IRC log for #asterisk on 20070914

00:03.14*** join/#asterisk wishes (n=wishes@60.234.20.178)
00:03.41wisheshas anyone ever had asterisk and mysql going ,and then had asterisk somehow just drop an entire table ?
00:03.59Nuggetno, but I've had mysql do that on its own.
00:04.06wishesmm really?
00:04.10Nuggetyes
00:04.20nDuffmysql ~= eeeevil
00:04.25Nuggetindeed it is.
00:04.47wishesmm ahh shit
00:04.48wishesnm
00:04.51wishesi just realized i did it
00:04.57wisheswhat a fucknut i am :/
00:05.01NuggetI guess you're evil, too!
00:05.15wishesi pasted the create new table, accidently pasted the 'delete if exists' on the line above
00:05.18wishesi guess i am :/
00:06.09Schreiber1337So, does anyone understand how MWI works with SIP..  I
00:06.45Schreiber1337I'm trying to find out if my phones are turning off the MWI light after 5 seconds or if the server is sending a clear MWI call...
00:08.55Schreiber1337crickets chirping...
00:09.19Nuggetsounds like a job for sip debug.
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00:31.22mistermochaokay, I've been having a hell of a time trying to figure this out....
00:31.39TJNIIBah.  I keep getting "ztdummy: Unable to register zaptel rtc driver" whenever I try to load ztdummy
00:31.40*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
00:31.47mistermochaI can connect to manager, but I can't figure out how to actively listen for events
00:32.27mistermochawhat would be the appropriate manager command to just see call events fly by?
00:32.37mistermocha(akin to watching the CLI)
00:33.35elixerat the end of your Action: Login block
00:33.36elixeradd:
00:33.38elixerEvents: on
00:34.33mistermochaoooh neato...
00:35.15mistermochaI'll give that a go... thanks
00:35.15elixeryup
00:36.31mistermochahmm... no go
00:37.19mistermochaI added Events: on and placed a call... but nothing came through
00:37.22elixerpaste your manager.conf to a pastebin
00:37.24elixer~pb
00:37.24jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:37.37elixermask out your passwords, please.
00:38.42rob0no way!! I want 'em.
00:39.29mistermochait's a trixbox pro... manager pwd's are mostly the same
00:39.42JTthis isn't a trixbox channel
00:40.02mistermochaI know... but it's still manager
00:40.16mistermochagive me just a coupla mins before giving me the boot
00:40.29elixermistermocha: manager.conf?
00:41.38mistermochaelixer: yah... as I look at it, I think I figured it out
00:41.53mistermochathe user I'm logging in as doesn't have any read perms
00:42.18elixerright
00:42.22elixerfigured as much
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00:45.21mistermochaoh snaps! that did it
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00:54.15tengulrehi,all
00:54.22tengulregood morning everyone~!
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01:21.56Fetchbleh, wtf does make install in zaptel need to overwrite zaptel.conf
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02:00.06WilliamKusb radio ALOT of issues in the svn version
02:00.13ZylkronI've a question, I have a phone connected to my router, and we register at asterisk server, our users uses g729 codecs and we want asterisk to passthru the calls to Openser , its that possible
02:00.51ZylkronIm not sure technically how would the diagram looks like really :P
02:01.02Zylkroncan anyone hekp
02:01.04Zylkronpls
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02:23.35TJNIIZylkron: You want asterisk to change protocols for you?
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02:24.32JTdidn't see any mention of changing protocols, TJNII
02:24.45TJNIIYea, I'm dumb
02:24.59TJNIII read it wrong.
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02:33.57ZylkronTJNII: nope
02:34.19ZylkronI just want asterisk to go into passthru mode, and let openser does the signal handling
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02:41.44[TK]D-FenderZylkron, What exactly is * DOING for you in this scenario?
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02:44.51riddlebox[TK]D-Fender, is there a good howto, on setting up a TDM card, with fx0?
02:45.49[TK]D-FenderZylkron, I'm trying to figure out why you are shoving * BETWEEN your phones & SER in the first place.
02:46.07JTZylkron: i thought that was crazy too
02:46.11JT[TK]D-Fender: even
02:46.17[TK]D-Fenderriddlebox, Plenty of guides out there for setting up zapata.
02:51.50Zylkronuhm
02:52.05Zylkronokay well
02:52.33Zylkronwe use * for registration
02:52.43Zylkronaccording to the diagram
02:52.57[TK]D-FenderZylkron, so far thats like inserting a 5th wheen in your car.  Whats to POINT?
02:53.05Zylkronthe point is
02:53.14Zylkronwe dont have g729 license :P
02:53.19[TK]D-FenderZylkron, why not just say "route all of our calls through CHINA".  that'd be just as productive.
02:53.23Zylkronso we use openser to handle the signal
02:53.37[TK]D-FenderZylkron, well if * jsut a PASSTHROUGH then it isn't adding ANYTHING <-
02:54.18Zylkronuhm
02:54.32Zylkronthis is definitely *not* helping me settle my problem :P
02:54.37[TK]D-FenderZylkron, * sitting between your phones and OpenSER in passthrough mode does NOTHING, so why put it there?
02:54.52JTsounds like your problem is an insane setup
02:55.06Zylkron* does voice mail, registrar and stuff, but it wont handle g729
02:55.09Zylkroninstead
02:55.09[TK]D-FenderZylkron, your description really sucks.  You don't have a "need" in it.
02:55.14Zylkronit will ask for g711
02:55.32JTstop being stingey and buy g.729 licenses?
02:55.41KrurstIt'll work though - * can passthrough 729 without a license
02:55.51ZylkronI dont have to, our routers have it :P
02:55.51[TK]D-FenderKrurst, back up while the getting's good...
02:56.08[TK]D-FenderKrurst, leave this to trained masochists :p
02:56.14iCEBrkr: |
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02:56.44Zylkronman =p
02:56.48Zylkronfucking useless =p
02:56.51Zylkronwtfever =p
02:56.55JT=p
02:56.57ZylkronI thought you're engineers :P
02:57.11JTengineers in sanity
02:57.15*** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir)
02:57.17[TK]D-FenderZylkron, You have not made an end to end description of WTF * is actually DOING<----------- for you.
02:57.18iCEBrkrWORTHLESS!!!
02:57.18JTto get your proposed setup working
02:57.32JTit will cost more to make that work, than to buy g.729 licenses
02:57.55Zylkron....
02:57.56[TK]D-FenderZylkron, "yay, register to * to to PASS THE CALL AS IS to OpenSER."  Translation : stis inbetween not converting ANYTHING or DOING anything.
02:58.24JTZylkron: they're like $10 each
02:58.38[TK]D-FenderZylkron, if all you're doing is doing G.729 from your phone to * and then right to OpenSER with G.729, then * has done NOTHING.
02:59.06JT[TK]D-Fender: he says it also does voicemail
02:59.07[TK]D-FenderZylkron, Try starting from the beginning and show where you want * to actually do something productive.
02:59.30[TK]D-FenderJT : Fine, so record your prompts in G.729, do your RECORDINGS in G.729 and be done with it.
02:59.34Qwellyou can only do voicemail if every user uses g729
02:59.44Zylkronuhm
02:59.57Qwellunless you transcode for some
03:00.27[TK]D-FenderQwell, So far all he's said was "G.729" so that seems consistent enough for me...
03:00.40Qwellfair assumption
03:00.46*** join/#asterisk lunaphyte__ (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net)
03:00.55Qwellso, why SER?
03:01.34[TK]D-FenderQwell, we were just wondering "wht ASTERISK" all this time?  Maybe the question should have been "WHY!?!?!??!!" in a "pleading to God" osrt of way :)
03:01.54Qwellwell, that would be as futile
03:02.24[TK]D-FenderQwell, yes, but we'd have just ignored it like God would and have made better use with our time :p
03:02.25Zylkron*sigh* =p
03:02.37ZylkronThanks for the Hekp
03:02.39Zylkronbitch =p
03:03.00Qwell<[TK]D-Fender> Zylkron, Try starting from the beginning and show where you want * to actually do something productive.
03:03.05Qwell^ a very fair request
03:03.14[TK]D-FenderZylkron, So if you want * as an application server without G.729 licenses make sure all recordings are in G.729 and will be made in G.729
03:03.19Zylkronso I dont get why they wont pay 10 bucks either, this one company has liek 200 phones waiting to be deployed :P
03:03.28JTliek fully
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03:03.33JTdon't call people bitches :P
03:03.37JTthey are trying to help :P
03:03.45[TK]D-FenderJT : gnarly!
03:03.45Qwell[TK]D-Fender: and, that wasn't quite what I meant by futile..
03:03.47JTyou aren't exlaining the scenario fully :P
03:03.51Qwelltake that as you will though :p
03:04.27Zylkronthats because I know its a stupid question which took me days to figure out =p, and I knew I'll be bitched at because of it :p
03:04.44QwellZylkron: so far, I haven't really even seen a question...
03:05.05Qwell"is <xyz> possible?"  yes, it's just software
03:05.08Qwellanything possible
03:05.21ZylkronTHATS MAYBE BECAUSE YOUR IQ IS 200% BETTER THAN ME O WIZARD OF QWELL
03:05.27Zylkronso I am DUmb
03:05.32Zylkronso what
03:05.38Zylkronzzzz
03:05.50*** mode/#asterisk [+b %Zylkron!*@*] by Qwell
03:05.52Qwellmoving on
03:06.06JTinstead of acting like an immature fool, you could give the requested info
03:06.07JToh well
03:06.10JerJerdamnit - did i miss a flame war?
03:06.24[TK]D-FenderJerJer, No, the embers still burn bright
03:06.28*** mode/#asterisk [+b Zylkron!*@*] by Qwell
03:06.31*** kick/#asterisk [Zylkron!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell)
03:06.35[TK]D-FenderJerJer, and trust me its the only "bright" thing going on ;)
03:06.35*** mode/#asterisk [-b %Zylkron!*@*] by Qwell
03:06.47JerJer!op me
03:06.55Qwell!op JerJer
03:06.58Qwelldidn't work :P
03:07.00JerJer:)
03:18.01Juggieholy shit
03:18.09Juggieis anyone else tired of the ast_frame_digital conversation? :P
03:18.15Juggieits been filling my inbox for like 2 weeks
03:18.20Juggieon asterisk-dev that is.
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03:34.56dudesbusy busy busy in here tonight holy batman
03:35.25[TK]D-Fenderdudes, SHH!!!! you'll wake the crickets!
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03:35.34*** mode/#asterisk [+o d3wayne] by ChanServ
03:35.38d3wayne~sipnat
03:35.38jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
03:35.53dudesIf you're looking for crickets I could help you out
03:35.58dudesyou don't mind if they are dead, eh?
03:37.37dudeswhat's NAT?
03:38.09dudesis that some bug that buzzes in your ear and pissed a homi off
03:39.05[TK]D-Fenderdudes, hukt on fonix werkt 4 u!
03:39.25dudesI don't speak french
03:42.04NuggetS.O.C.K.S!
03:42.30*** join/#asterisk dadicool (n=mrordaz@124.107.96.235)
03:42.47dudesI never got that hooked on phonics crap.  If the person was that stupid, how do they expect them to remember the alphabet?
03:43.56[TK]D-Fenderdudes, now you know your A-B-Q's!
03:44.18dudesyes I do
03:44.24dadicoolPlease help.... I have purchased G729 codec and am now having problems with the install network card.  I have already tried re-installation of my box and got to ask digium for a reset on my G729 registration.  Anybody here from digium? Please help. Thanks.
03:46.23dudescall them
03:48.11dadicoolI would have but that would cost me... I'm from the Philippines.
03:48.45dudesyou don't have termination to the US?
03:49.24*** join/#asterisk IgorG (n=FeedomPa@195.162.32.126)
03:49.40dudesif you message me, I'll give you access to my trunk so you can call
03:49.52Juggiedadicool, doesnt the default asterisk extensions.conf provide termination into the digium pbx misery?
03:49.53dadicoolhow do i do that? thanks.
03:50.58JuggieIAX2/misery.digium.com
03:51.05[TK]D-FenderJuggie, You keep GIVING away all out Seek-Rats!
03:51.20dadicoolpardon me for not being so techie... I am a newbie with VOIP. :-(
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03:53.15dudesyou know how to get that to work eh?
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03:54.16dudeswow, it's an outtolunc
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04:07.37dijungali just put a TE410P card in an asterisk install and i'm getting the following error "HDLC Abort (6) on Primary D-channel of span " 2/1 any ideas?
04:08.02dijungali've only configured port 1 & 2, since i'm only using those for now
04:10.20[TK]D-Fenderdijungal, pastebin "cat /proc/interrupts", "dmesg", your zaptel.conf and zapata.conf
04:10.22[TK]D-Fender~pb
04:10.22jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
04:10.53dijungal:O
04:10.58dijungalok hold
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04:16.39dijungal[TK]D-Fender: http://pastebin.com/d336f3ed0
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04:17.56[TK]D-Fenderdijungal, 2 T1 PRI's from the telco?
04:18.41dijungalfrom a cisco 3660
04:20.14[TK]D-Fenderdijungal, Ok, well set span 2 to 2,2,0 instead of 2,1,0.  otherwise you have 1 & 2 fighting for "primary" which might cause clocking issues
04:21.11dijungalhmmm
04:23.16dijungalok did that...., ran ztcfg -vvvv and restarted asterisk
04:24.14dijungal[TK]D-Fender:  so far no errors
04:24.27[TK]D-Fenderdijungal, Did you always get them immediately and constantly?
04:24.42dijungalyes on incoming calls
04:26.01dijungal[TK]D-Fender: so my third span should be 3,3,0 ?
04:26.11[TK]D-Fenderdijungal, yup
04:26.34dijungalok can u explain the timing column >
04:26.35dijungal?
04:26.52ManxPowerdijungal: think of timing as "sync source priority"
04:27.06dijungali thought it meant 0 - timing from telco and 1 - time internally
04:27.08ManxPowerwhere 0 means "never use this span as a source for sync"
04:27.37[TK]D-Fenderdijungal, 0 = PROVIDE timing
04:27.41dijungalok i just got the hdlc error again....
04:27.50ManxPoweror more correctly 0 means "never use this span as a source to get sync from"
04:28.01ManxPowerdijungal: contact digium support
04:28.27dijungalhmm..
04:28.44mmlj4hey ManxPower
04:28.48dijungalwhat if i put all spans timing 0
04:28.51dijungal?
04:28.52ManxPowerthat error means "got corrupted data from the pci bus"  It could be caused by many things
04:29.16ManxPowerdijungal: then asterisk will not have a timing source and audio will be terrible
04:29.35ManxPoweryour faxes will fail.
04:30.03[TK]D-Fendercalls may drop randomly and other craziness
04:30.13*** join/#asterisk zeeesh (i=zeeesh@202.166.161.45)
04:30.58dijungalk
04:32.27ManxPowerdijungal: basically you would get corrupted bits every once in a while.
04:36.56outtolunci'd say take that usb device off
04:37.33dijungalok thanks guys.. i've reverted to span=1,1,0,esf,b8zs
04:37.33dijungalspan=2,2,0,esf,b8zs
04:37.43dijungallet's see how it goes in production tomorrow
04:37.48dijungalwith 25 agents on :)
04:39.06dijungalI've also had an annoying issue where an agent will be on a call and might get another call on the phone, i've tried using call-limit=1, but that works for while, then seems to keep the agent in use, so no more calls go to that agent...
04:39.15dijungalthis is on asterisk 1.2.19
04:39.34dijungalis this a known problem? any work arounds, solutions?
04:39.39dudesdont send them a call then
04:39.45[TK]D-Fenderdijungal, limit the calls on the phone if you can.
04:41.25*** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca)
04:41.44dijungalhow so?
04:41.51[TK]D-Fenderdijungal, depends on the phone
04:42.02dijungalu mean on the softphone itself?
04:42.02[TK]D-FenderJunK-Y, I don't want to meet your mom!
04:42.32ManxPoweron polycoms it's something like "max.calls.per-appearance=1" in the phone1.cfg
04:42.49ManxPoweryou'll have to figure it out on the softphone
04:44.18dijungalk, it's the eyebeam phone
04:44.36ManxPower~softphones
04:44.46dudesthen disable call waiting in sip maybe?
04:45.01ManxPowerAll Softphones Suck (tm)(c) 2007 ManxPower
04:45.12[TK]D-Fender~softphone
04:45.13jbotsomething that should be drug out into the street and shot
04:45.13dudesyou're telling me
04:45.28dijungallol
04:45.29ManxPower[TK]D-Fender: you are such a jbot queen
04:45.32dudestheir update sucks
04:45.40[TK]D-Fender~[TK]D-Fender,
04:45.45[TK]D-Fender~[TK]D-Fender
04:45.46jbot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
04:45.51ManxPower~manxPower
04:45.51jbotyou are, like, your God. someone you should send lots of money to because he helps so many people.
04:45.51[TK]D-Fender^^^^^^
04:46.04JunK-Y[TK]D-Fender: thats what you said after what you did with her,last night, huh!
04:46.13ManxPowerI didn't know that said that, but I'm not complaining
04:46.35[TK]D-FenderManxPower, keep whoring away there!
04:46.37[TK]D-Fender:p
04:46.48dudesthen again eyebeam is done by Canadians
04:46.50ManxPower[TK]D-Fender: Everyone wants it for free!
04:47.36[TK]D-Fenderdudes, Apparently....
04:47.51dijungallol!!!
04:47.57dijungalu guys crack me up...
04:48.36dijungalok let me throw in another topic to gouge about "offshore call centers"... ok... and... GO!.
04:48.36[TK]D-FenderJunK-Y, Stacy's mom has got it going on!
04:49.19outtolunclook in the sky.. its a bird .. a plane.. no it's another phone being drop-kicked
04:49.36JTit's an iPhone
04:49.41ManxPowerdijungal: more and more companies are "onshoring" where they go into a rural area, train the locals and import any additional required people and open a call center.
04:50.04ManxPoweror a factory or a data center, etc
04:50.26[TK]D-Fenderdijungal, http://video.google.com/videoplay?docid=7362895170514642646
04:50.32*** join/#asterisk cmwt (n=bit_frog@adsl-76-200-102-66.dsl.pltn13.sbcglobal.net)
04:50.36dijungallol
04:50.40JTwhy would you put a datacentre in a rural area unless it was for disaster recovery?
04:50.50ManxPowerJT: The iPhone is cool, but not hundreds of dollars cool.
04:51.03JTit's not that cool :P
04:51.12ManxPowerJT: Why not?  cheap power, cheap realestate, put it near a university town and bring inthe bandwidth
04:51.22dudesI hate the iphone
04:51.28ManxPowercheap labor too
04:51.29JTdunno how cheap the power would be
04:51.35JTrural power is also shit power
04:51.37dijungali have a nokia e61i with SIP capabilities... kicks @@SS
04:51.46dudesjust another thing to pre-occupy peoples minds and crash into shit
04:51.47ManxPowerJT: put in your own power plant
04:51.55JTand there's no bandwidth in the middle of nowhere
04:52.00ManxPowerall these cool phones are GSM
04:52.33ManxPowerJT: I don't mean in the middle of nowhere, I mean like 20 or 40 miles outside of a larger city.
04:52.40ManxPowergoogle has been doing that
04:53.01JTwell i debate the ruralness then ;)
04:53.12JTbut still, near cities is the best for connectivity
04:53.37ManxPowerhttp://seattlepi.nwsource.com/business/280581_datacenter09.html
04:53.50dijungalahhh.. is "call-limit" flawed in *1.2 ?
04:53.56dudesyes it is
04:54.00dudeshahaha
04:54.23dudesI love beer
04:54.47dijungalriiighttt....
04:55.00dijungaland the price of rice in china just spiked!
04:55.13dudesthe US sent Rice to China
04:55.19ManxPowerhttp://www.thedalleschronicle.com/news/2007/08/news08-05-07-01.shtml
04:56.43*** join/#asterisk watchy (n=watchy@c-68-51-54-72.hsd1.ar.comcast.net)
04:56.56watchyanyone here use a solid state drive with linux?
04:56.56dudeshttp://www.usatoday.com/news/world/2006-10-19-rice-south-korea_x.htm
04:57.12dudesThat's so funny.  They sent Rice to China hahaha
04:57.16watchyim trying to install trixbox on a solidstate drive and getting mad problems
04:57.37ManxPowerwatchy: there was a discussion about that this week on the asterisk -users mailing list.
04:57.46watchywhat did they say?
04:58.48dijungaldudes... lol "US sends Rice to china"!!! that' as a good one!
04:59.15dudesI get a kick out of that everytime I think about it
04:59.17dudesthe Irony
05:00.59dijungalme tooo
05:01.00dijungallol
05:01.24dijungali'll email that one to the IT guys... "US sends Rice to China, due to shortage" :)
05:01.52dudesI like Condy though
05:02.04dudesshe a smart lady --- but her name takes the cake
05:02.53dudesanyone here from the Toronto area?
05:03.38dudeswhere you from dijungal?
05:03.53watchyim trying to install trixbox on a solidstate drive and getting mad problems
05:04.37*** join/#asterisk Joneser (n=Joneser@pool-71-170-201-50.dllstx.fios.verizon.net)
05:05.44[TK]D-Fenderwatchy, and you're asking HERE :p
05:05.57watchywell its a general question to all your smart losers :)
05:06.07watchyi think its linux issue personaly not trixbox
05:06.14watchygetting wierd hd erros
05:06.23watchyyou ever use a solid state drive with linux>
05:06.35[TK]D-Fenderwatchy, nope.
05:06.59[TK]D-Fenderwatchy, try another distro and see if its the kernel version
05:07.03watchyyou think they are a good idea for a * box?
05:07.39dudesI hate trixbox
05:07.48[TK]D-Fenderwatchy, SSD's should have the same long term writing wear out issues as flash>IDE dongles, so I'd consider thingfs carefully
05:08.00dijungaldudes: i'm in st. lucia right now
05:08.34watchytk: really?
05:08.34dudesa lot of babes imagine
05:08.37dijungalwatchy: there's a #trixbox
05:08.45*** join/#asterisk Rsaman2 (i=Alain@c1-169-15.tpr.isadsl.co.za)
05:08.46Rsaman2hello
05:09.08Rsaman2quick quiz : I am in asterisknow the shit gui version, stuck in some menu... how do i terminate it ?
05:09.26fujinctrl+alt+f1
05:09.28fujinlogin
05:09.33fujinkillall shitguiversion
05:10.20[TK]D-Fenderwatchy, its been discussed before due to CDR writes, VM, logging, etc.  you know flash has a limited write-life
05:12.10watchywell i dont think solid state disk are considered flash drives
05:12.19WilliamKhey [TK]D-Fender,  flash cards are essentially re-programable eeproms right?
05:12.38dudesdoes it matter in the end
05:12.41[TK]D-FenderWilliamK, I don't know the real tech nitty gritty...
05:12.50watchythis drives failure thing is 1,3 million hours
05:12.56watchythe mtbf is
05:12.59dudesno moving parts --- you loss the mystery
05:13.19WilliamKI've often wondered about taking 2 compact flash cards using an IDE adapter
05:13.22JTdudes: yes, it matters
05:13.23watchyi mean they put solid state in laptops for windows now
05:13.29dudespsst
05:13.33WilliamKsame way Global Technology Associates does for their firewalls
05:13.37watchyif solid stateonly supported so many writes you couldnt evfen use windows
05:13.43watchyit would kill the drive in a week
05:14.01JTwould depend on the solid state technology
05:14.10dudesI have drives from when I was 16 that are still rock solid
05:14.26JTwhat type of drives?
05:14.28WilliamKIt'd be interesting to load windows on an 8gb flash, and run a burn in util to see how long it'll run for
05:14.34JTand you could still be 16
05:14.34watchyjt: hmm hold
05:14.37dudesMaxtor drives
05:14.43JThard drives
05:14.44dudesI'm 23
05:14.59JTi'm sure everyone has drives that have lasted that long
05:15.03JTa lot fail too
05:15.08[TK]D-FenderWilliamK, not the same as firewalls.... the store their settings on flash, but load and run them from RAM.  Writting is very sparse.
05:15.09dudesmaybe
05:15.09watchyhttp://www.newegg.com/Product/Product.aspx?Item=N82E16820208316
05:15.12watchythat card jt
05:15.21watchyi mean that drivce
05:15.35dudesbut they have sat in a POS Athlon box for years and they still rock out
05:15.44watchyi cant get linux on it though
05:15.44JTsolid state is also less heat and less power
05:15.48WilliamKtrue, just thinking about the concept as a whole though
05:15.53JTi'm talking about for server/appliance use
05:15.54dudespowers cheap
05:15.58JTmore important than desktop
05:15.58JT...
05:16.05dudesand air conditioning fixes the head issues
05:16.10dudesor the window in the winter
05:16.28WilliamKpower is cheap depending on which part of the US you're in
05:16.30JTwtf are you going on about
05:16.49dudesthis that that
05:16.54WilliamKI don't know that it's cheap overseas - alot of places have power issues and asking for AC you might as well wish upon a star
05:17.01JTdudes: are you drunk?
05:17.18dudesno I'm quite sober
05:17.18WilliamKJT, I can only claim to have taken my meds
05:17.20WilliamK:)
05:17.46JTwatchy: says flash, wonder if it's write sensitive
05:18.02dudes<WilliamK> - where might one wish upon a star for AC?
05:18.02[TK]D-FenderJT : Looks like a duck....
05:18.17[TK]D-Fenderdudes, Walmart :)
05:18.19dudesI can imagine a few places -- but enlighten me
05:18.30dudesWal-Mart has a kick ass AC here
05:18.36dudesheater too
05:19.11dudesit's refreshing in the summer to walk into wal-mart
05:19.29dudesI really like when the lassies walk in heh
05:22.50watchyhey tk you tried a sangoma PCI-E card yet?
05:22.53[TK]D-Fenderok, time to hit the sack... later all
05:23.03[TK]D-Fenderwatchy, Nope, no need....
05:23.08watchywe bought a t1 PCI-E sangoma card to play with
05:23.26[TK]D-Fenderwatchy, have fun and don't let it run out of smoke...
05:23.44watchyyea, its for a customer were installing tommorow
05:24.16watchyif we can get this stupid linux on this solid state working
05:24.39JT1 day lead time? nice
05:24.53watchyjt: well we can always use a normal drive
05:25.05watchywe planned on being up all night doing it anyways
05:25.12watchyso we have plenty of time
05:25.20*** join/#asterisk Strom_M (n=strom@216.64.24.250)
05:25.35JTheh ok
05:29.32*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
05:30.04*** join/#asterisk CrazyTux[m] (n=CrazyTux@h460eb882.area3.spcsdns.net)
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05:34.16appelzaWow
05:34.21appelzaThe xorcom distro is AWESOME
05:34.22*** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net)
05:34.51dlynesmudirc
05:38.37remmoyawn
05:47.09Rsaman2arg
05:47.17dlynes?
05:47.17Rsaman2i am still stuck in that stupid menu
05:47.21Rsaman2for asterisknow
05:47.25Rsaman2when the linux boots up
05:47.29Rsaman2how do i terminate it ?
05:47.35dlyneswhat's the menu look like?
05:47.38Rsaman2and get a linux terminal
05:47.39dudesyou tell it to
05:47.45dudesthen you rub it and it does it
05:47.48dlynesalt-f2 maybe?
05:48.00*** part/#asterisk Shazoo (n=feng_me@60.216.14.2)
05:48.02Rsaman2still in
05:48.07Rsaman2i know its a hotkey
05:48.08dlynesare you in a console, or a gui?
05:48.09Rsaman2but i forget it ...
05:48.13Rsaman2well,,
05:48.23Rsaman2its the asterisknow console menus
05:48.27Rsaman2its the asterisknow console menu
05:48.35dlynesalt-f1, alt-f2, alt-f3, alt-f4, ...
05:48.52JTRsaman2: wrong channel, this isn't #asterisk-gui
05:49.04Rsaman2i know
05:49.07Rsaman2get it thanks
05:49.11dlynesor maybe you haven't gone far enough yet, to have the vt's loaded up
05:49.28dlynesRsaman2:  you could also try #asterisknow
05:49.42dlynesor maybe it's #asterisk-now
05:49.49dlyneslist *asterisk*
05:49.51dlyneserm
05:50.30dlynesyeah...#asterisknow
05:51.04dlyneserm #asteriskNOW....don't know if it makes any difference that the 'now' is in capslock or not
05:51.35JTno
05:51.39JTirc is case insensitive
05:51.40*** part/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net)
05:51.57JTbut people with good irc clients can tell how you capitalised it when you join
05:52.17JTuser and chan modes are case sensitive, however
05:53.52dlynesic
05:53.53dlynesbrb
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05:57.44*** join/#asterisk RsaMan (n=aa@196.210.155.3)
06:05.25awkhmm, is it vital for txgain to be upped in order to use hylafax/iaxmodem, etc
06:05.36awki'm getting a fax tone, but nothing as strong as it normally gets pushed out
06:06.00awkhence its not passing the fax through, only problem is I don't want it that people hear this intense sound on the other ed when we make a phone call
06:06.04awkso how do you balance it
06:06.15awksangona sugests you put the gain up to 8
06:06.18awkbut thats focking load
06:07.12Strom_Mawk: you call the telco's milliwatt test
06:07.32Strom_Mand you adjust your gain that way
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06:21.57mistermochaawk: are you using a dedicated fax line?
06:22.13mistermochayou can always just pump the gain on the one FXS channel
06:22.44appelzawhats the debian etch package for amportal called?
06:24.30*** join/#asterisk dominic1 (n=dob@213.221.82.242)
06:28.53watchyanyone ever put linux in a vlan>
06:29.38[hC]yeah
06:29.43[hC]its pretty easy
06:29.48[hC]easier depending on your distro
06:31.36watchycentos
06:31.38*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)
06:31.44watchywhat do i do to say put linux on vlan 1?
06:32.42[hC]vlan 1 is the default
06:32.49watchyok vlan 2
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06:32.54*** mode/#asterisk [+o Corydon76-home] by ChanServ
06:32.54watchyi just wanna know how to change it
06:33.12[hC]im not sure on centos, i use debian.. but the way it does it, its just a wrapper to vconfig
06:33.16[hC]search google for vlan settings centos
06:33.20[hC]im sure you'll find an easy howto
06:33.25watchythanks
06:33.49watchyhey hc you get ringing the base to work instead of call waiting beep?
06:33.59[hC]havent tried yet nope
06:34.18watchyyou ever use a solid state disk with asterisk
06:37.33*** join/#asterisk Strom_M (n=strom@216.64.24.250)
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06:41.08*** join/#asterisk RsaMan (n=aa@196.210.154.3)
06:41.10RsaManhello all
06:41.15RsaMani have a slight problem
06:41.19RsaManmy asterisk is buggered...
06:41.28RsaMan[root@localhost ~]# asterisk
06:41.28RsaMan[root@localhost ~]# asterisk -r
06:41.28RsaManUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
06:41.34RsaManwont load,,
06:41.48RsaManhow would i uninstall it ?
06:42.35RsaManrunning asterisk 1.2
06:42.39RsaMani dont just want to reinstall
06:42.46RsaMansurely that will break something ?
06:43.12Strom_Mhow about...
06:43.43RsaMan:)
06:43.43Strom_Mrun asterisk -cvvvvvvvg and see why it crashes on startup
06:43.58RsaManasterisk -cvg?
06:44.18Strom_M-cvvvvvvvg
06:44.52RsaManoh crap
06:45.18RsaManhttp://pastebin.com/d403ba263
06:45.22RsaManzap problems
06:45.37RsaMani dont get it
06:46.02Strom_Mare your drivers loaded?
06:46.04JTmake sure zaptel is running
06:46.16Strom_Mdid you run ztcfg -vv?
06:46.45RsaManoh
06:46.46RsaManran it now
06:46.47RsaManworking
06:46.54RsaMani ran zaptel make config
06:47.04RsaManbut didnt seems to load zaptel on start :(
06:47.19JTyou make some weird assumptions there
06:47.33RsaMan:(
06:47.35*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)
06:47.41RsaManwhat does ztcfg -vv do then ?
06:47.51JTconfigures zaptel
06:48.01JTreads /etc/zaptel.conf
06:48.12RsaMando i only need to run it once ?
06:48.31JTper boot, unless you unload zaptel stuff
06:49.23RsaManhow can i make it read each time i boot then ?
06:50.38watchycan i come vlan polycoms by provision?
06:50.54RsaManthats why i though make config in the zaptel source directory would do that
06:51.10JTRsaMan: do you know anything about the linux boot process?
06:51.16JTwatchy: err, say what?
06:51.40mistermochawatchy: I believe you might be able to...
06:51.44*** part/#asterisk dominic1 (n=dob@213.221.82.242)
06:51.56watchycan i setup polycoms on a vlan
06:52.03JTsure
06:52.04watchyusing ftp provisioning
06:52.49RsaManJT : does not seem like it :(
06:54.11JTeach distribution has a particular method of calling init scripts
06:54.55mistermochawatchy: the Polycom manual says you can from sip.cfg, but it doesn't say how
06:55.05*** join/#asterisk clandmeter (n=Carlo@81.175.82.2)
06:59.12watchyif i setup say 2 linux boxes on Vlan 2
06:59.21watchydo i have to hook them to a switch that supports vlans?
07:00.21[hC]it either has to support vlans, or be so dumb that it just passes whatever it gets along
07:00.34[hC]if it does support vlans, chances are you have to enable those ports to work on the new vlan
07:00.42[hC]otherwise it should just pass data if its a dumb switch
07:02.10mistermochathe vlan setting in the phone won't mean anything unless the networking equipment understands it
07:04.37*** join/#asterisk psk (n=psk@golia.caltanet.it)
07:07.49watchythanks'
07:08.24*** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no)
07:11.37Aursis this normal:
07:12.21Aurs[root@sipgw01 /usr/src/zaptel-1.4.5.1]# /etc/init.d/zaptel start
07:12.22Aurs[root@sipgw01 /usr/src/zaptel-1.4.5.1]#
07:12.41Aurs(centos5).
07:15.14litage|wwhy would asterisk write CDRs to /var/log/asterisk/cdr-custom/ but not /var/log/asterisk/cdr-csv/ ?
07:23.32WilliamKhey Aurs, shouldn't you be using /etc/rc.d/init.d/zaptel start
07:23.33WilliamK?
07:24.34WilliamKneat, looks like someone Alias'd it
07:25.23WilliamKoh by the way, if someone's paying any thoughts to the SVN...someone tacked some comments into the zaptel.init file that need to come out so it'll exec correctly
07:25.54litage|wn/m
07:33.33AursWilliamK: zaptel.conf was missing, so it just got to a exit 0 line
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07:49.09n3glvcan someone tell me, I see sip/2.0 on some things, I have an old zyxel wifi phone that will not connect, could it be running sip/1.0 (if there is such a thing)
07:49.17n3glvwould this cause 401 errors?
08:01.10penguinFunk~pb
08:01.11jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
08:01.47penguinFunkanyone know what would cause this: http://pastebin.com/m645dbea0 ??
08:01.56dan__tAnyone rockin' LTSP today by chance?
08:02.02penguinFunksomething looks flakey
08:02.05penguinFunk:(
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09:12.54yxaif my digium pri card has onboard ec, do I need to do anything else other than compiling and modprobing zaptel?
09:16.41JTwhat does the hardware ec have to do with zaptel?
09:16.57*** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl)
09:18.02NuggetYou still need echocancel=yes in zapata.conf to enable the hardware echo cancellation
09:18.08awkhm does asterisk record history by default for registered extensions, if so where?
09:18.14Nuggetall the other echo-related settings in zapata.conf are ignored in that case
09:18.17awkI need to get an ip for a phone that was registred and is now off
09:22.24ai-a[off]awk: turn it back on.
09:25.12Uatechi
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09:25.16Uateci reinstalled my asterisk box
09:25.36Uatecbut now the CLI is not showing the steps of my dialplan as a calls steps through it
09:25.52Uatecit used to say the line of the extensions.conf that it was on in purple
09:25.57Uatecso i could trace how my callplans were going
09:26.01Uatecbut it's gone
09:26.06Uatecdoes anybody know how i can get it back?
09:27.57ai-awhats your exact command line you use to run * Uatec ?
09:29.14*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
09:34.45Uatecumm
09:34.46Uateci don't konw
09:34.49Uatecit starts on boot
09:34.51Uatecalways has
09:34.52Uateclol
09:35.27Uatecit's there in chkconfig thoug
09:35.28Uatech
09:41.01Uatecok
09:41.09Uatecwell i've just restarted it with "asterisk -vvvgc"
09:41.18Uatecbut i can't exit the CLI without terminating asterisk
09:43.19Uatecahah
09:43.23Uatec"service asterisk start"
09:43.25Uatecw00t
09:48.02eniohUatec: no, you have to run asterisk without -c
09:48.12eniohand then connecting to the console with asterisk -r
09:48.29eniohso you will be able to leave with quit
09:52.28Uatecenioh, when i ran asterisk with the -c it just didn't give me the console
09:52.41Uatecbut i was still running it in that terminal
09:54.08eniohwith -c, you should have the console
09:54.22eniohtyping something doesn't do anything ?
09:54.55*** join/#asterisk duckz (n=duckz@81.180.83.75)
09:58.46*** join/#asterisk c1|freaky (i=alpha@team.code-1.de)
10:02.06c1|freakyhi all. can asterisk be used for only voIP talks - f.e.: I want to put asterisk on a server and let people talk with each other using some software ...
10:02.26c1|freakyprivate use
10:02.28c1|freakynothing else
10:03.06c1|freaky?
10:04.34Teln1100Ayea for sure... make sip accounts for them and assign extensions
10:05.00Teln1100Axlite is a good softphone but any other sip phone should work
10:05.12awkI asked earlier with no luck, when using hylafax is it vital to set your txgains up drmatically
10:05.15c1|freakyok, thank you :)
10:05.16awkdeegan: to 8db
10:05.23awkerrr eg!
10:07.14*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
10:07.51c1|freakyif i run an asterisk server, can ppl from other servers be reached?
10:08.02c1|freakyTeln1100A: thank you :)
10:08.22hwthey, i have an asterisk/meetme server that crashed hard the other day. i am using ztdummy on 2.6 as timing source. we have maybe 50-60 users on peak hours. could it be that ztdummy does not scale up to this number of users?
10:08.47hwtor is the server hardware unstable? the box just halted completely.
10:08.54awkwhat was your server load?
10:09.05awkwhre you using sip or iax peers
10:09.18hwtawk: 0.3 maybe, when we had ~20 users.
10:09.20hwtawk: SIP only.
10:09.33hwthow can i confirm that ztdummy actually works?
10:09.53hwti got reports about choppy sounds with ~20 users.
10:10.19awkcoppy sound
10:10.21hwtztdummy/zaptel is loaded without warnings.
10:10.27awkhmm, whats up with your link
10:10.31awkcould be a b/w issue
10:10.44hwtawk: but that shouldn't cause the box to die.
10:10.50awkno it shouldn't
10:10.54hwtwe have 100mbit
10:10.54awkbut could cause choppy sound
10:11.00hwtah, yeah.
10:11.17awkthe box dying would be hard without running a log
10:11.24awkyou can set asterisk to dump its core if there is a problem
10:11.28awkthen monitor that core log
10:11.41awkthe next time that happens
10:11.47hwtbut we had another box running there, with a zaptel card, running 1.2cvs whiched coped just fine
10:12.13hwti could not find anything in either /var/log/messages or in asterisk/messages
10:12.18awkhmm, i have had not problems using ztdummy, but that could just be me
10:12.25hwthow many users?
10:12.27hwthow much mem?
10:12.29hwtcpu?
10:12.29awknaaa, if it core dumps you wont find it in those logs
10:12.41awkp4 3ghz, 1 gig ram / 30users
10:12.41hwtbrb
10:12.48awkstarted getting choppy around there
10:12.56awkbut that our link i believe
10:13.12awkalso it takes strain on a box to do transcoding
10:13.16awkwhat codec where yo usuing
10:23.19*** join/#asterisk nohup- (i=hmmmph@203.81.206.134)
10:26.39*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
10:27.33c1|freakydo I neen an voip provider if i run asterisk?
10:28.20*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
10:30.53penguinFunkno
10:31.58penguinFunkonly if you don't plan to use an FXO card or digital card and still want to make calls to everywhere
10:32.15penguinFunki.e internet only based system
10:34.51c1|freakyi would like to run it for me and some friends on an dedicated internet server
10:35.07penguinFunkto be able to call anyone?
10:35.30penguinFunkare you going to install some hardware to inferface with a telco?
10:35.39penguinFunkinterface*
10:36.03penguinFunkif not then you need a voip provider
10:36.33c1|freakyno not to be able to call anyone - just each other
10:36.42nohup-if you're just wanting to run it for yourself and a couple of your friends on a dedicated server then no you dont need a voip provider
10:36.49penguinFunkah then you dont need a voip provider
10:37.09penguinFunkjust two sip clients both connecting to asterisk
10:37.16c1|freakyok very nice ^^ thank you for your help :)
10:39.54c1|freakyif i run my own server, can it communicate with other servers like f.e. skype servers?
10:40.03shido6muahaha
10:40.15shido6there are wrappers
10:40.16c1|freakyjust curious
10:40.17shido6u can use
10:40.40c1|freakyok, thx
10:40.45c1|freakyy r u laughing?
10:40.46shido6yes you can use it with skype  (difficulty level 4) 0 ......10
10:41.03shido60 = beginner 10 = expert
10:41.16shido6if u can read instructions its a 0
10:41.27nohup-yes you can use it with skype and fwd and other services
10:41.30c1|freakyi can ^^
10:41.35shido6but if you read instructions you wouldnt be in here :)
10:42.18c1|freakynah i've just heard about it half an hour ago
10:42.33shido6well, good morning to you.
10:42.37c1|freakyand i wanted some information ;D
10:43.09shido6theres all kinds of people here. Even really crazy ones
10:44.04c1|freakywell ive heard from it before, but ... some guy in #jabber asked for a possibility to let his users use software to talk to each other, than i remembered the asterisk plugin for openfire (jabber server) then i became curious and looked at the configuration options of the plugin which confused me, so i went to the asterisk site to find some information
10:44.08*** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu)
10:44.28c1|freakyas the documentation site with the book is down and so i dont have too much to read i first ask the essential questions for me to even consider running a asterisk server in here ;D
10:44.49McDouglasi need help sos: i ordered some dlink voip phones
10:44.53McDouglasand i cant transfer a call
10:45.01McDouglasis it possible at all with sip phones?
10:46.13ai-aMcDouglas: connected to an asterisk pbx - yes.
10:46.35McDouglasand if there is no transferbutton on the phone?
10:46.39McDouglas(cant find any)
10:46.51ai-atry #<ext>#
10:46.54McDouglascan i configure asterisk to use the * (asterisk) button to initiate a transfer?
10:47.35McDouglasai-a: nothing happens
10:48.00c1|freakyshido6: what do the really crazy people do with asterisk?
10:50.50Uatecdamn
10:51.46ectospasmc1|freaky:  crazy people use stuff like NFAS to share D-channels across multiple PRIs
10:52.04c1|freakyallright ;D i dont know what that means ;p
10:52.11c1|freakyNFAS, D-Channels, PRIs
10:52.19ectospasmin essence giving you extra channels on a T1/E1
10:52.33ectospasmNormally, with every PRI line you need to have one D-channel
10:53.17ectospasmNFAS, or Non-Facilities Associated Signalling, you share a D-channel or two across multiple T1/E1 lines.
10:54.22ai-aMcDouglas... watch...
10:54.23ectospasmThis means that what would normally be used as the D-Channel on some lines can be used as B-channels.  B-channel is a voice line, whereas the D-Channel is a signalling channel
10:54.52ectospasm(I'm taking the DCAP this afternoon)
10:54.55ai-aMcDouglas: what model are they ?
10:55.04McDouglasdph-300s
10:55.33McDouglasai-a: http://www.dlink.hu/?go=jN7uAYLx/oIJaWVUDLYZU93ygJVYLelXSNvhLPG3yV3oWIl5jqltbNlwaaRp7jgkFz2onGQTo48EBtfhzKHkK0gRse3Ya48=
10:55.48ai-aRussian ?
10:56.06McDouglaswell, yes thats the strange thing... couldnt find any manual in english
10:56.15McDouglasbtw, i'm in Hungary
10:56.22ai-aim English.
10:57.05ai-aMcDouglas: cant read it.. ask dlink how a sip transfer is done on this phone.
10:57.20McDouglasi did ask them
10:57.26McDouglaswaiting for their ansfer
10:57.30ai-aasterisk uses sip, which uses a standard protocol that supports transfer.  If the phone doesnt support that, then its not really an asterisk or sip issue.
10:57.37McDouglasinly problem is.. i already ordered 30 of these phones :\
10:57.46McDouglas*o
10:57.48c1|freakyectospasm: nice ^^
10:57.58*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
10:58.49shido6if u cannot return them
10:58.58shido6then u can use features.conf
10:59.07shido6and use like a *# or something
10:59.13ectospasmyeah, features.conf will allow you to set it up
10:59.16shido6or add a speed dial for "Transfer"
10:59.18ai-ayes, as shido6 said,, enable a feature.
10:59.27shido6so dont freak out..... yet
10:59.45JTstill
10:59.47JTd-link
10:59.47ectospasmI believe blindxfer is configured to be just # by default
10:59.48JTeww
10:59.52shido6who knows there might be a firmware update for your phone if it uses SIP
11:00.04ai-aMcDouglas: future reference... research and test before you spend your money.. I've been deciding about a new laptop for 3 years so far :)
11:00.19ai-aectospasm: he tried #ext, didnt work.
11:00.48ai-a#ext is for ATA phones, not sip right ?
11:01.03JTand never buy any d-link voip product, ever
11:01.20McDouglaswell, if i rpess the # key and the extension nubmer all that happens is that i hear the tones in the phone
11:01.22McDouglas*press
11:01.34ectospasmai-a:  some ATAs are SIP
11:01.34ai-aMcDouglas: feature not enabled ?
11:01.44McDouglashow do i enable it?
11:02.21c1|freakycan users have something like an answering machine on a asterisk server?
11:02.21ai-aso we saying his dlink is an ata box with a cordless phone ?
11:02.37ectospasmI would highly suggest you remap the blindxfer to be something other than #, because if someone is in an IVR and they hit # at the end of a digit string, the phone will try to xfer them
11:02.55ectospasmc1|freaky:  called Voicemail, sure
11:02.56ai-aMcDouglas: http://www.voip-info.org/wiki-Asterisk+config+features.conf
11:03.23c1|freakycool
11:03.24c1|freaky:DD
11:04.27McDouglasai-a: yes, this is a cordless dict phone with analog port, but it also has a voip port
11:04.42c1|freakycan they let asterisk speak their own text if they're not online f.e. "im currently not online please leave me a message" ... and can they put something like an own ringtone like modern mobile phone "networks" provide?
11:04.48McDouglasso if you press the first dial button it uses the analog line, but if you use the 2nd dial button you dial on voio
11:04.49ai-aMcDouglas: hence sip protocol isnt working.
11:05.05McDouglaswell, i could register the sip account
11:05.17McDouglasand call other sip hpnes
11:05.32ai-ayes, [asterisk] --- (sip)  --- [ata device] -- analogu -- [dict phone]
11:05.41ai-ayour dict phone isnt sip,, its the base unit.
11:06.19ai-ahowever our corless phones have a 'recall' button that support the sip transfer vai the ata boxes.
11:06.23ai-a*cordless.
11:06.26ectospasmc1|freaky:  yeah, there's unavailable and busy messages
11:07.00ectospasmYou can have distinctive rings, but I'm pretty sure they're governed by the phones in question, and asterisk doesn't have anything to do with ringtones
11:07.12c1|freakyok ;)
11:07.23c1|freakythank you :)
11:08.56McDouglasai-a: i put "blindxfer => #" in features.conf
11:08.59McDouglasbut nothing happens
11:09.10ai-areset asterisk ?
11:09.11McDouglashow can i check if it actually works?
11:09.16McDouglasi restarted it
11:09.24ai-apress # :)
11:09.36McDouglasi can hear the dtmf sound, nothing else
11:09.40ai-aMcDouglas: get a softphone, and check it works on the computer.
11:09.48ectospasmAre you bridged to anything you can transfer?
11:10.02ectospasmYou can't transfer if all you've got is a dial tone, there's nothing to transfer
11:10.15McDouglasi call phone B from phone A
11:10.22nohup-try this: just dial the extension number and then press # ?
11:10.25McDouglasand try transfering from phone B to phone C
11:10.36ectospasmIn the Dial string, do you have t or T as an option?
11:10.43McDouglasi have t
11:11.00ectospasmWho's trying to do the transfer?
11:11.07McDouglasphone b
11:11.18ectospasmah, then you probably need T as well
11:11.34ectospasmIIRC, t means A can transfer, T means B can transfer
11:11.51McDouglaslemme try it
11:15.16McDouglasai-a: i installed x-lite
11:15.30McDouglasif i call that extension and press X in xlite
11:15.34*** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3062ffe84f40f3ec)
11:15.41McDouglasasterisk says "transfer" and i can transfer
11:15.49ai-awhat about #
11:15.50McDouglasbut if i call a dlink extension
11:15.58kaldemart is for called party, T is for calling party.
11:15.59McDouglaserr, i meant #
11:16.02McDouglassry
11:16.13McDouglasso, if i call the dlink and press the # on it
11:16.16McDouglasnothing happens
11:16.20ai-aok, so transfer works.. just your phone doesnt allow it..
11:18.03ai-aMcDouglas: check the docs for the device... must be some way of doing it. Transfer is a high importance of a phone on a PBX. unless they just expected you to use the phone as a home phone connected to a sip service for calls.
11:18.18McDouglaslol, there was NO documents in the box
11:18.25McDouglashunted one from some russian ftp server
11:18.31McDouglasand it says nothing about call transfering :\
11:18.53JTsend them back
11:20.48ai-aMcDouglas: how much did you pay for each phone ?
11:21.11McDouglas$50
11:21.34*** join/#asterisk fujin (n=aj@unaffiliated/fujin)
11:21.55ai-aus$ ?
11:22.35McDouglasyes
11:22.45ai-awell, they are cheap. what do you expect ?
11:22.58McDouglasthey are not cheap
11:23.07McDouglasthis is 1/3 of the normal price
11:23.12JTthey are
11:23.17JTand they are still a waste of money
11:23.19McDouglasbecause we got some demo prices
11:23.22ai-a1/3 of normal price = cheaper than normal price.
11:23.23JT$85 buys you a polycom
11:23.27McDouglasbeing a reseller
11:24.00McDouglasai-a: i meant, if you wanted to buy this phone froma  retail you had to pay $150
11:24.13JTwhat a rip off
11:24.17JTsounds like a pile of junk
11:25.41*** join/#asterisk sniper[FOO] (i=Snip3r@217.27.214.111)
11:25.53sniper[FOO]hi there
11:26.12sniper[FOO]anyone having a couple of minutes to help me out?
11:26.21JT~question
11:26.22jbothmm... question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html
11:26.22ectospasmsniper[FOO]:  don't ask to ask, just ask
11:26.26JT~ask
11:26.27jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
11:26.41Wonkawhy not have transfer handled by the asterisk the phone is registered to?
11:26.50sniper[FOO]OK, got it :)
11:26.54*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582526.dsl.bell.ca)
11:27.09*** join/#asterisk zotz (n=zotz@24.244.163.157)
11:28.46sniper[FOO]my main concern is that my application has to detect a ringing tone in the media stream (SIP/g.711) and I ended up with no acceptable results using AMD and BackgroundDetect, both are giving a huge number of false positives
11:29.31*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
11:29.41c1|freakywhat are asterisk extensions? can i find a list somewhere?
11:29.44sniper[FOO]the point is to look for a regular ringtone in an already connected audio stream
11:30.00ai-ac1|freaky: you mean addons ?
11:30.03sniper[FOO]and do something if it succeeds
11:30.09c1|freakyai-a: yea, also ...
11:30.20ai-aextensions would refer to a phone extension :)
11:30.21c1|freakyi just want to find out more about the possibilities with asterisk
11:31.10sniper[FOO]I found an app called nvlinedetect
11:31.42sniper[FOO]but I gotta use the 1.4 branch and I saw several posts reporting it won't compile
11:31.47ai-ac1|freaky: http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons
11:31.56c1|freakythank you :)
11:32.30ai-ac1|freaky: or... http://www.voip-info.org/wiki/view/Asterisk+addons
11:32.36*** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg)
11:32.50*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
11:33.02ectospasm~thebook
11:33.02jbotit has been said that thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
11:33.31ectospasmyeah, asteriskdocs appears to be down
11:33.31ai-aectospasm: url failed :)
11:33.32c1|freakythanks
11:33.38JTsniper[FOO]: wow, detecting tones, sounds like you like to get frustrated
11:33.59sniper[FOO]JT: I'm already frustrated :)
11:34.14JTgive up?
11:34.17sniper[FOO]nope
11:34.30JTwhy are you detecting tones?
11:34.36ectospasmhttp://209.85.165.104/search?q=cache:X5x-an-2KkcJ:www.asteriskdocs.org/modules/tinycontent/index.php%3Fid%3D11+asterisk+the+future+of+telephony+mirrors&hl=en&ct=clnk&cd=1&gl=us
11:34.38ai-aisnt there a Ring cadence detection ?
11:34.49sniper[FOO]if I can't find a suitable solution in a couple of days, I'll write one
11:34.53ectospasmSome of those mirrors might still work.  Yay Google Cache!
11:34.55ai-aguess that cant detect tones ?
11:35.07ai-aectospasm ;)
11:35.10JTi have the book mirrored
11:35.23sniper[FOO]It's not that difficult to detect a known waveform in HQ PCM audio
11:35.44JThttp://210.14.110.96/~jon/asterisktfot.zip
11:35.52JTit is in asterisk
11:35.53sniper[FOO]but I hoped I won't have to get into * inside out
11:35.54ai-ai downloaded a copy of the Asterisk book and now its used as a door stopper for my virtual house.
11:36.16JTand it depends how clean you expect the waveform to be
11:36.33ai-asniper[FOO]: detecting tones,,, when, and why ?
11:36.56sniper[FOO]JT: the purpose is to OK the INVITE on a dumb gateway in the right time
11:37.29*** join/#asterisk klictel (n=klictel@atelka.info)
11:37.38sniper[FOO]eg. set the status to Ringing till ringtone comes
11:37.58JTwhat sort of piece of junk gateway is this
11:38.04JTdoesn't it have sip messages?
11:38.04sniper[FOO]and screen out the providers' message
11:38.11JToh right
11:38.14JTtoll evasion
11:38.16JTgsm gateway
11:38.27sniper[FOO]indeed
11:38.40sniper[FOO]toll evasion?
11:38.41JTnot a fan of gsm gateways
11:39.00JTit's at least evading terminating your calls properly :)
11:39.04sniper[FOO]this process is called toll evasion?
11:39.10sniper[FOO]sure :)
11:39.24JThigh density gsm gateways are a bad idea
11:39.35ai-atoll evasion doesnt sound legal.
11:39.46sniper[FOO]I'm pretty familiar with the issues :)
11:40.04JTsucky audio
11:40.09JTlack of cell capacity
11:40.20sniper[FOO]anyway, it's not my business, a friend runs the business and asked me to help
11:40.51sniper[FOO]audio is really decent
11:41.07JTnot compared to pri
11:41.13JTand transients can screw it up
11:41.15sniper[FOO]yep, right
11:41.23*** part/#asterisk nohup- (i=hmmmph@203.81.206.134)
11:41.31sniper[FOO]but it's somewhere near
11:42.35sniper[FOO]anyway, do you have a clue how to accomplish this?
11:43.10JTwrite c, or don't use asterisk :)
11:44.56sniper[FOO]OK, then I'm gonna tweak the nvlinedetect source to get it compile in the 1.4 branch
11:45.15sniper[FOO]...is it something people don't really want to share?
11:45.32JTno, it's just something people really don't do
11:46.13ai-asniper[FOO]: tried #phonefreaking
11:46.20JTwhat exactly does nvlinedetect detect?
11:47.19sniper[FOO]tones (ringing, congestion, busy/ignore, anything)
11:47.37sniper[FOO]nvfaxdetect does the same for fax tones
11:47.38JTyou can specify tones to detect?
11:47.56sniper[FOO]BT ringing tone
11:48.01sniper[FOO]sure I can
11:48.16*** join/#asterisk michael-i (n=michael-@Wb85d.w.pppool.de)
11:48.18*** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net)
11:48.47JTfreeswitch can detect tones you specify out of the box
11:48.59sniper[FOO]freeswitch?
11:49.06sniper[FOO]some * distribution?
11:49.07JTgoogle :)
11:49.09JTno
11:49.10sniper[FOO]IC
11:49.12JTnothing to do with *
11:49.15sniper[FOO]lame question
11:49.38*** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr)
11:51.44michael-iDoes anyone have any tips to prevent the first 100-200ms of my outgoing voicemail greeting from being cutoff? Calling in on a zaptel channel and reaching voicemail cuts a bit off.
11:56.47deegando a Wait(2) first?
11:57.01dan__thrm hrm hrm.... ALMOST have inbound calling working
11:59.06michael-iah yes, had a wait(1) in there but not in the right spot... (INSIDE the macro would be better)
11:59.15michael-ithanks for the jumpstart
12:04.00*** join/#asterisk cjk (n=loic@80.92.64.103)
12:04.55cjkhi, since a certain time i have problems to play wav49 files from my voicemail system under linux and windows. when i convert them using sox everything is ok. how do you play such files?
12:06.42*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
12:07.52dan__tAlright, I got incoming IAX2 calls to work, kindof
12:08.01dan__tI think my.. uh... dialplan is hosed
12:11.28*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
12:11.30drakomorning
12:11.50*** join/#asterisk klictel (n=klictel@atelka.info)
12:13.45dan__tsup, dood.
12:13.50*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:16.19*** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar)
12:16.58dan__tThis is so bad-ass.
12:17.45shido6must be on the right track, dan__t :)
12:17.46dan__tOk, so, it looks like the documentation over at Teliax suggests that I can use the phone number as the name or number of the extension
12:17.52dan__tGetting there man, I really am!
12:18.22dan__tI mean their examples for adding an extension included the actual phone number as the extension name
12:18.25*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
12:18.43dan__tSo I'd just keep using that "extension name" (really, the phone number) and build my dialplan out that way?
12:19.37McDouglasai-a: i got it working
12:19.54McDouglasthe dtmf signaling was at fault
12:20.13ai-asettings on the ata device ?
12:20.24McDouglasyes
12:20.30McDouglasit was set to inband
12:20.31ai-athats $50 * 30 saved in your pocket now :)
12:20.35McDouglashad to change it to rfc
12:20.41McDouglaslol, yes ;)
12:21.08[TK]D-Fenderdan__t: you will typically receive calls against an exten registered with your ITSP.  Once it arrives on that # you can then simply GOTO wherever else you want in your dialplan to actually begin processing the call.
12:21.14*** join/#asterisk coppice (n=chatzill@234.155.17.210.dyn.pacific.net.hk)
12:22.17*** join/#asterisk snk00sj (n=gnelisse@apollo.digitalbase.be)
12:22.19snk00sjhi all
12:22.24dan__toh ok.
12:22.33*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
12:22.37dan__tVery cool.
12:22.51snk00sji am using asterisk 1.4.11 for the first time, i created a SIP user, and now want to connect to phone to it
12:23.10snk00sjafter setting auth userid & pw, the asterisk log gives "no matching peer found"
12:23.28snk00sji am using the gui (web interface) to change settings
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12:30.49dan__tAnyone know of a .gsm format player for X, so I can mow through a few of these recordings?
12:31.35dan__theh, loading one just segfaulted audacious
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12:32.56*** mode/#asterisk [+o Corydon76-home] by ChanServ
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12:33.58jhiver_hi all, do you know how I could convert .wav files into .g729 to avoid transcoding when doing music on hold?
12:35.06jhiver_http://www.asteriskguru.com/audio_conversion.php thank you voip-info.org =)
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12:36.20[TK]D-Fendersnk00sj: This is not the GUI support channel.
12:36.47snk00sj[TK]D-Fender, i know, i just don't think it has anything todo with the gui
12:37.16[TK]D-Fendersnk00sj: Of course you don't, that would imply some sort of personal responsibility......
12:37.32[TK]D-Fendersnk00sj: but you see until you can show us something solid, its jsut you...
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12:37.54lirakismorning
12:37.54[TK]D-Fendersnk00sj: PASTEBIN is your friend...
12:37.56[TK]D-Fender~pb
12:37.56jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
12:38.45[TK]D-Fendersnk00sj: So go show us what you've got configured for that SIP device and lets see how * reacts with SIP debug enabled
12:43.42*** part/#asterisk Strom_M (n=strom@216.64.24.250)
12:43.57arekmif I drop E1 part of that config and leave Zhone part then asterisk starts and zhone works fine
12:45.29arekmhttp://pastebin.com/m7293149e - added zapata.conf here
12:45.31arekmany ideas?
12:45.42arekm(I mean zaptel.conf)
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12:53.55[TK]D-Fenderarekm: I don't think I've ever heard of E! & T1 mixed on the same card.  I recall there being a jumper you had to set to use E1.  To me that might devalidate the mix.  Maybe somebody can correct me on this...
12:54.15*** join/#asterisk anonymouz666 (n=anonymou@189.25.132.243)
12:54.23coppiceyou can mix E1 and T1 on many boards
12:55.00[TK]D-Fendercoppice: I should have specified my assumption of Digium there..
12:55.15coppiceyou can mix on digium boards
12:55.19arekmTormenta 2 (PCI) Quad T1 Card 0 Span 1
12:55.19arekmTormenta 2 (PCI) Quad T1 Card 0 Span 2
12:55.19arekmTormenta 2 (PCI) Quad E1 Card 0 Span 3
12:55.19arekmTormenta 2 (PCI) Quad E1 Card 0 Span 4
12:55.27arekmto 2xT1 and 2xE1 here
12:55.33coppiceexpect that one :-)
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12:55.40*** mode/#asterisk [+o Corydon76-dig] by ChanServ
12:55.42ManxPowerLearn to use pastebin.ca
12:55.54[TK]D-Fenderthats 4 lines... I wouldn't panic...
12:56.06arekmManxPower: learn to read (earlier ;)
12:56.07coppicethe tormenta 2 card has different chips fitted for T1 oe E1 operation
12:56.43file[TK]D-Fender: !?!!!!
12:56.53[TK]D-Fenderfile: !?!??!
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13:21.53henkoegema<PROTECTED>
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13:32.44kippihey
13:32.49[TK]D-Fenderho
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13:33.43kippireally simple question, I have this config http://www.pastebin.ca/697508 but when the line is busy it goes to voicemail, how can get it to go to 5 then 6 etc?
13:36.36[TK]D-Fenderkippi: the line isn't busy as far as # can see.  it got answered by the telco
13:37.03[TK]D-Fender*
13:37.20kippiwhat happends is, asterisk puts it to the voicemail because the handset is already busy
13:38.00ai-ayour not using your return 'r'
13:38.04[TK]D-Fenderoops.... ok, 5 should get called.  Show me a call where it isn't
13:38.16ai-aoh ignore me. :)_
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13:41.24kippihow do you mean called?
13:42.17ai-akippi: CLi log.
13:42.38ai-aprove 5 & 6 is not being called by showing us the log of the call.
13:42.59kippiok
13:43.08ai-a(side note, dont paste in here, use http://pastebin.ca )
13:43.24ManxPowerkippi: voicemail IS priority 5
13:43.38ManxPowersorry, I misread that.
13:43.46ManxPowerkippi: it should work as you expect.
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13:44.05ManxPowerkippi: what version of Asterisk are you using?
13:44.10kippihttp://www.pastebin.ca/697520
13:44.30kippi1.2.20
13:44.33ManxPowerkippi: the phone IS NOT BUSY!
13:44.49ManxPowerThe phone is forwarded to extension 1050
13:44.54ManxPowerturn off call forwarding on the phone
13:45.00kippiah ha
13:45.18ManxPoweralso remove the "r" from the Dial line.  It makes you look like an asterisk retard
13:45.49ai-alearn to read your logs.
13:46.04ManxPowerkippi: the phone MAY be configured to FORWARD calls to a voicemail extension you have configured for it when the line is busy.  Don't know.
13:46.08ManxPowerBut it is not a dialplan issue.
13:46.08*** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au)
13:46.52Pengguhi all. is there a 'camp' feature? eg you call someone, so you 'camp' on their phone.. then hang up. as soon as the person gets off the phone, your phone rings, you pick up, and it calls that phone
13:47.18Penggusorry i mised putting in there they the called party is initially busy on another call
13:47.22ManxPowerPenggu: you can write one in the dialplan.
13:48.10ManxPower"show applications like dial"
13:48.13*** part/#asterisk pointer (n=pointer@aj.catt.com)
13:48.15ManxPowernotice the retry version of Dial
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13:49.00ManxPowerSo if you Dial a device and it is busy (as determed by the value of DIALSTATUS or HANGUPCAUSE) then you can use the version of Dial that retries.
13:49.01Pengguta, i;ll look into it
13:49.14Penggui c
13:49.20Pengguso the called party can hang up in the meantime?
13:49.23ai-abut hes hanging up on the call, it needs bind transfering to a holding ext, then bringing back when the calling party is free.
13:49.35ManxPowerI dunno.  read the docs for the dial retry
13:50.03Pengguai-a: you mean call-parking style?
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13:50.39ai-ayep
13:50.41ai-apark the call.
13:51.03Pengguhmm
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13:55.26Penggui guess for the 'camping' that i described above, it'd probably be more desirable to use chanisavail() rather then dail()ing the extension, to avoid annoying the person too much
13:56.31Penggucould may be have an h extension to allow the hanging up to check for registred campers... and trigger a call back
13:56.42Pengguwould need somewhere to store campers
13:58.19ai-aPenggu: http://www.voip-info.org/wiki/view/Asterisk+call+parking
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14:02.58Zeeek[NEWS] in 90 minutes, there's the #asterisk-users-conference on the IRC channel of that name and info here: http://VoipUsersConference.org
14:04.29[TK]D-Fenderyangvnc: This isn't #Asterix
14:05.02elixeror #Asstricks
14:05.08elixerthats fun
14:05.11[TK]D-Fenderkippi: if the phone is forwarded like that you're screwed
14:05.33[TK]D-Fenderelixer: no, thats supposed tobe "That's hawt"
14:05.51ai-ayangvnc: oh, that was a great comic.
14:05.52elixerheh
14:07.08ZeeekRobin Williams is a great comic
14:08.33yangvncIt was ment as joke
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14:18.11*** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net)
14:18.55elriahHi all.  How stable is the Asterisk Manager interface in 1.2.x, I have an app that's going to poll every 3 seconds but that app may be in the hands of say 50 different customers.  Am I shooting myself in the foot doing this?
14:21.30dijungalhttp://asteriskdocs.com
14:21.42elriahdijungal: Is that to me?
14:21.52dijungaloops sorry.....
14:22.03dijungalwhat's wrong with http://asteriskdocs.com???
14:23.05lirakisdijungal: .. not sure.. but its been down for a while ( a few weeks at least i think )
14:23.09NuggetJust a guess, but I don't think the URL is supposed to have three question marks at the end.
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14:23.27dijungalohooo
14:23.36QwellNugget: RFC-8457 allows it
14:24.32[TK]D-Fenderelriah: What are they polling for?
14:24.35Nuggetheh
14:25.33elriah[TK]D-Fender: Calls (Status) and Peers (SIPpeers) and possible Queue info... In a Flex 2 based operator console hitting a php back end...
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14:26.15elriahThe app works great, but before I release it I need to test under load and do some research.. this channel is the first step..
14:26.19[TK]D-Fenderelriah: What I'd suggest : Make a SINGLE polling app that will retrieve & do the initial parsing and have it get picked up from THAT server.
14:27.06elriah[TK]D-Fender: Ahh... Such as you suggested yesterday, a daemon... Have my app poll every few seconds and write out XML files.. yep, that's works and should be easy to do on my end..
14:27.23elriah[TK]D-Fender: Where do I send your consulting fee, lol
14:27.52ManxPowerelriah: you send it to eric@fnords.org via Paypal
14:28.43[TK]D-FenderHide and Zeeek!
14:28.59elriah[TK]D-Fender: lol!!!!
14:29.43*** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
14:29.46*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
14:30.38[TK]D-Fenderelriah: For my Polycom MicroBrowser Idle live queue status I sued to have 5 phones polling every 10s with 2 access each.  my CLI would get spammed even though it wasn't a huge load.  Instead I made a master poller taht created a STATIC XHTML page which would get reloaded instead of creating extra AMI calls.
14:31.02elriahManxPower: Money sent.
14:31.03[TK]D-Fenderelriah: While this adds a few seconds of net lag to the reporting "currentness" its more than acceptable
14:33.13[TK]D-Fenderelriah: if you need something a little more live and want to reduce the frequency a bit you could also try the AMI Proxy so that its a single persistant connection.
14:33.52ManxPowerelriah: Uh, I was joking.  Also $1 is not a consulting fee, it is an insult.
14:34.40elriahManxPower: lol, sorry couldn't resist.  I'll get you next time.
14:35.00shido6wow
14:35.34kippithanks guys
14:35.36kippithat worked
14:35.50*** join/#asterisk juliux (n=juliux@ubuntu/member/juliux)
14:35.55outtolunci want $1 where do i get one
14:36.14*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
14:36.41juliuxhi all, i setup a new client and now i get this error dsp.c:1426 ast_dsp_process: Inband DTMF is not supported on codec gsm. Use RFC2833 all the time in my asterisk log, what did i wrong?
14:37.09ManxPowerelriah: you are the first person to send me money is at least 6 months
14:37.28ManxPowerjuliux: what you did wrong is configure inband dtmf when using GSM
14:37.34ManxPowerdon't do that.  it won't work
14:38.06elriahSeriously, this channel is incredibly helpful and open source doesn't = free.  I wish I could give back more and will as our products and services are successful.
14:38.46*** join/#asterisk defswork (n=andy@83.105.96.154)
14:39.16[TK]D-Fenderelriah: Share code where you can as well then.
14:39.27*** join/#asterisk ChrisN (i=ccn@72.46.131.18)
14:39.38[TK]D-Fenderelriah: Public document "how-to's" for the tricky stuff you got to work.
14:39.44[TK]D-Fender~sipnat
14:39.45jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:39.54[TK]D-Fender^^^ thats one of my contributions
14:40.24[TK]D-Fenderelriah: Alons with countless JBOT trainings.
14:40.55ChrisNAny comments on how to fix this error from Asterisk 1.2.13? channel.c: Avoided initial deadlock for '0x8121d00', 10 retries!
14:42.11ManxPowerChrisN: upgrade helps with many deadlock issues.
14:42.25*** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136)
14:43.23ChrisNI wonder why Debian is still using 1.2.13 in their provided asterisk package.
14:43.25*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
14:43.30ManxPowerUsing Monitor, MixMonitor, ChanSpy, and ZapScan can make this errors happen much more often.
14:43.33juliuxManxPower, thxs
14:43.41ManxPowerChrisN: We don't care.  Compile from source.
14:43.58[TK]D-FenderChrisN: Because glaciers are stable too.... onlyl thats too fast for Debian's liking ;)
14:44.15Wonkadebian has 1.4.11 already...
14:44.21Zeeekhere we go...
14:45.17CaT[tm]chrisn: if you don't want to compile it yourself check with beckports.org or apt-get.org to see if anyone else has done a newer version and made it available already
14:45.42ChrisNCool. Thanks for the help.
14:45.44*** part/#asterisk ChrisN (i=ccn@72.46.131.18)
14:46.10ManxPowerthe problem with using packages is that most of the packagers of Asterisk, Zaptel, libPRI have no idea what the requirements of the various pieces of software and how they interact.
14:46.18jcanfieldWhat is the secret to make calls ring down to the next line on a Polycom 550?
14:46.21ManxPowerAnd more importantly, WE have no idea how it is packages.
14:46.27ManxPowerjcanfield: there are at least two.
14:47.10ManxPowerThe one *I* use is that each line appearance is registered to Asterisk as a seperate SIP account, then I check the return value of Dial (using DIALSTATUS) and decide where to route the call based on that.
14:47.20CaT[tm]manx: well it might work out for him or it might not. if not he can always try and compile it himself.
14:47.30ManxPowerYou also have to set the max calls per line appearance in the sip.cfg or phone1.cfg config file for the polycoms
14:49.23jcanfieldManxPower, hmmm, so the polycom can't manage the line traffic?  I was hoping it would work like the softphones;  Setup one line and it rings down automagically .
14:49.43mattbolldoes anyone know something about create_addr: No such host: free/0247503054 ?
14:49.51ManxPowerI didn't say you can't do that, I said this is how *I* do it, as it give me TOTAL control of how calls are routed on the phone.
14:50.00[TK]D-Fenderjcanfield: setup reg1 to use X calls @ 1 call per line-key
14:50.11ManxPowermattboll: Your Dial line is screwed up.
14:50.12[TK]D-Fenderjcanfield: And its will ring down naturally
14:50.38[TK]D-Fendermattboll: pastebin your failed call attempt
14:50.40[TK]D-Fender~pb
14:50.41jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
14:51.51jcanfield[TK]D-Fender, I'll check it out.  Forgive me for being the FNG, but is reg1 a phone setting?
14:51.57ManxPowermattboll: chances are you have Dial(SIP/free/0247503054) instead of Dial(SIP/0247503054@free)
14:51.57*** join/#asterisk linagee (n=linagee@about/linux/staff/linagee)
14:52.08dijungali know call-limit is buggy in *1.2, has it been fixed in 1.4?
14:52.18mattbollhttp://pastebin.com/d302d7014
14:52.25ManxPowerAnd I am assuming you have  a [free] section/peer/user/friend in sip.conf
14:52.26[TK]D-Fenderjcanfield: By what means have you configured your phone?
14:52.39ManxPowerdijungal: call limit has been buggy since day 1
14:52.53linageeManxPower: call limit?
14:52.53dijungalk
14:53.02jcanfield[TK]D-Fender, web ui only so far....jsut started my dive into all this.
14:53.04[TK]D-Fendermattboll: Executing Dial("SIP/bmatthieu-0817f328", "SIP/0247503054@free/0247503054|300|TtW") <- indeed this is not a valid format
14:53.09dijungalis the agent module more stable in *1.4 ?
14:53.22mattbollManxPower: ManxPower yes I have
14:53.41ManxPowerdijungal: last I heard the calllimit is not reset when an agent transfers a call, until the call is hungup.
14:53.51[TK]D-Fendermattboll: What GUI created that dialplan?
14:54.01mattboll[TK]D-Fender: everything is configured with destar, I'll tell them ^^
14:54.18[TK]D-Fendermattboll: Verify how you filled in the blanks when configuring it.
14:54.19dijungalk
14:54.25ManxPowermattboll: we can't help you with GUIs.  I already told you how your Dial line should be.
14:54.30dijungalManxPower: and the work around is?
14:54.32[TK]D-Fendermattboll: Dial("SIP/bmatthieu-0817f328", "SIP/free/0247503054|300|TtW")  <- this should be valid.
14:54.50[TK]D-FendermattSee if the way you filled things in was the source and not the GUI itself.
14:54.51mattbollok thanks a lot
14:54.54ManxPowerdijungal: no idea.  I hate queues and don't normally use them.  I simulate simple queues using dialplan logic
14:55.40ManxPowerI had massive problems with queues - most of them were USER issues, but I had a few Asterisk issues too.
14:55.47[TK]D-Fenderdijungal: By & large works fine.  Was that an open-ended question or do you have an actual issue?
14:56.05ManxPower[TK]D-Fender: he has the classic call-limit=1 problem when using queues.
14:56.14ManxPowerand agents transfer calls
14:56.29ManxPowerI have no idea if/when that was fixed.
14:57.06[TK]D-FenderManxPower: I've seen the one where app_queue thinks the agent is busy, I've never head of chan_sip's counter being off because of it though...
14:59.25ManxPowerdijungal: describe to [TK]D-Fender what your issue is with queue and calllimit
14:59.41Uatechello, is there any way of making a queue wait 30seconds or so after an agent becomes available before their phone rings again?
15:00.06errrdoes _. match any incoming number?
15:00.36*** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
15:00.38[TK]D-FenderUatec: thats what wrapuptime is for
15:00.48[TK]D-Fendererrr: Dangerously so.
15:01.00errr[TK]D-Fender: thanks
15:01.02[TK]D-Fendererrr: _X. is safer
15:01.16errrok
15:01.22[TK]D-Fendererrr: but restricts you to 2+ digit #'s
15:01.31[TK]D-Fendererrr: actually I think _X! would be best
15:02.02*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
15:02.06ZaVoidhey guys
15:02.07errr[TK]D-Fender: what is the _ for.. does that mean its going to be an incoming call?
15:02.24[TK]D-Fendererrr: No it means that what follows is a PATTERN MATCH.
15:02.34errrah
15:02.40ZaVoidcan someone point me to a spot in the wiki or somehwere else where i can understand how to break out numbers in a dialplan from a variable?
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15:02.47errr[TK]D-Fender: thanks
15:02.48[TK]D-Fendererrr: You really should have know that part..... its beyond dialplan 101
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15:03.09errr:(
15:03.19[TK]D-FenderZaVoid: lookup "asterisk variables" on the WIKI and it will show all sorts of ways
15:03.43[TK]D-FenderZaVoid: And after review "show function CUT" for some funkier stuff
15:03.44ZaVoidtyeah i was reading http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
15:03.44errr[TK]D-Fender: looks like I need to brush up on http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
15:03.53ZaVoidbut it wasn't helping
15:04.19[TK]D-Fendererrr: not knowing that "_" at the start INDICATES that what follows is a pattern is jsut a little disturbing :)
15:04.29ZaVoidi got variable called bignumber.. and "345" is in the variable... exten => s,n,SayDigits(${bignumber})  and i want to say 3 first.. then play a sound file.. then 45
15:04.36ZaVoidlet me search on the show function cut
15:04.43[TK]D-FenderZaVoid: Be specific with your example of what you'd like to accomplish then.
15:05.23[TK]D-FenderZaVoid: Saydigits(${bignumber:0:1})
15:05.29[TK]D-FenderZaVoid: Saydigits(something)
15:05.38[TK]D-FenderZaVoid: Saydigits(${bignumber:1})
15:05.48ZaVoidahhhh
15:05.49ZaVoidi see
15:05.52[TK]D-FenderZaVoid: And yes, that WAS all in the first link
15:06.00ZaVoidmy link or yours?
15:06.10[TK]D-FenderZaVoid: yours
15:06.16ZaVoiderr ok let me go look again
15:06.46[TK]D-FenderZaVoid: Substrings   ${foo:offset:length}
15:06.56ZaVoidsubstring i see
15:07.00ZaVoidthanks man
15:07.03[TK]D-FenderReturns a substring of the string foo, beginning at offset offset and returning the next length characters.
15:07.05[TK]D-Fender<PROTECTED>
15:07.06[TK]D-Fender<PROTECTED>
15:07.08ZaVoidi won't ask that ever again :)
15:07.22[TK]D-FenderZaVoid: s'ok
15:07.37[TK]D-FenderZaVoid: At least it wasn't something we had to beat into your head :)
15:07.59*** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net)
15:08.01ZaVoidlol
15:08.02ZaVoidthanks man
15:08.30Dr-Linuxis there any link from where i can see/listen MP3 ads?
15:08.32dlynes[TK]D-Fender: I got that parking feature so that it rings multiple phones when it coms off of park done much easier
15:08.38*** join/#asterisk melbert (n=IceChat7@66.179.79.70)
15:08.41dlynes[TK]D-Fender: app_valetparking
15:08.41Dr-Linuxactually i wanna download one
15:09.58[TK]D-Fenderdlynes : Funny, my way was MUCH easier ;)
15:10.12dlynes[TK]D-Fender: your way to not do it at all?
15:10.24dlynes[TK]D-Fender: that's not a workable solution, though
15:10.28[TK]D-Fenderdlynesthat compiling that addon... but if you need to actually PICK IT UP, versus leaving them in limbo a bit, yeah ValetParking rocks :)
15:11.19[TK]D-Fenderdlynes : I told you a way to transfer them out, let them sit & circulate, and then call back after a timeout.  Pure dialplan, 4-5 lines tops, and no app to compile.  What could be easier than that? :p
15:11.46[TK]D-FenderDr-Linux>is there any link from where i can see/listen MP3 ads? <----- HUH!?!??!?!?
15:11.52dlynes[TK]D-Fender: yeah, but you said your solution would have issues with more than one caller
15:12.07melbertI am having trouble with the CLID changing after doing a match.  After doing a: "exten => 18885555555/4444444444,n,Hangup" to block a specific incoming number the caller ID for everything after that shows up as the 888 number and not the caller ID
15:12.56[TK]D-Fenderdlynes : No, I said YOUR idea of trying to compensate for an idiot user not doing a BLIND TRANFER wouldn't work.  And I challenged you to prove otherwise :p
15:13.01*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
15:13.18[TK]D-Fenderdlynes : Valet Parking with an attended transfer sucks jsut the same :)
15:13.20puzzledhi
15:13.23dlynes[TK]D-Fender: ah...anyways..whether they do a blind transfer or not, anthony's method will work
15:13.39linagee[TK]D-Fender: valet parking? do they keep the key? lol
15:13.51ZaVoidok stupid question what if my variable is 3.45 instead of 3.45  i treat "." i could "." as a field space.. i'll test.. just thinking out lound here
15:14.04ZaVoidignore me
15:14.21[TK]D-Fendermelbert: pastebin the entire context and we'll show you where you went wrong
15:14.32Uatechey, i'm using GotoIf(X = 1?3)
15:14.39Uatecbut whatever X equals, it's always going to three
15:14.53Uateceg
15:14.53Uatec<PROTECTED>
15:14.54Uatec<PROTECTED>
15:14.55[TK]D-FenderZaVoid: You'd use CUT to grab the left & right halves so you could say "point" in between.
15:14.58linageewhatever X equals, it's always going to be 42
15:15.03*** join/#asterisk |omni| (n=rob@net82.allied-security.com)
15:15.05Uatecoh
15:15.08dlynes[TK]D-Fender: btw...that page works wonderfully...thanks again
15:15.09Uatec?
15:15.12ZaVoidok let me go lookup cut
15:15.13[TK]D-FenderUatec: because that is not a valid EXPRESSION at all.
15:15.25[TK]D-FenderUatec: Go lookup "asterisk expressions" on the WIKI
15:15.30Uatecok
15:15.35Uateclet me rewrite the example
15:15.36[TK]D-Fenderdlynes : glad to help.
15:15.53UatecGotoIf(${command} = 1?6)
15:16.06Uatecwhere command is set by the Read() command
15:16.09[TK]D-FenderUatec: STILL not an expression at all.  Go read the WIKI page on them
15:16.14jcanfieldOkay i got a very natural ring down working with the Polycom500 by setting NumLineKeys  to 3 and Calls per line to 1.   Can you see any problems with this setup?
15:16.18dlynes[TK]D-Fender: Just need to get chan_alsa up and running now so I can do external paging and radio for music on hold
15:16.27jcanfield*Polycom550
15:16.46[TK]D-Fenderjcanfield: Works doesn't it?
15:16.55[TK]D-Fenderjcanfield: so you left 1 key for speed-dial?
15:17.18[TK]D-Fenderdlynes : For external paging I'd far sooner get an ATA +Amp.
15:17.22jcanfield[TK]D-Fender, no i was hoping to make that a call park key.
15:17.38[TK]D-Fenderjcanfield: Never going to happen.  Find something else to do with it.
15:17.55melbertHere is the context that it is in:  http://pastebin.org/2423
15:18.11dlynes[TK]D-Fender: Well, my boss wants me to get radio working as input for moh as well
15:18.27dlynes[TK]D-Fender: so getting the soundcard working on these machines is still important
15:18.35jcanfield[TK]D-Fender, crap!  so can't to call parking on BLF keys?
15:18.35dlynes[TK]D-Fender: I cringe at the thought, myself
15:18.36[TK]D-Fendermelbert: exten => 188855555555/4444444444,n,Hangup <- this won't work because you don't have a "1" priority for it.
15:18.46*** join/#asterisk |omni| (n=rob@net82.allied-security.com)
15:18.55jcanfield*do
15:19.02dlynes[TK]D-Fender: I really really hate trying to get the latest and greatest hardware to work under linux...and soundcards are one of those things that never seem to catch up
15:19.04[TK]D-Fendermelbert: exten => 188855555555/4444444444,1,Hangup <- this will work.  It does not INHERIT anything from the  exten => 188855555555,1 version
15:19.13Uatecahhh
15:19.15Uatec$[ ]
15:19.16Uatecthat's odd
15:19.20Uateci've not used it else where but it works
15:19.23Uatecthat's quite annoying
15:19.34[TK]D-Fenderjcanfield: Nope
15:19.50[TK]D-FenderUatec: Its been like this in * since its creation.
15:19.52dlynesjcanfield: you should be able to call pickup on blf, but ont call park
15:20.24jcanfield[That's a bit of a step back.  ...so how do you park calls?
15:20.42dlynesjcanfield: just call transfer to 700, by default, unless you've changed it
15:21.20jcanfieldHmmm...kinda like the old nortels.  :P
15:21.23[TK]D-Fenderjcanfield: Go read up on "call parking" on the WIKI to see how
15:21.29dlynesjcanfield: then you can blf:  exten => 701,hint,park:701@parkedcalls (I think)...check the wiki for sure
15:21.43dlynesjcanfield: and then exten => 701,1,ParkedCall(701)
15:22.10dlynesjcanfield: and that WILL NOT work in asterisk 1.2
15:22.15[TK]D-FenderdlynesYou never have to do that, it gets auto included if you use the context that gets dynamically created.
15:22.28dlynes[TK]D-Fender: including the hint?
15:22.48[TK]D-Fenderdlynes : the hint you will have to make, but not the pickup mechanism...
15:22.54dlynesok
15:23.03[TK]D-Fenderdlynes : thats get generated automatically.
15:23.26jcanfield[TK]D-Fender, k will do.  I know a lot of my questions expose my ignorance, but i do appreciate the direction.
15:23.54dlynesjcanfield: actually, nortel is *74
15:24.22dlynesjcanfield: which is what i program my systems for...most of my customers are nortel converts
15:24.42dlynesjcanfield: so even voicemail is *980 :)
15:25.08*** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-212-135.dsl.irvnca.pacbell.net)
15:25.10jcanfielddlynes, it's still a braindead code, that's why i switched to the Panasonic DBS systems years ago because they handled call park very nicely and users don't have to think.
15:25.38JerJerusers can think?
15:25.47ZeeekJerJer !
15:25.50dlynesJerJer: not really
15:26.03UnixDoghell I know some admins who cant think
15:26.18dlynesjcanfield: once the user's get used to a certain system, it's a real pain to get them used to a new system
15:26.26dlynesusers, even
15:26.35JerJerhehehehe
15:27.02jcanfielddlynes, that is what a lot of PBX guys say. :P
15:27.04dlynesjcanfield: I get so many damned users saying...nortel did it this way...how come your phone system doesn't do it that way?  and blah blah blah
15:27.16anonymouz666JerJer: did you fix your nat problem with openser+ast?
15:27.28Zeeekhas anyone hooked up GrandCentral to * ?
15:27.59JerJeranonymouz666:  not specifically proven fixed, no
15:28.29jcanfielddlynes, true, but if you set a system up properly the transition can actually be a good thing.  iswitch many nortels systems over and never had a complaint.
15:28.33JerJerthat project has all kinds of various anonying network situations - like 4 year old routers and first gen SIP gear
15:29.26JerJeranonymouz666:  we think some of the devices don't support symmetric rtp and others have very very old nat routers, who are not stateful
15:30.28dlynesjcanfield: yes, but you're in Atlanta; I'm in Canada...Nortel is king, here :)
15:30.36*** join/#asterisk melbert (n=IceChat7@66.179.79.70)
15:30.58anonymouz666JerJer: yeap, some devices aren't smart enough to be symmetric
15:31.05anonymouz666they suck
15:31.12JerJeryes they do  :(
15:31.15jcanfielddlynes, I'm in Tulsa.  ...you have a point though.
15:31.21melbert[TK]D-Fender - I lost my connection but I did want to say thanks for helping me.  That worked.
15:31.29JerJerso we had to swap out those bad customers with 3102s
15:31.50JerJeronce they swapped, everything worked as expected
15:32.03dlynesjcanfield: oh...you've got an atlanta ip block...so figured you were in atlanta
15:32.26dlynesjcanfield: oh...nvm...read that wrong...cox is in atlanta...not you
15:32.50*** join/#asterisk |omni| (n=rob@net82.allied-security.com)
15:32.51jcanfielddlynes, ya it's cox, they hand out ip's like candy.
15:33.24dlynesjcanfield: but it'd be like trying to switch over someone from avaya in the US
15:33.26*** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
15:33.52dlynesjcanfield: well...maybe...i don't know if americans pride avaya like canadians pride nortel
15:34.08QwellI've never met a canadian who "prided" nortel
15:34.13agxhi, i'm getting some like one-way-audio with chan_misdn and HFCPCI. The ougoing audio is ok, the incoming audio is distorted and noised. any idea?
15:34.25*** part/#asterisk melbert (n=IceChat7@66.179.79.70)
15:34.29dlynesQwell: you were talking to the telecom guys though, not the end users :)
15:34.38EchinosQwell: not these days...(canadian here)
15:34.52EchinosAlthough their head office is in my home town
15:35.06jcanfielddlynes, done, it.  Thing about us americans...not very loyal to a product.
15:35.31ZaVoidhey fender.. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut this page right?
15:36.03dlynesanyways...gotta run
15:36.07dlynesnice chatting with you all
15:37.11*** join/#asterisk `paul (n=vina@124.107.13.212)
15:40.36*** join/#asterisk BadPacket (n=John@unaffiliated/badpacket)
15:41.35`paulhi. i have a prob hope someone has a solution for it.   here is the set up....  local_number <calls>---> another_local_number --callforward--> toll free number --> enters asterisk -->queue(agents) the problem is we have lots of drop calls the reason being the original caller puts down his phone before an agent answers. testing this setup we noticed that it takes around 6 rings before the agents phone rings. is the problem with asterisk or before the call ent
15:41.42*** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
15:41.48drakowhats the way to tell queue.conf when using mixmonitor to use option b so it merge the files.
15:44.10thewiizlehi
15:44.17thewiizlewhats the config file that stores the trunk settings
15:46.23*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
15:47.10*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)
15:48.18*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
15:49.00WilliamKCan someone with SVN access please remove these lines from the zaptel init file.... it's causing zaptel not to start because the lines aren't commented... <<<<<<< .mine , ======= , >>>>>>> .r3017
15:49.12QwellWilliamK: what branch?
15:49.18WilliamK1.4
15:49.22WilliamKlatest SVN
15:49.27Qwelllooking
15:49.32WilliamKthis just happened like yesterday or so
15:49.56*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
15:49.59WilliamKthanks :)
15:50.01QwellWilliamK: that's all you
15:50.26WilliamKfile I took them out of was /etc/rc.d/init.d/zaptel
15:50.43QwellYour local copy is modified, so when you updated, it added those
15:50.51[TK]D-FenderZaVoid: Just read the CLI page on it
15:51.09Qwelldo an svn diff, and you'll see it
15:51.09WilliamKif [ -z "${MODULES}" ]; then  (is the line of code between it)
15:53.08*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
15:53.42WilliamKnice.... it's me
15:53.55WilliamKI rm'd the directory and pulled a new fresh copy
15:54.01WilliamKhave no idea why SVN did that
15:54.04Qwellyou could've just svn revert'ed those files
15:54.17WilliamKI just know it goofed the init script
15:54.18WilliamK:)
15:54.25Qwellit did it because you had modified one of those lines that got changed (deleted), so it didn't know what to do
15:54.50*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
15:54.52WilliamKjust enough to pull the files
15:54.53WilliamK:)
15:55.40WilliamKI bet it probably happened the other night with the TE120P issue I had
15:56.22jfitzgibbonit's less SVN-specific than the 3-way merge algorithm.  You'll get that any time you change the same part of a file that someone else does and your revision tool tries to do a 3 way merge and fails
15:56.26thewiizleanyway of making asterisk display which config files it is using
15:56.38jfitzgibbonthankfully SVN tries very hard to prevent you from checking in files that are conflicted
15:56.49jfitzgibbonCVS was ... less diligent...
15:56.58shinao1hi, im kind of stuck with a TDM844B card... and i found out i need some sort of astribank/channelbank to go with it.. and im all out of money. Also, the asterisk server must sit in a server room. I wonder if there is any kind of channel bank that doesnt have a T1 connections?
15:57.50Qwellshinao1: what?  Why would you need a channel bank for an analog card?
15:58.22shinao1that uses only FXO/FXS or ethernet to connect to the PABX?
15:58.54Qwellshinao1: take a step back.  start over.  why do you think you need a channelbank?
16:00.44defsworkshinao1: http://www.voip-info.org/tiki-index.php?page=Asterisk+Channel+Bank
16:01.08Qwelldon't confuse him more...let's find out the reason first
16:01.15outtoluncif the asterisk server must be in the server room, the card must be in the asterisk server <G> so the 'lines' will need to be *stretched*
16:01.30defsworkawww
16:01.33Qwell...
16:01.37defsworkyou upset him with hard questions :)
16:01.40outtoluncmy bad <G>
16:01.52defsworkWhy is usually my first question
16:02.09Qwellclearly somebody gave him invalid information
16:02.32defsworkwell with a channel bank he wouldnt need the TDM844B at all would he ?
16:02.42defsworkbut he said he'd ran out of money
16:02.46Qwellno, he would need something to plug it into though, heh
16:02.56outtoluncwhich he is 'out of money'
16:08.00thewiizlesip_registrations.conf
16:09.34n3glvhi
16:10.16n3glvif I have an old sip device, could it have problems with currant sip version? (I see sip/2.0 on messages)
16:11.35*** part/#asterisk `paul (n=vina@124.107.13.212)
16:11.51[TK]D-FenderQwell: Backwards is thinking very his hmmmMMMM?!?!?
16:12.11n3glvyofa?
16:12.18n3glvyoda even
16:13.25[TK]D-Fendern3glv: indeed
16:13.35*** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg)
16:13.58n3glvanyway to overcome this?
16:14.17n3glvit's an old (v.1) zyxel p2000w wifi phone
16:14.47Nuggetthose are the shittiest phones ever made.
16:14.50Nuggetworse than grandstream
16:15.05[TK]D-Fendern3glv: Whats your problem exactly?
16:15.35n3glvhttp://www.pastebin.ca/696841
16:15.40n3glvit was free btw
16:15.40thewiizleShould a registration appear when i type 'sip show channels' regardless of whether it is registered or not?
16:16.08n3glvthewiizle, if it's sending it will say reg sent
16:16.32thewiizlehmm
16:16.45thewiizleok
16:16.49thewiizlecan i include files from sip.conf
16:17.05thewiizleusing 'include sip_additional.conf' for example
16:17.17n3glvyes
16:17.20n3glvafaik
16:17.29thewiizlehmm doesnt seem to be working then
16:18.31thewiizlei have my registrations in a seperate file
16:19.04n3glvmany systems do
16:19.14n3glvsuch as freepbx instsalls
16:19.17thewiizle:)
16:19.19thewiizlespot on guess
16:19.22n3glvinstalls even
16:19.39thewiizlethe additional files however are not being read by asterisk, hense by channels and peers are not being configured
16:19.51ZeeekRussell et al, ladies, gents, and the rest of you: #asterisk-users-conference in 10 minutes. Thank you. http://voipUsersConference.org/join.php
16:19.59n3glvthey take a leading #
16:20.08n3glvthat's not a comment
16:20.41*** join/#asterisk VoicePulse_ (n=contact@unaffiliated/voicepulse)
16:21.47thewiizle:)
16:21.55thewiizlei knew there was something missing
16:22.09n3glvso, freepbx system?
16:22.28thewiizleindeed
16:22.35thewiizletotally hand compuled so far
16:23.15n3glvok, mostly I do distro's that use freepbx (that's a dirty word here)
16:24.39thewiizleyeh i can imagine
16:24.45thewiizleseems the best choice
16:24.54thewiizlefreely configurable system with a decent gui
16:25.09thewiizleasterisknow used to be tops before they locked it down to a full release
16:25.14*** part/#asterisk juliux (n=juliux@ubuntu/member/juliux)
16:25.25n3glvso, a device running old sip ver may have issues with sip 2.0?
16:26.18[TK]D-Fendern3glv: pastebin again, its expired
16:26.37*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu)
16:28.03n3glvoops
16:29.40n3glvhttp://www.pastebin.ca/697692
16:30.04*** join/#asterisk snk00sj (n=gnelisse@apollo.digitalbase.be)
16:30.37snk00sjhi, i am trying to compile asterisk from source (all the previous installations removed)
16:31.03snk00sjbut on the last step make config i get the error : System startup links for /etc/init.d/asterisk already exist
16:31.21snk00sjif i remove this it still returns the same error (although it's gone)
16:34.50[TK]D-Fendern3glv : SIP/2.0 401 Unauthorized <--- how many times does it have to say you've got the wrong auth info before you go and FIX it? :)
16:36.28n3glvit has all correct info
16:36.29n3glvas far as I can tell
16:37.04n3glvit says lower about md5, the server is not running md5 afaik
16:37.16*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
16:37.34n3glvWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c73c930"
16:38.03[TK]D-Fendern3glv: SIp is valid, your auth criteria do not match
16:38.44*** join/#asterisk sgarcia (n=sgarcia@78.Red-81-34-55.dynamicIP.rima-tde.net)
16:38.55sgarciahi everyone
16:39.26thewiizlehmmm
16:39.29thewiizle404 all day everyday
16:40.35[TK]D-FenderNOW its Miller Time (tm) :)
16:40.50n3glvGuiness
16:40.53n3glvreal beer
16:41.02thewiizlelol guiness isnt beer
16:41.13thewiizledraught ale at its best :P
16:41.14n3glvmiller is like making love in a canoe
16:41.23n3glv&*^%ing near watter
16:41.29thewiizlerather enjoyable with multiple possible outcomes?
16:41.29n3glvwater even
16:41.30n3glvlol
16:41.52n3glvnot an ale, it's a stout
16:42.18[TK]D-Fenderload res_beernazi.so
16:42.51thewiizleload why_do_i_get_404.so
16:43.05n3glvgrep 'where's my beer' fridge
16:43.13thewiizleno no no
16:43.19thewiizlegrep "beer fridge"
16:43.51thewiizleok ive added another extension
16:44.03thewiizlesurely i should be able to dial an extension in the same context
16:44.22thewiizlebut no
16:44.25n3glvthis is being sent to the device
16:44.26n3glvFrom: <sip:9667@voipcoop.org;user=phone>;tag=15746980534AF4DDE6
16:44.40snk00sjwhen i start asterisk using : asterisk -cvv it starts fine, although when i use /etc/init.d/asterisk start, it gives me this error : /usr/sbin/safe_asterisk: 161: Syntax error: Bad fd number => although everything compiled fine, where should i start looking ?
16:44.43n3glvis user=phone the * user?
16:45.08[TK]D-Fender9667 <--
16:45.23[TK]D-Fenderthewiizle: Self-explanitory...
16:45.29[TK]D-Fenderthewiizle: NOT FOUND <---
16:45.34thewiizleyeh
16:45.36thewiizlei dont get why
16:45.37[TK]D-Fenderthewiizle: PASTEBIN is your friend
16:45.39[TK]D-Fender~pb
16:45.39jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
16:45.40[TK]D-Fender^^^^^^^^^^^^^^^^^^
16:45.52thewiizlei have nothing to paste :(
16:46.18*** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br)
16:46.33[TK]D-Fenderthewiizle: Gee, fat load of good that does everyone don't you think?
16:46.49thewiizleWhy, what do you want to see?
16:47.45*** part/#asterisk sgarcia (n=sgarcia@78.Red-81-34-55.dynamicIP.rima-tde.net)
16:48.01n3glvthe user is 9667, that is correct
16:48.10n3glvbut what is user=phone in that line?
16:48.29[TK]D-Fendern3glv: forget that and start looking at your SIP setup for that device
16:49.01[TK]D-Fenderthewiizle: Gee, I don't know, how about SIP DEBUG showing the failed call attempt?
16:50.07thewiizleLooking for *43 in from-internal (domain 195.26.235.104)
16:50.08n3glvthat's all in the pasteing
16:50.10n3glvpastbin
16:50.10thewiizlethats about it
16:50.28n3glvwww.pastebin.ca/697692
16:51.12[TK]D-Fendern3glv: I wasn't asking about YOUR problem...
16:51.40[TK]D-Fenderthewiizle: Well clearly that exten isn't in that context, what more do you want?
16:51.50*** join/#asterisk M-I-A (n=yada@CPE00304827782b-CM0014f85e8abe.cpe.net.cable.rogers.com)
16:51.58thewiizlei dont think its a case of not in the context
16:52.06thewiizleit seems to be more like the context doesnt exist
16:52.12[TK]D-Fendern3glv: I that isn't your SIP SETUP, thats the CLI SIP DEBUG that shows that things don't match
16:52.19n3glvsri
16:52.26[TK]D-Fenderthewiizle: And why would THAT be?
16:52.38thewiizleno idea
16:52.54thewiizlei cant find from-internal in extensions.conf or extensions_additional.conf
16:52.54[TK]D-Fenderthewiizle: You tell a sip device to use a context and it doesn't even exist.  Why are you even wondering why it can't FIND ANYTHING?
16:53.03n3glvgoing try and reg with softphone, but, it does the same on two servers
16:53.08thewiizleThe context is a default one
16:53.15[TK]D-Fenderthewiizle: FreePBX debugging I see...... you are in the WRONG PLACE....
16:53.42thewiizlenah not so much debugging just picking this all back up
16:54.07[TK]D-Fenderthewiizle: Well if the context doesn't even exist well I guess FreePBX is screwed
16:54.19thewiizleai
16:54.21thewiizleit would seem that way
16:54.28M-I-AAm I allowed to advertise my Digium TDM2401e that I have for sale?
16:54.44[TK]D-FenderM-I-A: Sure.  Ebay works, or the mailing list.
16:54.46n3glvthewiizle, if it's freepbx, use context=from-trunk or from-pstn
16:55.16M-I-ATK I have it on eBay but have not had a hit for close to 20 days :(
16:55.39[TK]D-FenderM-I-A: and when does the auction close?
16:55.54M-I-A3 days from now... already had one auction close with no bids
16:56.29[TK]D-FenderM-I-A: either your rating suck, your auction info sucks, your location sucks, etc...
16:56.37M-I-ATK: not even one person watching the auction :(
16:56.57[TK]D-FenderM-I-A: and you can often go up to the last few minutes before seeing a shit-storm of bigs for people trying to grab it.
16:57.17[TK]D-FenderM-I-A: smart bidders don't announce they're watching...
16:58.19M-I-ATK what is a good asking price for that card?
16:59.22[TK]D-FenderM-I-A: Use some common sense.
17:00.26n3glvI just got a tdm400p with one fxo and one fxs for $100
17:00.51*** join/#asterisk Cresl1n (i=matt@nat/digium/x-d29f37af4fb0e6f1)
17:00.51*** mode/#asterisk [+o Cresl1n] by ChanServ
17:01.53hmmhesayshey [TK]D-Fender you want to hear that go awful noise I was talking about?
17:02.13[TK]D-Fenderhmmhesays: No, s'ok I believe you :)
17:02.33hmmhesayshaha ok, i'm going to post it on the support forums
17:02.58*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)
17:03.21M-I-ATK: I've searched the web for pricing but those are brand new cards, I used mine for three months. So I was trying to price accordingly. Could you take a look at my auction and see if I'm totally whacked
17:04.20M-I-An3glv: that seems like a really good price
17:04.30UnixDogrussellb: when they going to fix the freebsd g729 regtool
17:04.54j0is an x100p card considered stable enough to run a business line off?
17:05.04[TK]D-FenderM-I-A: You'll have to come to your own conclusion.  Keep in mind though, its a big card so only larger installs will want it and those with cases big enough to support it.  Then consider those who WOULD need it probably wouldn't want it USED.
17:05.11[TK]D-Fenderj0: No.
17:05.36j0[TK]D-Fender: thanks... is it worth trying to use an analog line at all?
17:05.43j0I can't get a t1 here
17:06.17[TK]D-Fenderj0: Sure, and if you HAVE the X100P may as well TRY it, just don't be surprised if you're disappointed if CID doesn't work, they audio is iffy, or have echo problems.
17:06.31*** part/#asterisk Cresl1n (i=matt@nat/digium/x-d29f37af4fb0e6f1)
17:06.32*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:06.51M-I-Aj0: I got a TDM2401e for sale  :)
17:06.53j0i have a knockoff one.. and unless it's just going to plain work, it's not worth the trouble
17:07.23j0M-I-A: might be on the high end for me.. what's on it?
17:08.02M-I-Aj0: 4 fxo and echo can. module
17:08.40j0funny, i was just looking at that auction.. lol
17:08.49M-I-ALMAO!
17:09.05M-I-Athats really scary
17:09.25j0these asterisk appliances look nice
17:09.28[TK]D-FenderM-I-A: And why are YOU selling it?
17:09.51M-I-Awe went PRI
17:10.05UnixDogI saw one on ebay for 1600
17:10.07UnixDoglol
17:10.15*** part/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net)
17:10.39j0has anyone had experience with the remotely managed appliances from digium or trixbox?
17:11.08UnixDogits easy
17:11.17UnixDogjust have to have ssh and www
17:11.32j0its easy to have them do everything?
17:11.55UnixDogdepends on what you mean by everthing
17:12.06UnixDogits a web appliance for the most part
17:12.18j0i don't want to do anything more than do the initial setup... everything other than minor configuration they can handle..
17:12.19snk00sjanyone know a mirror of asteriskdocs.org ?
17:12.32M-I-Amake the morning coffee is what i've been trying to get my box to do for me
17:14.46thewiizleyo
17:14.56thewiizlewhats the variable for number dialled when used in a context
17:15.43thewiizleeg, Dial _,1,Dial(SIP/$number)
17:16.58hmmhesayswhat the hell, you can't attach anything to an asterisk forum?
17:18.22*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)
17:20.06errrwhen I setup my sangoma card it mad a zapata.conf and a zapata-auto.conf . If I want to change the context being used for incoming calls to a test context would I change it in zapata.conf or in zapata-auto.conf
17:20.18errrs,mad,made,
17:22.03[TK]D-Fenderthewiizle: ${EXTEN}  This is SUPER * 101... go lookup "asterisk variables" on the WIKI
17:22.25[TK]D-Fendererrr: Depends on the CONTENTS of those files.
17:23.15errr[TK]D-Fender: would zapata.conf need to have an include => in it for it to be using the -auto.conf file?
17:26.10*** join/#asterisk |omni| (n=rob@net82.allied-security.com)
17:26.41*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:26.49errr[TK]D-Fender: http://fluxbox.pastebin.ca/697749 (zapata.conf) http://fluxbox.pastebin.ca/697751 (zapata-auto.conf)
17:29.49hmmhesayshttp://myweb.cableone.net/mattman21/sample2.wav <-- if anyone wants to hear what I'm talking about
17:30.11*** join/#asterisk Cybertoy (n=cybertoy@swillux.swill.org)
17:30.34Kwakwaerrr, it will read zapata.conf
17:30.41errrKwakwa: only?
17:30.46Kwakwaif you check the CLI> u can usually see what's loaded
17:30.53Kwakwayeah, there's no include in that
17:31.04errrKwakwa: ok. thanks
17:31.07*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:32.26*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
17:32.29errrto get zapata.conf to reload you must retsart asterisk??
17:33.36kaldemardepends on the settings you changed.
17:33.55*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
17:33.58kaldemara context change can be dealt with a reload.
17:34.14errrah nice, that worked, thanks for the help :)
17:34.21*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:38.12*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:39.51[TK]D-Fendererrr: zapata-auto.conf is not even USED anywhere.  It is irrelevant
17:40.40bkruseerrr: everything but signalling changes (besides channel declarations and their options)
17:41.34*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:45.40*** join/#asterisk sniper[FOO] (i=Snip3r@217.27.214.111)
17:45.43*** join/#asterisk naxeji (n=nax@M1097P021.adsl.highway.telekom.at)
17:45.51sniper[FOO]hi there
17:48.24sniper[FOO]had to revert to the 1.0 branch of * for a short time, does anyone remember how can I make a macro called from the Dial() app bail out and terminate the Dial() as if I set Set(MACRO_RESULT=CONTINUE)?
17:49.12sniper[FOO]now I use SetVar(MACRO_RESULT=CONTINUE)
17:50.12sniper[FOO]but nothing happens, the SetVar app gets executed and the bridge occurs back in the Dial app
17:50.31sniper[FOO]and that's something I really don't want
17:51.24KwakwaWhat version of * u on sniper?
17:51.33*** part/#asterisk Cybertoy (n=cybertoy@swillux.swill.org)
17:51.58Kwakwaahh, I didn't read it all.. 1.0, ouch
17:51.59*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
17:51.59[TK]D-Fendersniper[FOO]: pastebin all related code.
17:52.09sniper[FOO]OK
17:52.40[TK]D-Fender~pb
17:52.41jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
17:53.04sniper[FOO]I know what a pastebin is, thanks, though :)
17:58.39*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
18:01.16naxejihi, can someone recommend me a good (stable) linux distribution for asterisk. Should I use a special distribution like AsteriskNOW, Astlinux or Trixbox, or is a normal debian enough?
18:02.03chemikknaxeji: debian stable
18:02.16hmmhesayswhatever you are comfortable with
18:02.25hmmhesaysstability has a lot to do with the administrator ;)
18:02.45naxejiok, thx for the recommendation =)
18:04.39*** join/#asterisk Godsey (n=jason@pdpc/supporter/sustaining/Godsey)
18:06.21Godseyis there a way to log the ip of registrations?
18:06.59Godseyabout 3 days ago, my cdr-csv/Master.csv file started showing my device dialing out
18:07.10GodseyI'm trying to figure out how someone is registering as my device
18:10.19sniper[FOO]Kwakwa, [TK]D-Fender: http://pastebin.ca/697791
18:11.21*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:12.45shido6heh
18:12.55shido6iax bandit or a sip bandit?
18:12.57*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
18:14.56*** join/#asterisk bruder (n=sergio@201.21.180.98)
18:15.43Godseysip bandit
18:16.03Godseybut I'm at a loss
18:16.19Godseyin asterisk logs, it looks just like my device is calling out
18:16.47Godseyit's a pap2-na.. I just restricted access to it and added a password (I'm stupid for not having one before)
18:16.56Godseyand changed the passwords for my device
18:18.27Godseyit looks like they dial and get my voicemail
18:18.33Godseythen my extension starts placing calls
18:23.31*** join/#asterisk drwelby (n=mpfister@mail.enplan.com)
18:24.33[TK]D-Fendersniper[FOO]: Not sure....
18:24.43[TK]D-Fendersniper[FOO]: Maybe someone else will...
18:34.55*** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org)
18:37.39*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
18:37.56denontotally OT I know, but I'm in kind of a bind, any of you guys feel like lending a hand on an openvpn config question?
18:39.46jwhsup?
18:41.11bkrusei can give it a shot
18:41.51drakook another weird problem.... when i get the calls from the ISDN (BRI) interfaces and i put it on the queue it works perfect but if first i put a background and then waitexten before the queue, when the call reach the queue and is answer it comes with no SOUND from the Caller
18:42.07drakobut if i get rid of the background and waitexten it works.
18:43.05jwhGodsey: you do have guest calling disabled right?
18:43.15Godseyjwh: I'm not sure now
18:43.30Godseysip.conf [general[ first thing context=banned
18:43.54Godseywhich is exten => i,1,Hangup and t,1,Hangup
18:44.22*** join/#asterisk Peaceful (n=peaceful@70.98.162.62)
18:44.26GodseyI have this under my pap2-na device: insecure=port,invite
18:44.39Godseyand host=dynamic
18:45.08PeacefulSo I upgraded from 1.2.13 to 1.2.24, and DISA() seems to not work anymore.  Just gives a fast-busy after trying to dial.  ???
18:45.32[TK]D-FenderPeaceful: ......
18:46.07Peaceful[TK]D-Fender, you lost me.
18:46.24[TK]D-FenderPeaceful: Didn't take much...
18:46.33Peacefulhehe
18:46.46[TK]D-FenderPeaceful: Ho about showing us something USEFUL?
18:46.59Peaceful[TK]D-Fender, like the error it doesn't give?
18:47.03jwhGodsey: hm
18:47.05Peacefulwhat, exactly?
18:47.06*** join/#asterisk VijayG (n=vijay@58.68.47.120)
18:47.17[TK]D-FenderPeaceful: like your dialplan and its CLI output for what it DOES execute.
18:48.05Godseyjwh: when I set my ip I tend to get 1 way audio on incoming calls
18:48.24Peaceful[TK]D-Fender, Here's the dialplan (personal info changed):  exten => 5606,1,DISA(333333333,internal,"MyCallerID" <80155555555>)
18:48.27jwhnat?
18:48.34Godseyyes, I have nat=always tho
18:48.43[TK]D-FenderPeaceful: pastebin EVERYTHING.
18:48.55Peaceful[TK]D-Fender, alrighty.  Hold on.
18:49.03Godseyand i changed the port to 15061 and have that forwarded to pap2-na
18:49.06[TK]D-FenderGodsey: Where is your PAP2 relative to *?
18:49.17Godseyand pap2-na is configured to use 15061 and 15061 as ext-port
18:49.25Godseyasterisk server is not behind nat, pap2-na is
18:49.38[TK]D-Fender~sipnat
18:49.39jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
18:49.40[TK]D-Fender^^^^^ go read
18:49.57Godseywell, if I leave host=dynamic it works :)
18:52.20Peaceful[TK]D-Fender, ok, that's annoying.  I try it again to reproduce the console output, but DISA works this time.  Grrrr.
18:59.19*** join/#asterisk shellprompt (n=shellpro@unaffiliated/shellprompt)
19:00.36shellprompthello all - I have fallen at the first hurdle - before I have even installed!  it seems that the documentation for asterisk "first timers" is not there - http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 can anyone point me in the right direction?
19:00.48Qwell~tfot
19:00.49jbotit has been said that tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details
19:00.52Qwellerm
19:00.54Qwell~book
19:00.54jbot[book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf
19:01.00[TK]D-Fendershellprompt: Go read the readme's that COME with your source tarball
19:01.02hmmhesaysso who wants to listen to my crazy sound problem
19:01.07Qwellshellprompt: there's a mirror
19:01.20[TK]D-FenderQwell: Where?
19:01.24*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
19:01.30Qwell[TK]D-Fender: at your site? :P
19:01.55[TK]D-FenderQwell: For the BOOK yeah, but thats for 1.2  I'd like to assume they're going to install 1.4 which it does NOT cover :)
19:02.14[TK]D-FenderQwell: And Asteriskdocs did have guides for 1.4 IIRC
19:02.23shellpromptthanks for the pointer.  it has still not downloaded so I have not had a chance to review the tarball.
19:02.24[TK]D-FenderQwell: In articles seperate from TFOT
19:02.52[TK]D-Fendershellprompt: Always check the big neon sign labeled "README", or "INSTALL", etc :p
19:03.57shellpromptI apologise - no ignorance intended.
19:07.46*** join/#asterisk TicoTuco (n=matheus@200.250.100.25)
19:08.21jcanfieldCan i have an internal dialtone and an external dialtone?
19:09.36jcanfieldafter _9NXXXXXX switch to external dt. I found I can make it dt go away with ingnorepat.
19:11.23rickrossanyone here experienced with SpectraLink cordless handsets?
19:11.43*** part/#asterisk naxeji (n=nax@M1097P021.adsl.highway.telekom.at)
19:11.49[TK]D-Fenderjcanfield: ?!
19:12.01rickrosswe have recently switched to using Polycom SOundPoint IP phones and are DYING for a reasonable way to get rid of the cords
19:12.21[TK]D-Fenderrickross: INSANITY
19:12.38rickrossTK - I don't get it
19:12.57rickrosswe're insane for using/not using something?
19:13.04[TK]D-Fenderrickross: Soundpoit phones are the best... why are you looking to get rid of them?
19:13.05jcanfield[TK]D-Fender: after 9 is pressed, DT  need to change tone....I guess going away will work.
19:13.14rickrossoh, I am not!
19:13.19rickrossI love the SOundPoint
19:13.29rickrossbut I need some wireless handsets, too
19:13.50rickrossI am stuck at my desk now (cannot handle it :)
19:13.58[TK]D-Fenderrickross: haven't tried those DECT ones.... not sure about base functionality.  My Aastra 57i CT DECT is BLEH
19:14.20rickrossI wondered about the Aastras :(
19:14.36[TK]D-Fenderjcanfield: if you want dialtone top stop after 9 then stop using ignorepat.  And that only applies to zaptel channels
19:14.47rickrossI figured if Polycom liked SpecraLink enough to buy them, then maybe they're pretty good
19:14.58[TK]D-Fenderrickross: They are tied to their base and I recommend AGAINST them.
19:15.18[TK]D-Fenderrickross: SpectraLink may be better... depnds on how their bases work.
19:15.38[TK]D-Fenderrickross: Just saying Aastra's = ass
19:15.45jcanfield[TK]D-Fender: Okay, that makes sense.
19:15.47rickrossthx - appreciate the candor
19:16.03*** join/#asterisk Bentley (n=rcourtna@S010600195bb1a3a2.cg.shawcable.net)
19:16.19[TK]D-Fenderrickross: There are only certain bits of bile I spare this channel, the rest flies freely :)
19:16.39rickrosswhere would you buy Polycom's? We bought some from Atacomm, but have had a long delivery problem
19:17.10rickrossI'd love to know a good vendor who ships quickly with a good price
19:18.15russellbrickross: i can send you a phone very quickly for just $30
19:18.25russellbi can't guarantee type, or functional state.
19:18.31rickross:)
19:18.43rickrossthx, russellb !
19:18.46Bentleyhi all, does anyone here have a gpx2000?  If so, does blind transfer with the TRNF button actually work?  I've got 8 new phones where it doesnt
19:18.50russellbanytime!
19:18.58[TK]D-Fenderrickross: www.telephonydepot.com
19:19.45lirakisBentley: yes blind transfer works fine
19:20.06Bentleylinagee, are you running Software Version:    Program-- 1.1.4.18    Bootloader-- 1.1.4.6 ?
19:20.08lirakisBentley:  but i use the built in transfer ability of the phone.. i do not use "feature codes"
19:20.15Bentleysame here
19:20.18Bentleyhrm
19:20.44lirakisBentley: yeah it works 100% as normal... ive used gxp-2000's a lot .. ive never had a transfer problem before
19:20.55[TK]D-Fenderrickross: I and my clients have been very please with them
19:21.12rickrossbrowsing their site now, TK
19:21.14*** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl)
19:22.15*** join/#asterisk festr_ (n=festr@ns.regnet.cz)
19:22.37festr_hello, anyone using musiconhold random=yes? it does not work for me in recent 1.4
19:22.49festr_always start first file
19:29.07*** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl)
19:35.32*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
19:35.42dukihello
19:38.07dukiI just installed asterisk 1.4.5, and configured the minimum in sip.conf and extensions.conf.  I run asterisk without errors but when trying to call a sip phone (registred one), the CLI tells me:
19:38.11dukiNo application 'Dial' for extension (sip, 2124, 1)
19:38.34dukiI have this line in extensions.conf in the sip context:
19:38.50dukiexten => 2124,1,Dial(SIP/mustapha,20,tr)
19:39.26dukiIt seems Dial is not recognized !
19:39.41*** join/#asterisk Meaty (n=meaty3@office.abi.ca)
19:39.55dukithanks for any help.
19:41.06hmmhesaysanyone ever get "operation not permitted" when trying to modprobe ztdummy?
19:42.00*** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar)
19:42.02MeatyHi all, anyone know why asterisk may not send Authorization header in a REGISTER sip packet when asterisk is used as a sip client ?
19:45.42EchinosIs anyone else able to connect to FWD with IAX2 right now?
19:46.11EchinosI'm not sure if it is a config problem on my (and other users) end, or a server issue
19:47.22[TK]D-Fenderduki: Go verify your modules.conf
19:47.42Echinosie. IAX2 support on FWD is unstable, not just unstable in general
19:51.56duki[TK]D-Fender: Thank you,  the dial module was not loaded.
19:56.40*** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579178.dsl.bell.ca)
19:57.21*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca)
20:05.11*** join/#asterisk asdx (n=foo@adsl-152-79.click.com.py)
20:05.15asdxhi
20:05.40asdxmy isp is blocking sip, is there a way to bypass that
20:08.20Strom_Muse different ports?
20:10.14asdxthey don't block ports, i think they do packet sniffing
20:10.41Strom_Mwell...perhaps use IAX instead then?
20:12.54*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
20:15.38asdxStrom_M: IAX works but a friend is using some special adapter that don't have IAX
20:16.02Strom_Mtunnel SIP over something else then, maybe
20:16.10asdxyeh
20:17.25Meatyin ssh maybe
20:17.33[TK]D-Fenderasdx: or setup * over there and have their device talk to *
20:17.50[TK]D-FenderMeaty: Yes, VoIP over TCP = fun!
20:18.02*** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net)
20:18.03*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
20:18.10*** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com)
20:18.12Meatyonly sip over tcp
20:18.31asdxis SIP over SSH possible?
20:18.36tamp4xanyone in ct area need a job? priv msg me.  gui users do not.
20:19.30*** join/#asterisk anonymouz666 (n=anonymou@189.25.220.191)
20:21.46shido6thats a damn good question
20:21.50asdx[TK]D-Fender: yeah
20:21.55shido6i've never had the time to try it
20:22.04shido6i'll try it tonight
20:22.25Corydon76-digasdx: I'd suggest over openvpn
20:22.35Corydon76-digTunneling over TCP isn't the best of ideas
20:22.49asdxCorydon76-dig: ok
20:22.55Corydon76-digespecially for voip
20:22.58asdxthanks
20:23.06shido6ssh has a faster setup time tho
20:23.16shido6if you're in a hurry and need something pretty secure ssh might work
20:23.27Corydon76-digshido6: you can tunnel over openvpn with no encryption
20:23.27shido6bring a phone and a usb stick with putty
20:23.45shido6how long does that take to setup?
20:23.58Corydon76-digIn fact, if the ISP is truly filtering, then tunneling with no encryptiion will work fine
20:24.11Corydon76-digshido6: handshake only
20:28.47*** join/#asterisk juxhi (n=juxhi@241-82.97-97.tampabay.res.rr.com)
20:28.54juxhihello
20:29.13juxhii am trying to compile zaptel but i keep getting an error
20:30.20Strom_Mplease share this error with us
20:30.30juxhii am recreating it
20:30.37Strom_Mlest we have to merely guess at it
20:30.39juxhigive me a sec
20:30.40Strom_Mand you don't want that
20:30.47Strom_Mbecause we'll think up things like
20:31.00Strom_MERROR: YOU HAVE NOT ADDED ENOUGH SHARP CHEDDAR TO TEH RECIPE
20:31.09juxhihmmm
20:31.13juxhiit might be that
20:31.14QwellStrom_M: next error I wrote...
20:31.16Strom_Mto which the solution is clearly "eat more sushi"
20:31.24QwellI'm stealing that, and crediting you
20:31.27Qwellwrite*
20:31.33Strom_Mhahahaahah, kickasss
20:31.53Strom_Mmake it show up at, like, verbosity 700
20:34.44Strom_MVERY LOW SODIUM
20:34.52juxhiu sure u want it?
20:34.52Strom_MPLEASE RECYCLE
20:34.55Strom_Myes
20:35.07juxhi/usr/src/zaptel-1.4.5.1/wct4xxp/../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_channel.c:9799: internal compiler error: Segmentation fault
20:35.23Strom_Mooh, that's a yummy oen
20:35.27juxhiseg fault
20:35.29Strom_Ms/oen/one/
20:35.32juxhii love those
20:35.46Strom_MQwell: uh, you code!  you fix it!
20:35.52drako...ok another weird problem.... when i get the calls from the ISDN (BRI) interfaces and i put it on the queue it works perfect but if first i put a background and then waitexten before the queue, when the call reach the queue and is answer it comes with no SOUND from the Caller
20:35.57drakobut if i get rid of the background and waitexten it works.
20:37.49juxhiso what's up
20:39.48juxhiit does this even with the packages provided by the ubuntu repositories
20:40.06Strom_Mjuxhi: i dont know where to begin; what architecture are you compiling on?
20:40.37juxhi386
20:40.48Strom_Mwhich version of ubuntu?
20:41.17juxhialternative version 6.06 i think
20:41.24juxhimight be 610
20:41.26Strom_Mserver?
20:41.29juxhiyes
20:41.38juxhibasically it's server
20:42.14Qwelljuxhi: what version of gcc?
20:42.45juxhi4.0.3
20:44.43*** join/#asterisk rene- (n=rene@200.34.66.137)
20:45.20linageeBentley: i'm running the latest. it seems very buggy. :(
20:45.35linageeBentley: i would NOT NOT NOT TRIPLE QUADRUPLE NOT upgrade to the latest.
20:45.55Strom_Mwhich "latest" are you going on about, linagee?
20:46.03rene-hey, when the manager documentation says variables passed to originate must be on its own line, does it means one Variable: var=val\r\n per each variable one wants to pass or maybe Variable" var1=val1\r\nvar2=val2\r\n\r\n
20:46.04linageeBentley: don't be tempted if they say a new feature makes GOLD when you boot it.
20:46.08chemikki love you people :)
20:46.25Bentleylinagee, these phones came shipped with the latest
20:46.30linageeBentley: yuck
20:46.37linageeBentley: 1.1 or 1.4?
20:46.45Bentleyi actually found out why blind xfer wasn't working
20:47.12Bentley1.1.4.18
20:47.19*** join/#asterisk smace (n=chatzill@200.220.198.107)
20:47.26linageeBentley: yes. just confirmed. that's the crappy one.
20:47.33linageeBentley: does the phone boot every time you turn it on?
20:47.39linageeor like 25%?
20:47.44Bentleylinagee, yes - it's working fine
20:47.47linageeweird
20:47.55Bentleyall features work .. iw as just having probs with the xfer
20:48.09Bentleybut turned out to be related to a freepbx config (i think)
20:48.53linageeBentley: i think a lot of people are so pissed off as to start a class action lawsuit against them if they don't fix the "can't revert firmwares, latest one breaks it" thing. :-/
20:49.30linageeBentley: in which case i'm sure they would point and laugh and say, "the website SAID not to upgrade! lol"
20:49.32asdxis vpn something like samba/smb/cifs?
20:49.34Bentleyheh - the phone's inexpensive .. you have to give them that
20:49.46Strom_Mi'd start a class action lawsuit for foisting shitty phones on the market in the first place ;)
20:49.55linageeStrom_M: true. :(
20:49.57QwellYou guys must be talking about gs
20:50.09linageeQwell: lol. i'm not going to say the name until i put a flame suit on
20:50.14Qwell~gs
20:50.15jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
20:50.16linageeflame resistant
20:50.17QwellYou don't have to
20:50.19Strom_MQwell: OMG HOW DID YOU KNOW !?!?
20:50.47linageeStrom_M: have you used a digium IAXy? i want an iax solution that's not going to have horrid NAT problems.
20:50.55Strom_Mlinagee: yes
20:50.56linagee(ata)
20:51.01shido6they rock
20:51.01Strom_Mi love love love my digium iaxy
20:51.12linageeStrom_M: is the IAXy good, or is there a good phone that has native IAX support? (not gs)
20:51.19shido6get an iaxy
20:51.22shido6unless
20:51.22Strom_Miaxy
20:51.28shido6u live in a very very hot and sandy climate
20:51.41linageeStrom_M: i thought hard phones are always better than ATAs. more functions/features/fun
20:51.50shido6hah
20:52.03shido6that must have come from someone who doesnt putz with features.conf
20:52.12linageeStrom_M: i need something to distribute to family. *maybe* an iaxy would work.... :-/
20:52.34shido6pre provision your iaxy's BEFORE you ship them to family
20:52.51linageeshido6: LOL! voip-ing the family is not such an uncommon thing around here? lol
20:52.52shido6and the PAP2-TNA u can get to and view a web interface to make changes if necessary
20:53.04shido6PAP@t-NA, rather
20:53.08shido6sunuva new keyboard
20:53.28linageeshido6: anything to screw ma bell out of another $0.50, right? :)
20:53.43shido6well...
20:53.48shido6thats how it starts
20:53.56shido6then you look at your bill after the year is out
20:54.03shido6and go......... wow..... i spent $2k
20:54.07linageeshido6: lol. yup.
20:54.10Qwell"Why do I even have a land line?"
20:54.13shido6but I was on the phone for about 90% of my life
20:54.13linageeshido6: we could actually split the phone bill
20:54.20shido6yeah
20:54.24shido6why do I even have a land line
20:54.27shido6oh yeah.. 911
20:54.30QwellI don't have one :P
20:54.36linageeshido6: my parents are paying cocks right now. they have horrid compression going on.
20:54.37shido6in my old city
20:54.45linageeshido6: and caller id for cocks is not a standard feature!
20:54.49linagees/cocks/cox/
20:54.56shido6you could walk to the hospitol sit in the emergency room for 3 hrs and see a doctor faster than the ambulance would show up
20:55.12linageeyikes
20:55.12Qwellshido6: I lived in L.A.  You would often get busy signals calling 911.
20:55.20J4k39 1 1 is slow in your town
20:55.23J4k3;)
20:55.31linageeshido6: best just to go to an emergency room beforehand or something and get their cell number and drive there yourself. lol
20:55.33Qwelland if you did get through, you'd have to go through an IVR
20:55.35shido6gunshot wound to the stomach sitting in the triage getting your blood pressure checked is kind of hilarious
20:55.42shido6waiting on the doctor
20:55.44shido6im sorry
20:55.48shido6waiting on a room to see a doctor
20:56.04J4k3shit, I suggest carrying tampons
20:56.12linageeshido6: knowing triage first aid FTW. :)
20:56.24J4k3(tampons work wonders on gunshot wounds... unless its a shotgun...)
20:56.34shido6then u just need some salt
20:56.38linageeJ4k3: people live through shot gun wounds?
20:56.39UnixDoglooking at askozia vs the digium gui
20:56.39shido6since you;ve already been peppered
20:56.57UnixDogyes
20:57.14*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:57.17UnixDogit depeneds on how close you where and what is in the load
20:57.20J4k3linagee: yeah..  I know a guy who was randomly shot in the gut with a .410 (small) shotgun.  he recently got a transplanted (!!!) colon so he could actually poop right.
20:57.27J4k3he was using a colostomy bag for a couple years.
20:57.33linageeJ4k3: yikes
20:57.35putnopvutDidn't that guy with Cheney get shot in the face with a shotgun?
20:57.38J4k3yeah.
20:57.43J4k3putnopvut: yeah, from a HUGE distance.
20:57.46J4k3like 30 meters.
20:57.47shido6thats the kind of day that must have been great
20:57.49J4k3with an open choke shotgun
20:57.54shido6talking to his 5 yr old son
20:58.01shido6... dad why are you so happy?
20:58.21shido6(fill in the blank)
20:58.24J4k3haha
20:58.28J4k3ever seen Bad Santa?
20:58.54shido6not yet
21:00.09[TK]D-Fendershido6, open choke is great.... if you're shooting slugs or flechettes :)
21:01.20*** join/#asterisk ZackZ (n=zzumbaug@70.244.109.129)
21:03.21ZackZIf I have a PRI with 9 bchannels, how should I make my dialplan so it can rollover to the next available number when it dials out?
21:04.03*** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net)
21:04.08QwellZackZ: look up zaptel groups
21:04.09*** join/#asterisk tomcontr3 (n=tomcontr@92-175-28.dial.terra.cl)
21:04.36tomcontr3Hi,  Im still having problems with my TDM400P + 2FXO Card.... When I call I hear like static noise...
21:04.48*** part/#asterisk ZackZ (n=zzumbaug@70.244.109.129)
21:05.05[TK]D-Fendertomcontr3, Have you lowered your gains to 0.0 ?
21:05.09tomcontr3yep
21:05.18tomcontr3I even set it to -5.0
21:05.18[TK]D-Fendertomcontr3, no improvement?
21:05.23tomcontr3but same story
21:06.33tzafrir_hometomcontr3, and you say that a simple analog phone performs well there, so it is not an issue with the line, right?
21:06.33tomcontr3I thought it could be the computer where it was installed, because it was a little old (P3) so I change it to a Core Duo2 Server.  But the noise was still there
21:06.53tomcontr3correct
21:07.01tzafrir_homeDo you have any other FXO interface? an FXS module, some ATA, whatever?
21:07.08tomcontr3if I plug the phone directly to the phone line there is not noise
21:07.19tzafrir_homeone or two channels is more than enough even for a P3 server
21:07.25tomcontr3no,  just the TDM400P with 2 FXO modules
21:07.33hmmhesaysshit I ran 2 sip channels on a p233
21:08.53tzafrir_hometomcontr3, can you try removing the echo canceller for a while, to eliminate it as a source for any mishaps?
21:09.13tomcontr3this is my zapata.conf
21:09.13tomcontr3http://pastebin.ca/697957
21:09.59ManxPowerDo you have the HPEC?
21:10.09ManxPowerAlso, if your gains are too high you can get ECFO
21:10.12ManxPower~ecfo
21:10.12jbotEcho Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out.  Some users describe it as "screeching", "feedback", "static", or other useless terms.  If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ...
21:11.52tomcontr3ManxPower,  what is HPEC?
21:11.53tzafrir_homeyeah, only the gains are zero and below
21:12.07tomcontr3you can check my zapata.conf... my gains are set to 1.0
21:12.48tzafrir_homeechocancel=64 ?
21:12.49russellbtomcontr3: a software echo canceller ... you'd know if you had it :)
21:12.50tzafrir_homeThat's low
21:13.21tzafrir_homeanyway, what happens if you remove the echo canceller?   echocancel=no
21:13.33tomcontr3same thing..
21:13.38tomcontr3I try it
21:14.11tomcontr3russellb,  I have sip phones... I dont use softphones.
21:14.15tzafrir_homemake sure that this is applied. You should see in 'zap show channel NNN' in the middle of a call that the echo canceller has "0 taps".
21:15.38tomcontr3let me check
21:15.42ManxPowertomcontr3: professional quality Digium software echo canceler
21:15.47[TK]D-Fenderhe has STATIC, not ECHO
21:16.05[TK]D-FenderGeneral craapy audio quality
21:16.15ManxPower[TK]D-Fender: I only mention that because of ECFO issues I had.
21:16.25ManxPowerUsers report it as "static"
21:17.05[TK]D-Fendertomcontr3, freebie test : disable EC.
21:17.18tomcontr3I dont have the echo problem,  I dont hear my self with delay,.... I hear a sound like if a were at the beach
21:17.23*** join/#asterisk arguile (n=arguile@KTNRON06-1242488957.sdsl.bell.ca)
21:17.38*** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com)
21:18.31ManxPowerI give up.
21:18.44waverly360ManxPower: give up on what?
21:18.54ManxPowertomcontr3: regardless of what you think, you will do better to follow [TK]D-Fender's advice.
21:19.09waverly360Oooh..I must've missed something.
21:19.12tzafrir_homeOn our devices I heard such things from devices that were not properly grounded. But I'm not sure how this applies to a PCI card in the computer's case
21:19.41tomcontr3you mean echocancel: no ?
21:19.46tzafrir_homeright
21:20.15*** join/#asterisk MACscr (n=MACscr@adsl-75-23-96-108.dsl.peoril.sbcglobal.net)
21:20.26[TK]D-Fendertomcontr3, If it improves then there's part or potentially all of your problem.
21:20.57[TK]D-Fenderscientificmethod++
21:21.35*** part/#asterisk smace (n=chatzill@200.220.198.107)
21:22.36MACscrAnyone have any quick tips for using a single polycom 501 on an asterisk system using SIP? The asterisk system is remote and polycom is behind NAT. There is no stun server. I have forwarded ports on the polycom network, but i still cant even see any proof that the polycom is even hitting the asterisk server at all
21:22.52tomcontr3no,  now I have echo AND the static noice
21:23.04MACscrHeck, my cheap grandstream gxp2k was easier to setup that the polycom. Thats sad
21:23.20*** join/#asterisk rantsh (n=rantsh@190.36.185.139)
21:23.30rantshHey people
21:23.38rantshHow is everything?
21:24.01rantshI keep getting a problem with a queue since the other day
21:24.14tomcontr3this is quite disapointing, because Digium products are no very cheap ...
21:24.19MACscrI hate to go through a  bunch of bulk phone setup procedures if im only setting up one phone
21:24.44rantshI got help from nicchap here and decided to upgrade my asterisk 1.2.3 to 1.2.24 (on a test environment of course)
21:25.31tzafrir_hometomcontr3, well, at least you know that the EC is effective :-(
21:25.59[TK]D-FenderMACscr, here :
21:26.01[TK]D-Fender~sipnat
21:26.02jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:26.10rantshI posted my configuration here http://pastebin.com/m20ed3434
21:26.43rantshand still, I keep getting asterisk accept callers in the queue when there are no agents logged in
21:26.46tzafrir_homeMACscr, if just the phone is behine NAT, then just set nat=yes in the phone's entry in sip.conf .
21:27.09rantshfurther more it rings one of the agents direct number even if he's not logged in
21:27.15tzafrir_homeNo need to forward posrts and such
21:27.18rantshany ideas what could be going on?
21:27.21MACscrTzafrir_home : yes, its just the phone.
21:27.39tzafrir_homeUnless your packets travel through an aggressive NAT router
21:27.55tzafrir_homethat mangles SIP headers on its own
21:28.13MACscrI still do not see why i cant see any indications in the asterisk cli that the phone is even attempting to contact the server
21:28.39tzafrir_homeMACscr, have you enabled SIP debug?
21:28.45MACscryes
21:29.15tzafrir_homenext step: a sniffer (tcpdump / wireshark / whatever)
21:29.32MACscrAh man, that type of stuff is a bit over my head. grr
21:29.41rantshthis is the ooutput in the * cli ... http://pastebin.com/d68392573
21:29.43[TK]D-FenderMACscr, for your phone's sip.conf entry : "nat=yes", "canreinvite=no", and do NOT forward any ports to it.  Describe your * side now...
21:29.48tomcontr3tzafrir_home, something is something,  but the this is that I have to make this work here ate the office,... and we bought that card, becuase we thought Digium was good
21:30.11tomcontr3but anyway
21:30.24tomcontr3I will have t wait and see that the support guys say
21:30.30*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
21:33.34rob0IIRC, Digium includes support to get a device running. They did for us, even later when we went back begging for help.
21:33.45rantshcan anyone throw me a hand here?
21:33.51rob0but they're probably closed for the weekend.
21:34.42*** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com)
21:36.08Qwellrob0: still got another 2.5 hours, I believe
21:36.35rob0oh, ha.
21:36.52rob0Serves 'em right for sleeping in so long.
21:37.59*** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net)
21:38.20[TK]D-Fenderrantsh, pastebin "show agents" and "show queues"
21:39.04rantsh[TK]D-Fender: right away
21:40.15rantsh[TK]D-Fender: http://pastebin.com/d27c4e157
21:41.20rantsh[TK]D-Fender: I don't understand why it keeps ringing 9923181 :s
21:41.33[TK]D-Fenderrantsh, SIP/9923181 is available and because of that new callers don't get kicked
21:41.55rantsh[TK]D-Fender: but it's not declared as a member anymore
21:42.03[TK]D-Fenderrantsh, *I* see it there...
21:42.17[TK]D-FenderSIP/9923181 (dynamic) (Not in use) has taken 2 calls (last was 1382 secs ago)
21:42.23rantshI did change agents.conf to use 1001,1002,1003
21:42.44[TK]D-Fenderrantsh, doesnt' matter, that is a DYNAMICALLY added memeber based on that now disabled line from your dialplan.
21:42.53rantshI know, I saw it too; I can't understand why he's a member
21:42.57[TK]D-Fenderrantsh, You should use removequeuember it.
21:43.20[TK]D-Fenderrantsh, ;exten => 56446,1,AddQueueMember(queue1|SIP/${CALLERIDNUM}) <-------- this was why
21:43.38[TK]D-Fenderrantsh, You did this, then commented out the dialplan that added them in the first place
21:43.51rantsh:o
21:44.18ManxPowerWell there's a few mins of [TK]D-Fender's life he will never get back
21:44.21rantsh[TK]D-Fender: so it keeps remembering him as a member even though I've reloaded and restart many (MANY) times?
21:44.39[TK]D-Fenderrantsh, yup
21:44.47[TK]D-FenderManxPower, Cry for time lost
21:44.49rantsh[TK]D-Fender: :s that sucks
21:45.15[TK]D-Fenderrantsh, jsut think if you reloaded to take other changes and kept kicking out your agents!
21:45.21[TK]D-Fenderrantsh, This is a FEATURE!
21:45.52rantshso I should use removequeuemember in the dialplan to remove this agent, then scrape it out, right?
21:45.58rantsh[TK]D-Fender: I see your point
21:46.02[TK]D-Fenderrantsh, yup
21:46.26*** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga)
21:46.43rantsh[TK]D-Fender: I'll try that [TK]D-Fender, thanks for your help again
21:49.54*** part/#asterisk InsomniaCity (n=insomnia@raptor.ukc.ac.uk)
21:51.37*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
21:52.07rantsh[TK]D-Fender THANKS MAN IT WORKED
21:52.31rantsh[TK]D-Fender: You're the best dude
21:52.59[TK]D-Fenderrantsh, you're welcome
21:53.18*** part/#asterisk TicoTuco (n=matheus@200.250.100.25)
21:56.40_x86_HAHAHA! SCO files for bankruptcy!
21:56.49_x86_"on the eve of the Novell trial" ;)
21:56.53_x86_oh man this is awesome
21:57.09Qwellhow is it awesome?
21:57.17QwellIt means Novell doesn't get anything for a while now
21:58.03_x86_wasn't SCO the one chasing Novell for damages?
21:58.18_x86_IBM was the one chasing SCO
21:58.24_x86_iirc
21:59.05Qwellnovell has counterclaims
22:00.19_x86_ah
22:03.05*** join/#asterisk jsidhu (n=atomik@66.206.163.185)
22:06.27*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
22:06.37rantshI'm pretty sure anyone can help me with this n00b question
22:06.52jsidhui need some help with setting up FXS. I have an analog DID line, which uses Wink Start, OutPulse 4 digits DTMF.  How can I configure my TDM400's FXS module to work with this? I'd appreciate it if someone could point me in the right direction.. I'm sort of lost reading all kinds of different examples..
22:06.54[TK]D-FenderNEXT!!@!@@!@ (c) BKW
22:07.15rantshI want to play a message when an extension is busy/disconnected, how can I set it up?
22:07.19rantshI tried:
22:07.44rantshexten => 2050,105,Playback(tt-monkeys) ... but it didn't seem to work
22:07.55tzafrir_homejsidhu, FXS module?
22:07.55[TK]D-Fenderjsidhu, if its raw analog, Answer the line and dump the call immediately into an IVR with no prompting and a XXXX pattern to trap the DID
22:08.10[TK]D-Fenderjsidhu, and that should be FXO
22:08.20jsidhuFXO for an incoming DID line?
22:08.24jsidhuok
22:08.31rantshbtw, 4 is a Queue cmd
22:08.37jsidhugood thing i ordered the card with a fxo and fxs
22:08.41[TK]D-Fenderjsidhu, And as a fallback use "i" and "t" to default to whatever you'd consider your primary DID.
22:08.46jsidhuhow do i deal with the wink?
22:09.08jsidhui need to send a wink to get a dial tone on the line, its not a PSTN line
22:09.22[TK]D-Fenderrantsh, it doesn't priority jump.  just make it #5
22:09.37jsidhuerr.. i mean its not a regular phone line, its an analog incoming did line, that i need to send a wink on to get it to work
22:09.43jsidhuhopefuly im making some sense
22:09.43rantshok
22:10.46[TK]D-Fenderjsidhu, then right after you answer I think you should call "Flash" to "wink" the line
22:13.11*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
22:13.15jsidhuTK: i need to wink the line before I can answer
22:13.40jsidhuif I dont wink, I wont get a dialtone
22:13.54jsidhuno dialtone, no calls incoming that can be answered
22:13.56jsidhu:/
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22:17.51*** join/#asterisk NirS (n=NirS@84.94.145.166.cable.012.net.il)
22:17.58NirSgood evening everybody
22:18.34NirSanybody home ?
22:19.11*** join/#asterisk jsidhu2 (n=atomik@66.206.163.184)
22:19.20jsidhu2sorry, i timed out after my last reply
22:19.27jsidhu2any ideas td
22:20.09tzafrir_homeNirS, hi
22:20.18NirShey tzafrir, wassup ?
22:20.59NirSI'm having some really funky IAX2 problems over here
22:21.03*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
22:21.24NirSkeep getting INVALs for something that works in another location no problem
22:23.16[TK]D-Fenderjsidhu : whe n it rings, answer, then issue the wink to accept the call, then dump into IVR as I mentioned
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22:23.38*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
22:25.27NirSanyone has a clue for this with DIDx ?
22:25.27jsidhu2i think u misunderstand me, it will not ring unless I send a WINK on the line to tell whatever's on my telco's side that the line is hooked up to a PBX, when that wink is sent, I will get a dialtone and only then can calls come to that line.
22:25.28NirSRx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: INVAL
22:26.57[TK]D-Fenderjsidhuit sounds like you are mixing up your wording for an INCOMING call and that of an OUTGOING call.
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22:27.46jsidhu2i dont think so, why do you say that
22:27.56jsidhu2this line that I speak of is not a regular analog phone line.
22:28.17jsidhu2this is a DID analog line, which I must FIRST send a wink to, sending this wink activates the line.
22:29.12[TK]D-Fenderjsidhu2, ok, WHEN do you send this wink, and what happens NEXT?
22:29.38tzafrir_homeNirS, I really don't know IAX2 that well, but can you provide a more complete trace?
22:29.47jsidhu2I have to send this wink to "turn the line on". when that wink is send, then the telco side will start sending me calls on this line.
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22:30.06NirSthat's exactly it, there is no more trace
22:30.14iPod-nanoWhere are the config files located on a Debian system?
22:30.27NirSI wish I could see that the request coming in from DidX is correct, but I can't
22:30.39tzafrir_homeiPod-nano, /etc/<package-name>, and thus /etc/asterisk
22:31.21tzafrir_homewhich also happens to be the same sane choice made by Asterisk :-)
22:31.53iPod-nanoYeah, I'm trying to run my first Asterisk machine.
22:32.53iPod-nanoI'll be surprised if the ancient computer I'm using will even succeed.
22:33.44tzafrir_homeWhat computer is that?
22:34.01iPod-nanoUm... old.
22:34.04iPod-nanoOld Compaq.
22:34.12tzafrir_homeCPU? memory?
22:34.13iPod-nanoThat surprisingly can handle Debian.
22:34.21jsidhu2..
22:34.35iPod-nanoWhat does 60416 KB equate to?
22:34.44jsidhu260mB
22:34.46tzafrir_home64MB?
22:35.02[TK]D-FenderGTG, back in many hours :)
22:35.06jsidhu265536=64mg
22:35.13jsidhu2b
22:35.15iPod-nanoAnd no clue how to make this thing tell me what speed its processor is.
22:35.18tzafrir_homeminus some overhead
22:35.29tzafrir_homeiPod-nano, cat /proc/cpuinfo
22:36.27*** part/#asterisk Netgeeks-laptop (n=chris@204.11.231.198.static.etheric.net)
22:36.32iPod-nanoI don't know enough about Linux. :-P
22:37.17iPod-nanoWow, 225.030 MHz
22:37.24iPod-nanoThis thing is fast! :-D
22:37.29*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
22:37.31iPod-nanoI was expecting a 133.
22:37.34*** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com)
22:38.40jsidhu2crap. the TDM400p doesnt support Analog DID Lines??
22:39.38iPod-nanoYeah, it;s old.
22:39.50iPod-nanoThat's why it's been reduced to a console-only server.
22:40.30tzafrir_homejsidhu2, technically it is not the TDM400P that doesn't support it. It is Zaptel (from what I understand)
22:41.16jsidhu2i see, do you know of a way to use such Analog DID trunks with Asterisk?
22:41.30tzafrir_homeiPod-nano, a really borderline computer. Don't expect ot get too much from it. May make a nice toy PBX
22:42.02iPod-nanoIt's not going to be controlling more than one or two clients.
22:42.10jsidhu2it should be ok for 1 or 2
22:42.35iPod-nanoAnd for me it is just a toy, anyway.
22:42.43iPod-nanoThe whole point is to teach myself.
22:43.10iPod-nanoAnd then I might upgrade to my screaming fast 500 MHz box.
22:43.31tzafrir_homeYou also need there more memory
22:45.30UnixDogwhy 500 mhz is more then enough
22:45.43UnixDogthats a kickass pbx box
22:45.47iPod-nanoYeah, but that's not the machine I'm using.
22:45.53*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
22:46.10UnixDog500 mhz/256 megram and a laptop drive
22:46.20UnixDogdood that be a rocking box
22:46.21iPod-nanoI'm gonna start off simple with a computer that's useless for any other purpose.
22:46.32UnixDoga p2 266
22:46.41UnixDogor a p1 233
22:46.55iPod-nanoPentium I.
22:47.07UnixDogand what distro
22:47.18iPod-nanoIt was running Windows 98.  I gutted it and installed Debian.
22:47.27UnixDogok
22:48.03UnixDogwhat if I could give you a quick bsdinstall and pkgs that install everythign and have you up in less time then building on deb
22:48.43UnixDog+ the digium gui
22:48.54UnixDogor my gui I work with
22:49.14iPod-nanoGUI is practically out of the question with this machine.
22:49.40UnixDogwhy I run the asterisk-gui on a 486/100
22:49.46UnixDogand its fine
22:49.57UnixDogit uses the asterisk builtin httpd server
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22:56.50watchyanyone here do vlans with linux
22:57.06drakoUnixDog, whats so good with astersk-gui?
22:59.24UnixDogI use it just to save time
22:59.44*** join/#asterisk tsurko (n=tsurko@78.90.100.214)
22:59.44UnixDogask digium whats so good about it
23:00.01UnixDogthey have taken over a year to get ti tpo where is it now
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23:00.25MACscrThis channel is way to anti gui, its completely rediculous and ignorant
23:00.59iPod-nanoIt's not anti-gui so much as it's that GUI isn't an option for some of us.
23:01.12MACscrThat doesnt make it bad
23:01.34MACscrIpod-nano : many people in here are Anti-gui, not matter how you spin it
23:01.39outtoluncor maybe it isn't anti-gui it it is ANTI support for guis that do not support themselves <G>
23:02.31UnixDogthe big issue on guis is the fact they all use diff files
23:02.38iPod-nanoI couldn't live if I had to live only in the console.
23:02.48UnixDogand only 2 I know of even have fully functional dialplans
23:02.55iPod-nanoBut, GUIs take more resources, it's that simple.
23:03.20adorahwell, resources are dirt cheap these days..
23:03.26iPod-nanoAlso, if your administering a remote machine over a network, graphics are a very bad idea.
23:03.42iPod-nanoI'd rather ssh into my ancient machine.
23:04.13iPod-nanoMe, I try to find uses for older computers.
23:04.35UnixDogbut the big fact I still stand by this day. no one has put a fully functional dialplan out there for asterisk
23:04.49adorahwhen one get for ~200$ a machine that can handle E1 with 60 users, resources are not an issue..
23:04.55UnixDogand there for using asterisk out of the box is not functional
23:05.02[hC]any of you guys using an aastra 57i with expansion modules & BLF?
23:05.16UnixDogBLF in asterisk sucks
23:05.23UnixDogits still in need of work
23:05.35[hC]yeah no kidding, that doesnt change the fact that people need to use it :)
23:05.45[hC]i wonder how much better it is in 1.4, or if it even got any attention
23:05.45rob0Some of us are proud of being ridiculous and ignorant!
23:05.48UnixDogneed want
23:06.16[hC]UnixDog: i dont really view BLF as a want. if customers want it, that means i need it. :)
23:06.27rob0But the funny thing is when someone who needs help calls the people he wants to help ridiculous and ignorant. ;)
23:06.35UnixDogwell its time to shove the stick up asterisk groups ass and get them to understand that there needs to be a fully functional Dialplan
23:06.47adorahcustomers demand BLF
23:06.48rob0Hey all you R&I people, I need help! ;)
23:06.55rob0Here
23:07.11[hC]It works, but on this aastra it likes to crash and lock up. not sure if its asterisk blf, or the phone, or what
23:07.17rob0's the thing: my * server at home is on lousy power and Internet service ...
23:07.20[hC]Looking for someone else doing it to hear their stories.
23:07.43rob0... and I'm not home much (I work ~500km away from home)
23:08.09rob0... I got it so it comes back on when the power is restored.
23:08.42rob0... But: when the ISP is down and it comes on, I don't get online, and * doesn't like that either.
23:09.01rob0How can I make sure * will start even without an Internet connection?
23:09.42rob0See, I have an FXO there so the POTS line should be picked up.
23:10.02UnixDogit should install a startupscript in /etc/rc,\.d
23:10.08UnixDogit should install a startupscript in /etc/rc.d
23:10.30UnixDogif not you can always start it at the bottom of rc.local
23:10.39UnixDogsafe_asterisk
23:11.18rob0I do this. It DID try to start at boot time. But it barfed because no Internet probably broke the SIP and IAX channels.
23:11.31UnixDog?
23:11.40rob0(I am guessing, I have no way of knowing what it said)
23:11.41*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
23:11.55UnixDogthats sounds funky since asterisk does not require a internetconnection to start
23:12.19rob0some of my channels do
23:12.24dan__tAnyone use Teliax by chance?
23:12.36dan__tI have a pretty dumb question, but I can't find the answer heh.
23:12.39rob0yeah, I was surprised, the machine was up and running when I got here, but * was not.
23:12.50dan__tI guess I'm wondering if outbound calls are supported through the plans
23:13.04dan__tTo make sure that its just a case of me doing something dumb when configuring
23:13.13rob0I think Teliax is an outbound provider ...
23:13.29dan__tIs it uncommon to have a provider that only does inbound?
23:14.18UnixDogthey only do trunking
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23:14.29UnixDogthey offer no functionality
23:14.29dan__tI suspect that's all I need.
23:14.40UnixDogthere are other providers
23:14.41dan__tI've had the account for like two years and just finally started using it.
23:15.01dan__tI recall choosing it because they offered IAX2, and if I recall further, I believe that was not common back in the day?
23:16.03UnixDognufone
23:16.07UnixDogthey do iax
23:16.20UnixDogfwd has always offerd iax
23:16.27rob0dan__t: I'm just a small-timer, but for inbound I use Stanaphone and ipkall -- both are free.
23:16.49dan__tTryin' to rig a toll-free number out of the deal, too
23:18.31dan__tWelp, for paid IAX trunking, with the abiity to get a toll free number at decent rates, does anyone have any suggestions?
23:18.40dan__tI hate to poll like this, but I just want to see what's out there :)
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23:28.34rob0dan__t: Asterlink maybe, I use them for outbound
23:29.19rob0I never did get the inbound toll-free number working, but I don't really need it.
23:29.23dan__tahh
23:30.07*** join/#asterisk [ProB]CrazyMan (n=niethamm@pd907f938.dip0.t-ipconnect.de)
23:30.31[ProB]CrazyManhello
23:30.51dan__thi
23:32.37giesenhow the hell do you do a catchall extension
23:32.50giesenit's driving me nuts
23:32.59*** join/#asterisk Joneser (n=Joneser@pool-71-170-201-50.dllstx.fios.verizon.net)
23:33.39[ProB]CrazyManI have to upgrade my asterisk, therefor I also need to update spandsp (app_rxfax.so and app_txfax). does anybody know which spandsp version is the one whch is stable for asterisk 1.2.23 ?
23:37.22blitzragegiesen: _.
23:37.31file[TK]D-Fender: You are here.
23:37.38blitzragefile: you are there
23:37.47fileI am.
23:37.51blitzrageFalse.
23:40.33*** join/#asterisk Poehali (n=actionma@74.93.5.186)
23:40.39Poehalihey people! I got it working!
23:40.39*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
23:41.06Poehalitook me an entire week of trial and error
23:41.17Poehaliand I tink I broke the box in the process
23:41.54codefreezeIt's $180 for 1 OEM copy of Windows Vista Ultimate. (newegg). Am I a fool? What's so hot about ultimate vs. home vs. business?
23:42.08filefeatures and licensing
23:42.13filealthough OEM...
23:43.00TJNIIAnyone have a budgetone 100?
23:46.15TJNIINm, I found the option that was horking it.
23:47.37MACscrDoes anyone know if i can configure the voicemail button on a polycom 501 with its gui?
23:49.43*** part/#asterisk macli (n=macli@nmc.brc.ubc.ca)
23:50.03codefreezeI'm using #asterisk for a filter!
23:50.42codefreezeI make a simple statement, like "I found a budgetone 100 for $579.99! Is that a great deal or what?"
23:51.13codefreezeAnd if nobody calls me an idiot, or names a cheaper price, or offers me a bridge,
23:51.31codefreezeit must be somewhere near a fair deal.
23:52.37MACscrLol, your an idiot
23:53.38codefreezeEither that, or everyone is so stupefied, as to not know what to say! :)
23:56.00codefreezeMACscr: lol, that's a given!
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23:59.22dan__torly?
23:59.31dan__trob0, haven't seen you around in quite some time heheh
23:59.44rob0I haven't been in here.
23:59.52rob0(this channel)

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