that sissy GUI on XP?
11:27.56Zeeekmore or less
11:27.56RaYmAn-Bxyeah
11:28.06afeI wouldn't run XP with less than 512 MB and yes, you can turn off the sissy stuff :)
11:28.12ManxPowerSo it looks like a computer, rather than a hello kitty playland?
11:28.16coppiceyou *must* get rid of that GUI unless you have a fast machine
11:28.41coppiceif its was hello kitty like it would be pink
11:28.46coppicevery pink
11:28.49Zeeekin defense of XP, it handles stuff like USB a lot better than Win2k
11:28.57*** join/#asterisk cero64 (ruiner@fantab.ulo.us)
11:28.57Zeeekthey're all pink INSIDE
11:29.00ManxPower"Oh, look, such cute little icons,  *pet*  Ouch!  That icon bit me!  Oh god they are swarming!  Help!  Help!"
11:29.04afeI went the other way and installed lots of extra eyecandy :)
11:29.20coppicebut any third party USB driver with the windows logo certification will blue screen
11:29.23ZeeekManx it will popup and lecture you about having too many UNUSED icons
11:29.42ManxPowerzeedo, I really hope you are joking.
11:29.44Zeeekcoppice that'w why the blue screen driver was invented
11:29.55ZeeekMarxo, no not at all
11:29.57RaYmAn-Bxyou can just switch off desktop icons :>
11:30.01ManxPowerSomeone kick zeedo.  He's messing up my auto complete.
11:30.15ZeeekIt's Zeppo actually
11:30.32coppicei dunno what testing the MS labs do on these third party drivers, but its a joke.
11:30.43afeManxPower: unless you tell xp not too, it will eventually pup exactly those things :)
11:30.51ManxPowerAnyway, I didn't find anything suitable so I'll just live with it.
11:30.55afepup=pop
11:31.07ManxPowerI ran Win98 until 6 months ago.
11:31.09Zeeekyou tied the sourceforge dealie ManxPower
11:31.11Zeeek?
11:31.16RaYmAn-BxManxPower: considered a shell replacement? :> Like blackbox for windows or whatever?
11:32.01RaYmAn-BxI think they can do multidesktop thingy..and no startmenu (or desktop icons..)
11:32.04ManxPowerZeeek, Yes.  It happily installed, installed the "NT Service" then did nothing.  No icon, no taskbar, when I run the menu option for it it happily accessed the HD for a min or 3 and then...did nothing.
11:32.41ManxPowerRaYmAn-Bx, I basically want the GNOME windows manager and desktop pager applette for Win2k
11:33.08RaYmAn-Bxcygwin-X?
11:33.22ManxPowerRaYmAn-Bx, I was looking for something simple.
11:33.51ManxPowerIt's not like I'm going to spend a lot of time getting it to work, I only use the win2k machine when my linuxbox has hardware problems or when I'm on the road.
11:34.52ManxPowerThe Win32 "window manager" is pretty similar to my GNOME desktop.  A "start" menu, a task bar, an applette ssection.  It's just missing a desktop pager.
11:36.02afeanyone knows if it would be possible to connect a digium TE100P to a PRI interface in an old LG PBX?
11:39.12ManxPowerafe, We don't generally care what the far end hardware is, as long as it supports a protocol Asteriskk supports (like PRI)
11:39.29*** join/#asterisk dreamcode (~je@81.181.199.39)
11:39.59afeManxPower: ok, so I guess then it would be possible to use it to get 32 channels between asterisk and the pbx?
11:40.15ManxPowerafe, Well 30 channels, since 2 channels an an E-1 are used for signaling.
11:40.29ManxPowersorry, 2 channels on E-1 PRI are used for signaling
11:40.49afeoh, well that's enough :)
11:40.56coppice1 channel for signalling, and one for sync
11:41.12ManxPowerafe, The PBX could be doing something stipud, but if it's PRI, it SHOULD work.
11:41.16*** join/#asterisk jeffik (~jeffik@m807a36d0.tmodns.net)
11:41.30afeunfortunately, the expensive part to make it work is probably hiring someone to configure the old pbx ...
11:46.39Mavvie15:43:25.727288 10.192.15.229.1177 > 10.192.0.2.53:  48+ Type1907 (Class 29802)?. (33) [tos 0x60]
11:46.52Mavviethat's what my cisco phone (7970) uses for a DNS packet.
11:46.57dreamcodewhy is asterisk asking to autentificate a user from an SER server which is calling me ?
11:49.12dreamcodeor.. how do i set asterisk to match first the to then the from ip in sip.conf
11:49.27*** join/#asterisk gst (~gst@wireless.sysfrog.org)
11:49.54ManxPowerdreamcode, You really can't.  You want to either match on username/secret or match on ip address, but you don't want to do both.
11:50.37ManxPowerdreamcode, I seem to recall that SER does not authenticate at all.  You need to use allow/deny and match on IP address.
11:51.04ManxPowerdreamcode, How many thousand users will you have?
11:51.13dreamcodeManxPower: my problem is :that i want to foward call on ser server, and also to be able to receive calls from the same ser server
11:51.31ZeeekEntropy infers that nothing in the universe can ever be "unlimited"
11:51.32dreamcodei just use one user on the ser server
11:51.44ManxPowerdreamcode, don't use SER.
11:52.04ZeeekManxPower did you look at the control panel after installing any of those desktop apps?
11:52.07dreamcodebut that's my VOIP provider
11:52.27ManxPowerdreamcode, then use permit/deny to allow SER to connect.
11:52.57dreamcodeok.. thx.. at that i didn't thought
11:53.51lidlif i buy a cisco 7940g, will i be able to use it with * ?
11:56.00gstdoes with IAX the voice traffic between 2 clients always travel through the server or is IAX able to transfer the voice stream directly between 2 clients (like with SIP)?
11:56.33ManxPowergst, Yes.
11:56.47ManxPowerlidl, Yes, if yo have the SIP firmware (extra cost)
11:57.13gstManxPower: tnx
11:57.13ZeeekManxPower I just installed the VDM on SourceForge - it works beautifully
11:57.19ManxPowerlidl, Cisco phones only come with the phone, no power and no SIP firmware.  Both are extra cose ($45 for power, $125? for SIP firmware)
11:57.32ManxPowerZeeek, I've already moved on.
11:57.35ZeeekThe switching is donbe with Alt-1 2 and 3
11:57.43ManxPowerlidl, Consider Polycom.
11:57.59Zeeekyeah but it works - seems to do what you wanted
11:58.35ManxPowerZeeek, it puts a little pager between the windows on hte task bar and the clock in the lower right?
11:59.03ZeeekNO NONE OF THAT
11:59.06Zeeekooops
11:59.15Zeeekit doesn't put anythiong anywhere
11:59.23ManxPowerZeeek, Um, that's what the Linux pagers do and that's what I was looking for.
11:59.31Zeeekit just uses Alt1 2 or 3 to switch between three desks
11:59.49Zeeekwell, I installed it just for you :)
11:59.55ManxPowerZeeek, Thank you.
12:00.55*** join/#asterisk LorenzoMarouani (~LorenzoMa@AVelizy-112-1-27-252.w80-13.abo.wanadoo.fr)
12:00.56Zeeekfyi if you even need to install windows programs, there is usually docs in the directory
12:01.04lidlManxPower, thx for the hints
12:01.16Zeeekthey could have called it README but they chose desktop.html
12:01.18Mavviehttp://weblog.barnet.com.au/edwin/000094.html <- yippee cisco
12:01.28ManxPowerlidl, polycom comes with power and SIP firmware (if you get the right model).
12:01.29ZeeekManxPower a little ironic
12:01.32LorenzoMarouaniHi
12:01.40ManxPowerlidl, Neither Polycom nor Cisco support their phones with Asterisk
12:02.23LorenzoMarouaniSomeone can tell me if there is an handler on incomming channel event in astersik ?
12:03.24LorenzoMarouaniI need to dev a module, and pass out extensions
12:05.42afeyou guys know the typical street price of a polycom 600 in the US?
12:05.56afeor a cisco 7940 with SIP
12:06.10Zeeekafe I'd gueszs around $340
12:06.43lidlManxPower, so you wouldn't recommend policom either, would you?
12:06.48afehmm... that's about 50% of what it costs here (Sweden)
12:06.48ManxPowerlidl, If you want a cheap, but good phone, consider the SIPura SPA-841
12:07.04ManxPowerlidl, Polycom is the Official VoIP Phone for my customers.
12:07.30ZeeekI keep thin king this is a Polycom dealer : http://aticom.com/
12:07.49ManxPowerThe SIPura is not a "pretty" as Polycom or Cisco, but it works well.  It doesn't have a second ethernet port and it doesn't support PoE, however.
12:08.11ZeeekBT102 has a second port
12:08.23afeZeeek: Umm... that page was... well, interesting
12:08.39ZeeekI know there is a store that sells them with a URL near that
12:09.03ManxPowerYou can frequently get pretty good discounts on polycom stuff if you find the right dealer.
12:09.13afeI have a collegue going to NYC in a couple of weeks, but he'll only be there for 4 days, so no time to order it I'm afraid
12:09.27lidlManxPower, let's say I could pay 200Ī (or dollars) for a phone. is it policom a good choice?
12:10.00ManxPowerlidl, The polycom IP 300 should be available for under $200
12:10.04ManxPowerThe SIPura is only $100
12:10.28Zeeekhere's one afe: http://www.voipsupply.com/product_info.php?cPath=95_107&products_id=251
12:10.52Zeeek$199.95 no supply or shipping
12:11.44ZeeekManxPower the Sipura has had a few bad reports on the way the phone is built, sticking keys, etc
12:12.50ManxPowerZeeek, I know.  I've not had significant problems with my two SPA-841 phones
12:12.59ManxPowerAs long as you use the latest firmware
12:13.27Zeeekthe URL I was looking for is http://www.atacomm.com
12:13.42ManxPowerIt will prolly become the Recommended Official Manx Power Personal Phone
12:14.02ManxPowerNeither ATAcom nor VoIP supply seem to have great prices.
12:14.23ZeeekCisco 7960G with SIP $379
12:15.02Zeeekip500 $209 + ship
12:15.17Zeeeknot good?
12:15.22Zeeekwhere should I  look?
12:15.47ManxPowerZeeek, Maybe they reduced their pricing.
12:16.03ManxPowerIP 500 for $210 is not a terrible price.
12:16.21ManxPowerKeep in mind that if you want PoE on the 300 or 500 it's $30 for the special cable.
12:16.21Zeeekincludes the 110-250 PS
12:16.28ZeeekI do not want POE
12:17.07ZeeekI'm told I'd pay $220 including PS and shipping - I though that was decent
12:17.40Zeeekthe ip600 is about $100 more for the person asking about that model earlier
12:18.08afeZeeek: the first page had 7960:s for $299
12:18.33ManxPowerafe, I'm sure that's with the Cisco SCCP firmware.
12:18.45ManxPowerand no power supply, of course.
12:18.55Zeeek<PROTECTED>
12:18.59afeManxPower: actually, that was with the power supply
12:19.11ManxPowerthey must be getting them cheaper then.
12:19.24ManxPowerI should look at ATAcomm again if they have lowered their prices.
12:19.50Zeeekno wait $319.95 includes SIP
12:19.53coppicespeaking of atacomm, what happened to ipvolution?
12:20.11afehowever, if I would order it the shipping would cost a lot, and I might have to pay customs and VAT as well :/
12:20.24Zeeekafe no have your friend bring it back
12:20.48ManxPowercoppice, I don't know, but I suspect the usual problem of "Wow, this is a hell of a lot more complicated and expensive than I thought!" problem.  Which I refer to as "The Wasim Problem".
12:20.53afeZeeek: yeah, but I'm not sure he'd be able to get it his hotel fast enough
12:21.03Zeeek<PROTECTED>
12:21.17Zeeekusually they won't deliver to hotel anyway
12:21.41Zeeekhe's have to go pick it up +$40 taxi (assuling he takes the subway there)
12:21.59afethey're in NYC?
12:22.27ManxPowerFly to NYC for vacation, then write it off as a business expense since you were there to pick up the phone.
12:22.32Zeeekheh
12:22.54ZeeekI thought they were in Joizy actaulluy but I suddenly can't find that info
12:23.08ZeeekBuffalo, NY 14225 USA
12:23.17Zeeekclose
12:23.19ManxPowerThat's not even close to NYC
12:23.27Zeeekclose phonetically
12:23.37ManxPowerI think Buffalo is closer to Toronto than to NYC.
12:23.43afeon a 4 day holiday, I guess he might be reluctant to go to Buffalo :)
12:23.46Zeeekok, vacation in Toronto
12:23.53lidlhow about a clipcomm phone? has anyone tried http://www.voipsupply.com/product_info.php?cPath=95_106&products_id=307 ?
12:24.06Zeeekthere is no such thing a 4-day holiday for Europeans!
12:24.20afeunless I can convince him he needs to see the Niagara Falls :)
12:24.40Zeeekwhat about a cheaper Cisco?
12:24.57*** join/#asterisk feral_kid (~not@209.205.207.130)
12:25.03Zeeekah no, doesn't look very good
12:25.33afeI can get the low end ciscos for about $100 here, but they look like crap
12:25.43ManxPowerI was at Niagra Falls (Canada) last summer for a couple of days.  Pretty cool, but very touristy.
12:25.44Zeeekno SIP, no speakerphone
12:26.19Zeeekwatch out, those prices are sometimes for refurbished phones on voipsupply
12:26.48afeand to get a 7940 with SIP here, I'd have to get it from a certified Cisco dealer = $$$
12:27.04Zeeekor order from Russia :)
12:27.35feral_kidI just bought a Tiger Jet Network card of of E-Bay. Although ztcfg shows that one a channel is on, I can't get dialtone of the the card... Is there something about that card that I don't know about?
12:27.57Zeeekyou mean an X100P clone?
12:28.06feral_kidZeeek: Yes
12:28.10afeI can travel to russia for about $150 to get one, but I hate that place :)
12:28.10Zeeekthat's a FXO card to be connected tot he phone line
12:28.28Zeeekyou mean you can't make the card talk to PSNT?
12:29.07feral_kidIsn't that both a FXO/FXS card?
12:29.08Zeeekferal_kid it's connected to PSTN and you aren't able to get it to go offhook?
12:29.15Zeeekerrrrr no, not at all
12:29.24lidlon voipsupply a polycom ip500 is sold at 200$
12:29.26ZeeekI've never heard of such an animal
12:29.40Zeeekso we were saying lidl
12:29.50afeferal_kid: you can plug a phone into the phone jack on it, but it won't have anything to do with asterisk
12:29.51Zeeek$220 with power supply
12:30.04lidli would use POE
12:30.16ManxPowerWhich is really weird because Polycom always ships with a power supply, as far as I know.
12:30.23Zeeekferal_kid actually there is a phone jack but it's for when the card is not powered (PC off)
12:30.40lidlManxPower, so i'm compelled to buy the powersupply?
12:30.46ZeeekManx ya they do it to make it look cheaper - in fairness it's a good option
12:30.55feral_kidZeeek: Ah... That was what I was missing... :)
12:31.03Zeeeklidl not if you already have a good 12v
12:31.06ManxPowerlidl, It's not like the power spupply adds more than a few dollars to the phone.
12:31.10feral_kidZeeek: NOw it becomes clear...
12:31.28Zeeekferal_kid you need an FXS or an ATA (cheaper)
12:31.42ManxPowerlidl, I suppose they may have a version without power, but I've never heard of anyone getting the wrong product and screaming on the mailing lists about lack of power supply.
12:31.46afeferal_kid: actually, it works when it's powered as well, but it will "steal" the line from asterisk
12:32.05feral_kidZeeek: I have two Sipuras, but I was was just testing out the card...
12:32.06ManxPowerThe IP 300 and IP 500 both require special PoE cables.
12:32.12Zeeekafe does it ring when the line rings? I guess so
12:32.19afeZeeek: yes
12:32.38ZeeekI ran for a few weeks with phone in parallel
12:32.41Zeeekit sucks
12:32.43afeit's really not that bad to use as a backup
12:32.59Zeeekthe asterisk phones will keep ringing for one or two rings
12:33.02ManxPowerThe two ports on the X100P are hardwires togather.
12:33.07lidlManxPower, are the 'special' cables shipped with it, or si it an extracost?
12:33.11Zeeek$30
12:33.18Zeeekso you see
12:33.20afethe tdm11b is a much better card
12:33.26Zeeekno free lucnh anywhere
12:33.28ManxPowerlidl, PoE cables are $30 from what I understand and are a seperate product.
12:33.36Zeeekor lunch even
12:33.39lidl:/
12:33.50Zeeekexcept maybe in prison
12:34.01Zeeekspeaking of lunch....
12:34.05afeif now only digium support could get my dialback problem fixed :)
12:34.07feral_kidZeeek: The only reason I picked up the X100 clone was so I could play around with FWD-IN
12:34.20ZeeekFWD-IN ?
12:34.29ZeeekI know about FWD OUT
12:34.55Zeeekget a free software phone and use that for now
12:35.02*** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com)
12:35.04feral_kidZeeek: It is 4:30A, so by this time I don't know IN from OUT... :)
12:35.14Zeeekhaha
12:35.21Zeeekwatch behind you then ;)
12:35.25marlowehahah
12:38.29afeanyone knows if SJPhone or X-Pro for pocketpc works with a bluetooth headset?
12:40.12lidlon the specs i read the ip500 supports multilanguage, but it doesn't show what languages are available. does anyone know?
12:40.17marloweyesitdid on mine
12:40.52*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
12:41.36*** join/#asterisk [Paul] (~paul@80.100.33.108)
12:41.53[Paul]hi
12:42.26[Paul]i've just upgraded eyebeam to the newest version (3000W)
12:42.30[Paul]3004W
12:42.48*** join/#asterisk NosDe (~joernhall@c224016.adsl.hansenet.de)
12:42.52[Paul]when i try to make a call i always get a 407 error
12:43.20[Paul]when i reinstall 3002S (i think) it works fine
12:43.25NosDehi. anyone here with some skills in chan_capi ??
12:43.42[Paul]and when i'm in my lan it also works
12:43.49[Paul]but via NAT it doesn't
12:44.22ZgarbiI just run Asteriks for first time and dial to number 1234 but answer quality is very bad, like zigzaged voice. may it's because cpu=600MHz and RAM=128k on fedora core 3?
12:44.53[Paul]i'm running asterisk on 500mhz and 128mb and that works fine
12:45.32NosDei'm running * an an pentium 233 (down to 90 MHz) and 128MB
12:45.53Zgarbiso strange... :(
12:46.59ZgarbiI tryed to change codecs on my x-lite, but same, ping between mycomp and asterixhost less then 5ms
12:47.13afeI first ran * on a PII 333 and it was ok
12:47.43Zgarbiso what can be a problem?
12:47.45ZeeekZgarbi do you use ulaw?
12:47.48afe128 k (k???) might be a bit low ;)
12:48.12ZgarbiI don't konw what is ulaw, i'm newbie
12:48.19Zgarbi128M
12:48.36Zgarbisorry
12:48.37afeI use fedora 2 and it eats quite a lot of memory
12:48.50afeZgarbi: I guessed that, just teasing you ;)
12:49.15ZgarbiCpu(s):  1.0% us, 12.1% sy,  2.0% ni, 83.3% id,  1.6% wa,  0.0% hi,  0.0% si
12:49.15ZgarbiMem:    125704k total,   119472k used,     6232k free,     1084k buffers
12:49.15ZgarbiSwap:  1020116k total,    45864k used,   974252k free,    16520k cached
12:49.38ZeeekZgarbi the codec being used on X-Lite is shown during the call - what codec do you see?
12:49.49ZgarbiGSM
12:49.54afeulaw is a sound codec, and if asterisk does codec translation it might get CPU heavy
12:49.59ZgarbiI has changed it on 711
12:50.01Zgarbibut same
12:50.36afein the asterisk CLI, type show translation
12:50.39ZeeekIs that the onbe that is selected suring the call?
12:50.41NosDedoes anyone have some skills in compiling chan_capi (junghans) with capi20 (avm) for *
12:50.46ZgarbiI using x-lite to connect other host (not mine) and it's works perfect
12:51.48*** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl)
12:52.37*** join/#asterisk D1ng0 (~dingo@202.57.43.4)
12:54.44*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
12:56.07D1ng0hello from the warm tropical phillipines
12:57.19coppicehello from a cold wet miserable place just a short flight from you :-)
12:57.31marlowehello from nj.
12:57.33marlowehah
12:57.41Zeeekwhadya mean?
12:58.07afe~weather ESSA
12:58.30afeso don't complain about weather ;)
12:58.36ZeeekStockholm is a beautiful city
12:58.47coppiceits OK
12:59.32Zeeekcoppice your favorite city?
12:59.33marlowe~weather 07746
12:59.38afeI'm actually in Uppsala (70 km from Stockholm), but no weather code for that ;)
12:59.39marlowedarn
12:59.40marlowelol
12:59.58marloweill load weather.com
13:05.39marlowe~weather kttn
13:05.44marloweI knew that too.
13:06.23marloweIt should say snowing
13:06.29marlowe'cause it is out my window
13:06.45Zgarbiall right, I didnt understand what problem was but as I see there was 2 answer machine was executed together and sound was mixed. wher first finished second was fine
13:06.53*** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net)
13:08.18Eightaaaand BV is broken again.
13:08.22Eightsurprise surprise.
13:10.31Eightheh, funny you guys should be playing with jbot's weather feature.
13:10.48EightI'm in the middle of writing up a weather thing in python for Asterisk =p
13:10.58Eightit works with zip codes though =)
13:11.53Grooby~weather taiwan
13:12.02Groobydoh!
13:12.19coppicetaiwan is cool and damp right now
13:13.16Eighthttp://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice
13:13.29EightYou guys have any ideas as to why the second example might STOP working?
13:14.11GroobyBV can go out
13:14.15Groobylet me dial my self
13:14.21EightI can dial out, but not in.
13:14.29Zgarbibye all
13:14.40Eightand I haven't changed anything in the register => section in ages.
13:14.45afehttp://hem.bredband.net/b282251/images/IMAGE_00003.jpg = view from my window right now :)
13:14.49Eightsometimes it works, sometimes it doesn't.
13:14.56Groobyworks for me
13:15.03Groobyi never had any problems....
13:15.23EightGrooby: could you post your sip.conf as a third example on the wiki?
13:15.50Eightor atleast just pastebin for now.
13:16.06Eightand I'll work out the differences and make the changes to the second example.
13:17.45Groobyjust remember I am using AMP
13:17.52EightAMP?
13:17.56Eightjbot AMP?
13:17.57jbot[amp] an Audio MPEG Player.  [non-free]
13:20.21Groobytry that
13:20.52*** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz)
13:20.58afeumm... wouldn't that be asterisk management portal (or something like that) ... silly jbot :)
13:21.28Groobynah
13:21.31Groobythat's win AMP
13:21.38Grooby:P
13:21.50afe~kick jbot
13:21.52jbotbugger off sod!
13:22.16afe~hugs jbot
13:22.46carloshhowdy all: here trying to get  CallingCard Applications to work.. with postgreSQL.. having difficulties compiling I think because I haven't defied or configured postgresql client libary and header files, Anyone please care to help me doing this? thanks.
13:23.17carloshdefined
13:27.13Groobyyou there eight?
13:27.18Groobydid you see my pastebin?
13:29.36Groobyok..i am off
13:29.39Groobygood luck Eight
13:30.25*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
13:34.22feral_kidAnyone using Asterisk@Home?
13:34.35*** join/#asterisk threat|BX (threat@dsl-41.16.240.220.rns02-kent-syd.dsl.comindico.com.au)
13:34.38threat|BXG'Day
13:34.44afehowdy
13:35.46ZeeekBroadvoice questions, RFC: does BV encourage, discourage use of their service witrh asterisk (or are they indifferent)?
13:35.59carloshhowdy all: here trying to get  CallingCard Applications to work.. with postgreSQL.. having difficulties compiling I think because I haven't defined or configured postgresql client libary and header files, could anyone please help me with this? thanks.
13:36.25ZeeekIs there a compelling reason why BV or any provider should encourage (or not) the asterisk community
13:36.57afeZeeek: maybe they don't wan't their customers to be able to provide services for other people?
13:37.30feral_kidI tried to set up a trunk for FWD using iax2.fwdnet.net... I have yet to make a call through FWD (using iax2 or for that matter using just fwd)... Anyone know how to get this working behind a double NATted machine?
13:37.37EightZeeek: There was an article where a BV rep' discussed people using Asterisk with BV. Shouldn't be too hard to find. He welcome the market, but said 'you're on your own' for support.
13:37.57Zeeekwhat d'you think of his comment, Eight? (in a few words!)
13:38.37EightThat more service providers in more industries should have that stance.
13:38.52Zeeekno support but c'mon in?
13:38.58EightI dislike having a cookie cutter applied to me and getting parts I like lopped off.
13:39.12Zeeekis BV broken for everyone or just asterisk (several weeks now, all the postings)
13:40.34afeI'm using a Swedish provider (Rix Telecom) - they don't officially support asterisk, but has an asterisk forum
13:40.46afeand they're using asterisk for their service
13:41.32EightZeeek: I got it working again...
13:41.39EightI had to put 'insecure' in for them to log into me.
13:41.57EightI kept removing it because when I did so it KEPT working.
13:42.08afeEight: I had to put insecure=very for my provider as well - otherwise it didn't work
13:42.19Zeeekbut there has been so much talk about them recently, I wondered a lot about the compelling reasons on both sides
13:42.33ZeeekFWD requires that too IIRC
13:42.40Zeeekbut FWD is free as we all know
13:42.51afedoes bv limit the number of simultaneous calls (spelling? :))
13:44.39Eightafe: afaik they didn't used to, but do now.
13:44.57EightI can not, for example, call in and out at the same time.
13:45.03Eightatleast, however I tried it didn't work.
13:45.10afeEight: that kinda sux
13:45.40Eightoh, hey... I guess I can.
13:46.20afeat work we use asterisk for incoming support calls, and there's no limit in the number of calls for just one account :)
13:47.36afeour provider, however, refuses to sell their service to call centers and call marketing companies (for obvious reasons)
13:48.28threat|BXafe hows it going?
13:49.09afethreat|BX: just fine, thanks :) But, did you mean with anything in particular?
13:49.13ZeeekI was kind of worndering whether when possible we should strive to support service providers that try to lean twoards, not against asterisk
13:49.15Eightaha...
13:49.26EightI just did a PSTN->BV->Asterisk->BV->PSTN call.
13:50.00*** join/#asterisk Tili (~Tili@202-133-67-12-dialup.sat.net.pk)
13:50.09EightZeeek: I think that'll happen naturally.
13:50.19EightZeeek: if anyone gets a good rep it'll spread pretty quick.
13:50.59EightBV seems to be the leader now because they have unlimited minutes plans, and a fairly broad selection of DIDs.
13:51.11threat|BXafe I want to know more about asterisk :) in particular what I need to know for it to be a standalone PBX replacement
13:51.21threat|BXafe or if its feasable :)
13:51.26Eightthreat|BX: it is very feasible.
13:51.32threat|BXafe what hardware I need, etc..
13:51.35Eightthreat|BX: describe your current setup?
13:52.17*** join/#asterisk nix000 (~nix000@66.11.191.103)
13:52.18afethreat|BX: I suggest you take a look at the wiki - that's the best place to start
13:52.22afe~wiki
13:52.30Eightjbot wiki?
13:52.31jbothmm... wiki is http://www.voip-info.org
13:52.43nix000anyone can recomend a good ss7 gateway that works with asterisk ?
13:52.48*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
13:52.50threat|BXEight small business, four lines, probably need ~10 - 20 phones connected to the system
13:53.00*** join/#asterisk Goldenear (~Nicolas@d193.dhcp212-198-200.noos.fr)
13:53.10ZeeekEight yes that is the case (good/bad rep being known in the "community")
13:53.13Eightthreat|BX: keep your existing phones, or switch to new ones?
13:53.30*** part/#asterisk Goldenear (~Nicolas@d193.dhcp212-198-200.noos.fr)
13:53.48Zeeekthreat|BX here a few decent links with some into stuff
13:53.49ZeeekStarter tutorial:
13:53.49Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
13:53.49Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
13:53.49Zeeekhttp://www.automated.it/guidetoasterisk.htm
13:53.49ZeeekTHE reference of the moment:
13:53.50Zeeekhttp://www.asteriskdocs.org
13:53.50threat|BXEight This would be for a new branch office for this particular company, so all new equipment
13:53.58threat|BXZeeek thanx :)
13:54.03EightJust a fairly simple computer (anything modern), with atleast one TDM400P card in it to talk to the analog lines (4 ports per card).
13:54.11EightAnd then put SIP phones on the network.
13:54.23Eightthe card is ~320 bucks with 4x FXO.
13:54.40Eightthe phones are anywhere from 60->350 depending on how fancy the model is.
13:54.53threat|BXI see
13:55.03EightSome have pass-through ability so you can put them between the computer and the rest of the network on each desk, no extra wiring.
13:55.38EightAn' that's it.
13:55.40ZeeekEight good summing up!
13:56.04ZeeekSomeone needs to write a paper on the pbx user experience though :)
13:56.13Zeeekbelieve me, it's not protty
13:56.16Zeeekor pretty
13:56.18threat|BXok, I would need to get a price on a PBX system with phones for a comparison, but the astericks solution sounds pretty good
13:56.23Eightthreat|BX: be warned, however. The actually configuration is extremely arcane.
13:56.27*** part/#asterisk srt (~nobody@gw0-cgn.reucon.net)
13:56.52Eightthe software is free, the hardware is inexpensive, but the man-hours put into learning the configuration can be intensive.
13:56.52Zeeekthreat|BX you can set up an asterisk pbx for way less than a classic pbx, but there are disadvantages
13:57.17EightIf you find a local company that can do asterisk install/support, go for it.
13:57.28EightIf you have the time to dedicate to it yourself, go for it.
13:57.36*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
13:57.42Zeeekonly if you have no job and aren't married though
13:57.43EightIf you want to just plug it in and go... use something else.
13:58.08Zeeekservice providers sell plug and play including a day of setup and config
13:58.18threat|BXEight what are some common problems that I may encounter?  phone calls drop out? asterisk starts randomly playing voice messages at inapropiate times?? :)
13:58.30afeEight: I agree man-hours cost, but if you're true geek (like most of us) it's just fun :S
13:58.37Zeeekyup
13:58.43Eightya, I'm just playing with it right now.
13:58.52EightIt could turn into something more soon, but for now it's just neat =)
13:59.00EightI have all my friends setup with x-lite, and so on.
13:59.04threat|BXafe hehe i would find enjoyment out of setting it up :P
13:59.22afethreat|BX: when you get it working, you won't have any problems - its (in my experience) very stable
13:59.33Eightthreat|BX: since you're asking this on a saturday, I assume you have some personal interest in it.
13:59.40afethreat|BX: it's the first time configuration that takes time
13:59.51threat|BXafe can asterisk be comfigured for one line using a cheap ass modem? or is it more envolved then that? :) (basically just want voice messages if no answer etc.. :))
14:00.08afethreat|BX: yes
14:00.11Eightthreat|BX: now that you know it *can* server your purpose... try tinkering with it. setup a computer with it and tinker. See if it's something you want to learn yourself, or would rather pay someone else to do.
14:00.19threat|BXEight hehe :)  well actually its sunday morning 1:17AM :P so yes, your guess is correct :)
14:00.21afethreat|BX: if it's a X100P modem (or clone)
14:00.31Zeeekthreat|BX take a look at those pages I showed. They have a plain english explanation of all the basic concepts
14:00.44Zeeeka lot of answers to your questions
14:01.29threat|BXZeeek ok I will :)  I am still asking quick questions before I dedicate my unmarried and unemployed life to reading the docs :P
14:01.29Eightthe X100P works, but as far as I know 'you get what you pay for', and it can be a little flakey. Not sure what the issues are though.
14:01.29afekinda odd actually, this community is not always very newbie friendly ;)
14:01.30threat|BXafe X100P modem aye, so you need a specific type?
14:01.42Zeeekafe I would'nt say that - except in a very few cases I've seen
14:01.57afethreat|BX: yes, that's the cheapest option available, but it's not very good though
14:02.17afeZeeek: hmm... true, the mailing list is much more unforgiving
14:02.18threat|BXlol yes I learnt linux 5yrs back, I know how unfriendly people can get :)
14:02.46threat|BXafe well I would want cheap to test it out at home before implementing it else where
14:03.06afethreat|BX: if you have the dough, I'd suggest getting a tdm11b from digium (devkit) - works very good and allows you to connect an analog phone as well
14:03.07Zeeekthreat|BX you can do a lot of playing with no hardware at all
14:03.32threat|BXZeeek really? like what? (I would like to test it out also)
14:03.54Zeeekif you'd read the four page article you'd already know! :)
14:04.12threat|BXZeeek LOL ok already :) I will read it, grrr :P
14:04.22Zeeekit's worth the few minutes, believe me
14:04.33Zeeekyou'll have more complex questions after
14:04.39afewhat really made me like asterisk is the possibility to answer my home POTS from anywhere, and for that you need hardware ;)
14:04.49Zeeekno
14:04.58Zeeekyou can have a voip number at home
14:05.05Zeeeka lot of people are doing that
14:05.13Zeeekand have no POTS service even at home
14:05.20afeunfortunately, I can't here - I need a POTS for my DSL :/
14:05.25Zeeekme too
14:05.37Zeeekbut now we have offers of no-voice POTS lines
14:05.42Zeeek(for DSL)
14:06.05afeWe're still waiting for that to arrive here - but the main Swedish Telco owns all the lines
14:07.10ZeeekFT owns them all here too but they are foprced to capitulate
14:07.47Zeeekmost DSL offers now include voIP "unlimited" national calling
14:07.48threat|BXdamn monopoly phone companies :(  theres one here at AU too :(
14:08.08afethreat|BX: yeah, but it will probably change
14:08.23nix000anyone interfaced cisco gatways with asterisk ?
14:08.36Zeeekfor example there is Free who offers $40/mo 20M down DSL + unlimited phone
14:09.28afeI have an offer from my voip provider for about the same, but no traffic limit and free national calls
14:09.46Zeeekthe down was speed
14:09.51afehowever, atm I can only get 512KB from them
14:09.54Zeeek20Mbs
14:10.02afeah, ok
14:10.05Zeeekactually 16
14:10.09ZeeekATM
14:10.22Zeeekstill very good - I don't know the upload speed though
14:10.23afeI have about 8 MBs now
14:10.31afeand 800Kbps up
14:10.38ZeeekWe have 2 at the office and 1 at home
14:10.44Zeeek256 up
14:10.49Zeeekworks ok for my use
14:10.58Zeeeknaturally I'd like more
14:11.29Zeeekbut in many parts of the USA, 256K/64K is like $40/month
14:11.42aferight now we have some really aggressive anti-piracy stuff going on, which kinda limit the use of a high band-width ;)
14:11.58Zeeekyou mean like file-sharing?
14:12.08Zeeekthere si a big crackdown on that
14:12.10afethat, and private ftp archives
14:12.16Zeeeksame idea
14:12.49Zeeek~seen bacondoublechz
14:12.55jbotbacondoublechz <~bacon@69-162-37-142.stcgpa.adelphia.net> was last seen on IRC in channel #asterisk, 16h 52m 2s ago, saying: 'xantus, I was taking about the option in the call preferences menu'.
14:12.55afetwo days ago they busted in at one of the ISP:s and found one of the biggest archives in Europe ...
14:13.46afethose responsible will face a lawsuit of several hundred millions
14:13.48Zeeekafe yeah they're working on EU laws to make it mandatory for ISP to provide user info
14:14.25Zeeekhow large is your asterisk installation at the office?
14:14.54afelots of people have gotten raided at home, where they take the computers and you can get both jail time and have to pay millions
14:15.03Zeeekours is just 2FXO 3FXS plus three SIP phones outside
14:15.19Zeeekand 1 IAXy for now
14:15.31afeat office we use SIP only (for about 5 people)
14:15.44ZeeekI recently bought some music at Virgin store - 1eu per song
14:15.48carloshguys, what's definitely the best pre-paid app. for * from your personal experience?
14:16.12Zeeekafe and what in from POTS? anything?
14:16.31afeZeeek: atm, we just forward the pots line to a voip number :)
14:16.55Zeeekwaiting to see if voIP is just a passing fad?
14:17.33afenah, but it's a bit difficult to move the number to voip atm, so I put a recording there telling people to use the voip number instead
14:17.54afewe'll probably keep it since it's a lot cheaper, and we can record calls etc
14:17.55Zeeekthis stuff will change radically when every single household is connected by high speed and you TV screen (hi-res, huge, flat on wall) will show live scenes of a beach on another planet
14:18.18ZeeekI'll gibve that about 10 years - except for the planet part
14:18.21lidlis it possible to connect iaxy to a fax?
14:18.40afeeventually, we'll try to connect our old pbx to asterisk, so everyone can call out for free
14:18.41Zeeeklidl in theory absolutely
14:18.58lidlZeeek, thx
14:19.08afelidl: fax and modems over IP are a bit shaky though
14:19.10Zeeekbut results may vary depending on the lag etc
14:19.16*** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter)
14:19.27Zeeekwell it is ulaw, but still, it better be a good connection
14:20.07Zeeekhey I wonder if I could connect my laptop via modem to the IAXy and get faxes ?
14:20.11afeif the server gets busy with some codec conversion for some other call, that could possibly affect a fax transmission
14:20.38Zeeekbut then why would I want to do that...
14:20.45lidlafe, i'm going to use a p4 to serve 9 telephones and a fax, with two incoming lines
14:21.00afelidl: that should be enough :)
14:21.08lidl:)
14:21.21afewhy would anyone want to fax anything anyways? :D
14:21.27Zeeeklidl with 2 TDM400 ?
14:22.01*** join/#asterisk jeffik (~jeffik@m506e36d0.tmodns.net)
14:22.02lidlZeeek, yes, i was thinking about having a tdm400 with 2 fxo and 1 fxs dedicated to the fax
14:22.19Zeeekand the other phones would be SIP?
14:22.26lidlZeeek, yes
14:22.38Zeeeksounds like a good plan
14:22.51carloshanyone using the chinese IPPhone: HOP8T ? have some issues with hold music and also 178ms delay on my LAN ! (compared to 1ms using soft client)
14:22.57lidlwell 7 sip voip phones and 2 iaxy/atas for two pots phones
14:23.13lidl.. and 1 fax
14:23.50Zeeekfux sax
14:24.09Zeeekfax sucks
14:24.24afein 2004 I think we might have received a maximun of 10 faxes that actually meant anything
14:24.24Zeeekbut yes, some people need them
14:24.38Zeeekmy last two were both spams
14:25.01ZeeekI was so pissed I called one company and began an order for 10,000 of their machines to send to china :)
14:25.10afelol
14:25.32Zeeekthen at athe end I said and if I ever get a fax from you again, better lock your doors!
14:25.37lidlsadly the office i'm going to serve receives a lot of faxes :/
14:25.56lidlwith a mean of 1 x day
14:26.10afeif faxing is important, I'd probably just keep a separate pots for that
14:26.23Zeeeklidl the surest way wouldbe to have an FXS module and connect a faw to that
14:26.27lidlafe, so an fxs
14:26.28Zeeekfax
14:26.28lidlyes
14:27.01afeor skip the fxs and keep one analog line for the fax
14:27.01Zeeekhe meant separate line and just connect the fax to it, no FXS?
14:27.05Zeeekya
14:27.41Zeeekor use a service like j2 or efax and receive the fax by email
14:28.07carloshfor fax, I'd use an fxs direct to dedicated modem/hylafax => gfax (linux app) to get them in tiff format files...   or convert to tiff using something else, and get it in email..
14:28.27Zeeekcarlosh good plan also
14:28.42afethat would work aswell :)
14:29.20lidlZeeek, i would do so, but my customers are not that computer literate.. i'm sure they do prefer getting paper other than an email
14:29.39carloshthen, send tiff file to local printer.. lol
14:29.45lidlcarlosh, :)
14:29.49Zeeekheh
14:30.06afelidl: ouch... then I'd go the separate line route
14:30.14Zeeekcompuserve used to have a service to convert email to a printed page that they then mailed!
14:30.25*** join/#asterisk smurfix (~smurf@smurfix.developer.debian)
14:30.51Zeeekthe previous version was delivered by a lone horseman
14:30.53afeno need to involve the asterisk for faxes if you always want them on paper
14:31.58*** join/#asterisk eKo1 (~bernd@63.245.57.70)
14:32.02*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
14:32.12FaithfulHey, can we peer with skype?
14:32.18Zeeekno
14:32.21*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
14:32.23Faithfulboo
14:32.33Zeeekshame, in'it
14:33.07lidlafe, but using a dedicated pstn would not be feasible in their situation imho, since they just have 2 pstn lines :/
14:33.42carloshshame indeed.. guys, out of all the logged on the irc, there must be one using any prepaid application.. I have some problems setting any from the wiki.. any reccomendations ?
14:33.42lidlafe so connecting the fax to the fxs would let me to use both incoming lines for the two fxo
14:33.42afelidl: then I'd make one of the modules on the two TDM400:s an FXS port
14:34.00afecarlosh: sorry mate - never tried it
14:34.16Faithfulis G729 better than iLBC?
14:34.31carloshilbc works beautyfully on dial ups..
14:34.50Faithfulhence skype uses it
14:34.57carloshbetter handling of packet losses
14:35.10Zeeekskype will prolly never peer with anyone though - except Motorola (done)
14:35.10Faithfulbetter than G729
14:35.42Faithfulcan we connect to skype with *
14:35.46Faithfulat all?
14:35.50afeFaithful: no
14:35.56Zeeektwo handsets taped toghter might work?
14:36.02zoaomg
14:36.04Zeeekhead to tail
14:36.08zoathis jetlag is killing me
14:36.35afeFaithful: Skype might eventually get a pots number option, and that would of course work
14:36.36Faithfulwell you could get a Skype FXO and plug it into *
14:37.05carloshFaithful, true
14:37.06Zeeekyou can dial in ti DISA with skype paid PSTN service
14:37.21afeI don't really like Skype, since it's only purpose is to make its creator (a fellow Swede) more money
14:37.34Faithfulgood on him
14:37.41afeindeed
14:37.41ZeeekYou need more rich people in Sweden now that Abba left
14:37.44lidlafe, so 2 fxo and 1 fxs on the TDM400, right?
14:37.58FaithfulI like people with iniative
14:38.07afelidl: yep
14:38.13lidlok :)
14:38.17Zeeekwhy not build a skype pbx ?
14:38.33carloshzoa: are you the zoa i am thinking you are who just arrived from Italy?
14:38.40afeFaithful: yeah, I guess I'm just envious :)
14:39.05zoanah i just came from von
14:39.17FaithfulI don't know why we cant plugin
14:39.19carloshoh, sorry
14:39.23lidlwhat's the average uptime for asterisk?
14:39.42Faithfulabout 10min
14:39.45afelidl: mine is exactly 24 hours since I restart it nighly :D
14:39.46lidli mean, in your experience
14:39.50carlosha few weeks.. I got it installed at a customer's .. they have only had to reset it once...
14:40.01Zeeeklidl it's very hard to tell since most of us experiemnt so much
14:40.02carloshin 4 months..
14:40.02lidlafe, is it a good practise to restart it?
14:40.14Zeeekcommon wisdom is once a week
14:40.24Chujilidl : what kind of load are you putting on it?
14:40.30afelidl: I don't think it's really necessary, but the mpg123 might act up (if you use music on hold)
14:40.32lidlfor a production system, would you recommend to cron its restart?
14:40.45Zeeekmany people do that lidl
14:41.01Faithfulwhat about Alcatel PBX they run a form of linux I believe ... do they self reboot once a week or so (or just work)
14:41.01Zeeekas long as you can safely stop all service at some predefined time
14:41.04lidlChuji, not much load, 2 incoming lines, and 10 voip phones
14:41.10eKo1I don't restart, I just let it run until it dies.
14:41.18Zeeekso how ioften is that?
14:41.28eKo1Every week.
14:41.32Zeeekreally ?
14:41.35Zeeekof what?
14:41.46carloshso, no one experience with pre-paid apps... no one with HOP8T IP Phone (Chinese), anyone experience with SER?
14:41.48afelidl: the restart takes like 2 seconds on a small system, so if you do it night time it won't hurt anyone :)
14:41.48eKo1All sorts of misc. problems.
14:41.50Chujilidl : I wouldn't worry about it then
14:42.00ZeeekeKo1 tell me about this
14:42.01Faithfulone down side with OSS is that we patch so many disparate bits of software together the outcome is sometimes a little hard to stabelize
14:42.42lidldoes asterisk offer a something like a watchdog to check it is actually running?
14:42.50afeyikes... heavy snowing going on outside now
14:42.58*** join/#asterisk [Paul] (~paul@80.100.33.108)
14:42.58eKo1Zeeek: I've had so many problems, I can't remember any specific one anymore.
14:43.00Zeeeklidl the safe_asterisk script restarts it when it dies IIRC
14:43.12eKo1So far: System uptime: 5 days, 10 hours, 39 seconds
14:43.13Zeeekor can anway
14:43.26lidlZeeek, ok
14:43.40ZeeekeKo1 ok, I was wondering - mine usually is up until I recboot
14:43.54Zeeekbut then I do that at least once a week lately
14:43.56Faithfulfor instance, someone leaving voicemail on my system creates a 25% chance that the ISDN module will drop out of the kernel and the box doesn't answer any more calls
14:44.34afeouch
14:44.37lidlFaithful, that's quite unstable then
14:44.46Chujiafe : Where are you?
14:44.53afeChuji: Sweden
14:44.57eKo1Well, I think * should be stable enough to not warrant a restart at all.
14:45.05*** join/#asterisk dimmik (~kold@ip-92-216.first.gr)
14:45.07eKo1Given the right environment.
14:45.22carloshanyone experience with status management, so you know other parties are available or not (IAX or SIP) .. ? thanks.
14:45.27lidlsince the system i'm going to install is at about 1000km from where i live, i'd like a stable system
14:46.03afelidl: such a small system, with only tdm cards would probably be perfectly stable
14:46.09Chujicarlosh : How do you want to use this *status*?
14:46.21lidlafe, thanks that's reassuring :)
14:46.47*** part/#asterisk dimmik (~kold@ip-92-216.first.gr)
14:46.47*** join/#asterisk d00gster (~in_ter@70.48.207.77)
14:46.52Faithfullidl: stick to digium equipment
14:47.05lidlFaithful, for sure
14:47.17afedigium + fedora works perfectly for me
14:47.26carloshChuji: I want to be able to tell if a SIP or IAX user is available.. or not..  similar to firefly, when you change your status, the others will see you in a different color..
14:48.13carloshI use fedora 3, latest kernel, latest cvs.. always very stable..
14:48.55Chujicarlosh : what do you want to display this in?
14:48.56afecalisto: I use fedora 2, but always latest kernel and asterisk CVS head - works like a charm
14:49.21eKo1using CVS head is very risky for production use.
14:49.22afethat calisto should have been carlosh :)
14:49.32Faithfulexcept we talk about rebooting once a week... which has never been linux practice
14:49.37carloshChuji: Gnomemeeting (the last ver) also has some support for statuses.. but Asterisk would not do anything when you change your status...
14:50.05afeFaithful: I don't reboot (unless I upgraded the kernel) - I just restart asterisk
14:50.42FaithfulThat
14:50.52Faithful's not so bad I guess.
14:51.06carloshChuji: the soft clients should support this ... only firefly does, and if you use their private service to send these availability messages (add on or patched asterisk)..
14:51.24afeFaithful: restarting asterisk also restarts mpg123, which is what I believe is causing most problems for me :)
14:54.09eKo1afe: Yes, that causes problems for me too.
14:54.32lidli'm going to use a debian as a platform
14:54.51afebleh... my mouse just refused to work on my icemat
14:55.03eKo1I kill those mpg123 procs. after I stop * and start * thereafter.
14:55.22carloshyou should only get one hold-music process per hold music file on your system..
14:55.37Groobyok back
14:55.39PatrickDKno, it's not per file
14:55.51PatrickDKit's per musiconhold definition
14:55.59carloshok..
14:56.23carloshi have only two processes.. no problems at all..
14:58.03afeemergency over - mouse working again :)
14:58.19carloshthe only issue wih hold music I have at present.. is this IP phone.. it would not play it...  zaptel extensions and incoming callers no problem.. bt not this chinese IP phone..
14:58.45afecarlosh: that's probably a codec issue
14:58.47threat|BXhmm so how do these X100P PCI thingies differ from regular modems?
14:58.58carloshafe: ta, will check..
14:59.21*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
14:59.54TheBearIn order to use IAXTEL, can one use a SIP phone connected to an * server, or MUST it be a IAX phone ?
15:00.07Eightthreat|BX: They're basically sound cards.
15:00.28Eightthreat|BX: all the processing is done in the CPU, instead of on card. That's why they're so cheap (and require special drivers in windows to work as modems)
15:00.44Groobyeight
15:00.48Groobydid you fix your BV problem?
15:00.57EightGrooby: ya, 'insecure=very'.
15:01.01Groobyokie
15:01.04Eightwhen you remove it, it will continue to work for a while.
15:01.11Eightbut to kick it into working again you have to replace it.
15:01.14Eightthat seems to be the issue.
15:01.20Groobyok
15:01.28threat|BXEight oh I see, so the hardware versions of them are expensive? (how much extra are the hardware ones?)
15:01.36Groobyhehe..I remember having that issue but i figure everyone knew about it
15:01.45Groobyoh well..back to my Mythtv research
15:01.47threat|BXEight heh they wont be used in windows "{
15:01.48afeTheBear: you can use any phone that can talk to asterisk
15:01.50threat|BX:P
15:02.08TheBearafe: thanks,
15:02.18afeTheBear: asterisk will translate the call for you
15:02.36TheBearcool !
15:02.56Eightthreat|BX: there are a bunch of places that sell the TDM400P online, check froogle
15:03.11threat|BXheh ok
15:05.55TheBearAm I right that you can have an extension (44) that if I dial (44pstn_number) It will then be able to repeat dial the pstn_number until it is answered (assuming the first attempt) the number is busy ?
15:07.22*** join/#asterisk \Grooby\ (~sl9z@ip24-250-126-171.dc.dc.cox.net)
15:08.26afeTheBear: well, you would probably want to wait a few seconds until dialing it again, but yes
15:08.54*** join/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de)
15:09.00kodomohi folks :)
15:09.24kodomoCan anybody tell me if early B3 is possible with chan_misdn?
15:09.33carloshre this unavailable status, GM reports back (on the * console):  Got SIP response 486 "Busy Here" back from <IP ADDRESS> .. but .. I'd like to know somehow, beforehand like with firefly.. that the user is available or not...
15:10.21TheBearafe: right, if exten => _44.,3,Dial(${EXTEN:2}), then _44.,103,Wait(3) _44.,104,goto(3)  right to loop back on a busy signal ?
15:10.44threat|BXhmmmm ok so I need a TDM400P card for incomming lines and a TDM400P card with other modules to connect my phones to?
15:10.48threat|BXam I on track?
15:11.29kodomoanybody using misdn for the external device?
15:11.54afeTheBear: not sure... the 104 will kick in when busy, and you should probably keep the call somewhere else and then retry
15:12.27afethreat|BX: all tdm400 cards can be equipped with any module
15:12.57MikeJ[Jayden]~seen xkev
15:12.59jbotxkev <kevin@orbit.xmission.com> was last seen on IRC in channel #asterisk, 6h 20m 47s ago, saying: 'lousy mysql licensing'.
15:12.59TheBearafe: isn't it when a Dial() get a busy signal the next seq in the dialplan is 100+Dial seq number, thus 3 become 103 ?
15:13.24afeTheBear: its n+101
15:13.27threat|BXafe ok but basically I need a tdm400, and aport for every phone I want to connect to it?  or are ther eother ways? ":)
15:14.09TheBearafe: ok n+101, then 104 should be wait and 105 goto(3)?
15:14.28afethreat|BX: you could use a Sipura (or other) that makes an analog phone able to talk SIP
15:14.37afeTheBear: I think so, yes
15:15.03afeTheBear: never tried it though, so you better do :)
15:15.32afethreat|BX: but I think a tdm400 with the number of ports needed works best
15:15.33TheBearafe: thanks yeah I will just wanted to check it was possible, seems logical, but logical is not always possible
15:15.43lidlthreat|BX, http://www.voipsupply.com/product_info.php?cPath=96_118&products_id=33
15:15.59afethreat|BX: unless you go with SIP phones
15:16.54threat|BXafe ok, but for each SIP device I need to connect it to a port on a special type of PCI card? (E.g. TDM400?)
15:17.05threat|BXoh, SIP is good?
15:17.43*** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1)
15:18.16*** join/#asterisk Nohair (~Jez@srscomp.demon.co.uk)
15:18.20afethreat|BX: the sip devices connects over ethernet, and your server won't need anything but a ethernet connection for that
15:18.51afethe TDM400 is only needed for analog stuff
15:18.55threat|BXafe nice
15:18.58NohairHi can any one help I have asterisk running on Macos when I am left a voicemail its just white noise
15:19.46PatrickDKtdm400 is needed for a zaptel timing device in 3.3v pci systems
15:19.55PatrickDKunless you feel liking paying for the t1 card
15:20.09threat|BXhmmmmm
15:20.16threat|BXI cant get T1 here :(
15:20.28threat|BXE1 maybe, but I believe its uber expensive
15:20.32afePatrickDK: you don't need the timing device if you're not doing meetme, and you can use the ztdummy
15:20.34PatrickDKheh t1/e1
15:20.35Nohairany one help with the voicemail noise problems????
15:20.42PatrickDKztdummy sucks
15:20.42threat|BXÖP
15:20.46threat|BXÖP even
15:20.51threat|BXgaz kezboard is fscking up
15:20.57PatrickDKand meetme/iax trunking is required for me
15:21.25TheBearanyone know why? (1) I have subscribed to asterisk-user@digium  yet I have not received any acceptance email to approve my subscription  (2) I have registered with IAXTEL and also not received any email with my password ?
15:21.48CosmicRaysounds like your e-mail is hosed
15:21.59NohairAny one running asterisk on a mac????
15:22.26afePatrickDK: if only for timing I suppose any X100P clone would work
15:22.35TheBearCosmicRay: I can get junk and other mail fine :( in fact I've been checking every 5 sec. and keep receiveing mail I don't want, and not the mail I'm waiting for
15:22.38PatrickDKafe, I said 3.3v pci
15:22.39threat|BXafe, thanx for zour help Ö=  same with the others Ö=  I will havea read and think about setting up one of these szstems Ö=
15:22.50PatrickDKservers don't normally have 32bit 5v pci slots
15:22.55threat|BXPatrickDK 3.3v PCI isnt normal_
15:23.02afePatrickDK: my mistake
15:23.08PatrickDK3.3v pci is standard for servers
15:23.13threat|BXnice
15:23.18threat|BXwell BBL
15:23.36Nohairand one running asterisk on a mac
15:24.05afeNohair: running asterisk on OSX is not very common :)
15:24.39Nohairaft :-(
15:25.12afePatrickDK: do the X100P really require 5V=
15:25.22PatrickDKafe, the 3 I have do
15:25.37PatrickDKand so said digium
15:25.47PatrickDKthough the card has the 3.3v cut in it
15:27.35afePatrickDK: hmm... I know the TDM400 needs extra power, but I didn't think the X100p (or clones) did
15:27.46PatrickDKthey don't need extra power
15:27.50PatrickDKthey need a 5v pci slot
15:28.01PatrickDKmost 32bit pci slots are 5v
15:28.11PatrickDKand ALL 64bit slots are 3.3v
15:28.42tzangerwerd 'em up
15:28.48afePatrickDK: ah... sorry - now I see your problem (I'm a bit tired atm)
15:29.02*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
15:29.04coppiceall 66MHz slots are 3.3V. Not all 64bit slots
15:29.25PatrickDKheh, ya, someone always makes it different
15:29.37PatrickDKpci was made to do almost anything, makes it more confusin
15:30.38afePatrickDK: however, if you run asterisk on a Soho system you probably don't have those slots anyway ;)
15:31.06PatrickDKhmm, my soho systems do
15:31.59PatrickDKheh, 7 people in the office, 3 servers
15:32.04afeif that's a problem I would run asterisk on a desktop computer rather than a server :)
15:32.14PatrickDKnot reliable enough
15:32.24PatrickDKI perfer redundant cpu/memory/bios
15:32.46afePatrickDK: of course, most soho don't do that :)
15:32.48WhiteWlfstandalone old-school PBX'es didn't have much redundancy... but they never went down either
15:33.11PatrickDKwhitewlf, I have seen enough break
15:33.16WhiteWlfme too
15:33.17WhiteWlfheh
15:33.48afeif you need the redundancy, you can probably pay for a tdm400 I guess
15:33.48*** join/#asterisk Ron-Na (~ronald@203.70.36.126)
15:33.48tzangerwtf do you need 3 servers in a 7-person office for?
15:34.07Ron-NaDoes anybody has experience with Realtime?
15:34.13PatrickDKone for telephone
15:34.15tzangerWhiteWlf: actually they went down often enough...  usually the trunk cards (makes sense)
15:34.23PatrickDKone for harddrive shared storage
15:34.33PatrickDKone doing backups of all systems
15:34.45WhiteWlftzanger: the one at meh office is rebooted once a week and has 450 users and one of those oldsckewl tape backup drives in it
15:34.47tzangerI think we have 6 or 7 for a 50-person manufacturing org, could consolidate it down to probably 3
15:34.50PatrickDKhave had too many harddrive failures already in he last 4 years
15:34.58PatrickDKabout 1 drive every year
15:35.03tzangerPatrickDK: *nods*
15:35.04PatrickDKfor ide drives
15:35.12tzangerthat's what RAID'xs for :-)
15:35.13PatrickDKservers running scsi
15:35.18tzangerand hotswap, it just works too well
15:35.22PatrickDKhmm, you don't normally raid desktop systems
15:35.22tzangerpainfully well
15:35.27tzangerPatrickDK: no not desktops
15:35.29tzangeryou're correct
15:35.44Mocthe problem is hotswap is that it the cause of most problem
15:35.51tzangerwe've got a company who loves those diskless systems
15:35.56tzangerI can't think of the anme of htem offhand
15:35.57Mocbecause of the connection of the hotswap bay
15:35.58PatrickDKI have never had a problem with hotswap
15:35.59Ron-NaDoes anybody use realtime?
15:36.03tzangerRon-Na: not me
15:36.04WhiteWlfWinTerms....
15:36.11PatrickDKmoc, that is just compaq hoswap, many roblems with them
15:36.12tzangerWhiteWlf: yeah something like that
15:36.16fileooh twisted is unhappy at being woken up
15:36.19MocPatrickDK, Dell also
15:36.20WhiteWlfoh man do i hate winterms
15:36.20PatrickDKI haven't had a problem with my intel hotswap
15:36.24WhiteWlfThat leads me into a question about creating a dialplan... what if I want to prompt a user for an extension... then route them to voicemail... like directly to voicemail... how do I capture the user input at a menu to use it to send it to voicemail?
15:36.30PatrickDKhell, I would never trust dell
15:36.32tzangerfile: you shouldn't get out of bed so noisily then
15:36.43filewell it was bkw who slammed the door
15:36.46WhiteWlfdell is just a company that puts parts together... how the hell can you not trust it?
15:36.49tzangerfile: heh
15:36.57PatrickDKwhitewlf, theydon't put parts together
15:37.05PatrickDKhave you seen thier motherboard designs?
15:37.12lidlhow much it costs in your country a t1 or e1 line?
15:37.17PatrickDKthey have them made to their specs
15:37.23PatrickDKand isn't always reliable
15:37.26WhiteWlfyeah, and i can order exact replacements from the company they get them from
15:37.37MocCanada/Montreal, about 700$/month+ for 1 year contract
15:37.39MocCND
15:37.40WhiteWlfviva la asus, abit etc
15:37.42afedell is working perfectly fine for me (and for lots of our customers) - however, don't expect to be able to upgrade them with standard parts
15:37.49Ron-NaI want to charge incoming calls in ASTCC, does anybody know how to set this up?
15:38.20PatrickDKI'll stick to intel and supermicro systems
15:38.20fileRon-Na: astcc is a _calling card_ application
15:38.30filelol, twisted just freaked out at the TV because it was loud
15:38.34MocRon-Na adjust your outgoing price in consequence
15:38.37lidlMoc, canada, so t1, so 24 channels + signalling, isn't it?
15:38.45PatrickDK23+signal
15:38.48PatrickDK24 channels total
15:38.52Mocit 23B channel + 1 dchannel
15:39.01coppiceIf you are in a big company Dells are great. for a small business or home user results vary
15:39.05lidlyes, my fault
15:39.06Ron-NaMoc, I don't understand that
15:39.31WhiteWlfIt has 23 channels, and you have to have 1 signal for every set.... 23+1 = 24 total
15:39.52afecoppice: I wouldn't get a dell form home use (since I like to build my own computers, and I have 5 at home)
15:40.12WhiteWlfdid anyone understand my question or did I just like... do a morning?
15:40.13MocI got a Dell 1850 for my personal use, it aint bad
15:40.42afecoppice: but at work we just got 4 simple dells for $250 each - that really good if you only need them for surfing and office use
15:41.38coppiceMoc: its not so much he machine. its the support. if a big company has problems Dell's response is excellent. if you are small potatoes to them, they couldn't care less
15:41.50afedells High End computers are not bad either, but they can't be upgraded (easily)
15:42.20coppiceafe: in asia we never get any of those cheap deals. Dell machines are quite expensive
15:42.34lidlMoc, just a curiosity, for each voice channel do you get a different phone number?
15:42.37WhiteWlfI'd really prefer not to spout my question again... it's rude... but I'd like to know if anyone saw it?
15:42.44afeBig companies and government agencies don't upgrade computers, so for that purpose they're excellent
15:43.03Nuggetclearly the only rational solution is to buy a powermac.
15:43.16Moclidl, you can get as many phone number you wish
15:43.35afecoppice: here they vary from extremely cheap to very expensive :)
15:44.09afeyou can buy a machine one day, and it can be bought at half that price one week after ...
15:44.16coppicelidl: phone numbers are not normally related to channels on a PRI, unless its being used in a channel bankstyle to connect to 23 phones
15:44.20Moclidl, the B Channel is the channel that allow voice or data communication.  the D Channel have the 'signaling'. For example, a incoming call with DID 555 555 5555 is calling, so my system tell them ok use this B channel
15:44.35MikeJ[Jayden]hey Moc..
15:45.18coppiceafe: they endlessly advertise cheap deals in the newspapers. go to their site the day the paper is published and try the code - the system says it has expired. even those deals are not that cheap, though
15:45.19lidlMoc, coppice, thanks for the clarification :)
15:45.30Mochi MikeJ
15:46.55FaithfulI just experienced skype
15:47.20*** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter)
15:47.32FaithfulIt's got me worried
15:47.34MikeJ[Jayden]well.. I'm off to go have some exciting reading of Advanced UNIX Programming, second edition :)
15:47.41Ron-NaI use to say Skype is a toy, but it worked,.... on my Athlon it stopped to work
15:48.01lidlso, having a t1 let's you have a phone number which can handle 23 total calls (incoming/outgoing) concurrently
15:48.08Ron-NaThe service of Skype is zero, ...
15:48.22Faithfulthe issue for me is it works better than G729 does
15:48.45coppiceFaithful: of course it works better than G.729
15:48.46FaithfulMaybe I should change codecs
15:48.51Moclidl, yes,
15:48.56lidlok, thx
15:49.03Faithfulcoppice: why do you say that?
15:49.09*** join/#asterisk gonzo- (~gonzo@icc-nat.univ.kiev.ua)
15:49.20coppiceit uses a wideband codec at a much higher bit rate
15:49.33tzangermorning coppice
15:49.59tzangerI should soon be able to start the discovery phase of when the wctdm driver started sucking
15:50.00FaithfulI thought G729 was the VoIP codec of the gods
15:50.01coppiceif you don't have jitter and loss Skype should be considerably better than a land-line
15:50.18coppiceG.729 is the codec of the bandwidth starved
15:50.44FaithfulOh... so I bought it for nothing
15:50.45MocFaithful, the best codec in term of bandwidth and qualify, I think it g726
15:50.50coppiceits not great, its just pretty good for 8kbps
15:50.54*** join/#asterisk Mike (~mike@201.135.48.217)
15:51.10Mocqualify=quality
15:51.13Faithfulit suffers from jitter pretty bad
15:51.21coppiceG.729 is mostly something you need because the other end is using it
15:51.25FaithfulI gather that iLBC is better
15:51.51MociLBC is DEAD FOR ME ... it the worst codec !!! (mostly because the code is buggy)
15:52.06MikeMoc, buggy?
15:52.13FaithfulMy VoIP provider told me G729 is so much better than GSM
15:52.16MocI had terrible jitter with ilbc, and had issue that Asterisk go take 100% CPU also using that codec ..
15:52.23coppiceMoc: iLBC is fine. It must be how you are using it
15:52.37FaithfulSkype is iLBC correct
15:52.52coppiceSkype is iLBC wideband
15:52.53MocIm talking about Asterisk implementation of iLBC
15:53.07mikegrbskype is proprietary crap
15:53.17MocEvery time I get to use ilbc, problem appear ...
15:53.26WhiteWlfIn an IVR menu... how can I have it prompt an extension and then dial that extension?
15:53.39mikegrbWhiteWlf: by reading some damn docs
15:53.54mikegrbWhiteWlf: and not expecting everyone to hand you everything without work on your part
15:53.55WhiteWlfmikegrb: I'm not framiliar with that specific doc ;)
15:54.02tzangercoppice: I spoke to someone in France (I'm in Canada) over Skype (he called my POTS line) ... there was signficant delay but other than that it wasn't bad
15:54.04Moc~docs
15:54.05jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:54.12Mikehi coppice !!
15:54.22tzangerit was kind of funny, he was going on about how amazing voip is but he had no idea that ever single call we've been taking and placing over the past year has been VOIP :-)
15:54.32WhiteWlfThank you moc... mikegrb: this is a hobby... not a requirement of myself... so I enjoy the help I get when I get it. :D
15:54.33coppicetzanger: skype is designed to work really great when conditions are good.
15:54.42Faithfulcoppice: wideband?
15:54.45tzangercoppice: I find iLBC's voice quality subpar, at least compared to GSM
15:54.52mikegrbWhiteWlf: read some docs and you will get a lot more help
15:55.18MocWhiteWlf, if it a hobby, you should be searching by yourself even more ;)
15:55.27coppiceFaithful: 16k samples/second == 8kHz bandwidth. you can tell "s" from "f" :-)
15:55.29tzangercoppice: I'm not sure why though -- but every time I change my default codec from GSM to iLBC I get complaints... and the hardware's more than enough to be able ot handle it (Xeon systems)
15:55.51Moctzanger, yes, ilbc have weird problems
15:55.54coppiceMike: hi. how's things?
15:56.02Mocin asterisk atless
15:56.19Mikecoppice, after e1 worked they didnt call me no more:(
15:56.53MikeMoc, i use alot iLBC i find it really good
15:56.58MikeMoc, even better than gsm
15:57.31Faithfulso should I be using gsm over G729?
15:57.34Faithfulman
15:57.40coppiceMike: R2 seems pretty solid now. There are still a couple of odd things which happen infrequently, though
15:57.45MikeFaithful, anything is better than g729
15:58.03*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
15:58.04FaithfulOh, no... and I paid money for it
15:58.17FaithfulI was told the opposite
15:58.26MocFaithful, g729 is good with you got very low bandwidth
15:58.27coppicecomparing codecs is a complex thing. saying one is better is meaningless on its own. better at what?
15:58.47FaithfulI got jitter... what's good for gitter? iLBC?
15:58.53Mocbut g729 is CPU hungry, and qualify aint that great, but it aint that bad
15:59.29*** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com)
15:59.31Faithfulwhen I switched everyone I spoke to commented
15:59.40Faithfulwithout my saying
15:59.43coppiceiLBC is good for packet loss tolerance. jitter is a jitter buffering issue. not a codec one
15:59.53Faithfulquality went down from GSM
16:00.46Faithfulbut I gess the difference between jitter & packet loss can be undescernable at times
16:00.47MocI always use ulaw, except for 2office that I had to make 20 voice channel go throught a DSL, so I used g726 (sound nearly as ulaw)
16:02.13FaithfulMan I have so much to learn about all this stuff ... codecs etc
16:02.20lidlon voipsupply i see no gsm gateways. has anyone experienced them?
16:02.37Mocgsm gateway ???
16:02.58puppetthis is evil
16:04.32coppicethe nastiest things with codecs is mixing them. avoid transcoding at all costs
16:04.42lidllike this one: http://shop.voismart.it/proddetail.php?prod=GSM-1ANA
16:05.55lidlbasically, you put a sim inside, connect it to an fxs and apply least call routing
16:09.52filelalala
16:10.09lidlhttp://www.2n.cz/products/gsm_gateways/analog/analog_gsm_gateway.html
16:10.10puppetHave you all seen German Darth Vader?
16:10.18puppethttp://www.efterbliven.de/pics/ohhgawd23.jpg < German Darth Vader
16:10.25coppicethose GSM units require GSM<->analogue<->something digital with a quality loss. there are some units which allow digital audio to go straight into the GSM unit now.
16:10.41*** join/#asterisk RoyK (~roy@ti211210a080-2089.bb.online.no)
16:11.13lidlcoppice, any vendor?
16:12.06puppetcoppice: oh what do they cost?
16:12.15coppicedunno. try google. it used to be all the GSM modems only allowed audio to be fed in as analogue, but i was told recently some of them support digital audio now
16:12.38RoyKhmmmm
16:12.38carloshguys: what is aec ?
16:13.09RoyKautomatic echo cancellation?
16:13.14carloshta.
16:13.23coppiceWe used to use Falcom GSM modems for mass SMS. those were about $250. if units like that now support digital audio they should still be the same price
16:13.31coppiceacoustic echo cancellation
16:14.08carloshthanks.. been trying to tune up this IP phone, so I can hear the hold music... grrr..
16:14.29coppiceby which they usually mean voice on a line, rather than modem signals on a line. AEC sounds like it means cancelling rooms echos, but it usually doesn't
16:14.37RoyKif I have a queue with 10 SIP queue members, can I, when someone calls in, check wheather there are any SIP clients connected, and if not, play some message saying "fuck off" or something?
16:15.17`SauronHow polite.
16:15.36RoyKwhatever.
16:15.40RoyKis it possible?
16:15.59*** join/#asterisk _RaYmAn_ (user@213.237.12.147.adsl.vby.tiscali.dk)
16:16.09`SauronYou could read the documentation and find out.
16:16.10coppiceRoyK: I suggest you just tell them to go away. leave it up to them whether the use the time for sexual activities
16:16.23RoyK~lart coppice
16:16.57puppetcoppice: http://www.falcom.dk/fala2d-3.pdf ?
16:17.51Nuggethttp://lnk.nu/slacker.com/lt   <-- lart
16:18.32*** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com)
16:19.16`SauronNugget: It's nice outside. What are you doing not driving?
16:19.26Nuggetgearing up for sxsw.
16:19.36`SauronGoing today?
16:19.39Nuggetyeah
16:20.09`SauronHum.
16:20.23`SauronWith all the sxsw traffic, wonder how hard it's gunna be to get to Music Makers
16:20.27carloshsummary of my testing.. gee...  this ip phone only works kind of ok with ulaw and alaw.. g729 not fully licensed here.. and it won't play the holf music.. I had to allow only either ala or ulaw at a time in order to get good quality audio at the other end.. a software client...
16:20.58carlosh5:20 am here... off to bed.. :/
16:21.24filebe back tomorrow, going to... well... fly home
16:21.28carloshif anyone can recommend a good/affordable iphone, with ilbc support, much appreciaed beforehand..
16:21.47Zeeekbt100
16:21.58*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
16:23.07carloshZeeek: thanks.
16:23.23Zeeekmeets the two criteria you gave
16:23.39lidlZeeek, budgetone100?
16:23.53Zeeekto be precise BT102 is the one I rccomand
16:24.08Zeeekhas a hub for easier connection
16:24.14*** join/#asterisk jeffik (~jeffik@m5f7436d0.tmodns.net)
16:24.28lidla friend of mine said they look like toys, but are ok for residential use
16:24.42Zeeekmany, many of these phones look like toys
16:24.49`Sauronwhy ilbc?
16:24.50carloshsure, it has the two ethernet ifaces as well
16:24.59Zeeekthe cisco looks like it should be in the war room of the pentagon
16:25.11Zeeekand it probably is
16:25.16carloshZeeek: ilbc support ?
16:25.32Zeeekagain: meets the two criteria you gave
16:26.02zoahey royk
16:26.08Zeeekfile your photo has now been shown worldwide
16:26.11carlosh:o)
16:26.24`Sauronwhich photo?
16:26.30Zeeeknot the nude one
16:26.35Zeeekthat requires a password
16:26.39zoawhere is file ?
16:26.42Zeeekor an .htaccess hack
16:27.04Zeeekyou'll have to search
16:27.27Zeeekexten => hint,1,mailingList(recent)
16:27.42*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
16:27.42Nuggetheh
16:28.43puppethttp://www.efterbliven.de/pics/ohhgawd23.jpg < German Darth Vader
16:28.46puppetthat one
16:28.51puppetthats nugget
16:28.53puppetoh wait
16:28.55puppetwrong pic
16:28.56carloshZeeek: do you know what is the standard number for ilbc ?  gxxx ?
16:29.00puppethttp://slacker.com/photos/2002mroadster/IMG_1381
16:29.02puppetthere ;p
16:29.08Zeeekcarlosh no
16:29.34carloshthe pdf of the budgetone does not mention ilbc..
16:29.49Zeeekcarlosh it does ilbc
16:30.08carloshZeeek: thanks mate, off now, thanks all
16:30.11Zeeekintroduced around 5.11 firmware or something like that - a while ago now
16:30.57zoai have such a terrible jetlag now
16:31.06zoastruggling to stay awake
16:31.16Zeeekafraid of nightmares?
16:32.08Zeeeksearch for "pictures" in the asterisk users mailing list reveals a few interesting voIP personalities
16:32.36tzangerZeeek: eh?
16:33.22ZeeekAKA "Spring VON Asterisk Pavilion"
16:34.58`Sauronalright
16:35.05`SauronOff to do a brake job. How fun.
16:35.16tzanger`Sauron: doing it yourself or getting it done
16:35.18afemamma mu!!!
16:35.21`Sauronmyself
16:35.27`Sauronc'mon, I'm a real MAN. :)
16:35.27tzanger`Sauron: excellent
16:35.38tzangerI have to do brakes on my vehcile too
16:35.40tzangerI miss my jeep so much
16:36.52zoahehe
16:36.54zoai was there too
16:37.47Zeeeksomeone was talking about how linux uses RAM on the ML... I noticed free RAM went from like 400M to 100M between 4:30 and 4:45 - that's when the daily and weekly cronjobs are run
16:38.04ZeeekI've been monitoring RAM because ti appeared there was a leak
16:38.45Inv_arpif context A includes => B    and B includes => C            can A access C?
16:38.46Zeeekbut the cronjob that ran is just a stock script to archive logs
16:38.55NuggetInv_arp: yes
16:38.57Zeeekit would seem
16:39.41Inv_arphmm not good... when anyonecalls in they can access my outside lines by pressing 91NXXNXXX
16:39.49agave-txlinkheh
16:39.52agave-txlinkasterisk has memory leaks
16:40.03agave-txlinkeverytime I do a reload, mem usage goes up 2M
16:40.24zoahmm
16:40.26zoareport it
16:40.36zoaits not certain the actual mem usage goes up
16:40.43zoaas linux doesnt reclaim memory until it needs it
16:40.43agave-txlinkyeah, it's certain
16:40.48agave-txlinkonce I hit total mem used
16:40.50agave-txlinkasterisk crashes
16:40.54zoaaha
16:40.55zoayeah
16:40.58agave-txlinki'm running 1.0.6 though
16:41.00zoathats a memleak
16:41.01agave-txlinki'll try 1.0.7 before I bitch about it
16:41.17Zeeekby golly, he's right!
16:41.44Zeeekno, in fact it went down - 1.0.6
16:41.47EightI don't suppose there's magically a changelog for .0.6->.0.7?
16:43.33Inv_arpok context A includes => B and B includes =>C   anyway to prevent A knowing about C?
16:44.07agave-txlinkwhat are you trying to accomplish invarp?
16:44.12ZeeekInv_arp I think you need to build your contexts differently
16:44.47Zeeekwith the sooper power stuff included only in the priviliged users context
16:44.55Inv_arpagave-txlink: when anyone dials in my menus   "A" if they press 91  they get outside line "C"
16:45.03agave-txlinkokay
16:45.09agave-txlinkyour menus should have their own context
16:45.22agave-txlinkif you're using a DID, trap the DID and then do a Goto to your menu context
16:45.46agave-txlinkspeaking of asterisk bugs, if you do too many include => then asterisk will crash
16:45.57*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
16:46.01agave-txlinkhey kevin
16:46.40Zeeekhow many includes?
16:47.02agave-txlinkzeeek: i didn't count, it was a production system so I had to fix quickly, but I'd say around 100 or so
16:47.13Zeeekya well that is a lot of includes
16:47.33agave-txlinktrue
16:48.18Zeeeknot nested?
16:48.24agave-txlinkno, not nested
16:48.33Zeeekexcuse me I need to go open a bottle of Château La Mission Haut-Brion 1997
16:48.36agave-txlinkwhich is why I was able to switch to #include
16:48.55Zeeek#include rocks for making the dialpan hipper
16:49.18Inv_arpi seperate internal/external  using #include
16:54.21zoahey mark
16:54.24zoaim home already
16:54.28kramyay!
16:54.33kramglad you made it safe
16:54.44zoait was horrible
16:54.49zoacouldnt sleep all night
16:54.50kramhow horrible?
16:54.53krami sowwy!
16:54.57zoatook ages to get home
16:54.58zoa:)
16:54.59kramyou must have been thinking about asterisk
16:55.03zoahehe lol
16:55.51*** join/#asterisk bah (048830696@AC90B4BA.ipt.aol.com)
16:58.15*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
16:58.30*** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk)
16:59.26*** join/#asterisk Frantic (~ab@w020.z066088084.nyc-ny.dsl.cnc.net)
16:59.32*** join/#asterisk ws9455 (~ws9455@adsl-68-94-10-246.dsl.rcsntx.swbell.net)
17:11.32*** part/#asterisk jeffik (~jeffik@m5f7436d0.tmodns.net)
17:14.01*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjhe.dialup.mindspring.com)
17:20.38*** join/#asterisk n3tar (~geno@201.254.93.202)
17:20.41n3tarhi
17:20.47*** join/#asterisk spackle (~spackle@209.234.83.19)
17:22.09*** join/#asterisk Necko (~roy@IGLD-83-130-105-236.inter.net.il)
17:22.16Neckohello
17:22.43PTG123hey anyone awake know the polycom well?
17:23.00Zeeekhere we go again :)
17:23.07PTG123hehe
17:23.10PTG123well i unlocked it now
17:23.14Zeeekyou missed ManxPower
17:23.18PTG123now i just need to know how to make it register
17:23.21PTG123i can make a call
17:23.26PTG123its just not registering
17:23.32Zeeekyou were there this morning
17:23.45PTG123you mean he was here?
17:23.53Zeeekright after yuou left
17:23.55spackleAnybody know specifically what nat=yes or no does?  I did a search of the wiki and just find samples of it.
17:24.04PTG123yes
17:24.14NeckoIs there a version of asterisk that i can install on an exisitng linux distro?
17:24.25PTG123nat doesnt' listen to the headers for the ip address of the host, and uses the actualyip it comes from.. nat=yes should ALWAYS be on
17:24.28ZeeekNecko sure
17:24.41Zeeekuse cvs or download a tarball
17:24.47Neckothanks.:)
17:24.55spacklePTG123 does it send a keepalive or anything?
17:24.58Zeeeksee digium.com -> download
17:25.04Zeeekor
17:25.05ZeeekStarter tutorial:
17:25.05Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
17:25.05Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
17:25.05Zeeekhttp://www.automated.it/guidetoasterisk.htm
17:25.05ZeeekTHE reference of the moment:
17:25.06Zeeekhttp://www.asteriskdocs.org
17:25.21Zeeekthe automated stuff shows a whole procedure
17:25.35Zeeekso does asteriskdocs.org
17:25.43PTG123spackle: no
17:26.11*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
17:26.12Neckothanks!
17:26.20Zeeekcheers
17:26.31spacklePTG123: thanks
17:26.35Neckoanother little question ...can asterisk behave as an h323 gw?
17:26.47Zeeekit would appear that it can
17:26.56Zeeek~wiki
17:26.59TheBearI have my TDM10B (with FXS module) installed, and a std plugged in. What config files to I need to change to have * dial the std phone on an incoming call
17:27.08ZeeekI can never remember the URL
17:27.14PTG123ok i helped someone, so should karma make someone help me
17:27.19TheBearI know my extensions.conf must have Zap/2 probm but how do I define zap/2
17:27.34ZeeekPTG123 wait for Manxpower - wait do you use 1.0.6 STABLE?
17:27.39EssobiMorning peeps
17:27.57ZeeekPTG123 if so, go chase him down in #asterisk-stable
17:28.10PTG123hehe
17:28.16PTG123stable is for chickens
17:28.22ZeeekTheBear ZAP channels are defined in zapata.conf
17:28.34Zeeekpreted you use STABLE to find ManxPower
17:28.36TheBearZeeek: ok what about zaptel.con ?
17:28.38Neckoi mean,will i be able to create a dial peer of h323 which will get a request from an ip and if the destination pattern starts with ,03 forward it to the PSTN?(through a specified modem)
17:28.48PTG123Zeek: he only likes stable? :)
17:28.52Zeeekyes zaptel, not zapata
17:29.24Zeeekfxsks = 1,2
17:29.24Zeeekfxoks = 3,4,5
17:29.38Zeeekyou have statements like those ytwo in yours?
17:29.58TheBearfxsks=1 only yes, which is my X100p
17:30.06Zeeekso it's ZAP/1
17:30.10Zeeekand that's an end to it
17:30.40TheBearyes, now I have installed my TDM10B, along with the x100p
17:30.41Zeeekyou ain't got no ZAP/2 bro
17:30.47Neckoif i were to connect 4 phones to the asterisk machine i would have need 4 fxs(modem-line or phone?) right?
17:30.51Zeeekah
17:30.55*** join/#asterisk Gh0sty (~Ghosty@81.11.192.116)
17:30.59Zeeekthen you will need to install the driver
17:31.04TheBearI want to use the FXS module in the TDM10B to dial a std analog phone
17:31.07Zeeekthe name of which escapes me
17:31.26Zeeekyou have an FXS module. You need another statement in zaptel
17:31.35TheBearwctdm, I found that on wiki, but I'm not clear on what other conf files need to be changed
17:31.36Zeeeklike fxoks=2 ?
17:31.49Zeeekadd that
17:32.03TheBearok I just added that, what about zapata.conf it only have channel => 1
17:32.20Zeeekfirst you'll want to see if ztcfg says anything
17:32.35ZeeekI can't remember if you need to unload and reload the drivers first though
17:32.52Zeeektype this ztcfg -vvv and see what iot says
17:33.05TheBear2 channels configured.
17:33.10Zeeekyou got it!
17:33.12TheBearZT_CHANCONFIG failed on channel 2: No such device or address (6)
17:33.19Zeeeknow go read about zapata in the samples
17:33.53Zeeeki think you need to stop astrisk and unload/reload the zaptel drivers
17:35.10ZeeekTheBear check this out : http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
17:36.09TheBearok I'll try that
17:39.25TheBearreload the machine, and the modules, still the same messages from ztcfg -vvvv
17:39.39Zeeekanyone discoverdid you read the wiki page?
17:40.06*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
17:41.47*** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com)
17:41.51TheBearZeeek: also * fails to start "WARNING[6710]: chan_zap.c:848 zt_open: Unable to specify channel 2: No such device or address"
17:42.09Zeeeknot surprising that asterisk wouldn't like it
17:42.14Zeeekif ztcfg doesn't
17:47.14*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
17:47.23TheBearHow can I get zrtcfg to recognize the phone connected to the FXS module ?
17:47.26ariel_good afternoon everyone.
17:47.47ariel_TheBear, ztcfg -vv
17:47.49Zeeekit doesn't care about phones
17:47.57Zeeekonly the hardware itself
17:48.14Zeeekdo this:
17:48.18Zeeekcat /proc/interrupts
17:49.13PTG123Mar 12 10:45:41 NOTICE[6672]: chan_sip.c:8152 handle_request: Registration from '<sip:66.235.234.131@66.235.234.131>' failed for '68.106.24.139'
17:49.15Zeeekdo you see wctdm ?
17:49.21PTG123does that mean its specifying the ip as username?
17:49.23TheBearno
17:49.35ZeeekTheBear did you modprobe it?
17:49.51TheBearyes modprobe wctdm and it is listed in lsmod
17:50.03Zeeekand ztcfg gives an error?
17:50.09TheBearModule                  Size  Used by
17:50.09TheBearwctdm                 121728  -
17:50.09TheBearwcfxo                  10688  -
17:50.09TheBearzaptel                220516  -
17:50.21TheBear2 channels configured.
17:50.21TheBearZT_CHANCONFIG failed on channel 2: No such device or address (6)
17:50.40Zeeekdid you remove and then reload the drivers a few minutes ago like I asked?
17:51.09TheBearyes I even rebooted the machine and then did the modprobe from scratch
17:51.39Zeeekwhat distro and kernel?
17:51.49TheBeargentoo kernel 2.6.10
17:51.57Zeeeksounds like it's time to do some serious mailing list searching
17:52.15Zeeek2.6 kernel has issues that should be found somewhere
17:52.50Zeeeksomething about that here:
17:52.51Zeeekhttp://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
17:53.02TheBearI've tried to subscribe to the mailing list since thursday I'm still waiting for my subscription confirmation
17:53.08ZeeekSupport for Kernel 2.6
17:53.08Zeeek<PROTECTED>
17:53.18Zeeekdid you read that? ^^^^^^^^^^^^^^^^^
17:53.56Zeeekor this:
17:53.58Zeeek"Sep 2004: When compiling zaptel for ztdummy, be sure to link /usr/src/linux-2.6 to /lib/modules/2.6.x.x.etc/build so modprobe will succeed.
17:54.28TheBearyes I did the make linux26 I found that when I compiled and installed *
17:54.36ZeeekTheBear you can search the list without subscribing using google
17:54.42TheBearperhaps the acpi=off then ?
17:54.47TheBearlet me try that
17:56.28ZeeekIf you do a google search on "ZT_CHANCONFIG failed on channel 2: No such device or address" 45 pages come up
17:57.29TheBear<PROTECTED>
17:58.37TheBearmost that I have seen so far are talking about the USB version
18:01.25ManxPower~docs
18:01.26jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
18:01.42TheBearHi Eric
18:01.54*** join/#asterisk _tommyg_ (~tom@vsat-148-64-73-166.c119.t7.mrt.starband.net)
18:02.18ManxPowerTheBear, Does ztcfg -vvv show any errors?
18:02.34PTG123i got my phone working, woo hoo ;)
18:02.37*** join/#asterisk LorenzoMarouani (~LorenzoMa@AVelizy-112-1-9-62.w81-49.abo.wanadoo.fr)
18:02.41_tommyg_Any one out there tried a ShoreTel 530 with asterisk?  Having a prob with MGCP
18:06.00spackle~NAT
18:06.03jbotnat is probably Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
18:06.20TheBearManxPower: no just the two line about the channels and thos two give me a sec to bring the machine up and I'll display it all
18:06.52TheBearModule                  Size  Used by
18:06.52TheBearwcfxo                  10688  -
18:06.52TheBearwctdm                 121728  -
18:06.52TheBearzaptel                220516  -
18:07.05TheBearZaptel Configuration
18:07.05TheBear======================
18:07.09TheBearChannel map:
18:07.14TheBearChannel 01: FXS Kewlstart (Default) (Slaves: 01)
18:07.14TheBearChannel 05: FXO Kewlstart (Default) (Slaves: 05)
18:07.18TheBear2 channels configured.
18:07.23TheBearZT_CHANCONFIG failed on channel 5: No such device or address (6)
18:07.57TheBearI tried changing fxoks=5 from =2 no joy, also can't seem to get to the acpi=off
18:08.05EssobiI'm coming up with a new acronym for a project.. Someone give me a word that begins with F that fits... fields, forums, forms... F.. something that means website.. MMM
18:08.48Essobi_tommyg_ Why would you want to use MGCP?  No sip images for a 530?
18:09.18nestArisn't the shoretel just a Polycom IPX00?
18:09.29_tommyg_I do not believe so, if so I have no way of getting them
18:09.34_tommyg_Not sure
18:09.38EssobiHeh.
18:09.41_tommyg_docs only speak of mgcp
18:09.45EssobiNothing is impossible.
18:09.47puppetthebear: dont spam channel PLEASE
18:09.53puppet~pastebin
18:09.54jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
18:09.54EssobiMGCP is so durty
18:09.58puppetthebear: use that please
18:10.06tessierEssobi: Division by zero is impossible
18:10.25TheBearsorry
18:10.31TheBear:=(
18:10.39CosmicRaydoes anyone here have rules to support the 1010 carrier selection when dialing out?
18:10.51EssobiTessier Unless you redefine 0.  Or Division.
18:11.28tessierzero is zero. You can write it with a different character but the concept is still the same.
18:11.29_tommyg_OK I am gonna look for some SIP images, not sure if they are out there or not
18:11.34nestArthe shoreline IP 100 is a polycom, i bet the 530 is too
18:12.17*** part/#asterisk Ash (aaron@fudgecom.net)
18:12.19nestArhttp://www.voip-info.org/wiki-Polycom+Phones#comments <--- there's mention of the Shoretel 100 here
18:12.20TheBearLet me go read some more thanks
18:12.48_tommyg_Thanks, gonna check it out
18:15.57*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
18:20.56*** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
18:23.02*** join/#asterisk darby_t (~tom@dnk250.neoplus.adsl.tpnet.pl)
18:23.18*** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net)
18:25.03*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
18:25.03*** mode/#asterisk [+o twisted] by ChanServ
18:25.20twisted*sigh*
18:25.24ariel_~seen drumkilla
18:25.27jbotdrumkilla <~russell@12.21.241.80> was last seen on IRC in channel #asterisk, 9h 31m 42s ago, saying: 'ok ...'.
18:25.37mrgobyI'm having some issues with the Redirect command in the manager api ....    I'm trying to redirect a zap channel to a conference room....   it transfers to the context i specify and to the extension, but to the priority-1....  also, it doesnt actually connect to the extension ..  but it hangs until either input or timeout... any ideas ?
18:25.40ariel_hello drumkilla are you around today?
18:25.55ariel_hello twisted how are you doing today?
18:26.09twistedariel_, would be better if I wasn't sitting in an airport :P
18:26.22ariel_your heading back from Von?
18:26.24twistedyea
18:26.38shido6mrgoby
18:26.39twisteddrumkilla is in terminal C with bkw, and file is somewhere in terminal A here in san jose
18:26.41ariel_Did you like it? Was it worth it?
18:26.45twistedit was fun
18:26.47shido6mrgoby show me your dialplan
18:27.01mrgobyok, pastebin-ing it now
18:27.08ariel_I guess I will have to wait. I have a bug question.
18:27.19modulus_i'll show you mine if you show me yours
18:27.49twistedariel_, why is that?
18:28.42ariel_I have a question on bug 3577 which he put that it was not added to stable. But as far as I can see stable still has incominglimit active.
18:29.24twistedright....
18:29.43twistedthe patch wasn't added to stable to handle limit on peers
18:29.43ariel_so the bug 3577 is a fix and it should be added to stable.
18:29.59*** join/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il)
18:30.34twistedactually, it adds support for other items
18:30.38twistednot fixes any other bugs
18:30.47mrgobyhttp://pastebin.ca/7326
18:31.19mrgobythat is after the login and not including the logout on the manger api
18:31.22ariel_at least it looks like then this type of setting will stay with the product. I am having major problems with setgroup and getgroupcounts
18:31.48modulus_"Sorry, the requested ID value is way too large or too small."
18:31.55modulus_how brain-dead.
18:32.01modulus_pastebin.ca/0
18:32.10twistedariel_, what difficutlies?  I haven't had a problem using them
18:32.27mrgobyi was digging in manager.c  and i saw that it uses ast_asnyc_goto()  ... would this be why ?
18:32.51mrgobywhy it seems to hang, that is ?
18:33.14ariel_I am getting wrong counts from them and also it's not taking account when the users call outbound to exten only to pots or pri lines. It's just a very funny thing.
18:34.02adorahHave a query regarding setting a remote IAX extension: Once I added an extension in iax.conf do I set the extension in the extension.conf under context=default or open a new context say [iax]?
18:34.47mrgobyi also saw that on the other params, it is taking string length -1 for the value, which i assume is to get rid of null space and new line chars.... but with priority it just takes the val - 1... which i think may be a bug
18:34.52*** join/#asterisk santiago (~santiago@63.245.86.111)
18:35.06ariel_twisted main problem is when used in a rollover macro type it looses the callerIDNUM.  or it just keeps the orginal one.  Which I have been trying to reset when it goes to then next device.
18:35.26adorahHave a query regarding setting a remote IAX extension: Once I added an extension in iax.conf do I set the extension in the extension.conf under context=default or open a new context say [iax]?
18:36.48*** join/#asterisk Spooch (~rath@p549A1CF8.dip0.t-ipconnect.de)
18:38.14PTG123anyone have any idea why when i dial with my polycom it doesn't seem to look up a sip user?
18:38.19PTG123but yet it dials fine
18:39.56*** join/#asterisk file[airport] (file@dhcp64-134-126-76.sjca.sjc.wayport.net)
18:41.37twistedheh
18:42.02file[airport]don't you dare
18:42.05twistednah
18:42.11file[airport]I've got a mac mini and I'm not afraid to knock you unconcious with it
18:42.21Sedoroxahah
18:42.27Sedoroxdon't abuse it!!! sned it to me instred
18:42.30Sedoroxstead*
18:42.32file[airport]nah
18:43.20shido6make a fish tank
18:43.34Darwin35macmini wow
18:44.00ariel_mac  oh boy.  So did you change the os to a real linux build??
18:44.13Darwin35Darwin
18:45.34Sedoroxthe mini's are nice
18:45.52Sedoroxeven tho I like OSX.. I'll probably load gentoo on it
18:46.20ariel_is OS X not made from a version of BSD?
18:46.24file[airport]yes
18:46.36PTG123osx is freebsd basically
18:46.43mrgobydarwin is anyway
18:46.45Sedoroxwell the kernel
18:46.45Sedoroxyes
18:50.19*** join/#asterisk bah (048830696@AC892FC3.ipt.aol.com)
18:51.45PTG123you know i think i found a way to make a call on an asterisk server without registering, or authenticating
18:52.00PTG123no server is secure :)
18:52.01PTG123heh
18:52.28mikegrbsure
18:52.29*** join/#asterisk Dark40rce (~vince@dsl-17-106.cofs.net)
18:52.56ariel_PTG123, how are you doing this?
18:53.08PTG123put the username as the ip of the server
18:53.12EssobiMmm.
18:53.24PTG123seems to be letting my phone make calls no problem
18:53.27ariel_sample of the dial string?
18:53.38*** join/#asterisk lidl (~little@213-140-6-96.fastres.net)
18:53.46EssobiTwisted and File are going to throw down.
18:53.47PTG123actually one sec
18:53.51Essobi:)
18:54.07PTG123i think its setting the address as the server in the sip packet
18:54.08file[airport]haha
18:54.09PTG123not the username
18:55.02ariel_PTG123, so you have as username in lets say xlite with no password or any other settings?
18:55.11PTG123one sec checking
18:55.15PTG123put in some debugging in the code
18:55.16EssobiI got $5 on twisted
18:55.24EssobiWhat's the spread? :)
18:55.31PTG123yes its specifying usename as server ip address
18:55.35PTG123and its letting me make calls no problem
18:55.39PTG123i think i have a real password
18:55.51PTG123my polycom for some reason is doing it that way and its letting it make calls
18:56.01PTG123In find peer: 66.235.234.131
18:56.01PTG123TABLE: sipfriends
18:56.01PTG123==> name=66.235.234.131
18:56.01PTG123<PROTECTED>
18:56.01PTG123<PROTECTED>
18:56.02PTG123<PROTECTED>
18:56.04PTG123<PROTECTED>
18:56.05PTG123<PROTECTED>
18:56.08PTG123<PROTECTED>
18:56.10PTG123<PROTECTED>
18:56.12PTG123<PROTECTED>
18:56.14PTG123see its specifying my server ip as the name
18:56.16PTG123and it lets i go right in and dial
18:56.18file[airport]argh you silly person
18:56.50PTG123its whacky too
18:56.53PTG123it sets caller id to my ip
18:56.57PTG123er to the server ip
18:56.59PTG123when i call out
18:57.40puppetptg123: please dont spam channel..
18:57.45puppet~pastebin
18:57.47jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
18:59.21mrgobythat should be the topic
19:00.01mrgobypastebin ....   noahs of the irc-world ...  floods can go there
19:00.25EssobiUmm.
19:00.57EssobiI was just wondering.. Why is there no way so see the duration of a channels current status?
19:00.58ariel_PTG123, so if you try your settings I can use your server to make a call.. hummmm should I try?
19:01.13PTG123it should work on any asterisk server
19:01.23EssobiThat's kind of... smart to be able to get that from the AGI, Manager, or CLI
19:01.27PTG123i am more frustrated that the polycom isn't sending the proper username
19:02.23ariel_in most cases you can dial an extension@IP address and you have a good chance of getting it.
19:02.56PTG123but you don't understand, i can dial ANYWHERE in the world from my box
19:03.02PTG123without authenticating
19:03.06PTG123any phone number you want
19:04.31mrgobycan you dial santa clause at the north pole ?
19:05.06PTG123if i knew his number
19:07.25ariel_PTG123, yes your right. I just did it via your server.
19:08.27PTG123ariel: major problem isn't it
19:08.48ariel_yes it is in my view.  The dial string was so easy you would not belive it.
19:09.03PTG123ariel_: you want to submit it to mantis?
19:09.09*** join/#asterisk NYBLAZE10 (alex@ACA0FDF4.ipt.aol.com)
19:09.16*** join/#asterisk Dark40rce (~vince@dsl-17-106.cofs.net)
19:09.41ariel_I just did on my asterisk box.  exten => _1760.,1,Dial/sip/${EXTEN:4}@yourIPaddress.
19:10.29PTG123well i didn't have to do anything on my box to make it work
19:10.32PTG123just set username to ip
19:10.34PTG123then dial anything iw ant
19:10.39PTG123it goes through normal dial strings
19:11.10ariel_hummm I wonder if I can do this via someone else without logging in......
19:11.29*** join/#asterisk Tili (~Tili@202-133-65-63-dialup.sat.net.pk)
19:11.52PTG123yah you can i think
19:11.55PTG123try it on mine
19:12.00PTG12366.235.234.131
19:12.13PTG123but even if you had to be logged in
19:12.15PTG123still a major problem
19:12.20PTG123you could just get free calls
19:12.22PTG123and buy a $2 account
19:12.26PTG123on someones servers :)
19:12.41PTG123damn i should keep this bug secret, gives me ideas :)
19:13.54ariel_PTG123, make a guest account on your sip. and put it to a context=block and in the extensions.conf put [block] exten => X.,1,Congestion
19:14.15PTG123whats that gonna do?
19:14.46PTG123ok well i fixed my polycom so it sends the proper stuff
19:14.57ariel_It just might send these calls to that context instead of wide open server.
19:15.41PTG123well
19:15.47PTG123see if you can make a call through my server right now
19:16.05ariel_whats your exten?
19:16.37PTG123um
19:16.39PTG123call a phone number
19:16.43PTG123you don't need to know an extension
19:16.53*** join/#asterisk anthm (~anthmct@69.76.83.52)
19:16.53*** mode/#asterisk [+o anthm] by ChanServ
19:16.59ariel_I just made another ld call...
19:17.13PTG123yep
19:17.14PTG123it went through
19:17.19PTG123305 number?
19:17.23ariel_yes
19:17.26PTG123man bad bug
19:17.30PTG123ok trying your changes now
19:17.34PTG123now try
19:17.59PTG123bad bug
19:17.59PTG123:)
19:18.08jontowwhy do you think the makefile says to read SECURITY ;)
19:18.11ariel_PTG123, still worked.
19:18.20PTG123jontow: you got a fix for this?
19:18.31jontowmake sure your contexts are safe
19:18.38PTG123jontow: how would you do that?
19:18.41jontowand yeah.. put SIP traffic behind the firewall
19:18.55PTG123jontow: i need to allow people to use sip to my server
19:19.15jontowif nothing is accessible to [default] and your sip.conf says throw all traffic that isn't registered to specific phones in default or from-sip-restricted, or whatever the hell..
19:20.10Darwin35anyone done overhead paging with dsp
19:20.18PTG123ariel_: now try?
19:20.42ariel_ok it's blocked
19:20.53PTG123ok i bet this works on 99% of peoples servers
19:20.56PTG123heh
19:21.22PTG123free phone calls for everyone
19:21.27Darwin35ptg what you working on
19:21.56PTG123Darwin35: found a way to make free phone calls on everyones asterisk servers
19:22.00*** join/#asterisk Smythe (~Smythe@spock.cbcag.edu)
19:22.09Darwin35?
19:22.10ariel_PTG123, well at least there is a way to stop this.
19:22.21PTG123ariel_: true
19:22.23Darwin35I want the fix to block it
19:22.33Darwin35hehhe
19:22.34PTG123just set your default to context=block
19:22.38Darwin35just for safty
19:22.44PTG123and put a block context in extensions that dumps to congestion
19:22.57ariel_Darwin35, it's easy just have your default have only one line that says exten => X.,1,Congestion
19:23.02Smythedoes anyone have any expertise in tying an Asterisk box to a Mitel over a T1? (not PRI)
19:23.04*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
19:23.18Darwin35hh
19:23.21PTG123ok ariel wanna try it on another ip?
19:23.30Darwin35well back to overhead paging with dsp
19:23.54PTG123208.139.204.228
19:23.56PTG123try that one
19:24.30Darwin35need to be able to tie * into a pa system
19:25.16*** join/#asterisk nicholas_ (~nicholas@pD953A7A7.dip.t-dialin.net)
19:25.20nicholas_hi
19:26.28Smythemy Mitel SX2000 doesn't like calls initiated from the Asterisk box.  Any ideas?
19:27.16Ron-NaCan anybody help me with regex in ASTCC?
19:28.01modulus_how about just plain ol' regex?
19:28.01Ron-NaHow to setup a country and exclude some cities with different rates?
19:28.04modulus_i prefer POR
19:30.36*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
19:30.36Darwin35ok where is the console ansewr setup .. grr
19:33.22Darwin35grrrr
19:33.27Darwin35this should be easy
19:34.29_RaYmAn_PTG123, what was the actual problem? default context not being setup correctly? Or can it be done regardless of that?
19:36.44ariel_sorry PTG123 I was busy with my baby. did you try that ip address you just posted yet?
19:38.34*** join/#asterisk Gh0sty (~Ghosty@81.11.192.116)
19:44.41*** join/#asterisk mikegrb (~michael@thegrebs.com)
19:46.57*** part/#asterisk Smythe (~Smythe@spock.cbcag.edu)
19:50.09nicholas_are there any problems with authentication from kphone with asterisk? it seems generates the wrong response during digest authentication
19:51.22modulus_i prefer POR
19:52.50nicholas_linphone works
19:53.48jontow:)
19:54.26jontowi've yet to test a soft-phone under a platform other than windows
19:54.39jontowand i was told i need to have recommendations ;) sooo.. saturday at work compiling X.org
19:54.39jontowhehe
19:58.04nicholas_can i enable output why asterisk reject a SIP register? "sip debug" only shows what pakets are transmitted
19:58.59Darwin35why compile when you can install via pkgs
20:01.31jontowbecause i'm paid hourly
20:01.33rikstajontow: x-lite under wine
20:01.40jontow:P
20:01.45rikstai'm not joking
20:01.50rikstathats about the best
20:01.50jontowworks alright?
20:01.53rikstayep
20:01.53jontownice
20:02.03rikstathere's iaxcomm and a few others
20:02.05rikstathey are crap
20:02.11jontow:/
20:05.25*** join/#asterisk Gh0sty (~Ghosty@81.11.192.116)
20:06.08Darwin35hmm make a way for them to record and them play it back on th e pa hmm that might work
20:06.40*** join/#asterisk AhmedFouad (~xor@82.201.208.1)
20:06.46AhmedFouadhi all
20:06.52AhmedFouadi need an advice in something
20:07.17rikstadon't eat yellow snow
20:07.19AhmedFouadneed to connect 2 connect two seperate offoices overseas with asterrisk
20:07.38file[airport]lol I'm just sitting here and wireless APs keep appearing
20:07.54file[airport]originally at 12, now at 17... lol
20:07.57rikstafile[airport]: you wanna try living in student halls like me :)
20:08.17file[airport]oh well, I like my paid wifi
20:08.22file[airport]not NATTed
20:08.33rikstaya
20:08.52rikstacrazy how many unsecured APs there are tho
20:09.26file[airport]woot, fileserver at the airport! max out the bandwidth!
20:10.37*** join/#asterisk ionix (ionix@MTL-HSE-ppp184758.qc.sympatico.ca)
20:10.50ionixsup sup
20:11.12ionixAnyone has an idea on how to do prepaid for an asterisk<->asterisk solution ?
20:11.22*** join/#asterisk claint (~claint@195.174.25.120)
20:11.24ionixso that there might be multiple SIP/IAX2 channels at the same time
20:11.25file[airport]yes, it's called using astcc or writing code :)
20:11.28rikstawiki
20:11.35ionixastcc == calling card
20:11.40ionixI want to do SIP-SIP
20:11.43file[airport]astcc can be modified
20:11.44ionixor IAX2-IAX2
20:12.01ionixand this will allow multiple concurrent connections ?
20:12.06file[airport]sure, why not?
20:12.13ionixlast I check, it has multiple use prevention
20:12.21file[airport]you can modify it...
20:12.27file[airport]there is no package out there to do what you want, it's not that easy
20:12.30rikstait's silly questions night
20:12.33file[airport]you have to create or modify
20:12.40ionixyeh, I planned to create
20:12.40Qwell-- if (conn.count > 1) return;
20:12.41Qwell++
20:12.51rikstaill write it for you, for a fee :)
20:12.55ionixhowever, I was thinking about the best approach
20:13.10ionixlike, use asterisk manager, AGIs or wrote a custom module
20:14.08Darwin35http://pastebin.ca/7329
20:14.45wildcard0ionix, are you planning on gpl'ing it?
20:14.50ionixyeh
20:15.10ionixI hate private source code, this is why I won't use the NACT we have to do it
20:15.16wildcard0then i'll tell you how to modify asterisk/astcc to do it without changing huge amounts :)
20:15.24ionixhehe nice
20:15.33*** join/#asterisk Lee__ (~lee@69-203-206-248.nyc.rr.com)
20:15.46wildcard0lemme just pull up my notes then we can take this private so we don't annoy the rest of the channel
20:15.55ionixok
20:16.00rikstawhy not leave it public so we can chuck in our comments
20:16.16wildcard0sure.  i just didn't want to get too off topic with coding stuffs
20:16.33rikstai fail to see how that's off topic
20:16.35riksta:P
20:16.38wildcard0ok.  this is how i was planning on doing it:
20:16.40wildcard0riksta, :)
20:17.06wildcard0first, modify app_dial so that it stores the time for it's 'L' variable in shared memory
20:17.15wildcard0http://www.voip-info.org/wiki-Asterisk+cmd+Dial for reference
20:17.47wildcard0and give it an optional key also.
20:18.08wildcard0so that all the cc times can be stored in the same place
20:18.15file[airport]it's so very dead at this terminal
20:18.17file[airport]there's, like, nobody
20:18.36Qwellfile[airport]: San Jose?
20:18.41Darwin35http://pastebin.ca/7331
20:18.42file[airport]yes
20:18.46Qwellodd, for a Saturday
20:18.57QwellI'd bet LAX is extremely packed right now.
20:19.04zoaheyf ile
20:19.06zoahey file
20:19.06wildcard0then it's just a matter of modifying astcc to write to that shared memory location to compute it's times.
20:19.15file[airport]zoa: twisted and bkw hate you, they got what you had
20:19.19Qwellfile[airport]: See if you can't get a flight down to LAX or ONT. :p
20:19.24file[airport]<PROTECTED>
20:19.26zoafile how do you mean ?
20:19.31file[airport]zoa: they're sick
20:19.33zoathe bulgarian virus ?
20:19.36zoaHaha cool
20:19.37zoa:)
20:19.43QwellThey got sick from food zoa suggested?
20:19.49Qwellerm, nm
20:19.50zoai dont think it was infectuus though
20:19.51zoa:)
20:19.57file[airport]apparently same symptoms you had
20:20.06Qwellfile's next
20:20.07*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
20:20.09wildcard0it's not really that difficult.  i keep meaning to do it, but i haven't had the time yet
20:20.12zoathat will teach em from touching me all the time :p
20:20.16file[airport]:p
20:20.20ionixhmm astcc...
20:20.21ionix<PROTECTED>
20:20.21ionix<PROTECTED>
20:20.21ionix<PROTECTED>
20:20.21ionix<PROTECTED>
20:20.21ionix<PROTECTED>
20:20.23ionix<PROTECTED>
20:20.23file[airport]oh, c'mon... lemme on my fileserver
20:20.24file[airport]PALEEZ
20:20.25zoado they also have the horrible jetlag ?
20:20.28ionixso it means that it doesn't allow many calls
20:20.37file[airport]zoa: ha
20:20.43*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
20:20.51rikstaionix: dude, how hard can it  be to remove that functionailty.....i take it you are obviously not  much of a coder?
20:20.55wildcard0ionix, it's the 'inuse' variable
20:21.06*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
20:21.07wildcard0you can just set that to return 0 all the time
20:21.24wildcard0but if you do that, then calls can run over the max amount if there are concurrent calls at the end of the card's limit
20:21.41wildcard0the shared memory thingy fixes that correctly
20:22.01ionixk
20:22.02WhiteWlfwhat's the bitrate of ulaw?
20:22.06ionix64kbits
20:22.11WhiteWlfthanks :)
20:22.27ionixriksta: Problem is that I want to manage simultanious connections
20:22.34WhiteWlfGSM is ~13, correct?
20:22.35wildcard0ionix, otherwise it's just a matter of adding 'return;' as the first line in the checkinuse() function
20:22.40file[airport]frell frell frell
20:22.43ionixwildcard0: How it fixes that ??
20:22.56file[airport]this thing has such a crappy route to Canada
20:23.15ionixi.e: Will it have to adjust all timeouts on all connections for the account on each connect/disconnect
20:23.21file[airport]oh wait, someone is eating all the bandwidth
20:23.37wildcard0ionix, the shared memory thing will take care of that cause they'll all be reading from the same memory location
20:23.45_Vilerm -rf file
20:23.57file[airport]brb
20:23.57lesouvageI read on the digium site that the X100P is discontinued. What is the (productcode) of the new alternative?
20:24.25*** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com)
20:24.46Jer13261can someone help me in getting a channel var set in a dialplan that isnt working?
20:24.57ionixwildcard0: Hmm, let say I make a call to Canada for 1.1Ē/min. The app calculates 200 minutes timeout. Then an other call with the same account is made to Morocco at 10Ē/min. How will the shared memory update the Canadian call timeout ?
20:25.11jontowlesouvage; the alternative is a TDM400P card, iirc
20:25.24jontowor you can buy a cheap X100P clone on ebay, i hear..
20:25.39Jer13261yea like $7 each...
20:26.11wildcard0ionix, astcc will make that change in the db.  that's not difficult.  the hard part is changing the limits for the calls in progress
20:26.20*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
20:26.27ionixyeh excatly
20:26.30*** part/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com)
20:26.40wildcard0which is what the shared memory thing does
20:26.49*** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com)
20:26.52ionixI see
20:26.56wildcard0you need to modify app dial to do it cause it's passed in as a parameter
20:27.51zoado they also have the horrible jetlag ?
20:28.15*** join/#asterisk file[airport] (file@dhcp64-134-126-76.sjca.sjc.wayport.net)
20:28.40wildcard0ionix, basically you need to modify chan->whentohangup to be point to a shared memory location
20:28.46zoai gained 4 kg when i was at von
20:28.47wildcard0er -be
20:28.50zoadamn american food
20:29.12wildcard0heh funny.  i go down to the states and i lose weight
20:29.44ionixk, I'll try that
20:30.24wildcard0ionix, it'll take some mucking, but once it's set up there, you'll be able to modify that number directly from astcc
20:31.04wildcard0just make sure each card gets its own shared memory location for the time variable
20:31.33*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
20:31.51TheBearok solved my zap/2 problem the power cable had come out of the TDM card ???
20:32.18TheBearWhen I pick up the phone connected to the TDM card I get a dial tone, when I dial number it is not recognised, why ?
20:32.53*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
20:36.09*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
20:37.44lesouvagejontow:  a telephone line in and a network connection is all I'm looking for. What should be the benefits of spending $ 125 for a TDM400P, what are the extra's?
20:38.41zoadid anyone in here use app_icd lately ?
20:42.58eKo1lesouvage: Get an SPA-3000
20:43.18Jer13261is there any way to set a channel var without using an exten  line?...eg becuase the user hasnt dialed anything yet :)
20:44.00anthmsetvar => var=val in the friend def
20:45.03Jer13261setvar foo=bar
20:47.08Jer13261setvar isnt listed as a opt in sip.conf
20:47.58wildcard0globals?  kinda messy...
20:48.51Jer13261well i need to define a default area code on a per phone basis....its not even accepting me trying to set a global
20:49.35lesouvageeKo1: I 'm working out a plan to built a dedicated asterisk box based on the epia 5000 (mini) motherboard. I will need a kind of X100P card to connect the box to the world.
20:51.43jontowlesouvage; maybe you should buy a few of the ebay X100P clones then..
20:51.50jontowi bought 3 and have been experimenting slowly with 1 of them
20:51.51Darwin35ok it seems to work where is my speaker
20:51.58jontowmostly to test the netbsd drivers
20:52.33anthmline 197 of the cvs sip.conf.sample
20:52.40*** join/#asterisk marc32344 (~marc32344@65-39-197-107.dsl.teksavvy.com)
20:53.08*** join/#asterisk rowter (~Drake@201.135.26.29)
20:53.30rowteranyone has manage to work soyo G668 with *?
20:53.50file[airport]anthm: so, like, bkw is in the air
20:53.57rowterI have firmware version 1.42
20:53.58*** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net)
20:54.10zoaanthm, when do we finally get to meet ya ?
20:54.23anthmaug 3 when you come to cluecon
20:54.55file[airport]so, 1PM and this airport is not packed at all
20:55.28zoaanthm, what is the most recent version for app_icd and is it considered stable ?
20:56.00anthmprobably the on in it's cvs
20:56.12zoathats 10 months old ?
20:56.26zoacould that be ?
20:56.33zoaor am i looking at the wrong stuff
20:56.49anthmprobably
20:57.02anthmthe one on orson.callenish.com
20:57.39zoagot a full link ?
21:00.08*** join/#asterisk rvhi (~rv@66.175.65.89)
21:00.52*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
21:02.12anthmhttp://voip-info.org/wiki-ICD
21:02.24anthmthat'll be $50 google fee
21:02.27zoathnx i should have knowmn L)
21:02.30zoa:)
21:02.51*** join/#asterisk dev-null (~real@meitner.wh.Uni-Dortmund.DE)
21:03.28*** join/#asterisk darby_t (~tom@dnk250.neoplus.adsl.tpnet.pl)
21:03.29*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
21:06.14lesouvagejontow: I just did. There are plenty of x100p and clowns on ebay. It's my first ordered part of the asterisk-mini.
21:06.38*** part/#asterisk eKo1 (~bernd@63.245.57.70)
21:10.36*** join/#asterisk zippp (~zip@c66.190.109.98.ts46v-01.rckprt.tx.charter.com)
21:11.05*** join/#asterisk pryk (~tmalkut@fire2.orasoft.net.pl)
21:16.42Darwin35is there a list of all the functions in *
21:16.49Darwin35that can be mapped
21:17.20Juggiefunctions in the code?
21:17.24Juggieor dialplan functions
21:17.42Darwin35dialplan functions
21:18.00Juggieon the wiki
21:18.12Juggieuse google and search for dialplan applications
21:18.20Juggiethe google search on the page i mean
21:18.46*** join/#asterisk buskila (~buskila@CBL217-132-75-104.bb.netvision.net.il)
21:19.21*** join/#asterisk tessier (~treed@210.245.97.244)
21:21.17*** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net)
21:21.34*** join/#asterisk [spi] (~nothing@dD5E021D2.access.telenet.be)
21:21.39[spi]hi ppl
21:22.00*** join/#asterisk _Bradders (~Bradders@c220-237-64-249.fitzg1.qld.optusnet.com.au)
21:22.04_BraddersHeh all!
21:22.08Mochi
21:22.50[spi]I have the following error when trying to call a netmeeting client with sjphone in sip mode to test sip <-> h323: Spawn extension (default, #, 2) exited non-zero on 'SIP/mysjphone-e386'
21:22.50[spi]<PROTECTED>
21:22.50[spi]<PROTECTED>
21:22.50[spi]<PROTECTED>
21:22.50[spi]<PROTECTED>
21:22.52[spi]<PROTECTED>
21:23.34Moch323 is evil
21:23.46*** join/#asterisk zotz (~zotz@24.231.32.191)
21:24.23[spi]what are these security checks and how can I fix this?
21:24.40Darwin35ok I need to add *7-100 dialing
21:24.54Darwin35for each extension
21:25.01_BraddersJust doing a bit of a hunt around for a suitable solution to add SIP capabilities to GAIM for Linux
21:25.16_BraddersAnyone have any suggestions?
21:25.28Mocthat be cool
21:25.37Mocbut IAX in gaim might be better
21:25.47_BraddersThere are a couple of ports of linphone
21:25.48_Braddersphonegaim
21:25.49Mocespecially that GAIM was first made by Mark Spencer ;)
21:25.50_Braddersgaim-vv
21:25.54_BraddersIAX?
21:25.59Darwin35I wish yahoo would get voip in thier *nix client
21:26.11_BraddersMSN is _almost_ VoIP/SIP
21:26.27Mocwhinning again about BSD ? ;)
21:26.46Darwin35no
21:27.04Darwin35I have asterisk kicking ass on fbsd
21:27.37Lee__anyone have positive useage with asterisk on OpenBSD?
21:27.43Darwin35just adding feature/dialplan functions
21:27.45_BraddersOOh... IAX!
21:27.53Moc;)
21:27.58_BraddersWorks through NAT, no need for STUN!
21:28.03Mocoh yea
21:28.21Qwellcould probably borrow chan_iax, heh
21:28.30Mocwell there is a libiax I think
21:28.32Qwellas long as its GPL...which it probably will be
21:28.39Qwelldunno, that would work too :p
21:28.40_BraddersPerhaps I should be looking at AIX then!
21:28.45_BraddersIAX that is
21:28.47Qwelleww, not aix
21:28.47_Braddersfaux paus!
21:29.08Qwellyeah...if you release an iax plugin for gaim, I'll use it.
21:29.19Darwin35ok everyone over to Solaris
21:29.20MocI got samba + LDAP + Domain controler working like I want today
21:29.37Mocexcept for a minor Local Administrator issues
21:29.52_BraddersIs IAX supported on anything but Asterisk?
21:30.02file[airport]ya know what, I have yet to have been asked to turn on my laptop when going through security
21:30.10Mocyes, yat or something support it I think
21:30.15Darwin35use speex
21:30.20Mocfile, they dont anymore
21:30.20Darwin35or ilbc
21:30.41file[airport]I even snuck my Mac Mini past security
21:30.44Mocfile[airport], they use somekind of smell detector
21:30.44file[airport]they didn't ask me to take it out or anything
21:31.08_BraddersSo its really an Asterisk initiative?
21:31.16Mocbecause they know that doesnt mean the laptop power, it aint have nice C4 in the DVD Bay
21:31.23Qwell_Bradders: It stands for "inter asterisk exchange", so...
21:31.26_BraddersCan IAX go directly from client to client?
21:31.29_BraddersAhh...!
21:31.32_BraddersThat explains it
21:31.40Darwin35iax is best for trunking asterisk to asterisk
21:31.55_BraddersOk.. so I'm probably stuck with SIP
21:31.55Qwellbut it works great for clients too...NAT and all
21:31.58Mocyes trunking help alot, could double the number of channel with the same bandwidth
21:32.16Moc_Bradders, good luck ;)
21:32.20_BraddersI've gone to the trouble of writing STUN support for linphone, so I think I might stick with that :)
21:32.54_BraddersI am writing a wrapper for linphone, if It gets good I might write a IAX wrapper !
21:32.59[spi]I have a (Call ended due to security checks) error  which config files do I have to check for a sip to h323 client?
21:33.15Darwin35most *nix voip clients suck
21:33.19_BraddersWraps SIP in HTTP/TLS or STUN
21:33.45Darwin35asterisk is the best thing to come along
21:34.01_BraddersLinphone is kind of cool http://www.linphone.org/?lang=fr&rubrique=3
21:34.15*** join/#asterisk Stbjr[PuterShow] (~stboch@pcp0010759468pcs.howard01.md.comcast.net)
21:34.17_BraddersAsterisk is a full blown PABX, all I want is p2p messaging using voice :)
21:34.28_BraddersWe already have SIP comliant PABX's
21:36.49_BraddersJust need to the p2p client
21:38.07Lee__skype, but it isn't free software. some buddies of mines are raving about it.
21:39.07lindi-somebody should put together a free alternative indeed
21:40.32Qwellskype kinda sucks ;/
21:40.40Qwellproprietary protocols and all
21:40.56Lee__I didn't say my friends were above sucking  :)
21:41.04Qwellmost aren't
21:41.45Lee__I don't know anything about it except that they are using it to make international phone calls on the cheap and record the session with nothing more than a laptop.
21:42.08[spi]I have a (Call ended due to security checks) error  which config files do I have to check for a sip to h323 client?
21:44.16lindi-Qwell: point is that i have a feeling there already are free alternatives but nobody has put them together into something that an average windows user can install
21:44.55Lee__that assumes one cares about the average Windows user as a top priority
21:46.38QwellLee__++
21:46.52lindi-Lee__: i see your point but when it comes to communicating with people then it's quite important
21:47.00spackle~karma
21:47.00jbotspackle has neutral karma
21:47.30Lee__when it comes to making a commercial product it's important. communicating can be done for free for those who wish.
21:47.34*** join/#asterisk Mike (~mike@201.129.122.206)
21:48.00lindi-Lee__: er, those who have the technical skills :)
21:48.13Beirdoanyone here from the UK who can give my UK DID a ding so I can see how the callerID is formatted?
21:48.54Lee__wow. Icecast + asterisk!
21:49.09Lee__anyone here actually using that feature?
21:49.17QwellLee__: That exists?
21:49.21Qwelloh, for MoH?
21:49.30Lee__it's in the Debian package documentation
21:49.53Lee__<PROTECTED>
21:52.49rvhianyone use proftpd?
21:52.58rvhii created the account, but can't ftp into it
21:53.05rvhii can login
21:53.14rvhiso my phone can't get the config
21:53.15Lee__man proftpd
21:54.09rvhiproftpd config should be fine. it was running ok
21:54.14rvhii just added this new user
21:54.20rvhimy linux admin is on vacation
21:54.23rvhi:(
21:54.25*** join/#asterisk robbins (~robbins@adsl-068-209-107-007.sip.mia.bellsouth.net)
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21:56.17*** join/#asterisk kimc (~kimc@pcp04039944pcs.wbrmfd01.mi.comcast.net)
21:56.31robbinshey folks, i want to load only the modules i'll need for asterisk like an embedded system, is there a doc describing the modules and their associated functions?
21:57.12kimchello asterisk
21:58.34robbinsi've read asterisk slimming on voip-info, but it's pretty sparse
21:58.58kimcI'd be happy just get dialtone out of a digium port
22:00.00kimcI've got battery on the port but no asterisk dialtone..
22:04.37*** join/#asterisk l-fy (~diana@l-fy.developer.yate)
22:04.41l-fyihaaaaaaaaaaaa
22:05.23l-fybwahahahahaha
22:05.24*** part/#asterisk l-fy (~diana@l-fy.developer.yate)
22:12.52*** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net)
22:12.54Lee__so I think I have this right. I can set up an Asterisk server in my lab and have two computers with softphones connect to it just to test it's basic VoIP->VoIP capabilities, right?
22:13.13mrgobyi'm having dtmf detection problems using iax using nufone
22:13.18mrgobydoesnt work using ulaw or gsm
22:13.28file[airport]mrgoby: outbound or inbound?
22:13.34mrgobyout
22:13.42file[airport]it's not NuFone, it's your SIP phone
22:13.45mrgobyto my cell
22:14.02mrgobyit is iax and asterisk
22:14.05mrgobyso, i doubt it
22:14.20file[airport]outline how you're placing the call
22:14.31mrgobycall file
22:14.40mrgobyanswer
22:14.44mrgobywait
22:14.57mrgobyno extens can be dialed
22:15.09mrgobyaudio is fine
22:15.12file[airport]okay, well... uh... answer and wait won't accept DTMF...
22:15.20file[airport]you have to use background or waitexten...
22:15.20mrgobyecho works
22:15.23*** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net)
22:15.39mrgobybackground is what i'm using
22:15.44file[airport]see, you didn't say that
22:16.05file[airport]and the context is included in the one you're in so background can match against the extension?
22:16.14mrgobyi know,srry typing one handed :-)
22:16.31mrgobysame context,yes
22:16.43file[airport]then do an iax2 debug and see if you get packets for digits
22:16.50mrgobyok
22:19.56mrgobyis rfc2833 sip only ?
22:20.05file[airport]yes
22:20.22file[airport]IAX2 uses out of band dtmf all the time
22:20.28mrgobyok
22:20.37mrgobydebugging now
22:20.39mrgobyone sec
22:21.01file[airport]now how nice am I, debugging your problem while at the airport
22:22.08kimcfile: can you help me with a basic config?
22:23.25kimcTrying to get a port to deliver dialtone to a pots phone
22:25.12mrgobyhttp://pastebin.ca/7333
22:25.16mrgobyi appreciate
22:25.22mrgobyi know i should have a t exten
22:25.29mrgobythis is just for debugging though
22:26.08mrgobywhen i pressed the dialpad, no output from iax2debug
22:26.28mrgobyso, it appears it is not accepting it at all
22:26.35mrgobymy provider on this cell is sprint
22:27.03mrgobyit works fine if i dial the fxo line
22:27.24mrgobyfrom my cell
22:29.36mrgobyany ideas ?  i'm using switch-1.nufone.net....  just fyi
22:33.01mrgobyfile: http://pastebin.ca/7333
22:34.36*** part/#asterisk kimc (~kimc@pcp04039944pcs.wbrmfd01.mi.comcast.net)
22:34.39mrgobybrb
22:38.53*** join/#asterisk nwhit (nwhit@wsip-24-234-120-72.lv.lv.cox.net)
22:39.34nwhithello all... i have an easy (i hope) question
22:40.14nwhiti want to allow the incoming lines to dial an extension... there are about 60 sip phones connected... but i don't want to have to put them all in the dial plan
22:40.54nwhitsomething like: exten => _NXX,2,Dial(Sip/${EXTEN},20,m)
22:41.04*** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
22:41.08nwhitbut it needs to make sure that the extension that they enter is valid
22:41.13nwhitany way to check first?
22:42.06*** join/#asterisk jeffik (~jeffik@m8b7936d0.tmodns.net)
22:49.52*** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net)
22:51.16mrgobyi've got to get a new phone... crappy 2.4ghz phone wipes out my wireless everytime it rings... even though i have it set to like channel 32
22:51.48mrgobyso, anyone have any ideas about this nufone/iax/dtmf dealio ?
22:53.01mrgobydriving me nuts
22:53.19modulus_what's a voip?
22:53.29mrgoby~voip
22:53.30jbotextra, extra, read all about it, voip is Voice over IP
22:53.30*** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net)
22:55.53modulus_what's a IP?
22:56.01mrgoby~IP
22:56.02jbotmrgoby IP is a connectionless, best-effort packet switching protocol. It provides packet routing, fragmentation and re-assembly through the data link layer. [internet protocol]
22:56.28mrgoby~newbie
22:56.30jbotrumour has it, newbie is someone who is new to linux or debian, and should read the docs (/usr/share/doc/)
22:56.49nwhit~funnyguy
22:56.58mrgoby~justkidding
22:57.04nwhit~haha
22:57.05jbotheh
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22:57.08*** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz)
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22:57.36*** part/#asterisk Nebukadneza (~daddel9@i3ED6E179.versanet.de)
22:57.52modulus_who wants to see my calling card agi script?
22:57.58modulus_it's in perl
22:58.12mrgobypastebin dat mug
22:58.21Delmarsure modulus_
22:58.38modulus_http://pastebin.ca/7335
22:58.41modulus_tell me what you guys think
22:59.10Delmarbefore i look...
22:59.10nwhithey can someone help with a dial plan problem i am having
22:59.13Delmarwhat does it do :P ?
22:59.19modulus_calling card
22:59.29modulus_read it
22:59.57modulus_line 047 is my favorite line: if ($bal < 1.05){
23:00.24modulus_perl automagically knows a null value is less than "1.05"
23:01.12nwhiti want to allow the incoming lines to dial an extension... there are about 60 sip phones connected... but i don't want to have to put them all in the dial plan
23:01.17nwhitsomething like: exten => _NXX,2,Dial(Sip/${EXTEN},20,m)
23:01.34nwhitbut it needs to make sure that the extension that they enter is valid... what should i do to check first
23:01.56modulus_use exten => i,1,Playback(invalid)
23:02.04EssobiWASABI
23:02.05modulus_<PROTECTED>
23:02.43nwhitit just hangs up the line if, for example they enter 555 and Sip/555 doesn't exist
23:02.50mrgobyi dont know much about perl::AGI
23:03.01EssobiAGI ain't hard
23:03.10modulus_there's some fun regex in my code too:
23:03.11modulus_041   $bal = $dollars = $cents = $row[0];
23:03.11modulus_042   $dollars =~ s/(\d*)\.\d*/$1/;
23:03.12modulus_043   $cents =~ s/\d*\.(\d{2}).*/$1/;
23:03.12mrgobybut it would be helpful to make the provider a variable
23:03.21EssobiIt's actually pretty lame. :)
23:03.24mrgobyperhaps
23:03.31modulus_yeah asterisk::AGI is dumb
23:03.38mrgobywhy lame?
23:03.47modulus_it's just a module that does STDIN and STDOUT
23:03.51EssobiI wrote a CallerID/CMS lookup push screen in 20 minutes.
23:04.59EssobiHell.. it took me longer to get the XMLhttpget refresh code right then that AGI.
23:05.18modulus_that's b/c XML is brain-dead
23:05.27modulus_and so is html
23:05.31EssobiBut I tell you it's better then Cisco TCL.. Statefull/Callback TCL.. What a fricking joke.
23:05.54modulus_jbot tcl?
23:05.55jbotit has been said that tcl is at http://www.scriptics.com/ or in feed "http://handhelds.org/feeds/tcl"
23:06.18EssobiI mean really.. who the HELL thought it was a good idea to put a state engine in a halfass implemented TCL.
23:06.52*** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com)
23:06.52modulus_yeah hardware sucks in general
23:06.58EssobiThe felling blow.. I dled a sample script from the Cisco Engineering FTP site.. and the 2nd line in the program was a comment, stating.. "Does this work?"
23:07.41modulus_no perlers here want to comment on my code?
23:08.07EssobiWhat?  That regexp?  It makes me go blind looking at pattern matching positionals
23:08.40modulus_you should see my regex with pointers to arrays of associative array pointers
23:09.00modulus_only in the perl world
23:09.05EssobiFFS.. Just build multi-demenstional arrays.
23:09.12Essobiyou know what I meant.
23:09.21modulus_that's boring
23:09.28EssobiYea, and readable.
23:09.35modulus_and slower
23:09.50EssobiWTF are you running on? A 386?  who cares.
23:10.01nwhitany other suggestions?
23:10.02modulus_high volume load
23:10.03EssobiWOOOO 0.001seconds FASTER!
23:10.08nestAryour mom goes to college
23:10.16*** join/#asterisk jets (~jetsn@guardian.pmt.org)
23:10.22Essobihi vol is saved for C.
23:10.31modulus_not necessarily
23:11.27*** join/#asterisk cjk (~cjk@80.92.75.100)
23:11.41cjkhi, is there a limit of callgroups i can have?
23:11.51modulus_cjk, rtfs
23:11.56*** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net)
23:14.14cjkmodulus_, ok you see i know the word callgroup, so where to you guess did i found that. now think again. why might i have asked that question. and now you pretty cool rtfX-answering guy if you do not want to help me, than just do not answer. english is not everyone's native language.
23:14.16Essobihe Callgroup= setting defines call group for calls to this device.
23:14.18EssobiIn v1.0 and previous versions of Asterisk, call groups are numbered 0-31.
23:14.20EssobiIn v1.1dev and v1.2, call groups are numbered 0-63
23:14.35cjkEssobi, yeah thata what i read
23:15.03cjkso i can have 64 callgroups
23:15.56EssobiWow.. Call groups are kinda.... Retarded.
23:15.58EssobiHeh.
23:16.33modulus_cjk, thanks for your rather long response to my short, terse, and to-the-point response. i've always understood that it's just a suggestion. you're also correct about english, it's not my first nor native language.
23:16.39EssobiSo a call group is the inbound trunk of a pickup group?
23:17.42EssobiHey cjk .. WTF does engrish have to do with reading code?
23:17.52Essobi:)
23:18.21cjkEssobi, well i try to get the answer without reading the whole * sources, but thanks for the suggestion
23:19.20Essobiwho said you had to read the whole thing?
23:19.35EssobiI use grep everyday to find what I'm looking for and pull it apart.
23:19.38EssobiYou have to.
23:19.39modulus_jbot modular?
23:19.50EssobiA lot of code just isn't documented.
23:19.58Essobilike show channels concise
23:20.08modulus_essobi, it's not? *shock*
23:20.12nwhitfigured it out... using chanlisavail
23:20.13EssobiJust ain't documented.. ain't in the help CLI.
23:20.17modulus_essobi, like ALL of it isn't documented
23:20.25modulus_except stupid dial commands and functions
23:22.20modulus_damnit someone tell me how to write my perl code better
23:22.23*** join/#asterisk marc324 (~marc32344@65-39-197-107.dsl.teksavvy.com)
23:23.49Chujimodulus_ : Heh, don't ask me, I'm usually getting help from you
23:24.02Nugget`Sauron: http://nugget.livejournal.com/88629.html
23:24.04Chuji/j #perl
23:24.27modulus_yeah asterisk help
23:24.30modulus_not perl
23:25.15ChujiNuh uh, I don't need asterisk help
23:25.27ChujiYou've helped me with my lousy perl skillz
23:25.41modulus_i have?
23:25.42ChujiAsk anthm, he's a good perler
23:25.45modulus_*scratches head*
23:26.00modulus_you must be an ultra-ultra-noob then
23:26.07Chujihaha... yup
23:26.19modulus_'cause i'm pretty noobie at this perl stuff
23:26.45ChujiWell I'm getting better
23:27.06modulus_well i'm getting more obfiscated with my perl
23:27.09ChujiThanks to my safari subscription
23:27.43modulus_chuji, did you look at my calling card agi script?
23:27.57ChujiI don't think so, where's it posted?
23:28.29ChujiI made a calling card app too. Needed to have one that interfaced with our Intranet data in MSSQL
23:28.59*** join/#asterisk modulus_ (modulus@rm-f.net)
23:29.02modulus_damnit
23:29.04modulus_sorry
23:29.38modulus_http://pastebin.ca/7335
23:29.54ChujiDid you paste it to irc or something?
23:29.59Chujibooted cuz of flood
23:30.07modulus_pasted the wrong buffer
23:30.12ta[i]ntedmodulus_
23:30.18modulus_taintedulus_
23:30.19ta[i]ntedno money in calling card
23:30.24ta[i]nteduse astcc
23:30.35ta[i]ntedif u are that determined
23:30.43modulus_what's astcc?
23:30.51ta[i]ntedRTFS
23:30.53ta[i]ntedjk
23:30.58Chuji~astcc
23:30.59jboti heard astcc is the asterisk calling card platform.  There have been patches so that now you can use it in either a pre-pay or post-pay model.  You can find more information about it on the wiki (www.voip-info.org)
23:30.59modulus_jbot astcc?
23:31.00jboti guess astcc is the asterisk calling card platform.  There have been patches so that now you can use it in either a pre-pay or post-pay model.  You can find more information about it on the wiki (www.voip-info.org)
23:31.04ta[i]ntedjust giving u taste of own medicine
23:31.20modulus_astcc is too complex i've looked at this before
23:31.23ta[i]ntedastcc is all hooked up .. u just plug in values and go
23:31.29ta[i]ntedastcc is too complex?
23:31.37ta[i]ntedit's perl for christ's sake
23:32.15file[airport]this airport is interesting
23:32.28ta[i]ntedrtfc - it'll put hair on your chest
23:32.33ta[i]ntedfile[airport] which one are u at
23:32.42file[airport]San Jose
23:32.45file[airport]their wireless has a WINS server
23:32.52file[airport]so you can see all the other Windows boxes connected
23:32.56ta[i]ntedyea
23:33.00Essobiscarey
23:33.06ta[i]ntedi used to be able to get free net access at sjc
23:33.15modulus_tainted, compared to my script it's complex
23:33.26file[airport]it only cost me $6.95 for an unlimited day pass with Wayport
23:33.29file[airport]and that's full terminal coverage
23:33.31ta[i]ntedjust ssh'd into my own box
23:33.31Chujihow was von file?
23:33.38file[airport]loved it
23:33.40file[airport]miss everyone
23:33.50ta[i]ntedany highlights?
23:34.00file[airport]the Grandstream business phone is sweet
23:34.17ChujiI'm good for Von Boston, and Astricon, but I can't do anymore
23:34.21ta[i]ntedwhat about their other ATAs
23:34.48file[airport]I can do Astricon atlanta
23:35.00file[airport]ta[i]nted: it's just combos... but really sweet
23:35.04file[airport]like FXO/FXS combos, routers, etc
23:35.07ChujiAtlanta is pretty close for me. That is easy
23:35.16ta[i]ntednice
23:35.36file[airport]I have their flyer in my bag
23:36.04modulus_tainted, perl can get very complicated
23:36.22file[airport]it's fun to see all these people go through
23:36.34file[airport]hot guys and hot girls...
23:36.57ta[i]ntedhot girls at a voip convention?
23:37.04file[airport]yes there was
23:37.13file[airport]Tristan was nice, she was at the Switchvox booth
23:37.31hermiebooth babes or actual human beings?
23:37.39file[airport]actual human beings
23:37.45ta[i]ntedi hope a pic gallery pops up sometime
23:37.46file[airport]she designed the web interface/look of Switchvox
23:37.53file[airport]they went out with us a few times for dinner
23:38.04file[airport]Joshua and Tristan from Switchvox
23:38.11ta[i]ntedswitchvox has a nice website
23:38.13ta[i]ntedvery clean
23:38.33file[airport]http://www.desktopsummit.com/photos.php?category_id=19
23:38.34file[airport]Tristan is there
23:38.41modulus_astcc is borderline bloatware
23:38.46file[airport]Joshua is the guy with the glasses
23:39.22modulus_tainted, there are niches in credit cards still
23:40.01ta[i]ntedmodulus_ what do u mean credit cards?
23:40.08modulus_oops
23:40.11modulus_s/credit/calling/
23:40.17modulus_hahaha
23:41.02EssobiI need to get my boss to take me to VON next year.
23:41.03Essobi:|
23:41.37file[airport]it was a nice experience
23:41.39file[airport]lemme find pics
23:42.20file[airport]http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view
23:42.29file[airport]that was the Asterisk Pavilion!
23:43.10*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
23:43.21*** join/#asterisk onixx (1000@CPE0040f47145d1-CM000f9f7f2290.cpe.net.cable.rogers.com)
23:43.46onixxhi all, anyone else is getting "smsq.c:422: `POPT_ARGFLAG_SHOW_DEFAULT' undeclared " when compiling today's cvs ?
23:44.37EssobiMy boss said the pavilion was pretty tight
23:44.48file[airport]yeah, full of people
23:44.54EssobiSo what's the news on these new cards with built in DSPs?
23:45.00EssobiTight as in.. Nice.
23:45.19file[airport]what new cards with built in DSPs?
23:45.33file[airport]the only things we had was an echo canceller card for the TE410P and an IAXy version 2
23:45.49file[airport]which were both VERY cute
23:46.39file[airport]Essobi: I espically liked the carpet myself
23:47.30nwhithas anyone worked with cisco 7905g phones?
23:47.48EssobiIt was someone else's cards
23:47.55Essobi$3K a piece
23:48.50EssobiT1 with a hardware DSP
23:49.07Essobi410 with echo cancel ehh?
23:49.09EssobiMM.
23:49.24nwhiti am having trouble with the mwi on this cisco phone
23:49.27EssobiMaybe I can talk the boss into selling those 5400's after all. :)
23:50.21EssobiI just replaced two 5300's this week with an * box.
23:50.36file[airport]people in the US are odd
23:50.49EssobiLOL
23:50.57*** join/#asterisk tuxinator_linuxM (~tuxinator@m010e36d0.tmodns.net)
23:50.58EssobiNo shit.. We're all crazy.
23:51.12file[airport]tuxinator_linuxM: are you at the San Jose airport?
23:51.35tuxinator_linuxMfile[airport]: Ya
23:51.41tuxinator_linuxMwhre you at?
23:51.44file[airport]which terminal?
23:51.51file[airport]I'm in Terminal A
23:51.55tuxinator_linuxMC
23:51.57*** join/#asterisk bjohnson (~bjohnson@66.11.188.184)
23:51.58file[airport]between gates A8 and A9
23:52.02tuxinator_linuxMI have two hours to kill
23:52.15EssobiYou guys are sad.. Logoff and go have a beer.
23:52.16file[airport]I have, oh, 7
23:52.29EssobiOr seven. :)
23:52.37file[airport]haha
23:52.40tuxinator_linuxMman, I am at the expediat cafe, had laptop hookups
23:52.52file[airport]I just paid $6.95 for wifi access via Wayport
23:52.58file[airport]full terminal coverage ;)
23:53.07tuxinator_linuxMWayport is over here too
23:53.36file[airport]T-Mobile only covers a portion of here
23:53.48file[airport]the roaming is nice too, I've been through this entire place and haven't lost my connection
23:53.51EssobiPSssh... Log off and go have a beer together. Jeees.  High-ball, martini.. beer.
23:54.00file[airport]I can't drink here, or at home
23:54.02file[airport]I'm only 18 :p
23:54.06EssobiBaah.
23:54.07tuxinator_linuxMhe he
23:54.15file[airport]legal age here is 21
23:54.55tuxinator_linuxMDo I need to come over there file?  I just got comforatable.
23:55.07file[airport]you don't _need_ to
23:55.18tuxinator_linuxMyou have more time to kill, you come here
23:55.27file[airport]but I've already been through security once
23:55.40file[airport]once is enough for anyone
23:55.49tuxinator_linuxMdid they strip search you?
23:55.55file[airport]no
23:55.56file[airport]:p
23:56.14Essobifile[airport] WTF you from?
23:56.25file[airport]I really wonder sometimes if those things damage any equipment
23:56.27file[airport]Essobi: Canada
23:56.29file[airport]Atlantic Canada
23:56.55EssobiHah.. "Personal" equipment, or hardware?
23:56.57EssobiHehe.
23:57.02file[airport]both
23:57.11file[airport]:p
23:57.12Essobithey answer is both then
23:57.13EssobiHehe
23:57.36EssobiSo why was there no gear for the show?
23:57.43file[airport]it never arrived
23:57.46EssobiAhh.
23:58.00EssobiHell.. Shoulda hollered. I'd sent 10 phones with my boss.
23:58.07file[airport]oh we had phones and stuff
23:58.26file[airport]...yeah
23:58.30file[airport]and channel banks!
23:58.30tuxinator_linuxMfile[airport]: they won't let me trough security over there
23:58.42Essobihaha
23:58.43file[airport]tuxinator_linuxM: so we shall converse on IRC
23:58.48EssobiSHUTDOWN!
23:58.56file[airport]NO!
23:59.11Essobi"I don't know bob... This guy looks kinda shifty, with a loptop and all..."
23:59.17tuxinator_linuxMfile[airport]: So will be both be losers with our laptops
23:59.31file[airport]I need streaming TV on my laptop
23:59.46*** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com)
23:59.49tuxinator_linuxMI'm sure you could file a good porn movie

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