irclog2html for #asterisk on 20051009

00:00.16SarahEmmhihi
00:00.33dan__therro.
00:00.33[hC]I have two * boxes, one acting as the point for SIP connectivity, and another acting as my 'switch' - it has the PRI connected to it and acts as the master dial plan. when I place a call, and it comes back 'busy' from the PRI, and I execute Busy() - All it does is send back busy to the first * box (via IAX)
00:00.39*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
00:00.40[hC]at which point I have to call busy AGAIN from the first box
00:00.50[hC]is this normal, or should the switch actually play busy tones?
00:01.08[hC]I mean, both make sense for different reasons. :)
00:01.26[hC]Its no problem, I deal with it accordingly, Im just curious what other people do.
00:03.01[hC]PS. Who's stoked for astricon!?! :)
00:03.17SarahEmmheh
00:03.20SarahEmmi wish i could go :)
00:03.28SarahEmmand as for the former question, i'm not sure. i have a simple one-box * setup
00:03.55[hC]Yeah, most people do.. tis okay
00:04.04[hC]I debated going or not, and decided i should in the end
00:04.10KaBewMhey SarahEmm, fyi there seems to have been a regime change at sixTel. Their support is now superb
00:04.14[hC]Flying from florida though, made it a bit expensive.
00:04.34SarahEmmahh KaBewM
00:04.34dan__twhere's it being held?
00:04.38SarahEmmi'm happily with voctel :)
00:04.44SarahEmmsivana: speaking of which... callerid? ;)
00:05.03sivanaI need to ask some ppl.. can you email for a reminder? :)
00:05.07[hC]dan__t: anaheim, CA
00:05.16dan__tah.
00:05.16countmog_home: any luck?
00:05.22mog_homeone sec i got side tracked
00:05.23SarahEmmsivana: 'k. address
00:05.25SarahEmm?
00:07.09mog_homeset odbcstorage to the dsn name i think, and odbctable = name of table
00:07.12mog_homeif i am not mistaken
00:07.34mog_homethat will be tree fitty
00:07.35countok
00:07.39countlemme try that
00:07.42countI'll dcc you a beer if it works
00:07.45countscouts honor
00:07.46count:P
00:10.42Supaplexbeer.gif ;)
00:10.59Supaplexbeware of fools bearing gifs!
00:10.59mog_homeheh /me doesnt drink
00:11.04countbeer.can!
00:11.06countor bottle
00:11.49mog_homei have to run out for a bit
00:11.51mog_homehope it works
00:12.43countoh no!
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00:13.12countnope
00:13.17countit's still going to the file on disk
00:16.36NDTAnyone in here good with radius/gnugk/aquagatekeeper? Msg me..pay you for some phone time...
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00:37.43halogen8is it possible to use a Intel 537 based modem as a FXO card for Asterisk?
00:38.18KaBewMsure
00:38.19KaBewMwhy not
00:38.37Dr_Raybuy the dev kit
00:38.53Dr_Raythe help from digium is worth the $$$
00:39.26KaBewMhehe last time i gave digium money I got screwed, stupid iaxy
00:39.27Dr_Rayer, wait, I'm biased
00:39.37Dr_Raywhat happened?
00:39.42halogen8can you use any voice modem as an FXO port?
00:39.50Dr_Rayhalogen - no
00:40.23halogen8Dr_Ray: can you use any voice modems as FXO cards?
00:41.04KaBewMDr_Ray, i bought a iaxy. It didn't work right. I kept trying to get support from them and they told me to just mail the mailing list
00:41.08KaBewMwhich I had already done...
00:41.18Dr_Rayhalogen - no, just the intel chipsets
00:41.19KaBewMactually I didnt just buy 1 iaxy, I bought ten
00:41.27Dr_Rayand none of them worked?
00:42.02KaBewMthey ended up working ok, but it is severly lacking in features, security and everything
00:42.04SarahEmmjust a very specific intel chipset rather
00:42.07SarahEmma lot of intel chips don't work
00:42.21KaBewMbut the documentation stank, it didnt say to use a bootp server to configure the ip. I only had a dhcp server
00:42.31KaBewMit specifically said that dhcp would work fine
00:42.35halogen8SarahEmm: do you know where i can find a list of the intel chips that will work?
00:43.02halogen8SarahEmm: will these modems work good?  or would you just consider them for testing purposes only?
00:43.32SarahEmmMD3200 or Intel 837PU/PG
00:43.36SarahEmmany other chips won't work
00:43.39SarahEmm(EP, etc, don't work)
00:43.44SarahEmmhalogen8: i'm not the person to ask.
00:43.49SarahEmmi use asterisk for mostly TTY, not voice.
00:44.33KaBewMTTY?
00:44.51SarahEmmtext telephone.
00:44.57halogen8what about the Intel 537?  I'm looking around....and it looks as if it may work....just wondering if anyone has tried it
00:45.00KaBewMoh
00:45.02SarahEmmto type to others over the phone lines instead of voice
00:45.03halogen8http://www.voip-info.org/tiki-index.php?page=X100P+clone
00:45.28SarahEmmerr sorry halogen8, 537PU/PG not 837 :)
00:45.35SarahEmmthe PU or PG works, other ones don't
00:45.58SarahEmmKaBewM: * has a good amount of TTY functionality in it too :)
00:46.06SarahEmm(tho i wouldn't be surprised if i was the only one using it)
00:46.28Dr_Raytty as in services for the deaf, or tty as in teletype?
00:46.53KaBewMcool SarahEmm, never tried tty with it
00:47.06SarahEmmDr_Ray: the former
00:47.14KaBewMi use it as a replacement for  a home phone using my cable modem
00:47.17Dr_Rayvery cool
00:47.21SarahEmmtho i'm also into old computing gear ;)
00:47.27SarahEmmi was talking about the former
00:47.37Dr_RayI want a tty paper tape machine
00:48.11KaBewMi used an old trs-80 model 100 as a tty machine to get service faster from companies that would leave me on hold for hours
00:48.21SarahEmmDr_Ray: err.. as in the latter? :)
00:48.44SarahEmmKaBewM: ...
00:48.50Dr_Rayyes, as in the latter
00:48.59KaBewMthere was never any waiting for the deaf  . ..
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00:50.35SarahEmmKaBewM: want to bet?
00:50.55SarahEmmcertainly the opposite of my experience.. heh.
00:51.28KaBewMweird. wel this was in 91 or so
00:51.39KaBewMi've not had a model 100 in years
00:51.53Dr_Raythe model 101 was awesome
00:52.27KaBewMi had the original
00:52.39KaBewMbought it at a flea market for ten bucks
00:53.02Dr_RayA lot of reporters used them
00:53.02SarahEmmanyone know if you can turn off the dtmf sound when you dial on a polycom? replace it with just beeps or such?
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01:00.03JerJerLarge hammer
01:00.27SarahEmmlarge hammer? huh?
01:00.49lancey`busy:)
01:01.06JerJerthat would stop the dtmf sound, wouldn't it?
01:01.08SarahEmmlol
01:01.10SarahEmmyes it would
01:01.21lancey`busyhehehe :)
01:01.53SarahEmmnot quite what i had in mind tho :)
01:02.47*** join/#asterisk Derkommissar (n=Alberto@66.100.55.66)
01:11.16pauldywonder if you could make a bandpass filter to just cut out the audio for the frequencies used in dtmf
01:11.37SarahEmmuhh
01:11.41SarahEmmthat's a bit of overkill
01:11.45SarahEmmit was just 'cuz i prefer to not hear it
01:11.48SarahEmmnot because it's a big issue
01:12.34pauldyI know but it could be useful to sell for privacy concerns
01:12.45SarahEmmahh
01:12.48pauldyyou know the people who wear aluminum foil hats
01:13.30Dr_Raythey are not hats, they are liners
01:13.45Dr_Rayway to perpetuate a stereotype
01:13.46Dr_Rayer
01:14.39SarahEmmhehe
01:15.27pauldyok from now on I will phrase it as people with aluminum headwear
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01:42.01DerkommissarHas anyone attempted to run asterisk on OS X ?
01:43.23fileDerkommissar: sure, you can
01:47.44[dwC]how do i make a ZAP channel answer my phone line after 20 seconds rather than after 1-2 seconds like it is default
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01:48.57Derkommissarquestion is
01:49.05Derkommissarhow succesfull was it ?
01:49.07Derkommissarhehe
01:49.13Derkommissari just got an ibook
01:49.33Derkommissarim not sure if to leave OS X or to put debian on it
01:49.42[dwC]http://www.voip-info.org/tiki-index.php?page=MacOS+X
01:50.36DerkommissarCool
01:50.42[dwC]sorry this was the one i meant to paste
01:50.43[dwC]http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support
01:51.03DerkommissarCool
01:51.13DerkommissarMac OS X looks sounds and works cool
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01:51.44DerkommissarCool
01:51.48Derkommissarso otherr people done it
01:52.29[dwC]i heard of a couple of people running os x native on a PC with asterisk running 100%
01:52.32*** join/#asterisk maruz (n=maumar@host183-128.pool80104.interbusiness.it)
01:52.34[dwC]with pci support and all going
01:53.13Derkommissari though OS X was not released for x86
01:53.18[dwC]heh
01:53.21[dwC]not officially
01:53.30Derkommissarits supposed to be in 2006
01:53.45[dwC]http://osx86project.org/
01:55.17DerkommissarCool
01:55.46DerkommissarDint know that
01:56.07Derkommissari got to take my hat off, what people from mac have done with BSD its awsome
01:56.39[dwC]yeah well essentially when they move to intel x86 mac's will be PC's
01:56.54marc324what is the "./" in ./configure for?
01:57.15[dwC]you will be assimilated
01:57.20[dwC]resistance is futile
01:57.55DerkommissarLOL
01:58.10Derkommissar./ means execute the file
01:58.26Derkommissarwhen the file is not a /bin or /sbin directory
01:58.47Derkommissarso you compile a c program named doggy
01:58.54SarahEmmerr
01:58.59SarahEmm./ just means 'current directory'
01:59.14marc324configure <enter> returns command not found
01:59.22SarahEmm./configure
01:59.44Derkommissarsarah you would not be able to run an executable
01:59.47Derkommissarin a directory
01:59.52SarahEmmwhat?
02:00.02Derkommissarif the path is not defined or its on a /bin directory
02:00.09SarahEmmi know this
02:00.10Derkommissarso you got to xpecify it with ./
02:00.12SarahEmmso you have to specify the path
02:00.15SarahEmm... i'm aware ;)
02:00.16Derkommissarcorrect
02:00.19Derkommissarcorrect
02:00.37SarahEmmall i was saying is that ./ doesn't mean 'execute' it means 'current dir' :)
02:00.38SarahEmmmeow!
02:00.50Derkommissarif you add that directory to the /etc/profile
02:01.12Derkommissarit will work. by just typing it (Maybe) if it doesnt conflict with something else
02:01.35DerkommissarSarah sorry im not completely fluent in english
02:01.41marc324GOTIT
02:01.43QwellAre you saying to add . to /etc/profile?
02:01.45QwellI hope not, heh
02:01.52DerkommissarNo
02:01.54Qwellokay
02:01.58SarahEmmahh okay Derkommissar :)
02:02.01Qwell. in PATH is a massive security vuln
02:02.02Derkommissarthe directory where the configure script is at
02:02.02SarahEmmlol. that wouldn't be the best idea :)
02:02.12Derkommissarthat would brake your system
02:02.15Derkommissargood test
02:02.17Derkommissar:-/
02:02.21Derkommissarwhat would it do
02:02.35Derkommissaryou cant add . to your path
02:02.42Derkommissareven if you wanted to added
02:02.43QwellYou most certainly can
02:03.06Derkommissarit should be interesting
02:03.16Qwellits a big vulnerability
02:03.16Derkommissari will get back to you guys when i try it on my pc
02:03.23Derkommissarto see what happens
02:03.26SarahEmmsure you can... you just Never Ever Ever should
02:03.28SarahEmmit'll work fine Derkommissar
02:03.33SarahEmmbut it's a big security issue
02:03.41Qwellit won't "do anything", but if somebody drops a bad executable in /root/ or /var/tmp/ or whatever, it'll execute that first
02:03.42Derkommissarevery user
02:03.52SarahEmmheh.. oooooold slacks used to default to having . in path, iirc....
02:03.52Derkommissarwill be able to execute anything
02:03.56Qwellcp /home/Qwell/rootkit /root/ls
02:03.58QwellSarahEmm: heh, silly
02:04.02SarahEmmyeah
02:04.04Qwellanyhow, bbl
02:04.06DerkommissarLOL
02:04.06SarahEmmthis was in kernel 1.0 days or something
02:04.08SarahEmmttyl qwell
02:04.14DerkommissarSee ya'
02:04.53DerkommissarSarah sorry if my terminology is not apropiate
02:04.54Derkommissar:-)
02:05.16SarahEmmit's oay :)\
02:05.18SarahEmmokay even
02:05.28DerkommissarLOL
02:05.31Derkommissarsee ya later
02:05.54DerkommissarSarah, should i stay or should i go!!
02:06.05Derkommissari mean should i leave mac os x in my ibook
02:06.09Derkommissaror put debian on it ?
02:06.20SarahEmmdual boot
02:06.21SarahEmms'what i do
02:06.33Derkommissarinteresting
02:06.56Derkommissari have been able to compile execute and run all linux program and tools
02:07.05Derkommissarso lets see if its nesesarry
02:07.21Derkommissaranyways see ya la8ter
02:07.38NiveousCan some help me out? I'm trying to setup voicemail on pstn line.. I want it to be able to ring 20 seconds then got asterisk system then voicemail.
02:08.24GrubsI am trying to debug why rxfax does nothing on my system.  I'm using the |debug switch and also strace yet nothing is printed in the cli when rxfax executes and I cant find any log files to use to find out why the fax file is never created.  Any ideas?
02:10.27coppiceGrubs: when you use the debug option rxfax produces a lot of information in the main asterisk log
02:13.17Grubshi coppice.  I have checked  /var/log/asterisk/messages,  /var/log/asterisk/event_log and  there is nothing at all.
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02:14.35GrubsI am running in command line mode and can see the CLI and have verbose 6 and debugging 6  and no extra info is logged on-screen (same as messages log)
02:15.07coppiceif you see nothing you are not running rxfax at all
02:16.04Grubs-- Executing RxFAX("Zap/1-1", "/var/spool/asterisk/incoming/1234.tif|debug") in new stack
02:16.35coppicethen there will be output from rxfax in the log
02:17.02Grubsbut nothing further to say why the fax handshake fails (ie the sender never makes it to transmission stage)
02:17.33coppiceyou said you see nothing in the log. there will always be something if debug is on
02:17.45GrubsI have checked all thre log files that are in /var/log/asterisk and I  dont know what other log files I can check
02:18.08coppicelook at the console
02:19.36Grubs<PROTECTED>
02:19.53Grubsshould be /var/log/asterisk/messages
02:20.51Grubsfrom using strace  the next line after the  " -- Executing RxFAX(..." is "user hung up"
02:21.27GrubsI am wondering if the rxfax compiled into the debian package maybe doesnt understand the |debug switch and therefore does nothing.
02:22.32Grubs..I emailed the package maintainer to ask exactly that.
02:22.58coppicethe debug switch was added some time ago
02:23.14JamesDotComdid you strace -f?
02:25.24GrubsI did not do the "-f"   but am now
02:25.25coppiceYou should see something like this as the software starts, even if no communication occurs:
02:25.26coppice<PROTECTED>
02:25.28coppice<PROTECTED>
02:25.30coppice<PROTECTED>
02:25.31coppiceChanged from phase 0 to 1
02:26.28GrubsAll I see with verbosity 6 is:
02:26.46Grubs<PROTECTED>
02:26.47Grubs<PROTECTED>
02:26.47Grubs<PROTECTED>
02:26.47Grubs<PROTECTED>
02:26.47Grubspbx*CLI>
02:27.47coppicewhich version of spandsp are you using?
02:29.04GrubsThe debian package is "asterisk-app-fax (0.0.20050227-1)"  which uses libspandsp0 (0.0.2pre17-1)
02:31.31coppicethat is rather ancient. i think there should be something more up to date. the debian maintainer has been pretty keen on keeping things up to date
02:31.38Grubsthe only non-stadard thing I have done is to recompile Zaptel from the latest source as the Debian Stable did not include the newer TDM400P zap drivers.   May be this has broken the fax abilities?
02:33.20coppicethe debug option started with pre18 :-(
02:33.29Grubslol
02:34.17Grubslooks like I need to abandon debian stable and go for a more recent head build .... or try recompiling the whole caboose from the source myself.
02:34.45Grubscoppice - thankyou so much for your time.  I appreciate the willingness of this channel to help others.
02:36.10Grubsactually - pre17 *is* the unstable version.  my version is pre10-3  !!!!
02:36.28Grubslooks like I'll try a full source compilation.
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02:41.55marc324what is meant by cvs-head sources?
02:48.43Nugget"cvs head" refers to the state of the asterisk codebase as it exists this very moment in time.
02:49.02mog_homethe latest and greatest code
02:49.03Nuggetit's the development branch which reflects each coder commit to the source code as it occurs
02:49.12Nuggetit's the latest.  quite possibly not the greatest.
02:49.20websaeanyone here running ASTBILL?
02:49.21Nuggetoften it doesn't work at all.
02:49.27Nugget(well, sometimes)
02:50.42mog_homebah it very rarely "doesnt work" for me
02:51.22Nuggetthat may be, but it's still negligent to advocate cvs head for production systems, especially for end users who are unfamiliar with cvs and what that means.
02:53.06mog_homewell true
03:05.51Supaplexnesides, cvs gets plenty of head. same some for later ... ;)
03:07.08SarahEmm:P
03:17.21Dr_Rayin production even
03:17.27blitzrageevening y'all
03:17.42Dr_Rayblitz - in LA yet?
03:19.15blitzrageDr_Ray: yah
03:19.16blitzragedinner!
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03:30.39dan__twhat's up, kids.
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03:31.39dan__tEXCELLENT.
03:33.13Dr_RayI just ate my fortune from my cookie by mistake
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03:34.53dan__thaha.
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03:39.31dan__tbummer.
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03:53.48twistedmmmm
03:53.57twistedsinatra relaxes all the stress away
03:54.44SkramXanyplace that streams stuff like that fof free?
03:54.56twistedwww.shoutcast.com
03:55.01twistedpossibly
03:55.08twistedi have quite a few cd's tho
03:55.21NuggetiTunes has a lot of good streaming radio stations.
03:55.44twistedyeah
03:57.07zamslershoutcast is cool.
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03:57.48YaPhi
03:57.53zamslerhi
03:57.56twistedlo
03:58.04zamslertwisted, how have u been?
03:58.10zamslerbeen a while.
03:58.18twistedzamsler, heh... stressed
03:58.29twistedbut it's all fading away now ;)
03:58.34zamslerlol
03:58.37zamslermore beer.
03:58.39twistedno
03:58.49zamslerlol
03:58.54zamslerhave a sam on me.
03:59.02twistedis sam short for samantha?
03:59.06twisted:P
03:59.23zamslerlol
03:59.26zamslerSam Adams...
03:59.28zamslerbeer.
03:59.38zamslerbut a samantha would work..
03:59.48NuggetI like sam adams on tap ok, but in a bottle I'm not so fond of it.
03:59.52twistedi knew what you meant :P
04:00.44zamsler;)
04:00.59YaPi've a problem with asterisk, in my dialplan i send all incoming calls to a sip client and if nobody answer after the timeout the call is redirected to voicemail
04:01.17zamslerwow YaP sounds like a huge problem
04:01.22YaPif the sip client isn't connected the call goes to the voicemail immediatly
04:01.35zamslerof course.
04:01.38YaPi would like to wait, i tried to use Wait(30)
04:01.47zamslerthat doesn't sound like a problem.
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04:02.03zamslerwait for what? if no one is available, then it should goto voicemail.
04:02.04zamsler;)
04:02.07YaPbut now the call isn't redirected to voicemail
04:02.22YaPzamsler: wait for other phones not connected to asterisk
04:02.23zamslerwell if you would quit messing things up..
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04:03.06zamslerYaP, people like twisted have put a lot of time into asterisk. if you would RTFM and quit breaking things... it would make his time more worth wile.
04:03.24zamslersearch googole for macro dial.
04:03.27zamsler;)
04:03.38zamslererr asterisk macro dial
04:03.52YaPzamsler: i read the fine manual but i didn't found the answer
04:03.59YaPso maybe i didn't understand the manual
04:04.10YaPif you can give me an hint i'll appreciate
04:04.16zamslerfor real.. there are how many thousnads of asterisk servers up and running.. we don't have these problems..
04:04.35zamslerI know how to solve your solution..
04:04.40zamslerCisco Call Manager.
04:04.44YaPahahah
04:04.48zamslerthey have a nice web interface..
04:04.59zamsleryou would never be able to screw it up.
04:05.07YaPi like asterisk :)
04:05.11*** join/#asterisk nyquist (n=NyQuist@cp207-219-37-239.cp.telus.net)
04:05.22zamslerthen you should learn how to use it.
04:05.34Dr_Rayjesus, take a chill pill
04:05.35YaPi'm trying to learn
04:05.38zamslerlol
04:06.00YaPbut i don't understand this beaviour
04:06.08zamslerYaP, there is this really cool thing called astersik@home
04:06.12zamslertry that
04:06.22zamsler;)
04:06.23YaPi'm just looking for an hint to go ahead
04:06.29Dr_Rayvoip-info.org has a searchable wiki
04:06.33zamslerlol
04:06.46zamslerDr_Ray, ? = RTFM ??
04:06.49zamslerhahah
04:07.16Dr_Raywell, without being an asshole maybe?
04:07.34zamslerI am not a nice person. I am kinda like JerJer
04:07.43zamslerjust a bit quieter.
04:07.50Dr_Raywhy don't you try doing that now
04:08.43zamslerI am sure we would be able to look at it id you put it in pastebin YaP
04:08.44zamsler;)
04:08.51zamslerthat is normally how you get help
04:09.16zamslerbut if it is something stupid, you will get  > /dev/null
04:10.31YaPzamsler: It's just an extension with Dial, Wait, VoiceMail
04:10.51Dr_Raymaybe at the end of the Dial command?
04:10.53YaPVoiceMail is never executed if the dialed client isn't connected
04:11.13zamsleryap, if that is the case, there are 100 examples online for working exten-vm scripts.
04:11.40zamslerhave you searched google yet?
04:11.42zamsler;)
04:12.26Rowteranyone knows about the problem on wctdm (/lib/modules/2.6.8-2-686/extra/wctdm.ko) for debian 2.6?
04:13.18zamslerRowter, hmm.
04:13.32zamslerwhen was your cvs checkout?
04:13.41Rowterzamsler, yeah. zaptel
04:13.45zamslerI installed one last week and it worked.
04:14.05zamsler:-?
04:14.41Rowterzamsler, mmh, debian sarge 3.1
04:14.49zamsleryeah
04:14.58zamslerbut when was the zaptel checkout?
04:16.03zamslerkernel 2.6.12-1-686-smp worked last week
04:16.13RowterI just did it now..
04:16.21Rowterwell yesterday >)
04:16.30zamslerhmm
04:16.30Rowterlet me checkout again
04:16.40zamslerI am compiling now.
04:17.45RowterFATAL: Error inserting wctdm (/lib/modules/2.6.8-2-686/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg)
04:17.46websaeanyone here using ASTBILL?
04:17.51websaeim curious how it works out
04:18.41zamsleryup
04:18.43zamsler:(
04:19.05zamslerinsmod: error inserting '/lib/modules/2.6.12/misc/wctdm.ko': -1 Unknown symbol in module
04:19.11*** join/#asterisk NDT (n=me@cpe-24-195-219-245.nycap.res.rr.com)
04:19.14Rowterooh.. zamsler
04:19.23Rowterso now we both have the same problem
04:19.24zamslertwisted, thoughts?
04:19.26Rowterhehe
04:19.33zamsler:(
04:19.44Rowterlet's work it out..
04:20.14zamslerhmm
04:23.32twistedthoughts? from me?
04:23.40twistedi'm listening to sinatra whilst i clean my apartment
04:23.44zamslerlol
04:23.49zamslerI did that earlier.
04:23.49Rowterheh
04:24.23Rowtermmh, what should I do? use the  apt-get install zaptel-source?
04:24.47zamslerRowter, mostlikely you should submit a bug.
04:25.03zamslerbut I don;t know how to get any dumps for this.
04:25.19zamslerI am not a developer.
04:25.23Rowterok, but first I need to let this working hehe..
04:25.34Rowteras it was
04:25.36zamslerdo a cvs co from last week
04:26.33Rowterzamsler, could you tell me how to tell cvs to get last week?
04:26.49Rowterlet me google for that heh
04:27.03zamslerno.
04:27.05zamslerman cvs
04:27.07zamslerRTFM
04:27.09Rowterok :)
04:27.10zamslerDamnit
04:27.14zamslerLOL
04:27.52zamslercvs co -D 2005-10-02
04:27.53dan__towned.
04:27.54zamsler;)
04:27.56*** join/#asterisk |Vulutre| (n=Vulutre@211.119.205.68.cfl.res.rr.com)
04:28.08|Vulutre|Anyone here have a PRI in St. Louis, MO?
04:28.09Rowterhehe thanks
04:29.43zamslerthat doesn;t help either.
04:29.56Rowteryeah just notice that zamsler
04:29.56Rowtermmh
04:30.15zamsler;)
04:30.23Rowteron google says something about it for gentoo..
04:30.27zamslerlol
04:30.31zamslerno..
04:30.37*** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net)
04:30.37zamsleron your debian box.
04:30.40zamslerdo man cvs
04:30.40theblueHi all.
04:30.42zamsler;)
04:30.59zamslerRowter, you should use AAH also
04:31.00zamsler;)
04:31.02|Vulutre|hmm trying to find a ~ price on PRI Integrated
04:31.19zamsler|Vulutre|, who is the LEC there?
04:31.31zamslerSBC?
04:31.34|Vulutre|zamsler: I believe it is SBC
04:31.39zamslerouch.
04:31.41|Vulutre|yea
04:31.52zamslerif TDS is there then u can get ~$540
04:31.57|Vulutre|the one quote I like is at $640 for XO
04:32.00zamslerSBC wants ~$700
04:32.00|Vulutre|where is TDS?
04:32.01Rowterzamsler, why man cvs?
04:32.04theblueDoes anyone know of any free SIP or IAX based VoIP providers?
04:32.05Rowterahh?
04:32.18zamsler|Vulutre|, $640 is pretty good.
04:32.40zamslerhmm
04:32.42zamslernot really.
04:32.50zamsler$26/line.
04:32.59|Vulutre|zamsler: well its 1.44 Data/12lines
04:33.11|Vulutre|and it pulls from the data when a line is in use.. you know the integrated
04:33.20zamsleryeha
04:33.24zamslerI have it from TDS
04:33.30|Vulutre|and 100,000 LD mins... but its XO so its already VoIP
04:33.33zamslerbut I pay $179
04:33.37|Vulutre|JESUS
04:33.40|Vulutre|$1279
04:33.40zamslerlol
04:33.41|Vulutre|urg
04:33.43|Vulutre|179
04:33.43zamslerno LS
04:33.47zamslerer LD
04:33.56|Vulutre|unlimited local?
04:33.56zamslerI only take incoming calls on my PRI.
04:34.00|Vulutre|ah
04:34.04zamslerAll outgoing is VOIP.
04:34.10zamslerit is cheaper.l
04:34.17zamslernothing is local here.
04:34.18|Vulutre|who do you use for VoIP?
04:34.24zamslerthey charge by the mile.
04:34.31zamslerI have many providers.
04:34.31dan__tOk, even though we already have a championship, the F1 GP of Japan is on in 30 mins
04:34.32dan__tFYI
04:34.33dan__theh
04:34.39dan__thave a champion, rather.
04:34.45dan__tI need mroe beer, sorry.
04:34.50|Vulutre|zamsler: Ive used a few and I always get complaints cause the providers become sucky some days
04:34.52zamsler|Vulutre|, sixtel, voxee, voipjet, teliax, nufone
04:34.55|Vulutre|tried VPC
04:34.57Rowterwctdm: disagrees about version of symbol zt_receive
04:34.57Rowterwctdm: Unknown symbol zt_receive
04:35.00|Vulutre|Ill try some of those
04:35.05|Vulutre|asterlink is good
04:35.21zamslerI am ~5ms from nufone and sixtel
04:35.25zamslerthey work the best.
04:35.34zamslerI use voipjet for my canada calls
04:35.47zamslerand sixtel, voxee handles my bulk.
04:36.02zamslerI use nufone and teliax if needed.
04:36.28zamslerI am using AAH on debian, and I use the rollover feature so all my calls go out.
04:37.02zamslerI normally use ~5000 minutes a month outgoing.
04:39.19|Vulutre|zamsler: yea we use ~10k per office
04:39.28zamslerdamn.
04:39.33zamslerthat's cool.
04:39.39|Vulutre|still not bad
04:39.44|Vulutre|but CA raped me
04:39.48zamslerI hardly have issues terminating calls.
04:40.03|Vulutre|I want to get one or 2 of the XO unlimited LD lines
04:40.12zamsleryeah
04:40.15|Vulutre|then I can use them to throw all my LD over using DUNDi
04:40.20zamslerTDS will do that also.
04:40.31*** join/#asterisk Eight (n=blake@12-227-171-175.client.mchsi.com)
04:40.34|Vulutre|too bad TDS doesn't seem to be in any of my markets
04:40.41zamsleruhuh
04:40.48|Vulutre|CA, AZ, FL, SC, MO
04:40.59zamslerthey are giving good prices to piss SBC off.
04:41.00zamslerLOOL
04:41.12|Vulutre|need PRIs in St. Louis, Miami, and Phoenix
04:41.14zamslerin IL they own fiber to ever SBC CO in my area
04:41.26|Vulutre|damn
04:41.35zamslerSBC wants $800 for the same service.
04:41.42zamslerand I get special SBC discounts.
04:41.43zamslerLOL
04:41.46|Vulutre|damn
04:41.47|Vulutre|again
04:41.49zamsleruhuh
04:42.00zamslerI can get  a DS3 for 3000/mon.
04:42.05|Vulutre|I think $640 for the XO integrated isn't bad just want to make sure
04:42.07|Vulutre|WHAT?!!?!
04:42.10zamslerbut have to pay out the ear for a phone line.
04:42.13zamsler;)
04:42.24|Vulutre|Bellsouth introducted 45mbit LAN lines down here for $6k
04:42.28|Vulutre|I thought that was good
04:42.30zamslerI get bandwidth for 42.50/Mbit
04:42.44|Vulutre|zamsler: where the hell are you?
04:42.53|Vulutre|springfield?
04:42.58zamslerLibertyville.
04:43.01zamslerlol
04:43.17zamslerI go to the phone companies differently.
04:43.24zamslerI tell them whT I will pay.
04:43.31zamslerthey can choose to take my business.
04:43.48zamslerIf they do not, I use last mile to find what I need at my prices.
04:44.05|Vulutre|whats last mile?
04:44.08zamslersbc passed on my first deal.
04:44.46zamslerafter i forwarded my prices to them... they gave me a DS3 for 1200/mo, but screwed me on everything else.
04:45.02zamslerlastmile is a telco broker company.
04:45.08zamslerthey work if u know how to use them
04:45.13zamslerelse u will get screwed.
04:45.17|Vulutre|ah
04:45.29|Vulutre|I was using some random place I found in yellowpages in MO
04:45.47|Vulutre|Ive used Broadwing (terrible after support) and Xspedius
04:45.52zamslerI send them my requirements and a PO and they auction off my service.
04:46.14zamslerI have an awsome guy that does bandwidth.
04:46.16wunderkini just had a run-in with broadwing..
04:46.22zamslerhe is out of Cali.
04:46.31|Vulutre|wunderkin: how did it go?
04:46.37|Vulutre|wunderkin: don't tell me it was in CA
04:46.38enderhrm, We're using broadwing for our bandwidth too
04:46.58wunderkinit took a couple days to fix my pri.. they set it to SF instead of ESF and they kept blaming me.. no its in phx
04:47.08zamslerlol
04:47.13|Vulutre|took them 2 months to port in a DID
04:47.15wunderkinbut its for ld, i think their switch they said is in hayward ca
04:47.19zamslerdamn
04:47.19|Vulutre|one that was ON the order
04:47.29|Vulutre|didn't get ported in turnover
04:47.34|Vulutre|so I got a free month
04:47.37twistedomfg
04:47.42twistedthis desk is made of wood!
04:47.46|Vulutre|hahahaha
04:47.48zamslerlol
04:47.51|Vulutre|random
04:47.53zamslertwisted, ok?
04:48.04endertwisted: as opposed to what?  cardboard?
04:48.07twistedzamsler, yeah, just cleaning
04:48.08zamslerlol
04:48.11zamsleroh ok.,
04:48.30zamslerkinda like I find that my carpet is tan when I clean..
04:48.35zamsler;)
04:48.40twistedlol
04:48.40twistedyeha
04:48.44zamslerI have 19 computers in my apt.
04:48.51twistedlol
04:48.53|Vulutre|zamsler: and your power bill is...
04:48.54twistedi don't have THAT many
04:48.54zamslerIt is kinda hard to clean around them
04:49.01zamsler|Vulutre|, paid for
04:49.03zamsler;)
04:49.06enderyikes.
04:49.07|Vulutre|lol
04:49.07Dr_Ray19 computers in a studio apartment
04:49.08enderthats a lot
04:49.15twistedi have like..  6
04:49.21|Vulutre|PC+Laptop+Test Server
04:49.24twisted7 if you count the xbox
04:49.24zamslerlol
04:50.16zamsleri got a new remote today. it is awesome..
04:50.28zamslerlogitech Harmony 520
04:50.39zamsler$90 at wal-mart
04:51.01zamslerevery geek needs one.
04:51.06|Vulutre|zamsler: I got the other one the 800 or what
04:51.07|Vulutre|rox
04:51.18marc324what the download command in ftp?
04:51.22Dr_Rayget
04:51.28Qwellget
04:51.33Qwellerm, yeah
04:51.34Dr_Raybinary
04:51.35zamslerlol
04:51.42zamslerdamn.
04:51.46dan__theh
04:51.52zamslerthis is getting crazy
04:52.01zamslerhow can u be on IRC and not know how to ftp?
04:52.03zamslerFFS
04:52.06zamslerRTFM
04:52.12Qwellyeah, really...
04:52.17zamslerdamn kids
04:52.25zamslermirc kiddies
04:52.57zamslerI can see needing help compiling a kernel in debian.
04:52.59zamslerbut FTP?
04:53.03zamsler:!@#$!#
04:53.12*** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net)
04:53.18Qwellzamsler: newb :p
04:53.40Qwelloh, I read that as "I need help"
04:53.41zamsleryeah, we were all there at one time.. but we knew how to ftp..
04:55.33Qwellanthm: you know it
04:55.44zamslerlol
04:56.21enderthere are people that don't realize that the 'irc' of 'mirc' actually means something.
04:56.41Qwellender: most users of mirc
04:56.41enderthink think that 'mirc' _is_ the chat system.
04:57.23zamsler<PROTECTED>
04:58.37konfuzed;^?
04:58.54Qwellplease, the unzip command would have seg faulted :p
04:59.02Nivexender: I get that at work all the time.  We have a Jabber server and the preferred client is Gaim.  People come up to me all the time and tell me "I can't log in to Gaim."
04:59.02zamslerhahahaha
04:59.14zamslerI use gaim
04:59.16marc324150 Opening BINARY mode data connection for postgresql-8.0.4.tar.gz (13929175 bytes). it freezed
04:59.17zamslerlol
04:59.21zamslerbut not for IRC
04:59.31zamslerI used to use BitchX
04:59.31QwellNivex: Thats somewhat excusable
04:59.33zamslerLOL
04:59.39Nivexoh, well, the people who use Gaim for IRC are a whole 'nother crowd.
04:59.45zamsleruhuh
04:59.54NivexQwell: I usually let it slide since I know what they're talking about.
05:00.17NivexIt's the ones who tell me "My Internet is broken." that I want to thwap.
05:00.22QwellWhat bugs me is when people at my work call AIM "IM"
05:00.31Qwell"Are you signed onto IM?"
05:00.40endernod
05:00.47zamslerI am signed into 6 IMs
05:00.53NDTNo...it's the people that call a computer a piece of crap when they did something stupid to the OS that I want to thwap
05:00.54zamsleror 6 accounts
05:00.55enderWhich one?  I have 4....
05:01.00zamslerbut 1 IM program
05:01.14SkramXPeople at my school think Skype is the only VoIP platform. I'd like to slap them.
05:01.14zamslerPEBKAC
05:01.18SkramX:D
05:01.35zamslerID10T errors
05:01.41|Vulutre|SkramX: but it is
05:02.05zamslerI wish skype would allow 3rd party connectionsl
05:03.06|Vulutre|yea it would be nice to bridge that crowd.. I believe thats a bounty
05:03.49zamsleruhuh
05:03.51zamslerY@wn
05:03.56zamslerbedtime
05:04.15zamslernight all
05:04.20konfuzedwell cant you get asterisk to call up a skype account?
05:04.38konfuzedi never considered it before but ehy not?
05:05.55konfuzedif so then  you could set up like 4 skype accounts and then make a skype gateway ? mm maybe?
05:07.15Qwellkonfuzed: Asterisk doesn't support the skype protocol
05:07.28tzangerkonfuzed: many varied reasons, the biggest being that there is no protocol driver for skype for asterisk
05:07.44tzangerkonfuzed: also, the codec that skype uses isn't available on asterisk
05:07.57Qwellwhat codec?
05:08.08tzangerkonfuzed: also, there are some who take issue with the closed-source nature of skype
05:08.13tzangerQwell: wideband ILBC
05:12.16coppiceI think they call it something else now, like iSBC
05:18.55*** join/#asterisk loud (n=ariel@cypher.punk.net)
05:19.22QwellForecast for Astricon - sunny and hot
05:21.36Qwellhmm, 2.5 days away
05:26.13*** join/#asterisk Horshack (n=cdh@cpe-69-203-25-182.nyc.res.rr.com)
05:36.30SplasPoodAnyone here mind helping me with a Cisco ATA186 dialplan?
05:40.38*** join/#asterisk asterisk99 (n=chatzill@modemcable169.194-130-66.mc.videotron.ca)
05:41.18asterisk99anyone kow how to get *zaptel* to load on boot on a *Gentoo* system??
05:41.42*** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net)
05:44.28*** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net)
05:49.09konfuzedcant skype be used to call a SIP account via sipuser@Aserver.com ?
05:49.28konfuzedi can be a little slow some times
05:49.30konfuzed;^)
05:52.18tzangernot that I'm aware of, no
06:01.42konfuzedhmmm
06:01.48konfuzedhas anyone seen this http://www.jajah.com/en/features_protocols.asp
06:02.19konfuzedrather interesting part of the mix
06:05.34konfuzedhttp://www.jajah.com/en/features_codecs.asp is quite a list of support codecs includign speex and iax of course
06:08.11*** join/#asterisk ManxPower (n=eric@1Cust3844.an8.dfw28.da.uu.net)
06:08.31ManxPower~docs
06:08.33jbotdocs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
06:10.38*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
06:17.48*** join/#asterisk nnnnnn (n=killfill@pc-200-74-17-222.nunoa2.pc.metropolis-inter.com)
06:18.13killfillhi
06:21.12konfuzedok so I read some about the skype problems too
06:22.22konfuzedof course that always makes things a little Konphuzedless
06:24.47*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
06:24.58_Thorhello everyone
06:25.32_Thoranyone has worked with PrepaidCall (calling card add-on)?
06:25.57konfuzedhello * 295 back at ya
06:26.29_Thorhello konfuzed
06:26.56konfuzedhi
06:27.15_Thorhave you worked with prepaidcall?
06:27.23konfuzedmmmmmm no
06:27.31_Thorknow what it is?
06:27.40konfuzedyes
06:27.49Rowteranyone has seen this error fsk_serie made mylen < 0 (-23)
06:28.03RowterOct  9 02:27:14 ERROR[26392]: callerid.c:274 callerid_feed: fsk_serie made mylen < 0 (-23)
06:28.03RowterOct  9 02:27:14 WARNING[26392]: chan_zap.c:6031 ss_thread: CallerID feed failed: Success
06:28.10Rowteram not getting the callerid.
06:28.43JerJerplay with fxotune
06:28.45_ThorI donĀ“t know why in the world it ocurred to me to use it
06:29.09konfuzedjust for calling cards really
06:29.21konfuzedof go and creaatively utilize anything you want to
06:29.23konfuzed:^)
06:29.29_Thordo you know of any other solution for prepaid calls?
06:29.49konfuzedin context of calling cards
06:29.56_Thornot really
06:30.09konfuzedor prepaid minutes for extension use
06:30.15_Thorright
06:30.34konfuzedim sorta bolting one together
06:30.41Icemaannis there anyway to have asterisk detect certain DTMF tones during Dial()? If I have dialed someone, and we have talked for a while, and then I press *78, I want it to bring MOH into the channel (both sides would hear it).
06:30.42_Thorright, prepaid minutes per extension used, based on an existing balance
06:30.43*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
06:31.13_Thorwhat direction are you going to?
06:31.25Icemaannit actually wouldnt be MOH, but playback for both parties to hear. MOH was a bad example lol.
06:31.37konfuzedA Hybrid of Open Source Technologies
06:31.38konfuzedoooooooooooooo
06:31.57_Thorprepaidcall is open source
06:32.14_Thorwhy you didnĀ“t chose it?
06:32.23konfuzedconceptually
06:33.04konfuzedi have more accounting and billing to deal with besides asterisk
06:33.31dan__tAh
06:33.33dan__tdamn good race.
06:33.35dan__twith a last-lap takeover.
06:33.43_Thorumm
06:33.56konfuzedtheres also a big difference between billing and accounting
06:34.53_Thorbecause billing is on-line, and accounting is after the fact?
06:35.34konfuzedso it could be said that this series of systems talks to asterisk and other apps and sucks in reports from those apps too
06:36.44_ThorI got a couple of friends working in projects like that.   The idea is to integrate billing from different sources
06:37.23konfuzedand you cant do with out automated account activation (provisioning) payment (integrated ecomerce) and CRM Management (Collections/Suspensions)
06:37.43_ThorThe idea is to use Oracle for that
06:37.52konfuzedwell thats one idea
06:38.21konfuzedoracle is awesome but like database bloat ware
06:38.35konfuzedkeep it simple
06:38.47konfuzedits much less expensive
06:38.54_Thorif you ever need a nice little call accounting app for the client side, let me know
06:39.20konfuzedas in web page display of it or a reporting engine?
06:40.20konfuzedmaybe tool tray utility to monitor sip headers for call time ? hmmmmmmm
06:41.18_Thorwell, I wrote the first version 10 years ago... and to this date I believe a running exec performs/displays better than the current tendency to use web page displays
06:41.29*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
06:41.46konfuzedwell thats always been true
06:42.28_Thoranyway K, let me get back to my suffer some more
06:43.04konfuzedim curious to see the call acct app
06:43.18_Thoroh, by all means
06:43.37_Thorcontact me at c.savinovich@itntelecom.com
06:45.55konfuzedmmmmmmmm    interesting site
06:46.40_Thorthink so? thank you, it is all technology I was working on 5 years ago
06:46.57_Thorthe new web site will come out one day ;)
06:47.26konfuzedit has that hybrid technology appeal
06:47.44CoaxDany of you people total OpenVPN gurus?
06:48.16_Thorreally?, let me look at it one more time, <g>
06:49.07_ThorI was working on web-2-phone technoloy way before Mark Spencer finished high school <g> <g>
06:49.27konfuzedPDAs WEBCALL VoIP  ip-PBX
06:58.50*** join/#asterisk websae (i=websae@207-118-134-96.dyn.centurytel.net)
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07:05.40killfillhey, how do i make sure asterix can dial with my x100p clone card?
07:07.29JerJerdon't buy a clone card
07:13.09killfillhm..
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07:19.45*** part/#asterisk ^n0b0dy^ (i=KILLME@200-70-145-170.mrse.com.ar)
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07:27.42*** join/#asterisk leputois (n=alexis@toronto-HSE-ppp4326258.sympatico.ca)
07:28.19leputoisi just setup a new server ... everything is working fine except one thing ... no sound
07:28.52leputoisanyone have an idea ??
07:29.32alphaqueno sound from where ?
07:30.31djin_ibZaptel modules are loaded without problem by hand, but not at reboot. What does 'chkconfig -add zaptel' normally do? There is a /etc/rc.d/init.d/zaptel, but no /etc/rc.d/rc2.d/S09zaptel.
07:30.44leputoisanywhere ... if i put in my dialplan to play a sound i will hear it but if i make a call i cannot hear the other person
07:31.00alphaqueleputois: elaborate, show dialplan
07:31.15alphaquedjin_ib: i'm not a linux dude, so cant help u with that
07:31.24djin_ibleputois, does the other one hear you?
07:31.40djin_ibalphaque, np. thanks.
07:32.17djin_ibleputois, is the server behind NAT?
07:33.11*** join/#asterisk Nexis (n=nexis@12-219-60-252.client.mchsi.com)
07:33.29djin_ibSo, you connected directly and the other from outside (through NAT)?
07:35.10djin_ibMmm, leputois didn't quite understand when I asked him not to msg me :)
07:35.23alphaquedjin_ib: he msged me too
07:35.25*** join/#asterisk leputois (n=alexis@toronto-HSE-ppp4326258.sympatico.ca)
07:35.31djin_ibOk, there he is.
07:35.37leputoissorry
07:36.19djin_ibThe problem is probably in your NAT config. Make UDP ports 10000-20000 available for the outside.
07:36.39leputoisok i will try this
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07:37.19leputoisi will lost connection since i need to rebbot router for this brb
07:37.27leputoisreboot **
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07:43.08leputois_Still nothing
07:43.49djin_ibI'm pretty sure it is. Try it with two phones in the same lan.
07:56.24*** join/#asterisk _leputois (n=alexis@toronto-HSE-ppp4325288.sympatico.ca)
07:57.10_leputoisok i just pu my server in DMZ so if it was a port problem it should work and still no sound
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08:15.56NDTOf topic here but anyone good with gnugk?
08:20.43mepplguten morgen
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08:28.05Rowteranyone has the callerid problem with tdm400?
08:28.06RowterOct  9 04:21:51 NOTICE[29604]: chan_zap.c:6001 ss_thread: Got event 18 (Ring Begin)...
08:28.06RowterOct  9 04:21:51 ERROR[29604]: callerid.c:274 callerid_feed: fsk_serie made mylen < 0 (-15)
08:28.06RowterOct  9 04:21:51 WARNING[29604]: chan_zap.c:6031 ss_thread: CallerID feed failed: Success
08:28.07RowterOct  9 04:21:51 WARNING[29604]: chan_zap.c:6075 ss_thread: CallerID returned with error on channel 'Zap/1-1'
08:30.23JerJerFXOTUNE
08:30.24JerJeruse it
08:30.35JerJeri told you this the last time you flooded the channel with that crap
08:30.50brc_helo
08:30.57JerJermoo
08:31.00brc_woof
08:34.49RowterJerJer, sorry
08:35.02RowterJerJer, but I have no echo problem, think that will help?
08:35.16alphaquejerjer, is there a test suite for chan_h323.so ?
08:35.29alphaquejust to test if i got the configs right
08:41.52JerJeralphaque:  now that's funny
08:42.21JerJerRowter: then fiddle with the rxgain and txgain settngs
08:47.09JerJerand check for ring/tip reversal
08:52.30NDTAnyone think they can solve a gnugk/aquagatekeepr problem for a few bucks? heh cause I am at a loss...
09:00.00RowterJerJer, whats is this tip-reversal mmh
09:01.25swm_[away]Anyone know any good applications pertaining to VoIP on the Nokia Symbian Series 60 Models
09:03.37*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
09:24.39NDTugggg.. like impossible to find anyone anywhere that knows anything about aquagatekeeper...hate this thing lol...
09:26.08swm_[away]Anyone know any good applications pertaining to VoIP on the Nokia Symbian Series 60 Models
09:28.50*** join/#asterisk syle (n=blah@unaffiliated/syle)
09:38.33*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
09:40.18RoyKwtf?
09:40.18RoyKOct  9 11:39:00 NOTICE[5169]: chan_sip.c:10685 handle_request_register: Registration from '<sip:71398087@213.160.242.140>' failed for '213.161.229.84' - Not a local SIP domain
09:40.19RoyKvoicemail1*CLI> sip show domains
09:40.20RoyKSIP Domain support not enabled.
09:44.30swm_[away]WTF is sip domain support? lol
09:45.32RoyKswm_[away]: just read sip.conf in the source
09:45.39RoyKor t the fm
09:45.43RoyKs&t&r&
09:45.47RoyKs/t/r/
09:45.48*** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com)
09:45.49RoyK:P
09:45.57*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
09:46.45nexiscan anyone provide me with a few links to point me in the right direction on how to setup asterisk as a gateway to connect to a IAX service, and provide it as SIP to my hardware phones?
09:47.15RoyKargh
09:47.17RoyKcvs down?
09:47.28swm_[away]It's dead I cant even get my questions answered
09:47.35RoyKnexis: it's all in the manual :P
09:47.53RoyK~docs
09:47.56jbotfrom memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
09:48.43RoyKswm_[away]: questions?
09:48.56nexisRoyK, thank you.
09:49.30swm_[away]know any good applications pertaining to VoIP on the Nokia Symbian Series 60 Models
09:51.41RoyKno idea
09:51.50*** join/#asterisk newl (n=newlook@203-59-91-217.dyn.iinet.net.au)
09:56.17RoyK:%s/newl//g
09:56.58swm_[away]~karma oej
09:56.58jbotoej has neutral karma
09:57.17swm_[away]~kharma
09:57.22swm_[away]~karma
09:57.22jbotswm_[away] has neutral karma
09:57.25swm_[away]~karma bkw
09:57.25jbotbkw has neutral karma
09:57.32swm_[away]~karma Junk-Y
09:57.32jbotjunk-y has neutral karma
09:57.42RoyK~karma RoyK
09:57.42jbotroyk has neutral karma
09:57.50RoyK~karma jbot
09:57.50jbotjbot has karma of 7
09:57.53RoyKjbot--
09:57.59swm_[away]~karma twisted
09:57.59jbottwisted has karma of 2
09:58.04swm_[away]~karma drumkilla
09:58.05jbotdrumkilla has karma of 8
09:58.10swm_[away]~karma anthem
09:58.11jbotanthem has neutral karma
09:58.15swm_[away]~karma denon
09:58.15jbotdenon has neutral karma
09:58.57swm_[away]~lart RoyK
09:59.25swm_[away]~lart RoyK
09:59.40swm_[away]~lart RoyK
10:01.40RoyKpetefile == someone who likes pete? peat?
10:01.42RoyK:)
10:02.03swm_~jbot dict peatafile
10:02.08swm_~jbot dict petafile
10:02.17swm_~jbot petefile
10:02.22swm_~jbot dict petefile
10:02.29RoyK~swm_
10:02.39RoyKjbot: swm_?
10:02.41swm_~jbot dict pervert
10:02.41RoyKjbot: swm?
10:02.56swm_~jbot define royK
10:03.04swm_~jbot shutdown
10:03.06RoyKjbot: swm_ is the generic idiot
10:03.08jbotRoyK: okay
10:03.17swm_~jbot reboot
10:03.18jbotNot on your life cowboy :(, or are you using Window$?
10:03.25swm_~jbot kill
10:03.33swm_~jbot terminate
10:03.37swm_~jbot die
10:03.38jbotACTION takes two shots to the head and crumples to the ground, lifeless.
10:03.57swm_Ohh IGNORE COMMAND. WOW
10:05.22RoyK<PROTECTED>
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10:16.24andrownzcan anyone tel what's the best between a [sip client] and [skype/gtalk/...]  just to talk to a friend that's also on his computer?
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10:32.11andrownzno one can tell??
10:34.06pooh_Heya RoyK
10:35.43pooh_so WHEN is there no sound at all on * channels? Festival and milliwatt are working that is it
10:35.47pooh_RTP streams?
10:35.57pooh_Setup is on LAN so no NAT
10:36.58RoyKpooh_: give me that login info again. i can take a look if you like :)
10:38.21pooh_RoyK: hold on
10:38.42RoyKbrb. quick shower && 0xc0ffee
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11:21.41zuuluuis there a working cvs head mirror?
11:26.59*** join/#asterisk Inv_arp (i=junya@adsl-156-145-65.mia.bellsouth.net)
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11:47.44swm_know any good applications pertaining to VoIP on the Nokia Symbian Series 60 Models
11:52.06RoyKpooh_: what hardware?
11:52.15RoyKpooh_: tried with cvs head?
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11:58.42pooh_RoYK: QUADBri from Junghanns.NET
11:59.00RoyKic
11:59.08RoyKyou told me on msg :P
11:59.22pooh_just for the record ;-)
11:59.29RoyK:P
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12:06.01MRH2oi whats up with www.asterisk.org ?
12:06.53MRH2http://66.249.93.104/search?q=cache:Z77NU2BTL4QJ:www.asterisk.org/+&hl=en
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12:07.14MRH2vs
12:07.20MRH2http://www.asterisk.org/
12:10.01TiliMRH2: asterisk.org is busted
12:10.18MRH2k
12:14.01*** part/#asterisk Mw3 (n=mw3@daisy.chains.ch)
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12:31.17zuuluuis asterisk dead?
12:31.39coppiceits just taking a nap
12:31.58swm_anyone know any good applications pertaining to VoIP on the Nokia Symbian Series Models
12:32.47coppicemost symbian phones can't use the DSP for the codecs, and lack the processing power on the applications ARM to do much VoIP work
12:32.56zuuluuseems their cvs and .org code is all old 1.0 stuff
12:33.44coppiceoh, I see what you mean. the asterisk.org website. looks like someone screwed up a change
12:35.09zuuluucvs.digium.com too
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12:35.53swm_I have found 4 apps for the Symbian platform
12:36.11coppicewhat do they do?
12:36.34InfraRedstream porn
12:36.42*** join/#asterisk maager (n=mager@ua-83-227-134-28.cust.bredbandsbolaget.se)
12:37.02swm_SIP VOiP Applications
12:37.23swm_I cannot seem to find Buzz2talk ... an actual download. back in 2004 it was released .
12:37.37coppiceSIP over bluetooth?
12:37.40*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
12:38.13swm_SIP over GRMS
12:38.33coppicedo you mean GPRS? SIP over GPRS is rather nasty
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12:39.31swm_Okay buzz2talk uses bluetooth over your phone
12:40.11swm_I have a 4 BT AP's around my house and use them for my PDA but I'm wondering if I can do anything interesting with them via my 3620 Nokia
12:40.45swm_(Class 1 Access points)
12:40.51swm_100 Ft Range on each of them
12:52.24Tiliswm_: can u give me links to those apps you found. I think most of them are only for SIP text messaging and IM stuff and not VoIP usually
12:52.53Tiliswm_: Most of mobile phones also dont support InputStream and outputStream at same time. i tried once building such app for BT
12:54.18*** join/#asterisk Romik (n=romik_@212.143.5.146)
12:56.48pooh_With which * version is the latest mISDN compatible pls?
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13:02.19*** join/#asterisk magic_2 (n=magic_1@ndn-165-156-198.telkomadsl.co.za)
13:02.36magic_2lo all
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13:07.37magic_2how would i go about setting irc on my asterisk server
13:08.03netsurfermagic_2 - elaborate
13:09.15magic_2well read that i could use/setup irc on my asterisk
13:09.19jhiverhi all
13:10.12netsurfermagic_2 - <shrug> download and install BitchX or something
13:11.02magic_2excuse my ignorance but is bithx an irc client for asterisk
13:11.16netsurferits an irc client for linux
13:11.33magic_2cause i was thinking that i could setup a irc chan for my cleint to speak tp me or my tech
13:11.38magic_2o i c
13:11.57magic_2i am currently using linux
13:12.11magic_2but not sure what the irc client is
13:12.31jontowxchat is what you're using now.
13:12.44magic_2makes sens thanx
13:12.47jontow:)
13:15.52magic_2is there a way that i could set something like that
13:18.31jontowof course -- but you need to do some reading
13:18.42Chujimagic_2 : You can host your own irc server or use a public one. Plenty of info on the web
13:18.50Chujiand has nothing to do with asterisk :)
13:27.09*** join/#asterisk oling (n=chatzill@p54B8AF81.dip0.t-ipconnect.de)
13:29.40olingwhen i forward an incoming sip connection to an iax client, does asterisk decode and encode the audio?
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13:32.17olingdo i lose audio quality?
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13:54.12pooh_oling: encoding/decoding depends on the end peers
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14:04.32olingpooh_: if both use the same codec it's not reencoded?
14:10.38*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
14:17.00pooh_oling: nope, should be a bridged call
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14:25.46pooh_how the heck to I prevent USB to be active (Bios no help) I want to disable USB grabbing an IRQ or be active at all
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15:00.49fenlanderwhy is ZX81 sad?
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15:02.44ZX81~ping
15:02.45jbotpong
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15:06.04ZX81~ping
15:06.05jbotpong
15:06.11ZX81~pong
15:06.12jbotwheeeeeeeeeeeeeeeeeeeeeeeeee!
15:06.17ZX81:(
15:08.05*** join/#asterisk Craziman2 (n=Craziman@63.108.128.250)
15:09.26Craziman2I am running * 1.0.10 with Cisco 7960 phones.  I am trying to get calls that time out from being parked to ring back with a different ring tone.
15:10.22Craziman2I tried creating a parkinglot context and having a s,1,SetVar(ALERT_INFO=<Bellcore-dr1>)  but it didn't work
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15:12.39*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
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15:22.28Qwell1.0.10 was released?
15:23.34Craziman2qwell Connected to Asterisk CVS-v1-0-10/08/05-13:11:17
15:23.42Qwellhmm, no
15:23.58QwellThats cvs stable, v1-0, 10/08/05
15:24.45Craziman2Sorry... misread it.
15:26.42brookshirebut technically that was correct.. heh
15:26.51brookshirehowever 1.2 will probably be the next release
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15:27.14Craziman2hey
15:27.59brookshirehi
15:28.11file[laptop]bonjour
15:28.16brookshirehey file
15:28.22file[laptop]hiiiiii
15:32.49*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
15:33.41RoyKheh. just got an email from a quite pissed off customer after having called inmersat three times a few seconds and then getting billed about $35 for it :)
15:34.00SkramXThats not good
15:34.03RoyKcalling inmersat hsd isn't really budget stuff
15:34.18Qwellwhy would they even call there?
15:34.23RoyKno idea
15:34.33SkramXwhat is that
15:34.35RoyKbut he did, three times in under 12 hours
15:34.37SkramX:D
15:34.38*** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com)
15:34.46RoyKSkramX: satellite phones
15:34.52RoyKhsd is high-speed data
15:34.53Qwellinmerset is like...yeah...middle of freaking nowhere
15:34.55SkramXRoyK: Oh.
15:34.57file[laptop]RoyK: and you explained to him how expensive it is?
15:34.58RoyKwhich is extra costy
15:34.59Qwellall over the world though
15:35.05RoyKfile[laptop]: yes
15:35.12SkramXIckys.
15:35.15file[laptop]and then he still wanted a credit?
15:35.52RoyKfile[laptop]: i've just sent the answer to my boss for control. I'm not sure how boss wants the case handled
15:36.06file[laptop]ah
15:36.17QwellI wouldn't credit any of it. :p
15:36.26Qwellif I were to though, it'd be the first one
15:36.37RoyKif boss wants to credit the customer for dialing a number he possibly didn't know was that expensive, well that's up to him
15:36.55*** join/#asterisk gonzo- (n=gonzo@web.portaone.com)
15:36.55SkramXRoyK: How much do you all charge for it?
15:36.56RoyKit's only like NOK 210 (eur 25) or so
15:37.07RoyKabout nok 70/min
15:37.17SkramXthats about 45USD.
15:37.27QwellSkramX: 25 eur is like 30 usd
15:37.40QwellI just got a 25eur paypal the other day, heh
15:37.41SkramXReally? Okay.
15:37.49Qwellif its more, I'd be kinda pissed...
15:38.03RoyKNOK 210 is USD32,-
15:38.06SkramX25 Euros = 30.36 U.S. dollars
15:38.08SkramXYeah.
15:38.35RoyKbloody expensive
15:38.41SkramXQwell: Yeah, but paypal takes a nice chunk sometimes; I wish they posted their exchange rates.
15:38.48RoyKbut then, who needs high-speed data transfer in the himalayas?
15:39.42file[laptop]SkramX: they have a calculator
15:39.42*** part/#asterisk Craziman2 (n=Craziman@63.108.128.250)
15:39.57SkramXhehe
15:40.40file[laptop]like for Paypal... 25 euros is...
15:41.04QwellSkramX: dunno, the exchange was free to do
15:41.09file[laptop]$29.55 USD
15:42.30*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
15:47.02*** join/#asterisk Badenser (n=user@191-60-124-83.dsl.3u.net)
15:47.37*** join/#asterisk asterisk99 (n=chatzill@modemcable169.194-130-66.mc.videotron.ca)
15:48.31BadenserHelp! What does "  chan_sip.c:6865 handle_response: Failed to authenticate on INVITE to ..." mean? What did I do wrong?
15:48.46af_may I stuff two tdm400 in a box without troubles?
15:48.50Qwellaf_: sure
15:48.51asterisk99anysose ere usig Gentoo? i cann;t figure out how to have zaptel to load after a reboot
15:48.52RoyKBadenser: wrong password?
15:48.53Qwellusually, anyhow
15:48.58RoyKaf_: prolly
15:49.02Qwellasterisk99: lemme see how I do it..one sec
15:49.15af_I think to manage 6 fxo in a box
15:49.21RoyKasterisk99: modprobe.conf iiirc
15:49.31BadenserRoyK, no, otherwise this would be another error message. And I successfully registered etc. ... the pw is correct
15:49.44Qwellyeah, modprobe.conf indeed
15:49.56RoyKBadenser: hehe. not running 1.0.x or anything?
15:50.02Qwellasterisk99: make install on zaptel should have added them to that file for you
15:50.22asterisk99Qwell: moodprobe loads it OK.... but after a reboot, it woo;t reload automatically
15:50.28BadenserRoyK, Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by root@boomtime on a i686 running Linux
15:50.37RoyKbingo
15:50.50RoyKthere's a bad nonce handling in asterisk
15:51.02Badensermeans .. ?
15:51.48asterisk99Qwell: is there a file that linus referenced during OS boot to load the drivers?
15:52.52asterisk99Qwell: Egad... that is bad english ... Linus => Linux ... referenced => references
15:53.14RoyKBadenser: see http://bugs.digium.com/view.php?id=2746
15:53.34RoyKBadenser: the patch attached there works
15:53.55BadenserRoyK, thanks ...
15:53.56RoyKBadenser: iirc this bug is still not fixed
15:54.07Badenser*sigh* means compiling .... uuhhhh
15:54.08Badenser;-)
15:54.13RoyKi paid for having that bug written......
15:54.32RoyKBadenser: i could send you a patched-up tree if you like
15:54.53RoyKwith that patch plus a lot of other stuff, including a rather nasty RTP/UDP port leak
15:55.03RoyKs/a ra/a patch to a ra/
15:55.10BadenserRoyK, thanks ... mh, will try it first (using Debian sources etc.)
15:55.21RoyKBadenser: don't do that
15:55.36Badenser?
15:55.47RoyKBadenser: you still need to get that RTP leak fixed if you want to use it with any volume
15:56.01RoyKdo you need bristuff?
15:56.15Badenserehm .. what was bristuff again ... ?
15:56.19*** part/#asterisk Dr_Ray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net)
15:56.26BadenserI just did apt-get install asterisk :-)
15:56.31RoyKBadenser: drivers for hfc pci plus a lot of other stuff
15:56.40RoyKBadenser: asterisk is constant work in progress
15:56.46RoyKbetter get the bread when it's warm
15:56.52Badenser;-)
15:56.59Badensernice said
15:57.44RoyKalso, don't use bristuff unless you need it
15:57.49Badenseror is it in the cvs yet?
15:57.58RoyKbristuff?
15:58.00Badensercould do a checkout of warm bread
15:58.06Badenserno, these patches
15:58.09RoyKno
15:58.11RoyKthey're not
15:58.20RoyKbut I think most of it is fixed in CVS HEAD
15:58.24RoyKnot in 1.0, though
15:58.38RoyKwill you be using mysql integration or something?
15:58.41*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfim6.dialup.mindspring.com)
15:58.44Badenserno
15:58.53RoyKjust a small install?
15:59.01Badenserjust need SIP, IAX, voicebox ... and maybe isdn in future
15:59.35RoyKisdn might be misdn, zaptel, isdn4linux, capi and the forth-coming sangoma native driver.....
15:59.40RoyKisdn has lots of different drivers
15:59.48Badensersangoma ... never heard of
15:59.53RoyKdot com
15:59.58RoyKcanadian co
15:59.59BadenserI would use capi
15:59.59*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
16:00.05RoyKBadenser: what hardware?
16:00.06Badenserand chan_capi
16:00.12BadenserAVM FritzCard
16:00.15RoyKah
16:00.18RoyKright
16:00.40RoyKalso Herr Junghanns' code
16:00.44RoyKas with bristuff
16:01.35Badenseryes ... Junghanns (long ago I tested it once ...)
16:02.15Badenseron the bug page ... it is said, that it is hard to reproduce ...
16:02.34Badensermh, but I can reproduce it deterministic ..
16:02.35RoyKwhat bug?
16:02.52BadenserFailed to authenticate on INVITE to
16:02.56RoyKyes
16:02.57Badenserthe thing you posted
16:03.05RoyKsome say it's hard to reproduce
16:03.15Badenserbut its absolutely easy for me
16:03.24RoyKbut I've already sent my cvs asterisk tree to another guy to fix the problem at his place
16:03.24Badenserjust try to call out ... *booom*
16:03.29RoyKyes
16:03.37RoyKbut then it only happens for some clients
16:03.40RoyKafaics
16:04.28Badenserthe probem seems to be: I have 3 numbers in one account at my  SIP provider (and so: same domain)
16:04.40Badenserregistering isnt a problem ... and one number works ...
16:04.48Badenserbut the other two not
16:06.03*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
16:07.45RoyKBadenser: http://karlsbakk.net/asterisk/asterisk-1.0.7-briiz.no.tar.gz
16:07.54RoyKBadenser: that might solve your problem
16:08.06Badenserthanks
16:08.09Badenserwill check it
16:08.35RoyKthat also fixes that nasty rtp leak
16:08.59Badenserand if you once need someone to reproduce the problem in a deterministic way, just call :-)
16:09.24*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
16:09.41RoyKBadenser: just file a new bug with the info and add a note about that being a follow-up for my bug
16:09.51RoyKBadenser: it's pretty bad they don't fix bugs in -stable
16:10.06RoyKs/pretty/bloody/
16:11.21Badensermaybe I open the bug in Debian's system; maybe the "preassure" is a bit higher then ...
16:12.09*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
16:12.09RoyKBadenser: doubt it
16:12.18RoyKBadenser: i'd rather use bugs.digium.com
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16:21.08BadenserRoyK, thanks for help ... will check it in a calm hour
16:21.09*** part/#asterisk Badenser (n=user@191-60-124-83.dsl.3u.net)
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16:39.15cioHi all.  Is there somewhere I can set the default no-caller id text to something besides 'asterisk'?
16:39.31*** join/#asterisk Rav1974 (n=r@ool-457a17a9.dyn.optonline.net)
16:45.45Rav1974I have a CAC 600 + T1 card.  I can't get the trunks setup in asterisk. (For incoming & outgoing lines) please help
16:46.42pooh_sorry, no knowledge of CAC600
16:47.05Rav1974:( poohpy cock
16:47.10Rav1974sorry bad pun
16:47.31pooh_Which T1 card?
16:47.38pooh_what card ...
16:47.41Rav1974TE110p
16:47.54pooh_What is a CAC600?
16:47.59Rav1974its setup correctly, but I dont have any instructions
16:48.10Rav1974CAC/ADIT 600 is a channel bank
16:48.14pooh_ah...
16:48.25Rav1974I have a 1 fxo card & 2 fxs cards
16:48.29pooh_lsmod tell you the driver(s) are loaded ?
16:48.36Rav1974all zapata & zaptel conf files look good
16:48.42Rav1974i can ring on the fxs cards
16:48.49Rav1974from my sip phone
16:48.54Rav1974& vice versa
16:48.59pooh_but....?
16:49.20Rav1974the outgoing doesn't work
16:49.38Rav1974I'm not very well versed in linux/asterisk or AMP
16:49.54pooh_How doe syour dial command look like ?
16:51.31Rav1974pooh_: I actually took an easy way by installing AAH 1.5 so I'm not sure where to see the dialplan
16:51.41pooh_ewwwwww
16:51.54Rav1974sorry, but I think its the best
16:51.56pooh_should be at '/etc/asterisk/exensions.conf
16:52.04Rav1974not everyone is an expert
16:52.12pooh_people may have other thoughts over here ;-)
16:52.49Rav1974do you want me to paste here or at pastebin?
16:52.57pooh_pastebin is ok
16:53.11pooh_include zapata and zaptel
16:53.28pooh_make sure no private stuff is in there
16:53.35Rav1974k
16:54.53ciohrm...
16:55.07ManxPowerRav1974, Not many people can help you with A@H
16:55.11pooh_morning ;-)
16:55.26ManxPower'morning, pooh_
16:55.26Rav1974pooh_: http://pastebin.com/pastebin.php?dl=388236
16:55.27pooh_Heya ManxPower, correct, just giving it a try here
16:55.30cioI just realized I'm not getting any caller id info on my new TDM400P with CVS Zaptel.  Anybody know how to troubleshoot this?  My /etc/asterisk/zap...conf has usecallerid=yes..
16:55.35*** join/#asterisk adker (n=adker@67-51-237-234.dsl1.glv.ny.frontiernet.net)
16:55.49ManxPowercio, and callerid=asreceived
16:56.54*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
16:57.24ciohmm... No luck... Weird...
16:57.50Rav1974pooh_: hope you can help :(
16:57.54ManxPowercio, and you have both lines BEFORE the channel => lines?
16:58.16pooh_Rav1974: What does the CLI tell you when you make a trunk call ?
16:58.29pooh_set verbose 20
16:58.32blitzragemorning all
16:58.36pooh_morn
17:00.17pooh_Rav1974: looks like a config issue
17:00.43Rav1974pooh: Verbosity was 3 and is now 20
17:00.44Rav1974<PROTECTED>
17:00.44Rav1974<PROTECTED>
17:01.04Rav1974pooh_: thats exactly what it is.
17:01.45ManxPower'morning blitzrage
17:01.53blitzrageManxPower: how goes?
17:02.11pooh_Zap/21?
17:02.11ManxPowerblitzrage, My ISP goes down more often than Lynda Lovelace
17:02.17blitzrageManxPower: LOL!
17:02.35*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
17:02.46kl_pl_01Hi, has anybody tried to push T1 traffic (betwen two T1 devices) over E1 link (SDH) ?
17:03.00pooh_# channel 21, WCT1, unhandled for now
17:03.25ManxPowerblitzrage, I managed to get my Cisco to connect to a dialup ISP to at least get SOMETHING running.  Whoever at Cisco decided to default Ser0 in async mode to 8 data bits, no parity and TWO stop bits should be shot.
17:03.26pooh_Rav1974: Time to consult AAH fora
17:03.35Rav1974ok :)
17:03.36cioManxPower: Yep.
17:03.37cioWeird.
17:03.40cioIt was working -
17:03.59Rav1974pooh_: where is aah forum? I didn't see one
17:04.06ManxPowercio, having too low/too high of rxgain can make callerid break
17:04.15cioThere's nothing in /etc/zaptel.conf that would affect callerid is there?
17:04.22cioNo, haven't messed with the rxgain at all.
17:05.12cioEverything just says "asterisk" ...
17:05.20cioIt doesn't even log the callerid in the cdr logs.
17:05.27*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-137-137.dsl.sfldmi.ameritech.net)
17:05.29cioWonder if the feature got disabled somehow at the telco?
17:05.42ManxPowercio, "asterisk" means "I didn't get any callerid"
17:05.49blitzrageManxPower: TWO stop bits?!
17:05.57blitzrageManxPower: now that's fucked
17:05.57ManxPowerblitzrage, Yeah.
17:06.01Qwellblitzrage: y0, how's Anaheim?
17:06.03ManxPowerblitzrage, yeah.
17:06.10blitzrageQwell: r0ckin
17:06.11cioYea, that's what started me looking into it. I was trying to figure out a way to change that text to "No Caller Id" or something besides "asterisk", then I realized I wasn't getting *any* caller id?!?
17:06.30Qwellblitzrage: happen to see my question in #astricon earlier? :)
17:06.50blitzrageQwell: ugh.... no -- let me look :)
17:07.16cioAny other suggestions? Thanks for the help, btw.
17:09.02cioI think it's on the telco side.
17:10.05cioOn a related subject, is there a way to change the default text when there is no caller id to something besides "asterisk"?
17:11.32*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
17:11.38ManxPowerblitzrage, apparently the ISP uses Linux as their routers and they keep crashing
17:11.50ManxPowerI'm tempted to just buy the guy a cisco
17:12.09tzangerManxPower: who's that
17:12.18tzangerwe use linux for routers and it works very well
17:12.27ManxPowertzafrir, the two guy wireless ISP I use.
17:12.30Qwellmeh, my router crashes quite a bit these last few weeks
17:14.09*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:14.18ManxPowerHe seems like the typical "scrounge old hardware" geek, which I'm SO over.
17:14.24*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
17:14.59*** join/#asterisk quasi2k (n=Marcus@dsl-217-17-22-122.teliko.net)
17:15.01ManxPowerHis network diagram looks like the plate of spagetti I had for dinner last night.
17:15.12quasi2khi
17:15.37quasi2ksome german asterisk pros in here?
17:15.54tzangerahh
17:16.06tzangeryes there is a difference between linux rouers and linux routers on crap
17:16.16tzafrirquasi2k, try #asterisk-de (is there such a channel?)
17:16.34*** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be)
17:17.18quasi2knope
17:18.05*** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
17:18.26ManxPowerThe guy doesn't even run a routing protocol on his network
17:18.56Qwellsilly people...think they can make a whole ISP and not know what they're doing
17:19.17ManxPowerQwell, His whole operation reminds me of the first ISP I started.
17:19.21Qwellhaha
17:19.27ManxPowerUnder funded, under staffed, junk equipment, little experience
17:19.33quasi2kdoes anyone use hfc and avm combination
17:20.09quasi2kfor s0 intern line and avm extern
17:20.11ManxPowerIf I was planning on staying in this miserable excuse for a town I'd try to buy him out.
17:20.27QwellManxPower: heh
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17:24.16hhrphi
17:24.33*** part/#asterisk darkskiez (n=darkskie@host-84-9-237-20.bulldogdsl.com)
17:25.13hhrpwhere can i find info on how correctly configure oh323.conf, i need to be able to receive a call from h323 gw and forward it to an extention in sip.conf
17:25.41ManxPowerPics of the damage Katrina caused to my boss's mother's house (including many pipe organs) http://www.demajo.net/organ/katrina/index.htm
17:26.59marc324ne1 know how to download j2se from shell?
17:27.42Qwellmarc324: wget?
17:27.49Qwelllynx?
17:28.15marc324the files are behind a form
17:28.18SkramXmarc324: what distro?
17:28.33SkramXyou may want to use ports/portage/a binary
17:28.38SkramXlike an rpm, etc
17:29.07Qwellyou need to download the file with portage
17:29.18Qwelldownload it on your own, that is
17:29.59marc324https://sdlcweb4d.sun.com/ECom/EComActionServlet;jsessionid=3E73F1F8C1684E635AC3886090081F41
17:30.13marc324http://java.sun.com/j2ee/1.4/download.html#sdk
17:30.27SkramXjaja. THIS WEBSITE IS DEDICATED TO OUR LOUISIANA POLITICIANS, INCLUDING THE ORLEANS LEVEE BOARD, THE US ARMY CORPS OF ENGINEERS, AND THE CONTRACTOR WHO BUILT THE 17th STREET CANAL LEVEE. MAY THEY SPEND ALL ETERNITY IN HELL LISTENING TO KEN GRIFFIN PLAY THE 'BEER BARREL POLKA' ON A HAMMOND B-3."
17:30.32SkramX*haha.
17:30.58hhrpwhere can i find info on how to configure an incoming h323 gw?
17:31.09marc324wget is only when you know the url.
17:31.19*** join/#asterisk oej (n=Olle@dsl001-136-130.lax1.dsl.speakeasy.net)
17:31.22SkramXmarc324: what is the "form"
17:31.30SkramXwhats the link, ill figure it out for ya.
17:31.48marc324http://java.sun.com/j2ee/1.4/download.html#sdk
17:32.48*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
17:33.34SkramXat a bash prompt?
17:33.38ManxPowerSkramX, *nod*
17:33.48marc324yes
17:33.49InfraRedhhrp: tried the voip-info.org?
17:33.54SkramX$/# lynx http://tinyurl.com/cvud8
17:34.33SkramXit should open a little "window" asking if you want to download the file
17:34.39*** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
17:34.42SkramXI assumed you want english and for the linux platfor?
17:34.46SkramX*platform
17:35.36SkramXmarc324: working?
17:36.20InfraRedhhrp: also look into transcoding
17:37.06*** join/#asterisk chet-- (n=chet@cpe-065-190-056-004.triad.res.rr.com)
17:37.50hhrpi cant find any decent info on oh323.conf file, infrared
17:38.07hhrpi know how to config gw and end point in sip.conf no prob
17:38.13InfraRedjust a matter of interest, why are you using h323?
17:38.15chet--whats the concensus on openpbx?
17:38.34marc324skramx- works thx
17:38.43hhrpnetwork where my calls are coming from uses h323
17:39.15InfraRedAlternative implementation of H.323 protocol support for Asterisk PBX.
17:39.15InfraRedWWW: http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/
17:39.21InfraRedis this what you're using?
17:39.42hhrpi created a context in oh323.conf [gw] .. put ip etc but how to specify to forward incoming call from that gw to a sip extention?
17:39.48hhrpyes
17:40.27InfraRedi am talking out of my arse here but it could be useful
17:40.36InfraRedis it accepting the call on h323?
17:40.41hhrpyes
17:40.50hhrpi have samples
17:40.53InfraRedyou running h323 debug or some such command?
17:41.05hhrpand in sample it has all-prefixes
17:41.27hhrpi dont understand what alias= is for in there
17:41.36*** join/#asterisk acidfoo (n=acidfoo@66.11.160.156)
17:41.44hhrpim in console getting output
17:42.39*** join/#asterisk DukeOfURL (n=chatzill@67-41-211-223.brbn.qwest.net)
17:43.18hhrpit starts a call at all-prefixes
17:44.23*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
17:45.51SplasPoodAfter a Dial() when it sets DIALSTATUS to ANSWER, is there any way to determine which SIP/???? answered the call (assuming I'm dialing multiple channels at once)
17:46.08*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
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17:56.56hhrpdoes anyone have experience configuring oh323.conf?
17:57.04SkramXmarc324: sure.
17:57.13SkramXwhere can I email you the invoice?
17:57.15SkramXhaha
17:57.31hhrpyou talking to me?:)
17:57.47SkramXhhrp: http://www.voip-info.org/tiki-index.php?page=oh323.conf
17:57.49*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
17:58.01hhrphave probs cant figure how to make it work
17:58.11hhrpit has so little description no examples
17:58.37SkramXhhrp: sorry.
17:58.49SkramXNever used h323
17:59.08hhrpi got gws configured but i cant figure out what do i need to put in order for it to understand that an incoming call has to be forwarded to a particular extn
18:01.25SkramXhhrp:
18:01.26SkramX; Set the default context of H.323 calls.
18:01.26SkramX;
18:01.26SkramXcontext=voip-h323
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18:08.44shankygood afternoon
18:09.26SkramXHello.
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18:20.49*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)
18:23.23*** join/#asterisk darby_d (n=tom@dno226.neoplus.adsl.tpnet.pl)
18:23.31*** join/#asterisk mithro (n=tim@p548E84D5.dip0.t-ipconnect.de)
18:24.50*** join/#asterisk darby_t (n=tom@dno226.neoplus.adsl.tpnet.pl)
18:27.28hhrpand how would it know what exten it is supposed to go to?
18:27.45hhrplike in sip you for regiter = .... /exten
18:27.56hhrpwhat do you do in h323.conf?
18:29.07bjohnsonregister has nothing to with exten
18:29.18bjohnsonregister tells another server what IP address you have
18:30.54*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj1j.dialup.mindspring.com)
18:32.23hhrpwhen you use sip and regiter you have /exten parameter
18:32.58hhrpso when a call comes ot that sip gw you registered it knows what exten to forward it to in context you define
18:33.20hhrpso why are you telling me it has nothing to do with exten
18:33.25*** join/#asterisk MrMAGO (n=mglucksm@pdpc/supporter/sustaining/MrMAGO)
18:34.06*** join/#asterisk SwK (i=pmmzwj@12-219-144-126.client.mchsi.com)
18:34.25hhrpi need all incoming calls on oh323 terminate to a particular exten and i cant figure outhow to do it
18:34.51MrMAGOhello... can someone help me please? in astcc, how would I specify a route for a 7 digit number? ^XXXXXXX$ ?
18:35.28netsurferMrMAGO
18:37.12netsurfernevermind =)
18:37.24QwellMrMAGO: something like
18:37.34Qwell^[0-9]{7}$
18:37.35Qwellmaybe
18:37.54Qwell^[2-9][0-9]{6}$  realistically
18:38.08QwellI suck at regex
18:48.29[hC]Man.. I wish i knew exactly what the culprit was .. I presume its NAT timeouts, but.. I have messages all day, about phones registering, unregistering, registering, unregistering
18:48.33[hC]or becoming unreachable
18:48.42[hC]I have expiry set to 45!
18:52.03jontowhmm.. anyone used the Varion V400P?
18:52.30*** join/#asterisk ManxPower (n=eric@slip-12-64-90-191.mis.prserv.net)
18:53.42*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
18:57.00*** join/#asterisk Howitzer (n=aa@217.136.147.180)
18:58.39twistedbwahahahahahaha
18:58.40twistedhttp://www.shoplaser.com/index.html
18:58.47twistedwhat will they think of next?
18:58.56*** part/#asterisk quasi2k (n=Marcus@dsl-217-17-22-122.teliko.net)
19:00.31*** join/#asterisk brookshire (n=matt@esbrooks3.traveller.com)
19:02.12*** join/#asterisk MnxPower (n=eric@slip-12-65-168-175.mis.prserv.net)
19:02.28MnxPower64 bytes from 209.16.72.158: icmp_seq=437 ttl=49 time=6315 ms
19:03.53QwellMnxPower: very nice
19:07.15*** join/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk)
19:07.49ret28Evening
19:08.35ret28"Oct  9 20:03:52 DEBUG[31712]: Sending dtmf: 50 (2), at 82.71.120.242" from the logs
19:08.56ret28(was attempting to hit '2' in the voicemail system)
19:09.33twistedDEBUG is not a complaint
19:09.58*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
19:10.00ret28Well, the voicemail app doesn't recognise it :/
19:10.12ret28Another client manages to use voicemail just fine though (same user)
19:10.31ret28"Oct  9 20:01:46 DEBUG[31712]: Detected DTMF '1'" <-- success
19:12.41ret28Oh, now that's odd ... just forced the client to send it inband instead of rfc.... and it works
19:12.53ret28Thought the dtmf method was autonegotiated though
19:12.58Qwellno
19:13.02Qwellthey need to be the same
19:13.15ret28Hmm
19:13.16Qwellinband only works on ulaw/alaw
19:13.34twistedinband only works effectively on those codecs
19:13.48twistedi have seen strange shit where it works partially on codecs as compressed as g729 :P
19:13.52Qwellheh
19:14.01ret28OK, what's the most liberal dtmf method?
19:14.14ret28(I was forcing inband because of kphone being inadequate)
19:15.40ret28twisted: Heh. It screamed at me when I tried forcing inband on gsm :)
19:15.45ret28(well, where scream == massive log spam)
19:15.58fileinband on compressed codecs is bad, mmmk?
19:17.05ret28What should I be setting it to by default, the rfc or info ?
19:17.27filerfc2833 is good
19:17.34ret28'k
19:17.57ret28I'll do that for default, and coerce the kphone user with dtmfmode=inband and disallow=gsm
19:26.29drumkillafile: poke
19:26.33*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
19:26.35filedrumkilla: prod
19:26.39brc_ZAP!
19:26.43*** join/#asterisk toddf (n=toddf@ns0.fries.net)
19:26.44brookshiredrumkilla: <3
19:26.45brookshire:D
19:26.48drumkillaw00t
19:27.16drumkillahow are you kids today
19:27.28Juggiegot astricon setup yet? :P
19:27.30*** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net)
19:27.34filedrumkilla: oh, it's a fun day
19:27.36brookshiredoing gooood :)
19:27.51drumkillai still have to do laundry before I can pack :)
19:28.04twistedme too
19:28.10twistedyesterday was clean the apartment day
19:28.11drumkillaoh, and finish 2 different programming assignments for school
19:28.14drumkillain 2 different languages
19:28.17twistedwhich i have done, and now i have like 3 piles of clothing
19:28.20drumkillaprolog and verilog! yay
19:28.31Qwellmeh
19:28.34Juggiei guess i need all shorts heh
19:28.40Qwellat least you guys don't have to move a whole apartment before you can go :p
19:28.42Juggieqwell, its gonna be hot isnt it
19:28.43drumkillasomeone check the weather
19:28.48QwellJuggie: high 80s
19:28.49twisteddrumkilla, already have
19:28.56drumkillaQwell: I move a whole apartment a few times a year
19:28.59twistedlow-mid 80s, with the low in the lower 50's
19:29.02Qwelldrumkilla: yeah...
19:29.03twistedalmost all week
19:29.07drumkillanice
19:29.09Juggiewill tere be hot girls :P
19:29.14Juggie*there
19:29.17drumkillawe need to have a beach party
19:29.18ret28BTW, cheers for the help guys, and nn
19:29.20drumkillatwisted: you with me?
19:29.22twisteddrumkilla, hell yea
19:29.25drumkillaawesome!!!
19:29.31twisted"pasty white boys visit the beach"
19:29.37drumkillaat night
19:29.39*** part/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk)
19:29.40Juggiedrum, you going to go to the ducks game?
19:29.40drumkilla... and drink
19:29.40twistedah yes. ok
19:29.44twistedsounds good :)
19:29.47*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
19:29.48drumkillai have no clue what I am doing
19:29.52Qwellmake sure you find a beach where drinking is allowed
19:29.52drumkillai am showing up
19:29.58Qwellthere are a few where it isn't
19:30.08drumkillayeah, most probably don't allow it ...
19:30.11twistedwe could go to laguna beach :P
19:30.15drumkillawhere i'm from it's not allowed
19:30.19_ThorHi everyone
19:30.30Juggiewhats not allowed?
19:30.34Juggieoh drinking
19:30.39_ThorHas anybody worked with PrepaidCall?
19:30.45Qwelleven smoking isn't allowed at some beaches
19:31.43fileugh I'm tired
19:31.49ManxPowerIt's Calif.  Smoking is punishable by death.
19:31.50drumkillafile: red bull
19:32.01filedrumkilla: nooooooooooo
19:32.23drumkillatwisted: #astricon
19:33.44_ThorI really thought that PrepaidCall was a very well supported add-on when I chose it
19:34.26drumkillai think most people use ASTCC
19:34.33ManxPowerI liked Madrid.  You could smoke in the hotel.
19:34.42QwellManxPower: Vegas
19:34.59ManxPowerQwell, I've not been to Vegas in like 10 years
19:35.32_ThorAnybody here has used ASTCC?
19:36.05ManxPower_Thor, Sorry, I don't bill for calls.
19:36.56_ThorAnybody here bills for calls :)?
19:36.56drumkillaManxPower: I went there for my 21st birthday this summer :D
19:36.56ManxPowerdrumkilla, vegas?
19:36.57drumkillaManxPower: indeed
19:37.11ManxPowerdrumkilla, gamble, get drunk, get laid?
19:37.11QwellAstricon vegas, 2006?
19:37.18drumkillathe first 2, at least
19:37.22QwellManxPower: the latter is no longer legal in that county
19:37.34_Thordrumkilla: did you get lucky?
19:37.39ManxPowerQwell, I know, but shuttle service is provided to places where is IS legal.
19:37.44drumkilla_Thor: overall, no :)
19:37.44*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
19:37.47Qwellshuttle service?  wow
19:37.47drumkillabut had a great time ...
19:38.35ManxPowerI got married in Vegas.
19:38.45drumkillaawesome
19:38.48QwellI did too
19:39.20_Thordrumkilla: In Vegas, all you got to do is ask
19:39.41Qwellor get really drunk...
19:39.54_ThorGet drunk, and then ask anybody
19:40.44blitzrageheheh
19:41.03_ThorJust donĀ“t pay more that 100 bucks :)
19:41.08Qwellnah
19:41.12Qwelldon't pay LESS than 100 bucks
19:41.36fileeep blitzrage
19:44.00ManxPowerFor something like that, I don't think going discount is a good value.
19:44.56_Thorno way in the world I would ever pay anybody $1000 for that
19:44.56MrMAGO^\d\d\d\d\d\d\d$ =)
19:47.43*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
19:49.20ManxPowerSex, like water and air, shoud be free.
19:49.27tzangerno
19:49.36tzangersex is like programming.  one mistake and you support it for life.
19:50.21Qwellsex is like an unmanaged switch.  it doesn't matter which por..nevermind
19:50.26tzangerhahaha
19:50.32_ThorAs a matter of fact, a girlfriend is much more expensive than a one time Vegas shot
19:50.53Qwell_Thor: girlfriends can be inexpensive.  wives on the other hand...
19:50.55tzanger_Thor: don't I know it
19:50.59MikeJ[Laptop]heh
19:51.01tzangerQwell: I know that too
19:52.29QwellMrMAGO: Thats no good.
19:52.50*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
19:52.51QwellMrMAGO: If a country has a very small amount of digits, one could dial international with that
19:53.10Qwellfirst one should probably be [2-9]
19:55.29*** join/#asterisk clive- (n=pirch@ndn-165-140-03.telkomadsl.co.za)
19:56.14*** join/#asterisk cyun (n=martin@tor/session/x-d0ce532d0b8aa91f)
19:56.30cyunHello, anyone good with asterisk that can help me set something up? I just want to create a simple setup where I can put up a phone and dial "000" to route me to line 1 and "001" to route me to line 2. Can anyone recommend where I should start or point me in a good direction? thanks.
19:57.02jake1wow i got asterisk installed 2 days ago and i have NO CLUE how to use it
19:57.13Qwell~docs
19:57.16jbot[docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
19:57.17Qwelljake1: start there
19:57.43clive-cyun thats a very basic question, check the docs, you should find your answer easily
19:57.56*** join/#asterisk oej (n=Olle@dsl001-136-130.lax1.dsl.speakeasy.net)
19:58.03Qwelloej: afternoon
19:58.09*** join/#asterisk nnnnnn (n=killfill@pc-200-74-17-222.nunoa2.pc.metropolis-inter.com)
19:58.41cyunclive-, ya i realize it must be, but I'm not familiar with pbx terms and lingo... can you give me a keyword to search the docs for?
19:59.31*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
19:59.41clive-cyun extensions.conf
20:00.48SplasPoodIs there any way to make Dial() continue on in the exten if the CALLING party hangs up, not just the called party?
20:02.15*** join/#asterisk kiwnix (n=egarcia@82.158.159.104)
20:05.58ManxPowerSplasPood, see exten => h,1,Blah as well as "g" option to "show application dial"
20:07.44_ThorI have a dumb question... I have compiled file app_prepaid_call.c, and it all creates well corresponding .so files....now, how in the world asterisk correlates function PrepaidCall() to that module, since they donĀ“t spell the same??
20:07.50SplasPoodManx: yea, but g only works for when the called party hangs up, not the calling party
20:10.36_ThorQwell
20:10.46Qwell?
20:11.03SplasPoodManx: any other thoughts?
20:11.34_Thorwhat correlation is between a module app_dial.so an command Dial() ?
20:12.04SplasPooddirect?
20:12.45*** join/#asterisk Craziman2 (n=Craziman@63.108.128.250)
20:12.51_Thoris it that Asterisk looks for its commands in folder modules and assumes anything apps-*.so is a command?
20:13.21SplasPoodI'm not positive, but I believe the module registers itself as an application
20:13.30SplasPoodI don't think it has anything to do with the filename, per say
20:13.41_Thorbut how does it knows?
20:13.44SplasPoodOne of the devs would be able to give a better answer
20:13.48brookshireare you running cvs?
20:13.51blitzrageManxPower: any suggestions for traffic shaping based on interfaces / networks ?
20:13.55drumkillait has nothing to do with the name
20:13.58SplasPoodI'd guess something like register_me_as_an_app()
20:14.07ManxPowerblitzrage, what hardware?
20:14.14drumkillaevery module has a function called load_module() that gets called when it gets loaded
20:14.15SplasPoodbrook: me?
20:14.19_Thormeaning, if file is named app_dial.so, who come it correlates it to Dial()
20:14.22drumkillaand inside that function, it registers itself as whatever it is
20:14.23brookshirespals: yeah
20:14.31blitzrageManxPower: computer running Linux / or whatever
20:14.34blitzrageManxPower: with 2 interfaces
20:14.45SplasPoodbrook: 1.2beta1 I'm pretty sure, since it shows.. no version on show version
20:14.52blitzrageManxPower: Sempron 2200+ with 512+ of RAM
20:14.56drumkilla_Thor: inside app_dial.so's load_module function, it registers Dial(), the dialplan application
20:15.06blitzrageManxPower: so I can run anything that a computer can run...
20:15.09ManxPowerblitzrage, not really.
20:15.14drumkillaa loadable module can have multiple dialplan applications in it, or other stuff, too
20:15.15blitzragedamn...
20:15.30drumkillaand the name does not force anything
20:15.35ManxPowerblitzrage, Want to know how I do it?
20:15.36drumkillathat's just for convenience
20:15.53blitzrageManxPower: please
20:16.11blitzrageManxPower: just want to make sure we have a good network here for Astricon... but I'm not a big network admin guy really :)
20:16.27_Thorguys, I am frantically reviewing the code as we speak :)
20:16.31ManxPowerblitzrage, I set all my voip devices to use tos=0xb8 (DSCP Express Forwarding), then traffic shape based on the TOS.  This assumes you trust all your network devices.
20:16.48SplasPoodbrookshire: any thoughts?
20:16.49ManxPowerblitzrage, too bad I could not attend
20:16.50blitzrageManxPower: yah... I won't :)
20:16.55blitzrageManxPower: yah... very much too bad
20:17.06blitzrageManxPower: can't trust people coming to this thing :)
20:17.14drumkilla_Thor: join #asterisk-dev if you have more questions
20:17.24ManxPowerblitzrage, you can trust them to overwhelm the network and crash the hotel's router.
20:17.28ManxPowerThat much you can count on.
20:17.33blitzrageManxPower: I want to shape based on interface/network really -- so I can set a lower priority for wireless than over the wired connections (which I'll be able to control)
20:17.46blitzrageManxPower: we have our own router with T1s attached to it
20:17.55blitzrageManxPower: I have total control of the Internet access
20:18.03blitzrageManxPower: hotel has no control of it
20:18.08ManxPowerblitzrage, what router?
20:18.10blitzrageluckily enough :)
20:18.24blitzrageManxPower: Cisco 18xx (can't rmeember the xx part)
20:18.35blitzrageManxPower: has 2 T1 interfaces and 2 eth interfaces
20:18.36*** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
20:18.37ManxPowerblitzrage, nifty.  Want a copy of my 1750 configs?
20:18.48blitzrageManxPower: hrmm... don't really want to change anything on there :)
20:18.53blitzrageManxPower: Speakeasy set it up for us
20:18.53ManxPowerblitzrage, wuss
20:18.56blitzrageManxPower: lol
20:18.57_Thorterrific help.... thank you guys
20:19.08ManxPowerblitzrage, I don't use PCs for routers
20:19.10blitzrageManxPower: ok... we're going for lunch now -- gotta go :)
20:19.15blitzragepeas!
20:19.16pooh_anybody familiar with qozap quadbri?
20:19.47cyuncan someone help me in setting up a simple routing setup?
20:20.13pooh_shoot
20:20.42cyunI just want to create a simple setup where I can put up a phone and dial "000" to route me to line 1 and "001" to route me to line 2.
20:21.13cyunI realize its a simple question, but I have no experience with pbx's and only need to do that one thing...
20:21.20Qwellzap?
20:21.22pooh_take a look at the sample config files
20:21.33clive-cyun stop asking the same questions over and over
20:21.34cyunhowever I'm really not wanting to go through tons of docs
20:21.47pooh_ot look at www.voip-info.org sip.conf iax.conf
20:22.03cyunclive-, i really didnt get any answer the first time
20:22.12SplasPoodcyun: you could hire a consultant
20:22.14cyunthis is a support chan right?
20:22.17Qwell<clive-> cyun thats a very basic question, check the docs, you should find your answer easily
20:22.30cyunyep
20:22.34cyuni did check the docs
20:22.38Qwellcyun: support, yes.  not "do everything for you for free"
20:22.43cyunbeen checking them for a while
20:23.11SplasPoodthere are a ton of examples on www.voip-info.org and within the example configs..
20:23.32cyunyep
20:23.37cyunill look into it
20:23.51*** join/#asterisk Assid (n=assid@203.115.64.57)
20:23.57cyunbut voip is not what i'm wanting to do.. maybe it'll apply somehow to what im doing
20:24.07cioOn a related subject, is there a way to change the default text when there is no caller id to something besides "asterisk"?
20:24.13Qwellcyun: If you ask a real question...you might get a real answer.
20:24.27QwellWhat is line 1?  Why do you need to dial a specific prefix to hit that line?
20:24.51cyuncause the phone will have one line connecting to it
20:25.01Qwelldefine line
20:25.04cyunand it needs to be able to access both lines
20:25.08cyuntelephone line
20:25.22cyunnormal walljack
20:25.30Qwellone phone, two lines?  Why not just setup a group so it doesn't matter if one is free or not?
20:26.04cyunwhats a group?
20:26.11Qwell~docs
20:26.14jbotrumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
20:26.34cyunright
20:29.54*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
20:30.00killfillhey, how do i see ive my zaptel x100p clone card can make calls?, where do i test this?.. im new to asterisk... :-p
20:36.19SplasPoodSo can someone explain why the 'g' option to Dial() only works if the target of the Dial() hangs up and not either party?
20:36.26cioOn a related subject, is there a way to change the default text when there is no caller id to something besides "asterisk"?
20:39.06QwellSplasPood: because the person who added that option intended it to be that way
20:39.26QwellSplasPood: as was said earlier, look at extension 'h'
20:39.51SplasPoodQwell: I've looked at it, and already have it in use..  It doesn't get called either..  I'm more than happy to listen to suggestions..
20:40.13*** part/#asterisk eGnarF (i=mrbk@eris.hb.lu.se)
20:41.55SplasPoodDial() simply exits -1 and thats where it stops..
20:42.24*** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net)
20:42.35SplasPood<PROTECTED>
20:43.16ManxPowerSplasPood, you need a NoOp(DIALSTATUS=${DIALSTATUS}) after the dial
20:43.34*** part/#asterisk opus__ (n=opus@dahphish.org)
20:44.32SplasPoodManx: It never gets called if the calling party hangs up, hence the problem..   However, i've just discovered that exten => h does get called, just in the calling context, not within the macro..  So I suppose that'd be my mistake when it came to that suggestion
20:44.40SplasPoodI still don't understand why 'g' just doesn't work both ways
20:45.06Qwellwasn't intended to work both ways
20:45.17ManxPowerSplasPood, doing posthangup processing sucks
20:45.22SplasPoodyes, but is there a technical reason, or just for fun?
20:45.51SplasPoodManx: I'm seeing this...
20:46.20SplasPoodWell anyway, a exten => h, in the calling context does what I need in that regard
20:46.30SplasPoodI had it within the macro, and that wasn't working
20:46.51SplasPoodNext question would be how I could determine which of SIP/7001&SIP/7002&SIP/7003 answered the call
20:59.55cioOn a related subject, is there a way to change the default text when there is no caller id to something besides "asterisk"?
21:01.15drumkillait's hard coded in chan_sip right now, but that has been under discussion lately
21:02.02drumkillain cvs head, you can change like 328, which is "#define DEFAULT_CALLERID "asterisk" "
21:02.49mog_homeDRUMKILLA you out at *con yet
21:03.03drumkillanope
21:03.06*** part/#asterisk cyun (n=martin@tor/session/x-d0ce532d0b8aa91f)
21:03.10drumkillai fly in with you guys tomorrow
21:04.47cioCool.  Thanks.
21:04.50mog_homeoh really
21:04.55mog_homeyou gonna hit huntsville
21:05.05pooh_Ok guys, really need some help here with quadbri and qozap module
21:05.10*** part/#asterisk IPFOX^^ (n=barny@cpc2-char1-5-0-cust96.sot3.cable.ntl.com)
21:05.11pooh_getting desperate
21:05.56drumkillamog_home: nope, i'm just on the same flight from TX to CA
21:06.03mog_homeahh
21:06.05mog_homenice
21:06.11drumkillayah
21:06.11drumkilla:)
21:06.12ManxPowerpooh_, I vaguely recall some bug where all audio stops working if you have a Zap driver loaded, but no line connected into it.
21:07.03ManxPowerpooh_, might be ISDN specific
21:10.57pooh_ManxPower: so if no lines connected quadbri does not work?
21:12.43ManxPowerpooh_, As I said, I don't remember.
21:12.51ManxPowerTry a mailing list search.
21:13.13pooh_ok
21:13.59pooh_ewww, how can I search the archives pls?
21:14.57ManxPower~mailinglist
21:14.58jbotfrom memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
21:15.23*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
21:15.37pooh_got it thx
21:16.33cioAnyone here have a number I can call and maybe you can tell me what caller-id info you get?  I'm trying to figure something out.. I think I have a problem with my telco..
21:16.45cioCan't call them of course, it's Sunday ..
21:17.15mmlj4hey ManxPower
21:21.05pooh_MaxPower: care to do a search for me, can not find anything :-(
21:22.07*** join/#asterisk SwK (i=ijgtfn@12-219-144-126.client.mchsi.com)
21:25.39pooh_ManxPower: care to do a search for me, can not find anything :-( (typo)
21:26.13pooh_I read something about module-init-tools.... anyway related ?
21:27.43Qwellpooh_: module-init-tools is needed to run a 2.6 kernel, instead of module-utils, or whatever
21:29.21pooh_ok, using 2.4 here. Thx
21:30.36*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc15t.dialup.mindspring.com)
21:30.53*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
21:39.02*** join/#asterisk vvo (n=michal@bxj30.neoplus.adsl.tpnet.pl)
21:41.24*** join/#asterisk tainted_ (n=identd@adsl-71-129-47-53.dsl.irvnca.pacbell.net)
21:41.51*** join/#asterisk darby_t (n=tom@dno226.neoplus.adsl.tpnet.pl)
21:43.15vvoo, ktos z Polski (;
21:45.43pooh_no polski ;-)
21:45.46pooh_njet ;-)
21:55.04pooh_Dearly beloved, we are gather here today to say goodby to a ' good'  friend......
21:56.10tamp4xkurwa
21:56.11pooh_anybody have some last words for thispiece of junk ?
22:00.55InfraRedfreaks
22:01.36pooh_sttt. we are morning
22:10.27*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-90-1.tu.ok.cox.net)
22:10.52*** join/#asterisk mithro (n=tim@p548E84D5.dip0.t-ipconnect.de)
22:14.40*** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net)
22:14.45ManxPower100 packets transmitted, 85 received, +1 duplicates, 15% packet loss, time 99836ms
22:14.52ManxPowerApparently Texas qualifies as a 3rd world country
22:16.43pauldyuhm ?
22:19.13pauldyone shot one kill
22:25.18*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
22:25.42*** join/#asterisk nvz (n=nvz@c-24-3-200-17.hsd1.pa.comcast.net)
22:25.52*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
22:26.17nvzoh wow lots of people here :)
22:26.21harryvvgood web site to shop for wholsale rates?
22:27.13*** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com)
22:27.46InfraRedharryvv: www.voip-info.org
22:27.52InfraRed:p under wholesaaale providers
22:28.01nvzhmm I just remembered there is an asterisk-doc package available.
22:28.12InfraRedyou want someone close to you, someone on the other side of the world who's cheap is no good IMHO
22:28.34harryvvyes, trying to beat telus rates for some people.
22:28.47nvzno wonder there wasnt any good docs on how to use this thing. I didn't have them installed
22:28.49harryvvfrom canada to india
22:31.19InfraRedsend an email to asterisk-biz list
22:31.25InfraRedloads of people will reply i bet
22:33.20harryvvyes
22:33.37*** join/#asterisk syle2 (n=blag@unaffiliated/syle)
22:33.51*** join/#asterisk nnnnnn (n=killfill@pc-200-74-17-222.nunoa2.pc.metropolis-inter.com)
22:38.48nvzAll I am interested in right now is using my PC for a caller-ID unit, and maybe later down the road for a call-filter, speaker-phone..etc this software seems really confusing and the documentation keeps talking about Voip  that doesnt seem related to what I want to do. maybe i should be using something simpler like mgetty?
22:40.16dudesActually asterisk is well documented, and it's not all that complicated.  Most things related to SIP/IAX can be reversed in a similiar fashion as using zap
22:41.13nvzI am reading the documentation now, it might as well be written in hieroglyphs for all the sense its making to me right now. I dont know all these fancy TLAs and telephony terms
22:41.42dudesyou need to install it and play with it
22:41.45nvzI havent crossed a single document yet that seems to explain the simple nature of what I want to do
22:41.57nvzI have it installed, and I havent the slightest clue what to play with
22:42.44dudesyou start in /etc/asterisk
22:42.55nvzits installed and running right now for whatever its worth. its just a daemon that runs in the background and it has over a dozen config files. I have no idea what that does for me
22:43.05dudesconfigure your zap/sip/iax phones and then edit extensions.conf to do what you want it to do
22:43.26dudesthat's why you read the docs on extensions.conf and such
22:43.28InfraReddudes: asterisk documentation is out of date with every release/subrelease
22:43.34nvzsee, thats what I am talking about I dont know wtf a ZAP, SIP, or IAX is
22:43.52InfraRednvz: learn quickly
22:44.01InfraRednvz: asterisk is telecom, not IT
22:44.29InfraRednvz: voip-info.org has all the stuff you need to learn, or most ofn it at least
22:45.10*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
22:45.16InfraRedjust because you come from IT background don't expect asterisk to be walk in the park
22:46.12harryvvyes
22:46.15nvzI am confused. all I keep reading is using undefined terminology and TLA's to descripe this "PBX" and "Voip" stuff and I have no idea what any of that is or what it has to do with a window poping up to tell me who's calling. :P
22:46.25harryvvseems some administrators have a tough time understanding asterisk.
22:46.46sivanaJerJer: you around?
22:48.26nvzI am not trying to create some business telephony system here. I just want to know who is calling my single home line.
22:48.37dwmw2_gonenvz: then you probably don't need Asterisk
22:48.40nvzI am beginning to think this software is far too complex for my needs
22:49.00harryvvwho is like in CID?
22:49.00InfraRedprobably is
22:49.07dwmw2_goneAsterisk is what you need to run a full private exchange, with real phone lines and/or Voice over IP.
22:49.07InfraRedit's a PBX
22:49.20InfraReda complete switchboard
22:49.25InfraRedwell, almost complete
22:49.26InfraRed:)
22:49.36nvzbatteries not included? :P
22:49.40*** join/#asterisk mithro (n=tim@p548E84D5.dip0.t-ipconnect.de)
22:49.41harryvvThanks giving here in Canada
22:49.47dwmw2_goneIf all you want is a window popping up showing the incoming caller-id when you receive a call, you probably want something a _lot_ simpler
22:50.11InfraRedYOU NEED MICROSOFT TELECOM 6000 XP
22:50.17nvzeww
22:50.22harryvvwhat is that?
22:50.25InfraRedall you need to do is to setup.exe
22:50.26InfraRed:)
22:50.42InfraRedharryvv: no idea, i just made it up
22:50.51dudesYou can set CID in Asterisk if you have a PRI or a VOIP trunk that supports it
22:50.52harryvvi need to embed my firewall on a small form factor system
22:51.55killfillhey, i cannot make my asterisk work.. :-p  im with asterisk@home, and my SIP phone (X-lite) shows "Login failed! contact the admin"  whn i try to "login to a extension".
22:52.41killfillhow do i see whats happening
22:52.42nvzI am not sure even if I studied my TLAs if it would be worth learning how this works or not. I may be interested in other things.. but the reason someone suggested this to me, was I want a caller id. so when I am down here on the PC I know if its worth stopping what I am doing to answer the phone
22:53.23nvzI am sure if I learned how this works I could probably build something to interact with my phone system from any PC in the house and do some interesting things
22:54.26nvzI should probably look into mgetty it says it can do the basic things I wanted.
22:54.45*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
22:55.30killfillhey nvz, would you give me a hand?..
22:55.40nvzheh
22:55.45nvzyou've got to be kidding
22:56.23nvzthats about the only hand I can give you
22:56.50killfillthanks..
22:57.28nvzI asked about using my PC for a caller ID and someone recommended me to this vastly complicated asterisk PBX software. I am clueless
22:58.12nvzI just read through /usr/share/doc/asterisk-doc/ and I wasnt able to grok any of it. because its not in english. its in engrish acronym
22:58.25killfillah didnt understood what you typed before...
22:58.25harryvvnvz, yea dont use it
22:58.48harryvvnvz, why not just call your telephone carrier to turn on caller id?
22:59.23killfilldamn.. im sure i messed something up yesterday night... i cannot even connect the x-lite client...
22:59.27harryvvnvz, asterisk is way to complex for the new person to understand.
22:59.49nvzharryvv: we have caller ID, and upstairs we got one of them talking caller-id phones. down here I got a three-headed monster (3 clustered pcs) I figured rather than going out any buying another caller id unit I could just put them to work as a caller-id
23:00.31nvzI don't spend money on technology unless it makes me money. and I quit doing IT long time ago. all this hardware I have is free.
23:00.50nvzI figured it was fully capable of serving my convenience
23:02.31*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
23:04.49nvzI will try hack something together. Its pretty obvious this is overkill. Ideally I only want the phone down here to ring if its someone worth talking to. I dont want to be working on the PC and have the phone ring only to discover its a creditor for someone who hasnt lived here in 40 years :P
23:09.04killfillthe sip phone log shows me this
23:09.06killfillSIP/2.0 403 Forbidden
23:10.09*** join/#asterisk oej (n=Olle@dsl001-136-130.lax1.dsl.speakeasy.net)
23:10.41wunderkinwelcome to america!
23:15.18*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
23:17.42nvzamerukia f*** yeah!
23:18.05*** part/#asterisk nvz (n=nvz@c-24-3-200-17.hsd1.pa.comcast.net)
23:18.26killfillshow agents  Show status of agents
23:18.52killfillwhat an agent?..
23:18.52killfilla phone?
23:19.34wunderkinno, an agent is typically used in a queue
23:20.23killfillhow do i show the eveiable extensions?.. im trying to login with the sip phone, but cannot..
23:20.40blitzrageoej: !!!
23:20.47blitzrageoej: ALARM!
23:23.01kramhi blitz
23:23.07blitzragekram: hey oh!
23:23.15blitzragekram: wuz up?
23:23.55Qwellkram: afternoon
23:25.20blitzrageok... wireless network audit time!
23:25.34*** join/#asterisk mithro (n=tim@p548E84D5.dip0.t-ipconnect.de)
23:26.03killfillhey guys..  want to give me a hand?..
23:26.13killfillthis is the log of x-lite:
23:26.15killfillhttp://pastebin.com/388599
23:26.20*** join/#asterisk mcquaid (i=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca)
23:26.37killfillwhy do i get forbidden?..
23:26.52mcquaidhello, can asterisk operate behind a firewall, without mucking with the ports on the firewall itself?
23:27.13mcquaidI guess I'm wondering if asterisk can perform the same nat traversal magic that something like skype does
23:28.22mcquaidnot sure if asterisk can be used in this case, but basically i signed up for vbuzzer.com and it works fine with their softphone
23:28.37mcquaidbut all linux softphones i try fail to connect and i believe it's a nat issue
23:29.55mcquaidand i'm wondering if asterisk could help me get around this, but i don't have access to the firewall, so i wanted to know if asterisk could perform properly behind a firewall and perform this nat traversal for me and then i'd use a softphone to connect to the asterisk server
23:31.35*** join/#asterisk techie (i=gus@70.86.133.66)
23:38.54*** join/#asterisk AsteriskNoob (i=BoredBoz@207-114-232-10.gen.twtelecom.net)
23:39.01AsteriskNoobhey everyone!
23:40.18AsteriskNoobwho here has played with the BETA?
23:44.45twistednobody
23:44.48twistedbetas are for wussies
23:44.52twistedwe like pre-alpha here.
23:45.09CoaxDasterisk is always pre-alpha. :)
23:45.20CoaxDGenerating DH parameters, 1024 bit long safe prime, generator 2
23:45.20CoaxDThis is going to take a long time
23:45.32CoaxDgotta respect unix; at least it tells you the truth
23:46.10mcquaidcan asterisk function behind firewall without having access to the firewall to open ports? the server does have nat running
23:46.23CoaxD(and on a Linksys WRT54G, it is *really* gonna take a long time.)
23:46.47AsteriskNooblol@twisted
23:46.52CoaxDmcquaid: If you're using IAX2 as a communications protocol in its entirety, yes, long's the IAX2 port's open
23:47.22AsteriskNoobwell I installed the 1.2 beta cause i was having issues with the 1.0.9 all of the sudden, I love the new features of 1.2 but they screwed up some PRI or SIP stuff... trying to trace it
23:47.22CoaxDmcquaid: Otherwise, youre pretty much screwed.
23:47.59mcquaidhmm, well i signed up for vbuzzer.com which is sip based, it's softphone works fine in windows, just trying to get any softphone client to work
23:48.32mcquaidi assume it's a firewall issue thats stopping me,  so i guess their app is doing some nat traversal magic similar to skype. and I was hoping asterisk could do the same thing
23:48.53mcquaidsince none of the other softphones in linux i've tried seem to do nat traversal
23:48.54CoaxDmcquaid: I dunno. i always have to putz with it to get it to work in that scenario
23:50.38mcquaidok thx for your help
23:51.11mcquaidhmm, there must be some softphone that can get around this nat issue if asterisk isn't suitable for this
23:57.24*** join/#asterisk chet-- (n=chet@cpe-065-190-056-004.triad.res.rr.com)

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