IRC log for #asterisk on 20070629

00:04.31*** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net)
00:06.20*** part/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net)
00:06.38*** part/#asterisk karleeto (i=karl@gentoo.karlhaines.com)
00:08.29Hmmhesaysis there a good way to tell by callerid if a number came from north america?
00:10.00*** join/#asterisk coppice (n=chatzill@76.155.17.210.dyn.pacific.net.hk)
00:13.19HmmhesaysI'm trying to figure out how to set callerid with the country code all the time
00:16.44JTwell it's easy to fake in the US, so i dunno
00:18.14Hmmhesayswell the problem I see is some of the NPA's match country codes
00:18.57*** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)
00:19.13pipwerkso you can't trust incoming callerid's
00:19.26JTnot in the us, that's for sure
00:20.12pipwerkin that case it has no use normalizing them either I guess
00:20.40Hmmhesayswhat I want to know is how the telco's know when to add a 1 to a 10 digit callerid
00:21.08Hmmhesaysexample 972 is an areacode in Texas. and 972 is the country code for Israel
00:21.11coppicelook up NPI and TON
00:21.45Hmmhesayshow is that passed across sip though?
00:21.45*** join/#asterisk javar (n=javar@69.79.134.24)
00:21.52JTsip is not really telco grade
00:22.42Hmmhesaysit would be cool if they passed along type of number
00:23.07*** join/#asterisk CyBeR_GeeK (n=CyBeR_Ge@201.89.88.30)
00:23.26coppiceSIP is a mickey mouse system. how does a telco know how to interpret dialing to isreal from arizona? a prefix tells them its not for texas.
00:24.05*** part/#asterisk lymeca (n=lymeca@unaffiliated/lymeca)
00:24.13Hmmhesaysyes, but I'm talking about normalizing callerid
00:24.22Hmmhesayshow would you distinguish between israel and texas
00:24.26JT1
00:24.30JTa 1 in front
00:24.54JTor an international dialling prefit
00:24.56JTprefix
00:25.02JTit's 011 there isn't it?
00:25.04Hmmhesaysmany providers don't use a 1 if the call is from a north american phone
00:25.05coppicewhen you normalised it you should end up with a number plus and NPI and a TON. A leading + for international may be a substitute
00:25.20*** join/#asterisk JimAustin (n=JimAusti@cpe-24-27-122-207.houston.res.rr.com)
00:25.42Hmmhesaysthe problem is I don't have any control over the gateway that the call comes into..
00:25.51Hmmhesaysit comes to me via SIP
00:25.55Hmmhesaysthats the only place I can normalize it
00:26.14HmmhesaysSo I can't tell if the call came from Texas or Israel
00:26.34Hmmhesaysso I don't know if I should  +972 or 1972
00:26.54Hmmhesayssee what I mean?
00:26.59*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:27.01coppicewouldn't any international number be longer than any local number?
00:27.25JTi thought international callerid was flakey at best anyway
00:27.55HmmhesaysI just got a call from the UK,  44 and 10 digits
00:27.58coppiceyeah, often IDD comes through with a totally bogus caller ID, especially with cell phones
00:28.25HmmhesaysI'm just looking for a decent way to normalize this, I might just have to put up with the country code to north american area code matches
00:28.42Hmmhesaysdoes sip even have a mechanism for passing along npi and ton?
00:28.45coppicethe UK is a fairly big country, so the number length is distinctive. I'm not sure if all small places have enough digits to be distinctive
00:29.23Hmmhesayswell the good thing about north american numbers is they are length distinctive, you will always have 10 digits
00:29.38JTunless it's 911
00:29.39JT411
00:29.42JTwhat else
00:29.49coppice112
00:29.55JTon gsm
00:29.57Hmmhesays911 doesn't call you
00:30.05Hmmhesays411 doesn't call you
00:30.08Hmmhesaysetc..
00:30.12JTheh
00:30.20Hmmhesaysthats only in russia ;)
00:30.39Hmmhesaysdoes sip have a mechanism for passing ton and npi?
00:34.39JimAustinnewbie question: is asterisk a good solution for creating a conference bridge type solution for a dozen users? eight dialing in through the PBX to a TDM800 type analog card with FXO ports and four coming in as voip calls. or would you all suggest something else? and is the TDM800 the best way to interface to analog ports on the PBX if a digital connection is unavailable? will i have echo problems? do i need to buy hardware that
00:35.55*** join/#asterisk mtaht4 (n=m@165-115-62-200.enitel.net.ni)
00:36.06JT<PROTECTED>
00:36.06JT<PROTECTED>
00:36.18JTyeah, the ircd doesn't like being flooded ;)
00:36.34JTconferences should work alright if you have zap hardware
00:37.11JimAustinis the TDM800 zap hardware? or is this something else?
00:37.15JTyeah
00:37.18JTuses zaptel
00:45.13*** join/#asterisk remmo (n=junk@203.62.147.3)
00:49.55*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
00:50.21*** part/#asterisk mtaht4 (n=m@165-115-62-200.enitel.net.ni)
00:55.43*** part/#asterisk JimAustin (n=JimAusti@cpe-24-27-122-207.houston.res.rr.com)
01:01.31*** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com)
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01:08.24*** join/#asterisk wundaboy (n=pat@c-24-21-71-88.hsd1.or.comcast.net)
01:10.02wundaboywhen i call my phone from a landline (pstn -> gateway -> mypbx -> my ipphone), i cant hear anything on my ipphone, but i can on my landline phone
01:10.09wundaboywhat am i doing wrong?
01:10.56Strom_Mis your IP phone running SIP?
01:10.59Strom_Mis it behind a NAT?
01:11.05wundaboymy pbx is behind nat
01:11.12wundaboyand the phone is on the same network as the pbx
01:11.25wundaboyits a polycom ip500
01:11.26Strom_Mis the pbx talking sip to the gateway?
01:11.35wundaboythe pbx is talking iax
01:11.46JTto the gateway?
01:11.48wundaboyyeah
01:11.56JTcheck for firewalling to ip phone
01:12.14JTalthough 1-way audio is also possible in iax
01:13.04wundaboywhat do you mean?
01:13.30JTyou are describing a 1 way audio problem, are you not?
01:14.01*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
01:15.26wundaboyi am
01:15.42wundaboywhat would be causing it? a firewall inbetween the pbx and the gateway?
01:15.53*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
01:16.37JTpossibly
01:16.52JTso do calls to ip phone that don't go to gateway work fine?
01:18.11wundaboyumm
01:18.13wundaboyi dont know
01:18.17wundaboyi only have one phone
01:19.43JTmake an extension that does an Answer, then Echo
01:21.57*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
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01:29.28*** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com)
01:29.36watchyi got my iphone and its got sip bitches
01:29.46Strom_Mgood for you
01:29.53watchyi'm lying
01:29.59watchyi just wanted to be cool
01:30.38watchyiphones already hitting ebay at $1000
01:31.13JTwhat a rip
01:31.19Qwellheh, nice
01:31.28Qwelljust goes to show that people would have paid 1k
01:31.29Strom_Mi'm strongly considering going to the local apple store and photographing the schmucks waiting outside
01:31.37QwellStrom_M: do it
01:31.41JTStrom_M: nice
01:31.41Strom_Malright
01:31.44Qwellwell
01:31.45watchyhah i'm going to buy 2 iphones tommorow for ebay
01:31.46Qwellyou'd have to be
01:31.48Qwellyeah
01:31.48Qwellheh
01:31.53Qwellleave now :p
01:31.58watchyluckily i live in redneckville usa
01:32.15JTdid you know if you listen carefully enough to the speaker on an iphone, you can hear Steve Jobs saying "haha sucker"?
01:32.17watchyso i doubt anyones gonna be at my cingular store tommorow
01:32.21watchyhaha
01:32.33Qwellwatchy: You'd be surprised
01:32.38watchyi actually think the iphone is going to be cool as shit, i dunno if its worth $600 though
01:33.03watchythe comercials make it look amazing compared to other phones
01:33.19[TK]D-Fenderwatchy, the reviews so far say that for the most part, they are deserved
01:33.41[TK]D-Fenderwatchy, though its agreed that its connectivity is SLOW.
01:34.27watchyyou know whats fucking retarded about the iphone?
01:34.34watchyapparently it has a built in SIM card
01:34.40watchyyou can't just take yours and make it work
01:34.45[TK]D-Fenderwatchy, I want one that will have good performance on PDF (maps), normal phone, SIP(or IAX) phone, PDA stuff, music, and a DECENT camera (GOOD = bonus)
01:35.08watchyi have a cingular 8125 right now with windows
01:35.11watchyits quite nice
01:35.37wundaboywhats the difference between type=peer and type=user ?
01:36.44watchyyou guys read how much space the iphone OS takes up?
01:36.58*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
01:37.32[TK]D-Fenderwundaboy, Peer = outgoing, user=incoming
01:37.49wundaboyand friend is both?
01:38.20*** join/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net)
01:41.07wundaboyok i keep getting this why i try and dial out: Jun 28 18:40:25 WARNING[12902]: chan_iax2.c:7140 socket_read: Call rejected by 66.227.100.30: No such context/extension
01:41.15*** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net)
01:41.21wundaboyi cant figure it out, i have incoming working, but i cant figure it out...
01:43.43Strom_Mwundaboy: there's no matching extension or context for the inbound call in your dialplan
01:45.01wundaboyStrom_M, what do you mean? this is when im dialing from my polycom to my provider
01:45.25Strom_Mwell then you're sending them something other than what they're expecting
01:45.36Strom_Mdo they want eleven digits?
01:45.39Strom_Mten digits?
01:45.47wundaboythey want 11
01:45.50wundaboyand im sending 11
01:45.57Strom_M13115552368 ?
01:46.03wundaboyexactly
01:46.14Strom_Mshow me the dial() line
01:46.27wundaboyexten => _NXXNXXXXXX,2,Dial(IAX2/jnctn-out,1${EXTEN})
01:46.37Strom_Mwrong
01:46.38wundaboy<PROTECTED>
01:46.55Strom_MIAX2/jnctn-out/NUMBER-GOES-HERE,options)
01:47.35wundaboyexten => _NXXNXXXXXX,2,Dial(IAX2/jnctn-out/1${EXTEN})           ?
01:47.39Strom_Myup
01:48.01wundaboyoh man
01:48.04wundaboynow i feel like a moron
01:48.07*** join/#asterisk limaunion (n=limaunio@OL160-170.fibertel.com.ar)
01:48.11wundaboythanks man
01:49.57Strom_Mwelcome
01:55.20*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
01:57.40Strom_Mcatsex
01:58.34Hmmhesayssure
02:04.50watchyanyone wanna loan me the newest polycom firmware? i will hug you
02:05.02JTwill you give it back?
02:05.24*** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net)
02:05.28watchyyes
02:05.54JTcool
02:06.14watchynow give it up before i stab you
02:07.23*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com)
02:07.50watchyi dont see any firmware but i see you getting stabbed jt
02:07.50watchy:(
02:08.10JTi don't have it
02:08.27watchythe story of my life
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02:17.28flendershey, what does the "when convenient" option does when restarting again?
02:17.45Strom_Mwhen call traffic drops to 0
02:17.55flendersthanks
02:20.04flendersoh shit, when you do a restart when convenient, you can't do anything else on the CLI
02:20.08flendersis it a bug?
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02:23.03Strom_Mi dont know.  i always just wait till call traffic drops to 0 and restart now
02:23.52JTflenders: haha, that sounds non-optimal
02:26.52*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
02:29.06flendersit's terrible
02:30.24flendersso if you make any changes on zapata.conf, that can wait until there are no calls and you do a 'restart when convenient' but then you realize you need to reload the extensions, youre screwed
02:30.30flendersyou have to wait until it restarts
02:30.44Strom_Myou could also just do a zap restart
02:33.51flendersbut then I would drop all the current calls, souldnt I?
02:33.55flenderswouldn't I?
02:34.25JTyeah
02:34.37JTi think
02:35.03Strom_Mif they're zap calls
02:36.19JTflenders: so if you attach a console to it, it's non-responsive?
02:36.33*** join/#asterisk Krooks (n=Krooks@202.184.116.210)
02:36.37KrooksHello
02:36.39flendersany commands you type in, you get no response
02:36.49flenderseven a 'core show uptime'
02:36.49Strom_Mhello
02:37.00Krooksanyone using asterisk on centos 5?
02:37.19Strom_Mno one has ever used asterisk on centos in the entire history of hot dogs
02:37.33Krookshehe
02:38.02KrooksStrom_M: you running on cetnos 5 ?
02:38.06Strom_Mno
02:38.09Strom_Mi use debian
02:38.13Strom_Mbut what's your question
02:38.31Strom_Mor did you just want to form a knitting club
02:38.38KrooksWell I want to do iton centos 5 .I just want to know that its doable before I embark
02:38.44Strom_Myes it's doable
02:38.53Krooksok then
02:39.32KrooksIts would be nice to get a yes from somebody who _has_ done it on centos 5 though
02:40.29Strom_Mi've done it on centos
02:40.29Strom_Mi just like debian better
02:40.40Krooksoh ok
02:40.54JTasterisk runs fine on linux
02:40.59JTcentos is a linux distribution
02:42.03Strom_Mthe jizzy mcspurtface operating system
02:42.11Strom_Mfree as in boners
02:42.34KrooksStrom_M : should I svn checkout the trunk or should I just go with the tarred ones
02:42.52Strom_Mdon't use trunk in production
02:42.55JTjust use a release
02:42.59Strom_Mif anything, svn checkout the release branch
02:42.59Krooksok
02:43.01JTespecially if it's your first time
02:43.04Strom_Mbut the tarballs are a good idea
02:43.27Krooksok, 1.2.19 or 1.4.5 ?
02:43.32Strom_M1.4.5
02:43.38Krooksok I'm on my way
02:43.41JT1.2 :P
02:43.42Strom_Mwhee
02:43.50Strom_MJT: fuddy-duddy
02:44.15JTeh, might consider 1.4 in prodution when it reaches .8 ;)
02:45.06*** join/#asterisk Ergose (n=domainop@cpe-065-190-118-012.triad.res.rr.com)
02:47.26flendersfunny, that I've been running 1.4.x here for 4 months, and it's all good
02:47.35Strom_Mso yay, tomorrow i get to talk to a potential client about ripping out trixbox :)
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02:48.43Krooksok the zaptel and libpri are drivers for the fx cards , right ?
02:48.59JTlibpri is only required for digital cards
02:49.31Krooksok
02:49.36Krooksand zaptel too
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02:51.26JTyes, zaptel is also needed by FXO/FXS cards
02:52.36*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
02:52.54[hC]interesting. I thought i was having network problems but it turns out it was just a bug in OSX's tftp client.
02:53.05[hC]it cannot download files smaller than 512bytes.
02:53.21JTsounds like an apple
02:53.37*** join/#asterisk Ergose (n=domainop@cpe-065-190-118-012.triad.res.rr.com)
02:54.08[hC]on an unrelated side note, any of you guys done a factory reset on a cisco 7970 and have it sit at "Upgrading" --seemingly doing nothing, and not requesting DHCP?
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03:23.07NovceGuru<PROTECTED>
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03:23.20vnI hate music on hold :\
03:23.27vnespecially when played during 30mins
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03:26.58ErgoseHas anyone here ever had a problem with IVR not wanting to go more than one level deep before returning back to the home message?
03:27.22ErgoseJust a stab in the dark...
03:30.19ErgoseI believe the bug to be with the asterisk side of things because FreePBX seems to be configuring the extensions_additional.conf correctly, and asterisk does seem to acknowlege them...
03:31.43Ergosehowever for whatever reason, I press 1 for first option 1 or 2 for second and instead of going into the 3rd level voicemail menu, I get sent back to 1.
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03:32.19KrooksIf I'm not going to use my asterisk over a pstn line, then I don't have to worry about zaptel and libpri, right ?
03:33.13Ergosedon't think so, but I think freepbx was being pissy without the zaptel module
03:33.14JTKrooks: unless you need zap timing for meetme conferences or iax2 trunking or music on hold, then you need zaptel
03:33.29Ergosebut it was the dummty module
03:34.01JTyeah but freepbx is evil ;)
03:34.10Ergoseyeah tell me about it
03:34.21KrooksIs zaptel a hardware thingy, and the zaptel in asterisk is the driver for it ?
03:34.26ErgoseI still can't tell if my prob is with freepbx or asterisk
03:34.41JTzaptel is for interfacing to hardware normally
03:34.43KrooksI mean my box is just a standard
03:34.50KrooksI mean my box is just a standard PC
03:35.02KrooksI don't have anything called a zaptel in my PC
03:35.15JTyou don't get cards called a zaptel
03:35.21JTzaptel is the name of the software
03:35.30JTthere's plenty of info online about this
03:35.36Krooksok, thats exactly what I want to know
03:36.01Krooksand libpri is the same thing ?
03:36.15JTno
03:36.16VJFROMGThow can i adjust speex codec for 6 kbps?
03:36.22JTit's a different library
03:36.43JTfor pri circuits, they still need zaptel too
03:36.44Krooksno I mean its not a driver for a hardware thing
03:36.56JTwhat?
03:37.02JTask your questions more clearly
03:37.08Krooksok nvm
03:37.12Krooksthanks
03:37.23JTread the book
03:37.24JT~thebook
03:37.35jbotrumour has it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
03:37.36Ergoseso JT, any idea as to why I'm getting stuck in a loop on the IVR menus? I'm almost figureing something when wrong on the compile, but everything else works like a charm...
03:37.42KrooksI should read the manual first before I annoy someone and get kicked out :)
03:38.20Ergoselol
03:39.19Ergosepeople are usually cool on that kindof thing... atleast they used to be.... haven't been on IRC since way back
03:40.12Ergosebrb, going for a smoke before I kill my box...
03:40.49KrooksOne last try, I have a standard PC. you know, sound card , video card etc. nothing extra. I just wanna know if I need to compile zaptel and libpri also. Thats all. I just wanna play with asterisk. Not for any production. Just playing.
03:42.09GreenboxKrooks, the only reason you need zaptel is if you have one of the hardware cards
03:42.17Greenboxfrom digium
03:43.48KrooksGreenbox: ok perfectly understand now.
03:44.10Corydon76-homeOr if you need a timing source for one of the modules which requires it
03:44.19Corydon76-homesuch as app_meetme
03:44.47Ergoseyeah, all I was saying was that if using freepbx I had to load the dummy module, but that's more a freepbx quirk...
03:45.22KrooksCorydon76-home: you going to confuse me. If I don't have one of the zaptel hardware cards, then why would all that matter to me.
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03:45.37JTi explained when you need zaptel already, i guess it just needs to be repeated sometimes
03:45.40JTKrooks: for timing
03:45.49JTKrooks: please read the book
03:46.06Corydon76-homeHardware timing provides a more reliable source of evenly spaced interrupts
03:46.15Corydon76-homeSoftware interrupts just can't stack up
03:46.36JTsome of it is historical anyway
03:47.02KrooksSo even if I don';t have a zaptel hardware car, I still can make use of the zaptel module. Is that what you are saying ?
03:47.10Corydon76-homeCorrect
03:47.18JTyes, through ztdummy
03:47.33Krookswhich is in the zaptel module.
03:47.44JTno, zaptel is a package
03:47.47Ergoseyeah, I haven't had the time to go through the manual, I'm just trying to help the other admin couse I'm the "Linux guy..." :)
03:47.51Krooksoko zaptel package
03:47.51Corydon76-homeIt's a separate module, but it interfaces with the zaptel module
03:47.54JTthe zaptel kernel module is a part of that package
03:48.20Corydon76-homezaptel provides a consistent interface across multiple card types
03:48.35Krooksok I i'll jsut install zaptel and libpri. It won't hurt , right.
03:48.43Corydon76-homeNo, it won't hurt
03:48.51JTvery little chance you'll need libpri though
03:48.52Krooksok thats all I want to know
03:49.13Krooksok so forget libpri and install zaptel. Great.
03:49.22Ergosesounds good
03:50.53friedrich|good night.
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03:51.09Ergosegoodnight
03:53.03Krooksso I need the kernel source before I install zaptel , right ?
03:53.03ErgoseI just don't get it... the config is linking to the correct IVN sections, so I know the flow is right and it still doesn't work past the second menu... sends me back to the first instead of the third even though should goto 3rd menu...
03:53.24Krooksnvm
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03:55.22ManxPowerIVR (Interactive Voice Response)
03:56.44wundaboyhow do i setup a menu system?
03:57.11Ergoselol I'm having prob with our ivr...
03:57.48wundaboythat sucks
03:57.50*** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net)
03:58.16Ergoseare you using freepbx or just asterisk?
03:59.13Corydon76-homeMost IVRs are coded with Read.  For a simple menu, though, WaitExten is fine
03:59.27wundaboyim using asterisk
03:59.38wundaboyi just want a simple menu
04:00.44Ergoseooh, haven't heard about Read... maybe that'll help... I could compare the ivr sections generated and see a comparison
04:00.57Ergoseyou got the web address on Read?
04:02.37Ergoseheh... nvrmind stupid question...
04:02.44Ergoselol
04:05.36wundaboyok so i got a question
04:05.50wundaboyin extensions.conf is there any significance to the [default] context?
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04:06.49rob0IIUC not necessarily, but it's the default target for places like sip.conf, iax.conf and others.
04:07.10wundaboyok thanks
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04:08.42troy-when a call comes in on to a Cisco 7941 from Asterisk, for the first 2 seconds there is silence before you can talk, why is that?
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04:29.52wundaboyif i have someone in an IVR is it possible to then ring a phone on the pstn and transfer the call to that ringing?
04:29.58wundaboywhat command would i use with that?
04:30.34Strom_CDial()
04:30.37wundaboyok
04:32.34wundaboyhow do i specify how long it will wait for a dtmf response?
04:32.59wundaboywaitexten?
04:36.00wundaboyanyone?
04:36.54Strom_C<PROTECTED>
04:41.39MooingLemurtroy-: maybe it's round-trip lag between the phone and the pbx or other end of the call?
04:42.31troy-MooingLemur, hmm they are both on the same switch
04:43.01Aces1Upwhat are some good suggestions for cards, if i need 6 co lines as well as 6 station lines non IP.?
04:43.10JTtroy-: probably a sip/rtp setup problem
04:43.16Strom_CAces1Up: digium tdm2400
04:43.28troy-JT, how do i go about verifying that?
04:43.37JTtroy-: packet sniffing perhaps
04:45.39Aces1Upstrom do i need echo cancellation?
04:46.03Strom_CAces1Up: how long are your circuits?
04:46.32Aces1Uplength?  less than 200ft.
04:46.40QwellStrom_M: I got a video call going through the appliance :D
04:46.49Strom_Cyour telephone company's CO is less than 200 feet away?
04:46.55Strom_Cqwell: sweet
04:47.02Aces1Upstrom ohh i don't know.
04:47.12Strom_Ci'm still trying to figure out how to get the craft port to respond to my input :/
04:48.59Aces1Upstrom looking for solution for 6fxo/6fxs with 6-analog handsets for less than 2300.00 can it be done?
04:49.21Aces1Upaces including asterisk box.
04:49.23Strom_CAces1Up: how much is the TDM2422B?
04:49.26JThaha
04:49.30JTincluding asterisk box
04:49.31Aces1Up1515.00
04:49.33JTif it's free
04:49.35JTmaybe
04:49.41sweeperuh
04:49.42sweeperOR
04:49.47Strom_CAces1Up: let me offer you one piece of advice
04:49.49Strom_Cdon't be a cheapass
04:49.51sweeperyou could buy two 8-port gateways
04:49.56sweeper$300 each
04:50.01sweeperone fxo, other fxs
04:50.08JT$300, from where?
04:50.09sweeperhandsets are what, $20?
04:50.19Strom_C$20 for the crappy desk sets
04:50.20sweeperJT: the internets!
04:50.29Aces1Upahh well.
04:50.34*** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com)
04:50.35JTbetter not be much more for handsets or you mayaswell go sip
04:50.35sweeperI think the grandstreams go for less than 300
04:50.40Aces1Upmy client wants to go with a toshiba system.
04:50.41JTyeah
04:50.48sweeper300 is for the audiocodes
04:50.50JTbut using a grandstream for a business is braindead
04:50.53sweeperor the linkys
04:50.59sweeperso anyways
04:51.10sweeperthat leaves $1500 for the asterisk box
04:51.10JTcan you get 8 port linksys ATAs?
04:51.16sweeperoh yea
04:51.42sweeperthey require some tweaking
04:51.46sweeperbut they work
04:52.00JTwhat model
04:52.06Aces1Upwhats more reliable the gateways?
04:52.11sweeperSP4000? something like that
04:52.26sweeperAces1Up: more reliable would be a channel bank + t1 card
04:52.46sweeperthat would run you $1500 for both, tho
04:52.51sweepereh
04:52.51Aces1Uplol
04:52.59sweeperyou can build a decent system for $800
04:53.05sweeperI mean, for six phone lines
04:53.27Aces1Upyeh, thas all they need.
04:53.31Aces1Upwon't be growing at all.
04:53.37sweeperyou could do that
04:53.53Aces1Uphave a recommendation for the gateways?
04:54.05sweepernot grandstream
04:54.16sweeperpeople say they like audiocodes, but I've not had any luck with them
04:54.36troy-JT, any idea what this error means:  chan_iax2.c:3792 iax2_send: No private structure for packet?
04:58.21wundaboyso, when i have someone make a choice (press 1 for this, press 2 for this, etc), do i just put as one of the extensions, a Dail command?
04:58.39wundaboyfor instance: exten => 2,1,Dail(IAX2/jnctn-out/15415555596)
04:59.08JTif you want to dial someone, yes
04:59.21troy-JT, any idea on that error?
05:00.09JTno
05:00.19troy-seems really strange, its continuous
05:00.20JTexcept you some sort of packet error
05:00.32troy-how would i figure out what the problem is?
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05:01.36watchyanyone wanna loan me the newest polycom firmware?
05:01.57Strom_Cwatchy: your reseller should be able to provide that to you
05:02.21watchyvoipsupply is my reseller they are a bitch to get that shit from
05:02.42Strom_Ca good reason not to buy from them again, i suppose
05:03.02watchywell they are cheap and get me equipment fast
05:03.14wundaboywatchy, loan?
05:03.22wundaboylast time i checked you could just copy it...
05:03.31watchywunda: well copy it then
05:03.31wundaboyif i had it i would give it to you
05:03.37watchycp firmware firmware2
05:03.41wundaboyi have it on my phone
05:04.01wundaboygot it from my friend, just connected to his ftp server with the phone
05:04.09watchyah
05:04.19watchywell grab it off his ftp :/
05:05.39wundaboyi dont have that info anymore, it was a like a month ago...
05:06.10watchyi have indigestion and its making me sad
05:07.24cybergypsyi have just upgraded from asterisk 1.2 to 1.4 and the ilbc codec sounds much worse when calling through gizmo service , anyone else getting that problem ?
05:13.08[TK]D-Fenderwatchy, No, they are EXPENSIVE, and give shit service apparently
05:13.31Aces1Upwhat are the better brands for fxo gateways?
05:14.07[TK]D-FenderAces1Up, AudioCodes, MediaTrix, MultiTech (meh), Patton
05:14.15Aces1Upthanks tkd.
05:14.43watchygimme some service tk
05:14.59JT?
05:15.41*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
05:16.06[TK]D-Fenderwatchy, I don't workt he red-light district anymore.
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05:21.37Aces1Uptkd grandstream aren't good for gateways?
05:21.52*** join/#asterisk switch (n=switch@saya.attrition.jp) [NETSPLIT VICTIM]
05:21.53JTthey aren't good for anything
05:21.54*** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) [NETSPLIT VICTIM]
05:22.03JThow hard is it to spell his nick properly, btw :P
05:22.23Strom_Ctype [t and press TAB
05:22.37troy-[TK]D-Fender
05:22.38troy-there :P
05:23.14Strom_Cit amazes me how few people are aware of tab complete
05:23.27JTyes, not enough cli use
05:23.31JTmore cli osmosis required
05:23.36*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:23.47[TK]D-FenderAces1Up, I don't know how many times its going to take to get this through your head :
05:23.48[TK]D-Fender~gs
05:24.02jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
05:24.04[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
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05:24.35Strom_CAces1Up: let me offer you one piece of advice again:  DON'T BE A CHEAPSKATE
05:26.16Aces1Updaaaaaaaaaaaaaang
05:26.32Aces1Upcheck this out though  yehaaaa    http://www.asteriskguru.com/review_hardware_7.php
05:27.15Strom_Cyeah?  and see the bug in the middle of the review where it doesn't fucking work?
05:27.24Strom_Cthat's a pretty major bug right there :)
05:27.38Aces1Upheh, jeez.   you guys are so mean.
05:27.44Aces1Upi know just kiddin.
05:27.54Aces1Upgo with the audiocodes.
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05:29.25Strom_Ccocks, etc.
05:30.03JTi want to know what knok wrote that asteriskguru review
05:30.05JTe low price (approximately 450$) of the device makes it even more intriguing. The medium price of an AT is around 100$
05:30.17JTdollars symbols after the number is a major sign of retardation
05:30.21Aces1Upjt heh
05:30.32Strom_Cnot to mention mixing up medium and median
05:30.39Strom_Cthey're not even homophones
05:30.44Strom_Cthey're just similarphones
05:30.44*** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net)
05:30.51Qwellhell, they aren't even phones
05:30.57Qwellthey're like...similargrandstreams
05:31.01Strom_Chahahahaahahahahaha
05:31.04Strom_Cqwel++
05:31.05*** part/#asterisk zver (n=zveruga@ido-rtr2.distance.ru)
05:31.08QwellI *ALMOST* went for the other one
05:31.12Strom_Cs/l/ll/
05:31.13Qwellbut, decided not to
05:31.19Aces1Uphey, you guys are so funny  haha
05:31.26Qwellit would've been in bad taste
05:31.34*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com)
05:31.50Strom_Cqwell: I think after "cocks, etc." there's little you can do in the way of bad taste
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05:32.20Qwellwell, the alternative, would've been homograndstreams
05:32.20Strom_Cheheh
05:32.23Qwelland see, look, now the folks who just came in on a netsplit are like "wtf?"
05:32.35Aces1Uphey good one qwell hahaha  yeh hahaha
05:33.04russellbnetsplits are such teh silly
05:33.10*** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au)
05:33.10Aces1Up:ponders sarcasm in irc.
05:33.43*** join/#asterisk dansmith (n=dan@gw0.danplanet.com)
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05:33.49[TK]D-FenderAces1Up, So why are you even considering a SIP FXO gateway?
05:34.05Strom_Cbecause someone told him it was a better idea than a pci card
05:34.32Ergoseyeah, I haven't been on irc a while, but that was a massive dump of people....
05:34.55[TK]D-FenderErgose, welcome to FreeNode..... where Nodes..... run FREE.
05:35.06Strom_CFREEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE
05:35.11Ergoserofl
05:35.51Strom_Ctonight's episode of #asterisk is presented in technicolor
05:36.09Aces1Uphttp://voipspeak.net/index.php?option=com_content&task=view&id=113&Itemid=27
05:36.13wundaboyhow do i get voicemail registration to work with my polycom ip500?
05:36.17wundaboylike
05:36.22wundaboymake the light lightup when i have a voicemail
05:36.43Ergoselol
05:36.45Strom_C"The GXW-4104 is a slick looking box that is certainly one of the best looking products to ever have come out of Grandstream's engineering department. We wanted to put the GXW-4104 to the test so we decided to put it to use in a production environment in a small business using a trixbox server."
05:36.47wundaboyand makethe messages button connect to the voicemail system (rather than just having it dial an extension)
05:36.57Strom_Cthey care about looks and they're using trixbox
05:37.00Strom_Ci'd run from this review
05:37.05Aces1Uplol
05:37.24Aces1Upwhere are the good reviews at?
05:37.37Strom_CAces1Up: listen: forget Grandstream
05:37.39Strom_Cok?
05:37.52Aces1Upok.
05:37.56Strom_Cwe've been working with this stuff for a long time and we tend to know what we're talking about
05:37.56Aces1Upyes masta
05:38.22Aces1Upok doke.
05:38.25Aces1Upno problem.
05:38.32Aces1Upaudiocodes it is .
05:39.16Strom_Ci still say you should go for a TDM2422B, or a TDM808B and some IP phones
05:39.38Aces1Upwell kids, it has been fun, thanks for putting up with my questions this evening, bid you all a farewell and goodnight.
05:39.46Aces1Upstrom it would break the budget.
05:39.58Strom_Cre-evaluate your budget
05:40.06Strom_Cbeing a cheapskate == headache
05:40.08Ergosenight ace
05:40.08Aces1Uplol wish i could..
05:40.14Aces1Upnot my choice.
05:40.26[TK]D-FenderAces1Up, Yes it is.
05:40.34Strom_Cwell then you come back and say "we can't do it with this budget"
05:40.51[TK]D-Fenderexactly
05:41.00Aces1Upyep, thas whats i'm a gonna say.
05:41.36JT"we can do it, but it will be halfarsed and of poor quality and reliability, it will cost us in the long run"
05:41.58Aces1Upjt, sweet...
05:42.37*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
05:42.39Aces1Upthen i can say hey, why, don't you give me 3 more thousand dollars, and then we'll go get some cheesburgers.
05:42.59flendersAces1Up: what's the budget?
05:43.28[TK]D-Fenderflenders, "as cheap as possible" <- DUH
05:43.32Aces1Upwell, they are getting the toshiba cix40 system for 3300..   just looking to quote 1000.00 less than that :)
05:43.49flendershow many handsets? how many lines?
05:43.52[TK]D-FenderAces1Up, And what eother equipement is on this quote of yours?
05:43.57flenderssorry I missed it if you've already said
05:44.19Aces1Up6fxo, 6 handsets, voicemail, sla needed.
05:44.31[TK]D-FenderAces1Up, SLA?
05:44.39Aces1Upyes.
05:44.45[TK]D-FenderAces1Up, meaning?
05:44.54Aces1UpShared Line Appearances
05:45.03[TK]D-FenderAces1Up, LOL.
05:45.11[TK]D-FenderAces1Up, SO not happening!
05:45.16JTerr
05:45.16JTsorry
05:45.22*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
05:45.25JTHOW will SLA work on analogue phones?
05:45.33[TK]D-FenderAces1Up, "Put down the crack pipe!" (c) JerJre
05:45.43Aces1Upjt well i found that out, soo will run ethernet... so back to ip phones.
05:45.53JTah
05:45.56Aces1Uptkd its not cracked.
05:45.58JTyou only need a TDM800 then
05:46.01JTnot TDM2400
05:46.02Aces1Uplol
05:46.13[TK]D-FenderAces1Up, Your entire plan (yes, ALL OF IT) is on CRACK.
05:46.15JTand 6 Polycom IP320 or IP330
05:46.26Aces1Upmy pipe is not cracked :)
05:46.37*** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net)
05:46.39Aces1Upjt.. sweet.. lemme check da numbaz.
05:46.43JTi think it can be done, if the price doesn't involve your labour costs and a brand new server
05:46.48JTOK MAN
05:47.05Aces1Upjt yep, it does involve my labor, is only way i get paid, and the asterisk box.
05:47.13[TK]D-FenderAces1Up, You have set yourself up for end-to-end FAILURE.  Yuo have no clue about the equipment, nor even *'s capabilities.
05:47.28Aces1Uptkd I CAN DO IT!
05:47.29Aces1Uplol
05:47.47[TK]D-FenderAces1Up, "Denial.... it's not just a river in Egypt..."
05:47.50Aces1Upgive me a chance, I won't make a fool out of you.. yeh tom cruise, DAYS OF THUNDER baby :)
05:48.29[TK]D-FenderAces1Up, Thunder is actually a short way of saying "hot air that likes to make a big noise"
05:49.12Aces1Uptkd, i don't think i'm full of hot air... i know i don't know that much, and am well aware of my limitations...
05:49.30flendersAces1Up: seriously, don't do it too cheap... you'll regret it.
05:49.36flenderspersonal experience
05:49.36[TK]D-FenderAces1Up, You just don't know that of they gear & solutions you are basing your plans on.
05:49.57Aces1Uptkd, yeh, your right.  i haven taken your advice and will not go with the cheapo solution.
05:50.12Strom_Chttp://www.jerkcity.com/jerkcity300.html
05:50.15[TK]D-FenderAces1Up, More than that.... SLA <-  Not happening.
05:50.16JTAces1Up: [tk <tab> :D
05:52.19Aces1Upwell alrighty, you kids don't stay up too late..
05:52.25Aces1Uptalking bout phones and what not..
05:52.46Ergoselol yeah, no sleep for the admins...
05:53.02*** part/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net)
05:53.20Strom_Cwhat a nub
05:54.06*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)
05:55.24flenderswow, I'll never leave the channel
05:55.25flenders:)
05:55.25*** join/#asterisk denon (n=denon@tooth.decay.org)
05:55.25*** mode/#asterisk [+o denon] by ChanServ
05:59.30troy-Strom_C, what an atrocity
05:59.49[TK]D-Fender$850 Sangoma 86 FXO + HWEC. Phones IP 430 : 6 * $150 = $900, + Server ($1000).  can work.
05:59.58Strom_C[TK]D-Fender: he's already gone
06:00.07[TK]D-FenderStrom_C, I know.
06:00.28troy-i cant wait to pay bell $44 for a POTS line
06:00.46[TK]D-FenderStrom_C, Just proving a point.  But his approach sucks, and SLA is NOT happening, and he is all but certainly incompetant to set it all up.
06:00.52JTat least i'd be happy to be smashed in the face with a mediatrix or adtran, i'd have something usefuul to keep at the end ;)
06:01.00JT[TK]D-Fender: it had to include his labour costs too
06:01.04troy-haha yeah
06:01.11troy-unless he works for free
06:01.17[TK]D-FenderJT : You could cheap out the server.....
06:01.29JTyeah, second hand deskto p:P
06:01.34[TK]D-FenderJT : I could get a passable box for $100 techinally.  SFF to boot.
06:01.34troy-dell has servers starting at $400 right now
06:01.59JTheh, i think for really small setups without transcoding loads, embedded is the way to go
06:02.03troy-if configured reasonable you could make it for $700
06:02.07JTbrand new gear without 32838734 moving parts
06:02.20JTalso takes up less space
06:02.28[TK]D-FenderJT : Embedded actually COSTS more.
06:02.41JT[TK]D-Fender: more than what?
06:02.42[TK]D-FenderJT : And doesn't scale.  Never waste money on a DEAD END
06:02.45troy-JT, usually i buy rackmount boxes with rails and just stick it in a rack with the rest of the company's crap
06:02.49[TK]D-FenderJT : that a REAL PC
06:03.05JTpbxes are appliances,a ll some business need are an appliance
06:03.09troy-usually i spend roughly $1,200
06:03.21JT[TK]D-Fender: really depends on the specs of the PCs
06:03.29JTcheap ones will be much less reliable
06:03.32[TK]D-FenderJT : These days?  SERIOUSLY.
06:03.40JT[TK]D-Fender: not performance wise
06:03.43JTreliability
06:03.50[TK]D-FenderJT : uhhhh SURE...
06:03.52JTalso heat and power consumption and footprint are factors
06:03.53*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
06:04.05troy-JT, usually i will buy a P4 3.0 1G ram 2*80GB Raid 1 Dual GigE 1U Rails
06:04.08[TK]D-FenderJT : My server runs rock-sold for the past efw years 24x7
06:04.12JTget a shitty desktop, expect shitting reliability
06:04.19JT[TK]D-Fender: mine too, but they're server grade
06:04.25troy-exactly :)
06:04.26*** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au)
06:04.29phixhey
06:04.33phixJun 29 16:03:47 WARNING[8482]: codec_g729.c:249 lintog729_framein: Out of G.729 Encoder Licenses!
06:04.45JTtroy-: if you only need 6 extensions, getting a redundant everything server may seem like overkill
06:04.47[TK]D-FenderJT : its not a server unless its IBM's "iron" :)  Everything else is a glorified PC.
06:04.52troy-time to hand over everything you own to digium phix
06:04.57JTembedded is a niche
06:05.02JTmany won't fall into it
06:05.05phixHow expensive are ^.729 encoder licences?
06:05.06JTbut it's there
06:05.16troy-JT, disagree. a redundant everything box will fail a lot less
06:05.24Strom_Cphix: $10 per channel
06:05.27[TK]D-FenderJT : And there are 2 headed dogs.... Do you want one PERSONALLY?
06:05.37troy-i know plenty of companies with maybe 4 desks, that push serious sales / revenue and require high availability
06:05.48JTtroy-: you could buy a whole spare embedded box for the price of some servers
06:05.52[TK]D-Fenderok, bed time for sure.....
06:05.56JTtroy-: others don't have those requirements
06:05.57[TK]D-Fenderlater
06:06.02troy-night!
06:06.10troy-JT, guess it depends on the target market
06:06.37troy-i'm just laughing that i pay $10 for a 4 channel DID, and bell is charging $44 for single analog POTS
06:07.02Strom_Cwell that DID doesn't involve the leasing of the provider's outside plant
06:07.12troy-bell owns the plant
06:07.26Strom_Cmissing the point
06:07.45phixStrom_C: really? where from?
06:07.51Strom_Cphix: digium
06:07.56troy-yes i understand that a copper pair running between the CO and an office building costs money to install and maintain
06:08.08phixStrom_C: ok thank you
06:08.27phixStrom_C: $10 USD right?
06:08.30troy-ya
06:09.01troy-Strom_C, but that copper is a few decades old and was buried with thousands of other pairs
06:10.41troy-once that copper terminates at the CO its handed off on to Bell's SONET / ATM backbone and is basically the same thing as voip
06:11.01Strom_Clol
06:11.18troy-your argument is simply that the last mine costs money
06:11.22troy-*mile
06:12.06Strom_Cas does the SONET/ATM network, the class 4/5 switching infrastructure, the personnel to support it, etc etc etc etc
06:12.09*** join/#asterisk harlequin516 (n=sham@styk.net)
06:12.21Strom_Cwhereas your DID provider is probably three guys in a shack in calgary
06:12.47troy-you're telling me that carrier grade voip gear and personnel costs less?
06:13.06troy-bell charges CLECs a few cents for every DID, plus a few dollars per channel
06:13.36troy-bye bye :)
06:13.43Strom_Ccocks
06:13.48Strom_Cetc.
06:13.58troy-yes i have one
06:14.35harlequin516I have this problem that I have now finally traced.  I can only get a voice throughput from my SIP ATA to my destination if I play some sound at the Asterisk PBX Dialplan before attempting to Dial to connect to destination.  Is this a bug?  Or does anyone have a clue as to what may be wrong with my setup configuration?
06:14.52Ergosetime to sleep. later. been cool.
06:14.57*** part/#asterisk Ergose (n=Ergose@cpe-065-190-118-012.triad.res.rr.com)
06:15.10phixcorn dogs?
06:15.14harlequin516If I don't play a sound, then it keeps attempting to native bridge and never connects the voice.
06:15.43phixexcuse my ignorances but WTF is a corn dog? :)
06:16.28harlequin516Is it possible that Asterisk does not setup the voice path unless it needs to (in order to play that sound in the dialplan)?
06:17.20harlequin516Could this have something to do with NAT?
06:21.59brettnemwhat? bell doesn't charge CLECs for DIDs!
06:23.44Strom_Chttp://plif.andkon.com/archive/wc034.gif
06:36.37harlequin516brettnem : What does that mean?  They don't charge?
06:37.19brettnembell doesn't charge for DIDs. they don't even assign DIDs. they have absolutely nothing to do with DIDs. In fact, as a CLEC you tell bell what your DIDs are and you request that they route them
06:37.38brettnemnow, I'm talking about a facilities based CLEC. not a reseller
06:37.51Strom_Cassuming of course that you've received the DID block from CNAC
06:38.13harlequin516Hmm..  So does that mean that DID numbers are aquirable from some entity?
06:38.14troy-if you are facilities-based thats different :)
06:38.16Strom_Cor whatever the canadian numbering plan administrator's abbreciation is this week
06:38.18brettnemoh is this a canadian discussion?
06:38.28Strom_Cs/cia/via/
06:38.29troy-brettnem, yeah
06:38.45brettnemah... it's similar up there I believe..
06:38.51Strom_Cbrettnem: ok, so united states, s/CNAC/NANPA/
06:38.54Strom_Cand same deal
06:38.55troy-yeah pretty much
06:38.57brettnemactually
06:38.57brettnemI think you get your own DIDs for non-fac also
06:39.12troy-maybe?
06:39.32brettnemI'm pretty sure.. it's a real similar process.. I've done a lot of fac-based CLEC interconnection in the US
06:40.36brettnemI've decomissioned a non-fac based CLEC before.. so I've been through a lot of that paperwork. and i know for a fact that the CLEC owns the code and blocks. Bell doesn't have an ordering process for blocks and codes, just TQs and ASRs
06:41.23troy-could be, i'm busy trying to figure out why asterisk is producing this error
06:41.51*** join/#asterisk samarora (i=minesh@203.88.149.165)
06:42.08samarorahi there
06:42.35samarorai need some help regarding connecting two asterisk servers..
06:42.38troy-brettnem, bed time for me 2:45 AM :P
06:42.46brettnemyeah.. ugh.. time
06:43.02troy-must be up for work in 5 hours :(
06:43.14brettnemmy kids are going to get me up then
06:43.15brettnem:)
06:43.49troy-kids are the last thing i want, my dog is lazy and barely likes walking
06:44.33troy-he's usually ready to turn around before we get out the door
06:45.40brettnemhah
06:45.42*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)
06:47.01troy-brettnem, i finally understand why they call it man's best friend, if only he drank beer
06:47.13*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)
06:47.18brettnemI hear they drink just about anything
06:47.24*** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima)
06:47.35troy-maybe i'll launch a pilot project tomorrow
06:51.17*** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl)
06:52.00*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:52.30*** join/#asterisk stoffell_w (n=stoffell@fw.catsanddogs.com)
06:54.16*** part/#asterisk samarora (i=minesh@203.88.149.165)
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07:16.09*** join/#asterisk oej (n=olle@62.97.243.70)
07:34.06*** join/#asterisk basty (n=basty@212.218.65.247)
07:34.07bastyHi
07:34.36bastyWe have several Snom Hardphones on Asterisk 1.2.19, sometimes during a call (external/internal) it breaks up with following Warning in the Asterisk CLI: WARNING[18252] chan_sip.c: Maximum retries exceeded on transmission 3d2cca1cf1b3-c6cbjoxi8rvb@snom320-00041324A7FD for seqno 2 (Critical Response) - anyone knows why ?
07:35.00oejWe're not getting expected response on a SIP message - like it says
07:35.40oejIn an odd way though, but that is what it means.
07:35.43creativxpacketloss?
07:36.11JTusually it's packet loss
07:36.17bastyyeah - but why ? I mean I used to upgrade Asterisk 1.2.13 to 1.2.19 - allready googled. It seems that there is a change in chan_sip.c from res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2); to initreq 1
07:36.21*** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62)
07:36.32bastyon Asterisk 1.2.13 it worked just fine...
07:36.36JTerr
07:36.36*** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl)
07:36.47JTusually it means your network is stuffed, not asterisk :)
07:37.27*** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it)
07:37.42*** join/#asterisk yassaccan (n=yassacca@admin239.hgo.se)
07:37.55bastythats weirdo ;) Because it worked before for like 20 weeks without any problems...just after the update to 1.2.19 the "problems" happend
07:38.02bastyhappens indeed :)
07:38.25*** join/#asterisk matsk (n=mk@194.68.102.173)
07:39.26jeremy_gI am so pissed off
07:39.40jeremy_gwhy cant i perform make on one distro
07:39.55jeremy_gand make install on another to just put the binaries in there respective places
07:39.59*** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk)
07:40.00bastyso anyway..in your eyes, the error might be a packetloss from the snom hardphone to the server ?
07:40.06jeremy_gwhy does make install  require libs n crap
07:42.00JTjeremy_g: what are you talking about?
07:42.11oejbasty: Don't look in the source, look in the SIP dialogs. Some device is not responding properly. You have to check what type of message that isn't handled by turning on SIP debug and seeing what's going on.
07:42.57creativxbasty: thats why you dont upgrade if shit works :)
07:43.36jeremy_gJT:i want to compile asterisk on one machine loaded with compiler and all the libs and copy the binaries to another linux system that doesnt have a compiler and that many lib files. I ldd the binary and install the libraries it requires.
07:43.38bastycreativx: Well I had several cores in 1.2.13 ;-)
07:43.45jeremy_gi am making a live CD for asterisk
07:44.02*** join/#asterisk kova (n=kova@tech.quentris.com)
07:44.12JTjeremy_g: do you have a cross compiler?
07:44.21bastyoej: Some devices? It seems that ALL my Snom Phones (about 115) have the Problems :-(
07:44.29*** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com)
07:44.35JTbasty: are all the settings the same?
07:44.51bastyJT: Settings are the same...firmware could be different
07:44.57jeremy_gJT:both are linux, though different distros
07:45.06jeremy_gJT:same architectures
07:45.47oejWhat's the sip message you're sending? A 200 OK on an invite requires an ACK
07:45.49JTcan't you just copy the binares?
07:45.54oejso the 2 is correct
07:46.15*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)
07:46.34bastyoej: I dont know what the phone is sending..sounds like I have to turn on the SIP debug on Asterisk?
07:46.54oejbasty: That's the only way to figure out what's going on
07:47.18jeremy_gJT: i copied the asterisk directory in which i successfully executed make, to the other linux system and there did make install which gave this error: http://www.pastebin.ca/595003
07:47.54jeremy_gwhy does make install check for the libraries required
07:48.05jeremy_gi thought make install was suppose to compile everything
07:48.31bastyoej: just a general question....after "sip debug" how to turn the sip debug off ? ;-)
07:48.32creativxdamn swissvoice crap
07:48.39creativxnote to self: never buy swissvoice again
07:49.03bastyoej: no debug channel sip ?
07:49.18creativxsip no debug
07:49.20creativx:)
07:49.21oejbasty: Everything starts with "sip"
07:49.25bastyi am sorry ;)
07:49.27creativxnot very intiutive
07:49.32bastythanks creativx
07:49.36creativxshould be sip debug [on/off] :)
07:49.37creativxhehe
07:49.42oejin 1.6 it's going to be "sip set debug off"
07:49.45bastyyeah ;)
07:49.48jeremy_g:D
07:49.55creativxhehe
07:50.00creativxlooking forward to 1.6
07:50.06creativxor perhaps not.. 1.2 is working well
07:50.06creativx:)
07:50.11jeremy_gin 1.8 its gonna be sip configure debug set off
07:50.12jeremy_g:D
07:50.33JTsip no debug.. that style is pretty much cisco style
07:51.25creativxwho said cisco was intiutive :) hehe
07:51.26bastyis there any way to log the sip-debug into a file instead of the CLI ?
07:51.38creativx/var/log/asterisk/messages basty
07:51.47bastyyeah..but it is still in my cli ;)
07:51.56jeremy_gJT: any idea which library do i need? http://www.pastebin.ca/595003
07:52.00creativx/var/log/asterisk/messages is a file
07:52.23*** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net)
07:52.29bastycreativx: lol - I know that - but I dont want to see the odd sip-debug in the asterisk-cli...
07:52.34oejbasty: the SIP messages will always be to CLI, but you can copy directly to a file by doing asterisk -rvvvddddn | tee /tmp/mysipdebug.txt
07:52.34jeremy_gbasty:i always do it like that, you can forward all verbose output to a file
07:52.45JTjeremy_g: yes, kernel headers.
07:52.45*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)
07:52.53bastyah cool - thanks oej :)
07:53.10creativxahh basty :P
07:53.36jeremy_gbasty:or vi /etc/asterisk/logger.conf and add this line log-sip => verbose
07:53.47bastythanks!
07:54.37jeremy_gbasty:i then usually tail -f /var/log/asterisk/log-sip |grep 'anything-special-um-looking-for'
07:55.32creativxthen look forward to delving into a buttload o sip messages
07:55.32creativx=)
07:55.37bastyhehe
07:57.13JTtzafrir: about?
07:59.31bastyokay
07:59.55bastyI dont think logging all sip messages on this machine is a good idea
08:00.15basty-rw-r--r-- 1 root root 102M 2007-06-29 09:59 log-sip
08:00.32bastyin..hrm..10mins ? ;)
08:00.39JTwtf
08:00.45JTyou know why that is
08:00.49JTbecause it's failing
08:00.53JTand retransmitting
08:01.23bastyi have a lot of "Destroying call '763652823562007144222@10.46.0.5'" stuff in there
08:01.50creativxhow is your network?
08:02.25JTbasty: time to start packet sniffing
08:02.31JTsomething strange is going on
08:02.41bastywell...I have several Cisco Catalyst 3548XL Switches
08:02.54bastycreativx: or what do you mean by "how" ? ;-)
08:03.17creativx"how" as in any nasty equipment, low quality?
08:03.29JTcisco can be nasty
08:03.40JTespecially if you ever use a piece of rubbish known as "PIX"
08:03.43bastycreativx: nah...no low quality..only the best ;-)
08:03.58creativxhehe
08:04.05creativxglad i didnt buy a PIX.. netscreen <3
08:05.33bastyhrm
08:06.00bastyJun 29 09:58:29 VERBOSE[28550] logger.c: Sending to 10.46.3.100 : 2051 (NAT)
08:06.06bastywhy does it use NAT ?
08:06.12bastyI dont want to use NAT ;)
08:06.32creativxasterisk on a different subnet than 10.46.3 ?
08:07.01bastywell...the network is 10.46.0.0 / 255.255.248.0
08:07.06bastyso...the same network :)
08:07.42*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
08:07.43JTof course localnet= is setup correctly in sip.conf
08:07.47bastyyep
08:07.56JTmaybe you should set nat=no for the phones
08:08.17JTi thought this was the problem
08:08.30JTjust didn't think it was that important since you didn't pastebin anything
08:09.00bastyhrm
08:09.30bastyso I should insert "nat = no" in the general settings ?
08:10.02JTyou can, but if you have clients/servers over nat, you'll want =yes for them
08:10.12bastynah..its all in the same network
08:10.30JTso no voip on the Internet
08:11.08bastynope
08:11.19bastyjust 3 E1 Interfaces
08:11.35JTok
08:12.05bastyso the "nat = no" should fix my problem with the disconnecting calls during the conversation ?
08:12.37JThopefully, just give it a go already
08:12.42JTinstead of wasting time :P
08:12.52berktrbasty, do you have canreinvite=yes ?
08:13.01JTand no spaces on parameters in asterisk configuration files, least not in that section
08:13.25bastyberktr: canreinvite=no
08:13.39berktrok, and you still have disconnecting calls?
08:13.42bastyyep
08:13.59berktrlike after how many seconds?
08:14.08basty...well I dunno what the "nat=yes" did
08:14.19bastyberktr: differnt...lets say..every hour ? :)
08:14.20berktrnat yes wouldn't cause such a problem
08:14.25JTdid you set it to no?
08:14.31JTit can
08:14.37berktrno, i mean after you initiate your call, the other party answers, and the conversation beging
08:14.41berktrbegins*
08:14.50berktrcan you talk for like 15minutes?
08:15.21bastyberktr: oh okay..lets say...2-3 seconds ?
08:15.31berktri see
08:15.33bastyJT: nat=no - yep
08:15.44berktrbasty, is this a fresh asterisk installation?
08:16.12bastynot really...i used 1.2.13 for quiet a while....after some core dumps I had to upgrade to 1.2.19 :-\
08:16.42bastyi didnt change anything..just upgraded...and since that time..i have these weirdo problems.
08:16.44berktrthis most probably won't make sense but i had exactly the same problem with 1.2 after a fresh installation
08:16.58berktrand i was about to go crazy and upgraded to 1.4
08:17.06berktrnow i'm having no problems at all
08:17.23berktrtoo bad i couldn't figure out what caused my problem but
08:17.39berktrswitching to 1.4 solved it, and i had no time to research more
08:17.45basty:-(
08:17.57berktrwhy do u still insist on 1.2?
08:18.06berktrit can't even change rtp packet sizes?
08:18.08bastybecause 1.2 is actually stable.
08:18.26bastyi dont trust 1.4 for such a big installation
08:19.17*** join/#asterisk berktr (n=cn@teknopet.com)
08:19.24berktrsorry somehow freenode dropped me
08:19.54bastynp :) well..like I said..i dont really trust 1.4...
08:20.03basty1.2 is actually working fine...as in stable...
08:20.15berktri used to work with 1.2, but 1.4 is really much better
08:20.24berktrsip jitter buffer works perfectly
08:20.37berktrdo you use h323 there?
08:20.45bastynah...just SIP and IAX
08:20.55berktrokay then, you can switch to 1.4
08:21.01berktrit is really easy and takes less time
08:21.17bastyberktr: how many phones are connected on your installation ?
08:21.24bastyand how many E1/T1 Lines?
08:21.24berktr130
08:21.32berktrhere, it's different
08:21.36berktrno e1/t1
08:21.39bastyoh
08:21.41berktr130 internal phones
08:21.52berktrand we are directly connected to local telecom operator
08:21.55berktrover internet
08:22.04berktrwe have did numbers
08:22.04bastyoh okay
08:22.40berktrasterisk is handling conversations real good
08:22.48berktrinternal phones use ulaw in the network
08:22.55berktrpstn connections are g729
08:23.06bastydo you use FAX with Asterisk ?
08:23.13berktrdon't even ask....
08:23.23berktri couldn't get it to work :(
08:23.34berktr1.4 supports t38 passthrough
08:23.46berktrbut our telecom provider doesn't support it
08:23.50bastyoh okay
08:24.53bastywell..actually I work with mISDN or Zaptel...for E1 Interfaces...but as a matter of fact I had to realize the fax with Audiocodes Gateways
08:24.53berktrwe have some welltech gateways here
08:24.53berktrthey all support faxing
08:25.01bastyin T.38 ?
08:25.01*** join/#asterisk ramindia (n=ramindia@202.63.96.9)
08:25.01berktryes
08:25.01berktrwith ulaw
08:25.10*** part/#asterisk ramindia (n=ramindia@202.63.96.9)
08:25.56*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
08:25.57bastyberktr: where are you from ?
08:26.04berktrturkey
08:26.06bastyah okay
08:26.14berktrwhy did u ask
08:26.50berktrhttp://www.welltech.com/product_e_01.htm
08:26.50bastyI thought we could try to talk in German ;) Because my english isnt that good
08:27.06berktrahh i see, only english - turkish - french here
08:27.15bastywhat a shame :)
08:27.33berktrlol :)
08:27.45Zeeekich habbe kein camera
08:27.53Zeeeknoch en, bitte
08:28.26phez72I'm not english, turkish or french
08:28.38phez72:)
08:28.42Zeeekein Traum ist unser Leben auf Erden hier
08:28.54berktrbasty, if you want to go to a vacation, with no problems behind you, switch to freebsd, use ports, install 1.4, configure it, and forget about it
08:29.08*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
08:29.20*** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg)
08:29.29creativxheute is mein tag
08:29.41bastyzeek: hehe
08:29.59bastyberktr: heheh..I prefer FreeBSD also
08:30.04Zeeeknadle tsipful des Kafer bumser
08:30.07bastycreativx: "ist" :)
08:30.14berktryeah, it's cool, installs in 2 minutes
08:30.37bastybut still..there is no misdn/zaptel support for freebsd
08:30.40berktrgive me a brand new computer with no os, it will take only 1 hour for me to set asterisk up on it
08:30.56creativxverdammnt basty, i forgot :)
08:30.56Zeeekberktr that's about right
08:31.08bastycreativx: laugh
08:31.15berktri don't know about zaptel support though
08:31.20berktri thought it was solved in 1.4
08:31.43bastybrb....cigarette
08:31.47berktrk
08:32.16Zeeeksmoking is only permitted during the compilation phase
08:32.32berktr:)
08:32.53Zeeekand then only when compiling assembly language... from a cassette tape
08:33.35creativxehh.. what the hell am I thinking wrong here.. how do I invoke musiconhold on an extension? exten => s,n,musiconhold(class) right?
08:33.56berktrhere goes a moh question
08:34.01Zeeekyou don't invoke musicon hold
08:34.03berktrlet's say i changed the music on hold music
08:34.12berktrhow can i reload the moh module?
08:34.15berktrto switch to new music
08:34.25Zeeekrestart
08:34.29berktrgrr
08:34.32creativxbut I want to Zeeek :)
08:34.32creativxhehe
08:34.41Zeeekthan look up playback
08:34.47creativxyeah thats what im doing now
08:34.50berktrwithout restarting, is it possible
08:35.04*** join/#asterisk l2trace9999 (n=l2trace@fl-67-76-209-28.sta.embarqhsd.net)
08:35.17Zeeekberktr depends. If you have a bunch of mp3players already running as processes, no
08:35.31bastyback
08:35.56Zeeekyou're right not to inhale
08:36.20creativxhow old is this wiki page about cmd musiconhold() i wonder
08:36.27bastyokay...for right now..i dont have any calls broke up..after setting "nat =yes" so it seems to work...for right now.. ;)
08:36.35*** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net)
08:36.42l2trace9999any one know of an option to Dial like g but for the caller
08:37.30Strom_M?
08:37.36creativxZeeek: reason i wanted to invoke it was im using a ghetto parking extension.. and instead of playback(same file) each time.. i thought moh could handle it via a moh class
08:38.35Strom_Ml2trace9999: i don't understand your question
08:39.17l2trace9999the asteisk cmd Dial has an option to it
08:39.18l2trace9999g
08:39.33bastyokay..next question..anyone familar with call-queues ? I have setup a hotline queue...with 10 agents...during a 24/7 support only 2 agents are logged in...actually the queue works perfect. But if the 2 Agents are talking and the queue is actually busy...it rings for like 2 seconds...and then it is busy. In the Queue i had to setup "r". How can I fix that ? I dont want that the caller is getting a "ring" before the "busy".
08:39.46l2trace9999it allows the dial plan continue when the callee exits
08:39.57l2trace9999hangup or disconnected
08:40.10l2trace9999but it doesn't continue when the caller exits
08:40.19l2trace9999hangup or disconnected
08:40.50Strom_Ml2trace9999: thats because when the calling party hangs up, the call is torn down
08:41.02bastyoh and if I use "exten 1,1,Queue(technik) without the "r" - the caller hears nothing..not even the ringing.. :-(
08:41.10Strom_Mif you want to do post-teardown processing, use the h extension
08:41.38l2trace9999for outbound dialling ?
08:41.47Strom_Mfor any call
08:41.55creativxbasty: without r they should get moh if configured
08:41.55l2trace9999ok
08:42.21bastycreativx: exacly...but I want to have a "Ringing" instead of "moh"
08:42.48bastyso actually i HAVE to use the "r" - but with using the "r" it rings..even when the queue is busy...
08:43.09creativxyep i know
08:43.14kiscokidI installed Asterisk Gui on 1.4 but the login web page keeps reloading over and over even after I login
08:43.15l2trace9999thanks
08:43.21creativxi had some run ins with the queues myself
08:43.23Strom_Ml2trace9999: also, some terminology for you
08:43.29Strom_M"calling party"
08:43.33Strom_M"called party"
08:43.37Strom_Mnot "caller" and "callee"
08:43.49creativxi ended up with no r, moh silence and really short intervals for queue positioning messages etc
08:44.02dharriganSorry folks for asking here, I'm a right newbie. I'm hoping to setup asterisk shortly but what I don't understand about the last bit - if I have, say 20 internal analogue phones, how do I connect these to a fxs/fxo interface? These interfaces come with like 4 or 8 ports, do I need multiple cards?
08:44.16l2trace9999i know but i was using terminology from the voip-info documentation
08:44.20bastycreativx: So there is actually no fix for that? Because I disabled the queue position messages...
08:44.20dharriganand will asterisk communicate with all the cards?
08:44.33creativxbasty: i didnt have time to investigate
08:44.36Strom_Mdharrigan: you can get the tdm2400, which is a 24 port card
08:44.39Zeeekdharrigan yes you need multiple cards
08:44.44bastyah okay...dang :-(
08:44.48Zeeekor the 240
08:44.50Zeeekor the 2400
08:44.51bastyor like you said..."verdammt" ;-)
08:44.53Strom_Mor you can do a T1/E1 span and a channel bank
08:45.07Strom_Ml2trace9999: voip-info is not a reliable source for accurate terminology :)
08:45.26Zeeekdharrigan I think at the level of tens of phones it becomes cheaper to switch to SIP phones
08:45.36Strom_Mit depends
08:45.50bastycreativx: familar with sipsak ? :)
08:46.00Strom_Mif there are existing phones, the channel bank / tdm card option may be cheaper per-port
08:46.02dharriganSIP phones - these connect via the internal network yes, then that gets routed to the asterisk server?
08:46.12Zeeekyes
08:46.51dharriganCool, that's all for answer and clearing up some confusion
08:46.59dharrigans/that's/thanks/
08:47.21Zeeekdo you need to use existing analog phones?
08:47.27Strom_Mdharrigan: of course, your data network has to be up to snuff for SIP phones to work properly also
08:47.37Zeeekif so, it's more complicated to determine which is better
08:47.59creativxZeeek: i got it working now.. there is a way of invoking moh
08:47.59creativx;)
08:48.15Zeeektell us so we won't die stupid
08:48.35Zeeeks/stupid/ignorant/
08:49.04dharriganZeeek: Well, I think the office has about 5 phones, so perhaps easier just to buy sip phones?
08:49.19ZeeekIn my opinion it would be
08:49.25creativxfirst its a good idea to load res_musiconhold.so
08:49.31l2trace9999thanks now i feel like an idiot because I spent the last 2 days on that
08:49.33Zeeekbut it depends on a lot of things, like the users habits etc
08:49.38Strom_Mdharrigan: yeah, but whatever you do, don't buy grandstream
08:50.01Zeeekthe linksys phones are decent and the polycoms are good
08:50.11creativxso with res_musiconhold.so, you get application setmusiconhold(class) and musiconhold(class)
08:50.14dharriganHowever, they may not wish to expend money on replacing phones they already have, so I may have to go with analogue
08:50.17Zeeeklinksys used to be called Sipura
08:50.33Zeeekyou can mix and match. Keep 4 analogue
08:50.36Strom_MZeeek: no, linksys /purchased/ sipura
08:50.46Zeeekthe result is the same
08:51.01Zeeekwhat was SIpura is now called linksys (or cisco even)
08:51.16Zeeekthey're decent phones (941-2)
08:51.22Strom_Myes, but you made it sound like the one company simply became the other :)
08:51.33Zeeekwhat is your native language?
08:51.45Strom_Mmine?
08:51.48Zeeekya
08:51.53Strom_MEnglish
08:51.56creativxZeeek: are you prepared to die now then?
08:52.09Zeeekno because I'm still ignorant! :)
08:52.32creativxbut you could atleast die happy
08:52.41creativxknowing you can torture callers with moh at your discretion
08:52.41Zeeekbut I'm not happy at the moment
08:53.00Zeeekcreativx I have a much better torture for them: voice recognition!
08:53.09creativxouch
08:53.18Zeeekthat shit is REALLY irritating
08:53.24creativxyes
08:53.27creativxi was about to say
08:53.30creativxhas that tech matured enough
08:53.33Zeeekas anyone who's ever flown and reserved on the phone knows
08:53.34creativxbut you already answered :)
08:53.45creativxwhat? i order my flights online ;) hehe
08:53.46Zeeekno it works well
08:56.35kiscokidanyone using Asterisk Gui with 1.4?
08:56.53vltHello. Is there a dialplan pattern for the following regex?      /^555[0-9][0-9]*$/
08:57.22Strom_M_555XX.
08:57.28vlt_555X. doesn't work because it doens'n match 5550
08:57.40Strom_M_555X!
08:57.45Strom_Mthat'll match 5550
08:58.15vltSo I need two different extensions, right?
08:58.28Strom_M....?
08:58.36Strom_M! maches 0 or more digits
08:58.49vltAaah ... nice.
08:59.01creativxvery nice
08:59.03creativxmake sexy time
08:59.25vltSo _555X! matches 5550 and 55599?
08:59.25Strom_Mwas that a suggestion or a pickup line?
08:59.28Strom_Myes
08:59.44Strom_Mand 555999999929342394293492593452353623456
08:59.50vlt;-)
09:02.03*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)
09:02.36tzafrir_laptopkiscokid, ask something more specific
09:02.53tzafrir_laptopor ask on #asterisk-gui
09:03.38creativxStrom_M: hehe.. indecent proposals.. its friday innit
09:03.54kiscokidI tried installing Asterisk Gui on 1.4 and it seemed to install correctly but the login page keeps reloading even after logging in
09:04.45creativxcookie?
09:05.02Zeeekno thanks, I'm dieting
09:05.08*** join/#asterisk qdk (n=qdk@213.150.62.32)
09:05.09*** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net)
09:06.38kiscokidI'll try clearing cookies
09:07.16creativxhehe
09:07.22creativxinteresting.. 99,9% cpu usage
09:07.44Zeeekstop surfing pr0n sites
09:08.21Strom_Mhttp://www.jerkcity.com/jerkcity1578.html
09:10.33kiscokidcreative: that fixed it !
09:12.07creativxhehe kiscokid
09:12.30creativxglad to har
09:12.31creativxhear
09:13.14*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
09:13.59*** join/#asterisk Infested (n=infested@24.148.112.10)
09:16.37creativxexiting: 32253 root      25   0 21180 9916 4864 R 99.9  1.0   2757:06 asterisk
09:16.50creativxwhat have I done to my poor asterisk since its chewing 99% cpu
09:26.18*** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net)
09:27.42corruptorMy asterisk 1.4.5 after about 2 days of uptime works very slowly and consumes more and more memory. I relate this to Mixmonitor usage. Which info I have to post mantis? or how can i debug thisterisk after about 2 days of uptime works very slowly and consumes more and more memory. I relate this to Mixmonitor usage. how can i debug this? I need some info to post mantis but don't what can I post here. It's already compiled with dont_optimize and debug_th
09:27.42corruptorreads
09:29.25Strom_Mpaste mishap
09:29.26*** join/#asterisk key2 (n=Ritual@193.33.36.20)
09:30.16corruptoroops sorry for double post
09:39.30Zeeekand now... silence...
09:39.44Zeeekslowly I turend, step by step... inch by inch...
09:44.28*** part/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net)
09:47.40*** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com)
09:47.59creativx.mute=true
09:48.59ZeeekToggle(${mute})
09:56.47creativxset(mute=yes) would be proper cli syntax
09:56.47creativx;)
09:57.02creativxqueue pause sip/587 tnx.. lunchtime
09:57.03Zeeeknot with my toggle function
09:57.07Zeeekhttp://www.lulu.tv/?p=11368
09:57.49creativxhaha
09:57.50creativxghetto
09:58.09creativxglad I have to care about is the cat6 cable
10:00.08*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
10:01.23creativxdamn this chicken burrito was disappointing
10:01.26*** join/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com)
10:02.52Andri[DK]I'm currently in the process of upgrading my old Asterisk installation (1.0 branch) to a more modern installation. Are there any distribution recommendations. I'd prefer to keep everything within the packaging systems but I'd also like to have some sort of GUI for extensions and such.
10:03.18creativxonly option for gui is 1.4 (natively)
10:03.21creativxafaik
10:03.33*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
10:03.55creativxand going from 1.0 to 1.4 could be interesting
10:04.09creativxi would start reading about all deprecated functions and apps, and grepping like a madman
10:04.56Andri[DK]It's going to be a slow migration. I'll have a second server with a new SIP uplink and when that is working properly, I can start moving the phones
10:05.10Andri[DK]Can you recommend a Linux distro with 1.4 ?
10:05.55*** join/#asterisk vn (n=nostalge@bas5-quebec14-1177599731.dsl.bell.ca)
10:06.00Zeeekcentos 5
10:06.44*** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com)
10:06.57Andri[DK]is that packaged with Asterisk 1.4?
10:07.06Zeeekno
10:07.28Andri[DK]I left my company for school and I'm just here for the summer. So I'd like my inexperienced replacement to be able to update it :)
10:07.51creativxhehe poor sould
10:07.52creativx-d
10:08.06Andri[DK]so, there is no distro packaged with 1.4 yet ?
10:08.20Andri[DK]not even 3rd party repositories for debian or ubuntu ?
10:08.33creativxi doubt you are looking for trixbox
10:09.05Zeeekwhy package? Just get an os and do it
10:09.19Zeeeksomeone just said this morning it only takes about an hour
10:09.29mvanbaakAndri[DK]: try http://www.asterisknow.org
10:10.03creativxmake
10:11.25mvanbaakanyone from digium here?
10:11.30mvanbaakI cannot download asterisknow ;)
10:11.37mvanbaakIceweasel can't find the server at asterisknow.staging.digium.com.
10:13.29creativxasteriskLATER
10:13.31creativxplz try again
10:13.34creativx:)
10:13.54mvanbaaknope
10:14.07Strom_Myeah....like when it's not 5 AM
10:14.16creativxits not 5 am everywhere
10:14.21Strom_Mit's 5 AM in alabama
10:14.24mvanbaakFri Jun 29 12:14:23 CEST 2007
10:14.24Strom_Mand digium is in alabama
10:14.25Strom_Mtherefore
10:14.27Strom_MBONERS
10:14.32creativxits noon in norway
10:14.35creativxn for norwegium
10:14.37Strom_Mgood for norway
10:14.48Strom_Malabama is not in norway
10:14.51mvanbaakhhmm, the vmware images can be downloaded
10:15.06Strom_M(a-ha)
10:15.25mvanbaakwrong link there I guess
10:16.03creativxStrom_M: but norway is in usa
10:16.10creativxi think
10:16.28ZeeekNorwalk is in California
10:16.43mvanbaakyup
10:16.45mvanbaakwrong link
10:16.56Strom_Mspecifically, southern california
10:17.04*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
10:17.13Strom_Mlos angeles county
10:17.13Strom_Marea code 562
10:17.16mvanbaakif I copy-paste and replace asterisknow.staging.digium.com with www.asterisknow.org it's working again
10:17.18creativxWalk the walk
10:17.27Strom_Mcock the cock
10:17.33creativxshoot the gun
10:17.37Strom_Mbonerpills
10:17.59cpmsure as a ferret is a ferret
10:18.30Andri[DK]Zeeek: I'd like the pacacking for my replacement. I can easily install it from source but then it's going to be worse when it comes to patching security problems and such for the guys here that aren't that familiar with Linux or Asterisk
10:18.59creativxhow can they manage asterisk if they cant even compile from source :)
10:19.09creativxi didnt know the GUI was that far along.. hehe
10:19.09Strom_Mthose are some sysadmins
10:19.09Andri[DK]btw, if anyone is looking for a Linux related job, Iceland is the place to be... Too many jobs, not enough people
10:19.25Strom_Msunlight 24 hours a day during the summer
10:19.31Andri[DK]yeah
10:19.32Strom_Mscenic reykjavik
10:19.39creativxglitnir
10:19.43Strom_Mblblblbblblbbl
10:19.59Andri[DK]you guys know Iceland? :D
10:20.05Strom_Mskak mat 1972
10:20.13creativxi know its expencive as f there
10:20.19creativxand i know of arctic trucks
10:20.43Andri[DK]cool, I'm here now for the summer, but it's just because the salary is pretty good, even though it's expensive to live here
10:20.55creativxdo they understand danish?
10:20.59Strom_MI can spell ?
10:21.04creativxi can has cheezburger?
10:21.10Andri[DK]some do, but I'm Icelandic... I'm just studying in Denmark
10:21.35ZeeeksecurityR US
10:21.35Strom_Mand also ?
10:21.36creativxaha
10:21.37Strom_Mno, a-ha is norway
10:21.46creativxmisleading suffix there andri
10:21.48Andri[DK]take on me *sings*
10:21.52creativxa-ha sure is
10:21.56Strom_Mthe sun always shines on tv
10:22.03creativxthe sun shines in bergen too
10:22.11Strom_Mbut especially los angeles
10:22.19Strom_Mand what about cheesecake
10:22.19Andri[DK]creativx: yeah, sorry... I live in DK, didn't wanna register another nickname :P
10:24.33creativxso how do you like denmark
10:24.41creativxare they treating an icelandic well
10:24.44Andri[IS]very well, cheap beer, pretty ladies ;)
10:25.06Andri[IS]or was it pretty beer and cheap ladies... it's good anyway
10:25.08creativxyeah they got the ladies we didnt have room for
10:25.21creativxhere the beer is expencive and the ladies pretty cheap
10:25.22creativx;)
10:25.46Andri[IS]Iceland is like the run-away-child of Denmark. We can still come home to mom and dad, even if we wanna be independent
10:26.18Andri[IS]it's f.e. cheaper for me to study there, than home in Iceland
10:27.27creativxhehe
10:27.29creativxinteresting
10:27.51creativxi know a lot of the fish-business like to have meetings and trips to iceland
10:27.59creativxthey burn money like its 1995
10:28.38Andri[IS]yeah, those assholes are like the icelandic royalty, inherit daddy's fishing quotas and spend money like there is no tomorrow
10:29.10*** join/#asterisk key2 (n=Ritual@193.33.36.20)
10:29.26creativxmoney exists for one purpose
10:29.29creativxto be kept in circulation
10:29.33creativxe.g. use em while you got em!
10:30.13Andri[IS]yeah, somebody should tell those Norwegians... All that oil and still just keep the money in the bank
10:31.01creativxits the norwegian pension fund :)
10:31.07creativxnot even called oil money any more
10:31.30Strom_MIKEA
10:32.19creativxjarlsberg
10:33.16Strom_Mjohannesburg?
10:33.25Andri[IS]tuborg?
10:33.42Strom_Mkatzegeschlecht?
10:34.45Strom_Mkilling the conversation since 2004
10:34.46*** join/#asterisk ghenry (n=ghenry@212.159.59.85)
10:34.49creativxhehe
10:34.52creativxstill going strong
10:35.00creativxbumsenkönig?
10:37.01Strom_Mhollywood, california (not a real city)
10:38.43creativxmm
10:38.47creativxi can feel the friday vibe already
10:41.13creativxperhaps i should go get an icecream in ze fridge
10:42.40*** join/#asterisk javar (n=javar@69.79.134.24)
10:42.45*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
10:43.57*** join/#asterisk linagee (n=linagee@about/linux/staff/linagee)
10:44.09linageeyikes. better have your caller ID set right. LOL. $10,000 fine for doing it wrong.
10:44.25creativxin what messed up country?
10:44.30linageei wonder what happens if you have one of those silly phone cards that let you change it?
10:44.30creativxone that deals with dollars
10:45.10linageecreativx: what if it was like, "don't you DARE try to spoof your vendor assigned MAC address" (this is essentially what it is.)
10:45.31linagee(yes, i am saying a phone number is just as arbitrarily assigned as a MAC address)
10:48.17creativxyup
10:48.26creativxi know a guy who benefited from spoofing mac's
10:48.34creativxhe managed to bridge some 4-5 cable modems
10:48.41creativxand get sweet bandwidth for free
10:49.19*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
10:49.46linageecreativx: what is "spoofing an outgoing caller id" if for example, you have a call center that all send out your 800 number on the caller ID?
10:50.49linageemaybe they are somehow magically exempt because they are a business. heh
10:50.53creativxthis is what we would call a legal hell
10:55.57linageecreativx: hrm. someone asks a legit question on slashdot. :) does the victim of the spoofer get part of the $10,000? (heh. that would be nice.)
10:57.07dharriganHi peeps again! :-) I think I've figured out now about asterisk + analogue telephony. Now, my question is if I want to do voip from my place to another place, I need to go over the net right? I can't do that via an analogue line, so the asterisk server needs some type of network connection - it can route sip/voip calls across the net?
10:57.42dharriganso my asterisk installation can act as a gateway to traditional analogue and also do across the net voip calls as well?
10:58.58creativxdharrigan: correct
10:59.05creativxyou decide in your extensions where to route outgoing calls
10:59.19creativxbe it a sip/iax2 trunk over the interweb, or into a local analog line via zap
11:00.24linageewtf?
11:00.35linageethey intend on branding us all with "National ID cards"?
11:00.37linagee:(
11:00.52dharriganbeautiful. so, we have asterisk now routing traditional calls via "zap" and sip calls via the interweb (always liked that term :) What would be the
11:00.54linagee(USA)
11:00.57KrooksIf I install libpri and I don't need it. does it hurt ?
11:01.01*** join/#asterisk alomelo (n=wyginwys@85.105.204.40)
11:01.21dharriganrecommended upstream bandwidth requirments for say, 8 phones (and say worse case, 8 all answered at once) whilst doing SIP calls?
11:01.25linageemaybe they will associate your national ID number with your phone number. LOL
11:01.53linageedharrigan: 640 kilobits (and then some)
11:02.10key2?
11:02.13creativxdharrigan: depends on codec
11:02.14dharriganlinagee: thanks, is there some type of formula you used?
11:02.21linageeoh yes. and depends on codec. :)
11:02.35linageedharrigan: about 80 kilobits/sec for PCM/ulaw
11:02.41dharrigancreativx: can you help explain about which codec is best (I know what a codec is)
11:02.44creativxit also depends if you are trunking or not
11:02.45linageeafaik.
11:02.50linageeit might jump to 90 kilobits
11:02.53creativxdharrigan: theres some good articles on bandwidth calculations
11:02.58creativxi just dont remember them now hehe
11:03.01dharrigancreativx: great, can you link me?
11:03.04linageeymmv
11:03.15creativxasteriskguru had some i think, voip-info also
11:03.45linageedharrigan: whatever your ISP rates your upstream as, make sure you multiply by 80% or so for a real world number. ;)
11:03.45*** join/#asterisk skyphyr (n=alanj@135.196.58.222)
11:04.32dharriganwould 512kbs be enough for about 8 simultaneous calls?
11:04.40creativxdoubtfully
11:04.47linageekilobitseconds? whoa
11:04.54linageetime and bandwidth in the same dimension
11:04.54skyphyrhi all - sorry for the dumb question - I'm just setting up my second asterisk box and going with 1.4.5 for it - wasn't sure how much of my knowledge from 1.2 would be applicable (and how much of the O'Reilly book) - any particular gotchas I need to be aware of?
11:05.00creativx512 kb/s sure
11:05.00creativxhehe
11:05.01creativxKb
11:05.02creativxKB
11:05.04creativxkBit
11:05.05creativxetc
11:05.12linageeoh. kilobits PER second. :P
11:05.28linagee(yes it does matter)
11:05.44dharrigankilobits per second :-) (just getting the exact upstream figure now)
11:05.53linageecreativx: KB = kilobyte
11:07.01creativxI only deal in megabits
11:07.18dharriganat the moment it's 448 Kbps (this is what my router is telling me)
11:07.35linageecreativx: hehehe
11:07.39dharriganupstream and 4352 Kbps downstream
11:07.43linageecreativx: what about upstream?
11:08.01creativxlinagee: 10/10 fibre
11:08.06linageenice
11:08.07dharrigan448 Kbps upstream, 4352 Kbps downstream
11:08.26linageecreativx: route some of that fiber my way. :)
11:08.32creativxwouldnt wanna try 8 simultaneous lines on that dharrigan
11:08.39creativxyour users might be angry/disappointed
11:08.40dharrigancreativx: :-)
11:08.47creativxlinagee: np let me just reroute the linknet
11:08.52dharriganHow many lines, say about 3?
11:08.58linageedharrigan: like i said. no LESS than 640 kilobits/sec upstream for 8 lines
11:09.08linageeideally 720 kilobits/sec up
11:09.52linageedharrigan: i wouldn't run more than 4 lines off of your stated upstream.
11:10.04dharriganRight, I'm beginning to understand. It's a small office, so really learning lots here about the best way to take. Thanks guys.
11:10.06creativxi would take 448 Kbps with a grain of salt
11:10.19creativxyour ruter sure has a 448 Kbps link to the dslam
11:10.29linageecreativx: i would use traffic shaping to make sure that the 80 kilobits/sec was actually reserved for each phone IP. ;)
11:10.29creativxbut from there, good luck with maxing that =)
11:10.34dharriganlinagee: I think I need to find some formulas for working these things out
11:10.38creativxi would buy more bandwidth
11:10.39creativxhehe
11:10.44linageecreativx: indeed
11:10.48linageeif that's an option
11:10.51linageeget at least 1mbit
11:10.55linageeif not more. hehe
11:10.56dharriganyes, bandwidth is the key here I think.
11:11.04dharriganbig giganormous pipes
11:11.06dharriganfat ones
11:11.28Krookswhen I make linux26 on zaptel I get this -> grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file or directory
11:11.39linageecreativx: in the USA here, they (cox cable modem) would charge me about $100 for 1mbit up, and about $400 for 2mbit up.
11:11.40Krooksmake: *** No rule to make target `linux26'.  Stop.
11:11.53linageecreativx: waaaay higher incrementally more for a slightly more upstream. lol
11:12.05*** join/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il)
11:12.11linageecreativx: it's sick that upstream and downstream don't cost the same. (downstream is typically less than $20 per megabit)
11:12.53creativxinteresting
11:12.58Krooksoh its ok. I upgraded the kernel I forgot to reboot.
11:13.02l2trace9999about 4-6 concurrent providing that no one is doing anything on the internet
11:13.11Krooksthat sucks
11:13.24creativxlinagee: i think our fiber was some 2k usd pr month
11:13.28linageel2trace9999: 4-6 apples?
11:13.33creativxbut they had to dig it first =p
11:13.34l2trace9999yup
11:13.40l2trace9999about 2 smurfs
11:13.42linageecreativx: how much per megabit?
11:14.00linageecreativx: did you say 10 megabit up, 10 megabit down?
11:14.03l2trace9999or 1.66 smurfs
11:14.03creativxyes
11:14.08creativx10 full duplex
11:14.28creativxcant remember what we are paying today
11:14.30linageecreativx: i get like 30mbit up / 30mbit down from my dedicated colo server. ;)
11:14.47creativxwe are hosting in-house :)
11:14.54linagee(with only 500 gigs of total transfer though)
11:15.24linageecreativx: i think it's probably almost always more expensive to host in house
11:15.31creativxit sure is
11:15.42linageeso why do you do it? lol
11:15.44creativxi would prefer having our rack in a mountain hall
11:15.54creativxand remote access cards on all the servers
11:16.06creativxwe are slowly migrating
11:16.20creativxjust bought a new replication/live backup server that we are gonna host offsite
11:16.26creativxso i've started ;)
11:16.29linageecreativx: indeed. remote access cards on servers, service contracts for people to go onsite and repair stuff for you, server monkeys onsite to reboot machines that aren't rebooting through usual means
11:16.55creativxyep
11:17.06*** part/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il)
11:17.21creativxbut we have some internal servers as well as production servers
11:17.29creativxthat wouldnt be that pain-free to move to a different location
11:17.32linageecreativx: then you don't have to pay for CRACs, internet access, floor space, data center staff, etc, etc. (usually all included in one bill. sometimes they make bandwidth a seperate cost)
11:18.31creativxtheres always pros and cons
11:18.44creativxwith the new 2u im already running out of rack space within a year
11:18.55dharriganIf I was to use the 729 codec instead of ulaw, that would lower my bandwidth requirements
11:19.07dharrigan729 seems more efficient in encoding
11:19.30linageedharrigan: test it and see how it sounds
11:19.42dharriganI take it that it sounds, well, not that good ;)
11:19.54linageeymmv
11:20.34linageecreativx: can you believe the interweb (cable modem internet) went down today at 10pm? wtf were they thinking
11:20.53linageeright in the middle of a demo at this guy's house and everything stops working. fuzz on the TV, no cablemodem connection, LOL
11:21.00creativxlinagee: as long as nobody chops our fiber or any of the other backbone fibres im happy
11:21.16linageecreativx: is it dual ring? heh
11:21.18creativxthen our customers tend to get whiney
11:21.36creativxits sad to say it but we have a single point of failure fiber to our serverroom
11:21.45creativxhigh availability my arse
11:22.12linageecreativx: it's highly available as long as you don't cut the fiber and blind all the admins with the laser shooting out. lol
11:22.40linageeimagine the latency on all the admins losing their eye sight and having no other admins within a 500 mile radius. lol
11:22.48*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
11:23.01creativxi can live with blind admins
11:23.06linageehehehe
11:23.13creativxi would not have some 800 real estate agents calling us
11:23.18creativxasking why we are offline
11:23.20dharriganbtw all, here's the link for some bandwidth calculations : http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml
11:24.47linageecreativx: do people still think that when you have web panels, there's someone behind the scenes regenerating the way the website looks for you, then sends it back? LOL. millions of monkeys on typewriters.
11:25.14creativxi dont know what a web panel is :>
11:25.23linageecreativx: say like iGoogle
11:25.44*** join/#asterisk torch (n=logan@200.99.7.98)
11:26.28linageecreativx: hundreds of monkeys
11:26.40creativxah
11:26.58creativxi should get into web panels
11:27.29linageecreativx: i could say "DIV layer with some controls on it", but that's being a bit technical. i was trying to use a generic term. ;)
11:27.39creativxyes yes
11:27.50creativxi have a great need of simplifying our gui
11:27.51torchhi all ... just got my asterisk+digium TE110P working .... and now I'd like to have a integrated phonebook ...how can I do that? anyone?
11:28.36torchbasically ...I want all my pstn users and voip users ...with the same phonebook ....
11:29.31linageecreativx: igoogle panel for asterisk control! lol
11:31.02*** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca)
11:31.12tzangermorning
11:31.17tzangercitats: ping
11:31.25torchmorning ... :-)
11:31.36creativxlinagee: i was thinking more of the lines of our other app
11:31.54Krooksmake: *** No rule to make target `linux26'.  Stop.  <-- I get this when i make linux26 in zaptel's dir .
11:32.10Krookswhats the usuall reason for this ?
11:32.25tzangerKrooks: just use make
11:33.44Krooksno need linux26. The manual says do make linux26.
11:34.04Krooksmaybe thats old
11:34.11creativxyeah they fixed it in make
11:34.14creativxit autodetects
11:34.21Krooksah ok. thanks
11:34.46Krookssomone better update all the manuals out there. they all says linux26.:)
11:34.58tzangerKrooks: yeah but nobody is supposed to read the documentation
11:35.05tzangerthey're supposed to come in here and ask obvious questions
11:35.47Krooks:)
11:35.54tzangeryou're breaking ranks, private
11:36.21torchso guys ... no idea about how to implement a integreted phonebook ? (pstn<->voip)
11:36.54tzangerwhat do you mean, integrated
11:37.29Krookswow, make install zaptel downloading stuffs from digium.
11:37.40Krooksnever seen make install does that kind of stuffs
11:37.41tzanger?
11:37.46tzangerzaptel shouldn't do that
11:37.48tzangerasteirsk will
11:38.08tzangerthe ec binaries are in the tarball
11:38.23Krooksmaybe you should start reading the manual
11:38.50tzangerwhich manual are you talking about
11:39.08torchwell ... someone from a asterisk extension ..will have a phonebook (got from somewhere) ..will all pstn extensions ..
11:39.11torchand vice-versa
11:39.21KrooksI don't know. you're the general.
11:39.25tzangertorch: I'm still not understanding
11:39.29tzangerasterisk has a directory app
11:39.35tzangerand it's trivial to have speed dials
11:41.57torchtzanger. Well .. let's say I have my softphone ... and now I want to open my phonebook right?! ..I want to see all my pstn extensions without having to add them by hand ..
11:42.14tzangeruse a web based click-to-dial then
11:42.30tzangeror use a softphone which has remote directory capabilities (LDAP or provisioned like polycom's)
11:42.43Krooksis it easier to set up asterisk on Centos or on Suse ?
11:42.53tzangerKrooks: I find it trivial to install asterisk on slackware
11:42.54torchhmmm ..that's sounds much better ...
11:42.56tzangereverything else gets in the way
11:43.05tzangerbut I'm a crotchety old man
11:43.05torchand suggestions about which softphone ?
11:43.17tzangertorch: don't know.  don't use softphones for any serious work :-)
11:43.17torchI mean ..that has the remote directory capability
11:43.25tzangerI've used twinkle on linux and... well that's about it
11:43.33*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
11:43.36torchright ..right ...
11:43.43Krooksbut between Centos and Suse, which one would be more trivial ?
11:43.48tzangerno idea
11:43.58tzangerif you're installing from source, I imagine they'd be hte same
11:44.13tzangercentos seems to be what most of the asterisk know-it-alls here use
11:44.17KrooksYou know yesteday, I mentioned twinkle and nobody has ever heard of it. Some laughed at me.
11:44.23tzangerKrooks: I like it
11:44.27tzangerit's not a fucking gnome application
11:44.34tzangerand it's not half bad
11:44.38Krooksyeah fucking gnome
11:44.45tzanger(not half good either, but then again every softphone falls into that category IMO)
11:44.48Krookshate the interface.
11:44.57tzangerif the softphones skin themselves to look like a phone... instant trashbin
11:45.04tzangerif the softphones take over my desktop... instant trashbin
11:45.07Krookshate the gloomy color
11:45.18torchok tzanger ...one more question .. .would it be possible to have asterisk to use a ldap directory ?
11:45.19Krooksidedisk is good
11:45.25tzangeryes there is res_ldap
11:45.29tzangerI have yet to play with it
11:45.32tzangerbut it looks promising
11:45.39tzangeridefisk?  yeah but it's iax2 :-)
11:45.45tzangerI gave up on that protocol a long time ago
11:45.50tzangerI used to be a HUGE proponent of it
11:46.07tzangerthen Olle went and gave Asterisk symmetric RTP and almost every excuse I had to hate SIP went away
11:46.28KrooksI tried xlite with sip yesterday it wasn't as good as idedisk with iax
11:46.33tzangeraround that same time file introduced IAX2 multithreading and since I was using trunk, it caused me severe problems, so the jump to SIP was pretty much a no-brainer
11:46.52tzangerxlite falls into the "skins itself into a phone on my desktop" category
11:46.55tzangerfucking idiotic
11:47.01tzangerI wish I could bitchslap people over the internet
11:47.22tzangerwhat <slap> the <slap> fuck <slap> where <slap> you <slap> thinking <slap>
11:47.35Krookshehe
11:47.49tzangeroh yeah, if I can't control the softphone from dcop or dbus, instant trashbin
11:48.11creativxwhat is dcop
11:48.11Krookskde has no softphone app ?
11:48.13creativxor dbus
11:48.17tzangerbonus points if I can run the softphone entirely from CLI and use aforementioned dcop/dbus to make my own screen
11:48.20tzangerKrooks: twinkle
11:48.31tzangercreativx: message busses
11:48.34creativxah i see
11:48.36Krookstwinkle use kdelibs ?
11:48.43creativxyeah that would be the entire point of a softphone wouldnt it
11:48.44tzangerKrooks: I *think* so
11:48.45creativxfull control of it
11:48.54creativxthen integrate the crap outta it in your crm/whatever
11:49.02Krooksmaybe its just Qt,
11:49.09tzangerno it does not, it's only qt
11:49.13tzangerI just ldd'd my twinkle app :-)
11:49.20tzangerI thought it could use kde as an option
11:49.32tzangerI'm actually more or less happy with twinkle
11:49.37tzangerqt-only means win32-able
11:49.39Krooksjust Qt maybe bcuz for portibility sake
11:49.41tzangerand it's not too bad
11:50.32tzangerafctually
11:50.35tzangerit does have KDE integration
11:50.38tzanger1.0.1 (released 19 may)
11:50.39*** join/#asterisk matsk (n=mk@194.68.102.174)
11:50.43tzanger- Preload KAddressbook at startup.
11:54.59Krooksis there a asterisk-gui tarball ?
11:58.45Krooksok done asterisk -vvvvvr
11:58.50Krooksdone reload
11:59.02Krooksgot the CLI>  prompt
11:59.10Krooksnow what ?
11:59.27KrooksI guess I got to read the manual sooner or later.
12:04.27Krookswhats the thing I have to configure so that a softphone can connect to it ?
12:08.46Krookssip.conf is for sip and iax.conf is for iax. so if I'm using xlite. I edit sip.conf, right ?
12:09.37creativxyes
12:09.48creativxthis is covered by asteriskdocs, wiki, etc
12:13.58*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
12:15.22Krooksok I'm all set
12:17.14*** join/#asterisk magikxx (n=tawandax@196.15.164.11)
12:19.45*** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net)
12:28.33*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:30.59Krooksfuck fuck fuck
12:31.33Krooksx-lite want libstdc++.so.5 and centos 5 has .6
12:31.51Krookssoft linking it to .6 does not work.
12:32.21ManxPowermany distros have older verisons of that available
12:34.30Krooksah I knew its going to bite me for going wiht centos 5.
12:34.37Krookssomeone in here did warn me.
12:35.02Krooksidedisk does not use sip ?
12:35.05ManxPowerSo Centos 5 does not have the older library avialable?
12:35.14Krooksgood question
12:35.46Krookswill there be a conflict ?
12:35.53ManxPowerno.
12:36.01Krooksoh
12:36.08Krookslemme see
12:36.39ManxPowerI had to use Mandriva's older C++ libs for an application and it worked just fine, not conflicts.
12:36.42[TK]D-FenderKrooks: Yes, idefisk does SIP, and on *nix, you should be using Ekiga anyways
12:37.49KrooksManxPower: could I use centos4's libstdc++ rpm then ?
12:38.06ManxPowerKrooks: No.
12:38.11Krooks[TK]D-Fender: let me see
12:38.18KrooksEkiga
12:38.44KrooksI'm familair with xlite and idedisk.
12:38.49ManxPowerI just said that many distros also include older stdlibc++.  Do whatever you would do to search for all packages with libstdc++ in the package name.
12:38.53Krooksanyway, I'll give a try
12:39.05ManxPoweron Mandriva the command would be "urpmi -y stdc++"
12:39.37KrooksI'm on centos 5. urpmi won't work here
12:39.45[TK]D-FenderKrooks: X-Lite is crap and limited.  Idefisk is better & multi-protocol, Ekiga = 100% free and supports video, etc as well
12:40.10ManxPowerKrooks: Mandriva has both libc++ 5 and 6 available.  The default is to use 6, but you can also install 5 I think the Mandrive package is called libstdc++-compat or something like that.
12:40.31ManxPowerKrooks:  I know urpmi does not work on Centos.  I was just giving an example.
12:40.47*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
12:40.53KrooksManxPower: I'll search for it but [TK]D-Fender has kinda swayed my mind away from xlite
12:41.25ManxPowerAll softphones suck. (tm) (c) 2007
12:41.33KrooksManxPower: I know you know. I was just patronizing you, sorry.
12:42.15KrooksYou can't trademark that. That was going to be my line.
12:42.38ManxPowerKrooks: I've been using Asterisk much longer than you have 8-)
12:43.09[TK]D-FenderKrooks: True, all soft-phones suck, but X-lite is actually at the far end of the list :)
12:43.13Krookscan't argue with that
12:43.33*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
12:45.29*** join/#asterisk cayorde (n=flexable@host121-81-dynamic.17-87-r.retail.telecomitalia.it)
12:45.37*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
12:46.21KrooksWooo hoooo. ekiga is in the repository.
12:46.34KrooksThanks [TK]D-Fender
12:46.38[TK]D-FenderKrooks: np
12:47.10[TK]D-FenderKrooks: Keep in mind the only uses you should have for a soft-phone is for a roaming laptop, or for basic internal testing of *
12:47.36[TK]D-FenderKrooks: Any fixed location past that deserves at least an ATA for the pittance they cost.
12:47.46ReDNeQhey [TK]D-Fender how you doing..
12:47.57[TK]D-FenderReDNeQ: still breathing.....
12:48.18ReDNeQyou know the command to make the bots spit out the ports for nat or hte link to get me to the right doc
12:48.50[TK]D-Fender~sipnat
12:48.58jbotit has been said that sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
12:49.01[TK]D-FenderReDNeQ: 5060,10000-20000 all UDP
12:49.05ReDNeQthanks...
12:49.09ManxPower[TK]D-Fender: I disagree about using a softphone for testing Asterisk.   Softphones give Asterisk a bad reputation.
12:50.08purserjmany times, a softphone is all we've got to test
12:50.44[TK]D-FenderManxPower: This is a serious "don't bitch about it" item.  People want to experience * and do so for free for a tiny bit at the start without having to worry about wiring, costs, etc while not knowing what to buy.  And I cannot agree about it giving * a bad rep.... You don't see people in here complaining about them much.
12:51.10bastyanyone got a CCBS Script (Callback on Busy) ?
12:51.24[TK]D-FenderManxPower: Only points have been that "X-lite has no native transfer?!?!", "no G.729 for free?!", and even that is damn infrequent
12:51.39[TK]D-Fenderbasty: Don't bet on it.
12:52.13[TK]D-FenderManxPower: Softphone lets you get right into * fast and easy.  3 parms no wiring, no running out of switch ports, etc.
12:52.29*** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca)
12:52.49Krookshmm, when I run ekiga for the first time. It says there an app thats already using the sip port. Could it be its refering to asterisk cause asterisk IS running on that box.
12:52.51basty[TK]D-Fender: so..there is no solution for that feature ?
12:52.53[TK]D-FenderManxPower: BUT once they have used it through the end of that purpose they should go and get at least an ATA
12:53.05ManxPowerKrooks: That is correct.
12:53.17[TK]D-Fenderbasty: Sure there is, more like don't bet on anyone HERE having it coded to hand to you.  Get busy!
12:53.20ManxPoweryou don't run multiple applications using the same port on the same machine.
12:53.42basty[TK]D-Fender: So can you tell me where to get it ? =)
12:53.48jkiffI initially used couple softphones + my desk phone to test my first dialplan since I didn't have desk space/wiring/power for three phones at my desk.  :-P
12:53.49[TK]D-FenderKrooks: Run Ekiga on 5061.  in your soft-phones sip.conf entry you will specify "port=5061"
12:53.50jkiffIt worked well.
12:54.01KrooksManxPower: SO it mean generally I can't use a softphone on a asterisk machine.
12:54.09ManxPowerjkiff: I did that too.  I almost stopped using VoIP because of it.
12:54.16[TK]D-Fenderjkiff: And did you get PAST that and get some hardware?
12:54.25[TK]D-FenderKrooks: Yes you can, see above
12:54.41jkiff[TK]D-Fender: Indeed.
12:55.14Krooks[TK]D-Fender:  you mean in sip.conf for the user that gonna use Ekiga I put the line "port=5061" , is that right ?
12:55.30jkiffManxPower: How come?  My experience wasn't *that* bad.
12:55.45KrooksI mean in my case
12:55.48ManxPowerjkiff: terrible audio quality, horrid user interface
12:56.18[TK]D-FenderKrooks: Correct.  No issue
12:56.43[TK]D-FenderManxPower: I've had flawless quality.  It varies with super shitty sound cards and your headset.
12:56.57ManxPower[TK]D-Fender: Exactly
12:57.08[TK]D-FenderManxPower: But no argument that any user interface IS cumbersomes compared to any hardware.
12:57.13ManxPowerOther than the user interface issues, most softphone issues are hardware or OS issues.
12:57.39ManxPower[TK]D-Fender: Why?  You do not need a massivly confusing interface to just dial a phone number
12:57.41[TK]D-FenderManxPower: curmudgeon-- ;)
12:57.57ManxPower[TK]D-Fender: no.  laxy.
12:57.58ManxPower..er..
12:58.00[TK]D-FenderManxPower: C'mon.....
12:58.01ManxPowerlazy that is
12:58.20[TK]D-FenderManxPower: Ok, you're right.  The IAXY is a flaming piece of shit ;)
12:58.33ManxPower[TK]D-Fender: at least we agree on that point 8-)
12:59.17[TK]D-FenderOnly provisioned (?!) by * and only usable by it??!?! ISK
12:59.20[TK]D-FenderICP*
12:59.30[TK]D-FenderICK*
12:59.33[TK]D-Fenderkldjhfasafasdsfdasfdaslfdhfiosewyr
12:59.58ManxPower[TK]D-Fender: I would have said that lack of codecs and DNS was the biggest issue
13:00.09ManxPoweroh, and price
13:00.20ManxPowerBBIAW
13:00.29[TK]D-FenderManxPower: And look/feel/call features/number of ports
13:00.42[TK]D-Fenderevery/other/point/too~!
13:02.19jkiffManxPower: Oh yeah, I've never used a soft phone who's interface didn't want to make me vomit.
13:07.13Krooks[TK]D-Fender: on ekiga , the registra is the ip on asterisk ?
13:07.21*** join/#asterisk kannan (n=kannan@61.17.178.244)
13:07.27[TK]D-FenderKrooks: sounds about right
13:07.48[TK]D-FenderKrooks: Which would be "localhost" or the like in your case
13:07.57Krooks[TK]D-Fender: yep
13:08.30Krooks[TK]D-Fender: on themain screen after sip:
13:08.46KrooksI type the ip of the macghine also ?
13:09.30[TK]D-FenderKrooks: You mean to dial?
13:09.44Krooksyes
13:10.01[TK]D-FenderKrooks: No, you should be able to enter just teh # to dial and take its info from reg.
13:10.10Krooks[TK]D-Fender: I guess somewhere in Ekiga I have to tell it to use a different port too.
13:10.26[TK]D-FenderKrooks: shouldn't
13:10.29Krookslike 4321 for user test
13:10.59*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
13:11.22kannanhello , all . I want to use asterisk to connect outbound calls to an IVR of a bank that asks to enter account number followed by # key. I changed blinxfer in features.conf to ## in [featuremap] and re-started asterisk , but still the call gets blind transferred on a single # itself
13:11.24*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:11.37Krooksthe number to dial
13:12.11Krooks[test]
13:12.11Krookstype=friend
13:12.11Krooksusername=test
13:12.11Krookssecret=pwd_test
13:12.11Krookshost=dynamic
13:12.12Krookscontext=tutorial
13:12.14Krooksport=5061
13:12.18Krooksoops sorry
13:12.27Krookswhats the number to dial here ?
13:13.18KrooksI got user ivan already registered from one box using xlite.
13:13.39KrooksBloody user test is on centos 5 and on the asterisk server itself
13:14.36KrooksI'm so close
13:16.31[TK]D-FenderKrooks: Thats only your SIP device DEFINITION.  What you can DIAL is a whole other world.  That is extensions.conf and is the HEART of Asterisk.
13:17.48kannan[TK]D=Fender -> any ideas on why i am not able to change blind xfer keys?
13:18.08Krooksexten => 4321,1,Dial(SIP/test)
13:18.22Krooksso what do I dial for this.
13:19.29Krooksbut test is not registered yet
13:19.34Krooksivan is registered
13:20.11[TK]D-FenderKrooks: Doesn't matter
13:20.27[TK]D-FenderKrooks: You should at least be able to dial that exten and watch it FAIL.
13:20.44[TK]D-FenderKrooks: Fialing to have the call acknowledged is something else entirely
13:21.02[TK]D-FenderKrooks: You'd dial "4321"
13:21.13[TK]D-FenderKrooks: Pastebin is your friend <----
13:21.15[TK]D-Fender~pb
13:21.24jbotwell, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
13:21.25[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^^^^
13:22.26*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:22.38Krooks[TK]D-Fender: yes I can see it fail
13:22.45Krookslook
13:23.01KrooksDial("SIP/ivan-0943a948", "SIP/test") in new stack
13:23.01Krooks[Jun 29 21:20:37] WARNING[20538]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
13:23.18Krooksivan is calling test
13:23.24[TK]D-FenderThats absoluetly fine.  You yourself said that you don't have test yet.
13:23.42KrooksI think its the port thing
13:23.43[TK]D-FenderKrooks: And * can't dial them accordingly.
13:24.02[TK]D-FenderKrooks: If its not registered * can't dial them.  This is NORMAL.
13:25.00pigpen[TK]D-Fender, I am having issues with presence, polycom's and * 1.4.4.  Would you have some time?
13:25.08[TK]D-Fenderpigpen: Ask away
13:25.21pigpenk.  gathering facts...
13:27.02Krookswhy can't test register with asterisk ? is there somewhere I have to specify test to use port 5061 ?
13:27.34pigpenHere we go:
13:28.07pigpensip.conf:  limitonpeer=yes / call-limit=5 / busy-limit=1
13:28.32pigpenAsterisk CLI:  core show hints:   "Watchers 1"
13:28.45pigpensip.cfg:  feature.1.name="presence" feature.1.enabled="1"
13:29.00pigpenpolycom sip ver:  2.1.1
13:29.12pigpenNo buddy watch / presence.
13:29.25[TK]D-Fenderpigpen: pastebin it all.
13:29.28pigpenk.
13:29.30[TK]D-FenderKrooks: You too.
13:29.46[TK]D-FenderKrooks: Along with add : "sip show peers"
13:29.51*** join/#asterisk anthm (n=anthm@dhcp64-134-34-217.bwic.chi.wayport.net)
13:29.51*** mode/#asterisk [+o anthm] by ChanServ
13:30.33*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
13:30.33*** mode/#asterisk [+o mog] by ChanServ
13:30.47rob0I thought port was a global setting in sip.conf ... can you set that per section?
13:31.27[TK]D-Fenderrob0: Ye, it is phone specific.
13:31.41[TK]D-Fenderrob0: If you've ever set up a dual port ATA you should know that.
13:32.13*** part/#asterisk Andri[IS] (n=andri@hydrogen.calidris.com)
13:32.19*** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca)
13:32.31rob0I have a dual port ATA, but I didn't do anything special on the * side, they just register and tell * the port.
13:33.37Krookshttp://pastebin.ca/595321
13:33.42[TK]D-Fenderrob0: You should tell it the port...
13:33.49KrooksPlease take a look at http://pastebin.ca/595321
13:34.20[TK]D-FenderKrooks: ok, test isn't registered.  Thats your Ekiga, isn't it?
13:35.01pigpenhttp://pastebin.ca/595323
13:35.20*** join/#asterisk kvit123 (n=kvit123@203.209.31.86)
13:35.23[TK]D-FenderKrooks: Oh, and ditch the "username=" in your sip entries.  Not needed
13:35.30Krooksyes
13:35.31pigpenI did not past the sip.cfg, phone.cfg, nor the <mac>.cfg, if needed, let me know.
13:35.47rob0Hmmm, maybe that explains my 4-8 second delay when dialing ...
13:35.52Krooksis that the cause if the problem ?
13:37.36[TK]D-Fenderpigpen: move "subscribecontext=from-sip", "call-limit=5" into each phone's entry and out of [general], and ditch "busy-limit=1", and "limitonpeer=yes".  Also, your phones should now be "type=peer"
13:37.59[TK]D-FenderKrooks: Try dialing fred from your phone, and check your reg settings.
13:38.07pigpenk, did the type=peer change with * 1.4 ?
13:38.20[TK]D-Fenderpigpen: Yeah... various silly things...
13:38.20Krooksfred ?
13:38.25Krookswho is fred ?
13:38.33pigpenk.  kinda figured.
13:38.38pigpenfred is dead.
13:38.49[TK]D-FenderKrooks: Err..."ivan" ..... I have no explanation for the name shift ;)
13:38.50tzangerDave's not here!
13:39.05[TK]D-Fendertzanger: ".... what are you doing, Dave?"
13:39.12tzangerwrong dave
13:39.18tzanger"No it's be, Dave, I got the stuff"
13:39.21Krooks[TK]D-Fender: security check error
13:39.22tzangerer it's me
13:39.41Krooks[TK]D-Fender: test is not regiostered. how can he call ivan .
13:39.45pigpenDave is in line at the apple store, waiting for his iPhone....
13:40.05[TK]D-FenderKrooks: You don't actually need to be registered to place a call...
13:40.20Krooks[TK]D-Fender: Does this all got to do anyting with the port thing earlier on?
13:40.27[TK]D-FenderKrooks: enable "sip debug" and pastebin your reg attempt.
13:40.53[TK]D-FenderKrooks: it might.  You also didn't specify port 5061 in your entry, and I don't know if you screwed something else up in your Ekiga settings.
13:41.03[TK]D-FenderKrooks: I've used ti maybe twice... and don't have it here
13:41.19[TK]D-Fenderpigpen: OpenMoko FTW
13:41.24*** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar)
13:41.26[TK]D-Fenderpigpen: I'd buy one.
13:42.37Krooks[TK]D-Fender: didn't specify where ?
13:42.39*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
13:42.40[TK]D-FenderKrooks: OH... and ditch THIS :" bindaddr = 192.168.100.1" for this : "bindaddr = 0.0.0.0" and completely restart *
13:42.52[TK]D-FenderKrooks: under [test]
13:43.19*** join/#asterisk msetim (n=msetim@200.195.161.164)
13:43.27pigpenopenmoko?
13:43.32pigpenok...googleing.
13:43.52Krooksline 16   port=5061
13:43.52pigpenshit...what did you just get me into.
13:43.54[TK]D-Fenderpigpen: www.openmoko.org
13:43.59[TK]D-Fenderpigpen: www.openmoko.com as well
13:44.23[TK]D-FenderKrooks: vnm I'm f'n blind today ;)
13:44.29[TK]D-FenderKrooks: But check out the rest!
13:44.35[TK]D-Fendernvm*
13:44.37*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
13:44.57pigpen[TK]D-Fender, any hardware out for it?
13:45.29[TK]D-Fenderpigpen: it IS hardware.  Arrives this fall
13:45.49[TK]D-Fenderpigpen: One Smart-Phone to rule them all!
13:45.57coppicei wonder if it will ever really arrive. it seems rather late
13:46.01pigpenYeah..as you want it, program it.
13:46.14pigpenit has to sync with my mac!  haha
13:46.15[TK]D-Fendercoppice: iPhone isn't entirely here yet either, big deal.
13:46.32Krooksoh man
13:46.39[TK]D-Fender^%@# MAC.  Its pretty and all but I refuse to let my devices own ME.
13:46.57pigpenlooks sweet...I hope the openmoko comes with more storage.
13:47.07pigpenok..back to presence...
13:47.10[TK]D-Fenderpigpen: SD/microSD <-
13:47.14coppiceisn't the iPhone supposed to be here a few hours from now?
13:47.17[TK]D-Fenderpigpen: and more to come.
13:47.32syzygyBSDI have one right here
13:47.36pigpencoppice, yeah...6:00pm
13:47.44syzygyBSDok.. maybe not
13:47.49[TK]D-Fendercoppice: Oh... and like the iPhone doesn't own your ass with the bundled PLAN, lack of 3rd aprty apps, and do you trust their batter, or lack of SIM? :)
13:47.52pigpen[TK]D-Fender, thanks for the tip.
13:47.56syzygyBSDdirty $500 phones
13:48.05[TK]D-Fenderbatter*
13:48.06syzygyBSDthink I will just go to japan and get one from there
13:48.09[TK]D-Fenderbattery*
13:48.11[TK]D-Fenderaskldsdhfafdshasf
13:48.18coppiceI think the iPhone is more likely to cut your ass with broken glass
13:48.29syzygyBSDIphone doesn't have a sim?
13:48.37syzygyBSDI thought it was gsm...
13:48.39syzygyBSDhmmm
13:48.40coppiceit does have a sim
13:48.41[TK]D-FendersyzygyBSD: Built-in only.
13:48.51[TK]D-FendersyzygyBSD: Foget access or exchange :)
13:48.54KrooksI got it to work
13:48.58coppiceno. a normal plug in sim
13:49.02[TK]D-FendersyzygyBSD: Your provider will OWN YOU <
13:49.23[TK]D-FendersyzygyBSD: And have you read about the nasty termination fee even iff you paid FULL PRICE for it?
13:49.24syzygyBSDwhich is why I switched from cingular shortly after they bought at&t
13:49.32[TK]D-Fendercoppice: I've read differently on that.
13:49.56coppiceits a software locked SIM like almost every other phone in north america
13:49.58syzygyBSDhad the best plan I could imagine with at&t though
13:50.23coppiceliving where I do, I find it hard to imagine everyone tolerating locked phone, and tied packages
13:50.43syzygyBSDI haven't had a locked phone for 3 years
13:51.20pigpen[TK]D-Fender, buddy watch didn't work with a simple reload, so I am rebooting the polycom's, as this seemed to be a "thing" with previous ver's of *.
13:51.49coppicea locked phone would be about as popular as vista here
13:52.40syzygyBSDlol where you at?
13:52.54coppiceHK
13:52.58[TK]D-Fendercoppice: You shouldn't joke about Vista like that.  They sold dozens of copies!
13:53.02syzygyBSDahh, that makes sense
13:53.09syzygyBSDI have one of them!
13:53.19syzygyBSDcourse, it sits on my test box unused...
13:53.55*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
13:53.56[TK]D-FenderMy phone is rather locked, but very nice, was heavily subsidized and my plan is SICK :)
13:54.33naitram[TK]D-Fender: yesterday, you were trying to help me fix a problem with automon from and active sip call. You suggested the Read() app to see if i got correct dtmf tone awareness from *. I do, it works.
13:54.54[TK]D-Fendernaitram: Ok, well keep wiki-ing, I have no personaly experience with it.
13:55.22naitramThe monitor app also works from the dial plan, I just can't get anything invoked from an active call to work
13:56.26naitramanyone, have experience setting up 1 touch monitoring with sip clients
13:56.27*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
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13:59.46*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
13:59.58anthmI never set it up but I wrote that feature =p
14:01.27naitramanthm: any idea why I cant start any app on an active sip call via dtmf tones?
14:02.03*** part/#asterisk scud (n=scud@c-68-62-219-34.hsd1.al.comcast.net)
14:02.13anthmall I can remember is you need the flag in your dial app D or d or something for allow dtmf and a key defined in the config file
14:02.48anthmif you choose * you sometimes end up out of luck cos that one is hard coded in a few places esp before the call is answered
14:02.57BugKhaMI got  BPV count: 333        CRC4 error count: 4        E-bit error count: 986        FAS error count: 6
14:03.01BugKhaMin my    /proc/zaptel/3
14:04.10BugKhaMI just made E1 Cross over cable to link btw span 3 & 4
14:05.01naitramanthm: so, which sources files?
14:05.13kannananthm -> maybe you can surely help me, i set blindxfer to ##, but it still transfers on a single #
14:05.41kannaninstead of sending the # as a DTMF
14:06.17kannani have defined tT flags in the Dial cmd
14:07.36pigpen[TK]D-Fender, thanks...that did the trick.
14:08.12naitramso, are these features broken, mis-documented or what?
14:08.15torchguys .. got a weird noise when dialing from my pstn to my asterisk extensions ... the integretion is mad using a Digium TE110P
14:09.04torchwhen in asterisk CLI I do 'zap show status' the IRQ is quite high ... ideas?
14:09.10*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-154.buff.east.verizon.net)
14:12.54tzangerwhat... the... fuck
14:12.58tzangerI've got PRI DTMF issues again
14:13.19tzangerand my system hasn't changed or even bene restarted in what, 2 weeks
14:14.58*** join/#asterisk phillipk (n=pkey@216.248.143.87)
14:17.13*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net)
14:17.19BSD_Techok morning
14:17.29*** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com)
14:17.30*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
14:17.30BSD_TechI have a issue driving batty
14:18.05BSD_Techeverytime I hit the #key  it says please enter the extension then press pound
14:19.01*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
14:19.01*** mode/#asterisk [+o blitzrage] by ChanServ
14:19.01BSD_Techwhere is this defined I have nothing in my dialplan defining the # sign
14:19.01IOscannerYou have transfer enabled in the gloabal settings
14:19.01BugKhaMis it possible to make a cross over E1 to link between two ports of my quad cards?
14:19.01IOscanneryou using AH or Trixbox?
14:19.01BugKhaMI just tried and it gave me some errors
14:19.07BSD_Technope
14:19.14BSD_Techjust aasterisk 1.4.5
14:19.15_VoiceMeUp_COMBSD_Tech .. featurs.conf ?
14:19.39BSD_Techno # key enabled but double checking
14:20.35IOscannerI have a problem.  I have a queue and when people call in it rings once then sends the callers back to the IVR
14:21.08[TK]D-FenderBSD_Tech: "tT" in youd dial statement....
14:21.41BSD_Technot #  enables in feathers.conf
14:22.11IOscannerI even have ringall for the ring strategy.  I do know that fewest calls does this.
14:22.20[TK]D-FenderBSD_Tech: Are you using either of those dial options?
14:22.27BSD_Techno
14:22.36[TK]D-FenderIOscanner: pastebin is your FRIEND <----------------
14:22.42BSD_Techthis is a standard out of the box asterisknow setup
14:22.51[TK]D-FenderBSD_Tech: show some CLI output.
14:23.05*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
14:23.09[TK]D-FenderBSD_Tech: GUI?! *shudder* Trust--
14:23.52kannanBSD_Tech
14:24.11kannanBSD_Tech ->  it is the default blindxfer defined in features.conf
14:24.19kannanlol, i am asking about this too
14:24.32kannanhow to change the sequence in featuremap
14:24.37kannanit is not updating for me
14:24.52[TK]D-Fenderpigpen: You're welcome
14:25.09kannani need the tTo , as i am using vicidial
14:25.20[TK]D-Fenderkannan: pastebin EVERYTHING after restarting * and retesting.
14:25.25pigpenOpenMoko is pretty cool.
14:25.35[TK]D-Fenderpigpen: See? :)
14:25.37kannanok sure thanks a lot
14:25.50[TK]D-Fenderkannan: If you want help, we shouldn't even have to ASK <0
14:26.01BSD_Technope
14:26.27BSD_Techhold on brb doorbell
14:26.34[TK]D-FenderBSD_Tech: .... PASTEBIN :)
14:28.22Krookswhat the ccommand to shutdown asterisk ?
14:28.40Krooksexit ?
14:28.42[TK]D-FenderKrooks: "stop now"
14:29.17BugKhaM[TK]D-Fender: sorry to interrupt, u know if I can make a cross over cable and link between two ports of a quad card?
14:29.18Krooksstop gracefully
14:29.30[TK]D-FenderBugKhaM: Sure
14:29.35[TK]D-FenderKrooks:
14:29.42Krooks[TK]D-Fender: I found a laptop with windows I tested it.
14:29.47[TK]D-FenderKrooks: "stop with extreme prejudice" :D
14:29.47*** join/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com)
14:30.02KrooksWonderful . Its like a whole new world of stuffs to play with
14:30.20[TK]D-FenderKrooks: So you did a SIP>SIP call nice and happily?
14:30.24Krooksyes
14:30.26luckyonewhen tryin to connect to my running * process, I get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
14:30.48[TK]D-FenderKrooks: Congrats, not to embellish your setup, add VM, learn to make IVR's, funny prompts, conferencing, etc
14:30.54*** join/#asterisk pogo123 (n=hh@buero-gw.dortmund.loca.net)
14:30.56KrooksOne question . If I'm behind a firewall. The only port I have to open is 5060 ?
14:31.01[TK]D-Fenderluckyone: You either lack rights, or its not running.
14:31.11[TK]D-FenderKrooks: Here :
14:31.13[TK]D-Fender~sipnat
14:31.13jbotsipnat is probably for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:31.15BugKhaM[TK]D-Fender: and /proc/zaptel/* should not have any errors right?
14:31.22luckyonethis happens every time I reboot my machine - I recreated the directy /var/run/asterisk then chown'd it to user asterisk
14:31.25penguinFunkKrooks: 5060 for sip, 10000-20000 for RTP
14:31.26Krooks[TK]D-Fender: I will. Need to read the manual first. I did all this without reading the manual.
14:31.32[TK]D-FenderKrooks: 5060,10000-20000 all UDP and a whack of settings for sip.conf
14:31.55Krooksoh man
14:32.11[TK]D-FenderKrooks: not so bad.  I'm going to prime JBOT for this shortly.
14:32.30Krooksok thanks
14:32.32Krooksbye
14:32.47kannanhmm, i think i know the problem now, lol, lrt me try , brb
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14:34.45BSD_TechI will pick up when I get back
14:34.48BSD_Techthanks
14:35.38luckyone[TK]D-Fender: I think you helped me fix this a few weeks ago so I could reattach to it
14:36.02luckyone[TK]D-Fender: I know it is running, my sip clients can connect to is, calls work, voicemail is up, etc
14:36.54pogo123Hi! I'm trying to use a Zaptel device driver which was originally released for Zaptel 1.2.6 with Zaptel 1.2.16. But the kernel Oops'es as soon as I pick up the phone. Did Digium make any internal changes to Zaptel so that I have to adjust my driver to be compatible with the 1.2.16?
14:37.24luckyone[TK]D-Fender: I don't understand why /var/run/asterisk gets blown away on restart. Further, I don't understand what mods I need to make to /etc/init.d/asterisk so that start runs the application so I can always attach to it....
14:37.42[TK]D-Fenderluckyone: Who are you calling it as?
14:37.50luckyone[TK]D-Fender: root
14:37.53[TK]D-Fenderluckyone: And what distro?
14:38.01luckyone[TK]D-Fender: kubuntu
14:38.38[TK]D-Fenderluckyone: EEK.  They don't even use STD init anymore IIRC.  That new bootup sequence thingy...
14:40.21luckyone[TK]D-Fender: it is somehow getting called automatically, I don't have to run asterisk -vvvc -U asterisk -G asterisk or whatever to start it after reboot
14:41.13[TK]D-Fenderluckyone: This kind of item is out of my league.... maybe someone else will know (they'd have responded most-likely already).  Get goolging....
14:41.36luckyone[TK]D-Fender: hehe - yeah... =)
14:47.20rob0luckyone, it's just bash. Read the script and see what it does. Or, don't use it at all.
14:47.52*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
14:48.23luckyonerob0: I see what it does, it makes a call to safe_asterisk and when I look at that I don't understand why /var/run/asterisk/asterisk.crtl isn't being created
14:48.25rob0You can disable asterisk in whatver you use to manage your services, and then just start it in rc.local as you need.
14:49.50*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
14:52.44magikxxCresl1n around ?
14:55.07*** part/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com)
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15:12.24Traianahi to all, i was wondering if ther is anybody that could help me with a issue with the conference room option?
15:12.39Zeeekask away!
15:13.02Traianadoes the set marked user option allow the room to start when the marked user join the room?
15:13.28*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
15:14.23[TK]D-FenderTraiana: ....HUH?!
15:14.56Zeeekwhat huh?
15:15.03Traianai have turn the wait for marked user and set makred user on using the asterisknow gui but the room still will allow the conversation to start without a marked user joining.
15:17.37[TK]D-FenderTraiana: This is not the GUi support channel, please read the topic
15:18.02Zeeek[TK]D-Fender you're in a good mood :)
15:18.45[TK]D-FenderZeeek: Under my usual quote for capitalization and swearing ;)
15:18.46[TK]D-Fenderquota*
15:19.06[TK]D-FenderZeeek: And I have not figured out mIRC style font control with Chatzilla ;)
15:19.21Zeeekthere's font control?
15:19.35Traianaok, no problem , thank you for the assistance.
15:19.37[TK]D-FenderZeeek: Bold & colour, possibly italics.
15:20.06ZeeekI don't recall that, I only used zilla for a few times. It blew up too many times.
15:20.48[TK]D-FenderZeeek: Very stable for me, and saves on installing too many extra apps.
15:20.51Zeeekkrdian
15:21.00*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
15:21.12ZeeekI like extra apps
15:21.19Zeeekthey introduce nice security holes
15:21.45Zeeekin addition to those furnished with the os
15:22.04Zeeekkrdian hello
15:22.30kannanok i can't get to send the # to the IVR. the call gets disconnected. I have pasted in http://pastebin.ca/595492
15:23.02kannan[TK]D-Fender -> as i am very new to the chats , please excuse any mistakes in my part
15:23.11BSD_TechI found the issue
15:23.27BSD_Techits my macro-user-callerid
15:23.29kannanBSD_Tech , waht is it?
15:23.29[TK]D-FenderZeeek: Gruyere++
15:23.50BSD_TechI ment my issue
15:23.53BSD_Techsorry
15:24.09BSD_Techback to writing and fixing my dial plan
15:24.28anthony]Anyone here own a street bike?
15:24.45Zeeekif it had a GUI...
15:25.10[TK]D-Fenderkannan: What IVR, there IS no IVR in there.
15:25.26Traianais it possible to setup a user to be a moderator of a conference room ??
15:26.01[TK]D-FenderTraiana: the admin has a PIN or you can eter immediately as admin if you choose to do so in your dialplan.
15:28.27kannani meant that i am making an outbounf to an IVR where i have to send the # as dtmf
15:28.33kannanbut when i do it gets cut
15:28.45*** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk)
15:28.59[TK]D-Fenderkannan: Show me the call's complete CLI output at verbose 10
15:29.09kannanok sure
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15:32.01*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
15:34.00kannan[TK]D-Fender - > http://pastebin.ca/595526
15:35.05kannanvery sorry pl ignore i will re-send
15:35.10kannansorry
15:37.15*** join/#asterisk Qwell_ (i=north@pdpc/sponsor/digium/Qwell)
15:37.15*** mode/#asterisk [+o Qwell_] by ChanServ
15:37.32kannanhttp://pastebin.ca/595532
15:37.35*** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
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15:40.52kannanit just shows the call connected, when we send the digits all thew numerals are fine, when we hit # , it gets disconn
15:44.20[TK]D-Fenderkannan: For all I know its the OTHER SIDE that's choosing to hang up on you.
15:45.05kannan:( , when i register an x-lite directly to the service provider , it goes fine
15:45.26kannanthe callee is an automated IVR
15:46.10kannani set the SIPtrunk, so that the tTo is not used in the Dial cmd, but still it gets disconnected on pressing the hash key
15:46.53[TK]D-Fenderkannan: You'll need to include SIP debug info next and if that doesn't say why, then core debug
15:47.19kannanok then , sure thanks alot
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15:57.55BSD_TechZeeek whats the irc channel for the talk shoe
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15:58.21Zeeekasterisk-users-conference
15:58.45daveburrdoes anyone know of an app for scheduling asterisk meet-me sessions?
15:59.30[TK]D-Fenderdaveburr: ....
15:59.35[TK]D-Fender~toywy
15:59.35jboti heard toywy is The one you write yourself.
16:03.29IOscannerI have an IAX trunk between two boxes.  I need to pass a FROM_DID value.  for now it is passing s@....
16:03.38IOscannerhow can I define this in the IAX2 trunk?
16:06.48*** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net)
16:08.21javar:O
16:09.09[TK]D-FenderIOscanner: Look at your DIAL......
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16:16.45iruleh there, if I run install_amp on freepbs installation for the second time, it will not ask me any questions! how can I change this? thanks!
16:16.48*** join/#asterisk mihinomenest (i=aI3P@cerebus.clandestineresearch.com)
16:16.48jerurgh, i'm having a hellova time trying to turn off voicemail for a realtime user stored in a mysql database
16:17.28Corydon76-workUh, "turn off voicemail"?
16:18.29jeryeah. the user has requested to have no voicemail, i can't turn it off (i didn't set up this system ,and i'm only acting as a fill-in until we hire an actual experienced asterisk guy)
16:18.56jerjust driving me up the wall
16:19.10Corydon76-workIs your dialplan stored in the flat file?
16:19.23*** join/#asterisk Here_And_There (n=oldfart2@pool-68-238-252-162.phlapa.fios.verizon.net) [NETSPLIT VICTIM]
16:19.54Corydon76-workBasically, you just need to add a GotoIf prior to the call to Voicemail
16:19.55jerCorydon76-work, yes
16:20.07*** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net)
16:20.23Corydon76-workGotoIf($[${EXTEN} = 123]?hangup)
16:20.32Corydon76-workand then define a hangup label
16:20.51*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
16:20.54Zeeekif anyone has anything to say today, #asterisk-users-conference or http://x2z.eu
16:21.04jerhrmm here i am trying to do it a complicated way
16:21.06ZeeekBeginning in 10
16:21.09jerCorydon76-work, i'll give that a go
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16:21.27*** part/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net)
16:21.31Corydon76-workor, if your Dial timeout is set to a variable, Set(${IF($[${EXTEN} = 123]?dialtimeout:foo)}=9999)
16:21.32Zeeek~seen JerJer
16:21.34jbotjerjer <n=PhatJ@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk, 7d 20h 26m 49s ago, saying: 'dioedu:  that sucks'.
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16:22.14*** mode/#asterisk [+o Qwell[0]] by ChanServ
16:22.40Corydon76-workjer: and, if you want multiple people to be able to do that via a flag, you could create that flag in the db, then query it in the dialplan
16:22.56quidproHmm, anybody know how to stop the voicemail "alter tones" that pop up every couple minutes on Polycoms?
16:23.09Corydon76-workjer: I'd highly recommend that flag if a second person requests the same behavior
16:23.11quidprooops, alert tones
16:23.30jerCorydon76-work, ok, thanks
16:24.12bpiperI am trying to create a PHP script to call a person & read some digits back to them with the manager API but when I try running the script, it just shows "manager logged in" and "manager logged off" and does nothing. Here is my script... would anyone like to give me a hand?
16:24.12bpiperhttp://pastebin.com/939055
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16:30.42matdonhi
16:32.06[TK]D-Fenderquidpro: In sip.cfg set the sound entry to "silence"
16:32.32_VoiceMeUp_COM<PROTECTED>
16:32.37_VoiceMeUp_COMaint that a nat issue ?
16:32.52Traianadoes anybody have any experience in setting a conference room to wait for the moderator to arrive to start?
16:33.35drakoanyone here doing call monitoring with TDMs card?
16:33.59bpiperCan anyone give me a hand with a php script calling to the manager API? http://pastebin.com/939055
16:34.09*** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net)
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16:35.48quidproTK:  Hmm... I can't figure the exact tag... is se.appLocalEnabled="0"  the one?
16:37.07[TK]D-Fenderquidpro: <MSG_WAITING
16:37.15quidproTK:  Let me look...
16:37.26*** join/#asterisk RazaMetaL (n=razameta@200.93.220.27)
16:37.29RazaMetaLhi guys
16:37.47RazaMetaLI've two questions regarding my new asterisk install
16:38.06RazaMetaLis better to use alphabetic extensions or numeric?
16:38.11Strom_Mnumeric
16:38.17Strom_Myou can actually dial those
16:38.30*** join/#asterisk perf3kt (i=perf3kt@149.166.34.171)
16:38.37RazaMetaLi'm planning to integrate my asterisk with jabber and want to use alphabetic usernames at jabber
16:39.05[TK]D-Fenderbpiper: "fputs($socket, "Secret: *****\r\n");" <- Extra "\r\n" is called for IIRC to terminate the login.
16:39.23torchdoes anyone know any softphone to be used in a smartphone like a Palm Treo or P910 sony-ericsson?
16:39.31RazaMetaLI'm testing alphabetic usernames and giving one numeric extension for every username
16:39.33*** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net)
16:39.34RazaMetaLit is correct?
16:39.45twitchnlngood morning everyone
16:39.48bpiperTKD Fender, you're saying I need to make it \r\n\r\n?
16:40.10[TK]D-Fenderbpiper: I believe so
16:40.16bpiperok, thanks, I'll try that
16:40.25RazaMetaLI've another option that consist on install freepbx beta for 1.4 support
16:40.32Strom_MRazaMetaL: are you only going to be using jabber?
16:40.33RazaMetaLwhat do you think is better ...
16:40.43Strom_Mugh, don't touch freepbx
16:40.46iruleh there, if I run install_amp on freepbs installation for the second time, it will not ask me any questions! how can I change this? thanks!
16:40.52RazaMetaLStrom_M, yes.. in the future
16:41.37bpiperthat did the trick... thanks TDK-Fender
16:41.38twitchnlni was looking @ http://www.trixbox.org/forums/trixbox-forums/open-discussion/multiple-queues-agent-login-logout and am having difficulty getting the context setup so that extensions can dial them
16:41.55[TK]D-Fenderirule: You're repeating yourself like a broken record again, and  You already know its not supported here.
16:42.33[TK]D-Fenderbpiper: Np
16:42.48*** part/#asterisk bpiper (n=bpiper@ip-207-145-7-194.atl.megapath.net)
16:42.57drakoirule, #freepbx
16:43.03irule[TK]D-Fender complete silence sounds like nobody was arround at the moment of the first message ;)
16:43.20[TK]D-Fenderirule: You are being an ass and you KNOW it.
16:43.36Strom_Mtwitchnln: #trixbox
16:44.39irulethanks drako
16:44.41RazaMetaLit is recommendable to use freepbx? or you recommend to deal with the .conf files ?
16:44.43twitchnlnthanx
16:45.09perf3ktrazametal: its straight cli here
16:45.39RazaMetaLperf3kt, ok
16:45.45drakocan you use variables like in extensions.conf in voicemail.conf?
16:46.07*** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net)
16:46.45[TK]D-Fenderdrako: Nope.
16:47.06drakodamnit.
16:48.12[TK]D-Fenderdrako: Now trying thinking FORWARD.  What do you want to DO?
16:48.18Zeeekhttp://www.dataevolution.com/dectop%20info%202.htm
16:48.20Zeeekoops
16:49.12vn~book
16:49.13jbotrumour has it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
16:49.55drako[TK]D-Fender, I don't want to repeat this line on every entry ,attach=yes|saycid=no|envelope=no|delete=no|maxmsg=2000
16:50.23[TK]D-Fenderdrako: Copy&paste.. get over it!
16:51.50Strom_Mdrako: you can do it in the general settings too
16:51.54Jinglesyou know, that book is pretty decent for setup, laying out your dialplan, etc.
16:52.05Jingleshowever, there is one major bit of info it seems to lack.
16:52.09*** join/#asterisk guillote_GNU (n=guillote@190.7.30.135)
16:52.14Jinglestroubleshooting call quality issues.
16:52.30vnwe can't get everything!
16:52.31*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
16:52.42vngo back helping Santa Jingles :p
16:53.31Traianacan anybody help me ? I am trying to setup a user to become a leader of a conference room ? any idea on how to do that ??
16:53.52*** join/#asterisk ManxPower (n=manxpowe@90.sub-70-220-188.myvzw.com)
16:54.32[TK]D-FenderTraiana: Go read the INSTRUCTIONS : "show application meetme"
16:54.47drakoStrom_C, hmm good hint.
16:54.59drakoStrom_M, i thought it was only a per line params
16:55.40[TK]D-Fenderdrako: You should occasionally read the SAMPLE files ;)
16:56.21*** join/#asterisk techie (n=gus@antibala.net)
16:56.43drakoyeah
16:57.00drakobut most of the samples files sucks.
16:57.36*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
16:58.46Zeeekhttp://www.lulu.tv/?p=11368
17:00.14[TK]D-Fenderdrako: I never said "use as is", did I?  Hell no, but the options are DOCUMENTED there.
17:00.58drakoyes yes
17:01.04drakoi get ur point, is a good one.
17:01.58*** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net)
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17:03.21[TK]D-FenderZeeek: PRICELESS
17:03.41*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
17:03.47*** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net)
17:04.05Zeeekneeds a little trimming, but you get the idea
17:05.29[TK]D-FenderZeeek: Indeed.... those wires were WAY too long ;)
17:09.55*** join/#asterisk Dj_FlyBy (n=djflyby@mail.imonkeyit.com)
17:09.56*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:16.51_VoiceMeUp_COMwhen does cluecon end
17:18.38*** join/#asterisk Tako-san (n=Tako-san@216.232.147.102)
17:19.43*** join/#asterisk Toerkeium (i=oo@dcc-hq-host-200-59-45-188.dnsba.com)
17:20.46*** join/#asterisk drzed (n=drzed@synflood.homelinux.org)
17:23.42*** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com)
17:23.44ZeeekKarlito
17:24.38neoalexok... now I know most of you guys hate grandstream phones but I'm having a problem with one
17:24.46*** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus)
17:25.31neoalexa gxp-2000 connected to my asterisk, it works fine accept for one thing, when receiving calls from one of my providers I can't here anything
17:25.38[TK]D-Fenderneoalex: And you wonder WHY?
17:25.40neoalexeven though it works fine with all others
17:26.13neoalexI know I know... bought it dirt cheap before I got a chance to ask if it's any good
17:26.26*** join/#asterisk ixela (i=ixela@nat/digium/x-f665871c0ee1aaf5)
17:27.29neoalexso... any ideas... also if I receive the call from that provider on any other extension (softphone, ATA) it works fine
17:27.57*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
17:29.20[TK]D-Fenderneoalex: Then it would clearly sound like a newtowork issue, and I would bet a NAT one as well.
17:29.26*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
17:30.11neoalexthe phone is connected to the asterisk on the same network, local IP, no nat, just like all other clients
17:30.44neoalexand like I said, there's no problem receiving calls from any other provider as well on the grandstream
17:30.57neoalexso it's a pretty localized problem, which is what makes it weird
17:31.55drzedlittle ISDN question: is it correct that only for using a card as NT a HFC-S is necessary?
17:32.17[TK]D-Fenderneoalex: This is the point where I tell you that naturally I don't trust your configs for one second and you should have had them pastebin'd up for use to see before you even MENTIONED what the problem is...
17:32.18[TK]D-Fender~pb
17:32.20jbotA Pastebin is a place to paste your stuff without flooding the channel.  Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org
17:32.30[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
17:32.34IOscannerI have a sip trunk and I can't get it to use the contact as the account or username.  Instead it uses: Contact: <sip:s@69.18.209.xx>
17:32.56[TK]D-FenderIOscanner: Look. At. Your. DIAL.
17:33.31IOscannerThis is NATED, normaly it is fine, but when I nat it I loose the Contact info
17:33.54neoalex[TK]D-Fender: uoook
17:34.01neoalexjust a sec
17:34.17IOscannerI am just trying to registar not even call yet
17:34.41[TK]D-FenderGod helps those who help themselves.  *I* am considerably less forgiving ;)
17:34.58[TK]D-FenderIOscanner: Then look at your REGISTER.
17:35.41*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
17:35.47blitzrage[TK]D-Fender: I DON'T WANT TO KNOW YOUR NAME... I JUST WANT...
17:35.54[TK]D-Fenderblitzrage: ! ! !
17:35.58IOscannerThe registar string has the account number, but it is not using it as the contact when asterisk is natted and using externip
17:36.40blitzrageIOscanner: correct... Jared Smith and I found this at Geek Week in KC about a year ago... when you register, it'll try to register with 's'...
17:36.54blitzragelet me see how I got around that in TFOT2
17:36.58IOscannercorrect
17:37.01IOscannerplease...
17:37.09IOscannerI have to cases of this and it is driving me crazy
17:37.10IOscannerlol
17:37.22blitzrageya, you have to do something kinda funky to register 2 asterisk boxes together
17:37.36[TK]D-FenderIOscanner: Stop talking about it and PASTEBIN <-----------------
17:37.49[TK]D-Fenderblitzrage: Hardly :)
17:37.56IOscannerpaste what: Contact: <sip:s@69.18.209.XX>
17:37.59IOscannerthat is the problem
17:38.19[TK]D-FenderIOscanner: No, that is a sign of the SYMPTOM.  Your CONFIG is the problem (duh)
17:38.25IOscannerblitzrage seems to knwo what is wrong
17:38.50[TK]D-FenderIOscanner: And I'm quite sure as well, I just want the incriminating evidence :)
17:40.09neoalex[TK]D-Fender: http://pastebin.ca/595788
17:40.33*** join/#asterisk basty (n=basty@dome-city-rockers.sunblast.de)
17:40.36blitzrageIOscanner: http://www.pastebin.ca/595790
17:40.39bastyHi
17:41.01blitzrageIOscanner: fix it quickly -- that post will expire in 30 mins :)
17:41.34bastyI have around 115 SIP-Phones connected to my Asterisk PBX..now if I do a "sip show channels" I usually see 30-40 - is that normal?
17:41.43[TK]D-FenderIOscanner: Paster your damn REGISTER LINE.  Do I have to spell it out?
17:42.09blitzragebasty: depends how many NOTIFY's or OPTION's your Asterisk is doing -- you won't see channels just because a phone is registered
17:42.09[TK]D-Fenderbasty: You don't have channels if people aren't on calls.
17:42.40*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
17:43.00bastyMh, right now nobody is calling and i still see about 33 active SIP-channels...
17:43.06[TK]D-FenderIOscanner: Thats like pointing to the blood dripping on the floor as being the problem, and not the 14" craving kine that was plunged into your chest.
17:43.15[TK]D-Fendercarving knife*
17:43.33blitzragebasty: what are those channels actually doing, and what version of Asterisk are you running?
17:43.45IOscannerjust  a sec trying what blitzrage sent
17:43.54[TK]D-Fenderbasty: Pastebin it so we can see what it MEANS.
17:44.15bastyfor example: 10.46.3.80       (None)      3c26700f704  00101/21794  unkn  No       Rx: REGISTER
17:44.23[TK]D-FenderIOscanner: Pasting that single line would have taken 1/2 second and look what we are going through for this...
17:44.34[TK]D-Fenderbasty: those linger for a little bit
17:44.37*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
17:44.59neoalexmeanwhile, I'm waiting for [TK]D-Fender to get to me too :))
17:45.02[TK]D-Fenderbasty: then disappear.  You can try to reduce your register interval for that if you wish, but those aren't "calls"
17:45.08*** join/#asterisk THX2000 (n=bob@netblock-208-127-94-59.dslextreme.com)
17:45.23[TK]D-Fenderneoalex: canreinvite=no all around ([general] and phones, peers, EVERYWHERE)
17:45.36IOscannerregister=2056000055:password@registrar.carrier.com
17:45.41*** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com)
17:45.47[TK]D-Fenderneoalex: and "alexgs" is local to your * box?
17:45.55*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
17:45.57neoalexno... it's the phone
17:45.58[TK]D-Fenderio thats the problem, you didn't read the INSTRUCTIONS.
17:46.03[TK]D-FenderIOscanner: thats the problem, you didn't read the INSTRUCTIONS.
17:46.10[TK]D-Fender; Format for the register statement is:     register => user[:secret[:authuser]]@host[:port][/extension]
17:46.23IOscannerI typed it it is =>
17:46.29[TK]D-FenderIOscanner: You didn't fill in the /1235532whevere , so * fills in "s" FOR YOU
17:46.35*** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net)
17:46.49[TK]D-FenderIOscanner: No /extension <----------
17:46.50IOscannerThat is what I am trying
17:46.57bastyFender: ah okay - it just confuzed me because on voip-info it says "list all registed sip accounts". For right now I am trying to debug my sip stack, because after upgrading from 1.2.13 to 1.2.19, calls hangup during the conversation with aximum retries exceeded on transmission 52b24d6471faa4596be890cf624ec3ec@10.46.0.1 for seqno 104
17:46.59[TK]D-FenderIOscanner: thatsw what your line doesn't HAVE
17:47.01IOscannerblitzrage sent that with it
17:47.25IOscannerno I didn't have the /.... at the end adding it now
17:47.27blitzrageIOscanner: try again
17:47.36*** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net)
17:47.40blitzrageIOscanner: I did send it with a trailing /osaka
17:47.52blitzrage#
17:47.52blitzrageWe have a pair of Asterisk boxes we're going to call Toronto and Osaka that we're going
17:47.53blitzrage#
17:47.53blitzrageto have register to each other. We're going to use the most basic sip.conf file that will
17:47.53blitzrage#
17:47.53blitzragework in this scenario. Just like the SIP phone configuration earlier in this chapter, its
17:47.55blitzrage#
17:47.57blitzragenot necessarily the best way to do it, but it'll work.
17:47.59blitzrage#
17:48.01blitzrageHere is the configuration for the Toronto box:
17:48.03blitzrage#
17:48.05blitzrage<PROTECTED>
17:48.09blitzrage#
17:48.11blitzrage<PROTECTED>
17:48.13blitzrageoops :)
17:48.14*** mode/#asterisk [+b %blitzrage!*@*] by Corydon76-work
17:48.19IOscannercorrect I would then add the /accountnumber  it should use that instead of s
17:48.22*** mode/#asterisk [-b %blitzrage!*@*] by Corydon76-work
17:48.44*** mode/#asterisk [-o blitzrage] by russellb
17:48.52russellb:-p
17:49.01IOscannerokay things now it is a firewall issue upstream
17:49.02blitzrage:)
17:49.02IOscannerthanks
17:49.07basty[TK]D-Fender: you have any idea to the "Maximum retries exceeded on transmission" Problem?
17:49.17neoalex[TK]D-Fender: I put canreinvite=no in general (the phone already had it), still no go
17:49.17*** mode/#asterisk [+o blitzrage] by russellb
17:49.36blitzrageperfect... that was the last peice I needed in order to take over the world
17:50.13blitzrageman there are a lot of spiders outside my condo windows
17:50.45*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
17:51.35*** join/#asterisk LoveHatePassion (n=Nwm@office.xanter.net)
17:51.37LoveHatePassionHello
17:52.02*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
17:52.14*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
17:52.15LoveHatePassionanyone integrated skype to their  asterisk pbx?
17:54.09blitzrageLoveHatePassion: some people have.... check google for how to do it (you can only do 1 channel, and you have to run some weird interface thing)
17:55.24LoveHatePassionas far as I know there are only 3 gateways that do skype to sip
17:55.29matdoncan anyone point me at a simple example of a fast agi script wirtten in perl?
17:55.33rob0What are you going to do with the world now that you're in charge of it?
17:55.36[TK]D-FenderIOscanner: Just time just paste it, k? :)
17:56.14LoveHatePassionbut not sure which one to use so I wanted some advice
17:58.55[TK]D-FenderLoveHatePassion: What do you actually want Skype for?
17:59.47*** join/#asterisk gardo (n=gardo@121.97.211.162)
18:03.51*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com)
18:05.47*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
18:07.26kannanmatdon -> vicidial has a call logging script in perl
18:07.31kannaneflo.net
18:07.54*** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.20, 1.4.6 (June 29, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support.
18:11.20[TK]D-Fenderrussellb:  annnnnddd WHEE!
18:11.36russellbgettin' them bug fixes out there ...
18:11.56russellb90-something changes to 1.4 in 2 weeks, 40-something to 1.2
18:12.16Strom_Mi should upgrade
18:14.17LoveHatePassion[TK]D-Fender I want to use skype for international calls
18:14.40[TK]D-FenderLoveHatePassion: Why not use a normal ITSP to terminate?
18:14.47Hmmhesaysbah this sucks, it seems that channel variables are destroyed when asterisk redirects
18:15.02Strom_Mbecause the normal ITSP isn't "cool" like skype is?
18:15.10[TK]D-FenderHmmhesays: They should become INHERITED.
18:15.14Hmmhesaysor when asterisk receives a redirect
18:15.17Hmmhesaysbut they aren't
18:15.35HmmhesaysI'm sending a call out to an ata, the ata sends a "302" back
18:15.41Hmmhesaysasterisk redirects to a local channel
18:15.46LoveHatePassionwell because Skype is free
18:15.47[TK]D-FenderStrom_M: Skype is the bastard child of telephony.
18:15.54LoveHatePassionITSP does cost
18:15.56Strom_M[TK]D-Fender: i'm well aware of this
18:15.59Hmmhesaysthen my channel variables are gone
18:16.01[TK]D-FenderLoveHatePassion: Free to cal the PSTN everywhere?
18:16.08Strom_Mhence why i put cool in quotes :D
18:16.19LoveHatePassion[TK]D-Fender: no
18:17.04LoveHatePassionbut skype gave me free credits that would probably let me talk a very long time
18:17.10*** join/#asterisk gardo (n=gardo@121.97.211.162)
18:17.25BSD_Techok I am back still have the issue
18:17.43BSD_Techwhen I hit 3 it ask for the exten number
18:17.56BSD_Tech3 /#
18:18.04BSD_TechI dont have # defined
18:18.14[TK]D-FenderBSD_Tech: PASTEBIN <--------------
18:18.52[TK]D-FenderLoveHatePassion: probably?  You mean you're looking at this much pain and don't even know how long the ride is going to last?
18:19.22BSD_Techhttp://www.pastebin.ca/595892
18:19.32BSD_Techthats what happens when I hit pound
18:19.37LoveHatePassion[TK]D-Fender
18:19.53LoveHatePassionthe thing what I want to do is actually like this
18:20.43BSD_Techand I do not have a exten =# defined anywhere
18:20.43Hmmhesaysso this blows
18:20.54[TK]D-FenderBSD_Tech:  pastebin your actual dialplan
18:20.55LoveHatePassionI will be at the other side of the world for a long time at least 10 months. and I want to call my home and my home country so I have a TDM11P attached to my home telephony system where I can dial my home from a softphone SIP
18:21.19LoveHatePassionalso can call outside of my home using my existing phoneline.
18:21.23Hmmhesaysso how the hell am I supposed to authenticate a redirect call
18:21.35[TK]D-FenderHmmhesays: Whats to auth?
18:21.38LoveHatePassionbut I also want to call to other 3 countries
18:22.02LoveHatePassionand I though that I could do that with my free skype credits
18:22.15Hmmhesays[TK]D-Fender: when my ata redirects back to asterisk, asterisk sends to call into local/${num}@default
18:22.19BSD_Techhttp://www.pastebin.ca/595895
18:23.08[TK]D-FenderHmmhesays: Should send it to where the ATA is pointed to
18:23.27Hmmhesays[TK]D-Fender: the ATA redirects back to asterisk
18:23.59Hmmhesaysthats my problem
18:24.05*** join/#asterisk keulin (n=cray@AMontpellier-152-1-49-9.w81-251.abo.wanadoo.fr)
18:24.41[TK]D-FenderBSD_Tech: that is NOT funny
18:24.58BSD_Tech?
18:25.38Hmmhesaysso if asterisk sends the call back into  the default context, that means anyone sending in a sip call can terminate to my gateway
18:26.05BSD_Technot oll of it works
18:26.10BSD_TechI am rewriting
18:26.15Strom_MBSD_Tech: what the hell are you doing using # as the beginning of an extension name?
18:26.24Strom_M# means "I am finished dialing; put the call through"
18:26.32*** join/#asterisk ramindia_ (n=ramindia@202.63.96.9)
18:26.48BSD_Tech#6 is done by default inthe asterisk gui
18:26.54BSD_TechI did not get that
18:26.57[TK]D-FenderHmmhesays: The ATA is allowed to forward to anything it can dial
18:27.02ramindia_hey some one can asists me, my call disconnecting every 5min http://www.pastebin.ca/595894
18:27.03BSD_Techand 3 is used for directory
18:29.24Strom_MBSD_Tech: it's still a bad idea :)
18:29.26BSD_Techand some of th #XX are because they are used toi change functions from remote phones
18:30.02BSD_Techok
18:30.09BSD_TechI will make changes
18:30.34BSD_Techbut I still does not explai why when I hit just # its asking for a exten and passwd
18:30.57Hmmhesaysthere has to be some variable that is set when a redirect is sent
18:31.20[TK]D-FenderHmmhesays: ${BLINDTRANSFER}
18:31.28[TK]D-FenderHmmhesays: Quite likely
18:31.34BSD_TechI dont have bildxfer enabled
18:31.44BSD_Techblind even
18:31.53[TK]D-FenderBSD_Tech: Got a licences for that name change? :)
18:32.15BSD_Techwhat name change
18:32.26BSD_TechI have fatfinger this week
18:32.39BSD_Techbeen working on many projects
18:33.33BSD_Tech[featuremap]
18:33.34BSD_Tech;blindxfer => #1                ; Blind transfer  (default is #)
18:33.34BSD_Tech;disconnect => *0               ; Disconnect  (default is *)
18:33.34BSD_Tech;automon => *1                  ; One Touch Record a.k.a. Touch Monitor
18:33.34BSD_Tech;atxfer => 2*2                  ; Attended transfer
18:33.36BSD_Tech;parkcall => #72
18:33.39BSD_Techall disabled
18:33.53[TK]D-FenderBSD_Tech: I wasn't talking to YOU :)
18:33.57*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
18:34.13rob0Fat fingers mean you never have to say you're saoirrty
18:34.15BSD_Techsorry
18:34.38[TK]D-Fenderrob0: In lesbian terms that'd be "well hung" ;)
18:35.22*** join/#asterisk waverly360 (n=waverly@209.12.249.243)
18:35.37*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)
18:36.09irulehow can I control in what order are extensions interpreted?
18:36.26BSD_Tech?
18:36.33BSD_Techfor what function
18:37.12waverly360Does anyone know whether the operator setting in the voicemail.conf file does anything in asterisk 1.2.12?
18:37.28*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
18:37.41BSD_Tech1.2.12 is way old and I dont think it was in it
18:37.45[TK]D-Fenderwaverly360: Yes, just like every OTHER version.
18:37.47BSD_Techtry updating
18:37.52ramindia_is there any meetme/zaptel issues with asterisk 1.2.18 or SVN  ?
18:38.26*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
18:38.39BSD_Techtk any other Ideas
18:38.53BSD_Techwhere the issue might be
18:39.07waverly360[TK]D-Fender: Are you sure about that?  I've tried setting the review, operator, and contexts options...it still does nothing when I hit 0
18:39.22[TK]D-FenderBSD_Tech: that is a psychotic overincluded mess of over 1000 lines.
18:39.38[TK]D-Fenderwaverly360: Then clearly you are not doing it right.
18:39.50BSD_Tech?
18:39.58BSD_Techexplain
18:40.17*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
18:40.20ramindia_is the meetme stabled in asterisk 1.2.18, does any have problem with meetme rooms
18:40.30iruleI dial 321, and 1 is dialed, of course it is the only one valid, so, why is it skipping 3 and 2?
18:41.04Strom_Mramindia_: meetme should be fine in 1.2.18
18:41.32*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
18:41.33ramindia_Strom_M: i have low voice issue,
18:41.45Strom_Mwhat's "low voice issue"?
18:42.21[TK]D-FenderBSD_Tech: pastebin the actual context I should care about.
18:42.27jkiffOh my God...  a grand total of 2,572 lines.
18:42.29ramindia_Strom_M: when people in meetme room, they are not able to talk each other, very low voice not able to hear each otehr
18:42.31sweeperpendatry!
18:43.13BSD_Techbrb
18:43.28Strom_Mramindia_: what kind of phones?  what kind of connection into the asterisk box?
18:43.38Hmmhesayswell its kind of a hack, but it works
18:43.46ramindia_SIPTrunks
18:43.52ramindia_x-lite softphone
18:44.04Strom_Mramindia_: i'd blame the softphone
18:44.14Strom_Mwhat happens if you use real phones?
18:44.26ramindia_tried PAP2, same issue
18:44.47ramindia_but same X-lite directly calling works great
18:44.56Strom_Mpastebin your extensions.conf
18:44.59[TK]D-Fenderramindia_: You'r mics are all too low.
18:45.24ramindia_no its adjusted as per the test
18:45.27BSD_Techbbiab going to do some clean up and then retest
18:45.34ramindia_even increasing that is the same problem
18:45.43Strom_Mramindia_:
18:45.44Strom_Mpastebin your extensions.conf
18:45.46ramindia_when i listen the call, i can hear low voice
18:46.46_VoiceMeUp_COMJun 29 14:46:43 ERROR[7300]: pbx.c:5939 pbx_builtin_serialize_variables: Data Buffer Size Exceeded!
18:46.47BSD_Techolot of the file is documentation
18:46.49_VoiceMeUp_COManother nice one
18:46.55ramindia_let me do that
18:49.29ramindia_let me paste here
18:49.53ramindia_http://www.pastebin.ca/595941
18:50.00ramindia_here is my extension.conf
18:50.21ramindia_some time i get very choppy voice
18:51.23Strom_Mramindia_: ugh, haven't you heard of contexts?
18:51.34_VoiceMeUp_COMvicidial messes those up
18:51.54ramindia_yes i heard contexts .
18:52.06ramindia_why vicidial messup ?
18:52.23_VoiceMeUp_COMjsut saying that it looks like bits and pieces of the vicidial install procedure
18:52.34_VoiceMeUp_COMbut like strom_m said .. without any contexts
18:52.43Strom_Mramindia_: your dialplan is a royal mess
18:52.55_VoiceMeUp_COMso like a quick get it up and runnign , no toughts at all config
18:53.24ramindia_Strom_M: really ?
18:53.30Strom_Myes
18:53.36_VoiceMeUp_COMprboably security wholes in there too.., i imagine one getting intoa cofnerence and exiting to a zap
18:53.47_VoiceMeUp_COMsed/wholes/holes/
18:53.59waverly360[TK]D-Fender:  Well, I've been digging through voip-info for awhile now regarding the 0 out to voicemail problem.  When I'm in the person's voicemail box, hitting 0 does nothing..I don't even get any notification that I'm hitting it in asterisk.
18:54.05ramindia_Strom_M: any suggestions
18:54.14Strom_Mramindia_: rewrite it from scratch?
18:54.21[TK]D-Fenderwaverly360: You are falling into the usual trap....
18:54.30waverly360What are you talking about?
18:54.36_VoiceMeUp_COMthe 0 trap ;)
18:54.45ramindia_Strom_M: thats working config of vicidial
18:54.48[TK]D-Fenderwaverly360: the one of complaining about your problems without showing all the EVIDENCE
18:55.02Strom_Mramindia_: well then vicidial is a load of crap
18:55.03_VoiceMeUp_COMdoes any other dmtf get passed ?
18:55.04[TK]D-Fenderwaverly360: PASTEBIN <------------------------
18:55.22_VoiceMeUp_COMwaverly360 , try replacing voicemailmain by READ( and NoOp("
18:55.32_VoiceMeUp_COMand see if you at least have dtmf recog
18:55.36[TK]D-Fender_VoiceMeUp_COM: getting COLDER.
18:55.40_VoiceMeUp_COMhehe
18:55.55_VoiceMeUp_COMexten ,o,1,
18:55.56kannanramindia -> does not vicidial provide a ui to boost volume?
18:56.28ramindia_kannan: yes they does, but i have dialplan issue for that
18:56.32ramindia_i need to fix that too also
18:56.59Strom_Moverlapping extensions?  variable-length numbering?  godawful syntax?
18:57.05Strom_Mi'd toss that software pronto
18:57.09waverly360[TK]D-Fender: It makes more sense to me to just get a few suggestions before I go taking up everyone's time and energy with a wad of config files everytime.  If nothing obvious comes to mind, I like to play around a little bit more.  I don't like people holding my hand everytime I have a problem.
18:57.36Strom_Mwaverly360: do you have an "o" extension in the context from which you execute voicemail() ?
18:57.51*** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com)
18:58.06waverly360Strom_M: Yes.
18:58.12ramindia_Strom_M: u want me to use context to build my own
18:58.28Strom_Mwaverly360: and you're pressing 0 while listening to the user's recorded greeting, right?
18:59.03waverly360Strom_M: Yes.  I don't get any indication that asterisk is reading the '0' when I press it while watching the CLI.
18:59.30Strom_Mdoes asterisk get DTMF at any other point during the call?
18:59.50Strom_M(i.e. can you terminate voicemail by pressing #?)
19:00.28waverly360Strom_M:  Hmm...yeah I can.
19:00.58[TK]D-Fenderwaverly360: Just pastebin it all.  your description is not helping at all.
19:01.04waverly360Strom_M: Actually, # is the only thing it does recognize.
19:01.38mishehusorry guys, I've got limited browsing from this location.  is it possible in zapata.conf to do `channel => 1,3,4` instead of `channel => 1-4`, as I want to be able to remove a bad line from the zap trunk group
19:01.48*** join/#asterisk sharp (n=sharp@pool-72-94-91-143.phlapa.east.verizon.net)
19:01.53[TK]D-Fendermishehu: Yesw
19:01.55mishehuI can't try it yet as there's folks on the phone.
19:02.01mishehu[TK]D-Fender: I'll give it a shot.
19:02.09*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
19:02.19*** join/#asterisk kombi (n=kombi@213.160.14.18)
19:02.33[TK]D-Fendermishehu: You could also make multiple groups for your lines.
19:02.49mishehujust didn't want anybody going ballistic on me...   it's bad enough that all their bitching at me for all 3 days of cluecon turns out to be at&t's problem.
19:03.04kombiis a variable defined in the dialplan also available in agi?
19:03.15[TK]D-Fenderkombi: Yes
19:03.16kombihi Fender!
19:03.32[TK]D-Fenderkombi: y0
19:03.37Corydon76-workkombi: see GET VARIABLE
19:04.28[TK]D-FenderI need to get off my ass and learn AGI, Realtime, etc....
19:05.11kombiso like: say digits get variable(BluePill) ?
19:05.28mishehu[TK]D-Fender: if you were american I'd say that getting off your ass would be very unamerican, but at least according to your dsl, you're in canada instead...
19:05.36mishehuand likely a canadian
19:05.38waverly360Hah
19:05.52[TK]D-Fender<- Clearly Superior :D
19:05.58iruleis this valid? gotoif [${CALLERID(num)} = 5XX]?thisisanextension:thisisnotanextension  I want to send an error message and new dial tone to extensions and sent callers from outside to the IVR
19:05.58kombi..BluePill being some juicy digits of course
19:06.07mishehuwaverly360: hey, you took the stretch hummer back to the hotel the other night, didn't walk back
19:06.07[TK]D-Fenderwaverly360: Gimmeh mah PASTEBIN! :)
19:06.19[TK]D-Fenderirule: NO
19:06.25mishehu[TK]D-Fender: I'll let you believe that since it pleases you.
19:06.26LoveHatePassionwhy dont people use pastebin ?
19:06.34waverly360mishehu: Yeah, I know....two beers and 3 glasses of wine..why walk? :)
19:06.37[TK]D-Fenderirule: Go and completely reread there part of the book on EXPRESSIONS.
19:06.44mishehuwaverly360: gotta walk it off...
19:06.46waverly360[TK]D-Fender: actually, I think I may have found my problem.
19:06.51mishehuall that beer is lots of calories
19:07.04[TK]D-Fendermishehu: Not so much really...
19:07.08Corydon76-workkombi: No, you cannot embed one AGI command in another
19:07.10mishehuhow many bottles of beer is equivalent to a loaf of bread?
19:07.17kombiirule: few more params required there
19:07.27waverly360mishehu: less thinky, more drinky
19:07.33mishehu[TK]D-Fender: ok, so tehre was a lot of p izza eating too along with that wine and beer
19:07.38[TK]D-Fendermishehu: beer does cost a lof of 'bread' if you go to bars :)
19:07.49kombiCorydon76-work: so first write into an AGI var, then use that i take it?
19:08.01Corydon76-workkombi: correct
19:08.03mishehu[TK]D-Fender: indeed.  that's why I go for the hard stuff.  it's more economical.  more booze for the buck.
19:08.07mishehuerr alcohol
19:08.15waverly360[TK]D-Fender: I had my operator extension defined in the wrong place.  I thought I tried it this way once already..but must've neglected to reload or something.
19:08.26[TK]D-Fendermishehu: I'm a Grand Marnier stright kind of guy mayself
19:08.39waverly360I prefer dirty martini's :)
19:08.54[TK]D-Fenderwaverly360: "Evidence has been obscured to protect the GUILTY" :)
19:08.59Corydon76-workYick, too much salt
19:09.09irulekombi combi is the famous alias for the VW wagon in Mexico lol http://images.google.com.mx/images?q=combi
19:09.26waverly360[TK]D-Fender: Hah..you just want to give me shit about how horrible my dialplan looks :P
19:10.07[TK]D-Fenderwaverly360: No, I'd be perfectly happy to assassinate you on this flw individual of your myriad other sins ;)
19:10.10mishehu[TK]D-Fender: gin & tonic, screwdriver, vodka & cranberry, straight vodka, straight rum...  I used to do tequila but that does bad things to stomach and sense of balance...
19:10.11[TK]D-Fenderflaw*
19:10.18irulewaverly360 your dialplan is fine and dandy   ....hmmm, can I see it?
19:10.36*** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer)
19:10.50[TK]D-Fendermishehu: I'm good on most of that.  Mind you I'm not really a drinker.  That list alone accounts for more than my typical monthly consumption :)
19:10.53JerJerwhoohoo v1.4.6
19:10.55waverly360irule: Now why would you want to do that?  All my dialplan consists of now (well mostly) is a call to AGI
19:10.59*** join/#asterisk Tako-san (n=Tako-san@216.232.147.102)
19:11.16irulejsut out of curiosity ;)
19:11.40waverly360Corydon76-home: Martini's don't have much salt..the olive juice might have some in it..might you be thinking of margaritas?
19:11.44iruleI am so far 420 working lines :s
19:12.07JerJermmm 420
19:12.14waverly360irule: Mine used to be huge... [TK]D-Fender has already berated me on my lacking use of macros :)
19:12.18kombiCorydon76-work: is namespace completely separate in dialplan and AGI?
19:12.26mishehu[TK]D-Fender: heh, that's about my yearly consumption - one of each.  just toss in a few glasses of wine on top of that.
19:12.33irulewith beautyful formatting and ASCII art pointing all over for quick browsing throught the sections lol
19:12.37kombiirule: I like that car, I believe it is called that in south africa too
19:12.52mishehu[TK]D-Fender: we had somebody doing nose dive & face crashes at cluecon...
19:12.55Corydon76-workkombi: you're talking about two different processes, entirely, communicating through a pipe
19:13.00mishehusomebody couldn't handle the drinky
19:13.23waverly360[TK]D-Fender: Why didn't we see you at ClueCon..I figured you'd be all about a geek convention like that :)
19:13.33kombiCorydon76-work: ok, understood. so, yes so to speak..;)
19:13.40irulekombi yes combis rule, when I was a teenager a friend of mine had one and was always seeding it like crazy, it was hillarious
19:13.45[TK]D-Fenderwaverly360: Too far, too expensive and I'd gain too little :)
19:13.53irulespeeding
19:13.58waverly360[TK]D-Fender: Ah..you're already near perfect then? :P
19:14.18[TK]D-Fenderwaverly360: And that is a border I do not want to cross while the DHS, GWB, are still in effect....
19:14.22mishehu[TK]D-Fender: gain too little?  what's not to gain from hanging out with all of us?
19:14.31irule[TK]D-Fender just enjoy and show love :*
19:14.33[TK]D-Fenderwaverly360: No, I'm not in-depth ENOUGH actually.
19:14.52[TK]D-Fenderwaverly360: I've mastered the basics, but the cool shit is out of my league.
19:15.13kombiirule: they're actually called "bus" in germany (not very imaginative really), my brother had one and drove it through the sahara desert
19:15.15[TK]D-Fenderwaverly360: I'd go if it was cheaper and quicker to get to.
19:15.16irule[TK]D-Fender what is _THE_ cool shit?
19:15.26waverly360[TK]D-Fender: I'm still a newb no doubt..I've only been dealing with VoIP stuff for about a year and a half now.
19:15.30irulekombi awesome
19:15.33[TK]D-Fenderirule: It's not singular.  it's a collective term.
19:15.55irule[TK]D-Fender what are _THE_ cool shit for example then?
19:15.57waverly360[TK]D-Fender: That's why I'm always leary of pastebinning my stuff...I don't do well with such harsh scrutiny :P
19:16.15[TK]D-Fenderirule: and byt aht I mean Realtime, AGI, real AMI, SCCP/MGCP, billing channel dev, etc.
19:16.38[TK]D-Fenderwaverly360: my justice would be swift, and nearly painless ;)
19:16.41irulewaverly360 I always pastebin, I only take the positive criticism, the rest is down the toilet
19:17.24waverly360I really love the concept of AGI.  It's what we're using for our dialplan now.  My biggest problem is figuring out the difference in using certain commands from an AGI script, as opposed to using them in extensions.conf
19:17.49waverly360Like the directory...calling the directory function from my AGI script causes the directory function to do nothing.
19:18.11waverly360I had to create a special context in extensions.conf, and dial it locally to make it work.
19:18.35irulewaverly360 http://sciencehack.com/videos/view/DKivdMAgdeA#searchkeywords I always intentionally ot out of the masses in a sense of individuality
19:19.12waverly360Now I'm having issues with the Voicemail command.  if I'm calling voicemail from extensions.conf, hitting 0 takes me to the operator.  If I call it from AGI, hitting 0 hangs up the call...still workin on that one :P
19:19.22*** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com)
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19:19.30waverly360[TK]D-Fender: Hah.  nearly painless eh
19:19.51xo8oxguys anyone knows why the hell does my cisco 7960 keeps restarting ?
19:19.52[TK]D-Fenderwaverly360: No pain, no gain ;)
19:20.27mishehuxo8ox: why yes, I can think of about a million reasons to why your cisco 7960 keeps restarting.
19:20.37waverly360[TK]D-Fender: Yeah I know..but I don't like getting slack for shit that I didn't write.  And even then..both of us were brand new to the industry..it's like development.  you always look back at your previous code and sigh at how pathetic it was.
19:20.42xo8oxhow do I stop it from restarting so I can get into the configs to see whats wrong with the settings
19:20.51nDuffI've had a curious issue since moving to a new building and upgrading to Asterisk 1.4: On outgoing calls, two overlapping progress tones are audible. What might be the cause of this?
19:20.53waverly360Doesn't really make us bad programmers does it?
19:21.20xo8oxwell my question is how do u stop it from restarting so I can get into it
19:22.03BSD_Techok I found the issue and I am taking it up with the gui guis
19:22.10BSD_Techits a bug inthe gui
19:22.20BSD_Techpart of it is
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19:23.52[TK]D-Fenderwaverly360>[TK]D-Fender: Yeah I know..but I don't like getting slack for shit that I didn't write. And even then..both of us were brand new to the industry..it's like development. you always look back at your previous code and sigh at how pathetic it was. <- I have no intention of cutting you any slack ;)
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19:24.13[TK]D-FenderBSD_Tech: GUI has bugz z0mg noes!
19:24.48waverly360[TK]D-Fender: Y'know...if you weren't a punk like this to everyone, I might think you were....well wait..you're just a punk aren't you? :)
19:24.56BSD_Techwhen it creates a user exten in users.conf its setting context= not usercontext=
19:25.04BSD_Techso the dial fails
19:25.13[TK]D-Fenderwaverly360: Shup junior :)
19:25.34waverly360[TK]D-Fender: I'm not young enough for you to call me junior ;)
19:25.45[TK]D-Fenderwaverly360: All you have to be is YOUNGER ;)
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19:26.13waverly360[TK]D-Fender: You don't really have any basis for my age.  I might have gray hair :)
19:26.46[TK]D-Fenderwaverly360: I do, but I have professional help in lying about it ;)
19:27.03waverly360hah hah hah
19:27.08waverly360excellent
19:27.16kannanramindia_ , i cannot understand the dialplan issue you had mentioned
19:27.42[TK]D-Fenderwaverly360: NEVER think that you can do a decent job dying your own hair.
19:28.03waverly360[TK]D-Fender: LMAO  something tells me there's a great story there
19:28.08mishehu[TK]D-Fender: hahaha
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19:28.49[TK]D-Fenderwaverly360: Nothing "funny", just you miss stuff, and those box jobs rinse out at some point (they lie about that part.
19:28.59*** part/#asterisk galeras (n=root@200.31.204.42)
19:29.26[TK]D-Fenderwaverly360: 2 weeks ago I deceded to go and have it done by a pro.  Worth every penny, and it wasn't even that much.
19:30.06[TK]D-FenderI'v gotten comments on it from everyone so far...
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19:30.23waverly360[TK]D-Fender: I've had mine highlighted before..but I won't be doing that stuff by myself.  The most I've done is shave my head..that's pretty easy to do..and difficult to mess up.
19:31.03[TK]D-Fenderwaverly360: Highlights?  Dear God that is so not on the lit of approved things to do unless you've been spending too much time with BKW ;)
19:31.08[TK]D-Fenderlist*
19:31.37waverly360[TK]D-Fender: Hah.  Perhaps :)
19:31.46[TK]D-Fenderwaverly360: TMI <-
19:31.58Qwell[][TK]D-Fender: Why are you dying your hair to begin with? O.o
19:32.00waverly360[TK]D-Fender: lmao.  No worries...I'm about as straight as can be.
19:32.39waverly360Qwell[]: He's workin his mojo for the ladies.
19:32.50*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
19:32.51ramindia_kannan: the dial plan is for working
19:33.46*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
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19:34.50kannanramindia_ -> i wanted to ask whats the connection between the dialplan and the volume control in vicidial
19:35.21ramindia_kannan: volume increase with dial plan only
19:35.36ramindia_vicidial use app_meetme for the conference calls
19:35.58ramindia_so vicidial have tool which increase the volume in conference
19:36.55BSD_Techok now I knwo there is a issue and its not my dial plan
19:37.12BSD_TechI rm my dial plan and put the default asterisk dial plan back
19:37.22kannanBSD_Tech -> but what about ur featuremap? lol
19:37.33[TK]D-FenderQwell[]: You missed the very clear statement that I HAVE grey hair.
19:37.35BSD_Techexerything disabled
19:37.41Qwell[][TK]D-Fender: and you're...how old?
19:37.42*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
19:38.02BSD_Techand yes I restaryed asterisk
19:38.10[TK]D-FenderQwell[]: 32, but its genetic.  But in the long run I still get to KEEP it unlike some people I know :)
19:38.17*** join/#asterisk el_4_jinete (n=el_4_jin@mail.pulxar.com.co)
19:38.23BSD_Techand yet its still playing please enter the exten themn press pound
19:38.38BSD_Techwhen I hit just the # key
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19:40.29waverly360[TK]D-Fender: Jesus man..you're not even that much older than I am...wtf
19:40.29waverly360[TK]D-Fender: There's a 10 year requirement to call someone Junior and get away with it :)
19:41.19kannanramindia_ -> did u try the s option strin in meetme?
19:41.24kannanstring
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19:41.36[TK]D-Fenderwaverly360: .... shup Junior ;)
19:41.49ramindia_|s u mean
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19:43.21BSD_Techhttp://pastebin.ca/596038 this is all thats in the current dial plan
19:43.33BSD_Techand the # key issue still happens
19:43.51kannanyes
19:44.37waverly360[TK]D-Fender: hush old man :P
19:44.49BSD_Techso in some other file the # key is set
19:45.01BSD_Techkids goto seperate corners
19:45.04BSD_Tech<=== 39
19:45.15kannanBSD_Tech -> it is features.conf i guess
19:45.20kannan35 here
19:45.43BSD_Techeverythingg in features.conf is disabled
19:46.18BSD_Techand this only started with 1.4.5
19:46.27BSD_Techdid not do this in 1.4.4
19:46.30*** part/#asterisk pogo123 (n=hh@buero-gw.dortmund.loca.net)
19:46.37BSD_Techso I think some one hardcoded something
19:48.01BSD_Techits pissing me off
19:48.09msetimHi guys
19:48.27msetimsomebody already make a pre-provisioning with Linksys PAP2-NA
19:48.54*** join/#asterisk gardo (n=gardo@121.97.211.162)
19:49.01msetimWhat is the parameter that I need to put in DHCP server to say to PAP2-NA the TFTP server
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19:49.56[TK]D-Fendermsetim: Option 66
19:50.56*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
19:51.29msetim[TK]D-Fender: thanks TK but i already put it "option option-66 172.16.0.75;" but it doesn't work
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19:52.50[TK]D-Fendermsetim: Well thats the #, now if you're talking dhcpd.conf SPECIFICALLY, then its    "option tftp-server-name "ip-or-host";
19:53.25[TK]D-Fendermsetim: Next time be specific ;)
19:54.14msetim[TK]D-Fender: :-D O.k. Tks ;)
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19:54.25diclophis-workhello all
19:56.12diclophis-workhas anyone ever worked with a F5-BigIP device?
19:56.17diclophis-workfor load-balancing SIP?
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20:07.23errrif I am in the voicemail is there a way to hit some key to get back to the operator?
20:07.26errrId like to hit 0 and have it take me back to the operator
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20:09.51[TK]D-Fendererr : Yes, go read the instructions for VoiceMail and brush up on your Standard Extensions.
20:09.59*** part/#asterisk ramindia_ (n=ramindia@202.63.96.9)
20:10.19errr[TK]D-Fender: ok thanks
20:12.09claudiotainenis there anyone who could give me some help with my first asterisk configuration ?
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20:14.04Strom_Mclaudiotainen: depends on what kind of help
20:14.44kannanthey are categorized as paid and unpaid help, ?
20:14.49claudiotainenwell I am a complete newbie of asterisk
20:14.55claudiotainen:D
20:15.19Strom_Mclaudiotainen: have you read the book?
20:15.27Strom_M~book
20:15.27jbothmm... book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
20:15.45claudiotainenyes that's exactly the one I was reading
20:16.01Strom_Mwell, ask some specific questions and we can answer them
20:16.05claudiotainenit shows how to configure X-Lite
20:16.14claudiotainenI am trying to use Ekiga
20:16.27Strom_Mhopefully not on the same machine as the asterisk box
20:16.31claudiotainenI have both asterisk and ekiga installed on the same pc
20:16.37Strom_Mugh no no no no no
20:16.41claudiotainenoh ...
20:16.41Strom_Mdon't do that
20:16.48Strom_Mdon't run asterisk on any box that runs X windows
20:17.04claudiotainenah
20:17.20claudiotainenso I sould use a computer just to run an asterisk server ?
20:17.25Strom_Myes
20:18.54*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
20:19.23claudiotainenso as an example
20:19.29claudiotainenI could install asterisk on a pc
20:19.48claudiotainenthen connect this machine to the one where ekiga is
20:19.56Strom_Myes
20:20.08claudiotainenok thank u
20:20.13claudiotainener one more thing
20:20.18Strom_Msure
20:22.06claudiotainenwhen I am registering my account with ekiga I should use asterisk address as the registrar address
20:22.10claudiotainenis that correct ?
20:22.14Strom_Myes
20:22.21claudiotainenand asterisk addres
20:22.49claudiotainenis the IP address of the machine where asterisk server is running ?
20:23.05Strom_Myes
20:23.12claudiotainenoh ok
20:23.58claudiotainenmaybe that's why it didn't work when I tried to use the same pc for both the server and the front end
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20:25.27claudiotainenthank you strom
20:25.30claudiotainenbye all
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20:26.41[[blah]asfdI keep getting these errors on my server: Jun 29 14:25:19 NOTICE[4926]: chan_zap.c:8362 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
20:26.44[[blah]asfdversion 1.2.19
20:26.58[[blah]asfdwhat would cause that
20:27.12Strom_Mtiming issues?  irq conflict?
20:27.20Strom_Mwhat card are you using?
20:27.37shido6good start
20:27.37[[blah]asfdte410p
20:27.47[[blah]asfdi checked the irqs. they look fine
20:27.55[[blah]asfdtiming has been a discussion on this server.
20:27.58*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
20:28.08[[blah]asfdweather it should be provided by me, or the carrier.
20:29.04Strom_Mare you running anything at all besides asterisk on the box?
20:29.04*** join/#asterisk sabakas1 (n=solapus@66.90.121.129)
20:29.04[[blah]asfdonly asterisk
20:29.12Strom_Mno x windows or anything, right?
20:29.50Jingleswhat's wrong with running xwindows on an * box?
20:29.57Strom_MJingles: timing issues
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20:32.00[[blah]asfdok, here is my zapata.conf. have I made any mistakes there? I have 3 t1s in the card. the first has the dchan and the other two have none. they are in a trunk group.
20:32.09[[blah]asfdthis has worked for a while... just now become an issue.
20:32.12[[blah]asfdhttp://pastebin.ca/596159
20:33.28Strom_Mwhy are you defining spanmaps?
20:33.28Strom_MNFAS?
20:34.04Strom_Malso show me zaptel.conf
20:34.32[[blah]asfdyes, it is nfas
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20:35.44[[blah]asfdhttp://pastebin.ca/596170
20:35.48[[blah]asfdthere is my zaptel
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20:37.53Strom_Mso you're expecting timing on all four spans
20:37.53Strom_Mand the telco wants you to provide timing?
20:38.32[[blah]asfdi have it set to expect timing dont i?
20:39.55*** part/#asterisk naitram (n=ttech@216.77.58.40)
20:40.11[[blah]asfdwhat is happening is i reboot my server and then it runs good for about 10 minutes. Then i start to see hdlc errors and then everything stops working all to gether
20:40.36[[blah]asfdi get errors like: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
20:40.38[[blah]asfdwhen i dial outbound
20:40.44*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
20:40.50[[blah]asfdif i reboot that all goes away for about another 5-10 minutes
20:40.56[[blah]asfdtiming you think?
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20:43.09Strom_M[[blah]asfd: try setting it to not expect timing
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20:46.30[[blah]asfdStrom_M: ok, so would this be correct?
20:46.34[[blah]asfdhttp://pastebin.ca/596205
20:47.12Strom_Mno
20:47.56[[blah]asfdok... so I am not sure how that works then. I missunderstood
20:48.43Strom_Mspan = span_number,timing,buildout,framing,coding
20:48.49*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
20:48.59Strom_Mso span number is self explanatory
20:49.05[[blah]asfdso i have the wrong numbers in timing...
20:49.07Strom_Mtiming should be 0 if you're not looking for timing
20:49.18[[blah]asfdi have 1,2,3,4
20:49.27[[blah]asfdi should have either 0,0,0,0 or 1,1,1,1
20:49.28[[blah]asfdright?
20:50.07Strom_Mbuildout should be 0 if your telco's CSU/DSU is less than 133 feet away
20:50.09Strom_M0,0,0,0
20:50.15Strom_M1 means "primary timing source"
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20:50.20Strom_M2 means "secondary timing source"
20:50.21Strom_Metc
20:50.49[[blah]asfduhhh... ok. if the telco is the timing, then i want it to be what?
20:50.58Strom_M1,2,3,4
20:51.15Strom_Massuming you want to use span 1 as the primary timing source, et. al.
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20:51.24[[blah]asfdand if i am providing it then each one would be 0?
20:51.29Strom_Myes
20:51.34koosrooshi there
20:51.45koosroosam i alone in this asterisk jungle?
20:51.50[[blah]asfdok
20:51.53[[blah]asfdlet me try that.
20:52.00Strom_Mkoosroos: completely, utterly alone
20:52.22koosrooslol snom_m nice device name
20:52.25shido6:)
20:52.26Strom_Msnom?
20:52.28Strom_Mit's strom
20:52.29Strom_Mnot snom
20:52.32Strom_Mplease read carefully
20:52.46koosroosi know just sounds familiar
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20:53.03Strom_Mi picked the handle long before i knew about Snom
20:53.23koosroosk regular on asterisk?
20:53.34Strom_Mjust ask your question already
20:53.49koosroosi have all the answers
20:54.16koosroosi wish
20:54.45koosroosany asterisk and fax guru's around?
20:55.50Strom_M....
20:55.55Strom_Mask your question
20:55.55*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
20:56.22apturaHas there been a way to make the ip500s work behind nat? I know a discussion long ago saying that it cannot work with the soundpoint series.
20:56.40Strom_Maptura: my polycom works fine behind nat
20:56.45apturaohh
20:56.46nDuffkoosroos: like Strom_C said, ask your question.
20:57.06nDuffkoosroos: some of us know faxing, but we won't volunteer ourselves until we know what we're volunteering ourselves for.
20:57.24*** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com)
20:58.00koosroosdownloaded the bash script fax2mail but it does not work, and i don't know didlie about scripts, anyone know where i can find a bash script for just mailing the tiff file received with app_rxfax?
20:58.07Strom_MQwell Qwell[] russellb Corydon76-home file  can you please kick sunsmasher until he fixes his connection?
20:58.25*** mode/#asterisk [+b *!*n=Beamer@*.se.biz.rr.com] by Qwell[]
20:58.26*** kick/#asterisk [sunsmasher!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Please fix your connection.)
20:58.32nDuffkoosroos: not me -- I use HylaFAX+iaxmodem, not app_rxfax.
20:58.40Strom_Mthanks Qwell[] :)
20:58.54*** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net)
20:58.59koosroosnDuff: how do you intergrate halafax into asterisk?
20:59.03*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:59.11nDuffkoosroos: iaxmodem.
20:59.28apturaokay well then I guess I need to configure it so it can work. I am in the network part of the configuration and under Network Address Translation have the ip/signaling/media port setup. But wonder if both are 5060. Have RTP port setup with the range of 10k-20k but do not know what the forced port field is for.
20:59.29koosroosdo you only do ip faxing?
21:00.03nDuffkoosroos: no -- I don't do any.
21:00.28nDuff(don't do any IP faxing, that is; T.37 and T.38 are just too sparsely implemented).
21:00.37apturaMy guess is with these fields not complete it cannot register and so the display on the ip500 is flashing.
21:00.59Strom_Maptura: how are you configuring the phone?
21:01.25koosroosnDuff: I can receive faxes no problem, i just have a problem mailing it, it works then it just stops, but I still receive the incoming faxes in var/spool/asterisk/fax/ folder
21:01.33apturaRight now though its web page. But done it in xml and on the phone in the past.
21:01.44Strom_Maptura: do it in the xml file
21:01.48apturak
21:01.50*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
21:01.56Strom_Mwhat version of the firmware are you using?
21:02.03Strom_Moh god, he's back
21:02.34*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
21:03.22apturaStrom looking up version.
21:03.43aptura1.6.2.0041
21:03.48Strom_MQwell[]: it's happening again!   halp!!!
21:04.05Strom_Maptura: that's an extremely old version
21:04.07Qwell[]...
21:04.09apturaI know
21:04.13*** mode/#asterisk [+b sunsmasher!*@*] by Qwell[]
21:04.34*** mode/#asterisk [-b *!*n=Beamer@*.se.biz.rr.com] by Qwell[]
21:04.58nDuffkoosroos: hrm. I don't trust app_rxfax -- it doesn't have the robustness of HylaFAX's Class 1 implementation -- so I don't use it, so I don't know anything about its hook scripts... and since I'm not ever going to use app_rxfax for work, I'm not too inclined to learn how its hook scripts work without someone paying me to do that. Sorry.
21:05.02Strom_Maptura: who is your reseller?
21:05.04apturaNot a licenced reseller so do not have access
21:05.10apturato polycoms
21:05.19apturaattacom
21:05.30apturapolycoms web site.
21:05.32Strom_Mthey should be able to provide you with updated firmware
21:05.38apturak
21:06.12*** join/#asterisk mountainm2k (n=mountain@165.236.183.1)
21:06.12apturaIn the mean time for now want to see if I can make this phone register by passing out of my public ip and back in to my network.
21:07.32mcabaptura: what do you mean by "the display is flashing"? is it the entire display, or just the time?
21:07.55apturasorry the time is flashing and the lines are not registered.
21:08.32mcabaptura: well, the time is flashing because the phone can't contact an SNTP server; this is probably a seperate issue from the registration problem :-)
21:08.41apturaI figure that is the case with a incomplete NAT fields setup in the phone. Phone works fine so wanted it to register though the public ip instead of internally.
21:09.06mountainm2kI'm trying to make a menu that executes some system commands (it momentarily unlocks various doors in the office in case somebody locks themselves out -- it works from the command line)
21:09.21mountainm2khttp://www.pastebin.ca/596272
21:09.23apturamcab it works fine. I just made a minor change by adding nat info so while its incomplete it will not access the tftp server.
21:09.49Strom_Maptura: FWIW, my polycom uses the default files included with 2.1.2, and the only things i've added are the registration details and the SNTP server.  works fine behind NAT.
21:10.05apturaokay
21:10.19apturaso this is unessesary fiels then.
21:11.05apturawill go back to change the servers ip to external. fw is configured
21:12.30apturaalso anyone know of a west coast ntpd time server I can use?
21:12.41Strom_Mtick.ucla.edu
21:12.51Strom_Mis that west-coast enough?
21:12.54*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
21:12.57apturasure it
21:13.01apturais :)
21:13.06Strom_Mare you in los angeles?
21:13.44apturabc
21:14.10Strom_Mperhaps you'd better find an ntpd in BC then
21:14.17Strom_Mlos angeles is a bit of a schlep away
21:14.18apturawe are pacific time
21:14.22[[blah]asfdStrom_M: That did not fix my problem. I switch my timing to 0 and i still get the error
21:14.25[[blah]asfdwhat else can I try?
21:14.27apturait does not matter
21:14.28aptura;)
21:15.31shido6:)
21:16.24*** join/#asterisk boster (n=boster@38.98.147.68)
21:16.56apturammm just restarted ntpd and still says EDT
21:17.10bosteris there a way to compile ztdummy for a newer (2.6.18) kernel with SMP and have it not lose clock seconds?
21:17.16mountainm2kare you saying you're in the incorrect time zone?
21:17.20mountainm2kor that your clock is wrong?
21:17.24shido6use date
21:17.29bosterI can compile it, and it loads, but conferences get progressively laggy
21:18.56apturashido same thing
21:19.04shido6erf?
21:19.10shido6what did you put
21:19.11apturaFri Jun 29 17:18:48 EDT 2007
21:19.18shido6oh
21:19.19shido6zoneinfo....
21:19.21shido6let me think
21:20.32shido6goog /etc/localtime
21:21.27shido6http://www.linuxsa.org.au/tips/time.html
21:23.10[[blah]asfdok, so i am loosing my zap channels after about 5 minutes of server uptime... regarding timing and interrupts, here are my files http://pastebin.ca/596308
21:23.21[[blah]asfdcould anyone possibly see where I could be getting this issue from?
21:23.43apturadone
21:23.58Strom_M[[blah]asfd: why not call digium support?
21:24.07[[blah]asfdhmmm good idea
21:30.19apturainteresting. VM light is not lighting up.
21:31.07apturaokay this is a little odd. its happened before leave a vm but the vm light does not illuminate.
21:31.12Strom_Maptura: did you set mailbox= in sip.conf?
21:31.32apturayour right
21:31.41Strom_Mmy right what
21:31.45Strom_Mmy right....arm?
21:32.00apturaYou would think that would be built right into the code but in this case its not.
21:32.25Strom_Mwhich code?
21:32.41apturayea its setup mailbox=200@default
21:33.00Strom_Mand is there a message in 200@default?
21:33.03apturayes
21:33.23apturaleft two test messages. shows two attempted calls on the LCD
21:35.07Strom_Mis the MWI setting turned on in sip.cfg?
21:35.32apturaIt should be have not messed with it. Has always worked but let me verify it.
21:36.10Strom_Mis the phone set to host=dynamic in sip.conf?
21:37.02apturaThere is now
21:37.28Strom_M?
21:37.56apturaI just configured it. I think it was removed when doing some editing the other night
21:38.04apturak made a test call
21:38.04*** join/#asterisk Juxt (n=Juxt@c-71-196-42-107.hsd1.fl.comcast.net)
21:38.08Juxthello guys
21:38.12apturano mwi light yet.
21:38.25Strom_Mis the phone actually registering>?
21:38.45Juxtquick question: if i am dialing a bunch of numbers at the same time e.g. SIP/exte1@carrier1&SIP/exte2@carrier2) how can i make asterisk write separate CDR records for all the calls?
21:38.53apturayes
21:38.58aptura200/200                    192.168.10.50    D          5060     Unmonitored
21:39.24Strom_Mwait - isn't your asterisk box on a public IP?
21:39.40Dj_FlyByif I wanted to try asterisk would it be possible to try it in a Virtual Environment like VMWare?
21:39.44apturaboth phone and * are internal network
21:39.48*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
21:39.56Strom_MDj_FlyBy: sort of
21:40.09apturathis wmi mwi error preceded my attempt to make the phone access the gw externally
21:40.20koosrooswhat phone are you using aptura/
21:40.22koosroos?
21:40.24apturaip500
21:40.28Dj_FlyByStrom_C: would there be some docs I could look at for this?
21:40.36Dj_FlyByStrom_M: would there be some docs I could look at for this?
21:40.42Strom_MDj_FlyBy: don't spam.
21:40.50koosrooswhat seems to be the problem?
21:40.52Dj_FlyBynot spam, typo
21:41.00Strom_Mi can read, you know
21:41.13Dj_FlyBythen why accuse me of spam if you can see the typo
21:41.17Dj_FlyByanyways
21:41.26Strom_Mbecause you pasted essentially the same line twice
21:41.28Strom_Mthat's why
21:41.32apturawell i need to get other things done.
21:42.02apturaminor nusence that the mwi is not lighting up but will work on it another time.
21:42.07Dj_FlyBywell sorry I guess, never thought I'd have to apologize for fixing a typo
21:42.14Strom_MDj_FlyBy: anyway, running asterisk in a VM is doable
21:42.21Strom_Mbut i wouldn't run it in production
21:42.37Strom_Myou'll also have no access to hardware
21:45.39Dj_FlyBynot looking to run it in produfirstction *yet* just want to try it out
21:45.41Dj_FlyByack
21:46.03Strom_Mlol
21:46.03Dj_FlyBywell, you get what I mean I am sure :)
21:46.28Dj_FlyBy<-- having troubles with the touchpad on the laptop running in *nix, lol
21:46.47Strom_Mheh
21:48.08Dj_FlyBywell, thanks for the info, will search out some documentation for it, if there is any out there
21:48.10Dj_FlyBy:)
21:48.49Strom_M~thebook
21:48.50jbotextra, extra, read all about it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
21:49.02Strom_M^^^^^^^^^^^ read that
21:49.41Dj_FlyByyea, I've been reading it *just started* but have yet to see anything related to setting it up in a Virtual Environment, I assume though it is probably the same.
21:51.14Strom_Myeah, just for dicking around with it, you should have no trouble
21:52.08Dj_FlyByexcellent, thanks again for the help.
21:53.59*** mode/#asterisk [-bb brian!*@* #asterisk!*@*] by Qwell[]
21:55.36*** join/#asterisk vn (n=vn@bas5-quebec14-1177599731.dsl.bell.ca)
22:03.15*** join/#asterisk Primer (n=vi@sh.nu)
22:03.43PrimerIs chan_sccp deprecated in favor of something else? or is it just dead? It won't compile against 1.4
22:03.46*** join/#asterisk zeeesh (n=aadilism@202.125.143.67)
22:04.00*** join/#asterisk swagger (n=Steve@ip68-227-26-15.lv.lv.cox.net)
22:04.11mvanbaakPrimer: chan_sccp is dead
22:04.16Primerbogus
22:04.26Primerthere's no sip firmware for my 7920, last I checked
22:04.28Strom_MPrimer: see chan_skinny
22:04.38mvanbaakchan_skinny works ok
22:04.42PrimerI thought chan_skinny was even older/less maintained
22:04.43Strom_Mwhich, incidentally, is right there when you make menuselect
22:04.47mvanbaakI use it in production on a couple of sites
22:04.52Strom_Mno, qwell fixed it up
22:04.56Primerahh cool
22:05.14mvanbaakStrom_M: actually, chan_skinny is active again
22:05.17Strom_Mi'm trapped in a universe factory
22:05.19mvanbaaklook at the bugtracker
22:05.22Qwell[]quite active, actually
22:05.22Strom_Mcocks
22:05.36Strom_Mmvanbaak: thats what i said
22:06.12*** join/#asterisk Wag3Slav3 (n=gregg@71-32-119-21.bsmr.qwest.net)
22:06.33Wag3Slav3<PROTECTED>
22:06.44*** join/#asterisk GothAlice (n=amcgrego@209.161.123.42)
22:07.19GothAliceHow can I allow a user on a call to enter a command, say "112", and have asterisk perform an action, without disconnecting the call?
22:07.36mvanbaakGothAlice: look at features.conf
22:07.40Strom_MGothAlice: features.conf
22:08.02Strom_Mand let us not forget features.conf while we're at it
22:08.16mvanbaak;)
22:08.52GothAliceAaah... guess I shouldn't have deleted that file when I first set up the basic server... XD
22:08.53*** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net)
22:09.00waverly360Any of you guys played around with the Aastra 57i phone?
22:09.10__DAWwaverly360: yup
22:09.14knarflycan I record a call once it's started?
22:09.25Strom_Mand let us not forget features.conf while we're at it
22:09.29Strom_Mknarfly: features.conf
22:09.30mvanbaakknarfly: yeah
22:09.51mvanbaakknarfly: look at features.conf
22:09.51mvanbaak;)
22:09.52Defrazwhen a caller calls into a DID on my PRI, it seems to work great when I ring an extention, but I would like an incoming DID to go straight to a fax tone, and then convert that to a pdf when it is done receiving. Is this possible or does anyone know of some howto's on doing this.
22:10.00Strom_Mi should have an array of one-touch answers for this channel
22:10.06DefrazLike when someone dials 666 for a test fax tone.
22:10.10waverly360__DAW: Well..I have one here..and it has two 'sidecars'...one's an lcd display, and the other looks like a couple of columns of buttons that you slip paper into.
22:10.18mvanbaakStrom_M: irssi has /alias
22:10.18mountainm2kDefraz:  Yes, but not directly with Asterisk
22:10.24waverly360__DAW: can they both be attached to the phone at the same time?
22:10.29Strom_Mmvanbaak: eh, i'd like buttons
22:10.38Defrazmountainm2k: Really?
22:10.44mountainm2kDefraz: I have a solution that works using Hylafax -- check out "iaxmodem", read the docs...
22:10.45mvanbaakStrom_M: you can use F keys in irssi as well
22:10.49__DAWwaverly360: not really sure, ive never used the side car.  As with most aastra phone, im not terribly impressed.
22:10.55mountainm2kDefraz: Yup.  Each of our people have their own fax DID.
22:11.14*** part/#asterisk Wag3Slav3 (n=gregg@71-32-119-21.bsmr.qwest.net)
22:11.25Defrazcan you set the pri to split up the traffic for asterisk and for your solution.
22:11.51mountainm2kDefraz: You don't set the pri to do anything -- Asterisk handles the call, passes it off to an IAX endpoint
22:11.54Strom_Mdec: DNIS
22:11.55Strom_Mer
22:11.57Strom_MDefraz:
22:11.57DefrazI am checking out Iaxmodem right now.
22:12.01mountainm2kDefraz: which happens to be a soft-modem
22:12.08mountainm2kDefraz: (iaxmodem)
22:12.17Defrazhmmmm cool cool.
22:12.41mountainm2kDefraz: The docs for iaxmodem are pretty good -- read it, and get back with us if you can't figure it out
22:12.46DefrazBecause I have been using the fax detection deal and it works when the tone is provided but sometimes you get a 15 year old fax machine.
22:12.53DefrazOkay thanks a ton
22:13.00mountainm2kDefraz: (although I don't hang out here often -- maybe I should write a wiki)
22:13.25mountainm2kDefraz: I didn't try the fax detection stuff, but I knew I wanted actual DID faxing
22:14.13mountainm2kDefraz: I actually hacked the script in hylafax to do a database lookup (in the Asterisk Realtime DB) for the user's email address associated with their voice mailbox
22:15.21*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
22:16.02mountainm2kDefraz:  http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem
22:18.47*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
22:18.50*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
22:19.38Defrazsorry mountainm2k I was away reading.
22:20.27GothAliceLast question, I'm trying to run some commands before a prefixed call (_99.) then place the call back into the call tree to continue on its way (e.g. dial a local extension, dial out, etc).  Dial() isn't working for me.  And googling on voip-info.org isn't helping...
22:20.37*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
22:20.52sahafeezquestion - is anyone running on ubuntu-server?
22:20.53*** part/#asterisk Primer (n=vi@sh.nu)
22:21.12sahafeezi am upgrading a box on sunday and i was thinking about moving from slackware
22:21.40mvanbaaksahafeez: we use ubuntu-server only on the hardware
22:22.01mvanbaakit's because vmware has prebuild kernel modules for it
22:22.17mvanbaakso it's running on the steal, only to be the host for vmware
22:22.26mvanbaakall real stuff is running in a vm
22:22.45mvanbaakand all vm's run debian
22:22.52mvanbaakthat's on our linux platform
22:22.52sahafeezhum, i do not want that. i just want a single ubuntu box running asterisk with a PRI card.
22:23.16mvanbaakah, no idea about that sorry
22:23.25sahafeezthanks
22:24.32*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
22:28.02*** join/#asterisk electus (i=electus@asphaleia.SuxOS.org)
22:30.00waverly360Is there any way that a polycom phone and say an aastra phone can detect each other's presence?
22:30.23sahafeezwow. never seen it this slow
22:32.25[TK]D-Fenderwaverly360, Yes,
22:33.12waverly360[TK]D-Fender: How does that work?  The polycoms are specifically built to detect each other's presence using the buddy watch stuff....how can an aastra phone interact with that?
22:33.33*** join/#asterisk zeeesh (n=aadilism@202.125.143.68)
22:39.11Strom_Msahafeez: i run asterisk on debian, which is close enough
22:39.40sahafeezthanks. i need to read about how to get the zap kernel stuff in.
22:39.51Strom_Mit's easy
22:40.11Strom_Mjust make install zaptel, assuming you have the kernel headers installed
22:40.38mvanbaakapt-get your kernel headers
22:40.57mvanbaakand then you can follow the simple: make install
22:42.51*** part/#asterisk boster (n=boster@38.98.147.68)
22:43.31*** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)
22:44.18sahafeezok. cool. i normal do this on slack or gentoo so it is all by hand. not quite used to the apt stuff
22:45.33Strom_Mapt is sex on a stick
22:46.14sahafeezhum. i am a ports guy but ok.
22:50.58*** part/#asterisk mountainm2k (n=mountain@165.236.183.1)
22:59.02*** join/#asterisk dflow (n=pch@chaos.addr-arpa.pl)
23:02.58*** part/#asterisk dflow (n=pch@chaos.addr-arpa.pl)
23:05.45*** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)
23:05.45*** mode/#asterisk [+o Corydon76-work] by ChanServ
23:19.17*** part/#asterisk Juxt (n=Juxt@c-71-196-42-107.hsd1.fl.comcast.net)
23:30.43*** part/#asterisk swagger (n=Steve@ip68-227-26-15.lv.lv.cox.net)
23:35.16knarflythanks everyone...got the wW working...
23:35.58*** part/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net)
23:42.02apturaStrom ever look though sundancecom?
23:42.12apturasundance-com that is
23:42.37apturaTelcom guys commenting on asterisk and how it is not as serios as traditional PBX's
23:44.58*** join/#asterisk kombi (n=kombi@213.160.14.18)
23:45.57kombiis the path to the voicemail file accessable in some way (maybe even in a variable)?
23:46.43blitzrageaptura: typically spouted by people who don't understand it
23:48.26apturayea :) thay are putting down hosted PBxs which I can understand. The main gripe is support when its needed or a lack of.
23:48.39Innatechwell, if your definition of serious is "complicated and expensive." Sure.
23:49.22InnatechMaybe those guys want to buy the abandoned Merlin switch I tore out earlier today. Heh.
23:49.45apturaBTW my IP-500 lines are now registered to the outside GW but the time is flashing. Sounds like a ntpd port needs to be opened.
23:50.07apturaIts a very good site check it out sundance communications.
23:50.17blitzrageaptura: well, that makes no sense to me actually... hosted PBXs actually would be easier to support because now you can have a team of engineers who support multiple PBXs in a single physical location. But everyone has their own requirements. Sometimes it makes sense, sometimes not.
23:50.21apturaAlot of vetran Telcom techs are registered in the site.
23:50.47blitzrageveteran means old, and old telecom techs typically don't understand the future
23:50.53aptura:)
23:51.00apturasome are running asterisk
23:51.04blitzragethey're living in the past... I'm the new hotness
23:51.10blitzrageI mean Asterisk is the new hotness.... :P
23:51.14kombimay I repeat my sophisticated, yet highly intersting question?
23:51.29blitzragekombi: the question is still on my screen... wait a while
23:51.38Innatechztdummy still requires secret tricks on virtualized linux hosts to make it go. (last I checked.) Only some of the hosting companies have it working right, which doesn't help the reputation of hosted * .
23:51.47*** join/#asterisk ltd (i=foobar@nox.amused.net)
23:52.13kombiwhere is Fender?
23:52.27blitzrageInnatech: ahh... the way I've built hosted PBXs is not to virtualize anything
23:52.37Innatechblitz: yeah, that's the safe way.
23:52.51blitzrageguess it depends on your definition of hosted PBX
23:54.06apturaand if the dsl goes down then what? tanked unless the pbx is inhouse on a t1 or pots
23:54.21Innatechyup. Dedicated boxen or managed/shared large *'s for multiple organizations both work fine. But people want VPSes, which gets tricky.
23:54.48blitzrageaptura: your phones would not be functional, but all other services of the PBX would still function because it is hosted off site
23:55.18InnatechPeople could call in and get the IVR, they just wouldn't be able to get to your extensions.
23:55.27blitzrageVM would be functional
23:55.42blitzragehowever, if you had the PBX on site, you would not have any of those services
23:55.54blitzrageand small companies can't afford a T1 (hence why they are using a vPBX)
23:56.02InnatechI prefer to have a front and back * for those who want to most of their call handling offsite for BW savings by colocation or whatever. You use an internal Asterisk and an external hosted Asterisk. Tie em together.
23:56.30apturaOr finace the cost of a onside pbc
23:56.31apturapbx
23:56.33InnatechThus if your WAN link goes down you can still call internally and clients can still get IVR and VM.
23:57.07InnatechIf you're relying on a beefy hosted * to do most of the heavy lifting, a very modest * will do onsite for handling the extensions.
23:57.16blitzrageInnatech: yep, I've done that too so you can have 1 or 2 analog lines used when the Internet goes down
23:57.29Innatechyeah, typically the fax/DSL lines.
23:57.32blitzragesomething like a soekris box
23:57.34apturaor setup the pbx with a backup pots line for outgoing calls. I recently installed some cisco routers with DDR and thought that was interesting.
23:57.42blitzrageor a shuttle PC, or whatever
23:57.57InnatechI like commel mini-ITX boards.
23:57.59apturablitzrage how many channels can one of those soekris boxes handle with ulaw?
23:58.09blitzragesomething like 10
23:58.23apturaokay and that may include vm so mabey 8-9
23:58.30blitzragestill 10
23:58.37blitzrageVM is still a channel
23:58.45apturathats what im saying
23:58.46aptura:)
23:58.53aptura8-9 active conversations
23:58.54InnatechThe little Commels can do pretty much whatever you need.
23:59.11blitzragewell, if you're sending the call to a phone, you have 2 chanenls in use (bridged) -- with VM, it's only 1
23:59.24apturagood enough for small offices
23:59.47Innatechyup--very nice for small offices in fact.

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