IRC log for #asterisk on 20100321

00:00.12*** join/#asterisk DennisG (~DennisG@84.30.136.208)
00:03.07*** join/#asterisk jksM (jks@193.189.93.254)
00:11.54*** join/#asterisk fofware (~chatzilla@186.125.110.227)
00:12.11ChannelZquiet this evening
00:16.33TJNIIgrumbles about taxes. Specifically about how Iowa taxes your federal refund.
00:17.21*** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net)
00:17.59*** join/#asterisk bjhaid (~herbayjha@41.206.15.2)
00:18.17ChannelZhmm.  I don't taxed on my federal refund but the state taxes the state refund from the previous year
00:19.24TJNIIwhips out Colorado form 104 to look
00:19.58ChannelZthey send a card in the mail
00:23.53TJNIIForm 1099-G?
00:24.27ChannelZyeah I think thats it
00:26.05TJNIII have to do the long form because I got a 1099-R.  However, that made me realize I could deduct my moving expenses so it evens out.  Though it does suck having to do two state returns this year.
00:28.13hardwireok peeps.. chan_ooh323 isn't even sending the remote IP to libooh323c
00:28.26hardwireit's attempting to do peer matching for some reason on direct to IP
00:28.36hardwireand even when matching peers .. it fails to match them
00:28.40hardwirepile of poo?
00:31.21carraruse SIP
00:33.34hardwirenot an option atm.
00:33.35hardwiresigh
00:34.11Znuffthen use IAX
00:34.59TJNIII see the channel is in a helpfull mood tonight....
00:35.07carrarhaha
00:35.29*** part/#asterisk beta2k (1000@d24-36-68-97.home1.cgocable.net)
00:40.17*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
00:55.40dlynesWhy is the US federal and state tax returns so freaking complicated, anyways?
00:56.34dlynesHow do you even figure out which ones you need to fill out?  We got so frustrated last year with our corporate tax return (our first one), that we just hired an accountant...but they charge an arm and a leg
00:57.04dlynesSomething like $2100 for what should have been a straight-forward return
00:58.06TJNIIWell, there is legislation to simplify it.  But due to the wonders of the lobbying system it gets blocked by tax preparers.
00:59.56dlynesyeah...in Canada, we've only got one form to fill out, and a few sub-forms
01:00.10dlynesBut, when we had to file for our LLC's, it seemed to be a nightmare
01:00.43dlynesAnd the main form tells you which subforms you need, too
01:01.40ChainsawI'm in the UK. I don't need to fill in a form at all. It's automatic.
01:02.31ChainsawI don't think the corporate ones are quite as complicated as in the US either.
01:05.17dlynesChainsaw, so how do they determine whether they've overcharged you throughout the year or not, then?
01:05.30dlynesChainsaw, or they just screw you over, and you have no say about it?
01:06.39Chainsawdlynes: If you don't agree with what they've done, you can request the forms.
01:06.55dlynesChainsaw, ah
01:06.57*** part/#asterisk ryanlin (~ryanlin@you.better.not.nulroute.us)
01:07.52*** join/#asterisk ryanlin (~ryanlin@you.better.not.nulroute.us)
01:09.01carrardlynes, turn on c-span and watch it this weekend and you understand why
01:18.18*** join/#asterisk pentanol (~pentanol@77-35-13-094.pppoe.primorye.net.ru)
01:29.57*** join/#asterisk viq (~viq@unaffiliated/viq)
01:36.00*** join/#asterisk fofware (~chatzilla@186.125.110.227)
02:02.49*** join/#asterisk CrashSys (Kumba@azrael.crashsys.com)
02:03.10CrashSysIs there any way to get something more descriptive then asterisk not liking my dahdi config? http://pastebin.ca/1847736
02:03.51ChannelZ3 bullet points, that seems pretty descriptive
02:04.21ChannelZBut we're not clairvoyant. Do you have hardware needing DAHDI?
02:06.37CrashSysSangoma A104, as configured through wancfg_dahdi
02:08.11CrashSysMore descriptive as to what it didn't like about Dahdi. Like "dahdi devices configured but not found", or something
02:11.07ChannelZit's a timing error which means either the drivers didn't load, or you're misconfigured and it's not talking to the hardware properly
02:14.28ChannelZneeds dinner
02:14.37Kattywell.
02:14.41Kattyi got my nike + account setup
02:14.51Kattythis little app is pretty neat
02:23.06*** join/#asterisk fofware (~chatzilla@186.125.110.227)
02:27.04carrarPICS!!
02:35.52Kattyoh
02:35.53Kattyumm
02:36.17jayteeummm
02:36.50Kattyhttp://nikerunning.nike.com/nikeos/p/nikeplus/en_US/plus/#//runs/history/276281824/all// <- does that link work?
02:37.21*** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk)
02:37.34jayteeyeah it works
02:37.51Kattythat's a walk, not a run
02:38.29*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
02:38.32jayteethat's good because otherwise you'd fail physical qualifications in basic training. the Air Force used to require a mile in 8 minutes or less
02:38.59Kattyi'd probably fail that regardless
02:39.13Kattybut i do enjoy a leisurely walk
02:39.26jayteea mile in 8 minutes wasn't tough even for smokers. it was the fattys that had problems
02:39.59fileruns around Katty
02:40.29Kattywell i have been getting a bit around the front side due to winter laziness.
02:40.38Kattyand not the upper front side.
02:42.20jayteeI've lost 18 pounds just from switching to decaf and drinking mostly spring water and only allowing myself a couple cans of Sprite a day instead of drinking 4 or 5 cups of regular coffee and 6-8 cans of Coke. Plus I've been eating healthier foods too.
02:42.44jayteesoda really bloats your gut out
02:52.10Pan3Dquit soda two years ago and went down a pant size
02:54.09Kattymy problem isn't soda.
02:54.14Kattymine is not doing anythign when it's cold.
02:54.43Kattywhich adds up quick when you're used to running 3 times a week
02:56.09TJNIII'm really looking forward to the lakes thawing so I can go canoing again.
02:56.23TJNIII've got a busted knee so I can't run.  I canoe instead.
02:56.49jblackI'm starting to lose too. =)
02:57.15TJNIIshould go out to the Big Thompson Ponds tomorrow and see if they are thawed yet....
02:57.28Kattyoh man, i can't wait till hit gets in the upper 80s for floating down the river again
02:57.38Kattythat was so much fun, short of getting sunblock in my eyes
02:57.55Kattynever been canoeing before.
02:58.00Kattyhugs jblack
02:58.22TJNIIOnce you learn how to not fall into the lake, its a lot of fun.
02:59.02TJNIINarrow boats are not tolerant of side-to-side center of gravity changes.
02:59.59VoIP-PenguinI've been down the current river a few times.
03:00.07KattyVoIP-Penguin: do you live near it?
03:00.28VoIP-PenguinNot really.  We had to drive a ways.
03:00.49Kattyah, right.
03:01.01VoIP-PenguinWe had a big group, which makes things more fun.
03:01.08Kattywe're just a couple hours from Van Burren, mo...a major place for floating
03:01.25Kattywe just had 3 people last summer...it was still fun
03:08.37Slugs_.
03:09.13Kattyhi slugs
03:09.20Slugs_hi Katty
03:09.37Slugs_my h323 works!
03:09.57Slugs_avaya <=> asterisk!
03:10.22Slugs_this stuff is amazing, thx to hardwire, he helped me out
03:10.37Kattyyay (=
03:10.43Kattyhardwire is good peeps.
03:10.48Slugs_id say
03:11.22Slugs_Katty, what do you do w/ asterisk?
03:11.30Slugs_job, fun, both?
03:12.13*** join/#asterisk ruben23 (~ITadmin@122.55.48.243)
03:13.30*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
03:15.47ChannelZdamn I just spit a piece of ice into my keyboard
03:16.37ChainsawChannelZ: Disconnect it and air-dry and you should be fine.
03:16.58ChainsawChannelZ: Or if it's a laptop, turn it off before the ice melts, air-dry upside down and you should still be fine.
03:17.16Chainsaw(With the screen open, obviously)
03:17.20*** part/#asterisk ruben23 (~ITadmin@122.55.48.243)
03:17.21ChannelZwell it was a tiny piece I'm not too worried about it.
03:17.40ChannelZjust a wild party over here, spitting up on my computer
03:17.48Chainsaw*G* Alright.
03:17.55ChainsawLocal time's 3am, at any rate. Time for bed.
03:18.02ChannelZnighty
03:20.02KattySlugs_: mostly for work, but i do have one here at home i setup (=
03:21.40Slugs_home is all voip or not?
03:22.36Kattyi have a pots lined plugged into the server
03:23.28Katty1 polycom, and 1 snom
03:23.37Slugs_do you know of any free voip providers?
03:23.49Slugs_is there even such thing
03:25.25Kattyi've never heard of one
03:26.39*** join/#asterisk ruben23 (~ITadmin@122.55.48.243)
03:26.52ChannelZHmm anyone used nf_conntrack_sip ?
03:26.59Slugs_Katty, if i can make calls to particular extensions on another pbx does that mean i can make outside calls if i setup the dialplan correctly?
03:27.58Kattythe other pbx will have to have a registration for you
03:28.23Kattyit's the equivilent of registering with a sip providor
03:28.39Slugs_interesting
03:28.48Kattyif their dialplan allows, you can make regular outgoing calls off their pbx.
03:29.10Kattya lot of people here have a sip number setup for a meetme room for conferencing
03:29.11Slugs_would i have to do anything on the * end?
03:29.31Kattywell, you'd have to at least add the sip number you want to call to your dialplan
03:29.35ChannelZno it's all completely magic
03:29.48Kattyif you're planning on making calls, you'd obviously have to setup a registration in sip.conf to register with the other server
03:30.09Kattyand then setup your dialplan to 'dial' out that registration
03:30.35Kattyunless you have a person friend in the sip providor business this isn't really going to happen
03:30.41Kattys/person/personal/
03:31.01ChannelZDUNDi
03:31.15Slugs_so just because i can call from * to an avaya extension does not mean i can dail out via avaya, it needs to be registered
03:31.20Kattydundi can be useful in routing calls through a largescale pbx setup
03:31.43Slugs_this is all so interesting ;)
03:31.47Kattywell your server has to know what to do when you dial 1234567890
03:31.55Kattyelse it's going to sit there and go durrrrrrrrr
03:32.06ChannelZor the asterisk lady will yell at you
03:32.07Kattymost people do digit mapping.
03:32.14Katty7 digits, 10 digits, 11 digits
03:32.28Kattythey can all go out the same thing
03:32.41Kattyor you can send the 7 digits to pots lines, 10 digits to pri, 11 digits to sip providor
03:32.56Kattyor randomly send all 7 digit calls to uhhh pizza hut
03:33.06Kattyso any 7 digit number you dial, dials pizzahut
03:33.10Kattyor ejects the cdrom drive.
03:33.31Slugs_see, i figured i would just have to make a dialplan that allows me to dial for example, to send 7 didgets w/ asterisk, and since i can call avaya extensions it would work
03:33.32Kattyyou can execute scripts via System()
03:33.44Slugs_lol
03:34.12ChannelZIf you can dial out through avaya through whatever means you normally used to, you can make * do it too
03:35.00Slugs_ok, so im correct in what im saying then?  it's already setup on the avaya end or no?
03:35.14ChannelZThat's between you and avaya, I have no idea
03:35.15Kattyi am not familiar with avaya.
03:35.24Slugs_i see i see
03:35.28ChannelZAre they a telecom provider?  Like your phone service provider?
03:35.36Slugs_no its a pbx
03:35.44Slugs_like *
03:35.57ChannelZso it's a local thing you run
03:36.01Slugs_yes
03:36.20Slugs_avaya is setup to connect to the outside
03:36.25Slugs_my * is not
03:36.26ChannelZOk.. so normally your physical phones talk to this other PBX to make and receive calls
03:36.31ruben23hi
03:36.35Kattyhi ruben
03:36.38Slugs_ChannelZ, exactly
03:36.54ChannelZBut you're sending calls to it through *.. so yeah you should be able to dial anything you could dial from a phone if you configure your extensions.conf as such to do the proper thing
03:37.04ruben23safe_asterisk is part of the asterisk installation right..?
03:37.21Slugs_awesome ok hold on!
03:37.27Kattyi'm not familiar with safe_asterisk
03:37.30ChannelZruben23: should be
03:38.11ChannelZruben23: it's actually a big script
03:38.11ruben23ChannelZ: where i can find it..?
03:38.30ChannelZ/usr/sbin normally
03:38.32Slugs_exten => _4xxxx,1,Dial(H323/avaya/${EXTEN})
03:38.48ChannelZor it's in the contrib directory somewhere of the source
03:39.18*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
03:39.18Slugs_4xxxx are avaya extensions
03:39.19Kattyso 4 digit extensions starting with 4?
03:39.23ChannelZKatty: safe_asterisk is a script that you run instead of the asterisk binary.. it runs asterisk instead, and then re-runs it if it terminates for some reason
03:39.34Slugs_yes
03:39.41Kattywell that will be easy to change
03:39.46ChannelZSo what do you dial to dial a real phone number?
03:39.58Kattyi'm sure the avaya is already setup to process 7 digits
03:40.06Slugs_from avaya i dial 19 and rthe number
03:40.09Kattyso just _xxxxxxx,1,Dial,e tc
03:40.30Kattyokay so then: (H323/avay/19${EXTEN})
03:40.46Kattyso if i dialed 2004005
03:40.51Kattyit would dial 192004005
03:41.05ChannelZexten => _19NXXNXXXXXX,1,Dial(H323/avaya/${EXTEN}) for 10-digit dialing
03:41.13Kattymmmm
03:41.14ChannelZerr ${EXTEN:2}
03:41.17Kattywell
03:41.25ChannelZor no not :2 you have to send the 19 to them
03:41.28Kattyi guess it depends on how the pbx users are going to do it
03:41.28ChannelZso nevermind
03:41.40Kattythey can dial a 7 digit number, and asterisk can add the 19 for them
03:41.50Kattyor than can remember to dial 19
03:41.58Kattys/than/they/
03:42.00ChannelZthat too
03:42.01Kattyyou can really do it either way
03:42.24ChannelZexten => NXXNXXXXXX,1,Dial(H323/avaya/19${EXTEN})
03:42.33Kattywhatever way they're already used to will be the best
03:42.42Kattythe less the have to learn, the less frustrated they will be
03:42.58ChannelZI think this is just him
03:43.19Kattywell if it's just him he'll remember how he set it up
03:43.27Kattyinw hich case i'd just do plain ole 7 digit mapping and add the 19
03:43.30Kattyless work when dialing :P
03:43.51Slugs_which is this : exten => NXXNXXXXXX,1,Dial(H323/avaya/19${EXTEN})
03:43.56Slugs_?
03:44.06Kattythat's 10 digit
03:44.08ChannelZthat's 10 digit dialing yeah
03:44.18Kattywhat does 'N' mean?
03:44.19ruben23huhuh, this color cli-is making me sick..
03:44.20ChannelZwhether they require it or not you haven't said
03:44.21Kattyit's certain numbers
03:44.22Kattybut if orget
03:44.38ruben23<PROTECTED>
03:44.42Katty1.6.2
03:44.49VoIP-PenguinN should be 2-9
03:44.58Kattyi have also used 1.4.x and 1.2.x over the years
03:45.07Slugs_ChannelZ, i know when i make a call out i do, 195555555 from avaya
03:45.09ruben23ow, is the cli ok..? there is color hinting for agi, notice and error..?
03:45.10VoIP-PenguinZ is 1-9, and X is 0-9
03:45.35Kattyruben23: if your terminal supports color, it will give you color
03:45.41ChannelZSlugs_: so then NXXXXXXX
03:45.48Kattyruben23: disable color in the terminal and you should quit getting color
03:45.52Slugs_k
03:46.13Slugs_he just said N = 2-9 tho
03:46.15ChannelZruben23: wasn't it you who WANTED color earlier?
03:46.18Kattyif you dial a lot of numbers frequently, you can setup speed dials.
03:46.19Slugs_oh i see
03:46.20ruben23Katty: you mean terminal like ssh putty for windows.
03:46.31Kattyruben23: yeah whatever you're using
03:46.33VoIP-Penguinslugs_: You missed the _ in your above exten.
03:46.39ruben23ChannelZ yeah havent yet resolve it,
03:46.42Katty_ is for matching
03:46.43VoIP-Penguin(2243.51) <Slugs_> which is this : exten => NXXNXXXXXX,1,Dial(H323/avaya/19${EXTEN})
03:46.46VoIP-Penguinfail  ^^^
03:46.53Slugs_lol
03:46.53Slugs_k
03:47.07KattyVoIP-Penguin: he is new to this.
03:47.11ruben23Katty: putty do support color, but mine is not getting it..
03:47.16KattyVoIP-Penguin: failing happpens. repeatedly.
03:47.29KattyVoIP-Penguin: i still fail occasionally after years
03:47.31ChannelZruben23: echo $TERM
03:47.47ruben23xterm
03:47.50VoIP-PenguinIf it isn't pointed out, he won't realize it.  The fact that it was a fail is nothing to be ashamed of.
03:48.11KattyVoIP-Penguin: oh he'll get it
03:48.17KattyVoIP-Penguin: when he goes to test a number ;)
03:48.18VoIP-PenguinI often miss the underscore when trying to write examples.
03:48.29ruben23im getting color hinting for the OS itself using ubuntu-server, only with asterisk CLI- i dont have
03:48.31ChannelZruben23: ok.. do this, stop asterisk
03:48.44VoIP-Penguin...unplug the computer...
03:48.44Slugs_ok then this ---- exten => _NXXXXXXX,1,Dial(H323/avaya/19${EXTEN})
03:48.54Kattyyep
03:48.58ChannelZruben23: then run it manually (NOT the init script) with 'asterisk -c'
03:48.58Kattythat would send out a 7 digit number
03:49.04Kattyand add 19 to the beginning
03:49.08Slugs_omg im trying it ;)
03:49.10Slugs_one sec
03:49.10VoIP-PenguinThat will match 7 digits starting with 2-9.
03:49.12ruben23ok, run asterisk -c
03:49.38ChannelZIs it rainbow throwup now?
03:49.45VoIP-Penguinslugs_: What number do you want to dial on the phone?
03:49.50*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)
03:49.57Kattyi hope he dials pizzahut
03:50.00Kattyand orders me a pepperoni
03:50.02VoIP-Penguinhahaha
03:50.18KattySlugs_: you can set up ..kinda like speed dials, if you dial places frequently
03:50.28ruben23<PROTECTED>
03:50.28ChannelZmap 911 to pizza hut
03:50.39KattySlugs_: 42,1,Dial(H323/avaya/19pizzahut)
03:50.41ChannelZexcept that will kill you because you'll need to dial real 911 after all that grease
03:50.43VoIP-PenguinI'd rather set up speed dials on the phone.
03:51.05ruben23the color henting, so- when iused my asterisk i should run it by asterisk -c
03:51.13KattyVoIP-Penguin: then you do that.
03:51.23ChannelZ(henting?)
03:51.38Kattyruben have you been drinking?
03:51.52ChannelZruben23: well if you want to run it manually, yeah
03:52.09ChannelZruben23: if you want it to startup automatically, then you have to do so by other means.
03:52.31ChannelZruben23: asterisk won't output color, no matter what, if it's detached from the console and run in the background
03:52.31ruben23ChannelZ: why its not possible possible for asterisk -rvv...?
03:52.59ChannelZruben23: which is why I said you can use safe_asterisk.. safe_asterisk will detach so you can run it at startup, and then safe_asterisk it's self runs the 'real' asterisk with -c
03:53.11VoIP-Penguinitself
03:53.29VoIP-Penguin"it's self" isn't a sensical phrase.
03:53.31Kattyduring your compling process, you can run make config and it will setup startup scripts for you
03:53.38ChannelZNew nick, same old douchebag
03:53.53ruben23<PROTECTED>
03:54.01Kattyit's what i do
03:54.02Kattybut i'm lazy
03:54.16ChannelZKatty: the new startup scripts run asterisk, not safe_asterisk - so no color
03:54.26VoIP-PenguinBecause you are incapable of learning plain fucking English, I'm the douche bag.  Real genius, there.
03:54.27Kattydoesn't bother me a bit
03:54.33KattyVoIP-Penguin: that's enough.
03:54.44KattyVoIP-Penguin: be civil or take it elsewhere
03:54.51ruben23ChannelZ:  just one more, how do i make my asterisk init script run safe asterisk..
03:54.56VoIP-PenguinYou're my mother now?
03:55.31ChannelZruben23: change it? :)
03:56.04ruben23ChannelZ: yeah, but how would it be, like i replace my asterisk startup script
03:56.12ChannelZruben23: what distro are you running?
03:56.23ruben23im using ubuntu-server
03:56.28ChannelZok
03:56.35ChannelZhere I'll just post mine
03:57.07KattySlugs_: did it work? :>
03:57.25ruben23ChannelZ: thank you in advance.
03:57.35ChannelZhttp://pastebin.com/XcfLtCuE
03:57.39Kattyoh :<
03:57.40ChannelZuh oh
03:57.41Kattywell.
03:57.45ChannelZHe dialed a number and shut off the power
03:57.47Kattymaybe on the upside it worked, and he called an Ex.
03:57.50Kattyand uhhh
03:57.56Kattyyeah.
03:58.17*** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net)
03:58.19Katty:>
03:58.32KattyDID IT WORK!?
03:58.32Slugs_hehe one sec
03:58.32Slugs_:)
04:01.16ruben23ChannelZ: ill test this thank you again
04:01.43ChannelZsure
04:02.12ChannelZI should look at the source and see if it's easy enough to hack so it spits out color anyway if you run it forked
04:02.38ruben23http://pastebin.com/AjAKs2yY
04:03.17ChannelZps ax |grep asterisk
04:03.38Slugs_Katty, http://pastebin.com/nTwEEET0
04:04.03ChannelZSlugs_: did you 'dialplan reload' ?
04:04.12Slugs_i did 'reload'
04:04.14Slugs_whole thing
04:04.22ChannelZthats fine.  Then you have something off in your extensions.conf
04:04.22ruben23<PROTECTED>
04:04.38ruben23<PROTECTED>
04:04.40ChannelZruben23: ok so that means it didn't run correctly from the init script
04:04.46ruben23yeah
04:05.01ChannelZdid you get an error when you did /etc/init.d/asterisk start  ?
04:05.40ruben23http://pastebin.com/qr71pUWD
04:06.13ChannelZerr..
04:06.26Slugs_ChannelZ, this is the line added in extensions.conf
04:06.27Slugs_exten => _NXXXXXXX,1,Dial(H323/avaya/19${EXTEN})
04:07.21ChannelZok but the error you pasted, you dialed 10 digits
04:07.36ChannelZbut earlier you told us you normally just dial 7 digits
04:07.58Slugs_i said i dail 19 then 7 digits
04:08.06Slugs_im sorry
04:08.17Slugs_its 19 then 10 digits
04:08.17ChannelZCall from '1000' to extension '4045555555' rejected because extension not found.
04:08.24ChannelZThere you dialed 10... 404-555-5555
04:08.40ChannelZok so you want _NXXNXXXXXX instead
04:08.46Slugs_sorry thx
04:08.54ruben23<PROTECTED>
04:09.04*** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net)
04:09.15ChannelZruben23: I'm not sure what it's bitching about.. 'bad fd number8'
04:09.36ChannelZruben23: I'm looking at the script trying to find where that could be failing
04:10.21ruben23<PROTECTED>
04:10.29ChannelZhuh
04:10.35Slugs_.
04:12.24ChannelZruben23: hmm what version of ubuntu?
04:12.25ruben23<PROTECTED>
04:12.35ruben23ubuntu- 8.04 LTS
04:12.55ChannelZhmm same thing I'm running, not sure why it barfs for you
04:13.00ChannelZAre you root?
04:13.35ruben23yeahs.
04:13.36ChannelZor did you modify your old init script to run asterisk as a specific user?
04:13.38ChannelZhmm
04:13.47ruben23im using your script now
04:13.55ChannelZyeah
04:14.40ruben23but before im not- when i run   root# var/run/safe_asterisk ---- > i get this ---> bad fd number8'
04:14.42ruben23i gues
04:14.50ruben23ill try re installing it again
04:15.08ChannelZ/var/run/  ??
04:15.32ruben23sorry, it /usr/bin/safe_asterisk
04:15.42ChannelZah
04:15.55ChannelZhmm well I dunno why it's saying that
04:16.19ruben23ill let you know- after re installl and il use dyour script. thanks again
04:16.34ChannelZreinstall what
04:16.47ruben23i mean ill start form scratch again
04:16.50ChannelZWhat version of * by the way
04:16.59ruben231.4.27
04:17.19ChannelZscratches his head
04:17.29ruben23ow whats your version..?
04:17.55ChannelZ1.6.1.18 but I've run 1.4.x before with basically the same setup
04:18.26ruben23ow ok, you think your script is a conflict..?
04:18.41ChannelZshouldn't, I'm looking at 1.4's now
04:18.45ChannelZthey're pretty generic
04:19.02ChannelZin your case though it's actually the safe_asterisk script which is spitting out that error, I've never seen that before
04:19.44ruben23yeah, not the script maybe-bad source compile
04:20.10ChannelZI dunno.  Really depends on what actually is complaining
04:20.30ruben23thanks again bye..
04:20.33*** part/#asterisk ruben23 (~ITadmin@122.55.48.243)
04:20.39ChannelZeesh
04:20.54Slugs_ok ok
04:21.10Slugs_what do you think about this... http://pastebin.com/tDyuMUx9
04:21.42ChannelZ4045555555 can't be a valid phone number
04:21.51Slugs_im changeing it ;)
04:22.06Slugs_in pastebin
04:22.07ChannelZok
04:22.35ChannelZwell the call looks like it's going through but is coming back from avaya as busy/invalid
04:22.57Slugs_interesting
04:23.56ChannelZAre your real phones that are connected to this avaya pbx analog phones or voip phones?
04:24.02*** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net)
04:24.13Slugs_hold one sec, i think i got it
04:29.19ChannelZor not
04:29.45Slugs_ok well...
04:29.59Slugs_it's 91, then 7 didgits
04:30.05Slugs_im scrwing you up
04:30.35Slugs_exten => _NXXNXXXXXX,1,Dial(H323/avaya/91${EXTEN})
04:30.54Slugs_right?
04:31.40Kattyseven digits would be _nxxxxxxx
04:32.11Slugs_lol, im really fing you up
04:32.24Slugs_its 914045555555
04:32.31Slugs_thats what i need to dial
04:32.38Slugs_thats 10
04:32.45ChannelZCount much? :)
04:32.47Slugs_lol
04:32.52Slugs_tell me bout it
04:32.54ChannelZso yes what you pasted above
04:33.09ChannelZare you SURE the 1 is there?  (long distance?)
04:34.16ChannelZI only ask because it's kind of common for the steering digit 9 to reach an outside line.  But then you'd dial 10 digits like any other phone number, like if you were calling from your house phone.  The '1' would be additional if you were dialing an out-of-state area code
04:34.49Slugs_yep it works my friend
04:34.54Slugs_your the best
04:35.06Slugs_sorry for my tiredness
04:35.11ChannelZok just checking.. we don't know how your other PBX is setup
04:35.23Slugs_works perfect
04:35.40Slugs_ty so much
04:35.54*** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com)
04:35.55ChannelZhave fun
04:36.03Slugs_i will ;)
04:36.05Slugs_i am
04:36.05Slugs_;0
04:36.40ChannelZSo is this just playing or is there a practical use for doing this
04:36.50Slugs_practical
04:37.38Slugs_we have 5000 phones and the company does not want to pay for more extensions
04:37.48Slugs_so now i can add more ext's
04:37.58VoIP-Penguinextensions are free, phones, on the other hand, are not.
04:38.06ChannelZYou using softphones?
04:38.10Slugs_yes
04:38.16ChannelZah
04:38.25Slugs_avaya charges for everything
04:38.27Slugs_voicemail
04:38.31Slugs_you name it
04:39.14ChannelZYeah when I started my business my partner was looking at buying an old used phone system.. I told him NO WAY
04:39.32Slugs_lol
04:39.44ChannelZBuilt a computer myself for <$200, threw a TDM card in and built the thing myself.  Stupid paying for licenses for voicemail and crap
04:39.57Slugs_exactly
04:40.14Slugs_this company has paid well over 5mil for this system
04:40.22Slugs_5000 endpoints
04:40.24ChannelZI'm sure
04:41.41ChannelZDo you do IT for them, or is some computer nazi going to whip you for doing something they don't understand?
04:42.29Slugs_yeah im there linux admin, they thru me into pbx for some unknown reason
04:42.43Slugs_but im loving it
04:42.53ChannelZwell that's good at least
04:43.42Slugs_yeah, this has opened so my eyes to endless possibilites after messing with this for 2 weeks
04:43.52ChannelZhmm I think I found ruben's issue, too bad he left
04:44.01Slugs_poor ruben
04:44.35ChannelZhmm no Noteserv on this net
04:46.15Slugs_what part of the sip.conf shows up for callerid's on peoples cell?
04:46.27Slugs_callerid =
04:46.38Slugs_or the extension itself
04:47.34ChannelZwell that really depends on the phone system
04:47.46Slugs_makse sense
04:48.07ChannelZcallerid=Some Dude <5551212> in sip.conf (for a SIP device) associates CID for that device
04:48.28ChannelZwhether or not that makes it out into the public depends on your phone hardware/telco company
04:48.38Slugs_gotya
04:50.01ChannelZfor instance I just have 4 POTS lines, so outgoing caller ID is controlled by the phone company
04:50.39Slugs_yeah i just edited inside the <> and the caller id showed up as that
04:51.01Slugs_so i could call from 'what looks like' somebody elses number
04:51.08Slugs_how bout that..
04:51.12Slugs_interesting
04:51.56ChannelZIt's not impossible.  You're probably on some PRI for your service?
04:52.05ChannelZT1
04:52.08Slugs_yes
04:52.31Slugs_i called my wife from 1234567
04:52.34ChannelZSo it's not unlikely that you have control over your caller ID then
04:52.40Slugs_she was just like wtf
04:52.55Slugs_this is awesome
04:52.59ChannelZWith POTS lines or a lot of VoIP providers you can't
04:53.18Slugs_got ya
04:53.36ChannelZPRI is a bit more wild wild west :)
04:53.46Slugs_apparently so ;)
04:56.10*** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com)
05:01.36Kattyhi
05:03.11hardwirehi
05:03.54hardwirewow.. Slugs is picking that up quickly
05:04.05hardwirehis only problem was ooh323 is horribly broken
05:04.14hardwireasterisk-h323 (sigh) worked fine
05:08.15ChannelZso the 'ooh' part of 'ooh323' is really a verbal exclamation of disappointment
05:11.13hardwireno
05:11.20hardwireooh as in "ooh quit pinching me!"
05:11.33hardwireyou F***ing sh*jhing
05:11.36ChannelZheh
05:19.56otavioIt looks like I'm having trouble with 1.6.2 and format_mp3; it doesn't play mp3 files
05:20.02otaviogsm works fine
05:20.32otavioIt looks like it try to play a .slin file but the file was suppose to be mp3
05:22.33ChannelZis it showing up in 'core show file formats' ?
05:23.27otavioslin       mp3        mp3
05:23.34otaviodoes it make any sense?
05:24.41ChannelZwell yes in the sense that there is no support for mp3 native in the voip world.. so it's decompressing mp3 to convert to slin
05:25.55otavioOk; but it would then write it somewhere?
05:26.28ChannelZwell no ideally it's converting on the fly.. but I've never used mp3 in * (is that in asterisk-addons?)
05:26.37ChannelZIs this for MOH or you're trying to use them for prompts and such?
05:26.46*** join/#asterisk uqlev (~yuriy@91.184.221.31)
05:27.18otavioprompts
05:27.24*** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110)
05:28.19ChannelZSeems like a waste of CPU to do that but I guess it should still work.  But alas since I've not used it I can't say for sure
05:28.45ChannelZDo the files you're trying to play have unique names (IE they don't exist as *.wav or *.gsm etc in the same directory)
05:28.59otavioChannelZ: I used to use it with 1.4
05:30.05*** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire)
05:30.17otavioChannelZ: only the .mp3
05:30.45*** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire)
05:30.49ChannelZand it just tells you in the console the file isn't found or somethin?
05:32.31Kattyso...
05:32.34Kattywhat's happenin and stuff
05:33.40ChannelZa Katty hit & run
05:33.49*** join/#asterisk Katty (~User@adsl-68-91-212-133.dsl.stlsmo.swbell.net)
05:33.49Kattyrehi
05:34.30ChannelZrewaves
05:36.29otavioPlaying 'custom/office-hours.slin'
05:36.32otavioThis only
05:36.46otavioLooks like it has been played but it was not
05:36.58Kattyorly
05:37.00ChannelZhmm
05:37.19ChannelZthis is 1.6.x?
05:39.30otavio1.6.2.2
05:40.00ChannelZand it shows up in 'core show translation' too?
05:43.53otaviodunno; let me look
05:44.18otaviodoes not
05:44.19ChannelZit might not since
05:44.28ChannelZsince it's really outputting 'slin'
05:44.36ChannelZbuilds asterisk-addons for the hell of it
05:44.45hardwireotavio: I have a hint for you sir
05:44.56hardwire1.) format_mp3 will decode to slin
05:44.58otaviohardwire: I'd love to know it
05:45.03hardwire2.) you have slin files already available
05:45.09otaviohardwire: where?
05:45.12hardwire3.) it makes perfect since to use the .slin files
05:45.18*** join/#asterisk spartan07 (~spartan07@ip72-211-193-180.oc.oc.cox.net)
05:45.34hardwirenow.. does the .slin file exist or not?
05:45.37hardwireif it does.. then yeh
05:45.40ChannelZI think there's just something about his mp3s it doesn't like if he's getting silence but no errors
05:45.48hardwireif it doesn't .. then thats just how the debugging is presenting it
05:45.49otaviohardwire: no; it looks it does on the fly
05:45.55hardwireit probably is using your mp3 file sir
05:46.20ChannelZlooks like it's working for me
05:46.30ChannelZbut it's very quiet...
05:46.36*** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net)
05:48.04ChannelZit's probably working for you but you can't hear it.  strange
05:48.12hardwireheh
05:48.44ChannelZI need some non-music MP3s but they're playing back verrrry softly
05:49.14otavioconverting to gsm made them work
05:49.19otaviobut this is insane to me
05:49.50ChannelZwell as I said it's sort of insane to be using mp3s as prompts because it's a waste of CPU
05:50.04ChannelZthough if there's not a lot of traffic on the server it probably doesn't matter but still
05:50.40ChannelZhttp://lists.digium.com/pipermail/asterisk-dev/2009-May/038193.html
05:50.56Kattymaybe we should just go to Denny's instead.
05:51.01Kattyor waffle house
05:51.11hardwireotavio: are you sure format_mp3 can grok the mp3s you made?
05:51.57coppiceKatty: "waffle house" == "congress"?
05:52.09KattyOH I SEEWHATYOUDID THERE
05:52.28otaviohardwire: it used to do that when I was using 1.4
05:52.34hardwirebummer
05:52.40otaviohardwire: this looks like a regression for me
05:52.52Kattycoppice: les go get some hashbrownns.
05:53.05ChannelZThe phone number for the capitol is 1-877-SOB-U-SOB
05:55.37Katty18009687825
05:55.45*** join/#asterisk spartan07 (~spartan07@ip72-211-193-180.oc.oc.cox.net)
05:56.02Katty(1800yousuck)
05:57.00ChannelZif I change the value of OUTSCALE in format_mp3.c and rebuild it works a bit better
05:58.21hardwireKatty: agreed
05:58.49Kattyhardwire: what are we agreeing on?
05:58.51otavioA last question ..
05:58.57Kattyhardwire: are we agreeing to agree?
05:59.01ChannelZI want a milkshake
05:59.05otavioI'd like to use a different moh due language
05:59.09otaviois that possible?
05:59.11Kattymilkshake sounds good.
05:59.18Kattywe could do steak n shake. they make a killer grilled cheese
05:59.22hardwireKatty: hash browns
05:59.32Kattyhardwire: oh. right.
05:59.43hardwireor rather.. hashed browns
05:59.57Kattythat would also work
06:00.01ChannelZotavio: you can make a separate MOH 'context'
06:00.33otavioChannelZ: but for it; it's need to separate the queues
06:00.55otavioChannelZ: like q-lang-1; q-lang-2
06:01.11otavioChannelZ: in same context it doesn't look for language?
06:01.27ChannelZsorry context is not the right word.. I guess MOH is called 'class'
06:01.52ChannelZyou set the musicclass for your queues accordingly
06:01.56Kattyi have two classes on my server.
06:02.14ChannelZset the moh classes to feed off of different folders of files.  bagpipe music in one, techno in another, whatever
06:02.32Kattyi recommend late night alumni
06:02.48otavioChannelZ: the question is ... I can't use those depending on the language right?
06:03.07Kattyif there's a way to put the language into a variable you can
06:03.30ChannelZwell I don't understand your setup.. are you already putting people into specific queus based on their language (however you are determining that)?
06:03.33Kattyor perhaps setting the language at the beginning of that portion of the dialplan
06:03.46Kattypress 1 for english, 2 for spanish..i mean in that situation you could
06:04.08Kattyif they hit 1, set the moh class, then goto(onward)
06:04.41otaviohumm
06:05.17Kattywhat is the opposite of wire
06:05.26ChannelZstring
06:05.32KattyTHEORY
06:05.39ChannelZwireless
06:05.59Kattyi do not have a catchy comeback for that one
06:06.00Kattyyou win.
06:06.10ChannelZclaps
06:06.55carrarThats a fun game
06:07.13ChannelZfree word association
06:07.28ChannelZCACTUS
06:07.58seanjohnroot@mail named]# cd /etc/udev && grep -R urandom
06:08.25ChannelZwow
06:08.34ChannelZtell me more about your childhood
06:08.55ChannelZas that is not the response I'd expect to 'cactus'
06:09.29carrarMake sure to update that named server
06:09.57Kattyokay let's try something easier like...
06:10.04KattyCURLY FRIES
06:10.09ChannelZwant
06:10.19Kattymore
06:10.22ChannelZyes
06:10.25Kattynow
06:10.28ChannelZketchup
06:10.33Kattypepper
06:10.44ChannelZbelch
06:10.44carrarPOOP
06:10.48Kattybrown
06:10.54ChannelZstank
06:10.59carrarno POOP was in reponse to your fries
06:11.00Kattywhore.
06:11.02carrarheh
06:11.08Kattythis is word association
06:11.16Kattyit doesn't have to neccesarily apply to the previous ongoing topic :P
06:11.19Kattyjust the previous word
06:11.40ChannelZbetcha THIS doesn't happen in #freepbx
06:11.47Kattytrixbox
06:11.56carrarlets play associative array
06:11.59ChannelZwhore again
06:12.16Kattyhorde
06:12.20ChannelZroundcube
06:12.25Kattyrubixcube
06:12.29ChannelZbust
06:12.35Kattyuhhh
06:12.41Kattymaybe i outta keep it pg
06:12.42ChannelZnot like boobs but break
06:12.51KattySHATTER
06:12.54ChannelZcuz I'd just take my rubiks cubes apart
06:13.03ChannelZShatner
06:13.06Kattywilliam
06:13.16ChannelZtell
06:13.20Kattyoverature
06:13.31ChannelZhmm
06:13.41ChannelZviolin
06:13.45Kattyviola
06:14.03ChannelZvoila!
06:14.11Kattydinner
06:14.21ChannelZate
06:14.29Kattyeight
06:14.34ChannelZenough
06:14.46Kattysufficient
06:15.11ChannelZhmm
06:15.38ChannelZI guess I have no feeling towards 'sufficient'
06:15.44Kattyk
06:15.46Kattylet's try a new one
06:15.47ChannelZheh
06:15.47Kattywhiskey
06:15.49ChannelZblech
06:15.55Kattytequilla
06:16.00ChannelZdrunkard
06:16.09Kattywhiskey
06:16.13ChannelZrepet
06:16.16ChannelZerr repeat
06:16.19Kattyparrot
06:16.25ChannelZmonty
06:16.32Kattypythonnnnnnnnnn
06:16.36ChannelZ:)
06:16.42ChannelZexpired
06:17.53*** join/#asterisk tecnico (~tecnico@75.76.169.148)
06:18.00ChannelZviking
06:18.00Kattyturkey
06:18.05Kattyoh
06:18.06Kattyhorns
06:18.10ChannelZhonk
06:18.19Kattygoose
06:18.25ChannelZscream
06:18.30KattyLOL
06:18.33Kattyare you afraid of geese
06:18.40Kattythat's hilarious
06:18.41ChannelZhehe no
06:18.46KattyICESCREAM
06:18.59ChannelZmint
06:19.11Kattydairy queen thin mint cookie blizzard
06:19.20ChannelZmmmmmmm
06:19.36ChannelZtoo bad there's not one near me
06:19.51Kattyiraq
06:20.20ChannelZdirt
06:20.30Kattyworms
06:20.35ChannelZwet
06:20.51Kattyuhh
06:20.53Kattyrain. yes.
06:21.13ChannelZmaybe you don't want any more insight into my brain
06:21.18Katty:P
06:21.25Kattyseems pretty normal
06:22.03ChannelZI do want some ice cream now though.
06:22.36ChannelZhmm.. 7-11, Ben & Jerry's..
06:24.32Kattylol
06:24.37Kattypick me up a cream soda.
06:25.46ChannelZI need to go to the grocery store really
06:26.27Kattyme too
06:26.30Kattyget some real food.
06:26.36Kattyrather than chips and salsa and ramen
06:26.58ChannelZhehe
06:27.46ChannelZso if you're specifying host=xxx in iax.conf you don't need to register on the other end..?
06:33.18hardwireKatty: and hashed browns
06:33.37hardwireKatty: chips and salsa and ramen is how I stay a lean 280
06:33.38hardwire:P
06:36.04Kattypfffft
06:36.37hardwirepfffft
06:36.39hardwireexactly
06:38.23*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
06:40.00TJNIIpfffft
06:40.09TJNIIis drunk
06:40.28antiwiresir you are too drive to drunk.
06:40.35TJNIIIndeed
06:40.44TJNIIThat's why I stumble
06:40.49TJNIII maean walk'
06:40.53antiwireoccipher, I can't hear you.
06:44.55TJNIII hear rthey really hate ie when you call the ocipher.
06:45.09ChannelZor pig
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07:55.31khussein78when i receive call on ring group then one of this group transfer the call to me, CID which i got is the ring group ID, is that right
07:55.50khussein78or i should see the extension as CID
07:58.50*** join/#asterisk hipitihop (~denis@CPE-58-161-241-124.mzmy1.cha.bigpond.net.au)
07:59.02ChannelZI assume you mean the CID is of the phone that transferred, not of the call they transferred (the ring group number shouldn't be showing up)
07:59.56ChannelZThere is a 'blind transfer' and an 'attended transfer'.  In an attended transfer, the CID of the call coming into the transferred-to phone will show up as the transferred-from phone, because they are really the ones calling you.
08:00.35ChannelZIn a blind transfer, the CID of the call coming into the transferred-to phone will show up as the CID of the call being transferred as it came into the transferred-from phone
08:16.16*** join/#asterisk viq (~viq@unaffiliated/viq)
08:31.05khussein78ChannelZ, where can i choose between blind and attended transfer
08:31.53*** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire)
08:46.09ChannelZkhussein78: depends on your phone
08:48.10khussein78ChannelZ, OK thank you
08:48.15ChannelZif you're using analog phones and feature codes, it's configured in /etc/asterisk/features.conf - if you're using SIP phones that's dependent on the phone's config..
08:49.12ChannelZwhether it's a toggle, or two discreet buttons, or whatever.  Linksys SPAs for example use function softkeys, and they put axfr on the first screen and make you scroll over to get to bxfer (SO ANNOYING!)
09:08.12*** join/#asterisk sjobeck (~Adium@67.136.135.134)
09:09.57khussein78ChannelZ, i have SIP phone, linksys PAP2, and soft phones so i think i should leave the configuration as it is
09:13.23ChannelZthere's nothing to configure on the asterisk side really, it's all the function of the various phones
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10:48.20Dovidanyone know if Polycom supports STUN /
10:52.55*** part/#asterisk pentanol (~pentanol@77-35-13-094.pppoe.primorye.net.ru)
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11:59.12jblackDovid: I don't know of all their products, but I'm not aware of that capability, no.
11:59.24jblackhowever, it may have sip over tcp.
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12:11.17Dovideh.
12:11.32Dovidissue with polycom is that it sends it's local IP for RTP
12:12.12sawgoodDovid: sounds like a NAT concern
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12:16.42Dovidsawgood: why ? shouldnt the phone be smart enough to convert it over ?
12:16.52Dovidthe router shouldnt "have to" do it
12:17.07sawgoodno ... the Asterisk is the concern
12:17.14sawgoodhas this phone ever worked before?
12:20.24*** join/#asterisk sourcode (~code@ppp-58-8-236-98.revip2.asianet.co.th)
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12:38.52*** join/#asterisk Bondd (~Bond@dsl-202-45-105-166-static.QLD.netspace.net.au)
12:39.00BonddHey guys
12:39.17Bondd:( i have just spend 12 hours setting up an office and asterisk etc etc
12:39.27Bonddand im configuring the server
12:39.33Bonddand the zaptel stuff
12:39.36Bonddand i get this error.
12:39.42Bonddhttp://pastebin.org/119256
12:40.15Bonddanyone here able to lend a hand?
12:41.18Bonddim running debian
12:41.25Bonddits a fresh net install
12:41.33Bonddand i have a brand new digium tdm400 card
12:41.42Bonddwith 1 FXO in it
12:52.30ChainsawBondd: Absolutely useless without context.
12:53.27Bonddumm okay, sorry im sorta not that flahs with this
12:53.33Bonddwhat context do you mean?
12:53.51Bonddpabx:/usr/src/modules/zaptel# make install
12:53.54Bonddthats what i typed
12:53.58Bonddto get that error..
12:54.01Bondddoes that help?
12:56.43BonddChainsaw: im installing asterisk on a server and runing a make install in that DIR
12:57.22ChainsawBondd: I couldn't see what directory, I couldn't see what you typed to provoke the error, I couldn't see what you did before make install.
12:57.30ChainsawBondd: By removing all the context, you made the paste unusable.
12:57.43Bonddone sec
12:58.30Bonddhttp://pastebin.org/119268
12:58.38Bonddokay i added more lines of stuff
13:00.08Bondddoes that help?
13:00.47jayteedid  you do a ./configure and then a make without the install option first?
13:01.07Bonddyes jaytee i did
13:01.14Bonddsame error with the same
13:01.16Bonddmake*
13:01.58Bonddif i type make i get the same error
13:02.09Bonddand i did type ./configure
13:02.50*** part/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)
13:07.37Bonddis it worth trying asterisk now?
13:07.47Bondddes asterisk now come with the digium stuff in it?
13:09.37khussein78Bondd, did you tried to use elastix ?
13:19.05*** join/#asterisk admin0 (~admin0@cm73.delta128.maxonline.com.sg)
13:22.28admin0hi giys
13:22.32admin0hi all
13:23.12admin0this is my setting: http://asterisk.pastebin.com/ziSKJKG9   of sip.conf, extensions.conf and core show transalation .. i am using 2 PAP2 devices ..  when I dial, when the other party picks up and say hello, i get that initial hello and after that, there is no sound at all
13:23.21admin0please let me know how to troubleshoot this issue
13:23.26admin0both sides are using linksys pap2
13:23.32admin0all settings in http://asterisk.pastebin.com/ziSKJKG9
13:27.14admin0additionally , my server is in public ip,  both pap2 are behind nat
13:29.10jaytee~sipnat
13:29.11infobot[~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
13:40.58*** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:f9e4:43ec:892b:7e5a)
13:43.50admin0thanks guys .. i added qualify=yes in both and restarted, but still the same
13:45.09*** join/#asterisk Godfather_ (~Godfather@157.Red-88-11-88.dynamicIP.rima-tde.net)
13:46.25admin0is there a command to show codec negotations being used ?
13:48.15*** join/#asterisk neurosys (~neurosys@c-98-254-241-158.hsd1.fl.comcast.net)
13:50.19*** join/#asterisk quintana (~sylvain@aghnar.doowan.net)
13:58.57DennisGadmin0 try sip show channels =)
13:59.34DennisGor show channels verbose (but i think that you need the previous command)
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14:19.33*** join/#asterisk digilink (~digilink@tn-76-5-159-171.sta.embarqhsd.net)
14:24.59Dovidthis time of year again....... Can anyone help me with defunct agi's with php
14:28.10*** join/#asterisk tecnico (~tecnico@75.76.169.148)
14:33.18*** join/#asterisk voipmonk (~shido6@dsl-69-172-110-65.acanac.net)
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14:39.35*** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net)
14:40.31Slugs_hi
17:21.16*** join/#asterisk infobot (ibot@rikers.org)
17:21.16*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs
17:21.24VoIP-Penguinexten => 2000,1,Playback(somesoundfile)
17:21.29Slugs_ive got answer, playback and hangup
17:21.46Slugs_im just confused where the 2000 would come into play
17:21.56VoIP-PenguinPlayback() answers the channel, so you could technically skip an explicit Answer() in your plan.
17:22.19Slugs_good tip
17:22.45VoIP-Penguinexten => 2000,1,Playback(somesoundfile)  should be enough, actually.  When the sound is done playing, the channel will exit.
17:22.46Slugs_exten => s,n,Playback(you-will-be-transfered-menu)
17:23.04Slugs_so what does the s mean?
17:23.08Slugs_im mine
17:23.11VoIP-Penguinthe 's' extension
17:23.25VoIP-PenguinYou don't dial s from a phone.
17:23.43Slugs_ahh i think i see
17:24.01KattySlugs_: s is "startextension in context"
17:24.18VoIP-PenguinI've shown you the exact syntax twice already.
17:24.34KattySlugs_: it's used primarily for dialplans which you enter a context with no other extension information
17:24.43*** join/#asterisk sjobeck (~Adium@67.136.135.134)
17:24.44KattySlugs_: so it just /starts/
17:24.57*** join/#asterisk StarFyter (~ken@ip68-97-4-47.ok.ok.cox.net)
17:24.59KattySlugs_: there are other standard extensions like h.
17:24.59Slugs_VoIP-Penguin, and i appreciate, i dont really care about it working as much as the 'why' it works
17:25.00*** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2)
17:25.03VoIP-Penguin~s
17:25.04infobotWhen used with FXO ports (analog or T-1), "s" means "device too stupid to send number"
17:25.07KattySlugs_: h will do something if the person hangs up on the IVR
17:25.34KattySlugs_: and i is used for invalid. so if the user enters 9 and nothing on the IVR is for 9, it will go to exten => i,1,doseomthing
17:25.49KattySlugs_: usually plays somethign like "that was an invalid extension" then goes to s,1
17:25.57Slugs_right
17:26.01KattySlugs_: h is usually for cleaning up a call
17:26.17KattySlugs_: if the caller hangs up, it'll go to exten => h,1,whatever... usually hangup()
17:26.30Slugs_right
17:26.34KattySlugs_: but you could execute System() and eject the cdrom driver everytime someone hangs up, if you wanted to
17:26.42Slugs_lol
17:26.44Kattyor dial pizza hut
17:26.59Kattyor copy a call file into the spool folder, and execute a whole other section of dialplan
17:27.02VoIP-Penguinexten => 2000,1,Playback(somesoundfile)   creates extension 2000, which the first priority in the dialplan progression is supposed to run the Playback() command and somesoundfile is the name of the sound file.
17:27.26KattySlugs_: and if you have figured it out yet, Playback() and Background() are different
17:27.37KattySlugs_: Background() allows them to enter digits, playback doesn't
17:27.47VoIP-PenguinIf there is no next priority, the dialplan exits after somesoundfile is done playing.
17:27.53KattySlugs_: wait() and waitexten() are also different.
17:28.16KattySlugs_: wait is just that. a pause. waitexten is obviously waiting for something to be entered.
17:28.45*** join/#asterisk darkavanger (~Darkavang@41.225.98.39)
17:28.49Slugs_well im on page 155 of 604 of the asterisk 1.4 book, hopefully they get into that..
17:28.58Slugs_right
17:29.08Kattythere's also fax
17:29.16Katty[from-pstn]
17:29.23Kattyexten => fax,1,(dosomething)
17:29.42Kattyfax detection isn't all that great yet.
17:29.45darkavangerhi i ve prolem with inbound from did
17:29.53Slugs_got ya
17:29.55darkavangerthe communication cuts after 20seconds exactly
17:30.03Slugs_ty both
17:30.03Kattydarkavanger: check your timeouts
17:30.12VoIP-PenguinThat requires the other side of the call to send a call to the 'fax' extension.  I can't imagine that happens very often.
17:30.14darkavangerwhere can find that ???
17:30.19darkavangeri am a newby
17:30.22Kattydarkavanger: in your dialplan
17:30.24*** join/#asterisk LemensTS (~LemensTS@71.86.32.146)
17:30.32LemensTSshould my logs rotate when i do logger reload
17:30.39cuscoI have another question... telco sent us an email stating that our PRI chans have "Number temporary confidential (Always visible unless SETUP message states otherwise)"
17:30.43KattyLemensTS: no
17:30.44VoIP-Penguindarkavanger: Answer() the channel.
17:30.52KattyLemensTS: logger reload will recreate the files if you delete them tho
17:31.17cuscoso by this "SETUP message" I thought they meant SetCallerPres(prohib) would do
17:31.28darkavangerit answers i got the calls on a queue with timeout >>>> 20
17:31.29cuscobut it doesn't seem to suffice
17:31.32darkavangerbut it cuts
17:31.36LemensTSKatty: what controls them to rotate then? in /etc/logrotate.d/asterisk it says logger reload not logger rotate
17:31.56KattyLemensTS: i think what you're looking for is logger rotate
17:32.15VoIP-Penguindarkavanger: Show me something that contains the problem.
17:32.23cuscoeven tho it looks it does the same.. not sue (logger rotate/reload)
17:32.31LemensTSKatty: yea logger rotate works, but in /etc/logrotate.d/asterisk it says /usr/sbin/asterisk -rx 'logger reload'
17:32.31KattyLemensTS: optionally /usr/sbin/asterisk -rx 'logger rotate'
17:32.40Kattyah.
17:32.41Kattyfix it
17:32.44cusco:)
17:32.59LemensTSKatty: I was wondering if the rotate was setup somewhere else
17:33.18Kattynot sure, to be honest.
17:33.21darkavangerVoIP-Penguin: one minute
17:33.28Kattyi would just make a cronjob out of it, if it needs to be a regular thing
17:33.35cuscowould anybody know if Set(CALLERPRES()=prohib) is what I need?
17:33.43cuscowhat do they mea by SETUP message?
17:34.13Kattycusco: i'm not sure i follow
17:34.47Kattycusco: they could mean the outgoing callerid you're sending to them has to be set, or else it will show as confidental
17:35.07darkavangerVoIP-Penguin: http://pastebin.com/raw.php?i=BgENXYQW here is a part of the log where it hangs up
17:35.13StarFyterDennisG you around?
17:35.24Kattycusco: telcos are crazy folk.
17:35.34cuscoKatty: they had that option, we chosen to always show our number unless we wanted otherwise....
17:35.37cusco:(
17:35.44VoIP-Penguindarkavanger: You did not Answer() the channel before running Queue().
17:35.44Kattythat's a good choice.
17:35.58Kattycusco: we also send out the same number everytiem we dial
17:36.17Kattycusco: but you could have people calling that want to send out their did number as their main number too
17:36.24Kattycusco: or if your 'office' is renting other offices...
17:36.27cuscoI tried Set(CALLERID(num)=some-number); and it works (as long as the number is inside our range)
17:36.29darkavangerVoIP-Penguin: i ve received the queue's musiqu on my phone !
17:36.35VoIP-Penguindarkavanger: You did not Answer() the channel before running Queue().
17:36.35Kattycusco: yeah
17:36.41Kattycusco: some telcos well let you send out custom callerid
17:36.47Kattycusco: obviously pots lines won't let you do that
17:37.04StarFyterHey Dennis.  Could I beg your assistance with this SPA3102?
17:37.07Kattycusco: so if you take an incoming call,a nd forward it to a mobile--you can see who is actuallyc alling you..and not some number within your pbx's range
17:37.11darkavangerVoIP-Penguin: [Mar 21 18:16:00] VERBOSE[3424] logger.c:     -- Executing [100@ext-queues:2] Answer("SIP/from-didww10-04786ab0", "") in new stack
17:37.19cuscowell thats ok, but I can't figure how to send a call out without a callerid
17:37.32Kattycusco: not many telcos or sip providors well let you 'fake' the callerid
17:37.40cuscoI cannot "fake"
17:37.47cuscowe have 200 numbers from this telco we can use
17:37.53cuscoI can use one of those 200
17:37.57*** part/#asterisk LemensTS (~LemensTS@71.86.32.146)
17:37.58VoIP-Penguindarkavanger: Show me the dialplan.
17:37.58Kattynods
17:38.08cuscoerlse it goes via default number
17:38.17Kattycusco: i'm sure they have it setup where if you set your caller id number to /empty/ they just stick your main number on it too
17:38.24darkavangerVoIP-Penguin: what is a dialplan :-[
17:38.28cuscobut they mention the SETUP message...
17:38.32VoIP-Penguindarkavanger: extensions.conf
17:38.33Kattyyeah dunno.
17:38.37darkavangerah ok
17:38.40Kattycall their happy tails back and ask them what they're talking about
17:38.55cuscohah.. I guess I will have too...
17:39.09Kattyif you can get your brain around what they're wanting, and describe it to us, we'll have a better shot
17:39.21StarFyterInbounds roll thru to the Asterisk system.  Cannot make any outbound calls.
17:39.43KattyStarFyter: check the context your phones are in against the dialpan
17:39.48cuscoreading http://www.voip-info.org/wiki/view/CallerID has a .Note on the beginning
17:39.55cuscothat mentions initial SETUP message
17:40.13Kattylooks
17:40.20StarFyterHmmm.
17:40.52StarFyterGhostrider permission for a flyby!!
17:40.58cuscoit mentions it for other purposes... but I would like to know what is the SETUp message
17:41.10Kattycusco: that just looks like they're talking about setting the outbound callerid before the dial()
17:41.18darkavangerVoIP-Penguin: http://pastebin.com/raw.php?i=PjBcQRwR here is my "dialplan" :D
17:41.22cuscoyea, its some other purpose
17:41.35cuscobut I guess its the same SETUP message they mentioning there
17:41.46Kattycusco: well maybe someone else around here will have some idea about it
17:41.55cuscoI was hoping so...
17:42.02darkavangerVoIP-Penguin: context i am using to receive calls is from-trunk
17:42.17cuscoperhaps [TK]D-Fender, who seems to be a know-it-all...
17:42.19VoIP-Penguindarkavanger: FreePBX?  We typically can't support FreePBX here.
17:42.29StarFyterWhere might the context be?
17:42.30[TK]D-Fenderdarkavanger: FreePBX is NOT supported here.
17:42.37darkavangerok thanks
17:42.42[TK]D-Fenderdarkavanger: #freepbx <-
17:42.44StarFyter<-- Gross N00b!!!
17:42.47KattyStarFyter: where did you set it up?
17:42.54cuscoor not...
17:43.10StarFyterI'm looking at extensions and I see nothing about context.
17:43.16VoIP-Penguindarkavanger: There's no a single Queue() app in your entire file that you pasted for me.  THAT is why we can't support FreePBX.
17:43.21[TK]D-FenderStarFyter: [thisisacontext]
17:43.21StarFyterVia web gui.
17:43.24KattyStarFyter: context is [inhere]
17:43.34Kattyah, i see.
17:43.44[TK]D-FenderStarFyter: And being a GUI user, you are also in the wrong channel
17:43.45darkavangerVoIP-Penguin: thanks anyway
17:43.48StarFyterlemme check extensions.conf
17:44.00[TK]D-FenderStarFyter: No GUI's are supported in this channel. They each have their own
17:44.06KattyStarFyter: if you're using a web gui, i don't think i'll be able to help you
17:44.14KattyStarFyter: i don't use that sort of stuff. not familiar with it
17:45.01Katty[TK]D-Fender: do you have an ipod?
17:45.17StarFyterHmmm.  [from-sip-external] ?
17:45.32Kattythat is a context.
17:45.36cuscoKatty: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg13111.html they also mention what I need there: "You could set "Presentation Indicator" in setup message."
17:45.39Kattywhether or not that's the one you need is another story.
17:46.00StarFyteryeah.  They throw a lot into it when the gui makes it.
17:46.14KattyStarFyter: that's why it's so hard to troubleshoot
17:46.19KattyStarFyter: it's a mess.
17:46.43cuscoonce again Presentation Indicator seems to be what CallingPres does
17:46.59Slugs_StarFyter, thats y i removed freepbx and burned it.
17:47.11StarFyter:)
17:47.26cuscoow.. and it states I need usecallingpres=yes in zapata.. lets see if dahdi takes such option
17:47.32Kattya web gui can be nice for people who don't know anything about their phone system and just want to change the name on the phone or something
17:47.54Kattycusco: yes, it should. dahdi /is/ zaptel but with a different name due to licensing or some stuff
17:48.14StarFyterdebates the remove and burn...
17:48.24Kattycusco: not sure of the real reason they had to change the name
17:48.27Slugs_i think freepbx just hurts people, with no customization, and if something does break, gl w/ that
17:48.30cuscokk
17:48.48[TK]D-FenderKatty: no, I don't.  I don't like the cost or restrictions of Apple devices
17:48.50KattySlugs_: well you certainly can't eject the cdrom drive with it
17:48.53smooth_penguindahdi is an odd choice of a name
17:48.58Slugs_lol Katty,
17:49.01Katty[TK]D-Fender: i should go ask brian.
17:49.11Katty[TK]D-Fender: he's the local Apple Enthusiast
17:49.44[TK]D-Fendersmooth_penguin: Digium couldn't let Fonatily get all the gayest names ;)
17:49.55StarFyterNot sure how far I could get powering by pico.  I have watched the asterikast videos, but they are kind of old now.
17:50.04smooth_penguinheh
17:50.53cuscoah!! Worked !!!
17:51.03cuscohow cool!
17:51.22cuscolet me try our other telco.. I doubt it but.. heh
17:51.49Kattyhugs smooth_penguin
17:52.20smooth_penguingives Katty an icicle
17:52.23Katty:<
17:52.30Kattydiamond life, lover boy
17:52.40Kattyhe moved in space with minimum waste and maximum joy
17:53.27Kattyno need to ask he's a smooth penguin
17:53.31Kattysmooothhh penguinnnn
17:53.44smooth_penguinwaddles around the channel
17:53.53Kattycoast to coast, la to chicago, uhhh not western male
17:53.57Kattymiddle eastern male?
17:54.14smooth_penguinlol
17:54.48Kattycairo to baghdad!
17:55.20smooth_penguinbaghdad is too dangerous
17:55.29Kattyk...uhh
17:55.34Kattycairo to Amman
17:55.58smooth_penguincairo to tokyo would be fine too :p
17:56.24Slugs_do you always need the '_' in front of the extension?
17:56.55smooth_penguinonly if you are trying to regex match a exten
17:57.04Slugs_got it
17:57.20*** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net)
17:58.11Kattysmooth_penguin: i've always wanted to go to Mongolia
17:58.26smooth_penguinwhy mongolia
17:58.32Slugs_[playsound]
17:58.32Slugs_exten => 2000,1,Playback(you-will-be-transfered-menu)
17:58.48Kattysmooth_penguin: saw a pretty picture of it once
17:58.52Slugs_if i reload dialplan, y would it say extension not founf
17:59.10Kattyprobably because your phone isn't in the [playsound] context
17:59.16smooth_penguinId want to go to Madagascar
17:59.23VoIP-Penguinslugs_: Was there any reason you created a new context for the extension?
17:59.23KattyI LIKE TO MOVE IT MOVE IT
17:59.28smooth_penguinhahaha
17:59.55smooth_penguinwell I love green, and the wildlife variety is just insane
18:00.00VoIP-Penguinslugs_: You can't use an extension unless its context is somehow accessible by your device.
18:00.01Kattyhttp://www.youtube.com/watch?v=0x3W6hutEj8 <- Madagascar
18:00.12Slugs_VoIP-Penguin, just thought i would keep it separate, for appantly no reason ;)
18:00.29KattySlugs_: you could put exten => 2000,1,Goto(playsound,s,1)
18:00.37KattySlugs_: but that's kind of extra work
18:00.53VoIP-Penguinor  "include => playsound"  in your other context
18:01.06Slugs_ahh
18:01.16KattySlugs_: there's a number of was to do it. you can keep it seperate if you want
18:01.17Slugs_ok, thats wonderful
18:01.28Slugs_perfect thx
18:03.56Kattyhrmm.
18:04.02Kattyryan has been upstairs for awhile
18:04.08Kattygoes to see why it's so quiet
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18:30.55Kattyi need some new workout tunes
18:31.01Kattysuggestions, anyone?
18:31.48eppigyLady Gaga - Monster
18:32.16Kattyooh
18:32.18Kattygets
18:32.40eppigyYoung Jeezy - Grindin' Feat Lil Boosie & Rich
18:32.44eppigyif you have good headphones
18:33.39Kattyis it thumpy
18:34.18Kattyeppigy: can you link it from youtube? i'll dirpy it
18:36.00eppigyhttp://www.youtube.com/watch?v=eEzdSmykifs
18:36.04eppigyyes low frequency bass
18:36.37eppigyDeftones - 7 words
18:36.51eppigythat is basicaly the good workout music spectrum
18:37.08eppigypop/electric
18:37.16eppigygangster rap
18:37.19eppigyhard rock
18:39.29Kattyk, i added all those
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18:51.10ChannelZLady Caca?
18:51.12ChannelZvomits
18:51.21ChannelZGuess that's one way to lose weight
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18:54.25Kattywell i'mma go get some lunch
18:54.26Kattyafks
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18:56.21ChannelZmmm lunch
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19:06.07Slugs_.
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19:13.16ChannelZ!
19:13.23Slugs_.!
19:13.44ChannelZ.!..
19:13.54Slugs_;0
19:14.08ChannelZ@#*?!
19:14.19Slugs_that's not polite
19:14.21Slugs_;)
19:15.05Slugs_s/@#*/wtf
19:15.16Slugs_~s/@#*/wtf
19:18.13Slugs_ChannelZ, I'm not understanding context's
19:18.44ChannelZThink of them as little boxes to separate extensions into
19:18.54Slugs_how do i get a phone to accept a diff [context]
19:19.10ChannelZAnd the extensions can't reach other extensions in other contexts unless you specifically allow them to
19:19.50Slugs_and whrer do i allow that, in the dail plan or in the phone configuration .conf
19:19.52ChannelZI don't understand the question - a phone doesn't really know anything about contexts
19:20.06Slugs_ok...
19:20.48ChannelZA device, as configured in sip.conf or iax.conf or a DAHDI channel, has a default context.  When you pick up the phone in question and dial 555, it looks for extension 555 in that context
19:21.17Slugs_how can i make it use 2 diff contexts
19:21.22tzafrir(an incoming call starts the dialplan in the context of that device)
19:21.37tzafrir(a bit simplistic, I know)
19:21.52ChannelZYou can't, except to either include one context in another, or create the dialplan in such a way that ir jumps specifically to another context with Goto()
19:22.05Slugs_oh i see...
19:22.46ChannelZyou can "include => context2" inside another context, which makes all the extensions of context2 become a part of that context
19:22.50*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
19:23.16*** join/#asterisk joako (~joako@opensuse/member/joak0)
19:23.31ChannelZHere's a practical example;  I have a phone whose configured in "out_ld" (for 'out long distance').  My out_ld context has some extensions for dialing long distance.
19:24.05ChannelZThen I have an "out_local" context which has some extensions for local calls.  Then I have an "internal" context which has inter-building extensions.
19:24.25ChannelZout_ld includes out_local at the end; out_local includes internal at the end.
19:25.00Slugs_ok so then this should work right --> http://pastebin.com/ccbM7HVe
19:25.04ChannelZHowever I have a client phone who I don't want to be able to call long distance, so their context is set to just "out_local".  Which also includes all of the 'internal' extensions since out_local includes them.
19:25.20Slugs_ahhh
19:25.38Slugs_ok now that makes sense
19:25.40ChannelZyes.  I don't see your 'demo' context but the 'playsound' one looks fine
19:25.56ChannelZand I assume the device you're dialing from is set to the 'default' context
19:26.55Slugs_exten => 1000,1,Goto(default,s,1)
19:27.12Slugs_that's it's definition correct?
19:27.31VoIP-PenguinWhat?
19:27.36*** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net)
19:27.37ChannelZYes
19:27.42*** join/#asterisk Aven (~avenger@78.36.107.48)
19:28.06Slugs_i think im getting it
19:28.19VoIP-PenguinYou should have read the book -- all this is in there.
19:28.59Slugs_ty
19:29.06Slugs_im still reading ;/
19:29.08Slugs_sorry
19:30.25ChannelZignore him
19:30.44Slugs_im just getting to the sectoin on buiding an interactive dailplan
19:34.34*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)
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19:35.30Slugs_this stuff is too exciting
19:35.47*** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
19:36.01*** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr)
19:36.13ChannelZit's fun
19:37.04Kattyi'm all FRESH and CLEAN
19:37.17ChannelZI thought you went to lunch
19:37.20Kattyi did
19:37.24Kattyand then i had a shower
19:37.30ChannelZAh :)
19:37.38Slugs_did you eat in the shower?
19:37.47Kattyno.
19:38.07Slugs_kramer on seinfeild enjoys it
19:38.14Slugs_never tried it tho
19:38.40ChannelZtime for me to go get lunch now
19:38.49Kattyryan drinks beer in the shower
19:39.04ChannelZwow that's committed
19:39.34Slugs_lol
19:39.46Kattyyeah but he leaves the beer cans in there )=<
19:39.51KattyNot Cool.
19:39.54Slugs_even better
19:40.25Slugs_does ryan like *?
19:40.33Kattyno
19:40.35*** join/#asterisk ltd_wk (~z@sixified.transact.net.au)
19:40.37Kattyhe works with Nortel mostly
19:41.14Slugs_is ryan your husband?
19:41.34Kattynot legally
19:41.44Kattywe have not signed contracts
19:41.47Slugs_hehe
19:42.22Slugs_he propose ?yet
19:42.45Kattyno
19:43.05Slugs_due time
19:43.13Kattyno, not yet
19:43.51Slugs_how long has this sinful escapade gone on? jk
19:43.56Katty4 years
19:44.13Slugs_that's how long it took me
19:44.52Kattywell i'm  not ready yet
19:45.10Slugs_balls in ur court
19:45.22Kattypretty much
19:45.30Slugs_nice
19:45.40Kattybut i live here.
19:45.43Kattyand have for 4 years.
19:45.53Kattywe have 4 ferrets and a dog...
19:46.15Kattyreally not much difference, other than we file taxes seperately
19:46.22Slugs_eyah
19:46.31Kattyand i don't have a shiny rock on my left hand
19:46.38sawgoodwhat kind of dog?
19:48.52Slugs_Katty, does it go over in the book how to make an extension (which is a meesage) call out.  For instance i could record "happy birthday to my brother" ?
19:49.39Slugs_seems like possibilities are endless over here
19:50.11Kattysawgood: http://farm1.static.flickr.com/35/103097852_1c95917c3e_o.jpg <- german shepherd
19:50.16Kattyheh, go figure.
19:50.18Slugs_lol
19:50.41Kattythe book covers call files
19:50.48Kattywhich might work for what you're wanting
19:51.19Slugs_swwet thx, beautiful dog btw
19:54.34Slugs_.
20:01.07*** join/#asterisk zeppelin_ (~Zeppelin@187.4.23.23)
20:03.24VoIP-Penguinslugs_: Extensions will do _whatever_ you want them to do.
20:04.21Kattyhrmm. bored :<
20:05.36VoIP-Penguinslugs_: For an extension that calls a number and then plays a sound file, you'll probably want to look at the Dial command.  There are a couple Dial options that can achieve it.
20:06.28Slugs_ty
20:06.30VoIP-Penguinslugs_: "core show application Dial"  lists them.
20:08.27VoIP-PenguinCall files or the "originate" console command can also achieve it, but they are pretty exclusive of the dialplan (which is what I thought you wanted to use).
20:11.47VoIP-Penguinoriginate SIP/voipms/13149691077 application playback your-sound-file
20:19.12Slugs_got ya
20:21.52devoidWhat is zaptel?
20:22.06VoIP-Penguina channel tech
20:22.25devoidDo I need that to use asterisk?
20:22.35VoIP-Penguinmaybe
20:22.43*** join/#asterisk Heretic (~Fallen@dsl-246-107-142.telkomadsl.co.za)
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20:26.03tzafrirdevoid, "zaptel" is the old name of "dahdi"
20:26.18devoidoh. that makes sense.
20:26.18VoIP-Penguinor is dahdi the new name of zaptel?
20:26.38devoidare there any good tutorials for a complete newbie to asterisk?
20:26.44VoIP-Penguin~book
20:26.45infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
20:26.50VoIP-PenguinRead.
20:26.52devoidthanks
20:27.26tzafrirAnd dahdi is http://www.educ.uvic.ca/globalarts/pages/artexchanges/countries/indiadeenabandhugame/pages/page_7.html
20:28.28tzafriractually the book is rather outdated regarding that question
20:28.41tzafrirdevoid, what version of asterisk do you use?
20:29.50devoid1.6.2.6 I compiled it today.
20:34.28*** join/#asterisk smooth_penguin (~smoove@59.95.3.217)
20:35.43Slugs_.
20:36.08ChannelZno, not again
20:36.13Slugs_hehe
20:36.27ChannelZI will not be baited
20:36.33VoIP-PenguinAre you really that afraid of timing out?
20:37.18Slugs_i don't want to annoy w/ reconnections ;/
20:37.26ChannelZYour IRC client should be pinging
20:37.36VoIP-Penguinponging, even
20:38.20*** join/#asterisk Whtsup (~sssi@203.81.226.170)
20:38.54Whtsuphello
20:39.14ChannelZohell
20:39.25Whtsupcan i use speex codec in 3kb
20:40.05Whtsupper call
20:40.26ChannelZspeex is a separate thing, not sure how it's configured
20:41.03Whtsupany one knows
20:41.33tzafrirChannelZ, codecs.conf ?
20:41.53ChannelZI assume but I've never used it
20:41.54Slugs_do i need to build in mp3 support?  -->>  http://pastebin.com/73fQa5ee
20:42.06ChannelZSlugs_: I dunno, do you need mp3 support?
20:42.34Slugs_im trying to play an mp3 file, didint know if it was already there
20:42.49ChannelZwell in that case yes
20:42.59Slugs_got ya
20:44.05ChannelZI just built it for my box at home as a test for someone else last night.. it works but it knocks down the level significantly
20:44.06*** join/#asterisk andreas-- (~andy@unaffiliated/slacky)
20:45.36Slugs_interesting
20:48.11*** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur)
20:50.07Slugs_ChannelZ, i see how asterisk has built on wav to sln , but does it do mp3 to sln?
20:50.17Slugs_s/on/in
20:50.37VoIP-PenguinOh, soo close!
20:50.44VoIP-Penguins/soo/so/
20:51.05Slugs_damnit!
20:51.07*** join/#asterisk voipmonk (~shido6@dsl-69-172-110-65.acanac.net)
20:51.24Slugs_s/damnit!/darnit!/
20:51.28ChannelZSlugs_: yes that's how it works.. but the mp3 format support is separate, in asterisk-addons
20:52.03Slugs_thx
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20:58.18*** join/#asterisk uqlev (~yuriy@91.184.221.31)
21:02.53ChannelZI can tell you how to patch it.  I'm not really sure why it's this way in the addons source
21:03.29Slugs_should i patch or just convert to wav...
21:03.53Slugs_i have an mp3 to wav converter already
21:05.06ChannelZ? up to you.  Having them native at the right samplerate means * doesn't have to transcode on the fly
21:05.22ChannelZ(unless the channel thats playing them is something else too, like gsm)
21:05.41Slugs_can i patch even tho its debian pkgs?
21:07.00ChannelZno, not unless you're building the addons from source
21:07.13Slugs_figured
21:07.39Slugs_i think i added mp3 support which is weird
21:08.13Slugs_can files not exceed a certian size?
21:08.37ChannelZthe size of your disk I suppose.. ?
21:08.42Slugs_lol
21:08.49lmns972hello everybody
21:08.50ChannelZare you talking about mp3 support?
21:08.52Slugs_for playing the audio over the phone
21:08.58VoIP-PenguinThe file system will have a limit, but it's going to be quite large.
21:09.22ChannelZWell I've noticed format_mp3 doesn't like high bitrate files.
21:09.33lmns972i'm a newbie in asterisk and i search a how-to to compile asterisk with ldap
21:09.44ChannelZAt the end of the day it's going to wind up getting converted to 16-bit 8kHz if thats what you mean
21:09.52Slugs_yeah
21:10.23lmns972i use asterisk version 1.6.0.18
21:11.29*** join/#asterisk Z_God (~julius@ip80-101-232-43.hotspotsvankpn.com)
21:12.15ChannelZldap to do what?
21:12.54ChannelZhttp://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
21:17.26lmns972I want use ldap for authentication of users
21:17.44VoIP-PenguinJust for giggles?
21:19.16lmns972ChannelZ, yes i already look this page
21:20.09lmns972but I can not install
21:21.49lmns972this archive do not contain a Makefile
21:22.47lmns972when i use the makefile of asterisk packages, it is impossible to install asterisk
21:23.32ChannelZshrugs
21:23.43ChannelZlooks old, confusing, and unsupported
21:26.28*** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380)
21:26.43*** join/#asterisk jql (~jql@12.9a.344a.static.theplanet.com)
21:26.59lmns972then I must use a more recent version of asterisk?
21:28.11Slugs_unfortunatly I doubt it ;/
21:28.51Slugs_your a minior release from stable
21:29.05ChannelZNo I mean as in nobody is using this ldap thing and thus it's old and out of date and not maintained
21:29.21ChannelZBut I could be wrong, I can't even tell what it consists of and don't care enough to figure it out.
21:31.42lmns972if I use an older version of asterisk you think it can work
21:32.57lmns972for our final project study is based on a solution with asterisk mark a active directory
21:33.03ChannelZmaybe
21:33.40ChannelZBut I don't think I've heard anyone here talk about it.. so I can't imagine it's very common or that you will find much help with it here
21:34.51lmns972:(
21:35.28lmns972at least I tried
21:35.47ChannelZwell there's two things near as I can tell
21:36.04ChannelZOne is a Perl module which will generate asterisk config files from an LDAP directory
21:36.16*** part/#asterisk andreas-- (~andy@unaffiliated/slacky)
21:37.28ChannelZIf you want realtime config via LDAP (like MySQL) that's something else and will require actual integration into asterisk.. which is the bit that is probably out-of-date and unmaintained
21:39.20lmns972this is also not maintained for mysql ?
21:39.37*** join/#asterisk Netgeeks (~chris@173.11.68.155)
21:40.34ChannelZno, realtime config via SQL is kept up-to-date as far as I know
21:42.31lmns972ok
21:44.11lmns972I'll try the 1.4 version of asterisk and if it still does not work I'll have to go through the mysql
21:49.01*** join/#asterisk nickaugust (~anonymous@167.83.189.72.cfl.res.rr.com)
21:49.20*** part/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)
21:49.52*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)
21:58.10*** join/#asterisk Z_God (~julius@wlan226162.mobiel.utwente.nl)
22:00.21Slugs_.
22:00.51ChannelZhmm did the 'j' flag of Dial get depreciated in 1.6.x?
22:01.51*** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl)
22:02.14VoIP-PenguinMaybe deprecated.
22:08.59*** join/#asterisk jkroon (~jkroon@dsl-244-4-217.telkomadsl.co.za)
22:09.19*** part/#asterisk dgoner (~dgoner@mx1.repairpc.net)
22:09.53jkroonhttp://pastebin.co.za/96550 - that's the only indication I get in my logs that something is wrong with my dahdi configuration - how do I go about figuring out what asterisk thinks is wrong?  modules loaded is wct4xxp and wcb4xxp, 1 x quad pri card and 3 x quad bri.
22:10.08jkroonpri currently populated, not the bri's
22:11.04*** join/#asterisk lmsteffan (~lmsteffan@reef.ac-noumea.nc)
22:11.30ChannelZIf you look at the source code that comes out when it tries to read from /dev/dahdi/timer IIRC
22:12.29*** join/#asterisk lmsteffan_ (~laurent@reef.ac-noumea.nc)
22:15.11ChannelZI'd maybe start with unloading the BRI drivers if you're not using them.  Some sort of timing conflict perhaps, vomit on the PCI bus...
22:15.20*** join/#asterisk lmsteffan (~lmsteffan@reef.ac-noumea.nc)
22:15.28jkroonlooks like it yes.
22:15.38jkroonhowever, they got unplugged exactly because of this.
22:15.44ChannelZI don't know what happens with multiple hardware cards, who wins for the timing source (perhaps doc/timing.txt reveals)
22:15.57*** join/#asterisk lmsteffan_ (~laurent@reef.ac-noumea.nc)
22:17.02ChannelZWell if they are unplugged and not in use, don't load the drivers
22:17.08jkroonno, doc/timing.txt waffles about how important DAHDI is.
22:17.21VoIP-Penguinmmmmm... waffles...
22:17.25jkroonand it looks like (based on kernel source code) whether the timing source is per card.
22:17.43ChannelZhmm
22:17.47*** join/#asterisk lmsteffan_ (~laurent@reef.ac-noumea.nc)
22:18.55jkroonalso, my asterisk is the owner of /dev/dahdi/timer ...
22:19.24jkroonthe more i look at this the more I think that PRI card is bust.
22:20.07ChannelZIs this failure occurring after asterisk has been running, or are you not able to start up in the first place?
22:21.57jkroonabout to start up.
22:22.09jkroonit's been running fine for a couple of days and just failed suddenly this afternoon
22:22.17*** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net)
22:22.34ChannelZhmm.. well yeah the card could be semi-dead
22:22.46ChannelZyou've run some of the test tools?
22:22.48jkroonit actually starts up, i get that error.  no link state changes in dahdi_scan output (alarms) when the cable gets plugged/unplugged.
22:23.08jkrooni'm not physically there to wire in loop-back cables etc ...
22:23.25jkroonanything else I can do remotely with it plugged into the other equipment?
22:24.53jkroondahdi_test (not sure what it tests) is somewhat happy (>= 99.98 % accuracy ... not that I think that means much)
22:24.59ChannelZ?? sorry no experience with those cards
22:25.19*** join/#asterisk seanjohn (~john@173.50.101.10)
22:25.58ChannelZdahdi_hardware -v
22:26.41ChannelZor dahdi_scan, see if it says anything interesting
22:27.58jkroonnothing.
22:28.00Kattyhowdy
22:28.05jkroonhowdy
22:28.34ChannelZit says nothing, or nothing interesting? :)
22:30.40Kattyit says ORDER ME PIZZA
22:30.51ChannelZis making pizza later..
22:31.27Slugs_from scratch?
22:31.33Kattyyes
22:31.34Kattyhe does
22:31.43ChannelZhells yeah!
22:31.50Slugs_;) i own a pizzeria
22:31.50ChannelZgotta go make sauce in a bi
22:31.52ChannelZbit
22:31.56Slugs_i know all about that ;)
22:32.08ChannelZSlugs_: cool, where?
22:32.15Slugs_atlanta
22:32.24ChannelZNice.  What style of pizza?
22:32.26Slugs_salernos pizza
22:32.26Kattylet me pack my bags, i will be right over
22:32.28Slugs_ny thin
22:32.44ChannelZexcellent
22:32.47Slugs_not 2 thin tho
22:32.49ChannelZMight have to pick your brain
22:32.54Slugs_sure thing
22:33.39ChannelZYou do the whole cold fermentation for dough?
22:33.48Slugs_i make about 25 lbs a dough a batch so the percentages might be slightly confusing
22:34.02Slugs_yes
22:34.27ChannelZMost of the problem with making dough at home is mixing, and getting good high protein flour
22:34.31Slugs_i dont use dough sooner than 4 hours IN A EMERGENCY
22:34.56Slugs_high gluton is necessary, trump is ok
22:35.41ChannelZWow, 4 hours?  Mine doesn't get good until 24+
22:35.50Slugs_tricks !
22:36.13Slugs_its best 24+ don't get me wrong ;)
22:36.26Slugs_eaiser to strech, less dense
22:36.56ChannelZDo you use it cold?
22:37.17Slugs_slightly under room temp works
22:37.30ChannelZyah
22:37.30Slugs_cold is too difficult to streach evenly
22:37.41ChannelZIt also doesn't seem to bake right
22:37.53ChannelZthe crust will be very very smooth and hardly no rise it seems
22:37.56Slugs_true, do you use screens to bake on?
22:38.02Slugs_yep
22:38.17ChannelZYeah, I build on a screen and then bake it for a few minutes until it's stiff enough to remove from underneath
22:38.24Slugs_perfect
22:38.35jkroonChannelZ, nothing interesting.
22:38.38ChannelZMy think currently is coming up with a sauce that I like
22:38.46ChannelZs/think/thing
22:39.06jkroonok, i'm thinking hardware.
22:39.19jkroonwhen I get on site in a few hours ... loop back cable?  then what?
22:39.22ChannelZjkroon: unless the config changed, could be
22:39.36Kattyjkroon: swap it around with another ...hardware?
22:39.38ChannelZjkroon: or the config is wrong and was only working accidentally up until now :)
22:39.58jkroonChannelZ, i shudder at that thought.
22:39.58Slugs_i got a neat recipe if you don't make it from scratch, it's good but not as good as all fresh
22:40.29jkroonKatty, wct4xxp ... expensive cards, can only on tuesday...
22:40.42ChannelZjkroon: so you said the /dev/dahdi/* are owned by 'asterisk'.. and I assume your asterisk runs as the same user?
22:40.48jkroonjip.
22:41.06*** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net)
22:41.30ChannelZSlugs_: I'm trying to duplicate something from a local pizza place but have a hard time determining what exactly it is about it
22:41.55jkroonhere is an interesting one, if I comment the PRI configs out in chan_dahdi then it comes up clean.
22:42.02jkroonthat is, with the BRI cards configured.
22:42.09ChannelZSlugs_: visually it seems simple, just tomato with oregano I guess.. I think it's fresh garlic perhaps where a lot of flavor is coming from
22:42.12jkroonwith the PRIs I get that error.
22:42.16ChannelZor the distinct flavor anyway
22:42.24jkrooncan I pastebin my configs and let someone take a peek?
22:42.27Kattyjkroon: they may be epensive, but think of all the labor costs put into figuring out what's going on
22:42.45Slugs_well i have to go right now but if you need anything you know, i will share an easy recipe for sauce, you will like it
22:42.49Slugs_bbiab
22:42.57jkroonKatty, they're good cards, i'm not argueing that.
22:44.24ChannelZso, either some config changed with timing, or perhaps the card is semi-dead
22:44.39doneirwct4xxp aren't expensive, we just moved from dialogic cards ($5k a pop) to digium (700)
22:45.03doneirTE220 takes more ports too
22:45.06doneirheh
22:45.36ChannelZI've no experience with PRI so I can't really help much more, dunno how all the channels work, config terminology, etc
22:46.04jkroonhttp://pastebin.co.za/96559 - dahdi/system.conf, chan_dahdi.conf and users.conf.
22:46.39jkroon700 * 8 = R5600 ... about a third of what I pay for them.
22:47.47jkroonChannelZ, just take a look anyway please, maybe you spot something obvious that I'm not seeing.
22:48.21ChannelZI am but it's all greek to me
22:48.30jkroonthanks.
22:48.36jkroonfor at least trying.
22:49.06jkrooni are now properly confused.  i'm seriously thinking HW.  going to try switching off the crc4 stuff.
22:49.09jkroonsee what happens.
22:49.09*** join/#asterisk Cain (~Geek@unaffiliated/cain)
22:50.27jkroonPRI span 1/0: Provisioned, Down, Active
22:50.33jkroonstays Down.
22:50.39jkroonalarms says OK.
22:51.19ChannelZhmm
22:51.29ChannelZmaybe is your telco hosed?
22:52.26ChannelZaside: I thought T1 only had 24 channels?
22:52.41jkroonE1
22:52.50jkroonthis is ZA :)
22:53.06ChannelZahh
22:53.21jkroonand my telco is coming in on the BRIs.  which has now been moved direct to the pabx because of this.
22:53.33jkroonanyway, i'm off to bed.
22:53.51ChannelZhmm.. well good luck
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23:23.18ecristwill an FXS to SIP adapter allow a standard dial-up modem to function?
23:27.44VoIP-PenguinWon't the modem function by itself?
23:28.12VoIP-PenguinI mean, it would be worthless if it didn't function on its own.
23:31.22ecristI have a voip server with a PRI.  I have a server in the same rack that needs to dial out.  rather than getting a pots line just for that modem, it would be nice to make use of the PRI
23:32.10VoIP-PenguinYou want to use your PRI for dialup networking?
23:34.01ecristyes
23:34.10ecristwell, one channel of it
23:42.16VoIP-PenguinWouldn't you have to have a channel bank to be able to do that?
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23:44.03ecristthat's what the FXS to SIP adapter would be doing.
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23:47.14beekecrist: You *may* get away with it.  It depends on your network and how much latency and jitter you'll be experiencing.
23:47.40beekI know one company that uses an ATA across an RF link to its Asterisk box and it works fine.
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23:50.12ecristinteresting.  This would be gigabit on the same subnet
23:50.46beekecrist: Try it is the best thing I can suggest.   Modems can be quite finicky.
23:51.12ecristI think I'm going to cut my losses and just got a pots line
23:51.21ecrists/got/get/
23:51.55beekecrist: Doable, but the cost of an ATA is under $100.  I use the SPA3102s with great success and that's what the aforementioned company used, as well.
23:53.34ecristinteresting
23:53.45ecristperhaps I'll pick on of those up and try it out.
23:54.52beekGN
23:55.10ecristGN?
23:55.18beekGood night.. I'm outta here.
23:55.26ecristAmazon has one of those for $76 after shipping
23:55.30ecristI'll give it a shot
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