irclog2html for #asterisk on 20060219

02:43.29*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
02:43.29*** topic/#asterisk is Zaptel 1.2.4 Released ... More information available on http://www.asterisk.org
02:43.39*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
02:43.41LoonaTickI tried setting all channels as B channel, then cat /dev/zap/1 |hexdump
02:43.53LoonaTickon channel 8 there was actually data coming over the channel
02:45.06LoonaTickonly strange thing is, asterisk does not pick up the d channel is 8 (I have edited /etc/zaptel.conf saying so):
02:45.06LoonaTickFeb 19 03:44:35 ERROR[19222]: chan_zap.c:9043 start_pri: D-channel 16 is not in HDLC/FCS mode.  See /etc/zaptel.conf
02:45.06LoonaTickFeb 19 03:44:35 ERROR[19222]: chan_zap.c:10843 setup_zap: Unable to start D-channel on span 1
02:45.14MoutaPTno one works on fax to email?
02:45.34LoonaTickMoutaPT: I think there's quite a bounty on it
02:46.02MoutaPTLoonaTick, i got it already working on ast@home1.5
02:46.10MoutaPTwith some workaround
02:46.27MoutaPTbut now i thought it would be ok on as@home2.5
02:46.44Corydon76-homeLoonaTick: libpri currently does not support a d channel on anything other than channel 24 for T1 lines or channel 16 for E1 lines
02:47.13LoonaTickCorydon76-home: great :( thanks.
02:50.00Corydon76-homeChannel 32 is only used for timing and therefore doesn't appear in the lineup
02:50.26LoonaTickCorydon76-home: Should that be somewhere in the libpri sourcecode?
02:50.26Corydon76-homeHence, the first E1 is 1-15,17-31
02:50.50LoonaTickCorydon76-home: Its a fractional E1 line, i'm quite sure the D channel is on 8.
02:51.11Corydon76-homeLoonaTick: Sorry, the D channel has to be on 16
02:51.52LoonaTickCorydon76-home: According to libpri or technical telephony rules?
02:52.26Corydon76-homeLoonaTick: dunno
02:52.26LoonaTick(I mean: is it possible that the d channel is on channel 8 on a fractional line, or is that just my silly idea?)
02:52.39LoonaTickok, thanks very much:)
02:54.38*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
02:55.53*** join/#asterisk zaf (n=tfournet@rrcs-24-153-146-85.sw.biz.rr.com)
02:56.06*** part/#asterisk zaf (n=tfournet@rrcs-24-153-146-85.sw.biz.rr.com)
02:56.41Corydon76-homeLoonaTick: where it detemines the location of the D channel is in chan_zap.c, in pri_resolve_span
02:58.45LoonaTickthanks!
02:59.32LoonaTickI will try recompiling it now :)
03:00.14Corydon76-homeLoonaTick: actually, you might be able to do it with a trunkgroup
03:04.56MoutaPTi got the zip with ghostscript 8.5 could you help me with the commands to install?
03:05.12MoutaPTlast time i've installed through yum ...
03:05.15LoonaTickCorydon76-home: Let me try something with those
03:05.25Corydon76-homeYou must have confused this channel with #ghostscript.  Ask there.
03:05.51MoutaPTperhaps i'm confused, but the target is AsterFAX working fine
03:05.56MoutaPTwith PDFs
03:06.26Corydon76-homeI'm sorry, I didn't realize the Asterisk community was supposed to support that.
03:07.21MoutaPTthat is not supposed, was just a simple question to improve this target...
03:07.44MoutaPTi'm not a linux expert, but i tr
03:07.59MoutaPTi'm trying to solve the problem with Asterfax...
03:08.51*** join/#asterisk Davey (i=foobar@unaffiliated/davey)
03:09.11LoonaTickCorydon76-home: I have it like this now:
03:09.12LoonaTickgroup => 1,8
03:09.12LoonaTicktrunkgroup => 1,8
03:09.12LoonaTickspanmap => 1,1
03:09.25LoonaTickbut it still complaints on channel 16
03:09.34LoonaTickyou see any mistakes in that?
03:10.34*** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net)
03:11.58Corydon76-homeLoonaTick: I didn't say I knew how to use them; only that it appears like you might be able to use them
03:13.16LoonaTickah ok, thanks.
03:21.24delmarCan anyone help me with a sip phone registration problem? im getting like.. http://pastebin.ca/42093 ...I think the phone is the problem but i'm not sure.
03:24.05sivanadelmar: what do you get in the CLI?
03:24.53delmarsivana, on what.. Asterisk?
03:25.10sivanayes
03:25.25delmarsivana, with debug on, on the ip:5060 for the phone... I get messages like here -->  http://pastebin.ca/42093
03:25.40FuriousGeorgecan someone take a look at this and tell me why its not getting to the NoOp priority?
03:25.49delmarsivana, the BS thing about it is.. sometimes this problem eventually goes away all by itself...
03:25.59FuriousGeorgehttp://pastebin.ca/42118
03:26.02FuriousGeorgethere it is
03:26.59Corydon76-homeFuriousGeorge: shouldn't be a leading slash before "MeetMe"
03:27.00LoonaTickCorydon76-home: I have hardcoded the D-Channel on 8. I get an error now:
03:27.06LoonaTick*CLI> Feb 19 04:26:20 NOTICE[5750]: chan_zap.c:8175 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
03:27.06LoonaTickFeb 19 04:26:20 NOTICE[5750]: chan_zap.c:8175 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
03:27.07delmarsivana, i mean.. after a looong while.. but this time its not at all.
03:27.11LoonaTicketc.. (ongoing).
03:27.19Corydon76-homeLoonaTick: then that's not a D-channel
03:27.24*** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net)
03:27.26LoonaTickok, thanks:)
03:27.38trixterBT Fusion service, the first in the world to allow users to switch in midcall between cellular and Voice over Internet Protocol http://www.bergen.com/page.php?qstr=eXJpcnk3ZjczN2Y3dnFlZUVFeXk2NTYmZmdiZWw3Zjd2cWVlRUV5eTY4ODEwMTQmeXJpcnk3ZjcxN2Y3dnFlZUVFeXk3
03:28.05FuriousGeorgeCorydon76-home: thanks
03:28.21Corydon76-homeFuriousGeorge: oh, and the While condition isn't inside $[ ]
03:28.57delmarSo what I want to know is... what is that response... what is happening when it says Unauthorized in http://pastebin.ca/42093... whos fault is that.. it's the phone being a jerk right? it's not Asterisk... right?
03:29.13FuriousGeorgeCorydon76-home: oh yeah
03:29.15Corydon76-homeIf the While condition initially evaluates as false, it skips down to after the EndWhile
03:30.01Corydon76-homeFuriousGeorge: and same for your set at priority 9... needs to be within $[ ]
03:30.10*** join/#asterisk starwarez (n=feria@201.124.240.163)
03:31.00Corydon76-homeThe loop might still work, but C will be defined to 1+1+1+1
03:31.18Corydon76-home...which probably isn't what you want for the DB retrieval
03:31.46FuriousGeorgereally?
03:32.02FuriousGeorgeoh, you mean if i dont add the []
03:32.08*** join/#asterisk denon (i=denon@tor/session/x-6f3b78a3ac1559ce)
03:32.08*** mode/#asterisk [+o denon] by ChanServ
03:32.10Corydon76-homei.e. DB(Meetme/1+1+1+1) probably doesn't exist... you probably wanted DB(Meetme/4)
03:32.31FuriousGeorgebut ill be ok if i put the $[] aorund the counter increment
03:32.35Corydon76-homeYeah...
03:32.46Corydon76-homeI just realized though that it's a pretty neat trick
03:33.13Corydon76-homeI need to have Olle put that in an exam
03:34.01FuriousGeorgewhat mine?
03:34.24FuriousGeorgeits got some issues
03:34.39Corydon76-homeNeat, it evaluates correctly, but it won't work where you expect a simple number
03:34.45FuriousGeorgei.e. i gotta add another extension to kill the conference, if remote party decides to hang out
03:34.57FuriousGeorgeoh, nm
03:35.06FuriousGeorgeyou mean the trick i didnt mean to make
03:35.11Corydon76-homeWell, it also won't work if you use higher precedence operators inside the expression
03:35.29Corydon76-homeYeah, that trick
03:35.49FuriousGeorgemy plan is to use this with devstate to mimick an oldschool switched system
03:36.37Corydon76-homeUh, we already built that into Meetme, btw
03:37.00*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
03:37.07[av]bani\o>
03:37.09[av]bani<i/
03:37.19Corydon76-homeIf you use the 'e' option to MeetMe, it will select an empty conference automatically
03:37.31[av]baniFuriousGeorge: still trying to emulate key system?
03:37.42Corydon76-homeor E:  empty and pinless
03:37.50FuriousGeorge[av]bani: i just started again
03:37.55[av]banistarted?
03:37.58Corydon76-homeMore stuff I wrote...
03:38.01FuriousGeorgepurely for fun though, bristuff and devstate break my fxs
03:38.06[av]banio_o
03:38.15[av]banimaybe agi is the solution, or hacking app_meetme
03:38.42FuriousGeorgeCorydon76-home: what happens if i specify a conference thats not empty
03:38.59Corydon76-homeFuriousGeorge: the intent is to use it without a specified conference
03:39.09Corydon76-homeor with the d flag
03:39.16FuriousGeorgedoes it output which conference it picked
03:39.21Corydon76-home(or D)
03:39.32Corydon76-homeYes, it outputs the confno to the caller
03:39.39FuriousGeorgelike in a var, though
03:39.46FuriousGeorgesomething i can manipulate to set an LED
03:39.48Corydon76-home"You have entered conference number four"
03:41.36*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
03:41.37*** topic/#asterisk is Zaptel 1.2.4 Released ... More information available on http://www.asterisk.org
03:41.39[TK]D-FenderFuriousGeorge : Try using "devstate in the BRISTUFF patch with presence.  Set it from a script that will poll the room status on a timed interval.
03:41.45Corydon76-homeor listen to the manager interface for MeetmeJoin events
03:42.05FuriousGeorge[TK]D-Fender: i got bristuff installed, thats the plan
03:42.18FuriousGeorgeCorydon76-home: im not quite on that level yet
03:42.18_-Jon-_hey everyone, i need some help with a frustrating problem I'm having with Asterisk..  I can go into more detail of my setup, but the basic problem is after some time, one of the members in the call can't hear each other.  Is there any way to determine what the cause might be?
03:42.42FuriousGeorge[TK]D-Fender: unfortunately bristuff makes 2 of my zap channels stop accepting DTMB (dialtone persists)
03:42.48FuriousGeorgeso by monday its gone
03:43.19Corydon76-home_-Jon-_: how much time?
03:43.21FuriousGeorge[TK]D-Fender: correction i planned to use a counter and a fixed amount of rooms not a timer
03:43.46delmaranyone here using an IPPH-202 or had working with * ok?
03:43.59_-Jon-_Corydon76, it's quite random.  sometimes it doesn't happen.  Like can talk on the phone for about an hour and no problem, and just now, 3 times in about 10 minutes
03:44.17Corydon76-home_-Jon-_: SIP?
03:44.44_-Jon-_It's quite confusing, but SIP, IAX, then SIP again
03:45.05Corydon76-homeAny firewall between the phones and the server?
03:46.14_-Jon-_Well yes and no..  My Sipura is connected using SIP to an Asterisk server in my basement, which then uses IAX to connect to a dedicated I have, which is then connected to my provider using SIP
03:46.28[TK]D-FenderFuriousGeorge : wierd....
03:46.43Corydon76-homeYou're going to need to isolate this down to a particular leg of the call
03:47.29Corydon76-homeBecause it's probably happening on one leg or another
03:47.46_-Jon-_Hmmmm, my guess is it's the one in the basement
03:47.49Corydon76-homeYou might try recording your calls on both servers, as a starting point
03:48.05_-Jon-_ah that's actually a really good idea..  see which one is getting audio and which isn't
03:48.24Corydon76-homeThat will eliminate at least one leg as the culprit
03:49.02Corydon76-homeand not just which aren't getting audio, but where the audio quits in either direction
03:49.21_-Jon-_That's true.  I'll have to add that into my dialplan
03:50.06[av]baniFuriousGeorge: write app_keysystem.so :)
03:50.10_-Jon-_Is there any logging options in Asterisk?
03:50.13*** join/#asterisk brockj49464 (n=brockj49@63.87.56.235)
03:50.33*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
03:50.37Corydon76-home_-Jon-_: yes, but I suspect recording is what will benefit you the most
03:51.00Corydon76-home_-Jon-_: unless you're a developer, the logs aren't likely to be useful to you
03:51.12_-Jon-_I'm a very weak developer :P
03:51.27Corydon76-homeSee /etc/asterisk/logger.conf
03:52.07_-Jon-_Thanks, i'll take a look, but liek you said, probably won't be useful to me
03:52.31Ad-HocAnyone with Asterisk + Bluetooth ?
03:53.14starwarezhi all: what do i need to change to call to FXS cards instead of FXO on this document  http://www.voip-info.org/wiki-Asterisk+cisco+FXO
03:53.36starwarezanyone is calling to FXS interfaces on routers
03:54.11_-Jon-_corydon76, thanks for your help :)
03:54.38Corydon76-homestarwarez: FXO and FXS are simply signalling opposites
03:54.55Corydon76-homeFXO can only talk to FXS and vice versa
03:55.18FuriousGeorge[av]bani: funny :)
03:55.44FuriousGeorgeCorydon76-home: it appears that this wont work b/c after i exit the meetme it doesnt DB_DEL on the next priority
03:55.54starwarezCorydon76-home: I want to call from a soft-phone to the FXS port, can this be done?
03:56.23Corydon76-homestarwarez: probably, I've done it
03:57.45FuriousGeorgewhy doesnt it go to the next priority after the conference is empty?
03:57.52*** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net)
03:58.01starwarezCorydon76-home: based on this? http://www.voip-info.org/wiki-Asterisk+cisco+FXO
03:58.17*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
03:58.35Corydon76-homeFuriousGeorge: probably because they hung up?
03:59.20Corydon76-homestarwarez: no, I don't use Cisco crap
04:03.05FuriousGeorgeCorydon76-home: hmm, i dont see a workaround for that in the dialplan
04:04.20FuriousGeorgewell, acutally yeah i do, i can querry the rooms with meetmecount
04:05.36Corydon76-homeWhy are you lighting LEDs in the dialplan, anyway?
04:05.46*** join/#asterisk trixter (n=trixter@65.172.209.246)
04:05.57Corydon76-homeShouldn't you be using the manager interface to watch for appropriate events?
04:06.10Corydon76-homeYay, kram!
04:06.34FuriousGeorgeCorydon76-home: i suppose i should but ATM i just dont know how so meetme count it is
04:06.36Corydon76-homeHow's San Diego?
04:06.38kramgreetings tilghman
04:06.41krami'm in auburn right now
04:06.47Corydon76-homeOh, okay
04:06.51kramits been a lot of traveling this past week
04:06.55Corydon76-homeThought you might be with Kevin and file
04:07.47Corydon76-homeIs there ever a week when you aren't travelling?  ;-)
04:08.05Corydon76-homeSeems like you're perpetually on the road...
04:08.49*** join/#asterisk darkpixel (n=darkpixe@209.216.189.127)
04:08.57Abydos313anyone ever try the cisco softphone? curious how nice it is
04:08.59kramyah, i know, it does seem like that a lot
04:10.40harryvvkram have a choice hotel you like? :)
04:12.35*** join/#asterisk coppice (n=chatzill@246.206.17.210.dyn.pacific.net.hk)
04:12.51Corydon76-homekram: I need to call Debbie and find out which hotel we'll be at for VON... I'm getting a car...
04:14.09Corydon76-homeHaving a car was massively useful in Vegas
04:14.35Corydon76-homethanks to JerJer for letting us borrow the use of it
04:15.07coppicethey let you bet with cars? :-\
04:15.28Corydon76-homecoppice: we needed Red Bull on the convention floor...
04:15.36Corydon76-homeso twisted got two cases
04:16.03Corydon76-homekram: we'll make sure we get you some diet this time around...
04:16.40Qwellkram: hey
04:18.40coppicea convention floor is usually jam packed with bull
04:19.07QwellCorydon76-home: others have been saying Fremont
04:19.20Corydon76-homeQwell: eh?
04:19.23Qwellhotel
04:19.26Corydon76-homeOh
04:19.40Qwellfile hates the Fremont for some reason
04:19.45Corydon76-homeI need to get a roadmap so I can drive from the airport
04:22.22Corydon76-homeQwell: is that where the con is, where Digium is staying, or both?
04:22.56*** join/#asterisk Eggplant (i=No@64.233.107.198)
04:25.07mogormanwhere id digium, and where is redbull
04:25.11QwellCorydon76-home: afaik, it's on the same block - possibly connected
04:25.24Qwellmogorman: digium will be where the redbull is :D
04:25.33mogormanwell that is where mog will be
04:25.42Qwellmog == digium
04:25.47Abydos313evening everyone
04:26.27mogormanmog == digium?
04:26.29mogormanim not that fat
04:26.34coppiceeven if its morning?
04:26.36Qwell:p
04:27.06coppiceer, time flies. even if its afternoon
04:27.15Abydos313PST here
04:27.41coppiceCST here
04:27.47QwellBST
04:28.16Corydon76-homeCST here, although probably a different CST than coppice
04:28.42Corydon76-homecoppice: unless you're in the US...
04:28.52justinuchinese standard?
04:29.01QwellCoppice Standard
04:29.05Qwellduh
04:29.05Corydon76-homeQwell: how about the Fairmont?
04:29.11QwellCorydon76-home: That might be it
04:29.17Corydon76-home3 blocks away
04:29.40Corydon76-homeMarriott and Hilton are connected
04:29.57coppicewhy does the US need 5 time zones when 1.3B people live in just one? :-)
04:30.10justinuheh
04:31.24Abydos313so they can run commercials 3 times more
04:31.53glm2klol
04:32.20*** join/#asterisk denon (i=denon@tor/session/x-9b99d06e0b91b30e)
04:32.20*** mode/#asterisk [+o denon] by ChanServ
04:32.31coppiceI live in the world's most popular timezone
04:32.54implicitcoppice, cause it makes a HELL of a lot more sense than the chinese system, where during certain times of the year there is a huge variation in daylight for the single time zone
04:32.56glm2kmay i ask why?
04:32.57implicitalso it is more expensive
04:33.04implicitin terms of energy expenditure
04:33.13implicitand makes daylight savings much less useful
04:33.46delmarimplicit, that makes perfect sense to me.
04:33.55coppiceimplicit: then why do other places as far spread as the extremes of china use the same zone? (e.g. .ph and .my)
04:34.12delmarimplicit, i mean, why run the ass end of the day in the middle of the night or whatever.. needing more power for light/heat etc.
04:34.23implicitbecause the economic impact of having them in a different time zone would be greater than the energy conservation
04:34.34implicitsince they are geographically small
04:34.43glm2kimplicit: well from a military standpoint, and order made in shanghai for a launch at 5pm in mongolia, will always happen at 5pm in mongolia. no mistakes.
04:34.50implicitglm2k, for sure
04:34.55implicitI support UTC timekeeping
04:35.27implicitbut that doesn't mean I disagree with daylightsavings or with having multiple timezones in a country
04:35.43QwellI honestly don't see how dst helps with energy saving
04:36.04coppiceimplicit: so what is the economic advantage of malaysia and the philippinnes using the same time zone?
04:36.07delmarwell here is something even more boggling... why does the general USA have a time/date format of mm/dd/yy when the rest of the world have the much more logical dd/mm/yy format.. AND evne the USA military use dd/mm/yy also !
04:36.10Qwellwhere are they supposed to come from?  not having the AC on?
04:36.10implicithttp://www.energy.ca.gov/daylightsaving.html
04:36.16glm2kgets you up earlier so you feel sleepy earlier and don't stay up watchng tv :)
04:36.17Qwellgreat, instead, you have the lights on at night
04:36.23implicitcoppice, business hours
04:36.29Qwellit's total BS :p
04:36.32implicitbut not sure about their specific policies
04:36.42implicitit might be purely political
04:36.50glm2kcoppice: the philippines pretty much fits in a zone
04:37.14coppicebut its far from malaysia, yet uses the same zone
04:37.53justinuphillipenes is a 4 hour flight from malaysia, iirc
04:37.53*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
04:37.54Qwelland it you're on the wrong side of the timezone...it's actually dark when you wake up - which requires...*gasp* lights
04:38.03glm2ker, nope. manila time is the same as hongkong time
04:38.07impliciti've got a flight in a few min
04:38.11implicitso see you guys later
04:38.12coppiceand WTF does india have to be on a 30minute split zone? :-\
04:38.22Abydos313later implicit
04:38.28justinucuz it's 1337
04:38.44glm2kcoppice: heh, you're talking abnout a country that's socially split 7 ways (i think)
04:38.47*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
04:40.59coppicejustinu: next time i'm in india i'll have to complement them on their 1337 concepts of time :-)
04:41.50justinuheh, i bet they'll appreciate that
04:44.53FuriousGeorgeis this happening because im trying to delete a database entry that doesnt exist?
04:44.58FuriousGeorgehttp://pastebin.ca/42121
04:47.52[TK]D-FenderDBDel is depricated....
04:48.02FuriousGeorgewhat is it now?
04:49.08FuriousGeorgewhatever, i gotta rewrite this now anyway.  its become convoluted
04:50.24Qwellkram: hi..again
04:51.15kramhi qwell :)
04:53.18*** part/#asterisk darkpixel (n=darkpixe@209.216.189.127)
04:55.22*** join/#asterisk tuxinator_linux (n=tuxinato@m110e36d0.tmodns.net)
05:00.18*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167033017.pppoe-dynamic.nb.aliant.net)
05:01.08Qwellfile: back home?
05:01.15tuxinator_linuxfile: ? are you there?
05:01.17fileyes
05:01.20fileI am home.
05:01.25Qwelltsk, tsk, tsk
05:01.34tuxinator_linuxI am on the road to Phoenix
05:01.36fileflights went... weird
05:01.45justinuin canada, eh?
05:01.57tuxinator_linuxflights?
05:02.02*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
05:02.09justinuhey benk
05:02.12justinubenjk
05:02.17filetwo of my three flights were screwed up in some way
05:02.20benjkhi justin
05:02.23Qwell2 of 3?  ouch
05:02.31Qwell0 of 3 would still be ouch
05:02.43ManxPowerAnd people wonder why I don't like to fly.
05:02.58fileso much that if we didn't turn around for the second flight the stuff could have overheated and we would have lost pressure
05:03.05coppiceanyone who likes to fly doesn't do it very often
05:03.07Qwellfile: nice
05:03.26tuxinator_linuxfile: we're glad you're safe at home
05:03.26fileblargh
05:04.20Kattyfile :<
05:04.40tuxinator_linuxhey file, go to bed, you're overdue for some sleep
05:04.54fileI will soon
05:04.55Kattyi wish i could get some sleep
05:04.59fileI haven't eaten either
05:05.00Kattybut i'm in too much pain
05:05.43tuxinator_linuxKatty is 4 teeth short
05:05.55Kattyand allergic to her pain killers
05:05.59FuriousGeorge[TK]D-Fender: actually i was using meetmecount wrong, but i found a better way of doing it
05:06.13coppiceshe should keep away from those rough places :-)
05:06.57tuxinator_linuxwell, I will be in and out, need to get some work done
05:07.27*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
05:07.56FuriousGeorgeok, apparently meetme wont let the only person in a conference be an outside party
05:07.58Corydon76-homeKatty:  have some ibuprofen
05:08.25Corydon76-homeKatty: it will at the very least make the pain bearable
05:08.48Corydon76-homeKatty: take about 8 of them
05:08.58*** join/#asterisk HamYaI (i=HamYai@125.24.2.115)
05:08.58KattyCorydon76-home: the doctor said i could only have 3
05:09.05Corydon76-homeReally?
05:09.10Qwellhow many mg?
05:09.11Kattyyes
05:09.18Qwell3x200?
05:09.19Corydon76-homeThe dentist I had suggested 8
05:09.32glm2kmeds are dependent on body size last i remember
05:09.35Corydon76-homeSurgeon, I should say
05:10.15QwellKatty: 3x?mg per ? hours?
05:10.28Corydon76-home4 hours for me
05:10.40Corydon76-homebut I never took them that often
05:10.51glm2kthat's was some surgery
05:10.55FuriousGeorgei sooo have a better solution
05:10.57Corydon76-homeThen again, I'm not allergic to Vicodin ES
05:10.58HamYaIIs there anyway to limit SIP registration in * to one connection per extention?
05:11.12QwellKatty: not suggesting you take more :p  Just curious
05:11.22HamYaIs/extention/extension
05:11.37delmarKatty, get drunk :P
05:11.41glm2klol
05:11.41Corydon76-homeHydrocodone or Oxycodone is some gooooooood stuff
05:11.46Qwelldelmar: That would be VERY VERY bad...
05:11.47Kattyi don't drink, delmar
05:11.53coppiceoh, yeah. go ahead. trust someone with a medical degree over a bunch of anonymous people on the internet.
05:12.01Qwellalcohol == thins blood == bleed a lot
05:12.20delmarQwell, oh.. is that why i cant feel my feet most of the time these days?
05:12.33glm2kimagine bleeding to death from having your wisdom teeth pulled out.
05:13.13Kattyoh god
05:13.14Kattyglm2k: shut up
05:13.18Kattyglm2k: right now
05:13.24Qwellglm2k: quells
05:13.29glm2kmy thoughts exactly
05:14.32*** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net)
05:15.17[av]baniFuriousGeorge: still no app_keysystem ?
05:16.32znoGwow i weigh around 180-185lbs
05:16.45znoGdidn't realise I was that heavy
05:16.51znoGi'm used to measuring it in kilos
05:16.53znoGsounds a lot less
05:17.23[av]banitry it in KT, you'll weigh practically nothing
05:17.30glm2khehehe
05:18.25FuriousGeorge[av]bani: i just figured it out
05:18.40FuriousGeorge[av]bani: easier than i though
05:19.39[av]banio_o
05:19.41[av]banitell!
05:19.48HamYaIIs there a parameter to limit the # of SIP registrations for each peer? in sip.conf?
05:20.30HamYaII've searched through the wiki, but found nothing about this
05:21.43*** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com)
05:22.09*** join/#asterisk dmaust (n=dmaust@adsl-67-120-175-165.dsl.lsan03.pacbell.net)
05:23.25znoG[av]bani: :)
05:24.21FuriousGeorge[av]bani: a simple counter finds the first free room with meetmecount
05:25.37FuriousGeorgea remote party can be directly transfered to a meetme using transfer and snom button, but if that room is full (because someone x-fered someone to that line) itll either play congestion or stick it on the next 'line'
05:26.05SplasPoodIs there currently an enhanced parking solution for Asterisk 1.2 that allows you to setup seperate parking lots for different groups of exten?
05:26.12FuriousGeorgethe next guy who enters that room is the marked user, and when he leaves the room is closed, right?
05:26.19[av]baniFuriousGeorge: and the led?
05:26.34FuriousGeorge[av]bani: simple devstate in between
05:27.11FuriousGeorgewait a sec...  the led...  when someone leaves....
05:27.59FuriousGeorgeexten => h,1,set the devstate
05:28.09FuriousGeorgebooyah
05:36.31*** join/#asterisk cyonics (i=cyon@tx-71-52-77-114.dhcp.sprint-hsd.net)
05:36.40[av]baniFuriousGeorge: you should put a page up on voip-info :)
05:36.55[av]baniFuriousGeorge: snom shared line emulation
05:37.14FuriousGeorgewhen i get it working i will, just so i can hear why it is a bad idea.  im not gonna put it into production till bristuff goes stable so ill have time
05:37.15[av]banior rather, BLA emulator :)
05:37.53[av]baniso basically you have parking, park indicators, and able to pick up parked calls remotely
05:37.56[av]bani?
05:38.18FuriousGeorgeyeah
05:38.26[av]banithat will make people happy :)
05:38.43[av]banican anyone barge in too?
05:38.45FuriousGeorgethe tricky part will be preventing myself from transfering two remote parties into the same room
05:38.52FuriousGeorgeif i say you can :)
05:40.08[av]banihmm i think you could make it reserve a spot as soon as you hit transfer
05:40.17[av]baniand only after you hit the extension, complete it
05:40.22znoGwhat do you guys use the parking for? mostly?
05:40.47FuriousGeorgeit wont matter b/c im gonna use meetme count unless you are in the "bargers" context
05:40.49[av]baniso.. transfer picks a parking spot and reserves it.. would prevent accidentally transferring two parties
05:40.59[av]banihmm :)
05:41.09FuriousGeorgeiow, only bargers can enter a non empty conference, basically
05:41.15[av]banihow many snoms you got?
05:41.19FuriousGeorgeand incoming calls wont be bargers
05:41.31FuriousGeorgewe gonna have 6, im playing with the first and only one
05:41.36FuriousGeorgemaybe 8
05:41.39[av]banii am thinking of writign a replacement UI for it too
05:41.49[av]banisince you can replace idle, and line screens with xml
05:42.04[av]baniand remote control a lot with action URLs
05:42.11FuriousGeorgeplease do b/c the call id with number display is bad
05:42.24[av]banii am thinking you could fix a lot of the suck
05:42.29[av]baniyeah its like 300 point font
05:42.55FuriousGeorgelemme see if i can hammer out a working example soon so i can intoxicate myself tonight
05:43.14[av]banihttp://www.voip-info.org/wiki/view/snom+360
05:43.26[av]banimaye you can add your voice to the huge font feature request
05:43.39[av]baniright now its 1 for (me) and 1 against (sdjernes)
05:43.48[av]banii cant imagine why anyone would be for hugefont all the time
05:44.09[av]baniunless your snom screen are 15 feet away from your desk
05:44.09FuriousGeorgei certainly do that
05:44.15FuriousGeorgeor your blind
05:44.27[av]baniwell snom is inconsistent about it though
05:44.31[av]banithe idle screen uses tiny font
05:45.20FuriousGeorgeshoot, once i x-fer the incoming call its in my context....
05:45.46FuriousGeorgesmall setback
05:45.49FuriousGeorgejust realized
05:45.52[av]bani:o
05:46.14FuriousGeorgei dont want the button only to work to park a person, i'd like them to be able to join that way too, ya know
05:49.14[av]baniyes
05:49.16[av]baniheh
05:49.29[av]baniso youll need to make the scripts under two contexts?
05:49.41cyonicslooking for help w/ setting up asterisk to accept inbound calls via broadvoice, can anyone spare their time?
05:49.44FuriousGeorgethat would work but then its two different buttons
05:50.24FuriousGeorgei.e. led is mapped to *1, but they can access the line by hitting just 1
05:51.08cyonicsi can pay
05:51.41FuriousGeorgedont ask to ask, just force a pastebn of your sip.conf, extensions.con, and cli output, and someone will help you free
05:51.55FuriousGeorgeforce=post
05:52.09SplasPoodMy problem is that the current parking system works fine, cept nothing prevents one of my customers from picking up another's parked calls
05:52.34cyonicsI need more hand holding which is why I was looking to pay
05:52.48SplasPoodcyonics: I did it last night using the sample on www.voip-info.org
05:53.03FuriousGeorgeSplasPood: huh?  an outside party can pickup another outside party, while parked?
05:53.22*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167033017.pppoe-dynamic.nb.aliant.net)
05:53.25cyonicshmm, i'll look for this sample, thanks
05:53.31SplasPoodFurious: Not an "outside" party, simply another phone registered with asterisk..  Cept that phone is in another context, etc..
05:54.42SplasPoodcyonics: for inbound technically all you need is the register => line, assuming your global settings in sip.conf are proper ( context= set, proper codec allowed -- by default generally is..)
05:55.25SplasPoodcyonics: just make sure whatever context you have context= pointing to has an entry for the # you're placing at the end of the register => line
05:55.43FuriousGeorge[av]bani: i just need to use astdb again, so that the button works both ways.  after i've established that a room is empty, i can manually set an astdb value that causes the behavior of the extension to change (i.e. people in the barge context can now enter the room if membercount > 0
05:55.45cyonicswell I just reinstalled so I have a fresh install
05:55.53FuriousGeorgethis is such a learning experience
05:56.01SplasPoodFuriousGeorge: Think from a service provider point of view...  two different customers, phones are in diff contexts, etc...   But its all 1 parkedcalls
05:56.10FuriousGeorgeoh
05:56.11FuriousGeorgeyeah
05:56.14SplasPoodcyonics: well the defaults should be ok then
05:56.33SplasPoodcyonics: cept for context=, you'll generally wanna wipe out extensions.conf and start anew
05:56.36FuriousGeorgei hear thatll be fixed by next summer, something like what im doing may help you too
05:56.42*** join/#asterisk lalito (n=erg@201.154.202.128)
05:56.51SplasPoodFurious: yea...   Its a pretty big deal killer :)
05:56.56SplasPoodRight now I'm using meetme rooms
05:56.59FuriousGeorgemeetme's instead of parking lots
05:57.06FuriousGeorge:)
05:57.22FuriousGeorgeso it _/IS/_ possible
05:57.30SplasPoodyea, cept I was semi-reluctant to spend the time duplicating all the functionality of app_parking or whatever its called
05:57.49SplasPoodWell problem is there's no way for people to figure out where calls are parked
05:58.04SplasPoodAnd Ineed to switch meetme to automatically pick a room
05:58.20SplasPoodand then return the room # to the transferring party
05:58.34FuriousGeorgewhy not do what i am expoerimenting with and use either meetmecount, an astdb value, or a combination to do the right thing
05:58.44FuriousGeorgei keep getting to 90% there then i rethink it
05:58.49SplasPoodyea I could..
05:59.05SplasPoodthe status monitoring you're doing isn't really that important to me since I'm a polycom shop
05:59.11[av]bani:)
05:59.13[av]baniFuriousGeorge: AGI!
05:59.17SplasPoodand support for their stuff is coming (SIP-B) in 2006 too..
05:59.37FuriousGeorge[av]bani: i know asterisk dialplan, and i was toaught pascal 8 years ago, thats it.  cant code right now :(
05:59.50*** join/#asterisk tuxinator_linux (n=tuxinato@m190e36d0.tmodns.net)
05:59.51[av]baniSplasPood: not all phones support (or will support) sip-b. FuriousGeorge's stuff would work with almost any phone
06:00.07SplasPoodindicator lights and all?
06:00.14FuriousGeorge[av]bani: if i really can iron out all the wrinkles
06:00.21[av]baniFuriousGeorge: ael?
06:00.22FuriousGeorgethe meetme->led is well documented
06:00.36SplasPoodYea, but I thought that was semi snom specific
06:00.38FuriousGeorgeael=dialplan language?
06:00.43SplasPoodya
06:00.44SplasPoodI use it
06:00.52[av]baniseems to me ael would be a bit more flexible
06:00.52SplasPoodit's just extensions.conf with a bit nicer format
06:00.57FuriousGeorgeSplasPood: i dont think polycoms support devstate do they
06:01.06*** join/#asterisk thebonehead (n=bonehead@c-24-6-1-216.hsd1.ca.comcast.net)
06:01.10[av]baniFuriousGeorge: devstate is snom specific afaik
06:01.19SplasPoodFurious: wqhat av said..
06:01.35FuriousGeorgeerr, how long would it take someone of slightly above average inteligence and asterisk experience (and pascal 8 years ago) to master that
06:01.43SplasPoodAEL?
06:01.44SplasPoodnot long
06:01.51[av]banithe application hinting should work on anything though
06:02.06SplasPoodworst part is dealing with it's odd consistency or lack there of (common to much of asterisk..)
06:02.15*** part/#asterisk thebonehead (n=bonehead@c-24-6-1-216.hsd1.ca.comcast.net)
06:02.22[av]baniman, i should just integrate lua into asterisk
06:02.30[av]banithat would make things suck far less
06:02.31FuriousGeorge[av]bani: :(
06:02.52SplasPoodright now I'm working on LCR
06:03.17SplasPoodSince we're looking to start offloading minutes to other random providers via SIP
06:04.07SplasPoodFurious: you can do this with AEL
06:04.08SplasPood<PROTECTED>
06:04.08SplasPood<PROTECTED>
06:04.08SplasPood<PROTECTED>
06:04.08SplasPood<PROTECTED>
06:04.08SplasPood<PROTECTED>
06:04.10SplasPood<PROTECTED>
06:04.12SplasPood<PROTECTED>
06:04.14SplasPoodthat should sell you right there. :P
06:04.28SplasPood(there was another else I cut off)
06:05.47SplasPoodDoing the same thing in straight ext.conf is fugly
06:06.13SplasPoodeven tho AEL just converts to that anyway, internally
06:06.16SplasPoodor so it seems..
06:10.23FuriousGeorgehey im back
06:10.49FuriousGeorgeyeah, im gonna have to learn that
06:11.21FuriousGeorgebasically, its a more complicated more powerful scripting language for the * dialplan.  i read about it in the book but didnt think much more
06:11.39FuriousGeorgenow im going more complicated stuff...  no reason not to learn it, right?
06:12.25websaewhat softphone supports g729--anyone?
06:13.40justinueyebeam
06:13.48FuriousGeorgetrue
06:20.48coppicecommercial softphones generally support g.729
06:20.49coppiceopen source ones generally don't
06:21.17justinusnom360 softphone might
06:22.51justinusnom 360 softphone should do it
06:22.54justinuafk
06:32.06*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
06:34.11*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
06:34.51websaehow do i get my friend's asterisk server to register with mine so they can make calls out, i have a sip trunk to a PSTN gateway
06:35.57SplasPoodFuriousGeorge: Yea, with AEL you end up only loosing half the hair you would using ext.conf directly
06:41.12thazzawebsae: Research IAX2 to IAX2 for asterisks servers
06:43.55*** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net)
06:45.35websaesetting up a peer
06:45.38websaehow does one do that?
06:45.45websaei have a user setup in sip.conf
06:46.01websaeon my other machine that i want to peer to it
06:46.07websaedo i just put the same credentials
07:10.30thazzawebsae: www.google.com             > asterisk to asterisk
07:13.43*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
07:18.11*** join/#asterisk lalito (n=erg@201.154.202.128)
07:25.34*** join/#asterisk Pegger (n=peg@pool-68-163-174-106.bos.east.verizon.net)
07:27.51*** join/#asterisk astar` (n=astar@ANantes-154-1-59-20.w81-53.abo.wanadoo.fr)
07:28.50SplasPoodso why doesn't canada have it's own country code?
07:29.15SplasPoodbarring the obvious, 51st state jokes
07:30.44coppicewhat is .ca then?
07:31.14*** join/#asterisk camonz (n=camonz@200.8.20.210)
07:31.53*** join/#asterisk tengulre11 (n=tengulre@222.90.66.4)
07:33.21Qwellcoppice: california
07:33.31camonzhi
07:33.34coppicecalifornia is .ca.us
07:33.39Qwellcoppice: I'm kidding
07:34.21coppicea lot of people aren't when they say that
07:34.50Qwellheh
07:35.57SplasPoodcoppice: well I meant dialing code..
07:36.23coppice1 covers a lot more than the US.
07:36.33Qwellwhich is a dumb idea, IMO
07:36.42camonzi'm having 2 probs, 1st when dialing from the line1 iface in a spa3k the call takes like 3-6 secs to complete, the second is i cannot seem to configure MoH
07:36.42Qwell1+ dialing shouldn't cost you money...
07:36.48Qwellrather...not more than LD
07:36.57QwellJust like 7 digit dialing shouldn't cost anything
07:37.00EggplantAnyone used the Grandstream Budgetone 101??
07:38.52camonzany ideas on the spa3k prob?
07:39.08camonzi reduced the waiting time for digits on the configuration interface of the spa
07:39.23camonzwich reduced the time, from like 10 secs to complete a call
07:39.27camonzto 6
07:44.43*** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it)
07:46.25*** join/#asterisk robbie2 (n=rob@CPE-60-231-51-236.qld.bigpond.net.au)
07:46.41robbie2im suddenly getting
07:46.43robbie2wcte11xp: disagrees about version of symbol zt_receive
07:46.43robbie2wcte11xp: Unknown symbol zt_receive
07:46.43robbie2wcte11xp: disagrees about version of symbol zt_ec_chunk
07:46.43robbie2wcte11xp: Unknown symbol zt_ec_chunk
07:46.44IronHelixegg- need help with your barbiephone?
07:46.46robbie2....
07:46.46IronHelixi've used them
07:46.52robbie2whenever i try insert the module
07:47.01robbie2they worked before
07:53.17*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
07:55.14*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
07:56.27djin_ibQ: I'm running into something strange on a Debian fresh install:
07:56.28djin_ib/usr/sbin/safe_asterisk: line 48: /dev/tty9: Input/output error
07:56.49djin_ibAnyone any thoughts? Search didn't come up with something.
07:59.02[av]bani...
07:59.31Peggernever seen anything like it
07:59.51Peggertty9 must be your console
08:00.46camonztty9 is a console
08:01.15camonzprobably it's trying to access that file and it doesn't has enough permission to write to the file
08:01.27djin_ibcrw-rw----  1 root root 4, 9 2006-02-15 12:03 /dev/tty9
08:01.27djin_ibCould it be related to 'no proper rights'?
08:01.27djin_ibI'm starting safe_asterisk as root
08:01.48camonzthen you should have all the rights to the system
08:02.09camonzyep, it's related to no proper rights,
08:02.09shido6sudo su -
08:02.21shido6"sudo su - "
08:02.25camonzwhat you could do is chmod dev/tty9
08:02.46camonzto 662
08:02.58djin_ibOk, I'll try
08:03.09camonzwich is pretty much the same except that you add a write permission to the 'other' group
08:03.10[av]banicamonz: dialing digits takes time, you can reduce it [pstn line] -> PSTN Dial Digit Len: -> .05/.05
08:03.32[av]banicamonz: some LECs cant handle DTMF that fast, mine can :)
08:03.39camonzLEC?
08:03.47[av]banilocal exchange carrier
08:03.48[av]baniphone co
08:03.59djin_ib/usr/sbin/safe_asterisk: line 47: /dev/tty9: Input/output error
08:04.00djin_ib/usr/sbin/safe_asterisk: line 48: /dev/tty9: Input/output error
08:04.00djin_ibAsterisk ended with exit status 1
08:04.00djin_ibAsterisk died with code 1.
08:04.02djin_ibhmmmm
08:04.06camonzthanks!
08:05.24camonzdjin did you change the permission of /dev/tty9?
08:05.33djin_ibyes, tried 622
08:05.45camonztry 666
08:05.51djin_ibeven 666 for the fun of it
08:05.56camonz:->
08:06.06djin_ibNo change :(
08:06.18[av]banicamonz: you can also press # to send the number through immediately on line1
08:06.29[av]banicamonz: # is "enter"
08:06.58camonzso, 119# would inmediately send 119 to the pstn interface
08:07.07camonzwithout asking the * server?
08:07.09[av]banino
08:07.18camonzhmmm
08:07.19[av]baniit dials the extension immediately without timeout
08:07.41camonz:->
08:07.49*** join/#asterisk benjk (n=benjamin@h219-110-052-106.catv01.itscom.jp)
08:07.51camonzgonna try that now :->
08:08.14[av]banion other voip phones you would press "enter", but phones have no enter button
08:08.19[av]baniso spa3k uses # on line1
08:09.24camonzefectively it's a timeout issue
08:09.33camonzwhen pressin # the redirect it's inmediate
08:09.38camonz*pressing
08:09.38[av]banion the spa3k line1 partly
08:09.48[av]banithe other delay is dialing out pstn
08:09.51camonzbut the timeout value it's .5
08:10.12[av]banibut you can change the dtmf dialing speed with PSTN Dial Digit Len
08:10.22camonzthere's still a small delay when pressing # but like .5s
08:10.36[av]banithats the delay to setup the call
08:11.36[av]banitheres a way with voip to match extensions immediately, but the spa3k doesnt support it
08:11.37camonzand pstn dialing delay
08:11.42[av]baniits called overlap dialing
08:12.00[av]baniso with spa3k you have the digit timeout or press #
08:12.20camonzi reduced that value to .1
08:12.23*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
08:12.30camonzyep. exactly what * does on the dialplan
08:13.08camonzwhen matching an extension on same level priorities
08:13.28[av]banifor moh, it should 'just work', you just set a default moh in *
08:13.39[av]banithen you flash the hook, it should give you stutter tone and the caller is put on hold
08:14.37[av]baniif you hangup, the spa3k will 'quick ring' every few seconds to remind you a call is on hold
08:15.14camonzthe .1 thingie completely broke my connection with the outside :->
08:15.37camonzi've been trying to play several mp3s with MP3Player
08:15.41thazzaHey anyone here gotten a Sipura SPA-3000 to work correctly?
08:16.25camonzi'm kind of at it
08:17.05thazzaI am having trouble with asterisk answering the call. and all i have in the dialplan for the pstn side is exten = s,1,Dial(SIP/210,60,tr)
08:17.12[av]banicamonz: does your moh work at all? eg can you setup a test extension 7777,1,WaitMusicOnHold(60) ?
08:17.15thazzaAny reason why asterisk should be picking up the call?
08:17.19[av]baniand then dial 7777 from line1 and it plays
08:18.02[av]bani7777# if you dont want to wait :)
08:19.06camonzgonna try that now, as soon as i find a right value for pstn dialing delay
08:19.11camonz.1 and .25 is too low
08:19.37*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
08:19.58[av]banipstn dialing delay: 1
08:20.12[av]banii think .5 is the fastest any LEC can handle
08:22.58camonz.5 is still to low, gonna put it back in 1
08:23.29camonzbtw i tried the MoH with a softphone just a min ago and got no sound
08:25.42[av]banithen its not the spa3k :)
08:25.45[av]baniyou need a timer source
08:25.46camonzno sound when tested MoH from line1
08:26.03camonz?
08:26.09[av]baniztdummy
08:26.14[av]banimoh wants a timer
08:26.57camonzis that necessary for MP3Player as well
08:27.10[av]baniyes, for all moh
08:27.50camonz<PROTECTED>
08:27.51camonzFeb 19 04:27:24 NOTICE[10742]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0
08:28.00camonzthat's the prob i get from using mp3player
08:28.28camonzbtw, i've never seen this posted anywhere, but how do i uninstall *
08:28.37camonzto do a clean install of a newer version
08:29.58[av]banidont use mp3player
08:30.05*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
08:30.18[av]banieasiest way really is to use format_mp3 from asterisk-addons
08:30.59camonzand MoH, i've already installed format_mp3
08:31.27[av]banithen change to mode=files in musiconhold.conf
08:31.32[av]baniand restart asterisk
08:31.40camonzcould it be due that the mp3 is oversampled, i think it's a 44100hz, stereo, 192kbps
08:31.44[av]banino
08:32.11*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
08:33.13camonzhow do i install ztdummy?
08:33.52[av]banizaptel package
08:33.59wasimcamonz: edit makefile
08:34.00[av]banimake; make install; modprobe ztdummy
08:34.06[av]banirestart asterisk
08:34.10[av]baniand enjoy moh
08:34.25camonzi thought i needed to change something in the makefile for zaptel
08:34.40[av]banino
08:34.56Abydos313no make, just make install
08:35.15[av]baniasterisk: live the adventure
08:35.35camonzlinux:/usr/src/zaptel-1.2.3 # modprobe ztdummy
08:35.35camonzNotice: Configuration file is /etc/zaptel.conf
08:35.35camonzline 0: Unable to open master device '/dev/zap/ctl'
08:35.35camonz1 error(s) detected
08:35.35camonzFATAL: Error running install command for ztdummy
08:35.50camonz:->
08:37.12*** join/#asterisk colle (n=colle@c-fb50e253.021-27-6e796b10.cust.bredbandsbolaget.se)
08:37.38*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-94.claranet.co.uk)
08:43.38camonznedeed to change the makefile
08:44.02wasimtoldya
08:44.59*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
08:45.50camonz:->
08:45.53camonzbut still no sound
08:46.15camonzinstalled ztdummy and set mode=files in general context
08:46.31trixterturn your volume up
08:47.07trixterare your speakers actually plugged in?
08:47.13trixterthat often creates a one way audio path
08:47.51camonzyep
08:48.44camonzfinally!!!!!!!!!!!!!!!!!!!!11
08:48.50camonzhad to reload moh on the cli
08:49.18camonzdeleted on file on the mohmp3 dir and it was still trying to play it
08:54.43*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
08:57.29*** join/#asterisk stoffell (n=stoffell@d51A4D720.access.telenet.be)
08:58.35camonzFeb 19 04:58:26 WARNING[13534]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443302 ??
08:58.49*** join/#asterisk littleball (n=littleba@cm140.epsilon174.maxonline.com.sg)
09:00.27camonzhttp://lame.sourceforge.net/download/riaa.html
09:00.30camonzlol
09:02.54littleballhello, any relation between SIP express router and asterisk?
09:03.19X-RobNo, none at all.
09:03.48littleballasterisk also support SIP. then what is the difference?
09:04.25stoffelllittleball, all the rest ;)
09:04.45littleballstoffell, sorry?
09:04.45stoffellgood morning X-Rob ;)
09:05.11X-RobEvening, stoffell
09:06.24stoffelllittleball, SER does not handle connection to pstn and stuff, * does almost everything
09:07.40littleballstoffell, i want to know the advantage and disadvantage. Currently, i only use ZAP E1 of asterisk. i am planing to provide SIP services also. Can you give me some suggestions ?
09:07.44*** join/#asterisk freezer (i=freeze@66.29.46.127)
09:07.58freezerhi
09:08.07coppiceX-Rob: untrue. there is a clear relation - they doth support SIP, though only one does it properly :-)
09:08.22X-Robcoppice, and asterisk is _not_ the one that does it properly 8)
09:08.50*** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
09:08.54*** join/#asterisk mko-025 (n=korpim@p54988A82.dip0.t-ipconnect.de)
09:08.54CrashHDHello
09:08.59stoffelllittleball, depends on your needs, have a look on their websites, probably * can do it all
09:09.00littleballX-Rob, does it mean i had better to use SER for SIP service?
09:09.31coppicelittleball: that depends entirely on what you are trying to do
09:09.52CrashHDI can't figure out how to send dnis digits to a locally connected pbx (pri cross cable) from my * box. Can anyone help?
09:09.56littleballcoppice, then what do you mean only one does it properly? which points ?
09:11.06coppiceif you want to interwork with the PSTN, SER will get you nowhere. if you want many other things * can do, SER will get you nowhere. if you only want what SER can do, its probably better for you than *
09:12.00littleballanother question, in asterisk, can i get the callerID without answer the call?
09:12.32CrashHDheh
09:12.39CrashHDlittleball I believe the answer is yes
09:12.58remissthe answer is yes, just don't ask how
09:13.05CrashHDHAH
09:13.09CrashHDisn't that the truth remiess
09:13.31remissi thought it worked automagically... :-/
09:13.40coppicelittleball: for most PSTN protocols, the answer is yes. It really has little to do with *. its all to do with the protocol
09:14.05remissi have a question
09:14.10remissanyone here listen to herman düne?
09:14.16littleballcoppice, the problem is that asterisk always try to answer the call even the system doesn't answer explicitly. (E1/PRI)
09:14.21CrashHDnever heard of that remiss
09:14.49remissthe broken family band?
09:14.58CrashHDdoesn't ring a bell
09:15.23remissaight
09:15.42coppicelittleball: * won't answer until it looks things up and decides it should
09:16.10littleballcoppice, let me try it out
09:16.58CrashHDI know this is lame
09:16.59CrashHDbut
09:17.02CrashHDI'm desperate
09:17.13CrashHD100$ via paypal in it for the guy who figures out my issue
09:17.16X-Roblittleball, if you're using AMP, turn of 'Answer Immediately' in the trunk setting.
09:17.17CrashHDwith *
09:17.27X-RobCrashHD, what's the problem?
09:17.38CrashHDheh
09:17.46IronHelixwhats the issue?
09:17.47[av]bani\o/
09:17.50camonz:->
09:17.56[av]banicamonz: moh working now?
09:17.59camonzyep
09:18.03[av]bani/o/
09:18.04[av]bani\o\
09:18.06IronHelix(offering $100 will get a lot of helpers to pop out of the woodwork very quickly) :)0
09:18.09camonzhad to change the makefile
09:18.32CrashHDI have an altigen PBX connected to my * box via T1/PRI card. I need to be able to send dnis digits from the * to the altigen box.
09:18.32camonzstill some issues with not properly formatted mp3s,
09:18.43CrashHDI can't find a documented solution
09:18.57CrashHDdial(Zap/g#/#######) doesn't work
09:18.58[av]banicamonz: you dont send dnis, you receive it
09:18.59X-RobIsn't it just 'Set(CALLERID(..))' befor eyou dial?
09:19.12CrashHDI don't need to send caller id
09:19.15camonz[av]bani: Feb 19 04:58:26 WARNING[13534]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443302
09:19.17[av]baniX-Rob: dnis is the dialed number, not ani
09:19.21X-RobOh, there you go
09:19.32IronHelixcrash
09:19.32CrashHDI want the * to become our local trunk machine
09:19.34[av]baniX-Rob: dnis = ${EXTEN}
09:19.37IronHelixso you want to send dtmf tones?
09:19.51CrashHDwell iron I don't know how dnis works
09:20.01CrashHDI thought dnis was digitally sent
09:20.01[av]banidnis is sent in the signalling along with ani
09:20.03X-RobCrashHD, I'm guessing you've got pridialplan set incorrectly
09:20.11IronHelixhmmm
09:20.11IronHelixbrb
09:20.13CrashHDI have it set to unknown
09:20.14X-Robin /etc/zapte.conf
09:20.34CrashHDpridialplan is in /etc/asterisk/zapata.conf
09:20.37X-Robtried 'private'?
09:20.43X-Robyeah, yeah, I know 8)
09:20.52CrashHDwhat is the technical difference between them?
09:21.04X-Robdon't know.
09:21.07CrashHDlol
09:21.08X-Robas I said, I know nothing about T1's
09:21.16X-RobI'm offering you suggestions as to what to try
09:21.19CrashHDahh
09:21.21CrashHDthank you Rob
09:21.28CrashHDI've spent about 20 hours in this chair
09:21.32CrashHDtrying different things
09:21.37X-Robif you've got sync on the line, and calls aren't going through, then it's  something in your zapata.conf
09:21.46CrashHDcalls go through just fine
09:22.03CrashHDthe other system just acts as if it isn't getting any dnis digits
09:22.22X-Robuh, and how exactly is that?
09:22.29[av]baniCrashHD: you don't use dial() ?
09:22.54CrashHDbani: I've tried dial(Zap/g#/########)
09:23.01CrashHDthe call goes to the system
09:23.03CrashHDbut no dnis is sent
09:23.14[av]baniCrashHD: the altigen should use ###### as the dnis...
09:23.20X-RobCrashHD, you mean, no dnis is _received by the pbx_
09:23.23[av]banibecause you dialed it...
09:23.34CrashHDno dnis is received by the altigen correct
09:23.45X-RobSo
09:23.52X-Robpri debug span x intense
09:23.53X-Robdo a call
09:23.55X-Robpastebin it
09:23.58CrashHDI don't think dnis = dtmf (zap/g1/#####)?
09:24.12CrashHDok brb
09:24.22X-Roband paste your zaptel and zapata conf too
09:24.22[av]banihmmm... dnid is what * calls it
09:24.41X-Rob[av]bani, 'the number wot is sent to the uvva end of da cable'
09:24.46X-Rob8)
09:25.07IronHelixlol
09:25.09[av]baniX-Rob: if he's forwarding inbound calls through * to pbx ...
09:25.11[av]banisure
09:25.30[av]baniif he's originating new calls from * (say from voip phone) ...
09:25.53[av]baniinbound from PSTN anyway
09:25.54CrashHDthe scenario is I have inbound calls coming from a voip provider that is sending me dnis info
09:26.05[av]baniyeah, you'd want it then
09:26.07CrashHDI need to forward those voip calls to the altigen pbx over pri
09:26.49CrashHDwhat is interesting is
09:26.58CrashHDI think I have things setup right this time around
09:27.03CrashHDand the pbx won't accept the call
09:27.30robin_zOK, I have an SIP account with a provider for outgoing ... somehow in my dialplan, I want to route calls over that, if its not already in use .. and over a different one if it is in use ...
09:27.33robin_zclues?
09:27.43[av]baniCrashHD: have you printed DNIS to make sure its being set before your Dial() ?
09:27.48IronHelixrobin
09:27.53robin_zdude!
09:27.53IronHelixif chanisavail?
09:28.07CrashHDin * the dnis is just the extension number
09:28.15CrashHDplus in the pri debug I see that it is set
09:28.22CrashHD(I think anyway)
09:28.33[av]baniCrashHD: the dnis from the voip provider.. make sure it is still set when you dial(zap/blabla)
09:28.34robin_znot familiar with that .. ok, Il grep voi-info fo rhtat command, thanks
09:28.55CrashHD-- Executing Dial("SIP/71.16.179.175-0872ad60", "Zap/g2/6023571561") in new stack
09:28.58CrashHDya it is set
09:29.03CrashHDbeing sent in the dial command
09:29.15CrashHDwhere would the dnis be stored other than that?
09:29.18IronHelixer its something like that, might be part of an expression
09:29.28[av]baniCrashHD: in ${DNIS}
09:29.33IronHelixok maybe its chemicals firing in the brain instead of
09:29.34[av]banior DNID or whatever mangled thing * calls it
09:29.38IronHelixeh wrong past
09:29.38IronHelixe
09:29.55IronHelixhttp://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail  robin
09:30.00[av]banibut really, 6023571561 is your dnis right?
09:30.07[av]banithe pbx should at least pick that up
09:30.43*** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
09:30.46CrashHDlol sorry about that
09:30.57[av]baniif its not picking up the dialed digits... its busted
09:31.02robin_zIronHelix: yeah, that seems to say its not suitable
09:31.10CrashHDbut dnis does not = dialed digits?
09:31.22IronHelixi think you can use chanisavail in an expression
09:31.38IronHelixnm
09:31.40IronHelixyeah it will work
09:31.57robin_z"If you want to use it for limiting simultaneous calls to the peer, it will not work reliably for you. "
09:32.00robin_zso it says ...
09:32.35[av]baniCrashHD: try SetVar(DNID=${EXTEN}) before your Dial(zap/g2/${EXTEN})
09:32.45IronHelixuse ChanIsAvail(firstline&secondline&thirdline,s)    then dial(${AVAILCHAN}/${EXTEN})
09:32.52CrashHDok will try that brb
09:32.58IronHelixthe s command counts it as unavailble if its in use at all
09:33.13robin_zright ...
09:33.21IronHelixstores the first available channel as ${AVAILCHAN}
09:34.24IronHelixso that should do what you want right?  take a list of channels starting with your sip account, and if anybody is using your sip account then fall back to the 2nd choice
09:34.30CrashHDstill doesn't work
09:34.38CrashHDsame result
09:34.59robin_zok, and if the 2nd choice is a ISDN calling group?
09:35.03[av]baniwhy do you need dnis though? doesnt your pbx use the dialed extension?
09:35.30CrashHDI will be using * as a trunking solution
09:35.51CrashHDcombining multiple trunks from various providers and technology to a single pool for our multiple altigen boxes
09:35.57[av]baniok, and?
09:36.01IronHelixhmmm
09:36.09IronHelixthen you wouldnt want s because that will mark it unavail...
09:36.18[av]baniyou usually only need DNIS if you are accepting multiple calls on a single incoming DID #
09:36.24[av]banibut with voip, that wont happen
09:36.29[av]baniits irrelevant
09:36.41CrashHDit is not irrelevant
09:36.47CrashHDI have a single pri connection
09:36.58CrashHDbetween * and altigen
09:37.06CrashHDwhere 100's of numbers will be inbound
09:37.14CrashHDdnis will allow me to route the call
09:37.17CrashHDin the altigen
09:37.30robin_zIronHelix: basically, we a currently routing outgoing over ISDN ... we purchased a single sip account to try it, if its succesful over a week, we'll buy another 10 channels
09:37.30[av]banithe altigen can't route the call based on the dialed number?
09:37.41CrashHDit doesn't seem so
09:37.56IronHelixyou could do that more effectively with call limits robin
09:37.56remisswtf.. anyone but me get into an infite loop when entering asteriskguru.com ?
09:37.58CrashHDdoesn't seem like it support dtmf (dialed digits)
09:38.00[av]banithe only time you need DNIS is routing dialed call from pstn....
09:38.10robin_zIronHelix: call limits?
09:38.18IronHelixit used to be in sip.conf, checking...
09:38.20CrashHDin this scenario I want * to basically become the pstn
09:38.23[av]baniyou have 1-800-333-4444, and you have the phone co. routing 333-4445 and 333-4446 to 333-4444
09:38.34[av]baniand you want to tell what the person dialed
09:38.39[av]banibut with voip, you know already
09:38.45[av]banibecause you're dialing it!
09:38.52CrashHDno bani
09:38.59robin_zIronHelix: see this is where in depth knowledge helps :)
09:39.15CrashHDI have 12 ip trunks
09:39.16IronHelixhehe
09:39.22CrashHDwhere 100 numbers sit on
09:39.36CrashHDany one of those numbers are dialed they hit the * box
09:39.38CrashHDwith dnis info
09:39.43CrashHDtelling me what number they dialed
09:39.49[av]baniok, so your * receives it, fine
09:39.53CrashHDcorrect
09:40.01[av]baniso you dial that number out into the altigen
09:40.01CrashHDI just need to forward that to another pbx
09:40.19IronHelixrobin in the sip.conf entry for your VoIP provider
09:40.27IronHelixuse:  call-limit=1
09:40.28CrashHDthe altigen treats CO trunks and extension trunks differently
09:40.37[av]baniyou receive a call from ip trunk 1, dnis is 1-800-hot-sexx
09:40.45[av]baniso your * box picks up channel 1 of altigen, and dials 1-800-hot-sexx
09:41.03CrashHDthat does not work
09:41.13CrashHDdon't think the altigen supports it
09:41.16[av]banio_O
09:41.33CrashHDso I need dnis
09:41.35[av]banithats no different than routing on dnis. you're routing based on the extension
09:41.47CrashHDit is different in the altigen
09:41.52robin_zright ... and then just include it in the diaplan in the normal way? I tend to put each provider in  its own context in the dialplan, and then include them into a master context so I have control over the order when pattern matching ...
09:41.54[av]baniif the altigen cant route based on extension, its not much of a pbx
09:42.08CrashHDif I can mimic what the CO does on pri lines my problem is fixed
09:42.14CrashHDthat is what I am trying to do with the * box
09:42.15IronHelixyes robin
09:42.21IronHelixbut to provide failover
09:42.34robin_zand if its in use, it will just fall through and the next one will grab it?
09:43.00IronHelixyou can do something like ChanIsAvail(sip/provider&zap/g1)   then dial(${AVAILCHAN}/${EXTEN})
09:43.02[av]baniCrashHD: basically you're trying to use * as a switch, not a pbx
09:43.07CrashHDcorrect
09:43.16robin_zahhh ,, right
09:43.19robin_zthanks dude!
09:43.23IronHelixno prob
09:43.47CrashHDyou know what
09:43.55CrashHDthis is probably an altigen problem
09:44.02CrashHDI think the * is working the way it should
09:44.04IronHelixi was thinking that too
09:44.13IronHelixif zap debug says that its sending dnis
09:44.17IronHelixand the altigen isnt getting it
09:44.28CrashHDwell actually
09:44.32[av]baniCrashHD: is the altigen getting cid?
09:44.37CrashHDya it is
09:44.43CrashHDand I just pulled the logs on the altigen
09:44.50CrashHDan dI see called# indication
09:44.55CrashHDso it is getting the did
09:44.58IronHelixeasy way to test, i assume you have telco -> asterisk -> altigen...  plug telco into altigen and see if that works?
09:44.59[av]baniso the altigen is fucked up
09:45.02CrashHDbut the call is being immediately disconnected
09:45.12[av]baniprobably borked routing inside your altigen
09:45.25CrashHDiron I can plug the altigen into a pri line and it works fine
09:45.35CrashHDbut it won't support inbound VOIP trunks
09:45.43[av]banithe other possibility is it doesnt like the cid
09:45.54[av]banii've crashed commercial PBXes using asterisk
09:45.54CrashHDbani the cid works ok
09:46.05[av]banisending incorrectly formatted cid
09:46.06CrashHDI could prohib
09:46.13CrashHDbut the cid hasn't been the issue
09:46.18IronHelixohh ic so you're using * as a bandaid to add voip to the non-voip altigen
09:46.21CrashHDI always see that it comes through ok
09:46.27[av]baniIronHelix: * seems to be used a lot that way
09:46.32CrashHDIron that is exactly correct
09:46.44CrashHDthe altigen is actually a voip (extensions) pbx
09:46.48[av]baniCrashHD: what does the altigen say the disconnect reason is?
09:46.50CrashHDbut doesn't support Co trunking
09:47.05CrashHDwould that be in the asterisk? or on the altigen?
09:47.10IronHelixaltigen
09:47.25[av]banihow about using * for what it was designe for? a pbx not a switch :)
09:47.32[av]banithrow the altigen out and use *
09:47.36IronHelixagreed
09:47.40CrashHDlol
09:47.45CrashHDcome on guys
09:47.51IronHelixwe kid :)
09:47.55CrashHDlol
09:47.58CrashHDin a perfect world
09:48.07CrashHDI would agree
09:48.08[av]baniyou would take the altigen out back
09:48.13CrashHDHAHAH
09:48.14CrashHDohhh ya
09:48.15[av]baniand club it to death like a baby seal
09:48.18IronHelix'repair' it with a metal pipe...
09:48.19CrashHDthrow it off a damn cliff
09:48.23CrashHDexactly
09:48.25IronHelixseveral times repeatedly...
09:48.27CrashHDsee you guys are telcom people
09:48.29CrashHDyou understand
09:48.38IronHelix'course :)
09:48.51[av]baniCrashHD: the altigen(s) could "disappear"
09:48.58[av]bani"o noes we only have * now"
09:49.06IronHelixwhat a shame.....  (tear)
09:49.14[av]banisee, we are here to help
09:49.15*** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
09:49.17CrashHDshit
09:49.29[av]baniyay?
09:49.32IronHelixcatch up:
09:49.32IronHelix[04:48.50 A] <[av]bani> CrashHD: the altigen(s) could "disappear"
09:49.32IronHelix[04:48.52 A] * Quits: benjk (n=benjamin@h219-110-052-106.catv01.itscom.jp)
09:49.32IronHelix[04:48.54 A] * Quits: CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net) (Excess Flood)
09:49.32IronHelix[04:48.58 A] <[av]bani> "o noes we only have * now"
09:49.33IronHelix[04:49.06 A] <IronHelix> what a shame.....  (tear)
09:49.35IronHelix[04:49.14 A] <[av]bani> see, we are here to help
09:49.37IronHelix[04:49.15 A] * Joins: CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
09:49.46CrashHDheh
09:49.57CrashHDI tried to paste the altigen logs
09:50.02CrashHDit's gibberish to me
09:50.05IronHelixwww.pastebin.ca
09:50.06CrashHDlooks like t1 line code info
09:50.06IronHelixuse that
09:50.08IronHelixuseful+++
09:50.41[av]baniIronHelix: i could point out you should have pastebin'd that irc snippet... hypocrisy++
09:50.49CrashHDhttp://pastebin.ca/42130
09:50.53IronHelixi considered it
09:51.05CrashHDlol
09:51.08[av]banii get chastised when i paste anything over 2 lines here
09:51.14[av]baniso i dont see why you should get away with it
09:51.15CrashHDlife is full of hypocrisy
09:51.24CrashHDlol
09:51.44CrashHDbut having to open a whole other window
09:51.50CrashHDthat mouse click is just too tough
09:51.56CrashHDtoo much effort
09:51.59CrashHD*rolls his eyes*
09:52.13IronHelixhehe sorry bani  /me yells at self
09:53.18[av]banishrug.. looks like the altigen is convinced it was told to hangup
09:53.26remissthere are no letters in the mailbox
09:53.40CrashHDand that confuses me
09:53.45IronHelix--(Status_Indication) Cause = 0x11 (17)
09:53.47IronHelix?
09:53.51*** join/#asterisk tuxinator_linux (n=tuxinato@ip70-190-51-106.ph.ph.cox.net)
09:54.01[av]baniwould have to look that up in altigen docs
09:54.02CrashHDif I take out the dnis routing (from the altigen) for the particular number it goes through to the system
09:54.06CrashHDand the altigen picks it up
09:54.16CrashHD(altigen docs) wouldn't help
09:54.22CrashHDtrust me on that
09:54.32CrashHDnothing in there but point and click how to's
09:54.55IronHelix:\
09:55.45IronHelixhmmm
09:55.46CrashHDno fun
09:55.52IronHelixmaybe its looking for a short DNIS?
09:55.54IronHelixlast for digits?
09:55.55IronHelix*shrug
09:55.59IronHelixi dunno im guess
09:56.00IronHelixing
09:56.00CrashHDI was just thinking that
09:56.08CrashHDhalf way to setting up 4 digits atm
09:56.15CrashHDand upgrading the altigen
09:56.22CrashHDmy life would be easier with 10 at this point
09:56.31CrashHDbut if it will take 4
09:56.38CrashHDit will survive another day
09:57.03IronHelixor
09:57.09IronHelixhow bout random wierdo idea
09:57.16IronHelixswitch isdn signalling type or something
09:57.17[av]baniif thats a q931 cause code, 0x11 is line busy
09:57.19IronHelix*shrug
09:57.34CrashHDI've tried national
09:57.37CrashHDand 5ees
09:57.55CrashHDit's funny though
09:57.56IronHelix:\
09:57.59CrashHDbecause if I go to the altigen
09:58.06CrashHDand remove the entry for that dnis number
09:58.15CrashHD(meaning don't do any routing with that dnis)
09:58.17*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
09:58.23CrashHDthe call rolls into the auto attendant
09:58.26CrashHDjust fine
09:58.36IronHelixhmmm
09:58.39CrashHDand it shows in the logs the right called number
09:58.46[av]baniwhat does * say when dialing?
09:58.50IronHelixmaybe the routing entry is broken?
09:59.22IronHelixTBH this sounds like moreso an altigen problem than an asterisk problem
09:59.32CrashHDiron I agree with you
09:59.37[av]bani****Calling# 9167821300****   ****Called# 6023571561****
09:59.38[av]bani?
09:59.39CrashHDI think * is working perfectly
09:59.54CrashHDthat's the cid and dnis coming through perfectly bani
09:59.59[av]banilooks to me like * working perfectly
10:00.06CrashHDbastard altigen!
10:00.11[av]baniinternal routing issue in altigen
10:00.19CrashHDthink I'm going to cry
10:00.23IronHelixgood luck with that :(
10:00.30CrashHDsee I would have perferred an * problem
10:00.34CrashHDthat I could have fixed
10:00.35[av]baniwhat does * console say when you try to pass the call through? i'm guessing it's getting busy
10:00.36CrashHDlol
10:00.42CrashHDya
10:00.48CrashHDit gets a hangup request
10:00.55CrashHD-- Zap/25-1 answered SIP/71.16.179.175-09c8ad58
10:00.55CrashHD<PROTECTED>
10:00.55CrashHD<PROTECTED>
10:00.57CrashHD3 lines
10:00.58IronHelixyou could see if freenode has an #altigen
10:00.59CrashHDdon't shoot me
10:01.02IronHelixwouldnt count on it though :(
10:01.06IronHelixdont worry i dont care
10:01.10[av]baniok, so the altigen told * to foad
10:01.14CrashHD(was joking)
10:01.18IronHelixbesides its 5am :)
10:01.21CrashHDdunno what foad is
10:01.24CrashHD2 am here
10:01.24X-Rob8pm
10:01.26IronHelixF*** off and die
10:01.31IronHelixCurrent local time is Sunday Feb 19 05:01.31 AM -0500 GMT
10:01.32CrashHDya foad
10:01.34X-Robjust about to start in a night of AMP h4xx0r1ng
10:01.40CrashHDthat's my new favorite acronym
10:01.51IronHelixhehe
10:01.54X-RobCrashHD, PEBKAC is a good one
10:01.57[av]banisomeone should teach the bot foad
10:02.07X-RobProblem Exists Between Keyboard And Chair.
10:02.12CrashHDHAHAHHAHAHAHHA
10:02.25CrashHDrob that is getting put on a big poster for the office
10:02.32CrashHDall the employees can see that
10:02.46[av]banii want to get a big poster of the computer bomb story
10:03.02[av]banihttp://bani.anime.net/computer_bomb.jpg
10:03.12IronHelixbahahahahaha
10:03.15CrashHDlol
10:03.18IronHelixthats the dumbest hting i've ever seen
10:03.36[av]baniwe have that in the office
10:03.41[av]baniand custoemrs always ask if its real
10:03.47CrashHDlol
10:03.56CrashHDyou guys crack me up
10:04.02CrashHDI see myself spending a lot more time in here
10:04.07CrashHDwhat kind of systems do you all use?
10:04.21*** join/#asterisk saftsack (n=saftsack@p54A7E99A.dip.t-dialin.net)
10:04.29saftsackhi
10:04.33[av]banicomputermachines
10:04.40[av]banii use chatrooms on the intarweb
10:04.40CrashHDphone systems
10:04.41CrashHDlol
10:04.42IronHelixall asterisk
10:04.46saftsackis it possible to detect in a linux shell script, if there is a running call with asterisk?
10:04.49IronHelixim mostly a computer tech
10:05.03CrashHDiron, ditto here
10:05.06[av]banii listen to rumors on the internets
10:05.10CrashHDphone systems have been a neccessary evil for me
10:05.13IronHelixsaftsack- you could have it use a manager connection...
10:05.24IronHelixim getting into asterisk phones tho
10:05.25saftsacka manager connection?
10:05.26IronHelixgood market to be in
10:05.44IronHelixtelnet connection to * on funny port, uses simple commands, can do/control anything
10:05.44CrashHDwe have a product called the virtual office
10:05.59[av]bani2006 is the year of *
10:06.01CrashHDwhere we basically have offices ran entirely from our datacenter
10:06.02IronHelixagreed
10:06.03[av]banii can smell it
10:06.08[av]banior at least hear the SIT
10:06.12CrashHDcitrix, phones, etc
10:06.30CrashHDfrom what I gather from asterisk thus far, I'm impressed
10:06.40CrashHDthis is my first tangle with it
10:06.45saftsackthere is a bug so i have to restart asterisk every night. but i dont want to restart it, if there is a running call. so howto detect if theres a running call?
10:07.03X-Robsaftsack, use 'stop gracefully'
10:07.05IronHelixohhh
10:07.07IronHelixthats easy
10:07.07robin_zIronHelix: one last question ;) ... the calls to the ISND group g1 normall have ,B,30 at the end for early B3 etc, but the SIP channel would not want those passed to it would they?
10:07.13X-Robthat'll only shut down when it's empty
10:07.20IronHelixasterisk -rx restart when convenient
10:07.21CrashHDI really liked the php integration with asterisk
10:07.23CrashHDthat was sweet
10:07.38X-Robor yeah, restart when con
10:07.38saftsackthats cool :)
10:07.39saftsackthanks
10:07.52saftsackbut -rx restart wouldnt work because i have my own init script
10:08.25X-Robuh, actually, it's 'stop when convenient'
10:08.29X-Robnot stop gracefully
10:08.49IronHelixrobin, you mean in Dial()?
10:09.03saftsackso i would just do in my script asterisk -r then stop when convenient then reloading the drivers and then /etc/init.d/asterisk start
10:09.07saftsackdoes it work?
10:09.34IronHelixthat works, but it has to wait for asterisk to stop
10:09.35X-Robwhy would you reload the drivers?
10:09.40IronHelixnot just unload drivers and restart
10:10.22IronHelixalso crash
10:10.24IronHelix[04:17.13 A] <CrashHD> 100$ via paypal in it for the guy who figures out my issue
10:10.29IronHelixi figured out your issue:  altigen is broken
10:10.33CrashHDheh
10:10.38[av]baniyou have to split it
10:10.41IronHelixyeah
10:10.43IronHelix50/50
10:10.50IronHelixi kid, i dont really care
10:10.51X-RobI think I should get $5 for trying.
10:11.01CrashHDI don't mind paying up
10:11.04[av]bani$5CDN for X-Rob
10:11.06CrashHDI'll split it
10:11.08[av]banithats about US$0.35
10:11.11CrashHDafter I get it working
10:11.17X-RobWoo, that's like AUD$70!
10:11.21[av]bani!!
10:11.27CrashHDhaha
10:11.27IronHelixheh its cool i dont really deserve it anyway
10:11.28CrashHDhosting paid!
10:11.38IronHelixbani was with you for a lot longer than i was
10:11.41[av]baniX-Rob: a week of kangaroo burgers
10:11.45X-Rob8)
10:11.47X-RobAnyway
10:12.10CrashHDx-rob what language?
10:12.10CrashHDphp?
10:12.27X-RobCrashHD, Asterisk Management Portal is written mainly in PHP yeah.
10:12.34CrashHDsweet
10:12.39CrashHDthat's my favorite language
10:12.43CrashHDeverything should be php
10:12.52X-Robphp is t3h suxx0rz.
10:12.59X-RobI much prefer perl.
10:13.12CrashHDif suxx0rz is the best thing ever then ya
10:13.14CrashHDbooo
10:13.14X-Robbut, pfft, php has a faster startup and is easier for beginners to get into
10:13.17CrashHDboooo
10:13.22[av]banilanguage elitism is as cool as database elitism
10:13.28[av]baniand almost as cool as OS elitism
10:13.33CrashHDperl
10:13.34IronHelixlol
10:13.35CrashHDpffft
10:13.42X-Rob[av]bani, I've spent the entire weekend commiting PHP fixes to AMP
10:13.43[av]banimy OS has a bigger penis than your OS
10:13.48CrashHDHAHAHAHAH
10:14.06[av]baniX-Rob: i've programmed in perl for 10 years, php is a big relief in many ways
10:14.21[av]baniX-Rob: i mean in many ways perl is cooler, but it is also damned annoying
10:14.22CrashHDperl is old school
10:14.26CrashHDphp is the new and improved
10:14.29remissbah.... is there any usable sip-phones for linux?
10:14.32CrashHDas soo as it gets its cli figured out
10:14.35CrashHDit will be perfect
10:14.36*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
10:14.40IronHelix~softphone
10:14.42jbotsomething that should be drug out into the street and shot
10:14.45X-Robhttps://sourceforge.net/mailarchive/forum.php?forum_id=44690, see all those patches? That's me stopping php from whinging about stupid shit that it shouldn't be whinging about.
10:15.09CrashHDheh
10:15.12X-Robuh http:// not https://
10:15.20CrashHDmaybe not the language but the programmers of amp?
10:15.20CrashHD:)
10:15.21[av]baniX-Rob: bah, i spend all my time battling perl internal crashes because i18n is busted
10:15.22CrashHD*laugh*
10:15.36X-Robi18n is difficult.
10:15.50[av]baniX-Rob: whats worse is, their response "we have no idea wtf causes it"... for 2 years now
10:15.52X-Robbut since I'm a simple en_US user, it never bothers me.
10:16.01[av]baniX-Rob: "we hope it will go away in the next rewrite of perl"
10:16.10X-Rob[av]bani, you got a coredump?
10:16.27IronHelixhahaha
10:16.37CrashHDI'm rotflmao over here
10:16.41CrashHDthis is too fun
10:16.43[av]baniX-Rob: yeah, it dies inside perl's expression parser... in widechar code, which shouldnt be called because its en_US ...
10:17.01X-Robwierd.
10:17.02[av]baniX-Rob: afaik theres a couple bugs open on it since 2004...
10:17.14X-Robah well, not my problem.
10:17.22[av]banione of many perl bugs i battle
10:17.25[av]bani(and lose)
10:17.37CrashHDperl is like fighting mircosoft
10:17.40[av]banicpan is also no fun
10:17.42CrashHDthey just don't care to begin with
10:18.31[av]banino, perl cares.. they just dont know wtf is going on 8)
10:18.36CrashHDhaha
10:18.44[av]baniand theyre too busy redesigning the language from scratch to fix it ...
10:18.57CrashHDwell when it's broke you fix it
10:19.00coppiceASN.1 makes perl look like a model of clarity
10:19.06CrashHDwhen it's really broke you throw it away and create something new
10:20.38[av]baniit amuses me how short people's memories are too.
10:20.53[av]bani"php is insecure!"... completely forgetting the legacy that is guestbook.pl
10:21.35[av]baniand formmail.pl 8)
10:21.50*** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com)
10:22.05[av]baniwhich plagues the intarweb TO THIS DAY  :(
10:22.17littleballhello, when a incoming call to asterisk is hangup, is it possible to jump to another context to start another outgoing call?
10:22.32IronHelixyes
10:22.34littleballthrough exten=>h
10:22.38IronHelixput an extension h in the context
10:22.43xorsysdI have a question about MeetMe
10:22.50IronHelixexten => h,1,Dial(etc)
10:22.50tzafrir[av]bani, do an 'apt-cache search formmail' for better perl alternatives
10:23.01[av]banitzafrir: you miss the point completely :)
10:23.03coppicei feel insecure every time I plug in a rj45 :-)
10:23.04IronHelixask away xor
10:23.15littleballIronHelix, then this is one channel or two seperate channels?
10:23.20xorsysdThe application wasn't installed by default
10:23.30IronHelixone, the one you dial out
10:23.45IronHelixit *should* behave as if the script was triggered with a .call file
10:23.47CrashHD55] --(Status_Indication) Progress: Origination address is non ISDN.
10:23.49tzafrirxorsysd, seems like you don't have zaptel installed
10:23.51CrashHDthis is the altigen
10:24.02[av]banicoppice: try 10base2, it's *much* more secure
10:24.06CrashHDdoesn't show in the altigen log that is connected to our CO PRI
10:24.14X-Robbetter clips, you have to push _and_ turn.
10:24.19IronHelixbut keep in mind tho, once THAT hangs up it will loop back to exten => h,1 i think
10:24.26CrashHDany idea why * wouldn come across as non isdn?
10:24.41IronHelixorig 'address'?  hmmm
10:24.48IronHelixmaybe its whining about dnis format?
10:24.48tzafrirBTW: an article in freshmeat about SIP NAT traversal by someone from YATE: http://freshmeat.net/articles/view/2079/
10:24.49IronHelixi dunno
10:24.57coppice[av]bani: no its not. the cable pulls out of the BNC connector quite easily
10:25.08X-RobCrashHD, you know what's going to be funny? I set Set(CALLERID(..)) originally, and you said 'it's not that'.
10:25.15djin_ibHmm, still running into (small) diffulties with Asterisk on Debian. Does anyone here have experience in this combination?
10:25.25CrashHDrob?
10:25.31*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
10:25.46X-Rob"Origination address is non ISDN." - set the caller ID to some something reasonable.
10:25.59xorsysdI knew that was the problem so I recompiled and re-installed ,but it still isn't working
10:26.07CrashHDwhat is considered reasonable?
10:26.10coppicetzafrir: she likes telling everyone how to do NAT
10:26.15X-RobSomething valid.
10:26.21X-Rob'a phone number'
10:26.22CrashHDwhat is the function again?
10:26.38djin_ibtzafrir, hope you can help. safe_asterisk will not start. Getting:
10:26.38X-RobSet(CALLERID(num)=12345678)
10:26.39djin_ibtzafrir
10:26.43djin_ibsorry :)
10:26.52[av]banicoppice: "Origination address is non ISDN." means the connection is not end-to-end isdn, which is probably correct
10:26.53djin_ib/usr/sbin/safe_asterisk: line 47: /dev/tty9: Input/output error
10:27.09[av]banis/coppice/CrashHD/
10:27.20CrashHDhow would that be correct?
10:27.22[av]baniCrashHD: it's voip->*->isdn
10:27.31djin_ibstarting as root and rights on /dev/tty9 are "crw-rw----  1 root root"
10:27.36CrashHDwell the connection from * to altigen is end to end right?
10:28.10tzafrirdjin_ib, frankly safe_asterisk is not throughly tested. Can you try the non-safe_asterisk version and see if it works well?
10:28.13CrashHDX-Rob: so exten => ####,n,Set(CALLERID(num)=1234567)?
10:28.24X-Robis that a valid number?
10:28.33CrashHDthat is the format though right?
10:28.37*** part/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
10:28.43X-Rob(I'm guessing, but from what the other guys have said, that's probably not the solution)
10:28.59*** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
10:28.59[av]banihmm, its because of call progress indicator
10:29.00CrashHDooops
10:29.01X-Rob(I'm guessing, but from what the other guys have said, that's probably not the solution)
10:29.08CrashHDwait
10:29.09djin_ibtzafrir, what do you mean? Just starting asterisk as "asterisk -vvvv . . etc"?
10:29.30camonzwell..., i'm off to sleep
10:29.39camonzit's almost 7am over here
10:29.48CrashHDwhat is the command to wait in irc?
10:30.04CrashHDso I can msg nickserv and join asterisk in one line
10:30.11camonzsee ya all later :->
10:30.12IronHelixmaybe /timer?
10:30.14CrashHDaccounting for the delay it takes nickserv to indentify me
10:30.18IronHelixtimer works if you have mirc
10:30.24CrashHDhmm
10:30.28CrashHDusing trillian
10:30.28IronHelixuse like /timer 1 1 /commandtoperform
10:30.30IronHelixoh
10:30.31IronHelixhmmm
10:30.37IronHelixcuz its not a universal irc command
10:30.58tzafrirdjin_ib, better: /etc/init.d/asterisk debug
10:31.15tzafrirThis way you don't have to remember to add -U asterisk
10:31.37djin_ibtzafrir: bash: /etc/init.d/asterisk: No such file or directory
10:32.02tzafrirhmmm, not everybody installs from the deb, sadly...
10:32.08robin_zIronHelix: sorry, was AFK for a monent .. yes in the dial(${AVAILEXTEN}/${EXTEN}) I would want to pass B,30 to the isdn group, but not to the sip channel right?
10:32.26robin_zAVAILCHAN I meant
10:32.31djin_ibtzafrir, I look for that, but couldn't find a recent version :)
10:32.34tzafrirdjin_ib, how exactly do you start safe_asterisk?
10:32.49djin_ibSimply from the command
10:32.52djin_ibline
10:33.01brookshirehttp://www.backports.org/debian/pool/main/a/asterisk/
10:33.05brookshire1.2.1
10:33.07brookshiredeb
10:33.20IronHelixhmmm, i never seen the B flag on a Dial() and i dont see it in any of the docs...
10:33.26IronHelixwhat exactly does it do?
10:33.31djin_ibHmm, didn't I read something about a pre 1.2.4 memory leak?
10:33.44IronHelixchances are if its something isdn specific then sip will ignore it
10:33.50IronHelixeither way it prolly wont do any harm
10:33.53brookshireyeah.. but that might have been fixed in that
10:33.54robin_zIronHelixDial(${TRUNK}/g1/${EXTEN}/B,30)
10:34.09robin_zerly B3 on the ISND
10:34.09brookshiredfsg means the code has been changed
10:34.15IronHelixahhh
10:34.19CrashHDwhat is that?
10:34.26CrashHDearly b3 on isdn?
10:34.31CrashHDwhat does that mean?
10:34.43robin_zit means your B3 is early
10:34.47robin_z;)
10:35.00CrashHDlol
10:35.07CrashHDI'm having isdn problems it would seem
10:35.10IronHelixDial( ${TRUNK}/g1/${EXTEN}/B )
10:35.12coppicethe early B3 catches the worm
10:35.13CrashHDjust trying to get a grasp
10:35.18IronHelixDial( ${TRUNK}/g1/${EXTEN}/B  ,30)
10:35.24robin_zits to do with the busy, engaed, tones etc .. you actually hear whats on the channel, before the call connects
10:35.40IronHelixare you sure busy is a flag to dial?
10:35.49CrashHDahh ok
10:35.53IronHelixDial( ${TRUNK}/g1/${EXTEN}/B  ,30)
10:35.54IronHelixthere we go
10:36.11IronHelixits not B,30
10:36.18robin_zoh?
10:36.22IronHelixits dial ${EXTEN}/B
10:36.28IronHelix,30, dial for 30sec then give up
10:36.32robin_zyes ..
10:36.39robin_zbut it needs spaces?
10:36.45[av]baniok... its an issue with call progress indicators
10:36.49IronHelixno i was spacing it out to make easier to read
10:36.50[av]baniindications inband vs out of band
10:36.55IronHelixoh
10:36.56IronHelixhmmm
10:36.59IronHelixic the problem
10:37.04IronHelixsip cant deal with /B
10:37.10robin_zright ..
10:37.13IronHelixor at least i doubt it will want it
10:37.19robin_zexactly ...
10:37.25[av]banithe altigen is expecting ot of band indications, which * isnt providing
10:37.50[av]bani* needs to convert sip indications to out of band pri indications for the altigen i guess
10:37.50*** join/#asterisk ToTo (n=ToTo@host97-136.pool875.interbusiness.it)
10:38.11CrashHDbani?
10:38.14[av]banifor call setup supervision
10:38.29CrashHDI'm confused now
10:38.32CrashHDyou talking to me?
10:38.39[av]banijust mumbling in general
10:38.41robin_ztalk to the hand
10:38.44CrashHDlol
10:38.52IronHelixhmmm
10:38.53CrashHDI had priindication=outofband
10:38.54IronHelixrobin try this
10:38.56CrashHDstill didn't work
10:39.25[av]baniCrashHD: my guess is there will be some Dial() magic involved. what that incantation may be, i do not yet know.
10:39.42CrashHDbani those were my thoughts last night
10:39.51CrashHDso I dug through what I could find of the dial function
10:39.57CrashHDnot a whole lot to find
10:39.59CrashHDlol
10:40.12[av]baniCrashHD: tried a call from a local sip phone to the altigen?
10:42.00CrashHDwell
10:42.05IronHelix1: chanisavail(siptrunk,j)   2: dial(siptrunk/${EXTEN})   102: dial(isdnline/number/wierdflags)
10:42.11CrashHDhmm
10:42.13CrashHDno I haven't
10:42.26CrashHDguess I'm trying to stay on track
10:42.27X-RobIronHelix, *SMACK*
10:42.36IronHelix?
10:42.37X-RobNO +101
10:42.44[av]baniCrashHD: what is your dial command for sip trunk?
10:42.47[av]banier
10:42.51[av]banifor the altigen
10:42.56IronHelixlol i didnt say i liked it but it solves the problem really fast
10:43.20X-RobDON'T tell people to do that
10:43.21CrashHDI was just using dial(Zap/g2/${exten}) on the *
10:43.36X-Rob1,Dial(foo)
10:43.47X-Robn,GotoIf(${DIALSTATUS} =...)
10:43.59[av]banitry Zap/gd2/
10:44.03*** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com)
10:44.07xorsysd.clear
10:44.14CrashHDwhat is gd?
10:44.14xorsysdlol, typo
10:44.23CrashHDis my zapatel.conf
10:44.23CrashHDhttp://pastebin.com/562461
10:44.29IronHelixthat works
10:44.52[av]banier g2d i think
10:44.55IronHelixtwo ways to solve the problem
10:44.58IronHelixhis is more elegant
10:45.06IronHelixif you just start by dialing the channel
10:45.10[av]banig2 being group 2, and d meaning originate the call as isdn digital
10:45.11robin_zso dont even test chanisavail?
10:45.23CrashHDdude
10:45.24CrashHDd
10:45.25CrashHDd
10:45.34CrashHDok let me finish this altigen upgrade
10:45.36CrashHDI will try that
10:45.37IronHelixthen use gotoif and a contiditional statement to say 'IF the dial exited with channel unavailable, THEN goto next priority
10:45.39CrashHDthis could be it!
10:45.40IronHelixbrb
10:45.49[av]banio.o
10:45.50robin_zok....
10:45.56robin_zI will try that
10:46.37*** join/#asterisk admin0 (n=admin0@202.52.225.158)
10:47.01littleballIronHelix, when a incoming call to asterisk is hangup, is it possible to jump to another context to start another outgoing call?
10:47.15admin0hi all ..    i want to change sip port from 5060 to something else, but putting "port=" or port in sip.conf is not helping
10:47.16littleballIronHelix, i hope the first call will be terminated
10:47.26IronHelixsure, exten => h,1,Goto(newcontext,newexten,newpriority)
10:47.42IronHelixif you jump in that manner and the new context doesnt have an H then you avoid a loop
10:47.49littleballbasically, the problem is that the incoming call is still alive
10:47.55littleballjumping works.
10:48.14admin0is sip port hardcoded into asterisk ?   it does not seem to read sip.conf at all
10:48.25[av]baniCrashHD: also, you want signalling=pri_cpe
10:48.37admin0is it port or bindport :)
10:49.33littleballwhat i expect is that the originating call will be terminated. and then my system will call back
10:50.07robin_zok, the n,GotoIf(${DIALSTATUS} =...) seems to cause errors, is "..." meant to be somehting else??
10:50.24admin0using bindport helped .. thanks all
10:50.36IronHelixyeah dialstatus=unavail
10:50.44IronHelixnp admin (not that we helped)
10:50.44robin_zright
10:51.00IronHelixball- i assume you are setting up a callback thing for a calling card system
10:51.09littleballyes
10:51.25saftsackhowto write a script that can execute stop when convenient?
10:51.28littleballhow to terminate the originating call?
10:51.33saftsack#!/bin/sh
10:51.34saftsackasterisk -r
10:51.34saftsackstop when convenient
10:51.37saftsackthat doesnt work :()
10:51.46IronHelixasterisk -rx stop when convenient
10:51.48IronHelixall on one line
10:52.08saftsackthanks :)
10:52.34CrashHDI had signalling=pri_cpe
10:52.41CrashHDbut it complained that both systems were set that way
10:52.52CrashHDsince I'm making the * the make shift co
10:52.58CrashHDwouldn't I want it to be the net?
10:53.18saftsack#!/bin/sh
10:53.18saftsackasterisk -rx stop when convenient
10:53.28saftsackIronHelix, is that right? because it doesnt work
10:53.38CrashHDafk for a minute have to sit at the altigen console for this upgrade
10:53.50saftsackNo such command 'stop' (type 'help' for help)
10:53.51saftsackVerbosity is at least 3
10:53.57IronHelixyeah that should be
10:54.08IronHelixmeh, im out for the night
10:54.10IronHelixi need sleep
10:54.11IronHelixCurrent local time is Sunday Feb 19 05:54.10 AM -0500 GMT
10:54.19IronHelixcya all later, good luck Crash with your altigen woes
10:54.25*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
10:54.25saftsackcYa
10:54.35CrashHDlater iron
10:54.36robin_zsigh ... I think X-Rob's answer was incomplete
10:54.39CrashHDthanks for the help
10:54.39IronHelixand remember crash
10:54.51X-RobWhat was my answer, and what was the question?
10:54.51CrashHD*waiting*
10:54.52robin_zexten => _0.,1,dial(sip/8001/${EXTEN})
10:54.53robin_zexten => _0.,n,GotoIf(${DIALSTATUS} = CHANUNAVAIL)
10:55.00robin_zsurely it needs more than that?
10:55.01X-RobOh
10:55.02X-Robsorry
10:55.05X-RobYes it was
10:55.10IronHelixrack rails are really strong, really heavy, and make a really cool noise when you drop them off the side of your building onto uncooperative equipment :)
10:55.12X-RobThat was an example
10:55.13[av]baniCrashHD: if you are serving as a surrogate PRI, you want to be =pri_net
10:55.18[av]banii was reading it backwards
10:55.22coppiceall answers are incomplete, but some are more incomp
10:55.25tzafrirlittleball, asterisk -rx 'stop when convinient'
10:55.29CrashHDheh iron
10:55.33CrashHDoff this pbx goes
10:55.36CrashHDover the damn cliff
10:55.37littleballtzafrir, you misunderstand
10:55.40[av]banisignalling=pri_net
10:55.44CrashHDalthough I think bani might have nailed it
10:55.46[av]baniand zap/g2d/bla
10:55.46X-Robn,GotoIf($[ "${DIALSTATUS}" = "CHANUNAVAIL" ]?unavail)
10:55.50tzafrirnote the '-s . I may have had some typos. You may also use "sopy now"
10:55.51CrashHDmaking the call digital may fix all this
10:55.56X-Robthen
10:56.04IronHelixhope so
10:56.06IronHelixgood luck!
10:56.07X-Robn(unavail),NoOp(is unavailable)
10:56.12CrashHDbani let me get this altigen back up and going
10:56.16CrashHDand we'll try this out
10:56.59robin_zand thats it ?
10:57.06tzafrirhmmm, bristuff have added something for callback, didn't they?
10:57.49CrashHDbani where is this d thing documented?
10:58.11CrashHDI don't see it at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
10:58.21CrashHDso far that is my biggest complaint about asterisk
10:58.32CrashHDno single resource that is reliable for documentation
10:58.35[av]banihttp://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
10:59.20tzafrirCrashHD, try 'show application dial'
10:59.28tzafrirfrom the CLI
10:59.32CrashHDbani I still don't see it on that page
11:00.13robin_zso ..
11:00.14CrashHDlol
11:00.20robin_zsorry to paste .. but here gies ;)
11:00.23CrashHDI swear
11:00.32robin_zexten => _0.,1,dial(sip/8001/${EXTEN})
11:00.32robin_zexten => _0.,n,GotoIf($[ "${DIALSTATUS" = "CHANUNAVAIL"])?unavail)
11:00.32robin_zexten => _0,n(unavail),dial(${TRUNK}/g1/${EXTEN}/B,30)
11:00.34CrashHDI've looked over this same page you just showed me bani 20 or 30 times
11:00.44littleballtzafrir,IronHelix, my question is how to disconnect the incoming call and then the system call callback to the caller? this is a callback system
11:00.46CrashHDahh shit
11:00.47CrashHDthere it is
11:01.13CrashHDIf the letter d follows, and the channel is an ISDN PRI channel, then the channel will be "made digital", whatever that means
11:01.19CrashHDit's things like "whatever that means"
11:01.22CrashHDthat scare me
11:01.31CrashHDwhat is up with comments like that?
11:02.24CrashHDYou may also use the special modifier c to allow for clear channel connections between PRI ports.
11:02.26[av]banibecause the people writing articles for voip-info are tards
11:02.27CrashHDwhat does that mean?
11:02.43CrashHDwhat is a clean channel connection?
11:03.16[av]baniclear channel
11:03.21CrashHD?
11:03.26[av]baniprobably disables echo cancellers
11:04.04[av]baniif you awnt to use it as a data channel
11:04.38CrashHDand before that line
11:04.44CrashHD3 or 4 bullet points up
11:04.51CrashHDthere is another completely different use for c
11:05.03CrashHDIf the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user presses #.
11:05.17CrashHDdigium should fund an asterisk documentation project
11:05.18CrashHDhahahha
11:05.28robin_zok .. and another general * question .. if I set * -cvv I cant see enough verbosity fo watch the call progress .. if I set * -cvvv I cant see the call progress but its lots in piles of shite with phoens registering and hinting and ..
11:05.58freezertry -cvvvvvvvvv
11:06.02littleballhello, how to disconnect the incoming call to the * and then the system call callback to the caller? this is a callback system
11:06.26tzafrirlittleball, you could use the option G
11:06.41robin_zfreezer, err, no .. the trouble is vvv is too verbose, and vv not verbose enough
11:06.47tzafrirand use the caller ID or a code you got from the user
11:06.48littleballtzafrir G? which command
11:06.56tzafrirfor Dial
11:07.10littleballlet me try. tzafrir. thanks
11:07.35*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
11:08.59littleballtzafrir, what does G mean? i am reading doc, it is quite not clear
11:09.19robin_zgrr .. OK, so .. worse than that, how can I initiate two calls from the console huh?
11:09.55*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
11:11.02remissrobin_z: dial twice?
11:11.36saftsackhow can i look after asterisks uptime?
11:12.02stoffellsaftsack: show uptime ?
11:12.11saftsackthanks :)
11:12.57littleballtzafrir, what does G mean? i want to disconnect the incoming call first. Otherwise, the system canot call back to the caller
11:18.47[av]bani...
11:18.56CrashHDstill working on getting altigen going
11:20.25CrashHDHOLY SHIT!
11:20.27CrashHDdude
11:20.32CrashHDyou are the man
11:20.36CrashHDwhat is your paypal address
11:20.41robin_zsaftsack: MWRONG. next
11:20.56saftsackrobin, ?
11:21.05saftsackwhat do you mean? ^^
11:21.05CrashHDI shit you not I have working on this for probably 30 hours in the last 2 days
11:21.11CrashHDI shit you not*
11:21.24CrashHDand you cracked it with a single fucking character
11:21.29robin_zsaftsack: cant dial twice as the console is then "off hook" and your dial command doent dial but is passed to the call
11:21.49CrashHDgod I feel like a retard
11:21.55robin_zoh sorry, that was remiss, not saftsack
11:22.00saftsack^^
11:23.11CrashHDbani message me your paypal address
11:23.11remissrobin_z: i figured that out :p
11:23.15[av]bani:D
11:23.19[av]bani<- genius
11:23.22CrashHDyes you are
11:23.28remissrobin_z: is it possible to put it on hold?
11:23.39robin_zi dont know, is it?
11:23.45remissdon't know :p
11:23.47CrashHDwhy would asterisk not assume that on a pri trunk it shouldn't be digital already?
11:23.56[av]banibecause it doesnt usually matter
11:24.48CrashHDit does matter
11:24.49CrashHDdamnit
11:24.49CrashHDlol
11:24.52CrashHDI proved it!
11:24.56CrashHDwiat
11:24.57CrashHDyou did
11:25.02CrashHDlol
11:25.13CrashHDdude I'm jumping for joy over here
11:26.37CrashHDyou going to send me your paypal address
11:26.39CrashHDor what
11:26.41CrashHDyou deserve this
11:26.58CrashHDand I'm happy to send it
11:28.49*** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net)
11:28.53CrashHDhello?
11:28.59*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
11:28.59CrashHDchannel got quiet all of a sudden
11:30.26brookshireshhhh! everyone is sleeping :)
11:30.30CrashHDlol
11:30.35CrashHDwho needs sleep
11:30.41brookshirei should be in bed
11:30.51CrashHDyou'll get enough sleep when your dead
11:30.52brookshirebut i'm working on a server migration instead :)
11:30.55CrashHDno time to waste now
11:31.01CrashHDbrook you sound a lot like me
11:31.15CrashHDI'm always like "well I should be sleeping but....."
11:31.42brookshireheh
11:31.48brookshirei gave up sleep in college
11:32.09*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
11:32.26brookshirethere was always a test the next day, or a party to goto
11:32.30Qwellhmm
11:32.34CrashHDheh
11:32.35QwellIsn't it like...5:30 there? :p
11:32.51brookshireit's 5:32 here!
11:32.59CrashHD3:30 here
11:33.32brookshiresilly west coast, and it's silly time
11:33.39CrashHDlol
11:34.09brookshireeveryone should know by now, that central time is the only *true* time ;)
11:34.40remissCEST?
11:34.41CrashHDI think that lack of sleep has gone to your brain :)
11:34.48CrashHDGMT is where it's at damnit
11:34.53puppetGMT is the true time
11:34.58brookshire+6 GMT you mean
11:35.01puppetCET/CEST is just confusing
11:35.09CrashHDif I could only spell guinech or whatever
11:35.11CrashHDI'd be set
11:35.24puppetits fun when people get confused why it is +2h and not +1h ;P
11:35.25brookshire-6, whatever
11:35.46Qwellumm...yeah
11:35.48Qwellso, bed :p
11:35.53brookshiregreenwich
11:36.07remissi want summer :(
11:36.19brookshireqwell: i only sleep when i pass out
11:36.26puppetit is summer ;P summer time anyway
11:36.29CrashHDbrookshire I'm with you on that one
11:36.55brookshireand i'm not drunk tonight!
11:36.55Qwellyou need more alcohol then :p
11:36.55CrashHDI only sleep when I'm forced to
11:36.55brookshirehaha.. i could goto the bar
11:36.59coppicewhy is greenwich considered to be in the west, when its neither west nor east? :-)
11:36.59[av]baniwhee
11:37.16[av]banibecause its green
11:37.22remissbah... i have somewhat problems not comparing things to the dialplan :S
11:37.31brookshireit's soooo far west, it's east
11:37.32Qwellbrookshire: at 5:30am?
11:37.40Qwellno 2am cutoff?
11:37.45brookshireqwell: some bars don't close here
11:37.48brookshire24/7 bars
11:37.49Qwellhmm
11:37.54Qwell2am cutoff here ;/
11:38.00brookshireyeah.. most close at 2am
11:38.07brookshirebut a few can stay open
11:38.14CrashHDbrook you know you have your own personal supply at home nayway
11:38.16CrashHDwhy even go to the bar
11:38.18brookshireand right about now they are kicking too :)
11:38.29Qwellyeah...that's too much, heh
11:38.34remisswhere do we go when the booze runs out?
11:38.56Qwellanyhow...I need sleep, heh
11:38.58brookshirei love denver's liquor laws
11:39.02brookshire2 am cut off..
11:39.15Qwellyeah...2am sucks
11:39.16brookshireand 2 am, we will only sell you 3% beer
11:39.16CrashHDis autofallthrough set to yes by default in 1.2
11:39.17remissmore like 2pm in norway
11:39.34Qwellespecially when like...you decide at 1:45 "hey...let's get some beer from the liquor store"
11:39.45Qwell10 minutes later, you finally get around to it...
11:39.48brookshireit's the only place i've ever gotten sobber from drinking more beer
11:39.50Qwellend up there at 2:05...meh
11:40.03QwellThey can't sell ANYTHING after 2am here :(
11:40.10brookshireqwell: all the stores close at like 1am
11:40.11*** join/#asterisk newl (n=newlook@203-59-100-129.dyn.iinet.net.au)
11:40.17brookshirebut only on saturday!
11:40.21Qwellwell, 24 hour gas stations
11:40.28[av]banidennys is 24/7 here
11:40.29Qwellbut they can't sell booze ;/
11:40.31brookshireevery other day it's 24/7
11:40.33brookshirelol
11:40.39Qwell[av]bani: pretty much everywhere, afaik
11:40.43CrashHDdennys!
11:40.45[av]banino booze though
11:40.47CrashHDI love that place
11:40.49CrashHDin there all the time
11:40.55Qwell...sleep time
11:40.55CrashHDonly place I can eat this early/late
11:40.56brookshirewaffle house > denny's
11:41.01Qwellbed > waffle house
11:41.06[av]banidennys is like mcdonalds... mediocre, but its the same everywhere, so there arent any suprises
11:41.09CrashHDqwell you been saying that for 5 minutes
11:41.13QwellCrashHD: shaddup
11:41.16brookshirescatter, covered, and chunked :)
11:41.22CrashHDbani, agreed
11:41.25CrashHDmiddle of the road
11:41.29CrashHDalthough brook
11:41.33CrashHDI've been to waffle house
11:41.37CrashHDin ohio
11:41.41[av]baniits where you go to eat if you arent sure about anywhere else
11:41.43CrashHDI'll take denny's to waffle house any day
11:41.46[av]baniits 'safe'
11:41.59CrashHDand NO GRITS!
11:42.03CrashHDgrits are nasty
11:42.04brookshire:(
11:42.06[av]banilike if you go to bangalore
11:42.10brookshirecheese grits #1
11:42.17[av]banihmm i dunno bout those street vendors.. but mcdonalds...
11:42.21CrashHDanyone from CST is a grits fan
11:42.24CrashHDI don't understand
11:42.41CrashHDI guess they are so used to being stuck in the middle they will settle for something nasty like grits
11:42.41robin_zXrob: that solution does not work for various reasons .. firstly it seems to always choose the SIP channel, even when it already is in use, and it has no fallthrough when congested etc
11:42.41CrashHDlol
11:42.46brookshireit's because grits are awesome :)
11:42.51[av]banipoutine
11:42.55CrashHDboo on the grits
11:43.13remissi really need some help..
11:43.20remisshttp://mcgruff.org/funstuff/cyberbully.htm <-- i just don't get it...
11:43.20CrashHDremiss?
11:43.37CrashHDremiss are you bored?
11:43.38CrashHDlol
11:43.41remisshehe
11:43.48remissi'm trying to clean up my room :p
11:43.57CrashHDbig task I'm sure
11:44.17remissyeah.. i reach out and take every distraction i can get
11:44.23CrashHDdon't blame you
11:44.43CrashHDcleaning is for the maid
11:44.43remissif only i had
11:44.48CrashHDI use to say that
11:44.53*** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com)
11:44.54CrashHDthought it would be expensive
11:44.59CrashHDI shopped it around
11:45.10CrashHDsuprisingly there are people who are willing to work for dirt cheap
11:45.13CrashHDto clean up after me
11:45.16remisshehe
11:45.22CrashHDso I setup a couple webcams in the apartment
11:45.24CrashHDand showed them
11:45.34CrashHDso my stuff wouldn't get stolen
11:45.38CrashHDand they still clean
11:46.12[av]baniyou showed them where the cameras are so they know to avoid them
11:46.15xorsysdI got Zaptel stuff installed, but get errors and warnings when trying to enter a conference... *argh*
11:46.28xorsysdif someone can help, pm me please
11:46.35CrashHDheh
11:46.42CrashHDbani that could be
11:46.52CrashHDbut I'm pretty sure the cameras cover what I need them too
11:47.11[av]banihot maid on maid action
11:47.20CrashHDlol
11:47.25CrashHDthat would be too good to be true
11:47.30CrashHDI'm just happy when they clean
11:47.31CrashHDlol
11:47.43brookshirexorsysd: ztdummy?
11:48.15robin_zCrashHD: another solution would be to pay the cleaning people a fair wage, so they don't need to steal, the money saved on webcams would probably be enough, and they could eat for a month on it.
11:48.55CrashHDrobin sadly enough I actually did the math
11:49.04CrashHDwebcams = x.xx
11:49.10CrashHDfair wage = y.yyy
11:49.17CrashHDI come out on top
11:49.31CrashHDplus it was a cool excuse to install the kick ass camera system I've always wanted
11:49.46CrashHDso it wasn't just too watch them
11:49.47[av]banirobin_z: otoh, some people will be thieves no matter what you pay them
11:49.51[av]banirobin_z: (enron)
11:49.56CrashHDHAH
11:50.00CrashHDso true bani
11:50.10CrashHDit was just the perfect excuse for me
11:50.16robin_zyeah ... but I never believe in screwing hte last drop out of the people who can least afford it
11:50.26CrashHDit's not like that
11:50.34CrashHDI put a request for a quote
11:50.39CrashHDthey gave me their price
11:50.44CrashHDI choose the best one
11:50.46CrashHDdone
11:50.49xorsysd/msg brookshire WARNING[8526]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory
11:50.49CrashHDfair market
11:50.51CrashHDthat's all
11:50.51robin_zbest or cheapest?
11:50.58xorsysdlol
11:51.05CrashHDit ended up being 4th from the cheapest
11:51.12CrashHDout of 13 quotes
11:51.18CrashHD13 or 14 something like that
11:51.31robin_z14 quotes from cleaners ... coo.
11:51.59CrashHDmy condo homeowners association had a list
11:52.04CrashHDwas easy
11:52.08CrashHDjust started making calls
11:52.10CrashHDanyway
11:52.17CrashHDI'm going to go play with this extension.conf
11:52.22CrashHDnow that bani saved my life
11:52.26[av]bani\o/
11:52.34CrashHDbani you got my pm right?
11:52.45robin_zI have a different theory ... I pay people what I think they are worth .. oftne thats more than the "going rate" ... and I give them better conditions of service than the other local firms too.
11:52.46[av]banibah, i hate irssi
11:52.47[av]banijust a mo
11:53.10CrashHDrobin that is a good theory
11:53.13xorsysdERROR[8526]: chan_zap.c:7396 chandup: Unable to dup channel: No such file or directory??
11:53.20robin_zworks commercially too.
11:53.23[av]banirobin_z: i find introducing people to my pals guido and vinny helps keep people in line
11:53.37CrashHDbani I think our theories fall closer together
11:53.50robin_zyou'll never survive in business :)
11:53.57CrashHDlol
11:54.12CrashHDrobin that is an opinion
11:54.23CrashHDand a short sighted one at best
11:54.24[av]banirobin_z: is that a threat?
11:54.28CrashHDbut thank you for sharing it
11:55.10[av]banirobin_z: dat soundz like a threat ta me buddy
11:55.25robin_zwell, yeah ... let me put it this way, I just won a nice big contract, based on having two key staff, who I just poached from two competitors, they came when asked, because im known for being a good employer ...
11:55.47CrashHDI'm happy for your robin
11:55.56CrashHDthat's also a shitty way to get employees
11:56.02robin_zmore carrot, less stick ...
11:56.07[av]banirobin_z: you mean they didnt sign non competes? the other company was stupid :D
11:56.10xorsysdI'm guessing that my problem is that i installed other stuff and I needed to install ztdummy??
11:56.22CrashHDok I'm done with this convo
11:56.27CrashHDrobin you are chick right?
11:56.42robin_ztheres not much non-comptete stuff in the heavy fabrication/welding trade ;)
11:56.49[av]banihaha
11:56.50CrashHDLOL
11:56.54CrashHDahh man
11:56.55CrashHDtoo funny
11:57.02CrashHDwell when you jump into the IT field
11:57.05CrashHDcome talk to me
11:57.07[av]baniwas expecting it to be * or something
11:57.21CrashHDdifferent industries warrant different tactics
11:57.26[av]baniyou go disappoint us with manufacturing crapola
11:57.34[av]baniwhat a letdown
11:57.58robin_zCrashHD: im a 10 years linux admin/ perl/C++ guy .. who just happens to own a metal fabrication comapny ok
11:58.18brookshirec++ ewh!
11:58.21brookshire:(
11:58.39robin_zwell, its OK, ass OOP languages go, but Perl is better
11:59.02[av]baniperl is better than c++ ...
11:59.05[av]banihuh
11:59.17robin_zfoe OOP, yes
11:59.17brookshireany language that can map a '+' to a sql statement.. is a faulty in my opinion, lol
11:59.49robin_zoperator overloading is voluntary
12:00.00brookshirei guess c++ would be okay, taking out the operator overloading, lol
12:00.08robin_zthen you have Java ...
12:00.16[av]baniwrite once, run nowhere?
12:00.21robin_zshrug ...
12:00.30brookshirewell.. you would just have c, lol
12:00.35[av]baniyou are in a maze of twisty little JVMs, all different
12:00.44robin_zO develop on linux, with eclipse, then deploy on various MS and mac platforms ..
12:00.46[av]baniyou might get eaten by an exception
12:00.50robin_znot had trouble yet
12:01.13brookshirebani: that's why you develop for the most compatible one :)
12:01.13robin_zthsi is when Im not welding, obviously ;)
12:01.28CrashHDI'm going afk for a bit to play with dialplans
12:01.30CrashHDttyl
12:01.32brookshirec is more portable than java though
12:01.41robin_zerr ...
12:01.43[av]banic is more portable than anything :|
12:01.48robin_zI wish
12:02.06[av]banithe language is portable, the APIs may not be
12:02.19robin_zdeploying client side apps over the web, java is pretty good
12:02.25[av]banibut go ahead and write asterisk in java...
12:02.37xorsysdlol
12:02.41robin_zwell, that would be .... maaaad
12:02.46robin_zjava is shit slow
12:03.11robin_zbut it has its uses
12:03.15xorsysdthat's why they have a nice distributed computing library
12:03.17brookshiregui java is slow as shit
12:03.33brookshirejava itself.. is faster than perl, php, python, etc
12:03.55[av]baniheh, its funny how people go "wow azureus is a great example of java" err.. if this is great i dont want to know what shit is
12:04.21brookshirebut.. if you've ever tryed swt+java.. you'll see java is not really that slow
12:04.23stoffellhehe
12:04.27[av]banior eclipse.. man is that slow
12:04.34robin_zlovely IDE though
12:04.40robin_zshit slow on Wondows
12:04.44robin_zgreat on Linux
12:04.47[av]baniunusably slow for me... and i have a 2.4ghz amd64 with 2gb ram
12:04.54robin_zrunning doze?
12:05.05[av]banibut then, the x86_64 jvm is all levels of busted
12:05.11[av]banilinux
12:05.18robin_zoh, mines fine on linux
12:05.26[av]baniazureus x86_64 doesnt work at all...
12:05.32[av]banicuz the jvm is goofed
12:05.33robin_zvery very fast .. but unusebly slow on doze
12:05.59brookshireazureus partially uses c libraries
12:06.10[av]banibleh, its shit all over the place for me.. win32, osx, linux..
12:06.10brookshirefor the gui
12:06.15brookshirei could be that
12:06.18[av]baniwaiting for gui redraw...
12:06.19brookshirei = it
12:06.49[av]banicome on, its over 10 years and they are STILL fighting java performance problems
12:07.05[av]bani"yeah, the next java will finally be fast enough to use"
12:07.11[av]banibeen saying that like since forever
12:09.52brookshirealot of java performace problems can be traced back to crappy coding :)
12:11.23*** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk)
12:11.48[av]banithis does not speak well for the language then, since everything is slow
12:12.12brookshirejava is only slower than c/c++
12:12.18brookshirenothing else
12:12.27[av]baniehh... there are languages faster than java
12:12.33brookshirelike?
12:12.36[av]banilua
12:12.45brookshireand who uses that?
12:12.50[av]banilots of people
12:13.17[av]baniits a very popular embedded language, i am considering integrating it into *
12:13.28[av]banii've integrated it into some stuff of mine, its pretty neat
12:13.38brookshirelua is c
12:13.45[av]baniwhat?
12:14.04[av]banijava is c, if you get down to it
12:14.12[av]baniso is python, and perl
12:14.31[av]baniand php
12:14.33brookshireoh.. nm
12:14.35[av]baniso...
12:14.36brookshireLua is implemented as a small library of C functions
12:14.41[av]baniso is java!
12:14.44Dr-Linuxmost of people use PHP with *
12:14.46brookshirethat's what i read
12:14.53[av]banilua is a language
12:15.33[av]baniyou can integrate lua into your application, it has a c api
12:15.39[av]banibut then so does java...
12:15.49brookshirelua is very much live vb :(
12:15.57brookshires/live/like
12:16.03[av]banieh? not at all
12:16.13brookshirefunction factorial(n)
12:16.13brookshire<PROTECTED>
12:16.13brookshire<PROTECTED>
12:16.13brookshire<PROTECTED>
12:16.14brookshire<PROTECTED>
12:16.14brookshire<PROTECTED>
12:16.16brookshire<PROTECTED>
12:16.23[av]baniits closer to pascal
12:16.31brookshiretcl i would say
12:16.36[av]banino, tcl is evil
12:16.38brookshirenow tcl is faster than java, lol
12:17.00brookshireaol uses tcl, can't be *that* bad
12:17.01brookshirelol
12:17.29[av]banithis is like yahoo using lisp right?
12:17.42dpryolisp is great!
12:17.42brookshireyahoo uses lisp?
12:17.44hensemaaah, factorials, the classical example of when NOT to use recursion
12:18.10brookshireheh
12:18.11[av]banii guess the best example of java's failure is that sun has utterly failed to actually do anything with it themselves
12:18.20[av]baniand sun owns java...
12:18.29brookshiresun owns tcl
12:18.46[av]banieven microsoft is writing stuff in C# ... sun? hello sun?
12:19.25brookshireugh.. c# ruined java
12:19.33[av]banino, sun ruined java
12:19.37brookshirethey took what was good out, and over-hyped it
12:19.48[av]banithey micromanaged java and held it back
12:20.01[av]banic# went and addressed the complaints peopl ehad about java
12:20.14brookshirewhat was that? try and catch?
12:20.15[av]baniplus, gnu was actualyl able to implement a free vm...
12:20.17brookshireLOL
12:20.22[av]banisun wont allow that
12:20.45[av]banisun is killing java, they should let it go
12:20.54[av]banimicromanaging it to death
12:21.01brookshirewell.. they have to
12:21.11brookshirethat's why people use java
12:21.12[av]banitheyre going to strangle it
12:21.18*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
12:21.32brookshirethe whole advantage of java is it forces you to code better
12:21.48[av]banidoes it? eclipse and azureus laugh at you
12:21.56[av]banithe flagship examples of java
12:22.12brookshireeclipse and azureus are examples of people taking java.. and using c libraries
12:22.15dpryoIf you've written java, you know it forces you to code better.
12:22.27[av]banino, it forces you to do things the java way
12:22.35brookshirebani: DUH!
12:22.36[av]baniyou want forced to 'code better'? lisp
12:23.25brookshireif java is so horrible.. then why does anyone hardly use c++ anymore?
12:23.32[av]banium.. what?
12:23.34brookshireit's either c or java
12:23.40brookshireor C#
12:23.50brookshirec++ is dying
12:23.56[av]banidoesnt look that way to me
12:24.09[av]banihardly anything uses java... its all c or c++
12:24.21[av]baniwhat stuff are you using right now that is java, on your desktop right now
12:24.28[av]banieclipse and.... ?
12:24.31[av]bani(crickets)
12:24.35brookshirehttp://www.extremetech.com/article2/0%2C1697%2C1657948%2C00.asp
12:24.55[av]banilol 2001... still hasnt happened
12:25.03[av]banii can write any article i want and make any claims i want
12:25.10[av]baniand be considered authoritative?
12:25.43[av]baniso what applications are you using right now, written in java?
12:25.52[av]baniyour irc client?
12:25.56[av]baniasterisk?
12:26.05[av]banikernel? window manger? email client?
12:26.09[av]baniIM?
12:26.10brookshirei have tons of apps written in java
12:26.20[av]baniwhat are you using right now?
12:26.21Dr-Linuxwin98 :P
12:26.27*** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au)
12:26.29brookshirefor what, irc?
12:26.33[av]banianything, in java
12:26.48[av]baniweb browser?
12:27.33brookshirei have netbeans open :)
12:28.09brookshireoh.. and limewire
12:28.14[av]banithat article was written in 2001, predicting the doom of c++
12:28.26[av]banisojava should be dominating destkop applications by now right?
12:28.44[av]baniexcept if you look at the growth since then, its all been C#  :/
12:28.55[av]banisun totally missed the boat
12:29.26[av]banii predict in another 5 years java programmers will be about as much in demand as PDP/11 programmers
12:29.48brookshireand c++ programmers.. even less
12:29.57[av]baninah.
12:30.15[av]banithey were saying that in 1997, didnt happen
12:30.17dpryoThe game industries still embraces c++
12:30.20[av]banisaid it again in 2001... still nope
12:30.51[av]banicant say its grown, but it hasnt declined appreciably either
12:31.05[av]baniif anything, its java going down and c# taking up the slack
12:31.17[av]baninot that i particularly care for either
12:31.17brookshirei don't agree
12:31.27brookshirei think more people are hiring for vb than c#
12:31.36brookshireif anyone hires for m$ at all
12:31.54brookshirethere is no advantage over c# than vb
12:31.58[av]banieh? the jobs i see are for msvc
12:32.12brookshireso why hire for a harder language
12:32.15[av]banior php/perl/apache
12:32.21[av]banino java, no c# jobs
12:32.35nextimeand ruby and python?
12:32.39[av]banihah
12:32.44[av]banibig fat zero there
12:32.53brookshirei see a lot of hiring around here for java
12:33.00[av]banirails is too new
12:33.03brookshireALOT more than php/perl/apache
12:33.20brookshirethey have rails similar stuff now for php
12:33.28brookshirebut who cares.. frameworks like that blow
12:33.29[av]banirails is too gimmicky
12:33.32[av]banitoo constraining
12:33.33brookshireand are hard to maintain
12:33.39[av]baniyou do things rails way or not at all
12:33.53[av]baniwell you can break out of rails but they make it hard and its unmaintainable
12:34.01X-Robbloody hell. I've been coding for the last half hour, and you lot are _still_ on about languages.
12:34.06[av]baniits good for hacking up prototypes, thats about it
12:34.14brookshirewell.. it's unmaintainable by design
12:34.28brookshireyou runs some scripts which generate a framework of code
12:34.36brookshirewhat if that code it generates is buggy?
12:34.40brookshireand has an exploit
12:34.45brookshirebut you've modified it heavily
12:35.12brookshireugh.. i don't even want to think about it
12:35.20brookshireit's better to just use libraries
12:36.48brookshirex-rob: it's #asterisk, what else would we be talking about in here? surely not asterisk, lol
12:37.21brookshireyay! daylight
12:37.28brookshirewatching the sun rise #1
12:37.52*** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk)
12:38.14*** join/#asterisk miller7 (i=none@j242229.upc-j.chello.nl)
12:42.27*** part/#asterisk miller7 (i=none@j242229.upc-j.chello.nl)
12:43.02*** join/#asterisk davidcsi (n=davidcsi@181.Red-88-6-23.staticIP.rima-tde.net)
12:43.48saftsackoliver@oliver ~ $ man cron
12:43.48saftsackNo manual entry for cron
12:43.55saftsack:( i want to add something to my crontab
12:44.15saftsackaaaah :) man crontab is working
12:44.21remiss:p
12:44.45remissman 5 crontab is probably what you want
12:44.56brookshirecrontab -e
12:44.59brookshire:)
12:45.01saftsackNo entry for crontab in section 5 of the manual
12:45.11remissgoogle
12:45.18saftsack;)
12:45.36brookshire#minute hour    mday    month   wday    who     command
12:45.41saftsackthanks :)
12:45.53saftsackso what to do if i want to execute asterisksoftrestart everyday at 4 o clock?
12:46.05remissam or pm?
12:46.25saftsackin the night :) also am, or?
12:46.34brookshire0 4 * * * root asterisksoftrestart
12:46.36remissput a script in /etc/cron.daily/ instead
12:47.06saftsackthanks :)
12:47.13brookshire/etc/cron.daily doesn't exist for every os
12:47.30davidcsimilitary time
12:47.32davidcsino am or pm
12:47.37saftsackdebian:~# ls /etc/cron.daily/
12:47.42saftsackexists here :)
12:47.50brookshire16 = 4pm
12:47.55Dr-Linuxanybody works fine with AGI stuff?
12:48.03saftsackso i mean 4
12:48.09saftsackbecause i want to have 4 am
12:48.15brookshirematt$ ls /etc/cron*
12:48.24davidcsiagi is great
12:48.29brookshire/etc/crontab
12:48.31brookshire:(
12:48.44*** join/#asterisk lorinc (n=ang@caracas-2609.adsl.interware.hu)
12:49.08saftsack25 6    * * *   root    test -x /usr/sbin/anacron || run-parts --report /etc/cron.daily
12:49.12brookshirebut you could create a /etc/cron.daily pretty easy
12:49.27saftsackall cron.daylies are run at 6 o clock?
12:49.38brookshireanacron will run that task on reboot
12:49.45remiss6.25
12:49.46brookshireif you miss the scheduled time
12:49.54brookshireif that is what you want :)
12:50.06saftsackthats a little bit late ^^
12:50.12brookshiresaft: 6:25
12:50.29saftsackthere are many system tasks running in cron.daily. can i just change it to 4 o clock?
12:50.45davidcsibrookshire, use root's cron, it will run even is root is not logged on
12:51.04brookshireyeah.. it should run all the scripts in /etc/cron.daily
12:51.09remisssaftsack: that line... was it from /etc/crontab?
12:51.14saftsackyes it was
12:51.29remissokay.. mine is 02 4 * * * root run-parts /etc/cron.daily
12:51.47brookshiredavidcsi: i know.. haha.. i just don't have an /etc/cron.daily
12:51.50saftsackmaybe i should create a second one for all asterisk issues?
12:51.50brookshirei was just proving my point
12:52.10nextimedavidcsi : why the root one? He can use also the asterisk user cron
12:52.17brookshiresaft: i woudln't use anacron though
12:52.26davidcsibrookshire, so what? do a crontab -e and add your tasks there
12:52.35saftsackbrookshire, what is anacron?
12:52.38brookshiredavid: i do :)
12:52.43saftsackdo you prefer vixiecron?
12:52.44brookshireanacron is like cron
12:52.49brookshirebut if you reboot
12:52.49davidcsithat's it, then
12:53.04brookshireit will run all the scheduled tasks on bootup that you missed
12:53.04saftsackwhat is if i reboot?
12:53.25saftsackthat isnt bad for me
12:53.36brookshirelike.. if the box goes down at like 3:00 am
12:53.37davidcsiwhat? i've never seen that... my cron runs the tasks when it is supposed to
12:53.46brookshireand you have tasks scheduled for 3:30
12:53.53brookshireyou finally get it booted up at 4:00
12:53.55davidcsioh!
12:54.00brookshirecron would not run those tasks
12:54.04brookshireanacron would !
12:54.16brookshireit just depends on how you set it up
12:54.18saftsackyes but why is it bad if anacron does it?
12:54.24brookshireit's probably not
12:54.24davidcsii see your point now
12:54.43remisssaid the blind man
12:55.09remisssorry, i'm just bored....
12:55.39remisskilling dust bunnies isn't as fun as i thought it would be :-/
12:55.44davidcsi... my box never goes down... and if it does, there is a BIOS option which lets you set the state after a power failure... i set it on, and if the power fails when it comes back it starts automatically
12:56.28brookshirewill it reboot on a kernel panic too?
12:56.36*** join/#asterisk pengyong (n=lala@218.93.156.26)
12:56.43davidcsiyou get kernel panic????
12:56.52brookshirei have sometimes ;)
12:56.57brookshireback in the day
12:56.57brookshirelol
12:57.11davidcsiwow! its been years since i've seen one of those!
12:57.13brookshiremainly related to fedora core 1
12:57.16saftsackwhats the best ip phone in regarding the voice quality?
12:57.36brookshiredavid: i know!
12:57.36*** part/#asterisk Hertell (n=Rene@jumbo52.adsl.netsonic.fi)
12:58.07Dr-Linuxsaftsack: 7960
12:58.12brookshireit's still possible though..
12:58.26davidcsiif you get kernel panic, i recommend getting another distro
12:59.12brookshireyeah.. linux doesn't tend to die like that.. it just eventually gets slower after 400 or so days
13:00.15davidcsii have debian boxes which have been up for like 2 years... uninterrupted
13:02.14Dr-Linuxbrookshire: what do you mean? linux gets slower after 400 or so days?
13:02.20Dr-Linuxbrookshire: linux which distro?
13:02.32saftsackDr-Linux, what 7960?
13:02.33brookshirei mean.. a tad slower
13:02.35brookshireLOL
13:02.54saftsackdo you mean the cisco one?
13:02.59Dr-Linuxtad?
13:03.08Dr-LinuxCisco 7960G
13:03.10Dr-Linuxyep
13:03.29brookshireslightly higher loads levels, etc
13:03.47saftsackDr-Linux, but thats very very expensive
13:03.58[av]baniDr-Linux: better than polycom ip601?
13:04.14*** join/#asterisk pengyong (n=lala@218.93.156.26)
13:04.19*** part/#asterisk pengyong (n=lala@218.93.156.26)
13:06.28[av]baniscardinal: polycom ip601 is well regarded voice quality
13:06.35[av]banis/scardinal/softsack/
13:07.14brookshirehah
13:07.15brookshirethat's hot
13:07.34[av]banimeh
13:07.38brookshires/hot/cool/
13:07.54*** join/#asterisk Diogo_Melo (n=Diogo_Me@200.164.174.226)
13:08.17Diogo_MeloHi people, i'm from Brasil. I've been trying to setup my digium te110p to work with an e1 mcf/r2 line, but it doesn't work still. Does anyone could help me?
13:08.33*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
13:09.46Diogo_MeloHi people, i'm from Brasil. I've been trying to setup my digium te110p to work with an e1 mcf/r2 line, but it doesn't work still. Does anyone could help me?
13:10.24*** join/#asterisk backblue (n=moo@87-196-12-97.net.novis.pt)
13:10.25brookshiredid you get the r2 patch?
13:10.32brookshirehttp://www.voip-info.org/wiki/view/Asterisk+MFC+R2
13:11.27*** join/#asterisk pengyong (n=lala@218.93.156.26)
13:11.48*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)
13:14.15Diogo_Meloyes
13:14.27brookshireso what's the problem?
13:14.38Diogo_Melothe digium is working fine but i'm still getting this error msg  R2 prot. err. [2/       2/Group A       /Category req ] cause 32771 - T3 timed out
13:15.06Diogo_Melothis happen when i try to call to the phone number
13:16.02davidcsiDiogo, are you positive the e1 is up and ready to receive calls on the other side? can you ask the other side o send you a call?
13:16.45Diogo_Meloyes, it's working good
13:17.40davidcsican you ask the other side o send you a call?
13:18.11Diogo_Meloi'll call them now
13:18.37Diogo_Melodid you see the log i sent to you privately
13:18.38Diogo_Melo?
13:19.09brookshirei'm really unfamiliar with unicall
13:20.06CrashHDwhat is the new alternative to dials n +101?
13:20.16Diogo_Meloi've called to my telco yesterday and they did all tests and said everything was working fine
13:21.06davidcsithat might be true on layer 1 and layer 2... but 3? which is where calls and sent...
13:21.45brookshireyou can try contacting steve underwood :)
13:21.56brookshirehe wrote the r2 stuff
13:23.38CrashHDso why is dials n+101 feature being phased out? and what is the new recommended method?
13:26.21[TK]D-FenderCrashHD : Look up "stdexten macro" on the WIKI for a decent sample.  Its also in 1.2's sample extensions.conf
13:26.37[TK]D-FenderCrashHD : All based on the dialstatus variable
13:26.47CrashHDij
13:26.49CrashHDok
13:26.53CrashHDI'll take a look, thanks
13:34.10Diogo_Melothanks brook
13:34.35*** join/#asterisk ToTo (n=ToTo@host97-136.pool875.interbusiness.it)
13:40.06*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)
13:48.55*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.138)
13:49.10Kernel_corehi all
13:52.21*** join/#asterisk coppice (n=chatzill@140.203.17.210.dyn.pacific.net.hk)
13:56.46Kernel_coreI running debian , I want to enable H323 , where do I get PWLIB v1.9.0 ?
13:58.00Kernel_coreI am running debian , I want to enable H323 , where do I get PWLIB v1.9.0 ? : )
14:01.41X-Robwe all know it's a google-able answer, so no-one's bothering to answer.
14:01.44X-Robuse the fucking google.
14:02.18Dr-LinuxX-Rob: fucking google? :)
14:02.25X-Rob_fucking_ google I say.
14:02.34remissi wasn't me
14:02.35X-Robimages.google.com with safesurf _off_
14:02.45remissi had absolutely *nothing* to do with it
14:03.11remissbaby, you're so dark inside
14:03.58Kernel_coreX-Rob: didn't find anything useful , tried both google and sourceforge
14:04.33davidcsido you guys know if there is an ipphone outthere on which you could show the call costs as the call progresses?
14:04.38remisshow can you not find something useful when searching on pwlib?
14:04.43sulexKernel_core: wtf! did you tried at least an apt-get search
14:04.50*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com)
14:04.56littleballhello, i try to setup a callback system. the caller call the system, before answering the call, i try to disconnect the incoming call and then  call callback to the caller? any suggestion?
14:04.59Kernel_coresulex: tried apt-get toooooo
14:05.27littleballeven i hangup the incoming call, it still alive
14:05.41littleballand the system always try to bridge the call
14:05.55remisshangup?
14:06.01sulexKernel_core: I have it as first row on a ubuntu, cannot believe you did not find it for debian
14:06.04*** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca)
14:06.16dlublinkHello!
14:06.29dlublinkAnyone in chat??
14:06.32davidcsilittleball, immediate parameter?
14:06.39dlublinknice, people are here
14:06.42Dr-Linuxsulex :@
14:06.48sulexhi Dr
14:07.26Kernel_coresulex: take a look http://pastebin.com/562648
14:07.30littleballdavidcsi, for misscall callback, the system cannot answer the call. I already get all necessary info. But cannot hangup the incoming call because i try to dial out to the caller
14:07.49littleballdavidcsi, what do you mean immediate parameter?
14:08.05davidcsiis your system a zaptel?
14:08.08littleballyes
14:08.13littleballTE411P
14:09.06davidcsii think there is a parameter called "immediate" that answers the call as it comes in... if not, have you set upo a Wait(X) before making the call?
14:09.49*** join/#asterisk oej (n=oej@apollo.webway.se)
14:10.12littleballdavidcsi, it is not "immediate". The call is not answered. when the system detects incoming call, the system get the callerid, and then try to hangup.
14:10.31sulexKernel_core: exactly, your sarge seems supporting libpt-1.8.3, maybe for etch you'll get the 1.9.0
14:10.34davidcsiok, then what happens?
14:10.38littleballand then in exten=>h, jump to another context, and try to make the outggoing call
14:10.49dlublinkHey, I got my asterisk server setup yesterday and I want to test and see if I can accept incoming calls from the internet
14:10.55littleballthe result is that the incoming call is not hanguped
14:10.58dlublinkdoes anyone have a minute to try and dial my server?
14:11.00dlublinkIf so, let me know
14:11.01dlublinkthanks
14:11.30davidcsilittleball, the way i've seen callback is with call files, not on the dialplan
14:12.35littleballdavidcsi, i know callback file. but i try to make it simple. that is to say, i try to put it in dialplan
14:13.21davidcsilittleball, never seen it work with only dialplans...
14:13.30littleballonce the system get the callerid, then hangup the call and then call the caller
14:13.36sulexKernel_core: no luck, answer is 1.9 just through sources
14:13.38Kernel_coresulex: so libpt-1.8.3 is enough ?!
14:13.48Kernel_coresulex: okey
14:13.49sulexKernel_core: i donno, you know :)
14:13.57littleballi try to make it work :-) it should work, right?
14:14.30littleballi don't want to use agi to write the callback file. it should be done easily
14:14.33davidcsii don't think so... as far as i know, once the call up hung up it gets out of the dialplan
14:15.30littleballdavidcsi, no, it is not that case. the system hangup the incoming call immediately once the caller id is got. and then the outgoing call is triggered by exten=>h
14:15.38dlublinkAnyone here able to try and dial my server?
14:15.45dlublinkAll you gotta do is say whether or not it worked :P
14:15.49CrashHDanyone have some better documentation on retrydial? or care to elaborate for me?
14:16.09X-Robdlublink, here's a hint. Paste a SIP or IAX uri and say 'Try to dial this'
14:16.19X-Robsomeone will, rather than having to try to negotiate with you
14:16.24dlublinkAh ok
14:16.28davidcsilittleball, what happens when you try to make the call out?
14:16.55littleballdavidcsi, it works. but the system try to bridge the incoming channel and outgoing channel
14:18.24davidcsiright... but that's because of what i'm telling you... its a thread, so it doesn't care wether the call is up or not, it works only on that channel...
14:18.28dlublinkTry to dial this sip:guest@lublink.ath.cx;foreign
14:18.55littleballdavidcsi, i also think so. i am thinking whether it is possible to bypass this
14:19.32CrashHDhow do you set a global variable in extensions.conf?
14:19.36CrashHDwhat is the syntax?
14:20.20CrashHDoops
14:20.25CrashHDI was setting it in general
14:20.27CrashHDnot globals
14:20.29CrashHDnm
14:20.53davidcsiCrashHD: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
14:21.14X-RobThere you go, you knocked back my codec.
14:21.14littleballdavidcsi, maybe i can put it in agi
14:21.21dlublinkOoops
14:21.22CrashHDcan I send a numerical cadence to the retrydial command? instead of a file reference?
14:21.28davidcsiSetGlobalVar(foo=value)
14:21.34CrashHDthanks david
14:21.41davidcsilittleball, it would be better
14:21.45dlublinkI allow gsm ilbc and I just added ulaw
14:21.52dlublinkany others I should be supporting?
14:21.59littleballdavidcsi, not fastagi, but agi. it maybe work
14:22.09X-Robdavidcsi, that's depreciated I believe. It's now Set(__VARNAME=...)
14:22.33davidcsiin 1.2 yes, but not before that
14:22.39dlublinkTry to dial this sip:guest@lublink.ath.cx;foreign (fixed the codec)
14:22.39davidcsiand you can still use it
14:24.14X-Rob<-- SIP read from 24.201.38.114:5060:
14:24.14X-RobSIP/2.0 404 Not Found
14:24.16X-Robfrom exten => 889,1,Dial(SIP/guest@lublink.ath.cx)
14:24.37dlublinkoh
14:24.41X-Robusually the number goes before the @
14:24.44davidcsirer
14:24.44X-Robrather than 'guest'
14:24.47davidcsibrb
14:24.54dlublinkuh, I didn't know that
14:25.17dlublinkI am not sure than how the config files go for that, let me check that wiki site and I will be back
14:25.18dlublinkthanks
14:26.19dlublinkok, so if I say, for example 6000@lublink.ath.cx, where will it start in my dialplan?
14:26.50CrashHDwith the retrydial how do I just play ringing?
14:27.16CrashHDI'm doing exten   => s,1,RetryDial(ring:${HPBX1}/${ARG1})
14:27.24CrashHDand it keeps seg faulting on me
14:27.26CrashHDany ideas?
14:27.48dlublinkis there a website that would test out a sip url if given one?
14:28.24dlublinkI got it
14:29.05*** join/#asterisk rajo (n=rajo@bfs.cs.uni-sb.de)
14:29.43*** join/#asterisk loick (n=loick@APuteaux-151-1-27-119.w82-124.abo.wanadoo.fr)
14:29.51*** join/#asterisk rajo (n=rajo@bfs.cs.uni-sb.de)
14:30.04*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
14:30.31*** join/#asterisk davidcsi (n=davidcsi@181.Red-88-6-23.staticIP.rima-tde.net)
14:30.45CrashHDRetryDial requires the numerical argument <sleep>
14:30.49CrashHDwhat is that?
14:30.52CrashHDit won't go away
14:31.00CrashHDI have a numerical argument for sleep already
14:32.44dlublinkI defined a context in the general section of sip.conf, to allow foreign users to dial on my server, what do I put before the @ if I want them to go to the main menu? Do I need a guest account?
14:38.52*** join/#asterisk fugitivo (n=ajf@201.255.176.13)
14:39.38[TK]D-Fenderdlublink : you need an exten called "guest" in the context defined in [general] and "allowguest=yes" in there as well.
14:42.36dlublinkgreat
14:42.46dlublinkcan someone dial sip:guest@lublink.ath.cx please?
14:43.51*** join/#asterisk yogsoggoth (n=gnomer@blacksburg-bsr1-24-53-135-82.chvlva.adelphia.net)
14:44.19X-Robdlublink, that's not right. really.
14:44.38X-Robyou have [from-sip-external] (or whatever) with
14:44.43X-Robexten => guest,1,Answer()
14:44.49X-Robexten => guest,2,Dosomethingelse
14:44.49X-Rob?
14:45.09X-RobYou really don't want to do that
14:45.29dlublinkcan someone dial sip:1000@lublink.ath.cx please?
14:45.48dlublinkoops, gotta run
14:45.52X-Robwhat you want is in your sip.conf have one (and only one) type=peer with insecure=very and point that to a context that has exten => s,1,Answer() etc etc
14:46.08dlublinkok thanks
14:46.11dlublinkbye
14:58.08[TK]D-FenderX-Rob : No, the method using [general] and "allowguest=yes" works very well.
14:58.23[TK]D-FenderX-Rob : No need for a generic peer config
14:59.36*** join/#asterisk essaredee (n=srd@cpc1-lich2-0-0-cust257.brhm.cable.ntl.com)
15:00.06essaredeecan you do a gotoif based on say the current time ends in either 0,10,20,30,40,50 or 60 seconds?
15:00.21*** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com)
15:00.34[TK]D-Fenderessaredee : You can by breaking off that portion if the time with Cut
15:00.57essaredeewhat would be the easiest way to do that?
15:01.00xorsysdAnybody have problems with zaptel modules??
15:01.50[TK]D-Fenderessaredee : read how the "time" variables work and learn how to cut off the parts you want.
15:02.04*** join/#asterisk bkw_ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net)
15:02.30essaredee«[TK]D-Fender» know of any handy url that would have that information? or are you speaking in time() stuff?
15:03.39X-Robessaredee, voip-info.org and asterisk command 'show application Cut'
15:04.53*** join/#asterisk svenna_ (n=svenna@p548D14DE.dip0.t-ipconnect.de)
15:05.40xorsysdI was looking at the mailing list trying to find out why I can't load zaptel and ztdummy modules... But couldn't find a solution
15:05.52[TK]D-Fender~wiki
15:05.54essaredeebit confusing that page is
15:06.07[TK]D-Fenderessaredee : the wiki is just fine... read it...
15:06.10*** join/#asterisk RoyK (n=roy@242.80-203-45.nextgentel.com)
15:06.23essaredeei have, but I don't see any relation with time in it :)
15:06.44RoyK<PROTECTED>
15:06.53[TK]D-Fenderxorsysd : You need to modify the make file or something for ZTDUMMY, and it requires an appropriate USB controller.  Have you confirmed both of these things?
15:07.18[TK]D-Fenderessaredee : Cut is the FUNCTION.  then checkout "variables" on the wiki to show what * give you access to
15:07.34xorsysdI'm using 2.6 so I don't need the UHCI usb thingyh
15:07.40xorsysdand I uncommented the ztdummy
15:07.48[TK]D-FenderThen you shouldn't even need ZTDUMMY
15:07.50*** join/#asterisk pardove (n=pardove@217.219.250.24)
15:07.51xorsysdIt built the modules, but I can't modprobe them, error
15:08.01X-Robxorsysd, does it say I/O error?
15:08.08RoyKxorsysd: which 2.6?
15:08.09X-Rob(in dmesg)
15:08.32pardovewhat does this error mean: ERROR[3309]: pbx.c:1184 ast_custom_function_register: Function CDR already registered.
15:08.44X-Rob[TK]D-Fender, yes, he needs ztdummy if he doesn't have zaptel hardware, but he wants to use RTC, not USB.
15:08.45pardovemy box doesn't start
15:09.05RoyKpardove: seems two modules tries to register the CDR function
15:09.18xorsysd2.6.14
15:09.25RoyKpardove: make sure to rm /usr/lib/asterisk/modules/*.so before make install
15:10.05pardovei've just upgraded to latest trunk and got this error while starting...reverting back to last week svn solves the problem, any comments?
15:10.16pardoveRoyK: i'me sure that its done
15:10.36xorsysdWhen I attempt to enter a conference I get this
15:10.43xorsysdFeb 19 03:17:37 WARNING[11362]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory
15:10.43xorsysdFeb 19 03:17:37 ERROR[11362]: chan_zap.c:7396 chandup: Unable to dup channel: No such file or directory
15:10.43xorsysdFeb 19 03:17:37 WARNING[11362]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device
15:10.43xorsysdFeb 19 03:17:37 WARNING[11362]: app_meetme.c:464 build_conf: Unable to open pseudo device
15:11.28LoonaTickxorsysd: Did you insmod the zaptel modules?
15:12.31essaredeeso say I use ${DATETIME} how might i use cut to remove all bits but seconds, then define the remaining (seconds) as something to check
15:13.06xorsysd"invalid module format"
15:13.50LoonaTickxorsysd: Then your modules aren't compiled propery.
15:13.59LoonaTickxorsysd: Try recompiling them.
15:14.18X-Robessaredee, what part of 'show application cut' don't you understand?>
15:14.47xorsysdI've recompiled them several times
15:15.14X-Robxorsysd, then you're loading the wrong module.
15:15.16CrashHDhttp://pastebin.com/562724
15:15.16xorsysdI've upgraded GCC so, should I recompile my kernel?
15:15.20essaredeeI have looked at that but it has led me to a dead end
15:15.35CrashHDsomeone tell me why after the congestion application is ran everything starts all over?
15:15.38CrashHDhttp://pastebin.com/562724
15:15.39X-Robxorsysd, try insmoding the .ko files directly from zaptel
15:15.50xorsysdI just did
15:16.32essaredeeI understand it in principal but don't know how to practically use the function to get what I want
15:17.16LoonaTickxorsysd: Possibly
15:17.29LoonaTickxorsysd: I think your old gcc is also still there
15:17.37LoonaTickyou can try to compile the module with that gcc
15:18.59CrashHDwhy would that config I pastebin'd loop back and start from the exten => _X. all over?
15:19.17CrashHDwouldn't the congestion application stop processing that call?
15:19.25[TK]D-FenderCrashHD : make the last statement you macro executes be a HANGUP.
15:19.35essaredeeguess I'll have to consult someone else on that query, til then
15:19.48xorsysdWhen I finally get this damn thing workin' I'm going to have a drink in celebration
15:19.50[TK]D-Fenderand try using a limited congestion like "Congestion(5)"
15:19.51X-Robessaredee, bye.
15:19.57X-Robcome back when you're willing to try.
15:20.36xorsysdI'm going to write a script to interface with crafty to play chess over the phone :)
15:20.57CrashHDahhh
15:21.02CrashHDso congestion isn't enough
15:21.28xorsysd?
15:22.43CrashHDhmm
15:23.16xorsysdWhat is the "correct" way of removing Asterisk??
15:23.34*** join/#asterisk my007ms (n=my007ms@213.158.171.162)
15:23.39my007mshello all
15:23.55my007msmy asterisk stop loging to mysql server db
15:23.56xorsysdgreeting my007ms
15:24.00my007mswhere i can start from
15:24.19my007msgreeting xorsysd
15:24.47CrashHDthat didn't work
15:25.06CrashHD<PROTECTED>
15:25.06CrashHD<PROTECTED>
15:25.06CrashHD<PROTECTED>
15:25.12CrashHDjust loops back from the beggining again?
15:25.15CrashHDwhy is that?
15:25.47RoyKg2d?
15:26.07X-Robyou have a exten => _.,.. in the dialplan somewhere?
15:26.32my007msi see that my astersik stop loging to mysql server
15:26.38my007mswhere i can start from
15:26.39my007ms?
15:26.52CrashHDhttp://pastebin.com/562724 is my extensions.conf
15:26.54X-Robmy007ms, /var/log/asterisk/full
15:26.56CrashHDI use _X.
15:27.05X-Robyou got an exten => h,..something?
15:27.07CrashHDg2d = group 2 digital call
15:27.20CrashHDno h, anything
15:29.07my007msi see this in my log file cdr_addon_mysql.c: cdr_mysql: SQL command as follows ................
15:29.23my007msbut from where it's get user name and password of mysql server
15:29.52X-Robfrom /etc/asterisk/cdr_mysql.conf
15:31.10CrashHDshouldn't the call fall through
15:31.14CrashHDinstead of repeat?
15:31.23CrashHDsince macro should be returning to n + 1
15:31.25CrashHDnot n
15:31.28CrashHD?
15:33.34my007msafter insert i see this recored Feb 19 04:02:53 DEBUG[1444] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
15:33.44my007msis this mean that insert it right
15:35.57CrashHDwhy is the damn macro repeating
15:35.57CrashHDlol
15:35.59CrashHDstupid *
15:36.08CrashHD* is going to drive me up a wall I swear
15:36.59RoyKwtf is 'digital'?
15:37.07CrashHDisdn digital
15:37.23CrashHDfor some reason
15:37.34CrashHDwell
15:37.39CrashHDif you try to send dnis over that g1
15:37.42CrashHDit won't send
15:37.48CrashHDcorrectly anyway
15:37.54CrashHDit was a problem bani helped me out with
15:38.01CrashHDthe pbx my * is connected to
15:38.14CrashHDwasn't processing the call correctly if the dnis was being sent to it
15:38.32CrashHDit showed the inbound call (from * to other pbx) as non isdn
15:38.41CrashHDbtw
15:38.42CrashHDupdate
15:38.47CrashHDif you exit the macro
15:38.56CrashHDwith hangup or congestion
15:39.05CrashHDthe macro will not go to n + 1
15:39.09CrashHDbut rather just n
15:39.13CrashHDthus repeating itself
15:41.57my007msyes it's insert right
15:42.07my007msbut why cdr not viwe it right
15:42.09my007ms??
15:52.02*** join/#asterisk littlejohn (n=little@host162-8.pool82104.interbusiness.it)
15:52.11*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi6p.dialup.mindspring.com)
15:52.26*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfi6p.dialup.mindspring.com)
16:00.11*** join/#asterisk razu_ (n=razu@217-159-242-106-dsl.est.estpak.ee)
16:01.20*** join/#asterisk saftsack (n=saftsack@p54A7E99A.dip.t-dialin.net)
16:03.50*** part/#asterisk littlejohn (n=little@host162-8.pool82104.interbusiness.it)
16:06.04CrashHDis there an easy way to have a file with a list of numbers and somehow use a template or something to make those all extensions?
16:08.15CrashHDI have 100 or so inbound did's
16:08.20CrashHDI'd like routed the same way
16:08.23CrashHDvarying numbers
16:08.34CrashHDso can't do an easy expression
16:11.17CrashHDwill #include dir/* work?
16:11.23CrashHDwildcard includes?
16:11.29Corydon76-homeNot yet
16:11.40CrashHDok so only real files
16:11.47Corydon76-homeCorrect
16:12.27Corydon76-homeYou can do '#include relative/path/to/file.conf', though
16:12.43*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net)
16:12.52Corydon76-homeor '#include /absolute/path/to/file'
16:13.09CrashHDand relative being from /etc/asterisk correct?
16:13.14Corydon76-homeCorrect
16:13.18CrashHDperfect
16:13.27CrashHDlittle php script will do the trick for the mass numbers
16:13.41CrashHDI would hate to have to write the same extension line for 500 dids
16:13.47CrashHDthat would be crazy
16:14.04Corydon76-homeUh, why don't you use a pattern match?
16:14.11CrashHDthe did's vary
16:14.16Corydon76-homeSo?
16:14.19CrashHDwell
16:14.33CrashHDso out of that 500
16:14.44CrashHD100 are assigned to one client
16:14.48CrashHDanother 100 to another client
16:14.57CrashHDwhich may need to be routed to other pbx's
16:15.09Corydon76-homeYou still haven't said anything that precludes a pattern match
16:15.17CrashHDexample
16:15.27CrashHD6023571560
16:15.34CrashHD6023571561
16:15.38CrashHD2 different clients
16:15.42CrashHDlots of stuff like that
16:15.56CrashHDso I'd be writing a lot of different patterns
16:16.08CrashHDI'd like to be able to just shove a list of numbers in a file
16:16.27CrashHDand have those numbers do a certain thing
16:16.28Corydon76-homeexten => _XXX-XXX-XXXX,1,Goto(${DB(client/${EXTEN})},${EXTEN},1)
16:16.59CrashHDok you lost me
16:17.03CrashHDI'm a * newb
16:17.38Corydon76-homeTake any 10 digit number, look it up in the astdb for the client name, then go to a context of that client name, same extension, first priority
16:18.06Corydon76-homeWith one line, you've separated out each DID into each client
16:18.25Corydon76-homeAll you need then is 500 entries into the astdb
16:18.43davidcsiCorydon-76, usa an agi
16:18.47Corydon76-homeIf you need to change a DID, you make a single entry into astdb, on the fly, no reload necessary
16:18.48CrashHDhow do you access the ast db?
16:19.04CrashHDthat is not something I've read about
16:19.13Corydon76-homeCrashHD: database put client 2345678901 foo
16:19.25Corydon76-homewhere foo is the client name
16:19.41CrashHDcan you point me to some documentation on astdb?
16:19.43Corydon76-homedavidcsi: agi forks.  It's more efficient to use the dialplan
16:20.03Corydon76-homeCrashHD: use the Asterisk command line:  help database
16:20.12davidcsii'm doing about 25,000 minutes/Day using agi, no problem
16:20.38CrashHDanyway to access the asterisk db from outside asterisk?
16:20.50Corydon76-homeNot really, no.
16:20.56CrashHDwouldn't want to connect to command line everytime I needed to add commands
16:21.05CrashHDwouldn't want my technicians to have that access
16:21.21Corydon76-homeThere's the manager interface, too
16:21.32CrashHDagain way over my head
16:21.33CrashHDlol
16:21.55Corydon76-homeCrashHD: are you a programmer?
16:22.30CrashHDphp yes
16:22.32CrashHDperl
16:22.37CrashHDscripting languages mostly
16:22.38*** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com)
16:22.40CrashHDno OO
16:22.43Corydon76-homeOkay, the dialplan is just a language
16:22.50Corydon76-homeTreat it like a language.
16:23.07Corydon76-homeYou'll find there's all sorts of things you can do with the dialplan if you treat it as such
16:24.08CrashHDhmm
16:24.31CrashHDwell I noticed * has php support
16:24.38CrashHDguess I'll read into the interface docs
16:29.11*** join/#asterisk bmg505 (n=leon@dsl-146-45-45.telkomadsl.co.za)
16:29.50*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
16:30.59*** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk)
16:31.38my007msmy asteris work fine and log every thing to DB
16:31.55my007msbut cdr not viwe it any idea
16:31.57my007ms?
16:34.01*** join/#asterisk Teeli (i=Tili@202-133-67-236-dialup.sat.net.pk)
16:37.49my007mshello every one i have something wrong with time
16:38.34my007msthe  time in asterisk server is Sun Feb 19 18:35:38 EET 2006 but when loging it time 2006-02-19 11:36:42
16:38.39my007mshow this come
16:40.21luckyduckdoes anybody know a variable like ${DIALSTATUS}, which i could use in the standard extension failed? i use this stdext to call an agi script, in the case where a call which was initiated by a call file wasnt successfull. i would like to get the reason for that failure
16:41.04luckyduck${DIALSTATUS} doesnt work there. it seems, it only works after one has used the Dial Application
16:45.51*** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net)
16:45.54Teeliluckyduck: Yes ${DIALSTATUS} can be used only after calling Dial Application as it sets the value for it. What are you trying to achieve?
16:46.11*** join/#asterisk cyonics (i=cyon@tx-71-52-77-114.dhcp.sprint-hsd.net)
16:46.24*** join/#asterisk RoyK (n=roy@80.239.107.70)
16:47.27luckyduckTeeli: i would like to save the result in a database, more exactly the reason why it failed (BUSY, NOTANSWERED, and so on)
16:48.08Juggieluckyduck, thats called CDR its dont automatically
16:48.11Juggie*done
16:48.33TeeliJuggie: CDR of Asterisk sucks big time
16:48.53luckyduckif somebody answered the call, i play a defined message, to notify somebody about some kind of failure. i'm trying to build an 'notification system'
16:49.14luckyduckbased on mysql. in the end, i plan to integrate this into nagios
16:49.30Teeliluckyduck: it should work like that. If you cant get it in agi you can always get that value and use SET to set it another variable so that your agi script gets it
16:49.35sivana<PROTECTED>
16:49.35sivana<PROTECTED>
16:49.35sivana<PROTECTED>
16:49.40sivanasorry
16:49.44Teelisecondly, are you executing dial application from AGI itself or from extensions.
16:50.33luckyduckno, i just place a callfile in /var/spool/asterisk/outgoing, where i specify an extension, context etc, but no application
16:50.41Teelioh
16:50.47Teeliyou are not even using dialstatu
16:50.52Teelisorry dial applicaiton
16:50.59luckyducknot directly, that's right
16:51.12Teeliwell for outgoing calls like that, Dial app is not called.
16:51.19luckyduckah, ok
16:51.39cyonicsnewb question: when you make changes to the config files, do you need to restart asterisk?
16:51.56Teeliso no such variable. A nasty way would be to use outgoing for local channel and then use local channel to use Dial application to dial real number
16:52.11Corydon76-homeFor some things, yes.  For most things, a reload to the relevant section is sufficient
16:52.19cyonicsok, thx
16:52.26luckyduckTeeli: will try to go that way. thanks
16:52.53Teeliluckyduck: I have already warned its a messy way, but that's the asterisk way usually
16:52.58Corydon76-homeYou cannot, for example, change signalling on TDM lines without a restart
16:53.08luckyduckTeeli: :)
16:53.14Corydon76-homebut you can change your dialplan and just do an 'extensions reload'
17:09.30*** join/#asterisk astar` (n=astar@ANantes-154-1-51-142.w81-53.abo.wanadoo.fr)
17:09.31*** join/#asterisk denon (i=denon@tor/session/x-4be1d42c5079a13e)
17:09.31*** mode/#asterisk [+o denon] by ChanServ
17:17.14davidcsiis there any soft out there to view current calls via web?
17:20.36RoyKgastman?
17:20.43RoyKnot via web, though.....
17:20.46RoyKdon't think so
17:21.13RoyKwrite a small script to use with the manager interface
17:22.40davidcsiis what i did... just wondering
17:22.46RoyK:)
17:22.58*** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net)
17:23.07davidcsithe manager doesn't show much info though
17:25.56RoyKit doesn't?
17:26.16RoyKI thought you got it all there
17:26.18*** join/#asterisk ToTo (n=ToTo@host95-88.pool8260.interbusiness.it)
17:27.20Insanity5Anyone know what is up with nufone?
17:27.26*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
17:28.19davidcsino, call length is noty there
17:28.48RoyKno?
17:29.03RoyKstrange
17:29.06RoyKSMOP
17:29.20davidcsiin "show channels" or "show channels concise" or "show channels verbose".. no
17:29.43RoyKSMOP
17:33.43jeebusroxorsdoes anyone know the deps for zaptel?
17:34.53RoyKdeps?
17:35.09RoyKnot a whole lot iirc
17:35.18Kattyhi lads.
17:36.15jeebusroxorsRoyK; dependencies...
17:37.58*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
17:43.04*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
17:45.17*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-20-52.buckeyecom.net)
17:48.50*** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder)
17:49.03xbmodder_lappyhey guys
17:49.34xbmodder_lappyHow do I dial multiple extensions at once, as the DIAL command takes one extension
17:51.54*** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com)
17:56.59Kattyanyone up for a little target practice?
17:57.09Kattyi've got just the surgeon
17:58.36Kattyi hope this pain goes away soon
17:58.39Kattyoh
17:59.04iDunno:)
17:59.13iDunnohave you got pain killers?
17:59.19Kattyheh
17:59.26Kattyi had an allergic reaction to my pain killers.
17:59.42Kattyand i couldn't take ibeuprofen until the morning after less i risk drug interaction
18:00.02Kattybut yes, i'm on ibeuprofen now
18:00.29Kattyhave to wait another hour before i take them again
18:01.02iDunno:/
18:01.38iDunnoshouldn't last too long though, right? Back to normality (whatever that is) within a couple of weeks?
18:02.03Kattyi'm not looking forward to being in pain for 2 weeks
18:02.32*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167033017.pppoe-dynamic.nb.aliant.net)
18:03.37xbmodder_lappyKatty, you wouldn't have any morphine?
18:04.20iDunnoKatty: well, no, I'm not sure that anyone looks forward to being in pain... but as long as it works out in the long run it's probably worth while, maybe ;)
18:04.40xbmodder_lappysurgery for what?
18:05.22*** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca)
18:06.19Kattyxbmodder_lappy: no, i don't have morphine at my parent's house
18:06.52Kattyxbmodder_lappy: i just had my wisdom teeth removed. all 4 of them
18:07.47Kattycut out, not pulled
18:08.00Kattyone was impacting on another tooth
18:08.43acidchildO_O
18:08.45acidchildomfg
18:09.18*** join/#asterisk kivi (n=martin@fw01.eventcontrol.se)
18:10.36xbmodder_lappyOuchy.
18:11.13*** join/#asterisk izo (n=izo@193.202.114.43)
18:11.29izoanybody has access to caller name database ?
18:12.17xbmodder_lappyis there like a "_."  wildcard extension?
18:12.43xbmodder_lappyiaxy, irc via an iaxy? ;-)
18:12.46*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
18:12.55PakiPenguinevening
18:14.05iaxylappy, yes but very slow
18:17.20iaxyIs AMP portal worth setting up?
18:17.51iDunno~amp
18:17.52jbothmm... amp is NOT supported here! people using it should join #amportal
18:18.05iDunno(so, erm, no, it's not ;)
18:18.18*** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F4AD6.dip0.t-ipconnect.de)
18:18.27iaxythat must be why they are dead over there.
18:18.31iaxy:-0
18:19.16xbmodder_lappyKatty, how old are you?
18:19.21xbmodder_lappylike ~
18:19.44xbmodder_lappyI heard the younger you are the more it hurts...
18:19.45xbmodder_lappy:-(
18:20.09[ProB]CrazyManhello, I want to connect to an external SIP Server, but my Asterisk box (with an local IP) tells me that RFC1918-IP's are not allowed. coud I tell my Asterisk box to check every 5 minutes the external IP from an dns?
18:20.18xbmodder_lappyhehe
18:20.32xbmodder_lappyCDR?
18:21.04*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
18:21.07*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
18:21.44Kattyxbmodder_lappy: i'm in my early 20s.
18:22.11xbmodder_lappyAt least you get stronger painkiller!
18:22.19Kattyuhh
18:22.27Kattyi spent my first night after surgery with no pain killers
18:22.49Kattywe had to call my dentist because the surgeon wasn't returning his emergency calls
18:23.17Kattyand the pain killers were making me throw up blood.
18:23.58*** join/#asterisk Majestik (n=Majestik@S0106000024c058cc.ed.shawcable.net)
18:24.01iDunnoeep :(
18:24.14iDunnothat's never good :(
18:24.18xbmodder_lappy:-(
18:25.47*** join/#asterisk bbrdrgz (n=alex@p54B03156.dip0.t-ipconnect.de)
18:26.55*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
18:28.34MajestikI've got a weird problem, people can call into my asterisk, but they don't hear ring tone, all they get is dead-air. It's ringing the phones just fine.. the connection is via SIP to the upstream
18:28.41*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
18:29.01Majestikinternal exten calls ring just fine
18:32.06*** part/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
18:33.22andioMajestik: sounds like nat problem with sip.
18:34.18Majestikandio: the phone does ring at my end to my pap2.. it's just that the caller doesn't hear a ring tone
18:35.03andiolike i said, thats typically behavior for firewalling/nat problem. did you already try to use stun with your sip clients?
18:36.04Majestikandio: nope, didn't try that.. I'm confused as how networking would be that specific of a problem.. conversations through the connection are just fine
18:36.52*** join/#asterisk areski (n=areski@45.Red-88-5-214.staticIP.rima-tde.net)
18:36.52andioconversations work in both directions? but just no ringtones?
18:37.02*** join/#asterisk saftsack (n=saftsack@p54A7D964.dip.t-dialin.net)
18:37.11*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
18:37.31saftsackis it possible to get in any extension without doing anything? (i dont mean the s extension)
18:37.40Majestikandio: yup.. I'm just looking at my firewall now though, and I'm getting a bunch of blocked packets..
18:39.02andioMajestik: i still think it could be a nat/router/fw problem. otherwise you can try another codec just to be sure it's not an issue with codec.
18:43.49*** join/#asterisk denon (i=denon@synapse.subneural.net)
18:43.49*** mode/#asterisk [+o denon] by ChanServ
18:48.42Majestikandio: Any chance this could be a screwed up Dial line?
18:50.26*** join/#asterisk paxr0 (n=paxr0@200-126-99-63.bk7-dsl.surnet.cl)
18:51.12saftsackis it right, that there arent any cheap voip telephones with a good audio quality?
18:51.22xbmodder_lappywhat is this:
18:51.24xbmodder_lappyX-Asterisk-HangupCause: Unallocated (unassigned) number
18:54.39*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
18:57.12*** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com)
19:00.49*** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net)
19:01.07andioMajestik: don't know about your dial lines over there.. don't you have some more clients to test it?
19:01.51Majestikandio: just a pap2-na, and a single uplink for service.. it's just a home system
19:02.41MajestikI'm using stdexten for incoming calls
19:02.52andioMajestik: do you have the same problems when doing sip<->sip?
19:03.29firestrmive really got to find another long distance iax termination.. ive been having nothing but trouble with my current one for the last 3 months
19:03.38Majestikinternal? nope, it rings just fine, but that's port to port on my pap2
19:05.10firestrmanyone here using nufone? has it been stable for you lately?
19:05.11*** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-49-16.dialin.kamp-dsl.de)
19:07.29websaefirestrm: how are you doing?
19:07.40websaewe do iax termination
19:07.44websaeand sip
19:18.51*** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net)
19:19.22*** join/#asterisk MikeJ[Laptop] (n=vircuser@71-36-209-237.dlth.qwest.net)
19:22.00*** join/#asterisk Seggy (i=rbutler@tsss.org)
19:29.55*** join/#asterisk tbs_ (n=heste@0x57317642.hrnxx15.adsl-dhcp.tele.dk)
19:35.55*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
19:38.00*** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net)
19:38.34*** join/#asterisk Tamarisk (n=adrian@user-4200.lns4-c10.dsl.pol.co.uk)
19:42.35Tamarisk~wiki
19:43.05Tamariskwiki sip
19:43.27Tamariskwiki <sip>
19:43.33Tamariskduh
19:44.43*** join/#asterisk Pegger (n=peg@pool-68-163-143-28.bos.east.verizon.net)
19:44.55*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
19:45.20*** join/#asterisk darby_t (i=darby_t@djy42.neoplus.adsl.tpnet.pl)
19:47.12*** join/#asterisk astar` (n=astar@ANantes-154-1-57-68.w81-53.abo.wanadoo.fr)
19:48.06*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
19:49.38*** join/#asterisk darby_t (i=darby_t@djy42.neoplus.adsl.tpnet.pl)
19:53.54*** join/#asterisk tuxinator_linux (n=tuxinato@ip70-190-51-106.ph.ph.cox.net)
20:02.24*** part/#asterisk davidcsi (n=davidcsi@181.Red-88-6-23.staticIP.rima-tde.net)
20:06.53*** join/#asterisk MikeJ__ (n=vircuser@71-36-209-237.dlth.qwest.net)
20:08.20*** join/#asterisk svenna_ (n=svenna@p548D33BE.dip0.t-ipconnect.de)
20:12.02*** part/#asterisk Tamarisk (n=adrian@user-4200.lns4-c10.dsl.pol.co.uk)
20:13.26*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
20:13.51vetoanyone remember off the top of their head where to make g.726 work with sipura?
20:16.44vetonm, it's DEPRECATED_G726 in rtp.c
20:20.00*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
20:23.23*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
20:25.28PakiPenguin_anyone here uses gnudialer?
20:25.31*** join/#asterisk CrashHD_ (n=crashhd@70.96.98.254)
20:26.43fugitivoi could, but it only works with mysql and that sucks
20:30.08PakiPenguin_i see
20:30.21PakiPenguin_fugitivo, what do you use for a predictive dialer then?
20:36.19RoyKhttp://www.bbc.co.uk/sn/hottopics/climatechange/
20:36.28*** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net)
20:36.48bdunnIs there anyone here familiar with Polycom 601 phones?
20:40.25ManxPowerbdunn, they are mostly the same as the 50x and 30x phones.
20:40.51bdunnYes... but having a peculiar problem with one.  I can't seem to get the -directory.xml file to load from the FTP site.
20:41.02*** join/#asterisk Samoied (n=Samoied@201.14.157.210)
20:41.12ManxPowerwhat do the log files that phone uploads say?
20:41.46bdunnThe boot log claims that it downloaded the directory.xml file - no error.  However, the directory never shows on the phone.
20:43.28bdunnEver seen that before?
20:45.21jontowbdunn: directory formatted incorrectly?  have transfer logging turned on at the FTP server?
20:46.03bdunnDirectory is a copy of other -directory.xml files.  Same rights, etc.  Transfer logging turned on and it's getting them but then never showing up.
20:46.30bdunnIn the past, I have sometimes had to format the file system on the phone to get it to take the new directory.  I have tried resetting everything on the phone to no avail.
20:47.27[TK]D-Fenderbdunn : don't do a reset.  Modify the file directly, then pull the plug direct. if you reboot It'll try and upload against it IIRC.
20:47.45ManxPowerI seem to recall something similar.  on my 30x and 50x phones a few months ago.  I was never able to get the phone to overwrite the directory the user created using the keys on the phone.
20:47.53*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
20:47.58bdunnAhh.. pull the power cord.. good idea.
20:49.10bdunnI'm not using the 0000000000-directory.xml file - does that make a difference?
20:49.25bdunnI'm using [macaddr]-directory.xml files.
20:50.15*** join/#asterisk HeppyCat (n=unknown@cpe-70-115-78-74.jam.res.rr.com)
20:50.17dlublinkcan someone try calling sip:lublink.ath.cx and try calling extension 6002
20:50.25dlublinkcan someone try calling sip:1000@lublink.ath.cx and try calling extension 6002
20:50.47HeppyCatgood afternoon
20:53.30*** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net)
20:54.24*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
20:55.28dlublinkcan someone try calling sip:1000@lublink.ath.cx and try calling extension 6002
20:55.52HeppyCatdlublink: i havent even been able to get my phone to register...
20:56.13dlublinkyou tried on my server?
20:56.28HeppyCatno, im trying to get my phone to register to my server
20:56.35dlublinkah ok
20:56.37dlublinksip
20:58.32*** join/#asterisk blackgecko (n=blackgec@201.135.135.145)
20:59.58blackgeckohas someone seen areski ??
21:00.08[TK]D-Fenderbdunn : No if you pul the plug it should resync from the mac file
21:00.15PakiPenguin_blackgecko, i have :)
21:00.41bdunnIsn't that what I want it to do?
21:01.11blackgeckoPakiPenguin; do you know where can i find him
21:01.36[TK]D-Fenderbdunn : yes it is
21:01.59[TK]D-Fenderbdunn : I did that about 10 times modding the list on my receptionists IP601 + 2 modules till I got it perferct.
21:02.25bdunnDid it pull in your changes to the -directory.xml every time?
21:03.41[TK]D-Fenderyup
21:04.08bdunnCould you do me a HUGE favor.  To rule out an error in my -directory.xml file for that 601 phone, could you send me your copy?  Please?
21:04.34[TK]D-Fenderbdunn : Sorry, I'm heading out right now, perhaps if I'm on later....
21:04.39[TK]D-Fenderbbl (maybe)
21:14.07*** join/#asterisk ldnblk (n=Just@212.183.136.193)
21:28.16*** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net)
21:29.27*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
21:46.28*** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F41EA.dip0.t-ipconnect.de)
21:47.04[ProB]CrazyManhello, anybody here useing GMX with asterisk ?
21:54.45*** join/#asterisk mrdigital (n=Mrdgitia@pool-68-163-50-110.phil.east.verizon.net)
21:54.50mrdigitalanyone heard of gotvmail?
21:55.39*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
22:05.42*** join/#asterisk burtonez (i=mimx@w201.ljudmila.org)
22:07.19robin_zmeep?
22:07.28robin_zOK this SIP thing is still causing pain
22:07.47HeppyCatthat it is robin_z
22:07.58robin_zthe idea was to set call-limit=1 in the sip.conf for that peer,
22:08.10robin_zdial out on it ..
22:08.27robin_zand when its busy fall-through to the ISDN lines ...
22:08.48robin_zbut even when its in use, it tries to establish a second outgoing .. resulting in pain and suffering
22:08.55robin_zsigh.
22:09.31HeppyCatim just trying to get my phone to register right now :(
22:09.52robin_zexten => _0.,1,dial(sip/8001/${EXTEN})
22:09.52robin_zexten => _0.,n,GotoIf($[ "${DIALSTATUS}" = "CHANUNAVAIL"])?unavail)
22:09.52robin_zexten => _0,n(unavail),dial(${TRUNK}/g1/${EXTEN}/B,30)
22:10.22robin_zim wondering if "CHANUNAVAIL" is the right comparator ...
22:10.27robin_zmaybe CONGESTION?
22:10.57robin_zanyway, goddammit, it shouldnt dial
22:11.04HeppyCatway too advanced for me
22:11.55af_what does callprogress in zapata.conf?
22:13.13FuriousGeorgecan i do math with extension?  exten=exten+1?
22:13.56RoyKSet(EXTEN=$[ ${EXTEN} + 1 ])
22:14.15FuriousGeorgeRoyK: thanks
22:14.27FuriousGeorgei just wasnt sure how it would treat it
22:14.34Corydon76-homeUh, that won't work
22:14.39Corydon76-homeYou can't assign to EXTEN
22:14.45RoyKno...
22:14.57RoyKDial(Local/$[ ${EXTEN} + 1])
22:14.57RoyK:)
22:15.10Corydon76-homeOr just do a Goto
22:15.12robin_zsigh ..
22:15.27robin_zis call-limit a valid entry in a SIP peer configuration
22:15.31robin_z????
22:15.44*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
22:15.53RoyKdeprecated since around 1980
22:15.59Corydon76-homerobin_z: thought it was incominglimit and outgoinglimit
22:16.07robin_zhmmm ...
22:16.38robin_zno .. apparently its all_limit
22:16.55robin_zcall_limit even
22:16.55RoyKrobin_z: use SetGroup
22:17.07RoyK~groups?
22:17.10robin_zSetGroup? what does that do?
22:17.18robin_zexten => _0.,1,dial(sip/8001/${EXTEN})
22:17.18robin_zexten => _0.,n,GotoIf($[ "${DIALSTATUS}" = "CHANUNAVAIL"])?unavail)
22:17.18robin_zexten => _0,n(unavail),dial(${TRUNK}/g1/${EXTEN}/B,30)
22:17.21robin_zoopsy
22:17.30robin_zhttp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+incominglimit
22:17.32robin_zthats better
22:18.01RoyKrobin_z: just Goto(${DIALSTATUS}) instead :)
22:18.17robin_zseems outgoinglimit is replaced by call_limit from * 1.2 onwards
22:18.29robin_zRoyK: now, that wont work will it.
22:19.03robin_zthe problem is its trying to send multiplke calls down a single channel outgoing SIP
22:19.03RoyKof course it will :)
22:19.16robin_ztrust me, it doesnt
22:19.44robin_zplus ... we only have 1 G729 licence ...
22:19.52RoyK:)
22:19.58RoyKthey're SO expensive, you know
22:20.09robin_zbuying two would be a waste
22:20.51robin_zas we only have one outgoing channle capable of using it ..
22:20.54robin_zoh, hes gone
22:25.12ManxPowerI buy a min of 10 G729 licenses anytime we need it at a new site.
22:26.39robin_zthe "anytime we need it" is fine ..
22:26.49robin_zso long as we get this working OK, we'll buy 10 too
22:26.55robin_zmoney to a good cause and all that
22:28.45robin_zsigh .. even setting call_limit=1 still doesnt work .. it just tried to create two calls on that SIP peer
22:28.56kuku5how much is a license
22:29.03robin_z$10
22:29.05robin_zbargain
22:31.04*** join/#asterisk twyant (n=ghandi@68-188-139-162.dhcp.aldl.mi.charter.com)
22:32.46*** join/#asterisk n3glv (n=omega196@monrovll-cuda1-24-53-243-90.pittpa.adelphia.net)
22:35.22n3glvI installed asteriiskathome and can't seem to get softphone sip clients to connect to their extension number, am obviously missing something important
22:36.24n3glvboth server and client are behind consumer nat router
22:37.18n3glvnobody even want to yell RTFM at me?
22:37.24robin_zno
22:37.29FuriousGeorgecan anyone think of a way to make an extension behave differently with a transfered call vs a directly dialed call
22:37.42robin_zif server ands client on on the same lan  behind a NAT, then stop worrying about NAT
22:38.07robin_zsigh ... I set call_limit=1 in the peer in sip .conf
22:38.09robin_zbut ..
22:38.15*** join/#asterisk MikeJ__ (n=vircuser@71-36-209-237.dlth.qwest.net)
22:38.22robin_zCall limit   : 0
22:38.35robin_zin sip show peer <peername>
22:38.42robin_zbah
22:38.43n3glvcan't connect to private ip side or public
22:39.07robin_zerr ...
22:39.10*** join/#asterisk trym (n=trym@194.63.254.6)
22:39.12robin_zwhat does that mean ...
22:39.41FuriousGeorgethere is now way to differentiate in the dialplan between a call that is xferred to an extension and dialing it directly, is there?
22:39.54FuriousGeorge*no way
22:40.38*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
22:42.51*** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home)
22:43.05twyantI was going to come in here and start asking questions but I think I'll wait until I get my hardware later this week so I have a working system to bug you guys about.  :)
22:47.39twyanthow about this, does anyone know anything about asterisk clustering/failover/redundancy?
22:51.06*** join/#asterisk Corydon76-home (i=peuce@pdpc/supporter/sustaining/Corydon76-home)
22:51.19robin_zhmmm ...
22:51.21robin_z} else  if (!strcasecmp(colname, "limit")) {
22:51.21robin_z<PROTECTED>
22:51.44robin_zdoes that imply its using "limit" not "call_limit" in the peer settings in sip.conf?
22:56.10*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
22:56.49*** join/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com)
22:57.58[av]bani\o>
22:58.04*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
22:59.08puzzledevening all
22:59.59*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
23:00.52FuriousGeorge[av]bani: no matter how i slice this i cant make it work just right
23:01.16FuriousGeorgeit turns out you can transfer a person to a meetme room, but not a room to a person
23:01.25FuriousGeorgethat really screws things up
23:02.17robin_zsigh .. call_limit in sip.conf peers seems to be screwed
23:02.52robin_zlooks like the parser is looking for "limit" anyway, but even then, its borked in some way
23:03.15puzzledisn't that what SetGroup is for?
23:03.42robin_zno clue.
23:04.02robin_zsetgroup can limit the number of outgoing calls on a sip channel?
23:04.24rogierCan somebody give me a quick hint here? I'm trying to follow the examples in the O'Reilly book about asterisk.
23:04.34puzzledrobin_z: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
23:04.42FuriousGeorgerogier: what?
23:04.51rogierI'm trying to use a 's' extensions, but the softphone I'm using can't reach it and in the logs I get:
23:05.04rogiersent into invalid extension 's' in context 'incoming', but no invalid handler
23:05.13FuriousGeorgeyou dont call s,  s is where, for instance, an incomming call goes
23:05.27*** join/#asterisk backblue (n=moo@87-196-12-97.net.novis.pt)
23:05.36FuriousGeorgemake sure in incoming somewhere you have exten=>s,1,answer
23:05.47*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
23:05.52rogierI cannot get into that extension with a sofphone that is registered as friend ?
23:06.04FuriousGeorgewhat you mean get into?
23:06.10FuriousGeorgeincoming calls go in s
23:06.18*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
23:06.21FuriousGeorgethen exten=>s,2,dial(sip/you)
23:07.21rogierIs the 's' extensions not intented for people calling in from outsiide? Say PSTN or through voip provider ?
23:07.39robin_zpuzzled: setgroup is deprecated anyway
23:07.40FuriousGeorgeif you want to get into s in incoming, add this to your sip peers context:  exten => 1000,1,goto(incoming,s,1)  now when you dial 1000 you go to s
23:07.56puzzledrobin_z: yes but the link I gave you tells you about the new stuff too
23:07.59FuriousGeorgerogier: it can be for that...  thats the main use
23:08.00mrdigitalanyone heard of GotVmail?
23:08.09rogierForget about that 'not' I put in that sentence..... I mean, is the 's' extension only mentioned for people calling in from outside...
23:08.26FuriousGeorgemostly, yes
23:08.33FuriousGeorgealso in macros
23:08.41kingsleyCan asterisk provide voice mail if only the line from the phone comany's central office is plugged into the (Zaptel) card?
23:09.08rogierOk, thanks. That explains why it doens't trigger anything when I try to call it with a registered softphone.
23:09.18robin_z[uzzled: I think thats all to do with putting callers into groups, I just want to limit the outgoing calls on one SIP peer, to err, 1
23:09.19FuriousGeorgerogier: did you successfully authenticate with you sip provider, or set up your tdm, or whatever you are getting incoming calls right?
23:09.30robin_znot really related I dont think
23:09.38FuriousGeorgeincoming calls are in incoming context, when they ring, s gets called, and then s calls your peers
23:09.45FuriousGeorgepeers are in another context (outgoing?)
23:09.48rogiertdm?
23:09.57FuriousGeorgeits a card you plug phone lines into
23:10.32FuriousGeorgekingsley: can generate voicemail without being connected to the outside world at all
23:11.05rogierAha, I'm not using thath, I'm just connecting up some different vioip providers. On the other side I so far only have  one soft phone. Both the voip service providers  and the soft phone register just fine.
23:11.42rogierI can call  the freeworlddialup echo test through asterisk with my softphone.
23:11.46FuriousGeorgegreat, so your default context for sip can be incoming, and make all your softphones be in context outgoing
23:12.00FuriousGeorgenow when someone calls your sip account, s extension in incoming will happen
23:12.25*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)
23:12.31kingsleyFuriousGeorge: Thank you. Is my understanding correct that it's sufficient to use only one port on the (Zaptel) voice card?
23:12.35FuriousGeorgeyou can specify a different extension (or letter) but why?  when asterisk has no extension (i.e. the line is just ringing) it looks for an s extension in that line's context
23:12.56FuriousGeorgekingsley: i dont see what the zaptel card has to do with voicemail
23:13.08FuriousGeorgeyou can have 0 zaptel cards and generate,send,and receive voicemail
23:13.21rogierFuriousGeorge, aha, can I also use s for that. The exmple in the guide uses my fwd number for that referal I think.
23:14.22rogiereg: .exten => 705773,1,Dial(SIP/rogier)
23:14.28FuriousGeorgesome times that happens, but only if you explicitly tell it to with register=> blah:blah@blah.blah/HERE
23:15.17rogieraha, ok, thanks
23:15.48robin_zpuzzled: like this?? ::
23:15.51robin_zexten => _0.,1,Set(GROUP()=SWITZERNET)               ;Set Group
23:15.51robin_zexten => _0.,2,GotoIf($[${GROUP_COUNT()} > 1 ]?103) ;Exceeded?
23:15.51robin_zexten => _0.,3,dial(sip/8001/${EXTEN})
23:15.51robin_zexten => _0,103,dial(${TRUNK}/g1/${EXTEN}/B,30)
23:16.14puzzledrobin_z: no idea, never played with it
23:16.27FuriousGeorgekingsley: app_voicemail makes voicemail, not the zap card
23:16.44robin_zsigh ....
23:19.21kingsleyFuriousGeorge: Yes, of course you're right. Aside from voice mail, do you happen to know if asterisk will work if only one line is plugged into the card, and that line is shared by a plain old telephone and the line going to the phone company's central office?
23:20.18FuriousGeorgesure, but if someone answers that phone * will have no idea, if someone is on that phone * wont be able to dial
23:21.04FuriousGeorgeits not pretty
23:21.37*** join/#asterisk orlok (n=jwr@202-44-174-4.nexnet.net.au)
23:23.07kingsleySo is my understanding correct that unless asterisk's card is located between the phone and the line going to the central office, asterisk won't work?
23:23.21robin_zpuzzled: that seems to work ..
23:23.56FuriousGeorgewell, actually, if someone is on that phone asterisk will try to dial that line and play the tones if someone on the system tries to call out
23:24.06*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
23:24.09FuriousGeorgeit depends what you mean by work
23:24.31FuriousGeorgeyou can have sip, and aix peers
23:24.37puzzledrobin_z: nice
23:24.38FuriousGeorgethey can call out
23:24.45FuriousGeorgethey can answer calls
23:25.13FuriousGeorgebut as soon as someone gets on the ohone you decided to leave on the line before asterisk, it wont work right
23:28.19*** join/#asterisk Corydon76-home (i=seven@pdpc/supporter/sustaining/Corydon76-home)
23:29.00kingsleyWould asterisk be able to 1.) take voice mail 2.) log call detail records and 3.) send and receive faxes if only one line comes into the voice card, and that line is shared by the phone's hand set and the phone company's central office?
23:29.11*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)
23:29.19*** join/#asterisk dijit0 (n=dijit0@ppp-69-233-165-239.dsl.pltn13.pacbell.net)
23:29.22jontowsure, matter of configuration..
23:29.53dijit0would you guys say broadvoice is okay as far as quality and reliability goes???
23:30.16cyonicsi signed up with them last night
23:30.17kingsleyjontow: Were you answering my question?
23:30.23jontowyes.
23:30.42jontowperhaps, write an AGI script that'll hold the outgoing fax until the line is free, then send it
23:30.48cyonicsthey are ok so far, but have can't compare as they are my first service provider
23:30.58cyonicshave = I
23:31.29dijit0i c, and does the caller id with name work properly?
23:32.08cyonicsI'm able to see my home caller id info when I call my BV number if thats what u mean
23:32.34robin_zsigh .. poxy wiki software .. POS
23:32.44jontowrobin; mediawiki++
23:32.56robin_zwhat is the one on voip-info.org?
23:33.14kingsleyjontow: Do you happen to know if any special configuration is necessary for * to log call detail records from phones that share the line with the central office, but don't route calls through asterisk?
23:33.56jontowkingsley: its automagic.. look at the configs
23:34.35puzzledif the call doesn't go though an * box how can * generate a cdr?
23:34.37fileI think he means that the phones never touch Asterisk, and that it's an analog line
23:34.42jontowaha
23:34.45jontowgood luck with that, and no.
23:34.46jontow:)
23:34.47robin_zthe voip-info.org wiki is a bit sucky .. it shows the comment all wrapped when you preiview it, so you add <p> tags to format it a bit .. it looks OK on preview, you post it and find its full of <p> stuff ...sigh
23:36.19kingsleypuzzled: Yes, like you, I'm wondering how * would generate call detail records for calls that don't go through asterisk. I suppose it may be possible if asterisk can record whenever a line goes off hook.
23:36.34puzzledkingsley: it's not possible
23:37.05dijit0cyonics, did you do the BYOD from broadvoice?
23:37.31ManxPowerI told this to kingsley yesterday, but I guess he didn't listen
23:37.53ManxPowerkingsley, Asterisk CANNOT log calls that do not go thru it.
23:37.59jontowget some ATAs, or some IP phones, kingsley..
23:37.59kingsleyfile: I may have been too terse. As you surmised, I am assuming analog phones, an although they wouldn't plug into an asterisk card directly, they would share a line with the central office that plugs into the * card.
23:38.07jontowits the only way you'll really get what you want out of the deal.
23:38.19ManxPowerIn fact, analog Zap ports can't even tell if the line is off hook or not.
23:38.38jontowkingsley: in fact, good luck making it work correctly with * and analog phones plugged into the same line
23:39.21ManxPowerI don't know about the TDM2400P, but the TDM400B can't even tell if a line is plugged into an FXO port.
23:39.42kingsleyjontow: What is an "ATA"?
23:40.07ManxPowerkingsley, Analog<->VoIP adapter
23:40.26puzzledevening ManxPower
23:40.34jontowAnalog Telephone Adapter..
23:40.40jontowcheck the wiki (voip-info.org) for more info.
23:41.00ManxPowerpuzzled, hello
23:41.22*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
23:42.03kingsleyThank you for your thoughts.
23:42.36ManxPowerkingsley, Feel free to ask again tomorrow.  You'll get the same answers.
23:43.54jontow;)
23:44.41kingsleyManxPower: Perhaps the same answers from the same people, but not everyone will have the same answer.
23:44.57ManxPowerkingsley, in this case, everyone will have pretty much the same answer.
23:45.03puzzledkingsley: those people will be wrong
23:45.05jontowkingsley: your answer is, after understanding the question: "impossible.  try this instead."
23:46.22*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
23:55.40*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
23:55.44*** join/#asterisk RV-Dioxide (i=appleboy@ip68-231-211-153.oc.oc.cox.net)
23:55.44a1fahey
23:55.47afrosheenhello
23:55.54a1fawhat is that guys nick that built voicechanger app
23:56.03afrosheenhas anyone added fxotune to their startup scripts?
23:56.28ManxPowerafrosheen, yes
23:56.36afrosheenManxPower: where did you put it
23:56.41ManxPowerhas to be done AFTER zaptel and the drivers load, but BEFORE asterisk
23:56.52ManxPowerI edited /etc/rc,d/init.d/zapte
23:56.53afrosheenyeah I'm trying to figure it out on Centos..not sure where it goes
23:56.53ManxPowerI edited /etc/rc,d/init.d/zaptel
23:57.02afrosheencool I have that one :)
23:57.39afrosheenshould the rx and tx volumes be removed from /etc/zapata.conf when this is in place?
23:58.05afrosheenoops insert an /asterisk in that path
23:59.09ManxPowerno.
23:59.34afrosheenok, I thought something might interfere with this stuff, not alot of documentation on what it does or how it works

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.