irclog2html for #asterisk on 20060811

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00:13.52jarrodzc
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00:23.44vltI want to answer incoming voip calls with a voice "You are on hold on position x ..." and then hand the call over to an existing ISDN/analog PBX with 16 analog phones, 4 ISDN BRI channels (2 of them connected to external line, the other 2 internal). Can I do this with asterisk? What hardware do I need to connect asterisk to the ISDN lines (maybe to one of the external channels, too)? ...?
00:25.54*** join/#asterisk anthm (n=anthm@h4608a74c.area4.spcsdns.net)
00:25.54*** mode/#asterisk [+o anthm] by ChanServ
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00:44.25rushowrmurf, anybody who knows AEL2 inside and out
00:44.33rushowrneed a quick question answered
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00:44.40rushowrdesperately
00:44.43*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
00:45.28rushowranyone know the difference between the 'jump' and 'goto' statements in AEL2?
00:47.02rushowrI think i know but....ah fuck it
00:47.03rushowrcheers
00:47.22*** join/#asterisk chreese (n=chatzill@bridalveil.istep.com)
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00:51.43stinkpadanyone got zaptel compiled correctly on sparc64?
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01:01.04*** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net)
01:01.23jtexter3If I have two channels bridged, what is the best way to disconnect them?
01:01.30jtexter3ast_softhangup?
01:04.10Snake-Eyeshmm i have question about softhangup too
01:04.58Snake-Eyesyet to get clear answer about how softhangup should be used
01:06.14*** join/#asterisk trbldwine (i=troubled@71.194.161.170)
01:07.24jtexter3Snake-Eyes, what have you run into with softhangup?
01:08.40Lyfecan anyone tell me if the syntax for this is correct (to go to priority 20 if dialing 918001234567): GotoIf($["${EXTEN:2:4}" = "800"]?20)
01:09.30Lyfepresuming you're after the 800 part, that is.
01:11.36Lyfenevermind, answered my own question, noop() is my friend.
01:13.12nailbags|laptopi'm getting a weird echo from my tdm400 ... it only happens after an attended transfer! any ideas?
01:13.21nailbags|laptop:q
01:13.22Snake-Eyesjtexter3, Ive been trying to use it in a macro to get it to hangup all calls on a sip trunk, yet it doesnt seem seem to hang up a call
01:17.42*** part/#asterisk trbldwine (i=troubled@71.194.161.170)
01:17.44jtexter3looking at various pieces of code, maybe ast_softhangup only hangs up if its AST_SOFTHANGUP_DEV, or a time
01:18.31*** join/#asterisk michaelo (n=michaelo@adsl-147-45-179.gsp.bellsouth.net)
01:19.33Snake-EyesAST_SOFTHANGUP_DEV ?
01:22.01jtexter3well, looking at the code for the actual API, you can pass it several soft hangup reasons.  Looks like the SoftHangup dialplan app won't actually hangup an application
01:22.22Snake-Eyesthe really scary part is how no one is concern that softhangup might not work, seeing how its used for e911 /000 calls
01:22.37Snake-Eyeshmm
01:23.11*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
01:23.30Snake-Eyesjtexter3, I want it to hangup a call on a trunk, not app
01:24.18*** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au)
01:25.29Snake-EyesI want it to hangup acall on a trunk so a emergency call can go through yet in the test ive done its not hanging up any calls ...
01:27.06blitzragehas anyone used the Local/ channel in Asterisk Business Edition?
01:27.27blitzragethe 1.0 release
01:27.40P-NuTHi guys. I'm using a cisco 7905 phone, with asterisk. When I call somewhere with an IVR, i try to press the buttons for the options, but it doesn't recognise that i'm pressing any.
01:27.46P-NuTIs this an asterisk issue?
01:27.52P-NuTor something else?
01:29.15Snake-EyesP-NuT, have you checked what you have set the dtmf for in asterisk and the phone?
01:30.39P-NuTumm..
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01:30.39Snake-Eyesblitzrage, whats the difference between normal Asterisk and Asterisk Business Edition?
01:30.48P-NuTno. What should it be?
01:31.53Snake-Eyeswell, if you're using g729 it has to be rfc2833, else you can use inband
01:32.24P-NuTi'm using g711 and in sip.conf I have this..
01:32.25Snake-EyesP-NuT, I would set dtmf on both to rfc2833 first and if that doesnt work then inband
01:32.26P-NuTdtmfmode=rfc2833
01:32.50Snake-Eyesok, then in your cisco phone set it to rfc2833
01:33.09P-NuTumm.. actually a setting "ON" the phone?
01:33.15P-NuTor in sip.conf?
01:33.28Snake-Eyes"ON" the phone
01:33.46*** join/#asterisk _deg_ (n=deg@200.181.137.62)
01:34.37*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
01:34.42quid246is it just me or are alot of calling-card outfits run by shady guys?
01:34.55P-NuThmm
01:35.00P-NuTdont know if I can do that
01:35.04justinu|laptopits not you
01:35.29quid246justina:  I saw one board and seems like every 5th message says "don't deal Mohammed he'll rip you off".
01:35.32[TK]D-Fenderquid246 : Nope... most do evil stuff like hijack other people accoutns, run over credit limits, frequently bankrupt and restart....
01:35.42Snake-EyesP-NuT, look at the manual for the phone, most voip phones have a web interface
01:35.43quid246or insert name
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01:36.18quid246They are definitely a rung below cellphone salesmen
01:36.23justinu|laptopheh
01:36.44quid246atleast the cellphone guys wear a suit, albeit a cheapie
01:37.24Snake-Eyescellphone guys wearing suites ?
01:37.56Snake-Eyesall the ones ive seen where pants and a open collar shirt with something hanging from there neck
01:38.04Snake-Eyeswhere=wear
01:38.46quid246haha yeah true... I haven't dealt with them since the days of the brick, so they were more slick then
01:39.30quid246which reminds me... Cingular is charging a surcharge for people with old analog phones.  Way to take advantage of the retro-lover... haha
01:40.56quid246I wish I still had my brick... would make for a great sight gag.  Excuse me I have to make a call, and pull the behemoth out of your briefcase
01:41.08omalmy understanding is that an older phone costs them more to serice than a new one
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01:41.30quid246omal:  Well, yeah... since they are analog they take up a heck of alot more bandwidth than the digital call
01:41.58omalso i wouldnt really call it taking advantage
01:42.09quid246I was ebing sarcastic
01:42.31quid246long-live the bag phone
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01:46.21quid246SOIP?
01:46.51omalSarcasm Over IP :D
01:47.06[TK]D-Fender~book
01:47.09jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
01:47.24*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
01:47.35Snake-Eyes~booked
01:47.39omalits also found on my coffee table
01:48.01quid246haha that's what I thought
01:48.24Snake-Eyeshmm
01:48.30Snake-Eyes~bot
01:48.32jbotI ain't no stinkin' bot.  I am a finely tuned and hand crafted tool.  Oh wait... I guess I am a bot (that you should not abuse).
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01:48.43Snake-Eyeshaha
01:48.47*** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
01:48.50Qwell~botabuse
01:48.51jbotStop tormenting me!
01:50.30file~botsnack
01:50.30jbotthanks, file
01:51.53Snake-Eyes~botlove
01:51.55jbotACTION hugs CIA-5
01:51.55[TK]D-Fender~lart himself
01:52.06[TK]D-Fender:D
01:52.36Snake-Eyeshehehe
01:53.17harryvvAnyone seen a case of a working zap with no errors just suddenly not respond ? My cli output was right. Well I just restarted the server and now its working.
01:54.31rene1~lart twisted[asteria]
01:55.01*** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com)
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01:57.04rene1~lart brookshire
01:57.09rushowrhey all, got another question or two, shouldn't be hard...lemme know if you mind trying to help :)
01:57.14*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)
01:57.31murfrushowr: hey there, still got the AEL question?
01:57.36rushowrhel;l yea
01:57.38quid246it's funny when you look at it... with the exception of windowed clients... IRC hasn't really changed much in 10 years
01:57.40rushowrthere you are!
01:57.48rushowrlet me get you in a pm
02:02.09harryvvquid, actually it has in one respect. The servers dont crash as much from abuse.
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02:03.32quid246harryvu:  Touche on that one.  I remember having to regain my channel on splits.
02:03.52userdefinedhola. can anyone point me to an extensions.conf example showing how to route a call based on the calling party?
02:04.04userdefinedi've been googling for a while but apparently don't know the magic words =/
02:07.01hadsuserdefined: exten => 1234/5432,1,NoOp()
02:07.11hadsWhere 5432 is calling party.
02:08.04userdefinedah! thanks very much
02:08.09hadsnp
02:09.48harryvvquid, yea those were the fustrating days
02:16.24omal[22:15:06] <Damin> Digium just got 13.2 million in Venture Capital funding! ;)
02:16.40file13.8
02:16.45dlynes_laptopomal: that was a misprint; it's 13.8
02:18.10omalhooray!
02:19.26*** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
02:19.38dlynes_laptopwell, not a misprint; a misreport; they later restated it when Mark corrected them
02:26.03[hC]Cool, i wonder what they'll do with it
02:26.04[hC]:)
02:26.25filewe're going to buy $13.8 million dollars worth of muffins
02:26.36Nivexfile: here I was gonna say a really friggin' big keg
02:26.56filepfft
02:27.41Nivexhmmm... no updates to Pound Key lately
02:27.46harryvvvm needs a option to dial the number after listening to the vm
02:28.26fileharryvv: what if it already has it?
02:30.01harryvvfile I have not seen it. Which option is it? the
02:30.24file; callback=fromvm       ; Context to call back from
02:30.24file<PROTECTED>
02:31.01*** join/#asterisk foo (n=foo@unaffiliated/foo)
02:31.08harryvvIn options it would be nice to have it say press 1 to call this number, press 6 to listen to new messages ect.
02:31.22fooWe have a VoIP talk at our local LUG right now ... /join #sgvlug ... the stream URL is in the topic. Free, open source. :)
02:31.22mitchelocif you can stream this, KG is giving a presentation right now on asterisk ;) http://stream.sgvlug.org:8800/
02:31.28fooHAHA.
02:31.29mitchelocmeh
02:31.30foomitcheloc: :P
02:34.17harryvvfile this in voicemail.conf
02:34.42*** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
02:35.19fileI just grabbed it from the example config in trunk
02:35.54fileit is also in 1.2
02:36.41[hC]i have to go thru the example configs again some time
02:36.45*** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
02:36.52[hC]so many features since i did my 'standard' configs back in 2004 :P
02:37.35rushowr•murf• hey murf I remembered what I was gonna ask, if you've got a sec
02:37.37rushowrPM me
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02:47.20harryvvhard to believe that a ipod could be used to blow up a plane.
02:49.30mitchelocKerry_G live on webcam... -> http://stream.sglug.org:8800 (seriously)
02:49.44mitcheloc** Kerry_G live on webcam... -> http://stream.sglug.org:8800 (seriously)
02:49.46mitcheloc** http://stream.sgvlug.org:8800/
02:49.54mitchelocheh, clumsy me
02:50.38Snake-Eyes....
02:50.43*** join/#asterisk rrittenhouse (n=tad@24.55.244.254)
02:50.52[hC]not only may i say, "wtf" - but your link is dead.
02:51.36mitcheloc[hC]: use the last one, i missed the "v" twice in a row *doh*
02:52.15[hC]Guess it doesnt work on safari
02:52.47[hC]what is it anyways?
02:53.30mitcheloc[hc]: supposedly, you need a proper video player, like VLC or videolan, KG is doing a presentation on Asterisk to the LUG @ CalTech
02:54.54[hC]who is kg?
02:54.58*** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
02:55.05*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
02:55.17[hC]they're doing slides on freepbx right now
02:55.34mitcheloci agree....eww
02:55.43mitcheloche runs www.voipspeak.net
02:57.09*** join/#asterisk nortex (n=barracud@adsl-68-93-160-132.dsl.amrltx.swbell.net)
02:57.44[hC]he
02:57.47[hC]'he's not very funny
02:57.52[hC]and seems like a douchebag.
02:58.29mitchelocif you can do better you are welcome to
02:58.50*** join/#asterisk |dennis| (n=dennis@200.32.215.82)
02:59.46[hC]are you there?
03:00.45mitcheloc[hC]: yes
03:01.05fileam I Supposed to get audio with this?
03:01.14[hC]mitcheloc: are you in the frame?
03:01.19[hC]theres audio yeah
03:01.23[hC]im using vlc
03:01.23mitchelocfile: i think so, yes, join #sgvlug and ask
03:01.36[TK]D-FenderKG promotes Linksys & Trixbox a wee bit too much...
03:02.16mitcheloc[TK]D-Fender: he did a disclaimer before he started =P
03:02.22[TK]D-Fender:O
03:02.32[TK]D-FenderI'm trying to get in through VLC and failing..
03:02.45[hC]mitcheloc: are you the ibm laptop in the bottom left? i think i see bitchx
03:03.40mitcheloc[hC]: i  don't know, just a sec
03:04.04[hC]wow, a girl's there? haha
03:04.51mitcheloc[hC]: a couple, why are you being so negative?
03:04.56nortexWhat are ya'll up to?
03:05.28[hC]im not being negative... Didnt mean it in a negative way
03:05.30QwellWhat is this stream?
03:05.33mitchelocnortex: Kerry_G is webcasting, join #sgvlug and read the chan topic for the url
03:05.35[hC]Just surprising, not a lot of girls go to LUG meetings.
03:05.48mitchelocI'll wanr you all that he is doing a fair bit on TB/FreePBX
03:05.53mitcheloc* warn
03:05.59QwellThat stream quality is complete crap...
03:06.11dlynes_laptopno kidding
03:06.14dlynes_laptopand fwiw
03:06.28dlynes_laptopit works if you have mplayer plugin for firefox
03:06.32mitchelocQwell: probably because 90% of the people here in the room are watching it at the same.as seeing it live
03:06.41Qwell:p
03:07.06[hC].... why would you do that??? :P
03:07.13*** join/#asterisk AJaymn (i=AJaymn@70.59.126.198)
03:07.22*** part/#asterisk foo (n=foo@unaffiliated/foo)
03:07.24dlynes_laptop[hC]: and that's surprising?
03:07.34Qwellsomebody needs to shout something obscene
03:07.35mitcheloc[hC]: i don't understand the reasoning either as i'm not doing it
03:07.44mitchelocQwell: haha
03:07.45dlynes_laptop[hC]: have you been to any of the vanlug meetings?
03:07.53[hC]dlynes_laptop: a long time ago.
03:08.00[hC]I presented at one
03:08.21dlynes_laptopman, a lot of those guys are such a bunch of children
03:08.35dlynes_laptopmajor fanaticism
03:08.48[hC]yep, with nothing better to do than sneer over linuxy things.
03:08.48[hC]its the biggest nerd fest ive ever seen.
03:08.50dlynes_laptopexactly
03:09.03dlynes_laptopno realism; only linux fanaticism
03:09.07[hC]yep.
03:09.23Qwell"Programming"...PFFT
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03:10.20dlynes_laptopbtw
03:10.31dlynes_laptopwhere's 'sgv'?
03:10.43[hC]san gabriel valley
03:10.48Qwellhmm
03:10.48dlynes_laptopcalifornia?
03:10.49[hC]california
03:10.51mitchelocif anyone has questions for Kerry, I can go ahead and ask him? i.e. "why not use conf files?" would be a great one :)
03:10.51[hC]yeah
03:10.52dlynes_laptopah
03:11.02mitchelocit's near so-cal, "caltech"
03:11.40Qwellhey, I am in sgv :D
03:12.12[hC]"Would you say that trixbox is like, Asterisk for dummies?"
03:12.13[hC]:P
03:12.19Qwell~thebook
03:12.21jbothmm... thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
03:12.36mitchelocKerry said yes
03:12.37QwellAsterisk for...idiots
03:12.40[hC]HA!
03:12.46Qwelltrolling, kthx
03:13.02[hC]I didnt think you;'d actually ask :) teehee
03:13.42mitcheloci've only got 19 minutes of batter life left... *sigh*
03:13.53Qwellwoo, nano!
03:14.02[hC]I edit config files with a hex editor
03:14.11[hC]Ask if he edits extensions.conf with windows notepad
03:14.12[hC]:P
03:14.13[hC]j/k
03:14.41mitcheloci've actually used notepad...go figure
03:14.53Qwelljust unix2dos and dos2unix it
03:14.54[hC]what is a DID Alert Info? :P
03:14.55*** join/#asterisk tlow (n=tlowe@voip.terrorist.net)
03:14.59nortexSame here
03:15.22nortexSame as SIP Header I think.
03:15.23mitchelocthere you go :)
03:15.36QwellYou should mention that you're getting questions from IRC :p
03:15.37*** join/#asterisk laurence (n=laurence@DHCP-112-206.caltech.edu)
03:15.42Qwelland that Qwell says "w00t"
03:16.13mitchelocKerry carpooled with me, i'm sure he knows i'm not asking these questions...
03:16.17Qwellheh
03:16.22rrittenhouseI bet you get this a lot. I want to mess with asterisk pretty bad. I have no phone line coming in nor do i have any sip phones haha. What *could* i technically do with it (just curious)
03:16.23mitcheloc;)
03:16.30Qwellsnow-m?
03:16.34[hC]You may note that the questions im asking are coming all the way from san jose costa rica today
03:16.34[hC]:P
03:16.37*** join/#asterisk Dico_ (n=niko@60.51.217.61)
03:16.47[hC]Eek, stream stalled.
03:17.04nortexrrittenhouse: You could download a sip softphone and get an account with an ITSP
03:17.09rrittenhouseah
03:17.38rrittenhousewell time warner just took over our adelphia and im thinking about getting their "digital phone" if its worth it
03:17.44[hC]Is he talking about the spa941"?
03:17.45Qwellrrittenhouse: don't do it
03:17.49Qwell[hC]: yeah
03:17.51rrittenhousebut i need to look into an ITSP
03:17.54Qwellor 841?
03:17.55[hC]they're really not that good. :(
03:17.59Qwellnope...
03:18.03[hC]They're echoey
03:18.07[hC]crackly
03:18.08rrittenhouseoh man :(
03:18.17[hC]For the price, the polycom ip430 blows it out of the water.
03:18.22Qwellso I hear
03:18.37[hC]aw man.. my stream stalled right before he answered the coolness factor question
03:19.31[hC]Its not good for your eyes to work in a dungeon
03:20.01Qwellchan_skinny has unlimited softkeys
03:20.08QwellThank you very much
03:20.15[hC]ooh chan_sccp was abaondoned somewhat recently
03:20.23[hC]Im going to have to switch to chan_skinny after al
03:20.29[hC]that is, if i can find enough friggin time
03:20.45mitcheloctime, huh?
03:20.50[hC]........ he didnt like polycom?!?!
03:20.57Qwellpolycom sidecars
03:20.59[hC]aw man
03:21.00[hC]snom
03:21.01Qwellshort o..
03:21.13dlynes_laptopheh
03:21.18dlynes_laptophe's complaining about polycom
03:21.23dlynes_laptopand raving about linksys
03:21.24dlynes_laptopwtf?
03:21.31Qwelldude
03:21.32[hC]hello blackberry messenger
03:21.32dlynes_laptopoh....and now loves grandstream
03:21.34Qwellthere are like...5 people there
03:21.38nortexThere is not even a Poly on the desk
03:21.43dlynes_laptopwtf?
03:21.46Qwellmitcheloc: raise your hand :p
03:21.46dlynes_laptopTHE GRANDSTREAM SUCKS
03:21.52Qwellnon-chalant(sp)
03:22.00fileMyFirstVoIPPhone(tm)
03:22.03mitchelocI'm actually not in the picture
03:22.06Qwelllame
03:22.14dlynes_laptopwoah
03:22.17dlynes_laptopthe picture's clear now
03:22.21mitchelocsee the guy in the bottom right? i'm just behind him to the right
03:22.29Qwellscoot over, heh
03:22.38[hC]as far as what?
03:22.45[hC]oh asterisk
03:22.51rrittenhouseany suggestions on a softphone to mess with asterisk  (under linux)
03:22.58phifli_was anyone here at the con in europe
03:23.00[hC]what sorts of problems do polycoms have
03:23.01[hC]ask him that
03:23.05[hC]whats wrong?
03:23.10Qwellumm
03:23.13[hC]it has blf
03:23.14Qwellpolycom HAS blf...troll
03:23.20nortexrrittenhouse: Check out xlite from counterpath.
03:23.20Qwellcall him on it ;/
03:23.22mitchelocokay, i got a short battery life left
03:23.25filetrolllllllll
03:23.31rrittenhousenortex, thx
03:23.32mitcheloci'm going to sleep my laptop, come on in a bit and i'll wave at you all ;)
03:24.05nortexmitcheloc: later
03:24.17Qwell~blf
03:24.18jbotsomebody said blf was Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing.  hint extensions are static mapped to SIP or other channels.
03:24.18dlynes_laptopwtf?
03:24.18[hC]ask ummmmmm
03:24.22Qwelldude
03:24.24[hC]blind lamp function?
03:24.24QwellCALL HIM ON IT
03:24.25dlynes_laptopnow he's saying he'll use polycoms
03:24.41dlynes_laptophe says they suck, but he still uses them
03:24.43dlynes_laptopwtf????
03:25.25Qwelllaurence: yell at him, please
03:25.31Qwell"Busy Lamp Field"
03:26.02dlynes_laptopnow buddy in the green shirt's picking his nose on national tv
03:26.03dlynes_laptopwtf?
03:26.07Qwellhaha
03:26.09[hC]ask if he knows, will asterisk show my receptionist if line1 is in use, or line2?
03:26.11[hC]:P
03:26.26[hC]I think buddy in the green shirt earlier asked a question
03:26.32[hC]and i thought he was a 11 year old girl in the audience
03:26.34[hC]im not sure.
03:26.49dlynes_laptopand it's 'neesh', not 'nitch'
03:26.50dlynes_laptopsheesh
03:26.54nortexHe waived at the camera a bit ago
03:27.18Qwellooo, I like that PROXIMA fade in
03:28.07nortexA grand for a killer server???
03:28.31[hC]erm
03:28.37[hC]dell sc430 = approx 450 bucks
03:28.52nortexWould you call that a killer server?
03:28.59[hC]p4 2.6ghz with 1gb ram and 120gb sata drive
03:29.04[hC]its pretty damn good for the price.
03:29.25DaminWhere is the video stream for this presenation?
03:29.27Qwellwithout what?
03:29.28Qwellhttp://stream.sgvlug.org:8800/
03:29.56[hC]a 'restart now' works even quicker.
03:30.00Qwellomfg
03:30.01[hC]or, restart when convenient'
03:30.06Qwell"wait for no calls"
03:30.10DaminWhat the hell is it streaming with?
03:30.16QwellDamin: mplayer?
03:30.27dlynes_laptop[hC]: i thought a restart now has a potential for memory leaks?
03:30.28Qwellor, the server part?
03:30.30[hC]Damin: a Logitech Quickcam from 1994.
03:30.45laurenceDamin: You mean the codec?  Ogg/Theora.
03:31.00[hC]Hahaha.
03:31.01laurenceOr leave off the port and you get a java applet.
03:31.01[hC]'Wife'
03:31.04[hC]Have you seen the audience?
03:31.09[hC](no offense mitcheloc)
03:31.20[hC]These guys arent out of school yet!
03:31.21JTi'd want at least raid hard drives and a redundant power supply in my killer asterisk server
03:31.34Qwellask about e911
03:31.38Qwellwith voip
03:31.53[hC]oh ive got a good one
03:31.55filehaha
03:32.03[hC]"Ive got a Cisco 7970, but i dont have the firmware for it, can i get a copy from you?"
03:32.07Qwellheh
03:32.14[hC]:P
03:32.17[hC]</troll>
03:32.18nortex:P
03:32.40dlynes_laptopwhat's his nick on irc?
03:32.48Qwelldlynes_laptop: asterisk_nub, I imagine
03:32.53dlynes_laptopheh
03:32.56dlynes_laptopwoah
03:32.56[hC]Kelly_G or something
03:33.10dlynes_laptopAlmost at the end of the presentation, he finally mentions digium cards
03:33.14dlynes_laptopwtf?
03:33.25[hC]So far i still refuse to use digium cards
03:33.35[hC]maybe after this round of financing, they'll step it up
03:33.36dlynes_laptopheh
03:33.39nortexOtherwise he might have to explain fxo and fxs to then
03:33.41[hC]Ooh
03:33.43Qwellha
03:33.45dlynes_laptopxed2 admits to being the guy who picked his nose in the front
03:33.49dlynes_laptopsmart
03:34.13[hC]Ask him 'how can i solve the issue of my internal office network being saturated, say by downloads or a virus, and killing my phone calls? ie QoS..
03:34.19[hC]especially when you're using the phone as a switch
03:34.19[hC]:)
03:34.31mitcheloci asked about 911.... watching the room over laurences shoulder.... ;)
03:34.40Qwellmitcheloc: he completely avoided the question
03:34.56mitchelocmaybe he is leading up to it
03:35.02mitchelocgot to sleep laptop...
03:35.07[hC]chow
03:35.18[hC]Ask about sangoma :)
03:35.25[hC]hee hee
03:36.24Qwelland if your net connection is down, then what?
03:36.39[hC]Hah, or 'hey so ive got trixbox installed here but i cant get my xlite to register, can you help me?'
03:36.42[hC]:P
03:36.43DaminSPA 3000 lets you do 911 pass through..
03:36.55QwellDamin: true, if you have an analog line
03:37.00Qwellbut, he's talking specifically about SIP
03:37.02QwellITSP
03:37.06DaminIt also has a no-power cross connect between the FXO and FXS ports..
03:37.20DaminSo if you don't have power, it bridges the two..
03:37.26QwellDamin: yeah
03:37.33dlynes_laptopDamin: and a no-network cross connect too
03:37.51dlynes_laptopDamin: epygi's also do that
03:37.53[hC]ask him how much cpu per call
03:37.58Damindlynes_laptop: So that if the network is down it hardwires the ethernet to the FXS? :)
03:38.17dlynes_laptopDamin: if the network is down, it hardwires the fxs to the fxo
03:38.26dlynes_laptopDamin: also known as a lifeline
03:38.52Damindlynes_laptop: Hehe.. I know.. I was just messing with you..
03:39.19dlynes_laptopDamin: dood...you weren't messing with me...you were messing your pants...and now you stink
03:39.22dlynes_laptopDamin: get away
03:39.51[hC]Haha
03:40.32Qwellyeah, because emergencies don't happen during the summer when it's hot
03:40.35Qwell...
03:40.48Idlethey dont
03:40.53[hC]wonder what his biggest install is
03:41.00nortexAnd a hour of being down is no big deal :)
03:41.19Daminnortex: Unless you are a phone-sex provider..
03:41.21nortexHe said something about 90 phones
03:41.31Qwell"couple grand"?
03:41.34Idlehow can you cluster asterisk?
03:41.35[hC]Whhhhhattt
03:41.39QwellNO!
03:41.42Qwell1500
03:41.44[hC]This is why i buy sangoma.
03:41.45nortexOr you company likes to make sales by phone.
03:41.48DaminIdle: He meant "cluster fuck"
03:42.01[hC]a102d = $600 bucks.
03:42.05Idlewhat? I was actually asking how you would... :S
03:42.13[hC]THANK you
03:42.18Qwellheh
03:42.31dlynes_laptopnow he mentions sangoma :)
03:42.39[hC]...but dont use them, because that would be a travesty...
03:42.46dlynes_laptopno
03:42.54dlynes_laptopdon't use them, because file's heart will be broken
03:43.04[hC]er..
03:43.05[hC]./Setup install
03:43.07[hC]is pretty easy
03:43.08[TK]D-Fender[hC] : EC version of 102?
03:43.08[hC]:)
03:43.13[hC][TK]D-Fender: yeah!
03:43.24[TK]D-Fender[hC] :news to me.. going to look for now....
03:43.29[hC]I realized not too long ago I have a non EC a102
03:43.32[hC]and want to replace it
03:43.34[TK]D-Fender600 is unbelieveably cheap
03:43.43[hC]I got the price from my sangoma rep
03:43.48Qwellrhino channelbank does rock, I hear
03:43.52[hC]they announced the card a while ago but i couldnt find anywhere to buy it on line.
03:43.59[TK]D-Fender[hC] : thats the price of the NORMAL one...
03:44.13Qwellit IS NOT $2k
03:44.14nortexhC Voipsupply had it last I checked.
03:44.26fileI don't care what you use :P
03:44.28[hC][TK]D-Fender: I may be ~$100 off, I just remember it was around the 600 mark
03:44.30fileuse what works for you!
03:44.32Qwellbah, sissies need to call him on it
03:44.46Qwelllaurence: That means you :P
03:44.50[TK]D-Fender[hC] : I don't care if its $1000......
03:44.59dlynes_laptopor mitcheloc
03:45.04[TK]D-Fender[hC] : its much cheaper than 1/2 the A104D
03:45.12dlynes_laptopcome on mitch!!!
03:45.16Qwellgo up there
03:45.29nortexmitch's laptop went to sleep
03:45.37filelol
03:46.19[TK]D-Fender[hC] : the 102u goes for 770$
03:46.28[TK]D-Fender[hC] : I want to see it in print :)
03:46.35[hC]Where did you find the price?
03:46.45[hC]I get -25% from sangoma directly
03:46.46[hC]im a reseller
03:46.55dlynes_laptop[hC]: same here
03:47.03dlynes_laptop[hC]: i would never pay $770
03:47.23dlynes_laptop[hC]: about $550-600USD for a 102u
03:47.27[TK]D-Fender[hC] : Atacomm.
03:47.40[TK]D-Fender[hC] : I can't find the A102d listed anywhere...
03:47.54dlynes_laptop[TK]D-Fender: because it doesn't exist
03:48.07dlynes_laptop[TK]D-Fender: the a101d/a102d don't exist
03:48.25[hC]102d does infact, it was announced the same time as the 104d
03:48.26dlynes_laptopYET
03:48.32[hC]I just ordered one
03:48.36dlynes_laptopeh?
03:48.51dlynes_laptopwhy'd they tell me it doesn't, when I talked to my rep?
03:49.03filedlynes_laptop: you talked to someone in another universe
03:49.16dlynes_laptopfile: no...I talked to someone in the sangoma office
03:49.16[hC]when did you ask?
03:49.20[hC]I ordered 4 days ago
03:49.24[hC]although
03:49.24[TK]D-Fender[hC] : yeah I see recent references, but no official product page anywhere.  So its pending.
03:49.25dlynes_laptopabout 1-1/2 months ago or so
03:49.33DaminCreepy?
03:49.35[hC]I didnt do it personally, my hardware ordering dude did
03:49.41[hC]so I may need to confirm with him
03:49.49dlynes_laptop[hC]: umm...announced the same time as the 104d?  the 104d's been out for a while
03:50.13[hC]dlynes_laptop: it was announced at the same time, not released at the same time.
03:50.33[hC]is that mitcheloc?
03:50.34dlynes_laptop[hC]: ah...well, the a200 hwec was announced a while ago, too
03:50.38*** join/#asterisk niZon (n=ilt@S0106beefd4cecc3d.wp.shawcable.net)
03:50.41dlynes_laptop[hC]: but it didn't come out until June
03:50.46[hC]Yeah.
03:50.54laurence[hC]: It's not him.
03:50.55dlynes_laptopsame for the a108d
03:50.59*** join/#asterisk CoffeeIV (i=rgr@rrcs-67-79-2-146.sw.biz.rr.com)
03:51.15[hC]Blackberry can play wav
03:51.17CoffeeIVwhat does this mean: "Unable to find a codec translation path from g729 to slin"
03:51.19[hC]8700 can anwyays.
03:51.30fileCoffeeIV: your Asterisk install can't transcode G729
03:52.27laurenceQwell: I started nodding off, I'm not awake enough to ask your questions.  Plus I'm not feeling particularly mean, so I'm not going to "call him" on it.
03:52.35DaminCoffeeIV: It means that asterisk was unable to find a codec translation patch from g729 to Signed Linear.
03:52.37*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:52.41Qwellwell, he's giving incorrect information...
03:52.44Qwelland that isn't cool
03:52.50CoffeeIVfile: does that mean that inorder to handle calls, I need to either get g729 or make my provider send me calls in another codec ?
03:53.06DaminHe's freaking stealing my lines..
03:53.12dlynes_laptophehe
03:53.16phifli_haha
03:53.20DaminI use that line in my Intro to Asterisk presentations..
03:53.29QwellDamin: which?  "lowered the bar"?
03:53.36[hC]Qwell: haha.
03:53.37[hC]um
03:53.37fileCoffeeIV: either both sides need to use the same codec (g729) or you need to not use g729, or buy a license
03:53.40Damin"Thank God for cell phones, because they have lowered expectations"
03:53.43Qwellyeah
03:53.43[hC]vonage is not pronounced that wya
03:53.47CoffeeIVok
03:53.47[hC]for the love of god
03:53.50Qwellheh
03:54.04Qwelleven $900B in advertising can't get people to pronounce it right
03:54.08DaminDid he say "Vohnaj"?
03:54.15phifli_haha
03:54.17QwellDamin: Voh-naj-ee
03:54.45[hC]vonajh
03:55.02DaminWho is this guy?
03:55.08QwellDamin: some troll :p
03:55.16Qwellhe's telling like...half-truths
03:55.20DaminNo.. the Svlug guy..
03:55.20[hC]he runs voipspeak.net
03:55.34[hC]so, some nerd who has been using trixbox for 6 mos
03:55.36[hC]:)
03:55.40sumais there is any IAX FXO device available ?
03:55.58[hC]get outta here.
03:56.00[hC]gpl.
03:56.13[hC]yes because im sure customers give a shit about GPL.
03:57.01laurence[hC]: Chill.  He's saying that because it's a LUG.  It's a running joke for this talk. :-)
03:57.08[hC]I know :)
03:57.14Daminsuma: Not sure.
03:57.30Daminsuma: the Iaxy is just FXS..
03:57.51sumaDamin: yes, i'm familier with that, but looking for a FXO device
03:57.53Daminsuma: But I think that there was an australian company that did an IAX ata w/ FXo..
03:58.16sumaDamin: you got their name ?
03:58.20Qwelluntil the end of the year or so
03:58.59[hC]chan_twocansandstring.so
03:59.20[hC]I use it every day! :)
03:59.46[hC]I find it really ironic how he was doing the GPL joke at the LUG, yet his laptop runs winxp. :)
03:59.51Daminsuma: I'm looking..
03:59.53dlynes_laptopwho's the doofus up there speaking, now?
03:59.58dlynes_laptopis it mitcheloc?
04:00.04Qwellno
04:00.06[hC]apparently not
04:00.12laurence[hC]: See?  It's ironic! :-)
04:00.26dlynes_laptopoh
04:00.28dlynes_laptopit's laurence
04:00.35laurenceThe "doofus" is Matti.
04:00.39dlynes_laptophehe
04:00.44dlynes_laptopah...the asterisk developer?
04:00.54Qwellhuh?
04:00.57laurenceYah, I'm typing telepathically while you see me standing there. :-)
04:01.14dlynes_laptopnvm
04:01.29Daminsuma: It was virbiage: http://www.virbiage.com/3010.php But it is only an FXS device.
04:01.31dlynes_laptopi thought there was somebody in #asterisk-dev with the nick matti or something like that
04:01.54QwellYou can use vonage with asterisk...
04:02.00QwellIt's...SIP
04:02.24DaminQwell: But why would you want to? :)
04:02.31QwellDamin: he's saying you can't
04:02.51*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
04:03.13Qwellhaha
04:03.54QwellBS!!
04:04.04Qwell"Every cell provider"
04:04.27[hC]damn, what did i miss? I went to appease my wife
04:04.47fileQW
04:04.49Qwell[hC]: "You can set your cidnum as the callee, and get right into their VM.  It works with every cell provider"
04:04.50fileQwell: !!!!!!!!!!!!!!!!
04:04.56[hC]Hah
04:05.01[hC]Maybe in 1998
04:05.24Sedoroxsprint it does
04:05.27Sedoroxverizon I know doesn't
04:05.28Sedoroxnor nextel
04:05.30QwellSedorox: No it doesn't
04:05.31Sedoroxnor cingular
04:05.33Sedoroxhehe
04:05.34Qwellnot sprint
04:05.40Sedoroxwell thats what I was told
04:05.43SedoroxI haven't tried it myself
04:05.47QwellI have :)
04:05.53Sedoroxhehe so I'll go wirh your word then
04:06.07Sedoroxtmobile is dumb enough to do that
04:06.10laurenceQwell: So what date shall I put you down for your own Asterisk talk?
04:06.15[hC]tmobile does not
04:06.16[hC]i tried
04:06.19Qwellsome (most) providers, however, DO check "in network calling" via cidnum
04:06.19[hC]cingular does not
04:06.20Sedoroxlol
04:06.21Qwelllaurence: schedule me
04:06.30[hC]they all check ani now
04:06.30Qwelllaurence: I'm right here in wsco
04:06.44Qwellclaude...umm...who?
04:06.52[hC]junky?
04:06.53Qwellklaud, even?
04:06.54laurenceQwell: Easily done, the schedule is pretty open.  It's a popular subject, we could just wait a couple of months and it would go over big.
04:06.56Qwell[hC]: no :p
04:07.05Qwelllaurence: I'm only in wsco until the end of Oct or so
04:07.19Qwellwait, no, person I'm thinking of is "Klause", heh
04:07.27laurenceQwell: Are you close enough to Pasadena to join us at BC?
04:07.33Qwelllaurence: wsco..
04:07.52laurenceQwell: mitcheloc wants you to come buy.....
04:08.06laurencewsco?
04:08.09QwellWhat, tonight?  heh
04:08.12QwellWest Covina..
04:08.47laurenceQwell: Ah.  Well, go down the 210, Get off at Lake, go South to California.  We're often there until 1AM talking on the sidewalk after BC closes. :-)
04:08.55QwellBC?
04:09.07laurenceBurger Continental.
04:09.13Qwellnever heard of it
04:09.27laurenceIn spite of the name it's kind of middle-eastern.
04:09.50laurenceQwell: So, are you going to come by?
04:09.53[hC]haha. oh yes, security people dont use email.
04:10.07Qwelllaurence: another time, sure.  I don't drive :)
04:11.02laurenceQwell: Email me your contact info so we can set up a talk: dustin@laurences.net.
04:11.26filemoo
04:11.36[hC]ahh, internet taking a crap.
04:11.56Qwelllaurence: see msg
04:12.05Qwelland, keep that to yourself for about...2 days :)
04:12.22Qwell(between the two of you)
04:15.39QwellWTF?!
04:15.47QwellNo drivers for debian?!
04:15.51QwellIT'S LINUX
04:15.59Qwelllaurence: I rest my case. :)
04:17.35mitchelocokay real quick i'll wave haha
04:17.48Qwellnub
04:17.51[hC]mitcheloc was the bald guy?
04:18.03mitchelocis it still on?
04:18.06mitchelocno i'm not balk!
04:18.07Qwellyeah
04:18.08mitcheloc* bald
04:18.09[hC]yep
04:18.13mitchelocqwell you can see me then?
04:18.15Qwellyep
04:18.20[hC]you're kinda kevin rose lookin
04:18.20Qwellcan you see us?
04:18.30fileyay
04:18.43mitcheloci'm in the bottom right
04:18.44mitchelochaha
04:18.51mitchelocanyways battery is seriously low
04:18.52[hC]ooh that girl looks like she might be kinda cute
04:18.53Qwellit's off
04:18.56mitchelocQwell: too bad you can't make it tonight
04:19.02Qwellcome get me :P
04:19.04[hC]im still streaming
04:19.08mitchelocQwell: i can
04:19.10[hC]ah now it died.
04:19.16Qwellmitcheloc: I wish, heh
04:19.21Qwellwork tomorrow ;/
04:19.25mitchelocQwell: pmed my cell if you change your mind
04:19.27filelast day!
04:19.27Qwelland my toe is broken, heh
04:19.53Qwellindeed :)
04:20.07shido6erf?
04:20.08mitchelocQwell: don't be too anti-kerry, he may not have had everything right, but he still presented most of the basics ;)
04:20.22Qwellmitcheloc: I'm not..  just wish he wouldn't have misstated :)
04:20.49Qwellmitcheloc: PM me that # again
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04:28.51QwellWhat? ;/
04:28.55*** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home)
04:28.55*** mode/#asterisk [+o Corydon76-home] by ChanServ
04:28.56filewazzzzzzzup
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05:07.15Snake-Eyesany one want to have another crack at softhangup problem I have :)
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05:25.36x86maybe if someone had awesome telepathy they could know what the problem was, and if you were lucky enough, they might have the answer too... until then, you probably wont get much help ;)
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05:48.07Snake-Eyesha ha ha
05:48.54Snake-Eyeswell now that i have baited you :P
05:52.46Snake-Eyesx86, this  now the 3-4 time ive asked (loosing count now). Is softhangup ment to hangup all calls on a trunk ? eg I have macro that calls softhangup and doesn't seem hangup any other calls on the SIP trunk when its called  (tried SoftHangup(SIP/trunk-sx-1) SoftHangup(SIP/trunk-sx|a)
05:55.13hads|homeAccording to 'show application softhangup' that's how it should work
05:55.41Snake-Eyeswell its not .... :(
05:56.23Snake-Eyesmog was saying something about sub channels but he went to sleep and i couldnt get any more out of him
05:58.43Snake-Eyeshads|home, any ideas ?
06:03.33hads|homeNot really, looks like it should work.
06:06.16x86why not just use Hangup()
06:07.44*** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
06:08.24rushowrhey all, it's me again :) anyone in here have a good understanding of macros, particularly about variable inheritance?
06:09.12rushowrI've got macros calling macros, which in turn sometimes call other macros. It doesn't always work, and I'm figuring it's due to inheritance of variables
06:09.17rushowrbut looking for confirmation
06:09.33rushowrand yes I've read the Asterisk Macros page on voip-info
06:11.37Snake-Eyesx86, I don't want to hangup the call, I want to hangup another call/channel on the trunk
06:11.42Snake-Eyeshttp://pastebin.ca/126159
06:12.50hads|homeSnake-Eyes: You should get a warning message "Soft hanging %s up." for each channel
06:14.04phifli_set verbose 9999
06:14.05phifli_;)
06:14.13hads|homeAlso, regarding your dialplan logic, what if the call on that trunk is already an emergancy call? You will kill their call.
06:14.17rushowr•phifli_• who?
06:14.22phifli_do that on asterisk
06:14.25phifli_set debug 999
06:14.26phifli_lol
06:14.28phifli_if you wanna see all messages
06:14.30Snake-Eyeshads|home, not getting that warning
06:14.39rushowr•phifli_• I take it you're not talking to me......
06:14.57phifli_speaking in general to see error messages
06:14.58Snake-Eyeshads|home, that dial plan is for testing, ill change it later once i know softhangup works
06:15.05phifli_has|home said "Soft haing up o %s"
06:15.06phifli_whatever
06:15.07*** join/#asterisk _omer (n=_omer@202.38.51.2)
06:15.16rushowrah, yeah not me :)
06:15.26_omerhi
06:15.29rushowrI guess no one knows about the macro thing
06:15.46Snake-Eyesphifli_, ok
06:16.08rushowr•Snake-Eyes• Softhangup works
06:16.16hads|homeSnake-Eyes: Well does 'show channels' show a SIP/back-trunk-ulaw-1 channel?
06:16.20_omerhow do I know if the number or DTMF I am getting is in E.164 standard??
06:16.32rushowrtest it
06:16.55rushowr•_omer• the newer trunk revisions have the function REGEX
06:17.20_omerREGEX ???
06:17.22rushowr•Snake-Eyes• I use Softhangup constantly, it works
06:17.34Snake-Eyeshads|home, yes but i double check in a sec with debug set nice and high
06:17.41_omerlet me check wiki ...
06:17.42rushowr•_omer• As in Regular Expression, for testing your data
06:17.43Snake-Eyesrushowr, ok
06:18.09Snake-Eyesrushowr, can you give me example of your working softhangup?
06:18.20rushowr•Snake-Eyes• Softhangup(${CHANNEL})
06:18.32rushowrSE, hangs up the current channel
06:18.48_omerrushowr : I just meant to say...if given number is with COUNTRYCODE or without COUNTRY CODE..
06:19.20rushowr•_omer• test the length of the var, or maybe pass it to a context...thgere's many ways
06:19.26rushowr•sevard• Softhangup(${chan_to_hang}) would hangup whatever channel is spec'd
06:19.36rushowroops, I meant that for Snake
06:20.10Snake-Eyesrushowr, is there a way to hangup the first channel being used?
06:20.26rushowr•Snake-Eyes• hrm... explain what you mean?
06:20.34_omerif number is for United Arab Emirates .....and second number is for UK ....then shouldn't be same..:-/
06:20.34rushowr•Snake-Eyes• by first channel being used
06:21.01rushowr•_omer• there's soooo many ways to test the value of the DTMF
06:21.29rushowr•_omer• for instance: (I use AEL2, so bear with me)
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06:21.39Snake-Eyesrushowr, 1st person places a call then a 2nd person places a call, when 3rd makes a call using the macro the 1st persons call is hangup
06:21.48*** join/#asterisk neo (n=neo@kessel.ordrejedis.net)
06:21.51phifli_well use channel macros rather than global
06:21.58neohello :)
06:21.59phifli_read the newest README on macros
06:22.10*** join/#asterisk m_a_g_o (i=maxgluck@201.243.102.189)
06:22.18rushowr•Snake-Eyes• basically you just gotta pull the value of the channel name for the one you want to kill
06:22.31rushowr•Snake-Eyes• haven't ahd to do it yet, so don't have a fast answer on that
06:23.18rushowr•_omer• basically, I'm not trying to be an ass, but read up on asterisk dialplan functions on voip-info it's all there, all the crap you need to use for testing values
06:23.20m_a_g_ogood evening folks, I just updated the SVN version and suddenly I'm getting this messages when loading the g729 and g723 modules... Segmentation fault (core dumped)... anyone has a clue, any idea?
06:23.35rushowr•m_a_g_o• yeah I hear there's a prob with 'em
06:23.42phifli_heh
06:23.42rushowr•m_a_g_o• unfortunately that's all I know
06:23.50phifli_i'd hack up the source if i had a system to do it on haha maye another day ={
06:23.51phifli_=P
06:24.08rushowr•phifli_• *chuckle*
06:24.14Snake-Eyesrushowr, damn, i was hoping '|a' would do it
06:24.18JTman, what is with your nick addressing, rushowr?
06:24.27m_a_g_orushowr... :$
06:24.29neohum, has one of you already tested a Load Balancing Solution for Asterisk ?
06:24.30rushowr•JT• I've got some old IRC script running
06:24.50JTit looks like an inverse colour U is on each side of the nick
06:25.04rushowrover here it's white dot on the black backgroun
06:25.17rushowrthanks for the heads up though, I'll play with it
06:25.35JTwell i have a white on black console
06:25.39rushowrlol
06:25.46rushowrthat's fun, eh?
06:25.46m_a_g_orushowr, do you know how I can get the curl function with 1.2?
06:25.50neosame here :)
06:25.55JTyeah, standard console
06:26.04Qwellm_a_g_o: install curl-dev
06:26.05JTi'd wager that the majority of the channel does :P
06:26.09rushowr•m_a_g_o• hrm...have you tried asterisk-backports?
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06:26.12rushowrthe site's great
06:26.14rrittenhousei'm still trying to get xlite to connect to asterisk haha.
06:26.18Snake-Eyeshads|home, btw I will eventually have emergency calls on a seperate trunk but both trunks going to the same machine thus killing random calls on trunk A wont effect calls on emergency trunk :)
06:26.19rushowror at least the idea
06:26.23phifli_xlite works with asterisk
06:26.26phifli_make sure you change the auth user
06:26.38rrittenhouseyeah but ive never messed with asterisk before and im just kinda trying to get the hang of whats what
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06:26.51rushowrwell, anyway mates, I gotta run and get back to this code, have a good one :)
06:27.05m_a_g_orushowr: first time I hear about it... could you give me the address?
06:27.07rrittenhousei see you add a "friend" in the sip.conf and then an extention in the extensions.conf or something.....
06:27.13rushowrone sec mGO
06:27.14rushowrmago
06:27.22phifli_you can add em as user if you want
06:27.29phifli_just make sure you turn off incoming calls :)
06:27.45phifli_or you can add user/peer separately to get more contorl over it
06:27.47phifli_control
06:27.52m_a_g_oQwell, thanks, but could you point me where I can download the source?
06:27.57phifli_but friend is easier for not customizing the setup
06:28.00neosounds like nobody knows sip load balancing :'(
06:28.02phifli_www.asterisk.org
06:28.03rushowrasterisk-backports.org
06:28.04Qwellm_a_g_o: What distro are you on?
06:28.12m_a_g_ofc3
06:28.13rrittenhouseholy crap i got it :P
06:28.18Qwellyum install curl-devel
06:28.19rushowrcheers
06:28.48Qwelland obviously curl needs to be installed too
06:28.52rrittenhousewoo this is ..neat now that i got a user logged in hahah
06:29.32phifli_lol
06:29.38phifli_you'll get used to it =P
06:29.45m_a_g_oQwell: thx, installing now...
06:30.17neoQwell: sip load balancing, do you know something about it ?
06:30.36Qwellneo: could use something like ser
06:31.54neohum, yes
06:32.06neoi would try something more like IPVS
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06:35.01Snake-EyesI see why rushowr has rush in his name lol
06:37.46phifli_whys that
06:37.57phifli_owr should be hour =/
06:38.00phifli_even our haha
06:38.13phifli_gnite
06:39.54dlynes_laptop~seen kerry_g
06:39.57jbotkerry_g <n=Kerry_G@ip70-187-129-227.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 5d 5h 50m 41s ago, saying: 'no, you have spent 4 hours begging people to fix the problems you made'.
06:40.09Qwellheh
06:40.40Juggie~seen theplot
06:40.41jbottheplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 4d 12h 41m 50s ago, saying: 'I did set in the address field to match the username too'.
06:40.50Snake-Eyesnight phifli_
06:41.15phifli_take it easy
06:41.26Snake-Eyeshe rushes in and out of the channel asking and answering questions
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07:20.18sumais asterisk working wonderful ?
07:20.25sumawhu this channel is so silent ?
07:20.43Dico_hello everybody
07:20.58sumahi dico
07:20.59Dico_<PROTECTED>
07:21.04*** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it)
07:21.10sumawhat is your problem ?
07:21.11Dico_withouth the AMI ?
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07:21.18*** mode/#asterisk [+o denon] by ChanServ
07:21.21sumalinux they says, don't ask to ask
07:21.27Dico_humm, i've been told there is a problem
07:21.57Dico_suma,  hello
07:22.04Snake-Eyessuma, the asterisk channel was silent for 39 mins :P
07:22.07Dico_i dunno why the channe is so silent
07:22.20*** join/#asterisk Assid (i=assid@203.115.83.213)
07:22.26Dico_but yes asterisk works :)
07:23.07Dico_dlynes_laptop,  are you around ?
07:23.22Snake-EyesAU are going home, the US is still asleep and Euo is still waking up :P
07:23.40*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
07:23.47Dico_9.23
07:23.57Dico_it takes 23 min to drink a coffee ?
07:24.04e-ddieyeah
07:24.08e-ddiei didnt even get mine :(
07:24.20Dico_humm, may be : the time to enjoy the office managers attached to the coffee ;)
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07:25.03Dico_hehe : small survey :
07:25.08e-ddiegot it now :
07:25.09e-ddie:D
07:25.23Dico_in your company, what is an office manager : 1 person ?
07:25.27Dico_1 team ?
07:25.38Dico_is it equivalent to a secretary ?
07:25.47e-ddiethe one responsible for installing office
07:26.05Dico_lol, not the soft
07:26.11Dico_real people i mean :p
07:26.12e-ddie;)
07:26.27*** join/#asterisk inspired (n=mikael@85.221.7.59)
07:26.46Dico_I know there is cyber s... but real one is still better :p
07:26.52Dico_ok, anyway
07:27.13Dico_have you already tested the app_queue ?
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07:35.59teapotmorning
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07:38.53Snake-Eyesany one know if you can define a variable in sip,conf ?
07:40.30teapotonly time I've ever seen that is being able to define an environment variable in odbc.conf
07:40.35E-bolaMorning
07:42.07Snake-Eyeshmm
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08:01.20inspiredanyone else having problems with dtmf on 1.2.10? it worked fine on 1.0.7. after upgrading to 1.2.10 my apps seldom catch more than 3-4 digits before they just exit
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08:04.12kuku5which file do I specify the different 10000-20000 port range?
08:05.11darkgamer20is it possible to install freepbx on a custom installed asterisk system? where can I find the instructions to do so
08:05.23hads|homertp.conf
08:06.23hads|homedarkgamer20: freepbx.org?
08:06.39adorahhttp://www.freepbx.org/trac
08:06.53darkgamer20hads|homes: i tried the guides/help link but it shows something about trac
08:08.00Un1xhavent you guys read, the Topic
08:08.16hads|homeYes,
08:08.18Un1xFreePBX/Asterisk@home etc not supported here, join #freepbx
08:08.41darkgamer20oops sorry didnt notice that
08:14.01hads|homeUn1x: You can talk!
08:14.26Un1xlol, i was watching a movie debating weather i should watch another, or go sleep.. atm
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08:28.01teapotwhat's the 'correct' way to yield/sleep in a module?
08:28.33teapotI use sched_yeild() but grepping for it I find only one other instance in the code
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08:47.55phearlesshiiiiii
08:50.34phearlesshow can I set summer time / winter time on a cisco 7960 ?
08:51.05phearlessi am in london but in summer we are at GMT+1
08:54.02mitchelocphearless: i'm guessing that you can try setting up a time server and the phones can synchronize off of it?
08:54.37phearlesserr.. i do not know
08:54.52phearlesssntp_mode: "anycast"
08:54.52phearlesssntp_server: "asterisk"
08:54.53phearlesstime_zone: "CET"
08:54.57phearlessI use this for the monent
08:55.01phearlessmoment*
08:55.08phearlessasterisk is my asterisk server
08:55.14stinkpad"BST"?
08:55.27phearlessi will try BST
08:56.09*** join/#asterisk Alex|Work (n=hauntedu@gentoo/user/Alex)
08:58.48phearlessif I put BST in the SIPDefault, on the phone, it is GMT
08:58.54phearlessthis is weird
09:02.47phearlessi really do not know how this time thing works
09:03.37x86cant you use UTC ?
09:04.07x86hmm, probably no hard-coded timezone will automatically adjust for DST though
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09:06.10*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
09:06.13Bert-hello there
09:06.37Bert-just a little question : for what asterisk is listening on tcp port 2000 please ?
09:09.06hads|homeIt doesn't AFAIK unless you have told it to.
09:09.20Bert-AFAIK ?
09:09.30gandalfcomeI want to set up a linux server with two isdn cards, one to recieve calls from the existing isdn connection and the second isdn card two call with a normal isdn phone via voip. Is this possible? Is that doable in a reasonable amount of time with good linux knowledge? Are there better packages for this task? Thanks in advance
09:09.30hads|homeAs Far As I Know.
09:09.35Bert-lol ok
09:09.54hads|home2000 = # Sieve mail filter daemon
09:10.07Bert-??
09:10.20hads|homeThat's what /etc/services says
09:10.21Bert-in asterisk there is a mail filter daemon !?
09:10.25Bert-ho ok
09:10.40Bert-netstat -atnp shows that is asterisk listening onthat port
09:10.55gandalfcomewith asterisk of course.
09:11.17hads|homeTry `grep 2000 /etc/asterisk/*` see if it is set up to listen on that port.
09:11.28Bert-skinny
09:11.32Bert-done :)
09:11.58Bert-thx:!
09:12.10hads|homeAh, I don't use skinny. :)
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09:12.57Bert-neither di I
09:13.02Bert-s/di/do
09:13.27fourcheezeAssid: did you find out any more about realtime clustering?
09:13.41phearless<x86> cant you use UTC ? <-- london is not in UTC time
09:13.58hads|homeBert-: You can noload => chan_skinny.so in modules.conf
09:14.44Bert-ho
09:14.51Bert-I deleted the conf file to disable it
09:14.56Bert-bad way ...
09:15.32hads|homeProbably want to add a noload for it, otherwise it will just use it's default settings.
09:15.52Bert-hmm in fact if asterisk doesn't find the conf file, it disable the module
09:16.23hads|homeNot nessecarily I don't think
09:16.46Bert-for skinny, mgcp, sql i'm sure
09:17.08hads|homeOK, fair enough. Like I said I don't use them so I wouldn't know ;)
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09:23.54Assidfourcheeze: havent looked at it.. been busy trying to play with antispam settings
09:24.25fourcheezeAssid: I'm starting to hope that dundi may be the answer
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09:26.19fourcheezeAssid: http://forums.digium.com/viewtopic.php?p=4820&]
09:26.21phearlesswhat is the phone number to call to listen to the voicemail ?
09:26.26fourcheezeAssid: http://forums.digium.com/viewtopic.php?p=4820&
09:26.39fourcheezesecond one
09:27.01fourcheezephearless: whatever you make it
09:27.18phearlessi can't find this number in the config files
09:27.45Assidfourcheeze: well.. dundi still wont know where the user logged in from to route the calls
09:27.54phearlessI can see my messages in trixbox(asterisk recording interface)
09:28.06phearlessbut I can't find where is the phone number to listen to it
09:28.38fourcheezephearless: I have this:
09:28.41fourcheezeexten => 8500,1,VoicemailMain(${CALLERIDNUM}|s)
09:28.56Assidactually.. i added the @context as well
09:29.17Assidsince i got calls for 1 semi office within the main office
09:29.36fourcheezeAssid: I think dundi can be updated in real time, but I've not looked too hard at it
09:29.43Assidhrmm
09:29.54Assidwill check it in a bit.. im just too zonked to think ..
09:29.58Assidwas up till 5 or so
09:30.31fourcheezeAssid: what's your time now?
09:30.31Assidbrb
09:30.36Assid3.00 pm
09:31.43phearless200 => 123,alex,,,attach=no|saycid=yes|envelope=no|delete=no
09:31.45phearlessin /etc/asterisk/voicemail.conf
09:32.38phearlessfourcheeze: fro mwhich file does come from your line ?
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09:35.32yojanlI have a rather complicated problem I think, in short: I cannot call out with a grandstream gxp-2000, I think its an issue with the grandstream software and the asterisk version because I can call out with a sipsoftware in this network. But I cannot call out with the grandstream...
09:36.19yojanlincoming calls are working fine, only outgoing. Ive tried to understand the sip debug output and it looks like only invites are coming in, but i dont see an ack
09:36.34phearlessfourcheeze ?
09:36.49yojanlan other grandstream, with 1.0.1.9 software workes fine from an other network (havent tested it in this network)
09:36.54fourcheezephearless: extensions.conf
09:37.17fourcheezephearless: everything you dial goes in extensions.conf (or the realtime equivalent)
09:38.13phearless/etc/asterisk/applications.conf:exten => ${APP-MESSAGECENTER},5,VoiceMailMain(${VMCONTEXT})
09:38.13phearless/etc/asterisk/applications.conf:exten => _${APP-MESSAGECENTER}X.,4,VoiceMailMain(${EXTEN:3}@${VMCONTEXT})
09:38.13phearless/etc/asterisk/applications.conf:exten => ${APP-MESSAGECENTER-DIRECT},5,VoicemailMain(${CALLERIDNUM}@${VMCONTEXT})
09:38.13phearless/etc/asterisk/extensions.conf:exten => a,n,VoiceMailMain(${ARG1}@${VMCONTEXT})
09:38.23phearlessI got that stuff in my extensions
09:38.28phearlesssomething is missing ?
09:38.50phearlessI don't see the phone number for the voicemail in this, so something should be missing
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09:40.24pifyojanl : with the gxp you signed-up for a lot of pain
09:41.14yojanlpif: the other gxp I had worked fine, but now...
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09:41.30yojanlIm getting creazy!
09:41.35pifsome work, other don't
09:42.01pifthey're like women: moody
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09:44.32yojanlhaha, but I think it can be solved
09:44.40yojanli have 2 here, and they both do the same
09:44.56yojanlsomehow this version of the software f*cks something up with the request
09:45.01yojanlincoming is working smooth
09:45.22yojanlwhere could I go to solve this? does grandstream have tech support?
09:47.37phearlessI just added the module Voicemail in http://asterisk/admin/config.php?type=tool&display=modules
09:47.56fourcheezephearless: where did all that stuff in extensions.conf comefrom?
09:48.13phearlessi do not know
09:48.19phearlessfrom trixbox..
09:48.22fourcheezeahhh
09:48.34fourcheezeI think you want #freepbx then
09:48.44fourcheezesee the topic
09:49.04phearlessyes i know
09:49.16phearlessbut the config of the voicemail is in asterisk
09:49.17fourcheezeif you don't want to be using trixbox (and why would you) then just start off with a blank extensions.conf
09:49.24phearlessfreepbx is just an interface
09:49.34phearlessokay fourcheeze
09:49.37fourcheezeit's an interface that puts a load of stuff in your extensions.conf
09:49.45phearlessi will probbly do this when I will understand better asterisk
09:49.51phearlessprobably*
09:49.53hads|homeNo, it's an interface... Yeah, what he said.
09:50.25fourcheezehads|home: so trixbox doesn't need anything in your extensions.conf to work?
09:50.46fourcheezephearless: it's a steeper curve to go for plain old asterisk but you'll get there quicker :-)
09:51.09hads|homeNo, it does, I was halfway through saying what you said. But I'm slow tonite :)
09:51.19fourcheezeahhh
09:51.36hads|homemmm.. beer
09:52.57fourcheezeyeah, beer will speed you up ;-)
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09:56.37kanelbullarHello all
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09:57.29kanelbullarDoes anyone know if e&m signalling supports ANI?
10:00.37kanelbullarzapata.conf: siginalling=em
10:01.00kanelbullarzaptel.conf: e&m=1-24
10:05.38phearlessanybody got some fun cisco 7960 logos ?
10:12.36nibbler_deif i don't want to "Answer" a call. what was the other command again?
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10:21.36erwinismhello, i have cisco 7960, can anyone help me connect this phone to my asterisk box?
10:21.44erwinismit is brand new
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10:23.24Rawplayererwinism: why?
10:23.38Tommmofor RealTime is it necessary to define each realtime context in extensions.conf?
10:23.41Rawplayerjust add a client
10:23.52erwinismi dont have idea how to connect this thing. its not on its manual
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10:32.20shadebobhi. I have a little problem with the transfer of an agent call. When I tranfer the call, I have only silence
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10:35.44tinyviolinshadebob: at least it's easier to close the sale that way, without any objections
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10:41.30kuku5erwinism: you need the sip image
10:41.43kuku5voip-info.org
10:41.49kuku5or you can use sccp i guess
10:42.21erwinismkuku5: ok i will search on that
10:42.24phearlessokay *98 works for the voicemail now ... COOL !
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10:48.27mutmornin
10:48.29Winkieis it just me or is chan_agent the most horrible thing ever when using the manager interface?
10:48.30Winkiemornin
10:48.33Winkiei've been up all night
10:48.45Winkieand now i intend to hunt down whoever is responsible for zombie and masq channels and kill them
10:49.14mutheh *shrug*
10:49.22Winkieif you are in this channel and are responsible please pm me home address and list of greatest fears thx
10:50.08tinyviolinWinkie: wait 'til you got 40 agents on and it deadlocks and you're getting screamed at
10:50.25Winkietinyviolin: i'm putting way more than 40 onto it
10:50.33Winkiehowever we will have live backup systems
10:50.41Winkiei'm talking purely from a tracking point of view
10:50.43tinyviolinhow many?
10:50.48Winkie140 or so?
10:50.51erwinismkuku5:  thanks for the help
10:50.52Winkienot on a single box like
10:50.53tinyviolinone box?
10:50.55tinyviolinoh
10:50.58Winkieno, 3 most likely
10:51.04Winkie1x40 2x50
10:51.20Winkiehave you ever read the stuff it spews out on the manager interface?
10:51.36tinyviolinyeah, liberal use of UserEvent can help make it a little more sane
10:51.48tinyviolinthen you can disable a bunch of the perms and have way less crap to look at
10:51.54Winkieirrelevant in our situation unfortunately
10:52.10tinyviolinare you using cmd Queue?
10:52.13Winkieour calling plan is extremely simple, our agents do most of the work with transfers and internal bounces
10:52.17Winkieyes
10:53.01Winkiei don't have any problem with the queue portion, but when it tries to connect to an agent it pseudo bridges it using some horrific Local channel hackery it seems
10:53.38Winkieit seriously needs to be rewritten and if it does deadlock on me people will die
10:53.42tinyviolini use Bridge .. it's simpler, but you have the do the metrics yourself
10:53.52tinyviolinbut you have a lot more freedom
10:54.12Winkieindeed, i've already written one patch to add in an AgentAssociate event which allows you to link the inbound channel to the agent channel
10:55.24tinyviolincool
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11:01.50hads|homeI don't use it so this may be bogus, but aparently there is known trouble with queues and it is better in trunk.
11:02.31hads|homeerm... queues/chan_agent something. As I said, I don't use it.
11:02.32tinyviolinwell yeah, i shouldn't bitch, because the guys on our team that still use agents/queues haven't had the deadlocks in a while
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11:03.53Winkiewell hopefully i'll be writing a paper on this implimentation i'm doing, pretty much 99% linux, offices in 3 countries with over 200 asterisk managed callcentre workers
11:03.56Winkiegoing to be a while though
11:04.57tinyviolinyou know, i just looked and app_queue is 4000 lines of code
11:04.59tinyviolinwhy?
11:05.13Winkieyeah that's what i can't figure out, check out chan_agent
11:06.20hads|home2500
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11:11.46*** join/#asterisk Rawplayer (n=kevin@braadharing.oom-killer.org)
11:11.58Rawplayerdoes anyone of you guys have a snom 300 voip telephone?
11:12.25Rawplayerwhen i pick up the horn it makes a brummy noise
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11:25.14Tommmofor RealTime is it necessary to define each realtime context in extensions.conf?
11:25.35Tommmoi find that it only works when the config file has been updated to have the realtime context defined
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11:45.30ManxPower*sigh*  NPR is doing another story on New Orleans
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11:52.47phearlesshwo can i switch my message status between busy, unavailable,temporary ?
11:55.33ManxPowerphearless, for the most part you don't switch between busy and unavail in the system.
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11:55.51ManxPowerAsterisk plays the message type based on the u or b prefix to the call to voicemail
11:56.36phearlessi do not know waht is u and b
12:00.48ManxPowerphearless, Voicemail(b1234)
12:01.00ManxPowerin extensions.conf
12:01.11phearlessok i will have a look
12:01.26ManxPowerUnless you are using FreePBX/AMP/Asterisk@Home/Trixbox.  If you are using one of those, I do not know how you set the option.
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12:17.13danastb stands for busy and u stands for unavailable playbacks
12:18.39[TK]D-Fender"show application voicemail"
12:19.40danastI am using asterisk realtime voicemail, I cannot get app_directory to work and cannot change vm password from the phone. \n I am using postgresql
12:19.59danastany suggestions
12:22.33Ahrimanesanyone having trouble with nested macros ?
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12:34.20Tommmodoes the number of contexts defined in extensions.conf significantly impact the performance of asterisk?
12:34.30Tommmoeven if most contexts have nothing in them?
12:34.34Tommmo(or very little)
12:35.42ManxPowerTommmo, it should not
12:36.14Tommmothanks
12:36.41ManxPoweryou should, of course test that.
12:36.54Tommmowill try
12:37.52Assidhahahahaha
12:37.52Assid<PROTECTED>
12:38.04neoasterisk ?
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12:38.47Assidnope.. email
12:39.11neospamd ?
12:39.18[TK]D-FenderI can't imagine there being a reason it should place a load of any significane on a system.... (Aside from a WRT maybe in a worst case scenario)
12:39.26[TK]D-Fender(if even)
12:39.41ManxPower'morning [TK]D-Fender
12:39.51Tommmobecause it appears i can't create contexts on the fly for Realtime
12:39.56Tommmoit looks like they have to exist first.
12:40.25Assidspamd .. can do that to you
12:40.36neooh yeah
12:40.42*** join/#asterisk jaike (n=a@203.115.188.120)
12:40.46neocan kill your box so easily :)
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12:41.25jaikeanyone experiencng mixmonitor problems with 1.2.10? one-way audio or one side delayed
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12:44.06[TK]D-FenderManxPower: *yawn* mornin'
12:44.32[TK]D-FenderTommmo: That is something different....
12:44.52Tommmo[TK]D-Fender: is there another way around it?
12:45.04Tommmobecause it seems each context for realtime has to be put into extensions.conf
12:45.04Tommmoe
12:45.21Tommmoeg [CONV001-2]
12:45.22Tommmoswitch => Realtime
12:48.42fourcheezeAssid: what's your MTA - exim has an option to stop receiving mail when load is high
12:49.23Assidfourcheeze: qmail
12:49.34fourcheezeahh
12:49.44Assidits averaging around 12-14 right now
12:49.54Assidmostly because of clamd
12:49.58fourcheezeyeah
12:50.02fourcheezeBTDT
12:50.05ManxPowerThe problem with qmail is that I'm basically lazy.
12:50.16fourcheezethe problem with it is that it's so non-standard
12:50.17Assiddtbt?
12:50.21Assidbtdt ?
12:50.26fourcheezebeen there done that
12:50.39ManxPowerqmail's license does not allow distros to include the binary, which means I have to build it myself and, as I said, I'm lazy.
12:50.48fourcheezeahh yeah
12:51.05fourcheezeit also puts itself in stupid places IIRC
12:51.16*** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com)
12:51.17fourcheezemy first ever MTA was qmail - then I did apt-get install exim and saw the light
12:51.36*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
12:52.09Assidnah.. im happy wit the likes of qmail.. nice speed..
12:52.56ManxPowerd
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13:02.24ionixqmail is so gay
13:02.27ionixExim is better
13:02.35ionixit installs in like 20 dirs
13:03.35muto_O
13:06.58mutAssid... how much mail is that thing processing?
13:06.58Nivexpostfix > exim
13:06.58mutor is it just a crap server?
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13:07.28creativxhm
13:07.42creativxthe quickes way to grep for the queuememberstatus "status" codes?
13:07.58Assidmaybe around 10 mails per second..  including spam
13:08.49Assidrunning clamd/spamd etc on it
13:08.56fourcheezepostfix is good, but exim is a bit friendlier
13:09.05blitzragepffft... sendmail!
13:09.06fourcheezeexim has some good hooks in it for spam/virus etc
13:09.15fourcheezeha
13:09.43fourcheezeAny irc channel can turn into a MTA war
13:09.57quid246MTA?
13:10.42quid246I thought MTA wars only happened on the NCY subway
13:10.42quid246NYC
13:10.42fourcheezeNYC?
13:10.43fourcheezehttp://www.google.co.uk/search?q=define%3A+MTA&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-GB:unofficial
13:10.54Assidkoay i think that box doesnt wanna respond anymore
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13:11.14Assiduhho... __alloc_pages: 0-order allocation failed (gfp=0xf0/0)
13:12.12Assid__alloc_pages: 0-order allocation failed (gfp=0x1d2/0)
13:12.13AssidVM: killing process perl5.8.4
13:12.18Assidthats not good is it?
13:12.59*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
13:15.38fourcheezewell it's trying to help
13:15.56fourcheezeif you won't kill processes yourself ;-)
13:17.59Assidneed more ram?
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13:27.24fourcheezeAssid: will it let you type "free" right now?
13:28.05hmmhesayssevard
13:28.16hmmhesaysI need someone to test this calling card app
13:28.19fourcheezeAssid: but generally more ram would be good
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13:29.54SpaceBasshey folks
13:29.57Assidyeah.. tons of caching and buffering.. as usual
13:30.01SpaceBassgot a strange one-way audio issue
13:30.02yojanlanybody has great knowledge about grandstream gxp-2000's? I cannot make an outgoing call, very strange, response doesnt come back to the phone but in this network everything is okay. From a software client I can make outgoing calls so I guess its the software in the GXP
13:30.34SpaceBassI have 2 wifi phones on a subnet that is seperated by nat/firewall... when I call to the phone, audio works fine, when I call from it I get no audio in either direction....firewall is setup to allow all traffic
13:30.46SpaceBassbut here is what is strange, when I turn on rtp debugging, it works perfectly
13:30.58hmmhesaysSpaceBass:
13:31.07paryli'm having issues with agents... now and then one will try to log on, and it will get stuck... for lack of better words.  it sits there and nothing happens... so they naturally try to login again and again... and finally i have to restart asterisk altogether because there are 10 login tries just sitting there in the system.  is it possible to kill a specific call that's taking place from the...
13:31.08SpaceBasshey hmmhesays
13:31.08paryl...asterisk CLI?
13:31.31hmmhesaysSpaceBass: you want to test this calling card app for me?
13:32.24SpaceBasshmmhesays, why not?
13:32.42hmmhesayshttp://www.thelostpacket.org/tricks.php
13:33.23SpaceBasshmmhesays, give me a few
13:33.28SpaceBassand I'll let you know
13:33.50hmmhesaystesting your own stuff never works out well
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13:35.30cbrakeI have a Linksys SPA-941 and a cheap analog phone connected to a TDM400-FXS port calling out over NuFone.  People I call say the Linksys phone sounds garbled and the analog phone sounds fine.  I can't tell any difference on the receiving end.  Any ideas why the linksys phone does not sound as good on the receiving end?
13:36.10hmmhesaysum call yourself and test it
13:38.01cbrakehmmhesays: yeah, I've done that and the Linksys phone does sound worse.  Trying to figure out why, or what I can do to fix it.  Review suggest the SPA941 has decent voice quality ???
13:39.02cbrakehmmhesays: but the sound coming into the Linksys phone sounds fine.  Just seems to be an issue with sound going out of the linksys phone.
13:39.16hmmhesaysvad enabled on the linksys?
13:39.37parylplease help guys... is there a way to kill a call in progress?
13:39.44hmmhesaysctrl-c
13:39.59benjkyeah, hangup
13:40.24parylbut i have, and they still show up in 'show channels verbose'
13:40.39benjkzombies
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13:40.49parylhow do i get rid of them?
13:40.55benjkwait
13:41.03cbrakehmmhesays: what is vad?
13:41.12paryldude, i am waiting... but they are staying around
13:41.14hmmhesaysvad or silence suppression
13:42.29paryli have one 'zombie' that's been active 8 minutes now
13:42.48paryland no other calls can get in because of it
13:42.58benjkis this on a zaptel interface?
13:43.16parylactually... it agents who are logging in
13:43.32cbrakehmmhesays: silence suppression is turned off
13:43.41hmmhesayswhat voice codec?
13:44.15cbrakehmmhesays: G711u from the phone to asterisk
13:44.25cbrakehmmhesays: and GSM from asterisk to nufone
13:44.32hmmhesayscheck your packet time
13:44.52cbrakehmmhesays: can this be done w/ ping?
13:44.58hmmhesaysno
13:45.13benjkthe queue mgt system has a wind up parameter to specify the time an agent needs to clean up his stuff after each call
13:45.46parylbenjk... is that in response to me?
13:45.52benjkyeah
13:46.03hmmhesayscbrake: is is the amount of voice that goes into each packet
13:46.26benjkread up on configuration for agents and queues
13:46.53mutgod i hate to tell my bosses i told ya so
13:47.07mutbut they grounded all of our copper we laid in this town wrong
13:47.32parylyes... i have.  this isn't what you're thinking.  like i said above... it works great most of the time, but now and then the login process stalls, for lack of better words.
13:47.37parylcreating what you called a zombie
13:47.46benjkshould have listened to you and roll out fiber instead of copper :)
13:47.51parylthen every login/logout attempt after that simply stalls too
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13:48.12paryli just upgraded to 1.2.10 last night, and it's happened twice this morning
13:48.27benjkdid it work before?
13:48.38mutheh yea benjk
13:48.47mutwe shoulda
13:48.59benjkcopper prices are going up too
13:49.05benjksome strike in Chile
13:49.11cbrakehmmhesays: is packet time part of the SPA941 or asterisk config or both?
13:49.22mutit would just be the most hi tech city in northern michigan or something
13:49.34mutfiber to home before most of the rest of the state gets it
13:49.43mutin a rural town
13:49.56mutbe good publicity anyway
13:50.04benjkwell, you better get used to fiber because more and more countries rolling out fiber, its going to become not-quite-so-hi-tech-anymore
13:50.17mutheh
13:50.24mutwe're JUST getting dsl here
13:50.27mutand it's us providing it
13:50.32mutnot any telco
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13:51.00benjkcool
13:51.18hmmhesayscbrake: compare the two
13:51.22mutfrom what i hear now verizon wants to start doing dsl tho
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13:51.28mutbut that'll be 2 years off atleast
13:51.29parylhas anyone else seen the issue i described above?  please?
13:51.47benjkdid it work before 1.2.10
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13:52.41parylit works all the time, yes... this is just a periodic issue now and then.  it's just that after upgrading to 1.2.10, it's happened twice in the last hour
13:53.07paryli never could track it down, but now because of the timing, i have to
13:53.10benjkthen go back to the version you were using before
13:53.24parylwhich had IAX2 and vm issues
13:53.56benjkhigher version numbers are not going to cure your penis envy, in case you suffer from that
13:54.19parylgod, are you the channel troll or what?  dude, i'm looking for answers
13:54.37paryli said the previous version had major issues i needed fixed
13:54.43parylnow this is the issuee
13:54.49benjkpeople should be using the software that works for them, not just because the version numbers are fancy
13:55.27benjkno you said that your problems have become worse
13:55.28blitzragesnakes on a plane!
13:55.49blitzragehehehe
13:57.23tzangerfucking bluetooth
13:57.29tzangerwtf is up with that snakes on a plane hting
13:57.30tzangerI don't get it
13:58.17[TK]D-Fendertzanger: Its a new movie with Samuel L Jackson.
13:58.41Dr-Linux|worktzanger looks angry!
13:58.47[TK]D-Fendertzanger: It was a "working title first" and then when they wanted to change it, SLJ said NO WAY!  So it stuck, and its kinda catchy :)
13:58.50hmmhesayssnakes on a plane muthafuckah
13:59.05[TK]D-Fendertzanger: Thats also literally the plot line.
13:59.11filemoo
13:59.22*** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu)
14:01.28muttouch file
14:01.35hmmhesayswhoa...
14:01.40fileeep
14:02.13mut-_-
14:02.16Rawplayer-,-
14:02.29cbrakehmmhesays: SPA941 RTP Packet size = 0.03s (is this "packet time") ?
14:02.31*** join/#asterisk ToyMan (n=stuq@ool-44c7b88e.dyn.optonline.net)
14:02.34tzanger[TK]D-Fender: ahhh okay
14:02.45tzangerI thought it was maybe a web 2.0 thing
14:02.46hmmhesayscbrake yup
14:02.50hmmhesays30 milliseconds
14:03.17e-ddiei use web 3.1
14:03.34cbrakehmmhesays: ok thx, now to find the nufone size  ...
14:03.43hmmhesayscompare them between your two phones
14:03.53hmmhesaysnufone size doesn't matter
14:04.05creativxi use web 3.11 for workgroups
14:04.09creativxits AJAX2 powered
14:04.16hmmhesaysif your linksys is sending out larger packets, they might be exceeding the mtu size on your modem
14:04.22*** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it)
14:04.23*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
14:04.43Dr-Linux|workwho owns nufone?
14:04.47[TK]D-Fendertzanger: In there he supposedly says quite directly "I wan these MOF snakes off the MOFO plane!"
14:04.58hmmhesaysbwhaha
14:05.02[TK]D-Fendertzanger: Because he NEVER speaks inappropriately ;)
14:05.25*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:07.02blitzrageI can't believe you guys are still talking about snakes on a plane :)
14:07.18muti can;t believe they remade it
14:07.24mut:'(
14:07.32tzangerDr-Linux|work: jerjer
14:07.38tzanger[TK]D-Fender: indeed :-)
14:07.42tzangerahem
14:07.46tzangerREMADE snakes on a plane?
14:08.00hmmhesaysthat was all the rage on totalfark for like a month
14:08.30hmmhesaysso where do I post this calling card app so I can get some criticism on it
14:08.33trelane_SNAKES ON A PLANE?!?
14:08.47*** join/#asterisk sp0n9e_ (n=sp0n9e@phpurge.com)
14:08.53trelane_Dr-Linux|work, Jermey McNamera
14:09.44cbrakehmmhesays: how do I determine the * -> Nufone packet size?  Can't find any options in iax.conf and can't convince the console to tell me?
14:09.48tzangergod damn I hate Palm's insistence to lock you out of your own hardware
14:09.52tzangerit's almost as bad as Microsoft
14:10.03blitzrageaye
14:10.14hmmhesayscbrake: i meant compare between the two phones you are using
14:10.26mutyea
14:10.28mutfrom book
14:10.43muti can't believe all the publicity such a retarded movie is getting
14:10.57hmmhesayshey, slj is in it
14:11.02mutwell yea
14:11.07mutthat'll make it a winner right there eh
14:11.19hmmhesaysthe 51st state was fantastic
14:11.22benjkI hate Vodafone's insistence not to allow me to view any photos that haven't been taken with one of the phone's two cameras
14:11.30benjkand any other content for that matter
14:11.43benjkyou can send yourself voicemail by email attachment for example
14:11.57benjkwell, you can, but the phone will not play it, cause it isn't signed by Vodafone
14:12.16yatesyvodaphones branding really sucks
14:12.39benjkif it was just the branding I couldn't care less
14:12.41yatesymy dad ended up reflashing his phone back to manufacturer standard
14:13.19benjkI would like to know how to do that
14:13.22quid246I hate Cellphones... how they lock everything you do, a *real* phone would let you upload ringtones/pictures/etc via USB or BlueTooth with no "service fees"
14:13.35yatesyi can do that on mine
14:13.39benjkI don't care about ringtones
14:13.56benjkbut I want to be able to use voicemail by email attachments
14:14.27tzangerquid246: yep
14:14.37benjkits a Symbian based phone so I figured some Java guru could somehow remove the need for the signature
14:14.38tzangerthis new Nokia 6265i I have apparently only supports one bluetooth device at a time
14:14.47benjkbut every Java guru I asked said it wasn't possible
14:15.05tzangerif my headset is connected it does not participate in any other BT activities... i.e. I cannot use my Palm to dial through my phone if the headset is on
14:15.16benjkthat sucks
14:15.32Pj_polygamy rulez
14:15.34benjkmy Nokia doesn't do that
14:15.37tzangernot to mention that Palm refuses to let me have full control over the init and dial strings, so CDMA phones don't work as a general rule
14:15.57tzangerbut I can hack the palm library to use the correct BT profile (HFAG instead of DUN)
14:16.01benjkI can be on the phone via BT headset while my MacBook is using the phone for a data connection also over BT
14:16.02hmmhesaysbah cleaning out my old code folder, I got a lot of half written crazy sh1at in here
14:16.10tzangerbenjk: which phone is that?
14:16.18tzangerhmmhesays: :-)  We all have those kinds of folders :-)
14:16.30benjkI am not sure, I think 6680, here in Japan they change the model numbers
14:16.37tzangeryeah same here in Canada
14:16.41tzangerit's CDMA though which is the problem
14:16.52benjkthe Vodafone model number is Nk700MkII
14:16.56tzangerit's funny, it has a WCDMA SIM card socket but the manual says not to connect anything into it
14:17.01benjkor Nk702MkII
14:17.28benjkthis is a multi-protocol multi-band phone
14:17.47benjkit does both GSM in four frequency bands and WCDMA
14:18.16benjkyou need that over here cause Japan doesn't have any GSM whatsoever
14:18.29cbrakehmmhesays: other phone is analog TDM400 FXS channel.  How do I determine the packet size when using this phone?
14:18.51tzangerbenjk: NICE
14:18.52cbrakehmmhesays: I suppose I could also run some tests w/ xlite as well.
14:18.57hmmhesaysgood call
14:19.02tzangerI might have to send this phone back, this blows goats
14:19.09tzangerMotorola RAZR maybe
14:19.09hmmhesaysalso do an echo test, see if the problem shows up there
14:19.14ionixman, I hate coding forms
14:19.23benjkbut as I said, becuase of Vodafone's greed I cannot listen to voicemail attachments because my Asterisk server cannot sign them with Vodafone's digital signature
14:19.24hmmhesaysionix: yep
14:19.26ionixform validation, error notification and all that shit
14:19.38ionixsomeone should make a killer class for forms
14:19.58ionixlike we just specify the forms with preg match and validation tools and it does it automatically
14:20.03tzangerI hate form programming as well
14:20.09tzangerbenjk: yuck.
14:20.14benjkyeah
14:20.19ionixdoing a signup form is soo boring then you have to code a information update form after the signup
14:20.26ionixpukes
14:20.30hmmhesaysdreamweaver is pretty nice for generating forms
14:20.36hmmhesaysnvu is ok too
14:20.40ionixwell, the form is easy
14:20.43ionixthe form validation is shit
14:20.49hmmhesaysforms with associated validation
14:20.59benjkalso, OSX comes with a synchronisation utility which installs an agent on a Symbian phone by sending the Java applet to the phone by BT
14:21.02ionixin dreamweaver?
14:21.10tzangerbenjk: nice
14:21.16benjkI can't install this applet because its not signed by Vodafone
14:21.20ionixah wtf
14:21.24ionixI thought I was on ##php
14:21.25ionixsorry
14:21.27benjkand this means I cannot synchronise addresses and stuff
14:21.29hmmhesaysionix, LOL
14:21.34hmmhesaysnice
14:21.36ionixdamn, I am tired tonight
14:21.41*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
14:21.41hmmhesayswhere are you?
14:21.44ionixJapan
14:21.44tzangerI'm gonna see if I can fuck around with the phone with the nokia phone suite
14:21.45*** join/#asterisk murf (n=steve_mu@216.166.159.235)
14:21.59benjkWhen I tried to bring this to Vodafone's attention they said the phone doesn't support Mac connectivity
14:22.00*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:22.01hmmhesaysI have a linksys wip-300 that is a phone tof ark with
14:22.17benjkI said, Apple does support it, you block their support
14:22.24jtexter3I'm currently writing a module for asterisk.  If I have two channels bridged together, what is the correct way to break that bridge so I can go on and do other things with the channels?
14:22.27benjkthey don't get it
14:22.37benjkits not on the CD, so it can't work, they say
14:22.50hmmhesaysI'm guessing look at the transfer app
14:26.14*** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
14:27.00jtexter3unfortunately, app_transfer just uses the underlying technology to connect to a separate channel, i.e. a sip redirect
14:27.00fourcheezecan anyone see the writing on the wall with the windows communications server 2007?
14:27.16hmmhesaysok so who wants to test this prepaid dialplan
14:27.20hmmhesaysI need some input
14:27.29jtexter3it looks like may ast_softhangup is what I want, but I can't tell for sure
14:28.09brodiemjust curious, what causes the first part of the audio on a local extension to be cut off, i.e. if you are dialing an extension that just does a Playback(). Unless you put a Wait() beforehand the first .5 second or so of the audio is cut off. Is there a sip.conf parameter that will correct this?
14:28.39hmmhesaysthe first part of the audio for pretty much any extension will be cut off without a wait
14:29.07brodiemI've concluded that already :)
14:30.25*** join/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net)
14:30.38brodiemI would just love to know why it cannot just wait until the call is setup before it starts running through
14:31.32*** part/#asterisk jaike (n=a@203.115.188.120)
14:31.37benjkbrodiem, jtodds silence recordings are your friend
14:31.42*** join/#asterisk zedkatuf (n=audela@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk)
14:31.54benjkwww.loligo.com
14:32.09benjklook for sounds
14:32.14benjkdirectory silence
14:32.40benjkplay 1 sec of silence before your prompt and you'll never experience a cutout
14:33.20*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
14:33.22benjkWait() doesn't help, btw
14:33.31brodiembenjk why not just use Wait()?
14:33.41benjkwell, it doesn't fix the cutout
14:34.18brodiemi was just curious to know if this happens with everyone and if there was some parameter I could set that would start audio after a channel is completely setup or something
14:34.32*** join/#asterisk dyn (n=dyn@unaffiliated/dyn)
14:34.35brodiembenjk it always works for me?
14:34.36dynhi
14:35.05dynfaq: I have no "dial" command on the asterisk console.. which module am I missing?
14:35.05benjknever worked for me
14:35.14benjkapp_dial.so
14:35.15dynis it 'modem'?
14:35.24*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.118.Dial1.SanJose1.Level3.net)
14:35.28dynthanks benjk, checking
14:35.42*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)
14:36.02benjkmaybe your Linux box has a sense for tidyness
14:36.08dynbenjk:  loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_dial.so: undefined symbol: ast_bridge_call
14:36.40benjkapp_dial is amongst the most horrible code on the planet
14:36.47SpaceBasshmmhesays, I'm still going to try and test for ya
14:36.51MatsKdyn: do you have a soundcard and oss or alsa configed
14:36.51SpaceBassmy "day job" is getting in the way
14:36.53*** part/#asterisk jailbreaker (n=teodory@mail.jetfinanceintl.com)
14:36.56benjkast_bridge_call is in res_features
14:36.58SpaceBassnot to mention my one-way audio issue
14:37.07hmmhesaysSpaceBass: cool
14:37.16hmmhesaysit is pretty simple
14:38.04benjkmake sure you have res_features.so loaded
14:38.14dynMatsK: I have but now as you say, something is not right with my alsa config (cannot even invoke alsamixer though I have a /dev/dsp0). ok, I'll check this, thank you
14:38.20*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)
14:38.42*** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net)
14:38.46benjkthe error message you pasted has nothing to do with ALSA
14:39.24benjkyou have res_features.so missing
14:39.30dynokay
14:40.34*** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net)
14:40.43sudhir492Hi All
14:40.50sudhir492Anyone using PAP2-NA
14:41.00SplasPoodAnyone ever had an issue where *1 on a Queue/Agent call will cause the call to hangup?  I can duplicate it every time
14:41.13benjkits not an issue
14:41.17benjkits a feature
14:41.33SplasPood:P
14:41.36hmmhesayswell my yum is completely farked up
14:41.40benjkshow application dial
14:41.44benjkwill tell ya
14:41.59benjkjust don't pass the h and H flags
14:42.00SplasPood*1 starts on the fly recording
14:42.07SplasPoodand those flags are not passed
14:42.13SplasPoodthis is an Agent call
14:42.15SplasPoodin a queue..
14:42.32benjkthen there's a bug in app_dial or ast_bridge_call
14:42.45*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
14:42.58benjkbecause the hangup is caused by the code that is supposed to hangup when you press * if you pass those flags
14:43.07SplasPoodhrm yea
14:43.13SplasPoodlemme try remapping the feature
14:43.15SplasPoodto something else
14:43.22SplasPoodisn't it hangup on # tho?
14:43.30benjkapp_dial is pretty horrific, its possible that some circumstances cause it to confuse the status
14:43.40benjkno, # is for transfer
14:43.45SplasPoodnope
14:43.46SplasPoodyea
14:43.46SplasPood*
14:43.59benjk* for hangup, # for transfer
14:44.25macTijn~paste?
14:44.26jboti heard paste is see http://paste.husk.org
14:44.31macTijn~pastebin?
14:44.32jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts
14:44.34*** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net)
14:44.56benjkand with all them gotos and breaks in the app_dial, there are situations where it does something that it shouldn't do
14:45.22macTijn~pastebin is also http://paste-it.net
14:45.24jbotmacTijn: okay
14:45.31macTijn~paste is also http://paste-it.net
14:45.32jbotokay, macTijn
14:45.36macTijnredundancy++
14:45.37macTijn;)
14:46.37Rawplayer;)
14:46.40benjkSplasPood, you can always comment out the section of code in app_dial that handles the * DTMF, also in ast_bridge_call, then test again
14:46.45zeedoheh, the reply to ~pastebin floods more than mosts pates :-P
14:46.52benjkif the issue is gone, you know why
14:47.54SplasPoodbenjk: heh I *could* do that...  thing is I want them to be able to hit *1 to record the call
14:48.10benjkI am not saying you should do it as a fix
14:48.20SplasPoodok, yea
14:48.33benjkI am saying you should do it as a test, to find out what the culprit is
14:48.47*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)
14:49.27SplasPoodhrm, how can features.conf be reloaded
14:49.35SplasPood(short of a full reload)
14:50.20blitzragetry "reload res_features.so"
14:50.42blitzragethen do a "show features" to see if your changes took
14:50.49[TK]D-FenderSplasPood: You're using AgentLogin for your Queu aren't you?
14:51.19SplasPood[TK]D-Fender: Yes
14:51.45[TK]D-FenderSplasPood: You're kinda screwed... * hangs up a call in that app....
14:51.55[TK]D-FenderSplasPood: Pray you can remap that feature.
14:52.16*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)
14:52.32SplasPood[TK]D-Fender: Where's that, as part of the Agent/ channel?
14:52.39brodiemi find its just easier to record all calls :)
14:53.05SplasPoodbrodiem: thats the path I'm going to take if necessary
14:53.23SplasPood[TK]D-Fender: show application AgentLogin doesn't mention it
14:53.35SplasPoodoh
14:53.35SplasPoodyes
14:53.39SplasPoodit says 'the star key'
14:55.53SplasPood[TK]D-Fender: I'm actually using CallBackLogin
14:55.55SplasPoodbut same diff
14:56.10SplasPooddoesn't appear to be a way, short of hacking the source, to disable it
14:57.12benjkthat's what happens if you bolt on features without an overall design
14:57.38*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:57.41benjksooner or later features will step on each others' toes
14:57.47*** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se)
14:58.25sudhir492Anyone here using PAP2-NA?
14:58.57hmmhesayspaptuna
14:59.22benjkpapertuna
14:59.26benjkpapertiger
14:59.33hmmhesaysspiceytuna
14:59.34*** join/#asterisk _Guhit (n=amistry@am-productions.biz)
14:59.34benjkpeppertuna
14:59.38benjkheh
14:59.41hmmhesaysspeaking of spicey tuna
14:59.41*** part/#asterisk dasenjo (n=dasenjo@208.195.215.74)
15:00.17hmmhesaysword to the wise, don't mess around with your girlfriend after eating hot wings and not washing your hands
15:00.25SplasPood[TK]D-Fender: where would this code be, chan_agent?
15:00.31dynrotfl
15:00.41hmmhesaysshe WILL NOT be pleased
15:00.48filehmmhesays: or yourself
15:00.48*** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com)
15:00.56benjknot washing your hands or not washing your mouth?
15:00.57hmmhesaysfile: lol
15:01.22hmmhesaysbenjk whatever part of you that you might be sticking down there that still has hot sauce on i
15:01.22hmmhesayst
15:01.24benjkYou can always get a Korean girlfriend
15:01.30SplasPood[TK]D-Fender: nevermind, it is
15:01.31benjkthey should be used to it
15:01.34fileone of my friend's did that, he put a drop of something called "Satan's Blood" on his finger to taste it - went to the bathroom afterwards...
15:01.34filehaha
15:01.42hmmhesaysKorean girls like burning vagina's?
15:01.53hmmhesaysfile: LOL
15:02.07hmmhesaysthis happened to be "wild sauce"
15:02.07benjknot sure, but they eat this stuff called kimchi and other hot stuff all the time
15:02.12*** part/#asterisk yogurt2ungue (n=charlie@200.69.250.91)
15:02.30*** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au)
15:02.38malverianAnyone else getting frequent crashes with Asterisk 1.2.10?
15:02.51hmmhesaysmy 1.2.5 box crashed today
15:02.51malverianSeems to be a bug in channel.c that was introduced since 1.2.7
15:03.19malverianI get a bunch of messages about "avoided initial deadlock" and then a few seconds later it crashes.
15:03.24hmmhesaysanyone ever have the problem where you connect and you get no command line?
15:03.36*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)
15:03.37hmmhesaysLicense version 2 and other licenses; you are welcome to redistribute it under
15:03.37hmmhesayscertain conditions. Type 'show license' for details.
15:03.38hmmhesays=========================================================================
15:03.41hmmhesaysthats it
15:04.08brodiemi've never had a crash yet knock on wood
15:04.10brodiemroot@pbx:~# uptime
15:04.10brodiem<PROTECTED>
15:04.18*** join/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil)
15:04.21malverianbrodiem, I mean Asterisk crash.. not the actual box.
15:04.28malverianbrodiem, asterisk -r -x "show uptime"
15:04.33sp0n9edamn there's a lot of warnings in the wanpipe driver.
15:05.09brodiemmalverian, nothing impressive about mine there :) I had to restart when I added a PRI a couple of days ago
15:05.34hmmhesaysCLI> show uptime
15:05.34hmmhesaysSystem uptime: 49 seconds
15:05.34hmmhesays*CLI>
15:06.05hmmhesaysI just restared
15:06.12yojanlhi, what should I fill in in the field "Use NAT IP" in the advanced setting of the gxp-2000? the public IP of our network, the ip of our asterisk server or the internal ip of the grandstream?
15:06.23hmmhesaysnothing
15:06.28hmmhesaysdon't fill it in
15:06.33hmmhesaysnat=yes in sip.conf
15:07.16yojanlokay, but I have a problem here with outgoing calls, I was hoping the Use Nat IP could help me solve the problem
15:07.28hmmhesayswhat problem?
15:08.16*** join/#asterisk marv[work] (n=timr@64.89.118.139)
15:08.35yojanlstrange problem, outgoing calls time out, the grandstream gets no reaction from the server and the server cant get to the grandstream. First I thought it was something here in this network but Ive installed a sip soft client and i can call out with that sip soft client so my guess is its the grandstream...
15:09.00yojanlincoming cals work without problems...
15:09.13sp0n9eyojanl: do both of them have the same context?
15:09.18hmmhesaysyou have nat=yes in sip.conf?
15:09.39yojanlsp0nge: yes, I even used the same account for testing
15:09.44yojanlnat=yes is in sip.conf
15:09.52SpaceBassI have 2 wifi phones on a subnet that is seperated by nat/firewall... when I call to the phone, audio works fine, when I call from it I get no audio in either direction....firewall is setup to allow all traffic
15:09.55SpaceBassbut here is what is strange, when I turn on rtp debugging, it works perfectly
15:09.57hmmhesaysare you getting rtp back to the grandstream?
15:10.01yojanlits a very strange problem, im no beginner in asterisk but this is something ive never seen
15:10.17yojanlhow can i check if the rtp gets back?
15:10.21hmmhesaysSpaceBass: is it trying to reinvite?
15:10.23yojanlincoming works
15:10.33SpaceBasshmmhesays, good question...would I want it to, or not/
15:10.36hmmhesaysyojani, you also canreinvite=no
15:10.43hmmhesaysnot
15:11.07hmmhesaysSpaceBass: call an echo test from the phone
15:11.08yojanli think ive tried reinvite, but i will do it no just 2 be sure, ive changed almost all settings :-)
15:11.08SpaceBasshmmhesays, its set to no on both wifi phones
15:11.21ManxPowerIf there is nat involved then reinvites will NOT work.
15:11.22hmmhesaysSpaceBass: call an echo test in asterisk
15:11.37hmmhesaysManxPower: no, they are unlikely to work
15:11.47hmmhesaysyou can get them to work with nat though
15:12.05ManxPowerhmmhesays, the chances are so slow "will not" is a good term.
15:12.31hmmhesaysfor your average joe yeah
15:12.35SpaceBassi just cannot understand why enabeling rtp debug would fix it
15:12.43hmmhesayswhat version of asterisk?
15:12.54yojanlhmmhesays: reinvite no makes no problem
15:12.56ManxPowerhmmhesays, have you ever gotten reinvites to work without doing portforwarding on the NAT routers?
15:12.57SpaceBass1.2.9.1
15:13.07yojanlGrandstream GXP2000 1.1.0.16
15:13.14hmmhesaysyojani: what?
15:13.20yojanlAsterisk 1.0.7-BRIstuffed-0.2.0-RC7k
15:13.30hmmhesaysyojani: makes no problem?
15:13.42yojanlhmmhesays: sorry, typing to quick. gives the same problem i meant
15:13.50yojanltimes out again
15:13.58hmmhesaysManxPower: never tried
15:14.07hmmhesayswhen I use reinvites and nat I have control of the nat devices
15:14.18ManxPower*nod*
15:14.40hmmhesaysyojanl: is the call even making it to asterisk?
15:14.58yojanlyes, the call is getting to asterisk
15:15.12yojanli can send you the sip debug output if you think it helps
15:15.19*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
15:15.23hmmhesaysput your your wan ip into the use nat field in the gxp
15:15.30SpaceBasshmmhesays, having problems with echo test currently
15:15.40hmmhesaysproblems specific to that phone?
15:15.43_GuhitI've got a problem with asterisk (asterisk 1.2.9.1, zaptel 1.0, FreeBSD) not hanging up with my X100P.  After I modified the BATT_DEBOUNCE, I can see that asterisk does get a message like Hungup 'Zap/1-1' when the far end disconnects, even during the times that it does keep the line engaged.  Any suggestions on how to force the card to hangup when it sees the "Hungup" message?
15:15.45hmmhesaysor .. problems in general
15:15.51SpaceBassin general
15:15.54hmmhesayswhat?
15:15.58hmmhesayswhy?
15:16.01SpaceBassits not implemented correctly in trixbox I think
15:16.18SpaceBassgot it working
15:16.28hmmhesayshow is that possible? trixbox compiles asterisk from source
15:16.47SpaceBassyeah but the feature codes are odd
15:16.49SpaceBassregardless its working
15:16.52SpaceBassand I was able to hear myself
15:16.55hmmhesayswhat'd you do?
15:16.57*** join/#asterisk anthm (n=anthm@000-446-609.area4.spcsdns.net)
15:16.57*** mode/#asterisk [+o anthm] by ChanServ
15:17.11*** join/#asterisk UlbabraB (n=filippo@host241-43-static.72-81-b.business.telecomitalia.it)
15:18.00SpaceBassi have canreinvite=no on the lan phone and the wifi (natted) phone
15:20.03yojanlhmmhesays: makes no difference :-(
15:20.10yojanlwan ip in the field
15:20.18hmmhesaysso it is working now SpaceBass?
15:20.26hmmhesaysyojani, odd
15:21.06SpaceBasshmmhesays, for the time being...but yeah
15:21.50*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
15:21.56yojanlhmmhesays: its very strange, i have a gxp in a differnet network which works fine, but that gxp has an old version of the firmware, i think its a firmware issue
15:22.02hmmhesayscould be
15:22.04hmmhesaysupdate it
15:22.11hmmhesaysI got 1.2.9 running on this wrt here
15:22.13ManxPowerI suspect the inconing call is not matching a sip.conf entry.
15:22.27yojanlthe stupdig thing is, the old version works, this gxp has the newest version
15:22.38hmmhesayswith a usb audio card running
15:22.43ManxPowerThe easy way to fix that is to put context=INVALID in sip.conf [general] then context=thecorrectcontext in each SIP device [section]
15:22.47yojanland since i cannot downgrade to the old version im stuck
15:22.52hmmhesaysi've turned my wrt into an ip phone bwhwhaa
15:23.11*** join/#asterisk mivck (i=1000@200.114.70.228)
15:23.16ManxPowerI turned my WRT into a WAP
15:23.21hmmhesayshaha
15:23.30yojanlthe most important thing is incoming and that works, but it irritates my very very much that i cannot get outgoing to work, and outgoing normally is the simplest!
15:23.45hmmhesayscurrently i have my running asterisk with usb audio and a linksys wip-300 wifi phone registering to it
15:23.48ManxPoweryojanl, paste the Dial command for outgoing
15:24.17mutis there a way to turn off events in the manager api?
15:24.23mutso i can send me commands and get results
15:24.26mutbut not see everything else
15:24.28ManxPowerDoes "sip show peers" show the correct external IP address for the NATed phone?
15:24.31SpaceBasshow is the wip-300? b/c the 330 sucks ass
15:24.43yojanlManxPower: that works, if I use my other phone, or the local soft client on my laptop i can call outbound... the trouble is the grandstream doesnt receive the answer from the asterisk server
15:24.51*** join/#asterisk naif (n=xyz@85-18-35-21.ip.fastwebnet.it)
15:25.00*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)
15:25.08*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co)
15:25.12ManxPoweryojanl, and "sip show peers" shows the correct IP address for the grandstream phone?
15:25.35yojanlyes, the right ip and right port
15:25.59yojanlcause incoming is working perfectly, so there can be a connection, outgoing is the headache
15:26.34yojanlcontext etc is correct, my guess is the grandstream does something new to the sip packages and in combination with this router or something the package doesnt get back
15:26.42yojanli only see invites coming, no ack going back
15:26.46hmmhesaysSpaceBass: i like the wip-300 what is wrong with the 330?
15:27.08ManxPoweryojanl, incoming can work just fine even if there is no ip address listed in "sip show peers" for that device.
15:27.15*** join/#asterisk postel_ (n=jp@unaffiliated/postel)
15:27.22SpaceBasshmmhesays,  long laundry list....no stun which makes it worthless while traveling...inside my house (at least now) its working
15:27.31SpaceBasswpa supports only 27bit keys
15:27.41yojanlManxPower, okay I didnt know that
15:27.42SpaceBassseveral of the buttons just do nothing.....theres lots of little bugs
15:27.42ManxPoweryojanl, does your Asterisk have more than 1 ip address?
15:28.04yojanlManXPower, I use normally one 1, but there are several ips on that server
15:28.22ManxPoweryojanl, do you have a bindaddr or anything like that in sip.conf?
15:28.27fourcheezeyojanl: just use 1 ip number
15:28.38yojanlive tried to work with the hostname, and the ip, now its set to the ip
15:28.48ManxPowerremove the bindaddr.
15:28.54yojanlin sip says: bindaddr=0.0.0.0
15:29.02ManxPowerremove the bindaddr.
15:29.07ManxPowerthere is no need for it in your case
15:29.09yojanlokay, going to try now
15:29.19fourcheezeManxPower: why remove it ? Why not set it to the IP number he's trying to access * with?
15:29.51ManxPowerfourcheeze, beacuse bindaddr only applies to the SIP signaling packets.  It has no effect on RTP, and that causes issues.
15:29.59*** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18)
15:30.00yojanlmanxpower: no difference
15:30.19ManxPoweryojanl, I'm STILL waiting for you to paste the Dial command from Asterisk.
15:30.22fourcheezeyojanl: are you seeing any messages about stale nonces?
15:30.54yojanlManxPower: you mean the Dial command in extenstions?
15:31.03yojanlfourcheeze: what is a stale nonce?
15:31.03ManxPoweryojanl, correct.
15:32.26ManxPowerThose look OK.
15:32.39fourcheezeyojanl: is the IP number you're talking to asterisk on the main IP number on your server?
15:32.46fourcheezee.g. eth0 not eth0:4
15:32.59yojanlfourcheeze: what is a stale nonce? I see Retransmitting #1 (NAT):
15:33.06*** join/#asterisk Gregabyte (n=greg@gateway.digium.com)
15:33.34fourcheezefourcheeze: you get a stale nonce when the client doesn't get the message to reregister
15:33.52fourcheezei.e. the 401
15:33.56yojanlfourcheeze: yes, its the main ip
15:34.02sp0n9e<PROTECTED>
15:34.02ManxPoweryojanl, does it start working for a few mins when you reboot the natted phone?
15:34.34*** join/#asterisk monkey13 (n=monkee13@69.7.217.155)
15:34.42yojanlManxPower: it doesnt work all day, the phone is only a few days already and nothing, only incoming calls working like a charm
15:34.42*** join/#asterisk }btorch{ (n=btorch@208.63.19.179)
15:34.50fourcheezehmm
15:35.25}btorch{hey guys is it possible to create a conference room and while in conference call someone and make the person part of the conference
15:35.50fourcheezeyojanl: have you tried a stun server?
15:36.04yojanlyes, i think the stun.fwd.net or so
15:36.11*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
15:36.14Bert-hi again
15:36.21hmmhesayswhoa bug in the cc app
15:36.21fourcheezeyojanl: have you tried removing it
15:36.27yojanlbut ive tried so many things, ill try it again
15:36.29yojanlits now removed
15:36.31fourcheezeok
15:36.33quid246what kind of billing scales well.. alot of what I've read seems to speak against AGI?
15:36.33fourcheezetry it in then
15:37.04Bert-I have a little issue : with some phones, when we hangup, asterisk don't hangup with the other side, and the channel is still open
15:37.23yojanlis stun.fwd.net a valid one? and should i set the account to: NAT Traversal (STUN):  yes?
15:37.43Bert-it hangs some time after, like 1 or 2 min
15:37.44yojanlstun.fwdnet.net is correct right?
15:38.09benjk}btorch{ create an extension that calls the conference room, then calls somebody and transfer them to that extension
15:38.20benjkcorrection, call somebody
15:38.37hmmhesaysquid246: define scales?
15:38.44benjkfish scales
15:39.13}btorch{benjk, yeah that's what i did
15:39.15fourcheezeyojanl: I've never used that one
15:39.30}btorch{benjk, do I waste channels when doing that though ?
15:39.38yojanli tried it now with that one, but i doesnt make any diffence
15:39.38benjkwhy?
15:39.40}btorch{guess not if I hang up the call right
15:39.52fourcheezeyojanl: it seems to work
15:40.16benjkwhen you transfer the called party to the conference, they will be directly "connected" to the conference
15:40.22quid246hmmmhesays:  well... when you are running 200+ SIP/IAX channels with billing... won't be a serious drain on resources.
15:40.33yojanlI see something now, if I debug the sip ip a line saying: SIP/2.0 488 Not Acceptable Here is there
15:40.39hmmhesayswhat type of billing?
15:40.47quid246prepaid
15:40.52hmmhesaysusing what app?
15:41.04*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)
15:41.23quid246That I am not certain about... I've looked at the current offerings, a2billing being the most "clean" looking app... but may end up just doing somethign myself
15:41.36hmmhesaysI wrote a simple prepaid app with no agi
15:42.00quid246yeah I remember you meantioning you used the DP
15:42.05hmmhesaysvery basic, but its got the basics a prepaid app needs
15:42.06yojanlso I guess that must be the trouble, asterisk or the phone says:
15:42.06yojanlSIP/2.0 488 Not Acceptable Here
15:42.16hmmhesayscould be easily built upon
15:42.21*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
15:42.29hmmhesayshttp://www.thelostpacket.org/tricks.php if you want to give a whirl
15:42.38quid246okay... lemme check it out
15:42.51hmmhesaysi need some suggestions anyway
15:42.51quid246nice domainname BTW
15:43.01hmmhesaysquid246: thanks <chuckle>
15:43.11[TK]D-Fenderyojanl: Thats a codec mismatch error
15:43.22*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)
15:43.59yojanl[TK]D-Fender thats strange, cause I have nothing specific set, is there a way to debug this further?
15:45.36yojanlthe prefered codecs in the GS are: PCMU, PCMA, G.723.1, G729A/B, GSM, PCMU
15:46.08yojanland in sip.conf ive got
15:46.20yojanldisallow=all allow=ulaw allow=alaw allow=gsm
15:48.18yojanlso in short, it looks like it fails because asterisk says: Not acceptable here? Am I correct?
15:48.38fileyojanl: full sip debug is good
15:49.31yojanlfile: how can I enable full sip debug?
15:50.05filesip debug
15:50.06file:D
15:52.06yojanlfile: okay, im already doing this :-)
15:52.28filepastebin it so others can see
15:52.39yojanlhow do I pastebin?
15:52.45file~pb
15:52.46jbotmethinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/
15:52.51fileuse pastebin.ca
15:54.28SpaceBasshmmhesays, and my one-way audio problem is back :)
15:54.35hmmhesaysSpaceBass: lovely
15:54.39*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
15:54.40*** mode/#asterisk [+o Cresl1n] by ChanServ
15:56.06*** join/#asterisk smackus (n=ckwall@63.149.122.93)
15:56.45*** join/#asterisk funxion (n=nunya@63.214.236.169)
15:56.57smackushow can I make it so that in a context of extensions.conf, if someone enters a non existent extension it transfers to a particular extension(receptionist)?
15:57.35*** join/#asterisk morex (i=morex@host86-134-128-75.range86-134.btcentralplus.com)
15:57.51trelane_exten => i,1,<something goes here>
15:57.52morexHello all
15:58.04hmmhesaysexten => i,1,Goto(sexy-stripper-voice,s,1)
15:58.05*** join/#asterisk ToyMan (n=stuq@ool-45784f3b.dyn.optonline.net)
15:58.06funxionhas anyone ever used ooh323c?
15:58.07morexI'm getting messages like these in /var/log/asterisk/messages
15:58.11hmmhesaysfunxion: yes
15:58.12trelane_smackus, please look into exten => i (invalid) and exten => t (timeout)
15:58.12hmmhesaysdialy
15:58.17hmmhesays*daily even
15:58.20Assidi want sexy stripper whose voice that is
15:58.23funxiondoes it work good
15:58.33Assidsup file
15:58.41hmmhesaysfunxion, I don't have much volume but it works well for me
15:58.42smackusthanks
15:58.59fileWELL, nothing... at least that's what I'm saying - what's really sup? who knows
15:59.01funxionhmmhesays does it work with the current release? and is it no longer included in asterisk-addons?
15:59.09sp0n9ewooohooo.
15:59.10sp0n9e:(
15:59.11hmmhesaysits not?
15:59.16yojanlthis is what happens: http://pastebin.ca/126540
15:59.27funxionI couldnt find it
15:59.34hmmhesaysum it is in 1.2.3
15:59.35_GuhitArgh!  Ok, I can't believe I wasted sleep over this last night.  It seems that the X100P was sharing the IRQ with the sound card and that was causing problems.  Unloading the sound driver seems to have fixed all my problems with asterisk not hanging up and caller id issues. !!!
15:59.39morexcdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files
15:59.46sevardmorex: that's the problem with vi.
15:59.51funxionyeah
15:59.52*** join/#asterisk boch (n=root@201.216.241.97)
16:00.06yojanlthe second not acceptable here comes because theres already a first invite, and reinvite is no i guess
16:00.11morexAny ideas?  I think it's resulting in dropped calls...
16:00.16hmmhesaysit works in 1.2.10
16:00.20Qwellmorex: turn up ulimit
16:00.22morexI think Asterisk is running out of file descriptors...
16:00.28morexQwell: I thought I already did.
16:00.37Qwellbefore you run asterisk, and in the same shell
16:00.42funxionwhy can I not find it in 1.2.10
16:00.45sp0n9eany ideas why ztcfg is segfaulting on me?
16:00.53morexI have ulimit -n 65536
16:00.54bochi want to send a fax from command line, some .jpg or .bmp file, is it possible? where do i start reading?
16:01.03Qwellmorex: -c I thought
16:01.04morexin my /etc/init.d/asterisk script
16:01.06morexAh...
16:01.20morexHelpfully debian doesn't supply a ulimit man page
16:01.22hmmhesaysboch, you using spandsp?
16:01.45hohummorex: that's because ulimit is built into your shell
16:01.46yojanlfile: do you see anything that can help?
16:01.52morexOK
16:02.04hmmhesaysfunxion, it is in asterisk-addons 1.2.3
16:02.10funxionok
16:02.14morexulimit -c specifies maximum core size.
16:02.23yojanlafter further studying i think the grandstream doesnt receive or doesn respond to the confirm, it just start another invite
16:02.27hohumfunxion: :P
16:02.27funxiondoes it install with addons or are there special parameters
16:02.36morexulimit -n is the maximum number of open file descriptors
16:02.43morexFrom http://www.ss64.com/bash/ulimit.html
16:02.44bochhmmhesays: dont know, you tell me
16:02.57morexAnd I'm using ulimit -n...
16:03.12bochhmmhesays: i never dealed with asterisk and faxes
16:03.14hmmhesaysyou can accomplish it with spandsp and a perl or php script
16:03.32yojanlfile: and others helping me, I have to go for half an hour (bring the girlfriend with the car :-) so be back in half an hour, hopefully someone know whats going on!
16:03.46hmmhesaysnail her
16:03.59filehmmhesays: that's always on your mind!
16:04.04hmmhesaysfile: duh
16:04.37bochhmmhesays: ok thanks ill see some spandsp docs
16:05.02hmmhesaysfor some monetary compensation I could have a working demo for you in a couple hours
16:05.14*** part/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil)
16:05.14bochhmmhesays: do i have to re-build * to support spandsp ?
16:05.23hmmhesaysno
16:05.30bochawesome
16:10.42sp0n9edoes sangoma have an irc chat room?
16:10.48funxionhmmhesays will addons 1.2.3 work with asterisk 1.2.10?
16:10.57Qwellfunxion: yes
16:10.57sp0n9efunxion: yes.
16:11.02funxionthank you
16:12.30*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
16:13.03funxionhmmhesays how would I got about installing ooh323c from asterisk-addons?
16:13.33funxionI cant seem to find much documetation on it
16:13.39funxioncan someone point me in the right direction
16:14.13eKo1There should be documenation in the asterisk-addons package.
16:14.37funxionI did a grep -e ooh323c *
16:14.39funxionnothing
16:14.56funxioneven in the channels dir
16:15.34hmmhesaysfunxion
16:15.38funxionyes
16:15.41hmmhesayswhat the crap man, it is in asterisk-addons
16:15.44hmmhesaysnot asterisk
16:15.55funxionI know
16:15.59smackushttp://pastebin.ca/126550 when i play a meesage from exten => 5313 it offers the option to enter an extension... can anyone tell me what I have messed up so that it ignores the extension entered?
16:16.03funxionin /usr/src/asterisk-addons
16:16.06charles___ohh my freaking head
16:16.09morexSo ulimit -n doesn't work with Asterisk?
16:16.12charles___so easyyyyy
16:16.14smackusnevermind.... I know what I am missing.
16:16.36funxionif I just install addons it will work
16:16.48funxionor do I need to do anything special for ooh323c specifically
16:16.54smackuswait... nope. ok, question still stands
16:17.57funxionhmmhesays does it require open_h323 and pwlib?
16:18.40smackusbased of my pastebin, should i just be able to dial any extension in the same context as what I have called into? 5313?
16:19.17smackusso if i dial 5313, and I hear the message if you know your extension dial it now... then I dial 900, should that now work?
16:19.26bochanyone familiar with Sipura2100 and faxes through * ?
16:23.09*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
16:23.28*** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110)
16:23.52*** join/#asterisk toerkeium (i=oo@201.216.206.221)
16:24.06websaefaxes through asterisk can be difficult
16:26.56hmmhesaysspandsp works ok for me
16:27.59bochalways?
16:28.28hmmhesaysalways?
16:28.38bochmy * only passes some faxes, not all
16:28.51hmmhesaysoh yeah?
16:28.54hmmhesayswhy is that?
16:29.05bochthats a good question
16:29.10eKo1Mine does the same. Faxing is not reliable on *.
16:29.15hmmhesaysmine seems to work reliably
16:29.27hmmhesayswhat is your fax setup like?
16:29.35bochdont know if is a bw problem, or what.
16:29.41eKo1Most of my faxes result in POOR LINE CONDITION
16:29.50hmmhesayshmm not mine
16:30.04macTijnwhat's your uplink? SIP ?
16:30.07websaei find it better to outsource faxing to a third party...
16:30.10macTijnand if so, what codec ?
16:30.14sp0n9edepends on if faxes get caught up in the echo cancellation.
16:30.20sp0n9efrom what i've read.
16:30.27websaeand not come within 10ft of it and asterisk
16:30.35funxiondoes ooh323c require open_h323 and pwlib?
16:30.39hmmhesaysI use spandsp and sip reliably
16:30.41hmmhesaysfunxion:  no
16:30.45macTijnhmmhesays: what codec ?
16:30.55hmmhesaysulaw
16:31.02funxionwill it just install by make and make install asterisk-addons?
16:31.05macTijnthat'll work, if there's no packet loss
16:31.06funxionand specialconfig?
16:31.10eKo1My fax machine is connected to a SPA2002 set to use ulaw which sends the call to our main * box which then sends the call to our * box that is connected to our PRI lines.
16:31.12hmmhesaysfunxion read the config files
16:31.24macTijneKo1: any transcoding ?
16:31.31hmmhesaystranscoding fax?
16:31.31hmmhesaysright
16:31.38hmmhesaysfunxion read the docs
16:31.40coppicefaxing over VoIP channels will seldom work, whatever codec you choose
16:31.52eKo1macTijn: no because the main * box sends the call to the PRI * box using ulaw.
16:31.59eKo1So it goes ulaw all the way.
16:32.01macTijncoppice: works with ulaw or alaw
16:32.03hmmhesayscoppice I fax over SIP/Ulaw all the time
16:32.07macTijneKo1: that's good
16:32.27coppiceif faxing over VoIP works, its by luck, not design
16:32.36funxionhmmhesays Im only finding info on jerjer's h323
16:32.36eKo1coppice: ditto
16:32.37*** join/#asterisk kpettit (n=keith@adsl-70-241-67-181.dsl.hstntx.swbell.net)
16:32.51hmmhesaysfunxion it is in the asterisk addon package
16:33.07funxionin /usr/src/asterisk-addons/doc#
16:33.09coppicemacTjin: the codec is only slightly relevant
16:33.10websaeit's the downside of getting business clients to switch over to VoIP
16:33.19macTijnfor fax
16:33.28macTijnworks quite nice
16:33.55bochfucking outdated ppl who uses faxes
16:34.00macTijnuhuh
16:34.08*** join/#asterisk Lead_one (n=wont@dslb-084-058-191-079.pools.arcor-ip.net)
16:34.11coppicemacTjin: that is also by luck, since the digum cards don't sync
16:34.26macTijnapparently it's the only way to transfer legal-aimed docs
16:34.57coppiceits sad the kind of crap that is acceptable as a legal document
16:35.06macTijncoppice: heh, true :)
16:35.21macTijn* raised my error rate on faxes
16:35.28macTijnwhich doesn't surprise me at all
16:35.43coppicetelexes used to be legally acceptable, and any fool could fake one in a few minutes
16:35.58macTijnheh
16:36.26macTijnfaxes are quite fun to do a man-in-the-middle attack with
16:36.45brodiemi hate fax
16:36.53bochwhats the reinvite method for ?
16:36.53*** join/#asterisk Kyler (n=chatzill@74.132.200.229)
16:36.54macTijnso do I
16:37.10macTijnbut working at an ISP is not doable without fax
16:37.16coppiceboch reinviting. what else?
16:37.29macTijnin .nl we still need a bunch of signed docs to register a .nl domain name
16:37.34bochcoppice: but it is codec related ?
16:37.58macTijnboch: no, more NAT related
16:38.03brodiemI have to listen to people complain every day because a fax failed
16:38.05bochoh ok
16:38.11coppicereinvite can be used to change codecs, change routing, change between voice and T.38 FAX and other things
16:38.15websaei say just setup a proxy to a 3rd party fax service for you customers
16:39.26websae*your
16:39.37funxionwhy can I not find ooh323c in any of the svn tags?
16:39.39websaethat way you don't even have to deal with fax problems
16:40.13KylerA long time ago I recall seeing something about performing actions (in the dialplan) after the calling (SIP) party has terminated the call.  I'm not finding that now.  Pointers?
16:40.23brodiemwebsae, will that work while being able to still use a physical fax machine so that docs don't haveto be scanned/e-mailed?
16:40.24macTijnhmm
16:40.26macTijnisn't there a set of extensions for asterisk that permit FoIP ?
16:40.43macTijnshould just be a modification of rxfax
16:40.50coppicenot yet
16:40.59macTijnin the works ?
16:41.08coppicein testing now
16:41.14KylerI'm coming into this fax discussion late but...Gafachi says they have T.38 service.  I've been meaning to try it with Asterisk.
16:41.16macTijncool
16:41.33funxionok
16:41.36funxionIm a dumbass
16:41.39funxionsry hmmhesays
16:42.23coppicebut we have found the T.38 passthrough stuff in trunk has problems with the real world. someone will have to sort that out, because I won't
16:42.51KylerSo my fuzzy recollection is that there's no way (or only a tricky way?) to do anything in the dialplan after the caller terminates the call.
16:43.02hmmhesaysextension h
16:43.12benjkwon't work in a macro though
16:43.19KylerNo, it's not that easy...is it?  I thought there was a problem with that.
16:43.20benjkmacros don't have h support
16:43.31benjkalso, only if the called party hangs up
16:43.33SpaceBassanyone know why simply turning rtp debug on would solve audip problems?
16:43.34KylerAh!  That's why it didn't work for me.  O.k., workaround...
16:43.46Kylerbenjk:  Thanks!
16:43.54macTijnheh
16:43.57benjkif the calling party hangs you can't trap that
16:44.02macTijnskype on asterisk hack: http://slacker.com/~nugget/projects/asterisk/page12
16:44.06macTijnquite funny solution :)
16:44.15nortexCan AGI call an asp page on a server?
16:44.16benjkI have a patch for app_macro that adds h support
16:44.26macTijnnortex: AGI can do anything you want
16:44.38benjknope
16:44.50sp0n9enortex: i would guess you would have to write some type of wrapper for the http request
16:45.24benjkhangup by the calling party immediately terminates the pbx
16:45.35benjkbut I have a patch for that too
16:45.51nortexI really just want * to send the information to the outside app, not really needing a response.
16:45.58benjkso far its only working for calling party hangup before the bridge
16:46.17benjkstill working on the part that traps hangup by calling party after the bridge
16:46.30sp0n9enortex: you'll have to write an AGI script to do that, or get a cgi module for asp and use that
16:47.23sp0n9e#!/bin/php -q\nfile_get_contents('http://example.com/foo.asp');
16:47.33sp0n9ewhere \n is a newline (and i forgot <?php
16:47.45sp0n9ethat's a quick wrapper to request the page
16:48.52*** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net)
16:49.32macTijnsp0n9e: that's not reading headers, you need to do that first
16:50.02*** join/#asterisk marcster (n=rubber@203.87.184.218)
16:50.16sp0n9emacTijn: why would i need to read the headers?
16:50.19marcsterhi. what is a device like sipura 3000 called?
16:50.21sp0n9enot using any of the information
16:50.28marcsteris it a modem?
16:50.36bochnow i can send from pstn to a SIP endpoint, but not backwards, what can be the problem ?
16:51.24macTijnsp0n9e: hm, wait, that's only if you are also sending commands to *
16:51.28*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:51.28*** mode/#asterisk [+o Qwell[]] by ChanServ
16:51.43[TK]D-Fendermarcster: Its an ATA
16:52.14macTijnAnalog Telephony Adapter
16:52.17[TK]D-Fendermarcster:  But for the fact it also has an FXO port (ATA is typically reserved for FXS devices) you could say "SIP gateway"
16:52.27blitzrageone way audio in an internal network is awesome!
16:52.30marcsteri see
16:52.43macTijnblitzrage: firewall ? :>
16:52.58macTijnor is it on purpose ? :)
16:53.00marcsterthanks. a google search on 'sipura ATA' the results i needed to get started.
16:53.17macTijn\o/
16:56.01*** part/#asterisk a1fa (n=a1fa@207.210.210.202)
16:59.43blitzragemacTijn: no FW on the internal network
16:59.58blitzrageI think asterisk is forwarding the RTP frames to a different subnet than what it should
17:00.05macTijnblitzrage: ahh
17:00.08*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
17:00.08*** mode/#asterisk [+o mog] by ChanServ
17:00.09macTijn<- gone
17:00.14macTijnl8r all
17:01.48[TK]D-Fenderblitzrage: Missing localnet clause is what it sounds like....
17:01.50*** join/#asterisk ajaymn (n=Ya@70.59.126.206)
17:03.10*** join/#asterisk nDuff (n=ccd@64.128.31.220)
17:03.54*** join/#asterisk ki2k (n=ki2k_@207.231.83.242)
17:04.07nDuffCan I specify a behavior to take place immediately whenever all members of a queue are already on the line?
17:04.13*** join/#asterisk Beighto (n=chatzill@64.160.113.130)
17:04.22ki2kanyone know if FWD.net is broken for new accounts?
17:05.20*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
17:09.05nortexIt still suprises me what sound files are missing, things like night mode or closed
17:10.02ki2kanyone know of a way to allow the use of the @ sign as part of a password?
17:10.35eKo1ki2k: good question. Maybe enclosing it in "" will help.
17:11.08ki2knope
17:11.18ki2ki'm thinking of using html encoding
17:12.29ki2knope, %40 doesnt work
17:12.42ki2kwelp, thats a bug
17:12.55ki2kcant use : and @ as part of your password
17:13.00ki2kin asterisk
17:13.46*** join/#asterisk [hC] (n=hardcore@190.10.12.97)
17:14.40sb_mxki2k, we use : as part of our sip peers' password
17:14.45eKo1ki2k: when you way password, what do you mean?
17:14.53eKo1s/way/say
17:15.09ki2kregister username:password@sip.server.com
17:15.27eKo1Ah. Try single quotes then.
17:15.31eKo1Maybe that'll help
17:15.37[hC]anyone have an example of having a recording play, instead of hearing a caller hear 'ringing'.  maybe something that says 'please wait while we connect your call'
17:15.53ki2ktry register username:'p@ssword'@server.com ?
17:15.59eKo1[hC]: that is pretty easy to do.
17:16.11eKo1ki2k: yeah
17:16.25[hC]Yeah, i imagine it is.. Is it a dial option, or do i have to use playback first?
17:16.27ki2keKo1: nope
17:16.41ki2kiax2 show registry doesnt even see it
17:17.10[hC]oh i can use special hold music too
17:17.12[hC]that works
17:17.15[hC]so it will auto-repeat
17:17.31*** join/#asterisk Gregabyte (n=greg@gateway.digium.com)
17:17.42ki2keKo1: know of any other ways to escape out the @ ?
17:18.01ki2kmy voip provider set up an account for me but he's now on vacation
17:18.03[TK]D-Fender[hC]: Give them actual MoH after playing the annoucment once
17:18.08ki2ki cant have him change hte passwd
17:18.11ki2ktill monday
17:18.38[hC][TK]D-Fender: yeah. now, playing the announcement once.. (i realize this probably sounds stupid but..) can I have dial play it, or do i have to hack it in some other way?
17:19.02[hC]I dont see any options in Dial() to do that.
17:19.06sp0n9eki2k: encode @ into %40
17:19.20sp0n9ep%40ssword
17:19.35sp0n9eit's a URI not a URL, but it works in some URIs
17:19.36[TK]D-Fender[hC]: Just sove it before your dial.
17:19.43[TK]D-Fendershove*
17:20.01*** join/#asterisk dasenjo (n=dasenjo@208.195.215.74)
17:22.47[hC][TK]D-Fender:  yeah that works, the only thing im curious about in that way, is if people dial extensions off an ivr, i just do an include of the local sip context... I guess i could hook like... _XXXX, play the thing, then goto ${EXTEN} of the local sip context.
17:24.59*** join/#asterisk Wazb^ (n=wazb@199.243.74.220)
17:25.04Wazb^hi
17:25.10*** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net)
17:25.11*** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br)
17:25.39[TK]D-Fender[hC]: Yup, you acn front-end it like that.  and on failed GOTO then send them to "i"
17:25.41Wazb^does anybody know any opensource wholesale billing with asterisk?
17:26.14BeightoIs there any way to put a delay on input dtmf?  I am having a problem where if someone dials a 4 digit extension it would go the a 3 digit extension and then go to an invalid option immediatly after.  Setting the digit timeout doesn't seem to effect this.
17:26.27ki2ksp0n9e: already tried it
17:27.08}btorch{hey guys I have some horrible echo now aftern upgrading my zaptel and asterisk to the latest
17:27.21}btorch{echo on iax2 to iax2 calls
17:27.58eKo1check the latency
17:28.16}btorch{how ?
17:28.51}btorch{I'm using a one ear piece headset (cheap one) and idefisk .. whenever I lower the speaker volume echo seems to go away
17:29.29}btorch{but if you try to hear it  really hard you can .... MG2 seemed to make the echo on my zaps worse too before I used MG1 i think
17:29.52smackusI am using addqueuemember to log my agents into the phones... one issue I have is that when on a call, the queue will still deliver the calls to that phone, even though it is not idle. is there a way to stop that?
17:30.03[TK]D-Fender}btorch{:  That's accoustic echo and has nothing to do with Zaptel.
17:30.29blitzrage[TK]D-Fender: the phones aren't talking to anything outside the network though -- seems a canreinvite=no resolves the one way audio (phones are SIP, but only talk to other SIP phones on the LAN or to chan_zap)
17:30.29[TK]D-Fendersmackus: Using Local channel?
17:30.58Wazb^does anybody know any opensource wholesale billing with asterisk?
17:30.59[TK]D-Fenderblitzrage:  You implied it spans subnets (even though local).  My guess is the phones don't know how to route back.
17:31.00hmmhesayswow did I drink waay too much last night
17:31.24*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
17:31.27Beightoanybody using voxee?
17:31.35blitzrage[TK]D-Fender: the phones are on the same subnet though -- I was just theorizing that Asterisk was getting confused and was sending the audio out a different route the phones couldn't be reached at
17:31.36smackus[TK]D-Fender: I dont know how to do that
17:31.39[hC]I wonder if its cause ive mucked with the tonezone stuff here.. Calling users dont hear ringing indication when * is ringing sip phones
17:31.40[hC]very odd
17:32.03blitzrage[TK]D-Fender: I might still try the localnet in there later to see if I can get reinvites to work
17:32.16*** join/#asterisk Akhilesh (n=akhilesh@203.76.184.38)
17:32.24blitzragebut thus far at least I've got a production box working again
17:32.26}btorch{[TK]D-Fender, I believe that is true on the iax <-> iax calls but what about a call that comes in from a zap line to a meetme room
17:32.43*** join/#asterisk feld_ (n=feld@12.148.212.157)
17:32.49*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
17:32.49*** join/#asterisk riddlebox (n=blah@24-207-167-238.dhcp.stls.mo.charter.com)
17:32.49*** mode/#asterisk [+o anthm] by ChanServ
17:33.01[TK]D-Fenderblitzrage: That won't help on re-invite.  The phone still needs to know where to go.
17:33.22[TK]D-Fender}btorch{: Who in the room gets echo?
17:33.24feld_if I park calls I dont get any info from asterisk telling me where I parked it -- I'm just starting to try to config this feature... all I have is include => parkedcalls in my extensions.conf and briefly looked over features.conf
17:33.31feld_what am I missing? I dont see it in the docs.
17:33.35}btorch{maybe I shouldn't have added aggressive suppressor and changed to MG2
17:33.40}btorch{everyone
17:33.54AkhileshHi guys, I need some information.
17:33.56}btorch{[TK]D-Fender, the zap caller hears his own echo
17:34.00[TK]D-Fenderfeld_:  Screw Parking (at least the one that comes with *).  Get app_valetparking from pbxfreeware....
17:34.01brodiem}btorch{ doesn't agressive just cause it to go half duplex?
17:34.09}btorch{[TK]D-Fender, and me (iax) and the other (iax) user too
17:34.11[TK]D-Fender}btorch{: When he's ALONE?
17:34.14AkhileshAs asterisk works only on linux, is there PBX on windows ?
17:34.14feld_[TK]D-Fender: i'll check it out thx
17:34.26}btorch{[TK]D-Fender, did not test that
17:34.31}btorch{will do that .. brb
17:34.48ki2kAkhilesh: i'm running asterisk on a windows server
17:34.55ki2kAkhilesh: Microsoft Virtual Server
17:34.57ki2khahahaha
17:35.01ki2kwith ztdummy
17:35.08AkhileshWhere did u get binaries on Windows ?
17:35.20ki2kI didnt
17:35.21Akhilesh:) Ki2K.... I need it
17:35.28Rawplayero god..
17:35.29[TK]D-FenderAkhilesh: LOL.  There are no "windows" binaries.... its all virtualized Linux....
17:35.34ki2kyep
17:35.35sp0n9eyep
17:35.36ki2kit works
17:35.47sp0n9eyou can get MS Virtual PC for free now
17:35.52ki2kyep
17:35.53sp0n9edon't know how well that would run it
17:36.04ki2ksp0n9e: testing now
17:36.05smackus[TK]D-Fender: ok, so is this what you are pointing me to? AddQueueMember(MyQueue,Local/123 at my-context) )
17:36.08ki2kit seems to be ok
17:36.16ki2kit's a dual core ath 64
17:36.50AkhileshIs there soft PBX which is natively compiled on windows ?
17:36.51sp0n9enow if only i could get my sangoma card working :)
17:37.25eKo1Akhilesh: * + cygwin
17:37.39blitzrageyou'll just have timing issues regardless in a virtualized environment
17:37.40sp0n9ehow's it run on cygwin?
17:37.57ki2ki dont even wanna bother with cygwin
17:38.00*** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com)
17:38.07ki2kAkhilesh: why do you want a windows machine?
17:38.14AkhileshNative ? Without cygwin ?
17:38.21sp0n9ehttp://www.asteriskwin32.com/ <<< looks pretty bad.
17:38.53[TK]D-FenderAkhilesh:  Forget about * on windows.  You want a windows VoIP server?  Odds are it will cost you and not be this flexible.,
17:39.11smackusis there a way to do something like this: exten => *1,2,AddQueueMember(test1,Local/<some variable for device>)
17:39.41[TK]D-Fendersmackus: Clarify that var....
17:39.42feld_ARGH...
17:40.02feld_my boss bought Grandstream phones after I specificly told home to get Polycoms
17:40.10[TK]D-Fenderfeld_:  I've recently implemented the valetparking and it just rocks.....
17:40.16[TK]D-Fenderfeld_:  ZOMG!!!!
17:40.20feld_these things suck ass and they dont send DTMF right
17:40.28feld_they are 101 in the config and they wont work with voicemail
17:40.54smackus[TK]D-Fender: am i following you correctly on using Local channel to correct my queue issue?
17:41.59[TK]D-Fenderfeld_:  Get them returned...
17:42.20[TK]D-Fendersmackus: Not sure.... not a qualified sample fo code to examine...
17:42.44bcnl[TK]D-Fender: valtparking?
17:43.15bcnlI still no longer have MoH when ppl are parked, I get a codec translation error, eventhough the calls are ulaw
17:46.39*** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net)
17:46.55ki2kfeld: fire your boss
17:48.38[TK]D-Fenderfeld_:  Just return them.  End of story....
17:50.38*** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
17:52.42smackusok... the local thing is the answer to my queuing issue, thanks [TK]D-Fender:
17:52.50smackusnow i just need to get the correct variable set up :-D
17:53.06smackusi have it so that it uses whatever i dialed to be the extension... which I dont want.
17:53.15ki2kso anyone else have any ideas about my password problem?
17:54.52feld_ki2k: he already breaks enough stuff around here... and he has some financial issues that just surfaced to us employees, so i will be leaving as soon as i get another job lined up
17:55.26feld_[TK]D-Fender: found that dtmfmode=auto fixes the issue... works for these phones and out xlite phones, so it will suffice... but man... just reading the asterisk wiki on these things makes me want to puke.
17:55.34feld_*and our xlite
17:56.05jsaunders* You were kicked from #yate by killall-9 (killall-9)
17:56.05jsaunders-
17:56.05jsaunders#yate unable to join channel (address is banned)
17:56.14jsaunders:D  Heheh.  Wow, she really hates me.
17:56.25filewhat did you do?
17:56.33fileand please don't do it here
17:56.34file:P
17:56.36jsaunders:)
17:57.34jsaundersI don't know what I did.  I was given no reason.
17:58.38jsaundersI suspect I know why...  because of differing opinions on a certain topic between Diana and myself.  Was just a little suprised to see it from killall-9.  Curious as to wether it was him who actually did it, or just her using his account.
17:58.45feld_rofl my cow-orker changed his ringtone to Duke Nukem's "Get back to work, you slacker!"
17:58.58jsaundersHeheh, beauty.
17:59.01jsaundersI loved that game.
17:59.18feld_yeah I cant wait for DNF to be released. Actually, nobody can wait.
17:59.51jsaundersOh?  DNF you say.  News to me.  *googles*
17:59.52mogjsaunders, what was all that about?
18:01.00*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
18:01.00ki2kDNF? ha
18:01.11jsaundersDiana claims to have given me all this help with regards to our skySwitch project.  Apparently, according to her, we had an agreement that I would release our code if she helped us.  That was never the agreement, yet she's adamant it was.  I have several logs of conversations clearly showing otherwise but she's ignorant & stubborn and truly believes she is right.
18:01.49jsaundersIf I was spiteful I'd put up a public website w/ the logs but alas...  I don't think that would do much.
18:02.06mogahh
18:02.12mogwas funny argument
18:02.18jsaundersIt's beyond all of that anyways.  She simply doesn't like me now cuz I argue with her.  :)
18:02.24jsaundersOh, you saw?  Heheh.
18:02.29mogyeah
18:02.34jsaundersMan, that's one of many, let me tell you.
18:02.38mogim in all pbx channels
18:02.48jsaundersRediculous.  It's been going on for over a half a year now.
18:02.50jsaundersHeheh
18:02.59jsaundersI try and be in all.  Down 1 now.
18:03.05jsaundersAh well, that's what bnc is for.
18:03.22nDuffMy SIP phones appear not to be informing the server when the user is busy, but instead trying to push the incoming call through on an additional line. Is this typically fixable behavior?
18:03.22filemog: you spy you!
18:04.09jsaundersmog: Did she say anything after my last comment and before I was kicked?  I don't have the channel logged and my client closed the window when kicked.
18:04.10puppet20:01:09 < diana> hi puppet
18:04.14puppet;P
18:04.29mognothing really
18:04.33jsaunderskew, tnx mang
18:05.05smackusbased on the wiki... (${EXTEN}: The current extension ) what do i use to setvar to the extension i am dialing from?
18:07.08feld_what the hell
18:07.10feld_load average: 13.41, 5.14, 1.96
18:07.29feld_^^ from when I put 4 grandstream phones on the system
18:09.00jsaundersheheh
18:10.51*** join/#asterisk vpanayotov (n=vdp@213.91.154.185)
18:11.46Seba_soyhello, anyone has configure realtime sip in astrisk?
18:11.47*** join/#asterisk ^Xypher (n=bentley@arnor.fornost.com)
18:12.16*** part/#asterisk ^Xypher (n=bentley@arnor.fornost.com)
18:12.21quid246seba; yeah, just follow the wiki... pretty easy
18:12.37Seba_soycan I register a sip account using realtime?, I can make calls but I can't registre the user
18:13.02quid246seba;  you mean create a user in the SQL table?
18:13.07*** join/#asterisk jeebusmobile (n=jeebusmo@12.180.154.130)
18:13.32*** join/#asterisk Un1x (i=Sean@CPE123456789123-CM0011ae8a7b0a.cpe.net.cable.rogers.com)
18:13.33Seba_soywell, I have x-lite, so I want to register it so it can see if it have voicemail and all that...
18:13.59quid246then insert it into the table
18:14.05Seba_soyI have it inserted...
18:14.36quid246if it's in the table & x-lite is configed properly, then check your DB settings in extconfig
18:15.03Seba_soyWhen I try to register, asterisk said NOT FOUND, but if I send a call directly, it goes OK, it uses extensions and all config I put on DB
18:18.27javarwhen modify manager.conf, i need reboot the server?
18:18.53*** join/#asterisk Ixitxachitl (n=m@209.151.130.10)
18:18.59Seba_soydone
18:19.16Seba_soyI have to replace type=user by type=friend
18:19.18Seba_soy:)
18:19.20Seba_soymy mistake
18:20.57smackusok... what am i doing wrong? http://pastebin.ca/126692 I need to have the variable for Local/$<variable> show the extension that I am logging in from how can I do that?
18:21.22benjk${variable}
18:21.30benjknot $<variable>
18:21.57IxitxachitlWondering if anyone has any ideas: i've got a PRI connected to digium card, and inbound calls can be received and heard just fine. However, outbound calls are garbled to the point of incomprehension, but only to the phone calling out (the receiving line hears audio properly)
18:22.00smackussorry... what am i doing wrong? http://pastebin.ca/126692 I need to have the variable for Local/${variable} show the extension that I am logging in from how can I do that?
18:23.44*** join/#asterisk NDT (n=noone@cpe-24-195-66-214.nycap.res.rr.com)
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18:24.08sb_mxsmackus, what version of * are you running? i've found trunk doesnt support SetVar anymore
18:24.39*** join/#asterisk _deg_ (n=deg@200.163.193.247)
18:24.45smackusAsterisk 1.2.10
18:26.37sb_mxsmackus, found your error
18:27.34sb_mxchange the AddQueueMember line to exten => *3,3,AddQueueMember(test1,Local/${CHANNEL}) and the next ones add +1 to their priority
18:28.27smackusahhh crap
18:28.50puppetANyone know how it is with g729 in europe? do you need to pay license here since it is software patent?
18:29.04benjkyes you do
18:29.14ki2khow does the licensing work?
18:29.23benjkyou purchase the licenses from Digium
18:29.26ki2kany why havent anyone come up w/ a open src version?
18:29.27puppetwell
18:29.30puppetthere are other ways
18:29.35puppetthe intel open729 thing
18:29.36benjkbecause it is patent encumbered
18:29.48puppetwell there are no software patents in .eu
18:29.51ki2kwhat intel thing?
18:29.55benjkyou have to pay the royalties to the patent holders no matter if its open source or not
18:30.03benjknothing to do with software patents
18:30.04*** join/#asterisk mkrufky (n=mk@68.160.103.77)
18:30.04puppethttp://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
18:30.11c4t3lhence lic fees
18:30.16benjkfor educational purposes only
18:30.24benjkany other use is illegal
18:30.30ki2kah ok
18:30.33c4t3lhence lic fees
18:30.37benjkEU has signed trade agreements with the US
18:30.38puppetwell isnt it really software patent? since it is a software they want lic fees for
18:30.50benjkthat means all US patents are valid in the EU through the backdoor
18:30.56puppetgah
18:30.59puppetthat sucks :(
18:31.07benjkand vice versa
18:31.09*** part/#asterisk Ixitxachitl (n=m@209.151.130.10)
18:31.14puppetwell well, then i just have to pay up and increase the thought price
18:31.38smackusit is still trying to set that variable to the extension *1. which is the dialed extension, not the extension i am dialing from.
18:31.45benjkor avoid using patent encumbered codecs in the first place
18:32.05benjkthere are very good free codecs out there
18:32.15puppetbenjk: speex?
18:32.20c4t3lyes
18:32.23benjkSpeex has the same low bandwith footprint and quality
18:32.27justinu|laptopbenjk: you live!
18:32.35benjkhi justin
18:32.41justinu|laptophow the heck have you been?
18:32.43puppetso i should forget about g729 and go speex?
18:32.56benjkyes, if you can
18:33.10ki2kalso, since codecs are so cpu intensive, why hasnt people made coprocessor cards that sit on pci/pci-e or even on the digitum cards
18:33.12benjkI was busy and away
18:33.21puppetki2k: thats true
18:33.25puppetki2k: should be good
18:33.36benjkyou never called me up, did you actually come to Japan after all?
18:33.41justinu|laptopno :(
18:33.54ki2kthats why dedicated pbx's are faster, dedicated hardware
18:34.31c4t3lki2k: you have to get hardware manufacturers motivated to do so
18:34.48}btorch{[TK]D-Fender, wierd the echo did go away once us both iax client disconnected from the meetme room and connected back using a regular phone
18:34.49benjkkik2, Digium has just announced a g729/g723 PCI card
18:34.50puppetc4t3l: well why cant digium do it? ;P
18:35.01*** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au)
18:35.02cybertrickleI am trying to install zaptel, and its not making the .so files (thus I cant modprobe it). Anything I can try ?
18:35.09c4t3l:D
18:35.39filethe TC400P
18:35.43benjkyep
18:35.45}btorch{[TK]D-Fender, maybe IDEFISK isn't doing such a good job on echo cancelling or maybe it may be the echo acoustic due to the cheap headset
18:35.48ki2kbenjk: is support in asterisk yet?
18:36.11benjkof course if its from Digoum it will be for Asterisk
18:36.22benjkbut its not shipping yet
18:36.24ki2kyet as in already done
18:36.31ki2kalready in the current distro
18:36.31benjkI heard it will ship within a month or so
18:36.50filecurrent distro? some of the components required for it are in trunk
18:37.12ki2kah ok
18:37.25ki2kanyway, what's w/ the mmx optimizations?
18:37.36ki2kthem not compatible w/ AMD cpus?
18:38.34cybertrickleWhen you make zaptel, the drivers are in the .so files, not .ko files right ?
18:38.45filecybertrickle: uh... no
18:38.53brodiemdepends if you're on 2.4 or 2.6
18:38.55smackusso i am dialing *1 from extension 3562. I want it to log in Local/3562, not *1... what am I doing wrong? http://pastebin.ca/126707
18:39.18cybertricklefile, ko then ??
18:39.32fileko for 2.6, o for 2.4 I believe - but I haven't been on a 2.4 system in a longggg time
18:39.40brodiemthat is correct
18:39.48*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)
18:41.06cybertricklethese ko files go in /usr/lib/asterisk/modules, correct ??
18:41.11sb_mxsmackus, you want it to go to Local/{the-extension-you-dialed-from} right?
18:41.18smackuscorrect
18:41.31filecybertrickle: no, those are for Asterisk modules
18:41.34sb_mxsmackus, instead of {CHANNEL} use {CALLERID(num)}
18:41.42cybertricklefile,  where then ?
18:41.43filecybertrickle: all you should need to do for zaptel is make and make install, and the modprobe to load the modules
18:41.54*** join/#asterisk dasenjo (n=dasenjo@208.195.215.74)
18:42.01fileprovided your install is capable of building kernel modules
18:42.09cybertricklefile,  modprobe cant find them. But I do see them in the zaptel folder.
18:42.39cybertricklemodprobe says "FATAL: Module wct4xxp not found.
18:42.39cybertrickle"
18:42.45filedid you do make install?
18:42.49cybertrickleyeah
18:43.18filewhat distro?
18:44.29*** join/#asterisk |dennis| (n=dennis@200.32.215.82)
18:44.29*** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net)
18:45.00cybertricklefile, Fedora Core 5
18:45.24filehave you called technical support to get your installation support? they know the tips'n'tricks for the different distros
18:45.27ki2kwhats wrong w/ fedora?
18:45.42cybertricklefile, no I have not.
18:46.10yojanlhi im back, anyone seen my pastbin?
18:46.14cybertricklefile, if I copy the .ko files manually to the kernel modules folder I get the error "WARNING: Error inserting zaptel (/lib/modules/2.6.15-1.2054_FC5smp/zaptel.ko): Invalid module format"
18:46.25cybertricklefile, Ever seen that error before ??
18:47.11fileI don't use FC5, just Debian but people may have had that problem before... it would be on voip-info.org, the mailing lists, or wherever
18:47.19fileand technical support has problem encountered it with someone
18:47.26smackusok, CALLERID(num) will not work either, cuz its a call center..  all of the caller ids are set to the same number. i needs to be extension.
18:48.19smackusthere has to be a variable that is for the device called from but what I have done it does it as SIP/ I need it to be local.
18:49.16sb_mxsmackus, this is for inbound or outbound calls
18:49.35sb_mxsmackus, or internal calls for that matter (ie: apps, ext2ext, etc)
18:49.36smackusinbound. using channel I can get it this close: Added interface 'Local/SIP/3562-0de3'
18:49.47smackusit is an inbound queue
18:50.12sb_mxsmackus, and instead of SIP/3562-0de3 you only want the 3562?
18:50.20filesmackus: setvar in sip.conf for each device, export a variable to the dialplan which is like the Local string to reach them or something
18:50.57smackushmmm... ok interesting.
18:51.02smackuslemme try a few things.
18:51.15filesetvar=MY_AGENT_CALLBACK=100@agents
18:51.26fileAddQueueMember(support,Local/${MY_AGENT_CALLBACK})
18:51.58yojanlI get this: Via: SIP/2.0/UDP [[my wan ip]]:6050;branch=z9hG4bKde22e969cb66db45;received=[[my wan ip]];rport=33297
18:52.09sb_mxcybertrickle, do you get an error message when compiling zaptel?
18:52.12fileit's a Via header, determines the path the packet took
18:52.24filepacket... SIP message... meh
18:52.35cybertricklesb_mx, nope
18:52.36yojanlwhat is the last rport? Maybee this is my problem, in a succesfull request the last rport is the same as the first port
18:53.27yojanlsuccessfull: Via: SIP/2.0/UDP [[wan ip]]:6100;branch=z9hG4bKbb378c42e1365306;received=[[wan ip]];rport=6100
18:55.24yojanlcan I change a setting in my grandstream so it will trigger a differen rport?
18:56.49*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:57.24smackusfile: so the setvar=MY_AGENT_CALLBACK would be done in the extensions.conf? or on each entry in the sip.conf?
18:57.31filesip.conf
18:57.35smackusawesome, thanks
19:02.12*** join/#asterisk [hC] (n=hardcore@190.10.12.97)
19:03.43AkhileshI need some small help
19:04.13SkramXInteresting.. a client wants me code a web interface and asterisk type script to work with http://liarliar.sourceforge.net/ ..
19:04.22SkramXcould be cool... could be super hard
19:04.31*** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
19:04.39fileAkhilesh: asking a question goes a long way towards getting help
19:05.09Lyfeanyone have any ideas why i can receive a call out of a standard fxo card but can't place one?
19:05.10AkhileshSince I am testing on a single machine,  I can use only one softphone. But to test, is there anything I can do ?
19:05.11Rawplayerhow does * perform on sparc64 compared to x86?
19:05.13*** join/#asterisk ionix (n=ionix@p3101-ipbfp05miyazaki.miyazaki.ocn.ne.jp)
19:05.23AkhileshCan I hear my own voice after some delay ? Like delayed echo ?
19:05.28*** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131)
19:06.41ArkonaDev-MikeAnyone played with the IRC server imbedded into Fonality's Asterisk solution?
19:06.55[TK]D-FenderDelayed echo.. is that like a redundant repetition? ;)
19:07.07*** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net)
19:07.27AkhileshI mean, I want to setup a SIP user , to whom if I call I should hear back after some delay, say 3-4 seconds.
19:08.01[TK]D-FenderAkhilesh: Depends on the latency of the connection for Sip>SIP
19:08.24AkhileshI wanted to test how it works... I am a total newbie to this voice stuff.
19:08.31AkhileshI have setup two users
19:08.51c4t3lhello
19:09.02AkhileshI registeresed on sip and connected using x-Lite.
19:09.09Akhilesh*registered
19:09.56AkhileshThen, I dialed some random number, it said "The person u r calling is unavailable. Please try again."
19:10.18[TK]D-FenderAkhilesh: Means that you are not dialing a valid #
19:10.48[TK]D-FenderAkhilesh: that message is created by X-Lite, not #
19:12.56Akhilesh# ? Whats that ?
19:13.05*** join/#asterisk x86 (n=x86@p3m/member/x86)
19:14.32ArkonaDev-MikeAnyone ever user Fonality's HUD software?
19:16.39NDTArkonaDev-Mike: Saw screenshots never had any use for it though so never tried it
19:16.44Akhileshoops... my context in sip.conf was wrong :)
19:16.52AkhileshThanks D-fender.
19:17.05file[TK]D-Fender gets a cookie!
19:18.20*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
19:18.22[TK]D-Fenderarf!
19:20.57[TK]D-FenderMeant to say * on the 2nd one there...
19:21.02*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
19:21.02*** mode/#asterisk [+o anthm] by ChanServ
19:21.20[TK]D-Fenderanthm:  Killed that peer good you did!
19:21.30anthmlol
19:22.09cybertrickleI got past my zaptel issues. I upgrade the kernel
19:22.10[TK]D-FenderThats the best part about my God complex.... no peer pressure :D
19:22.30Lyfeanyone have any ideas why i can receive a call out of a standard fxo card but can't place one?  (hopefully?)
19:22.30cybertrickleHas anyone ever had issues doing asterisk -r after its running ?
19:22.45[TK]D-FenderLyfe:  Show use your dialplan.
19:22.55Lyfeexten => _91NXXNXXXXXX,n,Dial(Zap/1/${EXTEN:1})
19:22.55Lyfeexten => _91NXXNXXXXXX,n,Hangup
19:22.55Lyfeexten => _91NXXNXXXXXX,n+100,Hangup
19:22.56[TK]D-Fenderus8
19:22.58[TK]D-Fenderus*
19:23.08[TK]D-FenderLyfe:  PASTEB IT.  ALL OF IT
19:23.19[TK]D-Fender~pb
19:23.22jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/
19:23.24[TK]D-Fenderdfsfhjvhnhnfasdlkjnhkdrnuqwe9rityvnioufytvg
19:23.26[TK]D-FenderBLARG
19:23.31Lyfewoops, forgot the noop on the top.
19:23.51moghey i have a question
19:23.52*** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org)
19:23.57moganyone here know stuff about hints
19:24.03Lyfed-fender: is there a preferred place to paste 5 lines?
19:24.06[TK]D-Fendermog : I do, what of them?
19:24.11*** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu)
19:24.13filemog: I'll give you a hint...
19:24.18mogooh thanks
19:24.19[TK]D-FenderLyfe: pastebin.ca
19:24.37fileut roh
19:24.47[TK]D-FenderYeah file.... you work there... they have exclusive rights to you!
19:25.40Lyfed-fender: http://pastebin.ca/126750
19:25.48brif8I have a remote SIP phone (cisco 7960) which has registered with the local * system, as ssen by sip show peers. (1) the phone can dial an extension on the * box and place an outbound call.  (2) The phone will ring when it's extension is dialed but NO voice on incoming calls. and I get "SIP response 481 "Call Leg/Transaction Does Not Exist"  What am I missing
19:26.16shmaltzLify, change the dial line to read:
19:26.18shmaltz_91NXXNXXXXXX,n,Dial(Zap/1/ww${EXTEN:1})
19:26.39Lyfeww?  interesting.
19:27.25shmaltzLyfe, it inserts a pause, so that dialtone is present before it dials.
19:28.51shmaltzLyfe, did that fix the problem?
19:29.02brodiemanyone recommend a good CRM w/ easy * integration?
19:29.17Lyfei need a moment, to test this better, cause now it's sitting there, apparently.
19:29.19*** join/#asterisk aydiosmio (n=aydiosmi@65.213.70.43)
19:29.31shmaltzbrodiem, define easy
19:30.04aydiosmioyou should use the User Logon to join an extension to a queue right? My phone keeps saying my extension is unavailable?
19:30.38shmaltzaydiosmio, your phone? then take it up with the manufacturer of your phone
19:30.57aydiosmioI mean the extension, asterisk reports it
19:31.06shmaltzohic
19:31.16shmaltzaydiosmio, what does *asterisk* say?
19:31.30aydiosmiophone 200 is currently unavailable
19:31.39yojanlokay, one try again. I have a GXP-2000 and in this network I can receive calls but I cannot place calls, I have tried an other gxp-2000 here which works in a different network but i doesnt work here, the strange thing is that in this network if I try to call with firefly_thirdparty in the failing network i can place calls. The problem lays I think in the GXP with the Network here. This combination fails. What can I check?
19:31.43shmaltzaydiosmio, thank you, pb
19:31.46shmaltz~pb
19:31.48jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/
19:31.50*** join/#asterisk e-MAC (n=YMironek@82.207.61.237)
19:31.50[TK]D-Fenderaydiosmio: Clarify what he perceives as "unavailable".
19:32.38yojanlso outgoing is failing and I cant figure out why, incoming works like a charm
19:32.46shmaltzyojanl, what does your network topology look like?
19:32.59shmaltzyojanl, pb your sip show peers
19:33.18aydiosmiothe featurecode plays back is-curntly-unavail.gsm
19:33.22aydiosmiothat's it
19:33.36shmaltzaydiosmio, that means that asterisk is operating nicely
19:33.38shmaltz:P
19:33.39aydiosmiomy softphone is registered and I can dial to it
19:33.42yojanlthe asterisk server is on a private ip, then I have the wan with a router which forwards the udp ports to the static interal ip of the grandstream
19:34.02shmaltzyojanl, again your sip show peers
19:34.02brif8For those who know sip debug here is the SIP debug peer 650 I get   http://channels.debian.net/paste/3409
19:34.02Akhileshhow to read voice mail >?
19:34.07[TK]D-Fenderaydiosmio:  Show use the CLI output leading to this error....
19:34.14AkhileshI dialed 8500, then it asked for my mailbox.
19:34.24[TK]D-FenderAkhilesh: "show application voicemailmain"
19:34.49AkhileshI pressed 5190 as in sip.conf, I have written mailbox=5190
19:35.09AkhileshThen it asks for password, how do I dial password as my password is teststring !
19:35.32yojanlhttp://pastebin.ca/126768
19:35.38[TK]D-FenderAkhilesh: that makes no sense... its what you put in voicemail.conf
19:35.56Akhileshnothing
19:35.59*** join/#asterisk Bullseye_Network (n=info@216.143.192.69)
19:36.18shmaltzyojanl, which one is the gpx?
19:36.21Akhileshoops, realized my mistake.
19:36.25[TK]D-FenderAkhilesh: Go read up on how to configure your mailbox in the first place....
19:36.30yojanlits the zs1
19:36.32AkhileshThanx D-Fender.
19:36.34yojanland the zs2
19:36.56[TK]D-FenderAkhilesh: Great idea to DL the book first.  It will show you the basics you may have skipped.
19:36.57[TK]D-Fender~book
19:36.59jbotbook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
19:37.06shmaltzyojanl, on a different network, by that you mean to this asterisk, or another asterisk?
19:37.42Lyfeshmaltz: unfortunately, adding the ww in front of the number did not solve it.
19:37.48yojanlline 1 & 2, in my dialplan I can dial the sipura without problems with firefly_thirdparty in this network, but when I call in this network with the grandstream it fails
19:37.56shmaltzLyfe, what do you get on the line?
19:38.02aydiosmiooh, I'm not set up as an agent.
19:38.02shmaltzand what does the CLI report?
19:38.24Lyfei get a fast busy signal, and the only thing i see from the CLI is what's in the second part of the pastebin url i gave.
19:38.26shmaltzyojanl, can you read and answer what you are asked?
19:38.49yojanlshmaltz: asterisk is an external network
19:39.16yojanlbut on a public IP
19:39.17shmaltzyojanl, thats *not* what I asked last
19:39.27Lyfe('cept that now there's "ww" in front of the phone numbers)
19:39.28shmaltzyojanl, on a different network, by that you mean to this asterisk, or another asterisk?
19:39.44yojanli have only 1 asterisk
19:39.55yojanlthat asterisk is on a public internet server
19:40.02yojanli am behind a router
19:40.10shmaltzyojanl, so those phones work on a different network with the *same* asterisk?
19:40.13yojanlyes
19:40.36shmaltzyojanl, what does sip debug show?
19:40.52shmaltzdo a sip debug peer zs1, try making a call from zs1, then pb the cli
19:41.12yojanlk
19:41.13yojanlhold on
19:41.33brodiemshmaltz, didn't see your response.. easy as in I don't have to develop my own code to listen for manager API events
19:41.40ki2kanyone know what docs digium gives you when you buy their business package? what's in those neat binders?
19:41.53shmaltzbrodiem, try FOP
19:42.12Lyfewish i was simply bored.. i'm merely confused.
19:42.20brodiemi know FOP can launch a CRM, but isn't a CRM itself
19:42.25shmaltzki2k, if you don't know how to read they wont help you, if you do know how to read then you don't need them :P
19:42.28brodiemand I don't like FOP :)
19:43.22ki2kshmaltz: huh?
19:43.32ki2kshmaltz: i just wanna know what they're bundling
19:43.41shmaltzki2k, I got no clue
19:43.50ki2kthey call it "Asterisk Technical Manual"
19:43.57ki2kand a quickstart guide
19:44.02ki2khttp://www.digium.com/en/products/software/abe.php
19:44.13yojanlhttp://pastebin.ca/126778
19:44.53*** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41)
19:45.29GerbilWrkAnyone familiar with the Grandstream Handytone 286/386 or 488 products?
19:46.10shmaltzyojanl, you are having nat issues
19:46.23shmaltzyojanl, what does your sip.conf look like?
19:46.45*** join/#asterisk TK9 (n=Miranda@p54B28EA9.dip0.t-ipconnect.de)
19:47.53shmaltzyojanl, pb your sip.conf
19:48.15blitzrageif I perform a SetGroup for a particular extension, is it possible for me to perform a CheckGroup() for that Group I just set from a different extension?
19:48.34*** part/#asterisk TK9 (n=Miranda@p54B28EA9.dip0.t-ipconnect.de)
19:48.39blitzrageI don't think that's going to work because SetGroup just sets a channel variable... which is what CheckGroup is probably looking for...
19:48.52yojanlhttp://pastebin.ca/126784
19:49.02shmaltzblitzrage of course
19:49.28blitzragethe easy way for me to do what I need is to use the Local channel, but of course thats no available in ABE 'A'
19:49.58*** join/#asterisk kchrist (n=kwilson@bridalveil.istep.com)
19:50.22shmaltzyojanl, what type of router do you have there?
19:50.39blitzragethe issue is basically that I need a method to determine if a phone has a channel in use, and if so, not to include it in the group I'm dialing from an extension
19:50.43*** part/#asterisk kchrist (n=kwilson@bridalveil.istep.com)
19:50.54shmaltzyojanl, I would disable the port forwarding to the zs1, it shouldn not be needed
19:50.56Lyfeanyone have any ideas for why I can receive calls from a zap channel, but cannot place calls on that zap channel: http://pastebin.ca/126750 ?  I get  fast-busy, that apparently is generated by asterisk.
19:51.05shmaltzblitzrage, use chanisavail
19:51.27yojanlsmaltz: dont know for sure, its a corparte one, they have forwarded the ports i wanted to the static ip of the grandstream
19:51.30yojanland incoming works fine
19:51.41blitzrageshmaltz: does that not just check to see if a channel is available (i.e. registered) and not whether it is using a line or not?
19:51.44yojanlI guess theres a win nt server somewhere
19:51.48shmaltzyojanl, it appears that that might be the problem
19:52.15shmaltzblitzrage, nope I use it for ecactly what you want to use it, I use it in my page context for app_page
19:52.24yojanlwhat do you think it can be? i can call the corporate network guy but I have to know what he has to change
19:52.48yojanlif you want i can post a succesfull outgoing call via a sip soft client on my laptop in the same network as the non functioning grandstream
19:52.50shmaltzyojanl, the firefly also has ports forwarded?
19:53.17shmaltzdoes that softphone have the ports forwarded?
19:53.33yojanli tried on a random port and one of the forwarded ports, and in both situations firefly works for outgoing, incoming I havent tested
19:54.02yojanlsorry, the forwarded ports are on the grandstream so i tried it only on ports which havent been configured to point to my laptop
19:54.10shmaltzyojanl, change the ports of the gpx to one thats *not* forwarded
19:54.17blitzrageshmaltz: thanks -- unfortunately I'm building a DP for someone running ABE, and of course I don't have a copy of it, so I'm not exactly sure which features it actually has :)
19:54.34shmaltzblitzrage, doesn't Digium do that?
19:54.43yojanlshmalz: i already tried this, but it gives the same result. Just to be sure I can try again on a very random port, hold on
19:56.56*** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr)
19:56.58jhiverhi all
19:57.20jhiveri have a strange issue: this morning my PAP2-NA was working fine, sending and receiving calls
19:57.34jhivertonight, when i try to send calls, i see this on the asterisk CLI
19:57.37jhiverAug 11 23:57:28 NOTICE[90384]: chan_sip.c:10372 handle_request_invite: Failed to authenticate user <sip:jhiver@lcr.ykoz.net>;tag=b1680c61c06a4e9fo0
19:57.52yojanlschmalts: gives the same result
19:58.15jhiverany ideas what's going on?
19:58.38yojanlits so strange that only the grandstream has issues and not the firefly
19:58.51yojanlare the special settings on a win nt server that have to be set?
19:59.30shmaltzyojanl, I'm sure, but I can't tell you unless you tell me what type of router/firewall you have
19:59.55yojanlim trying to find out, but I can only telnet and it gives no info :-(
20:00.05yojanland i dont have the telnet pswd
20:00.25shmaltzyojanl, looks like a cisco to me
20:00.42shmaltzyojanl, tell the netadmin (if it's cisco that is) to do a no fixup sip
20:00.42yojanltheres no http connection possible
20:01.55yojanlshmaltz: thank you very very much, im not a beginner in asterisk but this thing has got me troubled. I hate it when i cannot fix issues like this!
20:02.05*** part/#asterisk neo (n=neo@kessel.ordrejedis.net)
20:02.14shmaltznp anytime
20:02.56ki2kanyone have a clue what error 423 means?
20:03.05ki2kinterval too brief
20:03.09yojanlshmaltz: what exactly does no fixup sip do? you think it defaults to a different ip?
20:03.32yojanland, sorry if i want to know too much, how come firefly has no troubles with this?
20:03.42yojanlis it a specific grandstream issue?
20:03.46shmaltzyojanl, it's a cisco firewall feature that does some things with sip traffic
20:03.51shmaltznope
20:03.56nDuff...not that grandstreams don't have plenty of issues.
20:07.24*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
20:07.38*** join/#asterisk wunderkin (n=wunderki@216-19-202-9.getnet.net)
20:09.39*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
20:09.39*** mode/#asterisk [+o anthm] by ChanServ
20:09.50[TK]D-Fenderblitzrage: "show application chanisavail"
20:11.07shmaltzblitzrage, you want to use it with the s option
20:12.46*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
20:12.50Lyfeanyone have any ideas for why I can receive calls from a zap channel, but cannot place calls on that zap channel: http://pastebin.ca/126750 ?  I get  fast-busy, that apparently is generated by asterisk.
20:12.56[TK]D-Fenderbi bi :(
20:13.11Lyfe(yes, i know i'm repeating myself, but i'm not finding any luck on the web, either)
20:14.00[TK]D-FenderLyfe: tried plugging a normal phone on that jack and dialing the #?  Also please pastebin your zapata.conf
20:14.22*** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com)
20:14.28Lyfeyes, the line has been tested.
20:14.30dos000howdy
20:16.00shmaltzLyfe, plug in a phone while asterisk goes offhook, listen to make sure you get all the 11 digits, also, if you have a butset try using the moniter to make sure that asterisk is not dialing to soon after going off hook, also try using relaxtdmf
20:16.16shmaltzLyfe, is this a tdm400?
20:16.33Lyfenope, it's a cheapo 1-port card.
20:16.53shmaltzhmmmmm, so I don't realy know,
20:17.03Lyfeand, since i'm 80mi away, i can't really plug a phone in and test it :\
20:17.43*** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
20:17.45Lyfei had the guys at the datacenter test the line, but they might not take to well to finding a splitter, and listening while i try to dial.
20:17.46shmaltzLyfe, why gas is soooooooooooo expensive :P
20:18.08Lyfebecause visiting there once a month is cheaper than hauling t1 lines out here.
20:18.11*** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com)
20:18.27Lyfenot to mention the power & cooling requirements.
20:18.49shmaltzLyfe, where are you located?
20:19.02Lyfein the worst possible place to buy a t1 for internet connectivity :P
20:19.30shmaltzLyfe, IRAQ?
20:19.34*** join/#asterisk MindHack (n=mindhack@unaffiliated/mindhack)
20:19.48Lyfeok, guess i'm wrong about that.
20:19.59Lyferockford, IL, USA.
20:20.00shmaltzLOL
20:20.13shmaltzand the colo? Chicago?
20:20.24Lyfemilwaukee, actually.
20:20.29Lyfeprior contacts.
20:20.39shmaltzMilwaukee, is in IL?
20:20.45Lyfesorry, milwaukee, WI.
20:20.50shmaltzoic
20:21.10websaelyfe: you live in milwaukee?
20:21.17Lyfenope.
20:21.20*** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net)
20:21.36Lyfewe just have contacts with a company that we like who's datacenter is in milwaukee.
20:22.06*** join/#asterisk wunderkin (n=wunderki@216-19-202-6.getnet.net)
20:24.16Lyfeanyway.. getting back to it, i was asked to pastebin zapata.conf, and i swear that the only thing useful out of it (given that the [general] section is default, and the other sections are commented out) is this section that's in the url i pasted already.
20:26.51*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
20:26.59shmaltzgtg, cya guys
20:27.04shmaltzenjoy ur weekend
20:28.24Lyfeanyway..s ince it's requested: http://pastebin.ca/126808
20:30.17*** join/#asterisk zpertee (n=zach@cpe-65-25-51-117.neo.res.rr.com)
20:30.57zperteehey does anyone have a moment to answer a couple of questions about g729 codecs
20:30.58[TK]D-FenderLyfe: that is not zapata.conf
20:31.06Lyfeoh, my mistake... i read wrong.
20:31.23eKo1zpertee: what ?
20:31.55zperteeeKo1, i have 2 channels of g729 but how do i support that and regular gsm
20:32.20zperteeeKo1, i want it to default to g729
20:32.22Lyfed-fender: my apologies, i've got too many dumb things going on around me right now and it's affecting my brain.
20:32.46eKo1zpertee: disallow => all and allow => g279
20:33.08zperteeeKo1, ok thank you so much for your time.  I truly appreciate it.
20:33.13Lyfed-fender: http://pastebin.ca/126811
20:34.43*** part/#asterisk zpertee (n=zach@cpe-65-25-51-117.neo.res.rr.com)
20:34.59*** part/#asterisk mkrufky (n=mk@68.160.103.77)
20:36.11*** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110)
20:36.58*** join/#asterisk sxpert (n=sxpert@vau75-1-81-57-130-155.fbx.proxad.net)
20:37.02sxpertheya
20:37.55*** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca)
20:45.24sxpertanyone home ?
20:45.31*** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com)
20:45.32Lyfethere used to be.
20:45.48Lyfehell, if my brain was working, i might've even gotten a response to my initial inquiry.
20:46.47NDTwas just a split they will be back
20:47.24Lyfethere was?
20:56.01MindHackIm with an organization looking to have more effective conferance calls. Skype is proving to be quite limiting, and quality is really bad when communicating with people overseas.
20:56.18hmmhesaysi'm not home
20:56.24hmmhesaysasterisk contract?
20:56.38MindHackIs it possible, or sensible, to use asterisk to only communicate pc to pc? With perhaps only one node actually dialing out?
20:58.01hmmhesayswhy not?
20:58.09sxperthmmhesays: yeah.. some old friend of mine requests me to create and invoice a brand new system for him ;
20:58.10sxpert;D
20:59.00*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
20:59.29Lyfeso, all that pasting and stupidity over a comma.
21:00.01sxpertok, so what should I use for the base PC (I'll only be using G711) ? AMD64 or intel ?
21:00.22Lyfeor something.
21:01.11Lyfenevermind, guess not.
21:01.19hmmhesayssxpert: I see
21:01.40benjkI'd say WRAP or Soekris board
21:01.42hmmhesaysso we'll see you in here on ${install_date} + 1
21:01.50hads|homeheh
21:01.53hmmhesayshalf bald
21:01.55sxpertbenjk: nah, it needs to look serious ;D
21:01.56hmmhesaysno sleep
21:02.17hmmhesaysI see quintum fixed their asterisk prolem
21:02.20benjkan embedded system looks a bazillion times more serious than a foolish desktop toy box
21:02.22sxpertbenjk: it will need to handle like 4 E1 plus some analog ports for testing / monitoring
21:02.41sxpertbenjk: that'd be a rack machine ;D
21:03.02*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
21:03.07benjkfair enough, I'd still settle for four embedded boxes though if it doesn't need to transcode
21:03.26sxpertit will need to do multiple people conferencing with lots of conferences, something like 24 conferences of 5 people
21:03.39benjkok
21:04.29hmmhesaysyep sxpert you will be pulling your hair out
21:04.49sxperthmmhesays: heh
21:04.50sxpertlol
21:05.04hmmhesaysthis your first install?
21:05.11sxpertyeah
21:05.22hads|homeGood luck then.
21:05.26sxpertlol
21:05.27Lyfehe's right then, you will be.
21:05.48sxpertobviously, everything *will* go wrong ;D
21:05.57hads|home4 E1's is definitly throwing yourself in the deep end.
21:06.00Lyfe<PROTECTED>
21:06.26sxpertheh, we will start with 1 E1 anyhow... then add more E1s if the service is successful
21:07.41sxpertthe E1s will either come from Completel or Colt
21:09.33*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
21:14.07*** join/#asterisk c4t3l (n=c4t3l@69.15.174.114)
21:14.32*** join/#asterisk fiber0pti (n=John@207.114.199.107)
21:15.31fiber0ptiI'm using the manager API to create a small java application. I'm trying to take a channel and transfer it directly to voicemail but I don't know how I can do it. I can transfer to other extensions, how can I transfer directly to a specified mailbox?
21:16.45*** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il)
21:18.12eKo1First of all, you can't use the manager API to create anything.
21:18.23SplasPoodHey..  Can anyone suggest to me how I might be able to hook Cepstral up to Asterisk in such a way so that I could send text to Cepstral via some external source and have it speak that text via an existing call..    Ie, I want to make something where one party of the call can be typing and doing TTS
21:18.32eKo1Second of all, you just need to make your channel dial into a context that will enter voicemail.
21:19.55eKo1SplasPood: There is no built-in functionality that will allow that.
21:20.03eKo1You're going to have to hack it out.
21:20.23SplasPoodeKo1: I know, I'm more than willing to do that
21:20.40SplasPoodeKo1: just looking for suggestions
21:20.51SplasPoodeKo1: Ideally I'd like to avoid coding app_mycepstral.c
21:21.00eKo1app_cepstral already exists.
21:21.06*** join/#asterisk dasenjo (n=dasenjo@208.195.215.205)
21:21.09SplasPoodyes, but not the way I need it
21:21.16eKo1Then hack it.
21:21.24eKo1You can't avoid coding.
21:21.34SplasPoodcan an AGI be used that'd call Cepstral() app_cepstral when it got events from some external source?
21:21.39quid246I wish Cepstral had a sultry voice... then I could develop a "dirty talk phone service" based on AI
21:21.49SplasPoodeKo1: I can try and avoid coding C which is not my area
21:21.58eKo1quid246: hehehe
21:22.13eKo1SplasPood: Not your area? Then what is your area?
21:22.22SplasPoodJust not my language
21:22.29*** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
21:22.37eKo1I take it you're not a programmer.
21:22.45eKo1Your AGI idea seems feasable.
21:22.45SplasPoodNot by trade, no
21:23.00SplasPoodOk
21:23.31benjkSplasPood, you can still use app_cepstral
21:23.57benjkCepstral(${TEXT})
21:24.41benjkand do asterisk -rx database put tts text "foo bar baz"
21:24.52benjkactually with quotes
21:25.23benjkasterisk -rx "database put tts text \"foo bar baz\""
21:25.28fiber0ptieKo1: How would I specify the extension that I want to transfer too?
21:26.10benjkthen have a small macro that does DBget to read the text from astdb and stick it into ${TEXT}
21:26.42eKo1fiber0pti: Are you using the raw manager interface?
21:26.44[TK]D-Fenderfiber0pti : Make an extens for each mailbox...
21:27.06fiber0ptieKo1: I'm using the asterisk-java API
21:27.09benjkso, no c coding needed, only a handful of dialplan
21:27.20benjkno AGI needed either
21:27.26eKo1fiber0pti: sorry, I'm not familiar with that one. I work in raw mode only.
21:27.28[TK]D-Fenderfiber0pti : exten => #1000,1,VoiceMail(u1000@default) , etc....
21:27.52GerbilWrkHas anyone been sucessful sending CallerID with Voxee.com termination services?
21:27.58fiber0pti[TK]D-Fender: I understand what you're saying but I don't understand how to integrate that into the application I'm developing
21:28.25fiber0pti[TK]D-Fender: How can I dynmically pass the extension?
21:28.43*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
21:29.01*** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl)
21:30.35GerbilWrkAnyone at all? We just get Unknown for the callerid with the Voxee service, it's kinda annoying
21:31.16TrixVoxName or number?
21:31.44*** join/#asterisk supjigatr (n=syslod@152.53.17.26)
21:32.01mitchelocGerbilWrk: voxee should provide free support to their customers
21:32.02supjigatrHello.
21:32.22GerbilWrkmitcheloc, they do, through trouble tickets that will take a while, i'm trying to get something quick, just in case
21:32.34TrixVoxNo phone support?
21:32.40GerbilWrkTrixVox, all my cell phone shows is "Unknown"
21:33.14GerbilWrkno phone support apparently
21:33.46quid246Well, even if you set it... it might not show depending on who in the termination pool Voxee connects to.
21:33.52benjkthey probably have phne support but they don't accept calls from "Unknown" callers :)
21:34.04supjigatrAnyone have tips on troubleshooting Credit card and fax machines not working from FXS ZAP to PRI Zap bridge?
21:34.39TrixVoxjust use VoicePulse, CID works perfectly
21:34.46eKo1supjigatr: faxing is not reliable
21:34.51websaeVoicePulse ~ yuck
21:34.59websaemy opinion
21:35.08websaefaxing, just use a third party for that...
21:35.10TrixVoxyeah, because you're a competitor
21:35.31supjigatreKo1:  I think you are refering to spandsp.  I have a real fax machine and a few credit card machines on a analog channelbank and they are calling out on the PRI.
21:35.57GerbilWrkand how much does Voicepulse charge per minute?
21:36.04supjigatrI'm not using spandsp.   I can't get a zap to zap bridge to work with a modem.
21:36.04benjkwho cares
21:36.24benjkrates are low enough not to have to care about fractions of pennies
21:36.33eKo1Supaplex: The faxes go through * right?
21:36.50GerbilWrk1 cent a minute, versus 3 cents a minute on 8000 minutes a month, makes a difference
21:37.02websaewhat's the quality like?
21:37.02benjkif its a little more than elsewhere, it won't hurt, but it ensures that you get better service because all the suckers signed up with the ones that are 0.01 penny cheaper
21:37.12websaebenjk: exactly :)
21:37.14TrixVox0.005 - 0.019 depending on where you call, i average about 0.007 aka 7/10th of a penny
21:37.31supjigatrEko1: They are a native ZAP bridge.  NO IP involved.
21:38.21GerbilWrkTrixVox, and they work with Asterisk, and have unlimited channels at once?
21:38.25benjkand how much time are you now wasting trying to troubleshoot and you don't even get phone support for your 8000 mins
21:38.27benjkpffft
21:38.38adorahdo u know where can I get cheap fxs modules for digium tdm400 card?
21:39.07*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
21:39.12TrixVoxGerbilWrk: http://connect.voicepulse.com/ , everything is spelled out there
21:39.14GerbilWrkbenjk, I JUST signed up for their service to try it out, i don't know how quick their support is, or if they have a phone support line I can reach them at
21:39.28TrixVoxnot www.voicepulse.com, that's the vonage-like service for home users
21:39.44supjigatrEko1: I suspect my problem is timing but no real docs on taking timing from port 1 on a 104d and outputting on all other spans synced.
21:39.46benjknah, you want connect.voicepulse.com
21:40.36supjigatrIt seems all the 104d ports are not synced by default and run independant.
21:40.44eKo1supjigatr: interesting.
21:45.10*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
21:45.23TripleFFFFguys.. what that option to group cdr into one entry
21:45.27supjigatrWhat I need is port 1 to accept timing, ports 2-4 to sync to 1 and transmit to slave Channel Banks.
21:45.32javaradorah: maybe here http://shop.ifax.com/-c-32_26.html?osCsid=8d5dae981104dce7243f3f6b6663734b
21:45.43supjigatrHas anyone done this with sangoma or digium card?
21:47.26[TK]D-Fendersupjigatr : I do it on my A104d
21:48.38supjigatrHow do you define timing with the A104d?
21:48.55supjigatrI want to accept timing on Port 1 and sync that with 2,3,4.
21:49.30*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)
21:51.41GerbilWrkso TrixVox, do you work for Voicepulse or something?
21:51.48*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
21:51.53[TK]D-Fendersupjigatr :pastebin your configs
21:52.40TrixVoxnah, i've tried everyone and they all have their problems, the guys at vp have always picked up the phone when i called and at least made some effort to fix the problem, i'm done with all these bullshit email-only support providers
21:53.01GerbilWrkTrixVox, have you used Teliax before?
21:53.14supjigatrhttp://pastebin.ca/126892
21:53.35GerbilWrkTrixVox, and how many times have you seen voicepulse go down?
21:54.29TrixVoxit's gone down before, i don't think there's any provider that hasn't... but they answer the phones during an outage and there's a message up on the website pretty quickly with a status
21:55.01[TK]D-Fendersupjigatr : and now your wanpipe confis....
21:55.18TrixVoxthey used to have this crappy sip server early on that went down all the time (the iax ones didn't), but they upgraded everything a few months ago and it's been smooth sailing since
21:56.15GerbilWrkok, the plan is to you use two servers for outbound calls, so I doubt they will both be down at the same time
21:56.23*** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com)
21:56.30TripleFFFFguys.. my asterisk is not producing the cdr logs correctly...
21:56.36TripleFFFFjust the first thing it does..
21:56.52TrixVoxyeah, you register to two servers for incoming and send outgoing through two servers
21:56.54TripleFFFFso 1,dial,(SIP/blah)
21:56.54TripleFFFF2,dial(sip/1234123123@host)
21:56.59TripleFFFFwill only make cdr for first
21:57.06Mattwj2005hey guys
21:57.14JuggieTripleFFFF, because 2 will never run
21:57.38Juggieunless 1 fails
21:57.39GerbilWrkwell incoming is just an 800 numbers which will forward to our local phone switch in the cities we can accept it, and then come in through junction networks most likely for the rest of the cities
21:57.45TripleFFFFyes but 1 fails
21:58.09TripleFFFFok i meant.. Look.. i get hmm
21:58.35TrixVoxjunction's prices are crazy for a newbie service provider
21:58.41TripleFFFFi mean.. it should make a cdr log for every operation.. not just update the cdr record with latest operation
21:58.58TripleFFFFso.. if dial, dial ,dial and voicemail
21:59.02TripleFFFFhmm
21:59.05TripleFFFFok i get it
21:59.13GerbilWrkwell most of our 800 usage is in areas we can get the calls for free, we may spend $10 a month, even with their prices once we get everything squared away
22:01.40Mattwj2005is it possible to hook up a bluetooth to usb adapter to an asterisk box and call directly from the server?
22:02.03*** join/#asterisk NativeOnRye (n=terry_si@206.163.1.131)
22:02.04eKo1Mattwj2005: call where?
22:02.15Mattwj2005call anywhere
22:02.20eKo1and what does the adapter have to do with anything?
22:02.51*** join/#asterisk postel (n=jp@unaffiliated/postel)
22:03.07Mattwj2005well that is how I would enable my server to have bluetooth abilities
22:03.54eKo1I don't get it.
22:04.41Mattwj2005I was thinking maybe I could dial out with festival
22:06.36supjigatrPastebin appears to be locking up.
22:06.58eKo1Festival does text-2-speech. How will it help you dial out?
22:07.45Mattwj2005oops....doesn't asterisk have some speech2text abilities?
22:08.51TripleFFFFactyally
22:08.51TripleFFFF<PROTECTED>
22:12.02sxpertshould I have lots of memory for an asterisk box, or the minimum is enough ?
22:12.53sxpertthinking either 1G or 2G per processor here (considering how cheap ram is these days)
22:17.35eKo11G is good.
22:18.12Mattwj2005it is possible to forward calls to a bluetooth headset?
22:19.20[TK]D-FenderMattwj2005 : Sphinx.
22:19.22eKo1forward calls to a headset?
22:19.39eKo1that doesn't make sense
22:19.48Lyfea guy i work with used his bluetooth headset as an audio device for his softphone on his laptop.
22:20.05eKo1Lyfe: that makes sense
22:20.14Mattwj2005well for example lets say someone gives me a call.....I pickup on a softphone on my computer....can I transfer to an extension (bluetooth headset)
22:20.42eKo1the headset isn't a softphone Mattwj2005
22:20.58eKo1or a phone
22:21.11eKo1I've never seen a voip bluetooth phone
22:21.16eKo1that would be killer
22:21.47Mattwj2005I am just trying to understand chan_bluetooth and chan_btp
22:22.40*** join/#asterisk fgwaller (n=frank@65.105.5.40.ptr.us.xo.net)
22:24.21carrarMy cell phone bluetooth ear piece also works with my laptop as a audio in/out device
22:25.03Lyfecarrar: that's what the guy i work with was doing.. same unit he used for his cell phone.
22:25.28Mattwj2005yeah that is probably what I'll do....it wanted it to be cool (and geekier) than that
22:25.29Mattwj2005:)
22:25.30TripleFFFFme
22:25.31TripleFFFF;)
22:25.42TripleFFFFlaptop with spinx + voice recgno + asterisk
22:25.47TripleFFFF+ bluettoth
22:25.59Mattwj2005lol
22:25.59carrarand a hologram imager
22:26.00TripleFFFFso i click.. recognize call home..  and idal home
22:26.01*** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com)
22:26.12sxperteKo1: ok. should I expect running the stats postgres on a second processor in the same box, or on some other box via the network ?
22:26.35Mattwj2005and a holodeck and repacator
22:26.38carrarobi one kanobi, you're my only hope!
22:26.46Mattwj2005:P
22:26.56fgwallerTwo call at the same time having the same UniqueId, is that a Bug or a Feature... (colission in PRI channel)
22:27.16Mattwj2005a Stargate or two would be nice
22:27.22Mattwj2005lol
22:28.30TripleFFFFtheroicaly an atom is in 2-3 ~ places in same time till you look at it.. ( quantum crap says it anyhow) so .. jjust dont look and the object will be elsewhere
22:28.31TripleFFFF;)
22:28.52Mattwj2005lol
22:29.15*** join/#asterisk _DAW (n=bob@adsl-35-242-196.msy.bellsouth.net)
22:29.28Mattwj2005wow we are all a bunch of nerds :-P
22:29.29_DAWHI all..
22:30.04eKo1TripleFFFF: not atoms, but its constituent particales
22:30.18eKo1err, particles
22:30.58fgwallerI only understand Quark
22:32.04TripleFFFFmaybe its neutrinos also
22:32.58eKo1and that is due to the probabilistic nature of quantum mechanics
22:33.25eKo1the position of any particle, say an electron can only be described by a probability density function.
22:34.14TrixVoxIt's called the Voxee Uncertainty Principle -- there are 2-3 customer service reps, but if you try to find one, they've moved elsewhere
22:34.30benjkTripleFFFF, those particles are doing that on purpose, just to piss you off
22:35.04*** part/#asterisk dasenjo (n=dasenjo@208.195.215.205)
22:36.05sxpertfgwaller: do you want to play dabbo at quark's bar ?
22:36.52_DAWHas anyone here seen dtmf problems with asterisk over a very high latency circuit (satellite in this case).  I have a ds1 to a satellite carrier with an IVR on my * box.  When I call in and enter the extension I want to dial I get variations of the digits I dialed.  Usually it is truncated.  Suggestions?
22:37.16sxpert_DAW: press the dtmf key longer ?
22:38.08Lyfeheh.. very high latency indeed.
22:41.17*** join/#asterisk chreese (n=chatzill@bridalveil.istep.com)
22:45.05Lyfeanyone setup a non-pri t1 that can tell me how to get the callerid information from the line (my provider says they're providing as much callerid information as they're receiving about it)
22:45.35fgwallerNo, my wife would hire some Klingons to hun me...
22:45.52Lyfei would think that simply setting "usercallerid=yes" in zapata.conf would solve this, but i eem to be mistaken.
22:46.03*** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net)
22:46.22benjkpri debug is your friend
22:46.43benjkit'll show you the raw PRI messages coming in
22:46.51Lyfepsst.. "non-pri"
22:46.55benjkthat'll tell you if caller ID is sent or not
22:47.06fgwallersorry only PRI here
22:47.07benjkso what's the protocol then?
22:47.11Lyfee&m wink
22:47.22fgwaller(bushdrums)
22:47.29benjkthere gotta be some debug options for that
22:47.34fgwallerwith CAS?
22:47.53eKo1e&m wink <---- yuck
22:47.57sxpertanyone has experience with interfacing with completel or Colt E1s ?
22:48.11LyfeeKo1: yeah, i know, we're migrating to a pri in a couple weeks, but it's bugging me righ tnow :P
22:48.24justinu|laptopin my experience, E&M wink doesn't work on asterisk
22:48.39Lyfeworks fine.  no callerid though.
22:48.51justinu|laptopthen it doesn't work fine, does it
22:49.06benjkis there no debug option for e&m ?
22:49.08Lyfefine, yes.. not good though
22:49.37justinu|laptopi actually fixed a bug regarding E&M a while back, but it was for immediate start
22:49.44justinu|laptopi gave up on trying to get winkstart to work
22:49.51benjkbut does it have a debug option?
22:50.01justinu|laptopno, i added me own to debug the problem
22:50.13benjkpah, that really sucks then
22:50.25benjkshouldn't use anything that has no debug
22:50.37Lyfewell, i didnt' know any better wheni started.
22:51.07benjkdebugging and logging are the most important items for anything software related
22:51.35benjkif you can't see whats going on, you're blind
22:51.58benjkwould you start a job as a cab driver if you're blind?
22:52.16Lyfeactually, that might work, given cab drivers.
22:52.22benjkheh
22:52.23eKo1hehehe
22:56.02*** join/#asterisk |dennis| (n=dennis@200.32.215.82)
22:56.27twisted[asteria]actually e&m debugs just fine.
22:56.41twisted[asteria]and yes, e&m wink does work on asterisk
22:56.42twisted[asteria]we use it here
22:56.57Lyfehave information on how to tell what kinda stuff you're getting from it so i can see if i'm getting callerid info?
22:57.21twisted[asteria]you should get data in the form of *CALLERID*DNID*
22:57.21twisted[asteria]i do believe.
22:57.28Lyfeis it a variable?
22:57.30twisted[asteria]no
22:57.38twisted[asteria]turn on debug
22:57.42twisted[asteria]set debug 9
22:57.43twisted[asteria]set verbose 9
22:57.48twisted[asteria]call into one of the e&m wink channels
22:57.52twisted[asteria]you will see the digits pulsed to you
22:58.32sxpert*pulsed* ???
22:58.35sxpertwow...
22:58.43Lyfehrmm.. on the console you'd see it?  (i don't see anything on there)
22:58.44twisted[asteria]pulsed, toned, wtf ever
22:59.00twisted[asteria]you have to turn on debug in the logger
22:59.11*** part/#asterisk sp0n9e (n=sp0n9e@phpurge.com)
22:59.34twisted[asteria]but i'll give you a little hint
22:59.41twisted[asteria]our provider *SOMETIMES* provides caller id
23:00.04twisted[asteria]so you may or may not get it
23:00.13twisted[asteria]at first they told us they couldn't provide callerID on e&m wink
23:00.19twisted[asteria]but i called them out
23:00.29Lyfewell, my provider has said they are providing callerID on it.
23:00.44twisted[asteria]so turn on debug and see
23:01.22*** join/#asterisk VetteC6 (n=info@216.143.192.69)
23:01.48twisted[asteria]e&m wink is so old and slow ...  i despise it sometimes
23:02.10Lyfeso, set debug 9, set verbose 9, and hope that i see debug info?
23:02.14twisted[asteria]no
23:02.17twisted[asteria]turn it on in the logger
23:02.23twisted[asteria]you have to turn it on in the logger
23:02.39twisted[asteria]once you turn it on there (logger.conf)
23:02.45Lyfeahh
23:02.50twisted[asteria]reload the logger, then do set verbose 9 and set debug 9
23:02.55twisted[asteria]and viola, you have debug
23:04.25*** join/#asterisk ogbi (i=ogbi@71.194.90.124)
23:04.27*** join/#asterisk MatsK (n=mats@83.233.97.229)
23:05.14Lyfewell, how about that, just dnis digits.
23:05.53twisted[asteria];)
23:06.28twisted[asteria]now you can call your provider and complain :)
23:06.35TripleFFFFhehe.. hey
23:06.40TripleFFFFhow you get glboal vars ?
23:06.45TripleFFFFi tough they where globals by default
23:06.47twisted[asteria]${VARNAME}
23:06.51TripleFFFFnah
23:06.51Lyfewell, the question is, is the provider of the phone i tried to dial out with providing them with the callerid.
23:06.53TripleFFFFits not liking it
23:07.12TripleFFFFi do       SET       OldContext=${CONTEXT}
23:07.15twisted[asteria]Lyfe, good question.
23:07.21twisted[asteria]TripleFFFF, no, that won't work.
23:07.23TripleFFFFthen a LOCAL/1234@mynewcontext
23:07.29TripleFFFFand in there i try to get Old
23:07.35TripleFFFFoh
23:07.45TripleFFFFsetvar ?
23:07.50TripleFFFFchanges all the time
23:08.17twisted[asteria]TripleFFFF, show application setglobalvar
23:08.55*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
23:09.02Lyfecause this is what i should be expecting to have the callerid too, right?
23:09.03LyfeAug 11 18:08:53 VERBOSE[581] logger.c:     -- Starting simple switch on 'Zap/1-1'
23:09.06LyfeAug 11 18:08:54 DEBUG[581] chan_zap.c: DTMF digit: 1 on Zap/1-1
23:09.08LyfeAug 11 18:08:54 DEBUG[581] chan_zap.c: DTMF digit: 5 on Zap/1-1
23:09.11LyfeAug 11 18:08:54 DEBUG[581] chan_zap.c: DTMF digit: 4 on Zap/1-1
23:09.12*** mode/#asterisk [+b %lyfe!*@*] by twisted[asteria]
23:09.20twisted[asteria]sorry, you're going to wind up flooding the channel with that
23:09.22twisted[asteria]use pastebin.ca
23:09.33TripleFFFFAs of v1.2 SetVar is deprecated and we are back to Set.
23:09.34TripleFFFFlol
23:09.36TripleFFFFgot
23:09.44*** mode/#asterisk [-b lyfe!*@*] by twisted[asteria]
23:09.53TripleFFFFthey say to use set with ,g
23:10.01twisted[asteria]ah k
23:10.08*** join/#asterisk VetteC6 (n=info@216.143.192.69)
23:10.18twisted[asteria]see, you found your own answer ;)
23:13.24*** join/#asterisk file (n=file@neutrino.joshua-colp.com)
23:13.24*** mode/#asterisk [+o file] by ChanServ
23:13.26*** mode/#asterisk [-b %lyfe!*@*] by twisted[asteria]
23:13.41twisted[asteria]strange
23:13.45twisted[asteria]i could swear i undid that already
23:13.50Lyfefreenode's weird.
23:14.07twisted[asteria]yeh.
23:15.31chreesehi, i was wondering if anyone could help w/ a fax issue.  sending from  a fax machine on an FXS card out via an FXO card, calling to a phone number  on the same machine which is connected to an FXO. hope that makes sense.
23:15.35*** part/#asterisk VetteC6 (n=info@216.143.192.69)
23:15.49chreesehere is my ztmonitor stuff: http://pastebin.ca/126995
23:15.50*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
23:15.50*** mode/#asterisk [+o Qwell[]] by ChanServ
23:15.51Lyfesorry about the spam, wanted 3 lines, got 4, got caught. :P
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23:41.56bkw_ALONE
23:41.58bkw_wasabi peeps
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23:42.57NativeOnRye<chreese> are you using the g.711 ulaw codec?
23:43.45_GuhitI'm trying to setup receiving faxes with my X100P and all I can seem to get when using rxfax is a small 8 byte file.  There seems to several people with the same problem (via. google) but no solutions.
23:44.00bkw_rxfax isn't ment to even be used for faxing
23:44.12bkw_if you get 8 bytes then your libtiff is hozed
23:44.49_Guhitbkw_: What is rxfax supposed to be used for?
23:45.14bkw_it was a simple test application to try out spandsp
23:45.24bkw_it would be more prudent to use iaxmodem with hylafax
23:45.28bkw_so you gain software ECM
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23:45.36*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
23:45.59_Guhitbkw_: hmmm...ok, I'll try that then
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