irclog2html for #asterisk on 20060108

00:00.12[hC]i love the thin line that people operate between
00:00.29trixterand spreading falsehoods about guis.  the gui in astlinux is basically a web based text editor but oops its a gui so it must be evil and not let you do everything and ...
00:00.31[hC]most people that say serious admins dont use a gui, are the people that have their 3-4 unix boxes that are tuned out the ass
00:00.39[hC]i would not call that serious admin
00:00.54[hC]anyone who manages 100, 500, 1000, 10000 machines will know that you need an interface to manage all of that
00:01.12[hC]doesnt mean it has to be a stupid dinky interface, but you need *something*
00:01.31trixtersaying that the free ones overwrite manual changes, when amp itself doesnt even do that ...  and saying that you cant be serious if you use either tool just for some ego related issue..
00:01.34Qwellvi + rsync
00:01.35Qwellheh
00:01.42mog_workamen brother qwell
00:01.50[hC]:P
00:01.52mog_workbut guis arent bad in general
00:01.53Qwelltrixter: amp most certainly does overwrite manual changes
00:01.55trixteryes amen lets restrict cohice!
00:02.00[hC]good guis are good, and bad guis are bad
00:02.00mog_workbut just no good ones yet
00:02.05[hC]look at windows and osx for example.
00:02.09QwellMy guis are good
00:02.10mog_workexactly
00:02.15trixterqwel: no it doesnt.  becuase it doesnt overwrite sip.,conf for example it writes sip_additional.conf
00:02.17trixterwhich is included
00:02.20mog_workwhen i see a good gui for asterisk i will praise it
00:02.22QwellI have a way to manage queues...and it's done RIGHT.
00:02.25mog_workbut i havent see one yet
00:02.27Qwellbut...it won't be released. :(
00:02.37*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
00:02.37[hC]amp is getting *better* but its still young.
00:02.45mog_workcode tease
00:02.53Qwelluntil AMP stops writing to it's own database...it'll forever be junk in my eyes
00:03.00mog_worki think complete realtime would make a gui doable
00:03.01[hC]and is still geared towards the simple everything-on-one-box user.
00:03.06Qwellif it wants to write to a database...great, use realtime
00:03.06mog_workbut editing flat files is dumb
00:03.08trixtermy understanding is that the mac asterisk group has some nice guis, although I havent seen them just  heard about them
00:03.13Qwellmog_work: I use realtime in my guis
00:03.16Qwellextensively
00:03.20QwellYou should see my dialplan
00:03.20mog_workits only way to do it right
00:03.29mog_worklol denon
00:03.33mog_worki dont quite believe that one
00:03.46denonoh, and cpu is at 2% util
00:03.50Qwellmog_work: Did I show you my EVIL func_odbc call?
00:03.52trixter[hC]: lets face it that is how most asterisk installs are done by volume (of installs not necissarily calls).
00:04.03trixtereven the 'real admins' tend to run everything on one box :P
00:04.29razu_hi ... anyone using realtime ?
00:04.32[hC]im not sure 'real' is the word i'd use :)
00:04.39Qwellrazu_: get with the program :P
00:04.44trixter[hC]: heh..  why it was in quotes :P
00:04.54razu_:)
00:05.03[hC]asterisk wise, alot of stuff makes sense to go on the same box.. sometimes.
00:05.08[hC]depending on what you're going for.
00:05.24theblueDo 'user's take calls or make them?
00:05.47fugitivo_Sam--: 6 servers? and you consider yourself a real admin? bah
00:05.50razu_Qwell : so i have this kind of problem ... i have phones behind different nat-s and all i see is the damn public address ... not the nat address ... :(
00:05.55OloBolain the following example, how would I get "yada" from within my php/agi script? exten => 1,2,agi,test.py|yada
00:06.02Qwellmog_work: see msg for very EVIL fund_odbc call
00:06.07Qwellfunc_odbc rather
00:06.07fugitivoi want to see that guy administering 100 servers with a GUI
00:06.23Qwellfugitivo: never gonna happen
00:06.46[hC]again, it depends on the gui
00:06.47razu_Qwell : why is that ?
00:06.53[hC]if you have a gui for each server
00:06.55[hC]that is insanity
00:06.59fugitivo[hC]: any gui
00:07.02[hC]if you have a gui that represents each server inside one interface.
00:07.05[hC]thats different.y
00:07.06*** join/#asterisk Chiardon (n=yo@200.71.58.39)
00:07.15fugitivo[hC]: it's faster vi, scp and rsync
00:07.25[hC]It depends what we're talking about here
00:07.29[hC]Im talking about servers in general
00:07.31fugitivoin general
00:07.32[hC]not necessarily asterisk ones.
00:07.35Qwellmog_work: I had to expand the default buffer of 512 chars
00:07.43mog_workheh
00:07.49fugitivoi have 10 years of linux admin experience
00:07.51fugitivonever used a gui
00:08.02trixterand that is your choice
00:08.08[hC]I have about the same, and I use a mix. the gui's are usually ones ive written myself
00:08.11trixterbut to mock people who want to give others the choice doesnt seem like a good idea
00:08.21[hC]having to push the same config changes out to 100 boxes simultaneously
00:08.38fugitivotrixter: people can use whatever they want, i just don't like when they use a gui and come here for help
00:08.41[hC]it all depends on what you're trying to do.
00:08.43trixterall I am saying is that people should be given the choice and falsehoods about guis and mocking of people that would write them, or use them doesnt seem wise
00:08.53Qwellfugitivo: I think you just summed up the argument...
00:09.00trixteraparently people cant use whatever they want without ridicule however
00:09.01*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
00:09.04QwellIt makes it difficult for US, when YOU use a GUI. :)
00:09.13Qwellbecause we prefer not to use them
00:09.18trixterpeople that use a gui know no more about asteirks than someone who just installed it
00:09.20fugitivoQwell: that's what i wanted to say
00:09.26[hC]The problem with people using asterisk guis, is that the people in here do NOT use them, therefore do not know how they work under the hood, making it hard to help.
00:09.31[hC]whereas if you do it by hand, you know.
00:09.32OloBoladamn gui queers
00:09.43trixterto think that people that are predisposed to use a gui somehow are born with less knowledge than someone who isnt predisposed is silly
00:09.49trixtereveryone starts with 0 knowledge
00:10.05xhelioxOloBola: wtf?
00:10.09QwellI knew how to config asterisk, when I was born
00:10.17Qwellit just...you know...wasn't invented for 20 more years
00:10.20trixterthat explains a lot about your comments here
00:10.21fugitivotrixter: people using a gui, come here asking question related the gui, using terminology used by the gui, they think we know what they're talking about
00:10.27[hC]i agree with both sides. sometimes its much more convenient to NOT use a gui, particularly when you have no CHOICE, and you have to use a gui rather than do it by hand, when you know it would be easier to do by hand
00:10.41trixterahh so you dont like people talking about something you dont know about
00:10.43theblueHow do I change the context for a register statement?
00:10.45[hC]and in other cases, doing it by hand are a time waste, and having a custom gui you've written to manage something can save tremendous amounts of time.
00:10.46trixterrather than them not knowing something you do
00:10.49trixterI start to understand
00:11.20trixter[hC]: the mac asterisk gui has a templating engine to do most common tasks for people very quickly
00:11.21OloBolaI was suddenly reminded of the end of the movie  brokeback mountain, sorry
00:11.30trixteror so they said I dont use a mac so I dont know
00:11.40*** part/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net)
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00:11.45trixterinfact I dont think I have even seen a screenshot of it, just heard its some cocoa app
00:11.58fugitivotrixter: this is #asterisk, asterisk has plain text config files, i like people that come here asking for those files, not a GUI
00:12.18trixterok so we cant talk about asterisk addons either
00:12.21trixterbecuase that isnt plain asterisk
00:12.28fugitivook
00:12.31Qwellbut many people here use it
00:12.31fugitivoenough, i have to go
00:12.34trixteror any questions that relate to connecting asterisk to any non asterisk device
00:12.38trixterthat makes this channel quite useless
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00:13.13fugitivotrixter: look, if you like your web interface, go ahead, do it for regular users, i'm not telling you not to do it, just make sure you create a support irc channel for your gui
00:13.17[hC]trixter: now you're getting it.
00:13.18[hC]:)
00:13.27Chiardonhey, anybody knows how do I find out which R2 variant do I have?
00:14.05trixter[hC]: exactly that and people that are opposed appear to have said clearly they dont like people asking questions about stuff they dont know
00:14.15trixterbecuase they want to be able to answer every question rather than admit they dont know something
00:14.24trixterso rather than learn they want to attack people who do something they dont know about
00:14.31alephcomThat's like the "outtolunch" guy the other night.
00:14.31trixterI think I finally understand the state of things
00:14.38[hC]trixter: its pretty comon.
00:14.47[hC]trixter: in life in general.
00:15.08trixteryeah, why it takes people like me to speak up for choice and freedom rather than lack of choice and lack of freedom
00:15.23trixterbut then I am not all about removing choice from people or making them feel worthless for doing something I dont know about
00:15.30trixterI like to learn, maybe that is the difference
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00:16.58*** part/#asterisk Tiger (n=navajodu@adsl-69-106-247-196.dsl.pltn13.pacbell.net)
00:17.27fugitivotrixter: NOBODY is removing choice from people, if they choice to use AMP, then they should ask questions in #amportal, not here, get it?
00:17.30[hC]trixter: yeah, same for me, ive tried amp. it doesnt suit me, but i tried it out to see if it would.
00:18.26trixteryou remove choice by attcking people for *writing* guis, you remove choice by attakcing people that use them
00:18.31blitzrageGUIs are not meant to give you total functionality of the system. The point is to make the commons tasks easier to do. You still need an administrator. Just like you still need a Windows admin in a networked server environment
00:18.36trixterget it?
00:19.03trixterblitzrage: I tried to bring up the 80% thing and got attacked for suggesting that
00:19.05blitzrageI wrote my own GUI to control several Asterisk boxen from a central portal
00:19.08fugitivotrixter: ok, i'm a bad person
00:19.10trixtersomeone else was attacked for saying they were writing a gui
00:19.23fugitivo"attacked"
00:19.28blitzragetrixter: you can't take things too personally in here
00:19.33trixteryeah told they arent real admins for writing software
00:19.34fugitivoyou should be in politics for the way you talk
00:19.40trixtertold that they are somehow inferior
00:19.42trixterthat is attacking
00:19.47fugitivook
00:19.51fugitivoi'm a bad person
00:19.55fugitivoand i attacked you
00:19.57fugitivobah
00:22.13*** join/#asterisk jbot (i=ibot@rikers.org)
00:22.13*** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org
00:22.15blitzragetrixter: I've been on IRC for years, I know where you're coming from. My statement still stands.
00:22.23Qwellblitzrage: You just HAD to start the os/distro war, didn't you? :)
00:22.26trixtereven though iuts inaccurate?
00:22.30trixteryou dont know where I am coming from
00:22.34[hC]Qwell: blasphemy! osx rules too! you suck for not liking it!
00:22.36blitzrageQwell: just changing the flame war to something else, then I'll leave :D
00:22.43Qwellblitzrage: you're a pro
00:22.45mog_workqwell is infaliable
00:22.48mog_worklike the pope
00:22.53Qwellwhich one?
00:22.56trixterI am trying to gain understanding of the mindset that inhibits choice, that actually takes a bit of backbone
00:28.40*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
00:28.40*** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org
00:28.44mog_workim not just closing em
00:28.54JohnnyCAnyone works with phpagi ?
00:29.18wunderkinhes putting little smilie faces on them too
00:29.18trixterI have
00:29.32trixterahh well I would rather have a smiley face than working code
00:29.51trixterwhy I have 5M of patches to asterisk that I have to install before delviering a system
00:29.53mog_workhey qwell your a marshall... you looked at http://bugs.digium.com/view.php?id=5443
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00:30.18mog_work5M of patches?
00:30.24trixteryeah for various things
00:30.25mog_workthats like half size of asterisk proper
00:30.29mog_workwhat are you doing
00:30.38mog_workare they 0 padded?
00:30.39trixterfiuxing a lot of broken stuff
00:31.01Qwellmog_work: I don't even see files on there
00:31.06mog_workyeah i know
00:31.09Qwellerm, nm, heh
00:31.13QwellI didn't read the lines before that
00:31.13mog_workbut if i assigned it to you
00:31.17mog_workwould you deal with it
00:31.18trixterthe CDR problem where a call is marked failed when its really completed is one fine example that needs to be fixed before distributing a system
00:31.24Qwellmog_work: I don't use that skinny
00:31.26mog_workor just assign it to yourself
00:31.30QwellI use chan_sccp from berlios.de
00:31.30mog_workand... ^_
00:31.31mog_work^
00:31.47[hC]is skinny even worth keeping in *? why not pull in chan_sccp?
00:31.51mog_workand you havent gone and fixed it because?
00:32.14Qwell[hC]: I've been trying, heh
00:32.15mog_workit is as the one from berlios is in legal contest i thought
00:32.43mcquaidhow do i install chan_sip2?
00:33.02[hC]urg. legal contest as in how he got it to work, or licensing conflicts with *?
00:33.09mog_workno not on our end
00:33.19mog_worki wasnt sure if the code was all legal or licensed correctly
00:33.21mog_workon their end
00:33.26[hC]ah
00:33.26mog_worki rember reading that months ago
00:33.51*** join/#asterisk anonymouz666 (n=lynx@200.218.193.6)
00:34.14Qwellwell, I asked the guy about it...I forget what he said
00:34.24mog_workoh?
00:34.36mog_worki imagine the fargo would care about it qwell
00:35.00QwellI think it originally came from asterisk proper
00:35.09Qwellthen it turned into something else, then into what he's got
00:35.18mog_workthats what i heard too
00:35.21Qwelllemme peek at the copyright notices
00:35.28[hC]man. i just need a reliable ssh tunnel that will restart itself if it gets killed. this is not behaving.
00:36.00OloBolashould I just parse agi_request to get variables passed from dialplan to agi? exten => 1,2,agi,test.php|I_NEED_THIS
00:36.06Qwell* The original chan_sccp driver that was made by Zozo which itself was derived from the chan_skinny driver.
00:36.07Qwell<PROTECTED>
00:36.32ManxPoweOloBola, That is not a variable, that is an arguement
00:36.44OloBolaargument
00:37.12ManxPoweso look up how to access "command line arguements" in PHP.
00:37.21Qwelldoes chan_skinny even work?
00:37.24Qwelllike...well?
00:37.58mog_workyes
00:38.07mog_workbut not with features is my understanding
00:38.24Qwellchan_sccp has a bunch of nice features
00:38.27mog_workdo you have a 7920?
00:38.31QwellI wish
00:38.35QwellI'd LOVE to test one of those
00:38.39mog_workdamn i think we have one
00:38.45mog_workim gonna see if i can replicate bug
00:39.57brockj49464Got a question...I need to use dtmfmode=rfc2833 before my register string but when I use AMP it forgets about that.  Any other place I can put it?
00:40.18*** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it)
00:40.42[hC]I have a 7960, 7914, and 7970 running on chan_sccp right now
00:40.43Ariel_brockj49464, in the sip.conf file at the top part you can put dtmfmode=rfc2833
00:41.13Qwell[hC]: cool
00:41.20Qwell[hC]: the berlios one?
00:41.26QwellTry my realtime patch. :)
00:41.28[hC]Qwell: yup.
00:41.31tzafrir_laptopbrockj49464, detete that peer and add it in sip_custom.conf :-(
00:41.34[hC]I dont use realtime yet.
00:41.49[hC]I keep getting ready to try it, and people go 'eeeeaaghh realtime'
00:41.52[hC]so i put it off
00:42.21QwellI love realtime
00:42.32[hC]I'll have to pick your brain about it sometime soon
00:42.45[hC]Im gonna be redoing our infrastructure a bit soon
00:42.48[hC]to use realtime
00:42.56[hC]and be very db driven for config and provisioning
00:43.07[hC]tie it into our client database, etc
00:43.28Chiardonhewllo
00:43.37trixteris there a list of all the things that are different with realtime, especially things that no longer work quite the same
00:43.40Chiardonhey, anybody knows how do I find out which R2 variant do I have?
00:43.41trixteror at all
00:43.56[hC]What the hell is an R2
00:43.56trixterI recall people talking about that on the mailing lists but didnt know if anyone compiled a list in one place
00:44.01[hC]I'd tell you if i knew what it was
00:44.02*** join/#asterisk viLeR (i=1000@66.128.47.232)
00:44.06trixter[hC]: prolly ccitt5 r2 signalling
00:44.27trixteras opposed to R1 which is basically what a bluebox did
00:45.22Chiardonsome kind of legacy protocol
00:45.46wunderkinnvbackgrounddetect isnt working with sip, it does on a pri though :(
00:45.53troybhow do i dial an @ sign?
00:46.17marcus2how 'stable' is AEL?
00:46.41Qwellmarcus2: it 'works'
00:46.48Qwelltry the new AEL2 parser
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00:47.02Qwellhttp://bugs.digium.com/view.php?id=6021
00:47.10mog_workit works just fine, it converts it to dial plan
00:47.18mog_workthere is no 'stableness' to it
00:47.25mog_workhowever not everything is implemented
00:47.28marcus2ah ok
00:47.34mog_worklike im not sure if hints made it in yet
00:47.47mog_workbut you can still do all of the things that are not done in extensions.conf
00:47.48QwellI think AEL2 does hints...
00:47.50mog_workas it reads both
00:47.52QwellI'm not sure
00:48.37marcus2maybe i will just stick with old style dialplans
00:48.54marcus2i'm finally getting around to upgrading my personal asterisk box from 1.0.7
00:49.52mog_workahh
00:49.56trixterael2 has the ability to include other files and even has a cli parser so you can check syntax before laoding into asterisk, and claims to have a better syntax checking, although I havent tried it yet the aditional stuff looks nice
00:49.58mog_workael is cool as crap
00:50.20marcus2maybe i will migrate my work install to ael sometime tho
00:50.57bn-7bcto any developers in here: thank you for a great peace of software, I installed it just as a test, and it works like a dream, this is IMHO a vary good axample tha FOSS works
00:51.08mog_workbut as my dial plan here is 4 lines
00:51.12mog_worki dont have much need for it
00:51.16mog_workyay! bn-7bc
00:51.59tzangermog_work: hahaha
00:52.06tzangerwhat's your dialplan, set callerid and dial?
00:52.09mog_worki am debating realtime
00:52.19tzangerdon't do it man
00:52.25marcus2i've been pondering realtime for work too
00:52.29tzangerit's evil... EEEEEEvil
00:52.33mog_worksend caller id to jabber, see if i want to take call, take call or send it to vm
00:52.33Qwellrealtime <3
00:52.33marcus2but really i'm more interested in some good ldap integration
00:52.45tzangerbut yeah the recent changes you have made to the realtime interface eliminate pretty much all my arguments against it
00:52.52tzangermarcus2: yes that too
00:53.00tzangeronly really for callerID lookup for me though
00:53.05tzangerI wouldn't store a dialplan or IVR in LDAP
00:53.06Qwelltzanger: seen func_odbc?  It's wonderful
00:53.11tzangerno
00:53.49marcus2yeah, callerid lookup is a primary use, but also sip/iax2 auth and some aspects of the dialplan
00:54.11tzangeryes that too, I had forgotten about that which is actually troubling :-)
00:54.36QwellAriel_: missing out
00:54.44Ariel_Qwell, not really
00:54.50marcus2almost all othre aspects of our IT infrastructure "just work" for users once they are added to lda
00:55.11marcus2it would be nice if they could use a sip phone and have their extension routed to it as soon as they wre in ldap
00:55.11tzangerI have recently put my CDRs in postgres
00:55.34marcus2alos, voicemail configs in ldap could be nice
00:56.03tzangerI need to write a patch for cdr_pgsql that will queue 'em up in batch mode if for some reason it couldn't write to the db
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00:57.00Ariel_it's getting too cold tonight. argh can you guys up north of here shut the door on the frig.. please
00:57.14troybhey guy's for some reason im having trouble calling SIP url's
00:57.24tzangerAriel_: :-)
00:57.26troybit's a cisco 7940
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00:58.16Ariel_~weather ktmb
00:58.38alephcomAriel: We're trying to help you get rid of the bugs
00:58.49Ariel_52 is cold for me.. it's suppose to get donw to 40...
00:59.07Ariel_alephcom, I don't have any heaters in my house.. argh
00:59.12tzangerit's -1C here right now
00:59.25Qwell68F...gotta love CA
00:59.34tzangeryeah I love .ca  :-)
01:00.28Ariel_Qwell, nice... but it's only for 2 days that it gets this cold.
01:00.46alephcomAriel:  I can understand you then. :-)
01:01.27Ariel_alephcom, I like cold weather if your homes are able to take it. But most down here are not really made for cold weather...
01:01.49troybAriel_ more like if you can afford the heating bill *grin*
01:01.50QwellAriel_: You're in what, FL?
01:02.45Ariel_Qwell, South Florida. actually Homestead....
01:03.44mcquaidis it still possible to compile chan_sip2? or necessary?
01:04.26Ariel_why???
01:04.45QwellI need JD, and quick
01:04.57tzangerI'm exhausted
01:05.05tzangerI was out in the snow with the kids
01:05.11Ariel_it's 2nd game start time.....
01:05.29tzangerI found it is actually easier to weed in the winter than the summer
01:05.46tzangerpulling out bigass burr bushes
01:06.00tzangerheh
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01:08.04slappingtwW had a foot of snow in December here in Missouri.  Today it was in the 60's.
01:09.05Ariel_slappingt, our high was 65 that is the lowest it's been for a high in a very long time here.
01:11.44slappingtcold in the 60's there and people here broke out the shorts and short sleeves.
01:13.07Ariel_burr
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01:17.35*** part/#asterisk slappingt (n=randygre@pcp03933849pcs.sthind01.mo.comcast.net)
01:23.51harry8has anyone had problems installing zaptel on FC4?
01:23.59Qwellharry8: none
01:24.04QwellWhat problems are you seeing?
01:24.04harry8really?
01:24.20harry8I tried compiling from source but the problem has to do with the kernel sources
01:24.26harry8and then when I tried an RPM
01:24.28Qwellso install the kernel sources
01:24.32harry8i did
01:24.38harry8but i keep getting this error
01:24.48harry8Module zaptel does not exist in /proc/modules
01:25.00harry8missing /dev/zap!
01:25.02Qwellon an rmmod
01:25.11Qwellharry8: udev?  read README.udev
01:25.28harry8what's udev?
01:25.35Qwelllike devfs, but not
01:25.44Qwell(any new distro likely uses it)
01:25.44mcquaidis there an up to date listing of all cmds for * besides at the wiki
01:26.04Qwellmcquaid: type help at the CLI, or press tab, or ?, and show applications
01:26.07Qwelland show functions
01:26.08Qwelletc
01:26.27ManxPowemcquaid, you mean like "show applications" in the Asterisk CLI?
01:26.59mcquaidsorry i mean all variables, that you could use say in sip.conf
01:27.16Qwellsip.conf.sample
01:27.52harry8thanks taking a loook at README.udev
01:29.03mcquaidQwell, outboundproxy isn't documented in sip.conf.sample
01:29.21harry8Qwell: did you compile the zaptel devices or use an RPM?
01:30.12Qwellharry8: compile
01:30.23Qwellweekly, heh
01:30.41Qwellseveral times a day for *
01:30.50harry8so do you have to do a full linux compile of the kernel before you use zaptel?
01:31.05harry8fc4 doesn't come with the kernel-sources rpm
01:31.28harry8so I find that to even compile zaptel i have to go through the entire kernel compilation process
01:31.34harry8FC1 was flawless
01:32.00Qwellkernel-headers
01:33.41*** join/#asterisk redd (n=redd@81-178-129-92.dsl.pipex.com)
01:33.46*** part/#asterisk redd (n=redd@81-178-129-92.dsl.pipex.com)
01:34.21harry8hmm
01:34.23mcquaidi'm trying to ensure outboundproxy is set, i should see that when i start * no?
01:34.34harry8i have installed glibc-kernheaders-2.4-9.1.94.i386.rpm
01:35.16harry8but the linux/version.h file only gets made when I do a make
01:36.30Qwellharry8: Does FC4 still have a kernel-headers package?
01:37.01harry8when I did a search for kernel-headers on rpm.phone.net
01:37.04ManxPowemcquaid, Asterisk does not support "outbound proxy" in the traditional sense of SIP.
01:37.18Qwellharry8: no, use yum
01:37.26harry8?
01:37.29harry8oh
01:37.43mcquaidManxPowe, isn't that what this is?: http://www.voip-info.org/wiki-Asterisk+SIP+chan_sip2
01:37.56mcquaidnear bottom
01:38.09ManxPowemcquaid, I believe chan_sip2 is still in development.
01:38.21ManxPoweyou can check sip.conf.sample in the asterisk source
01:38.35mcquaidyes but i read that most of chan_sip2 is ported to chan_sip
01:38.40mcquaidi checked sample, no mention
01:38.41harry8i always wondered what yum was for heheh
01:38.43filechan_sip2 doesn't exist anymore
01:38.52mcquaidbut outboundproxy and port is in chan_sip.c
01:39.24mcquaidi realize that chan_sip2 no longer exists, trying to confirm if this variable made it over
01:39.37mcquaidit seems like it did as it's in the c file
01:39.54mcquaidbut can't find it documented anywhere else
01:40.27mcquaidi need it for my voip provider
01:40.46ManxPowemcquaid, then you would be one of the few.
01:41.02Qwelloutboundproxy is in chan_sip, yes
01:41.07mcquaidhmm, maybe i'm on the wrong path
01:41.18mcquaidright now, i can call my * box with my sip client fine
01:41.28mcquaidi can call out to my voip to a pstn number
01:41.41Qwell"You can't have a dynamic outbound proxy, you big silly head"
01:41.42mcquaidbut if i call in from pstn i see asterisk answer it but hear nothing either way
01:41.45Qwellheh
01:42.03ManxPowemcquaid, What is your provider?
01:42.05Qwellit's a general option
01:42.13mcquaidvbuzzer.com
01:42.31Qwell[general] \n outboundproxy=ip \n outboundproxyport=1234
01:42.51ManxPowein sip.conf, of course
01:42.53mcquaidit definitely was a general option for chan_sip2 but according to wiki it's a peer option now
01:42.58*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
01:43.03Qwellmcquaid: according to the code, it's a general option
01:43.10mcquaidok i'll try there again
01:43.19mcquaidhow can i verify it's being set?
01:43.31Qwell<PROTECTED>
01:43.31Qwell<PROTECTED>
01:43.34Qwelldunno
01:43.52*** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
01:44.10mcquaiddamn, i see asterisk answering it, but no sound
01:44.15Qwellokay, it can be a peer option too
01:44.23Qwellbut it's kinda funky, and very similar to host
01:44.29ManxPowemcquaid, no sound is usually a codec mismatch or a NAT problem
01:45.10mcquaidwell, in the two clients i've got working with this voip provider (linphone, twinklephone) it only worked once i plugged in the outboundproxy
01:45.17mcquaidi didn't need to enable nat nor stun
01:45.30mcquaidtrying to get the same going with asterisk
01:45.49mcquaidi'm new to this so i might be on the wrong path, but i can't think of any other reason why it's not working
01:45.59mcquaidwhen i debug rtp i see packets going back and forth
01:46.08harry8kernel-smp-devel-2.6.11-1.1369_FC4.i686.rpm fixed the problem thanks
01:46.14mcquaidand as i said, when i call out i can hear/speak fine
01:46.17harry8now i'll try to figure out how to use YUM
01:46.23Qwellharry8: kernel-devel...silly Redhat
01:46.29ManxPowemcquaid, Is the IP info inside the packets the internal or external address of your network?
01:46.30Qwellnever would have said to install that
01:47.02marcus2wtf
01:47.06marcus2i have an 888 DID from nufone
01:47.22marcus2when calls come in on it now, the callerid is set to some number i've nevr seen before
01:47.26marcus2always the same, it seems
01:47.46mcquaidchecking
01:49.03mcquaidManxPowe, internal
01:49.19mcquaidbut i was trying to even just have asterisk to answer and play a file
01:49.30ManxPowemcn, there's your problem
01:49.33mcquaidfor those it said my voips ip obviously
01:49.35ManxPowemcquaid, that s
01:49.53mcquaidany suggestions how to fix? i've already tried setting the externip and local
01:50.29Ariel_nat=yes
01:50.44mcquaidtried that
01:51.47*** join/#asterisk fgravato (n=frankie@mail.serversforless.com)
01:52.26*** part/#asterisk fgravato (n=frankie@mail.serversforless.com)
01:52.30*** join/#asterisk fgravato (n=frankie@mail.serversforless.com)
01:52.33fgravatohello
01:52.45fgravatoanyone know if you can do more then one bindaddr in iax.conf?
01:52.57fgravatofor both internal & external ips
01:53.56Qwellfgravato: 0.0.0.0
01:54.01mcquaidor could siproxd maybe help here?
01:54.10Qwellthen let iptables take care of blocking
01:54.47mcquaidi thought for sure it was outboundproxy, but that doesn't seem to help
01:54.55Qwellmcquaid: pastebin an rtp debug
01:55.01mcquaidok
01:55.02QwellYou just get no audio, right?  Everything else works?
01:55.06mcquaidyes
01:55.14mcquaidno audio on incoming from outside
01:55.25Ariel_ports blocking
01:55.27Qwelland you're positive you've got the rtp ports open?
01:55.32QwellAND forwarded?
01:55.34mcquaidaudio on outgoing calls to pstn, audio if i call from sip phone to * server
01:56.32Ariel_incoming from outside voip provider
01:57.20mcquaidQwell, http://pastebin.ca/36146
01:57.32Qwellanswer the other question, then I'll look at it :p
01:57.41mcquaidand i am behind a firewall, i have no access to it.
01:57.50QwellThen you're toast
01:57.56mcquaidi thought i can get other sip clients working fully, why not *
01:57.57X-Robheh
01:58.15Ariel_X-Rob, long time no see. how are you doing?
01:58.16QwellWhat is 10.1.0.12?
01:58.30Qwelland did you change your RTP ports?
01:58.30mcquaidmy ip behind lin server
01:58.34X-RobAriel_, not bad. Been busy over xmas/ny
01:58.34Ariel_canreinvite=no
01:58.39file[laptop]...hi
01:58.44Qwellfile[laptop]: http://pastebin.ca/36146  fix
01:58.45Qwell:P
01:59.01file[laptop]externip/externhost AND localnet set? RTP ports forwarded?
01:59.05file[laptop]if not - ha too bad
01:59.08*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
01:59.25Qwellfile[laptop]: No ":("
01:59.33Ariel_file[laptop], he has no access to his fw
01:59.40mcquaidi've tried setting externip, and local
01:59.50file[laptop]then your firewall is probably blocking the RTP stream
01:59.52mcquaidi've done nothing with rtp ports
02:00.01file[laptop]no audio foo you!
02:00.12file[laptop]Hackers is on
02:00.12mcquaiddoubtful, i can call out.  and as i said why do sip clients work fine
02:00.47file[laptop]doubtful? there's no RTP packets coming in to you from the provider :P
02:00.51Ariel_mcquaid, outbound opens the rtp port for outbound calls but it's not opening for incoming
02:01.11mcquaidin the sip clients i've tried, i dont enable nat or stun, just put in the outbound proxy and they both worked
02:01.17Ariel_firewalls most of the time will allow outbound first then inbound. But not inbound first
02:01.42file[laptop]oh well, I've said what I've said...
02:01.43mcquaidthen i wouldn't be able to use sip clients directly to the voip no?
02:01.59mcquaidif it was a firewall issue
02:02.07file[laptop]do this
02:02.16file[laptop]make sure localnet and externip/externhost is set
02:02.19file[laptop]do a sip debug
02:02.27perlmonkeeafter placing a call, and having it connect, I press *1 - the asterisk console tells me that I have started recording.
02:02.30file[laptop]and see what Asterisk says to use for the IP and port in the SDP content
02:02.43perlmonkeeThe .wav files appear in the proper place under /var/spool
02:02.47mcquaidok
02:02.50perlmonkeewhen I stop recording, the files are merged.
02:02.52file[laptop]if it's your public IP address, and an rtp debug still says no RTP coming from your provider - then chances are it's getting blocked at your firewall
02:02.57perlmonkee(automagically by asterisk it would seem)
02:03.02perlmonkeeI now try to play this resulting wav file
02:03.07perlmonkeeand it contains several minutes of total silence.
02:03.17mcquaidi was setting externip but not externhost before
02:03.28file[laptop]externhost is if you use a dynamic DNS host
02:03.37file[laptop]externip is for IP address
02:03.42file[laptop]you just need one
02:03.44mcquaidno fixed ip
02:03.48file[laptop]but you have to use it in connection with localnet
02:03.51file[laptop]or else it's useless
02:03.57perlmonkeecan anyone help me with automon and call recording?
02:04.13file[laptop]Qwell: wanna help me hax0r teh g1b50n?
02:05.49Qwellfile[laptop]: k
02:06.14mcquaidfile[laptop], ok i did what you said
02:06.28file[laptop]pastebin, pastebin, it's so cool
02:06.39mcquaidone sec
02:07.30Qwellperlmonkee: What are you trying to play it with?
02:08.13mcquaidfile[laptop], http://pastebin.ca/36147
02:08.35*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)
02:10.29file[laptop]you are indeed sending public IP in SDP
02:10.53mcquaidthats good right?
02:11.08jahanihow can i use asterisk to send fax by email?
02:11.23file[laptop]yes, it means they are sending audio to you on that IP
02:11.33Qwellbut the firewall is eating it!
02:11.43mcquaidweird, cause i hear nothing either way
02:11.57file[laptop]jahani: well, first you research... as there is documentation, and scripts and stuff to do... exactly that!
02:12.46mcquaidwhy isn't the firewall eating it, with a basic sip client connected directly?
02:13.16mcquaidthere must be some way to mimic what these sip clients are doing differently
02:13.57jahanifile[laptop] where i can download the script?
02:14.07file[laptop]jahani: I don't have links here, why don't you Google?
02:14.24Ariel_argh it's too early to be this cold outside tonight...5.0°C (41°F)
02:14.26file[laptop]for example I just typed "asterisk fax e-mail" into Google, and the first result is:
02:14.33file[laptop]http://www.voip-info.org/wiki-Asterisk+Fax+to+email
02:14.46file[laptop]which has a macro, and a script
02:14.51file[laptop]to do exactly what you want :)
02:15.34jahaniok thank you very much
02:16.37mcquaidfile[laptop], my sip.conf if you could look and see if something doesn't look right
02:16.40mcquaidhttp://pastebin.ca/36148
02:18.10file[laptop]never used an outbound proxy myself
02:19.21Qwellfile[laptop]: watch with me!
02:19.33file[laptop]Qwell: you just want to escape your family
02:19.37QwellI do
02:19.40file[laptop]toooo bad!
02:19.45Qwellpwned
02:21.03mcquaidhow well does * work with siproxd? i'm thinking on trying that
02:21.20Qwellmcquaid: Does it follow standards?
02:21.27mcquaidnot sure if i can run siproxd behind the firewall
02:21.33mcquaidi believe so
02:21.38Qwellmcquaid: You'll probably have the same problem with that
02:21.59mcquaidin it's documents it mentions * not supporting outboundproxy but they're probably outdated
02:22.14mcquaiddamn
02:22.30mcquaidbut yet vbuzzers client works fine, so does twinkle, and linphone
02:22.37*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
02:22.38QwellFor incoming calls?
02:22.41mcquaidhow are they getting around nat/firewall issues
02:22.43mcquaidyes
02:22.45QwellIt's a whole different deal with incoming vs outgoing
02:22.49mcquaidthey work for both in/out fine
02:23.42*** part/#asterisk Naturalblue (n=Kay@195.26.12.229)
02:24.06harry8hey Quell, what is a good development tool to use for developing asterisk?
02:24.21harry8i am new to the linux dev arean
02:24.25harry8arena
02:24.49harry8is Eclipse a good editor for c?
02:26.30sivana<PROTECTED>
02:28.47Qwellharry8: I just use nano, heh
02:29.35harry8where is a good place to get started just by looking at the source?
02:29.45Qwellharry8: doxygen docs
02:29.51tzangerharry8: somewhere where you have an interest
02:29.55tzangerharry8: I started in zaptel
02:30.02tzangerthen pulled in chan_zap
02:30.04harry8I am interested especially in the part i mentioned earlier about the soft video client + hard phone
02:30.06tzangerjumped into chan_iax2
02:30.07harry8integration
02:30.15tzangerdropped down to channel.c
02:30.18harry8similar to the cisco VT Advantage thing
02:30.23tzanger(or rose to channel.c, I guess it's higher level)
02:30.29tzangerdid some pbx.c stuff
02:30.31Qwelllower level
02:30.40tzangerbut you need to start somewhere where you have interest
02:30.41tzangerQwell: ??
02:30.45tzangerchannel.c is hte highlevel channel code
02:30.52tzangerthen you drop down as you get closer to the wire
02:31.01tzangerwhether that wire be cat5 or cat3.  :-)
02:31.04Qwellokay, long day :p
02:31.04*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net)
02:31.04harry8yes that's the part i would like to figure out not sure how difficult it is
02:31.16harry8or if it's even possible to do with asterisk..
02:31.22tzangerharry8: asterisk is not easy to understand but in its own sick and twisted way it does make sense
02:31.32mcquaidQwell, any other suggestions in the light that other clients work?
02:31.54Qwellmcquaid: I saw it was sending rtp to 5062...is that what asterisk expects them on?
02:31.59harry8tzanger: how would you approach this scenario: I have a hard phone + a soft video client
02:32.10tzangereek
02:32.11Qwellor maybe that's just the source port
02:32.12tzangernot exactly sure
02:32.15mcquaidi noticed that too, i don't know why it's doing that
02:32.15harry8I want to have the application so that when I receive a video call on my hard phone
02:32.21mcquaidi bound my local ip to port 6000
02:32.23harry8I get a video pop up on my screen
02:32.25tzangerit'd definitely be SIP and chan_sip.c eats babies, this is a documented fact.
02:32.25Qwellit surely sees them
02:32.29harry8and vice versa
02:33.04*** part/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt)
02:33.08tzangeranyway I am totally wiped, I'm heading to bed
02:33.11harry8from a high level i would guess that there would be some way to configure your SIP phone to send a message to your computer to launch a video screen
02:33.18harry8that would be a cool feature :0
02:33.23Qwellharry8: You have canreinvite=no, right?
02:33.34tzangerharry8: nah, just before you Dial() the sip phone you do something (System?) to notify the PC
02:33.36Qwellerm
02:33.38Qwellmcquaid:
02:33.46mcquaidyes
02:34.16Qwellmcquaid: What happens when you get an incoming call?
02:34.21Qwellin your dialplan
02:34.30QwellDo you Dial() a SIP phone or something?
02:34.46harry8qwell: hmm not sure let me check
02:34.50harry8eg:
02:34.52mcquaidexten => _4.,1,Dial(SIP/home)
02:34.54QwellTry an Answer() then an Echo()
02:34.57harry8i have a 7960G phone
02:35.03harry8if I pick that up
02:35.10mcquaidbut before i was trying to just have * answer and play monkeys
02:35.12*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
02:35.22mcquaidit says it's playing it but no audio
02:35.34QwellI thought it was incoming audio you didn't get?
02:35.35harry8then a software client (like xten eyebeam?)
02:35.44Qwellharry8: wrong person, sorry
02:35.48harry8oh
02:35.51harry8heheh duh!
02:35.52perlmonkeecan anyone help me with automon and call recording? I end up with wav files containing silence.
02:35.56*** join/#asterisk jyukes (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)
02:36.20mcquaidon incoming calls from outside in, i get no audio either way.  if i call out to pstn i get full audio both ways
02:36.21Qwellperlmonkee: How are you playing the files?
02:36.26xhelioxperlmonkee: Maybe no one is talking. ;)
02:36.36Qwellmcquaid: okay, I see
02:36.48perlmonkeeQwell: I tried with `play`, `mplayer`, and `xmms`
02:36.56Qwellperlmonkee: try to play them from asterisk
02:37.02perlmonkeexheliox: There was plenty of talking =/
02:37.03jyukeswhen i do Ringing in dialplan it gives SIP msg 180 with no audio.... how do u get 183 Ringing with 1 way audio?
02:37.25xhelioxperlmonkee: Maybe it was telepathy and you just don't realize it.
02:37.41harry8tzanger: so technically, could you overload the Dial() functionaly to add dialing to video? or create a new function called DialWithVideo()
02:37.45perlmonkeexheliox: amusing but not helpful.
02:37.46harry8or something like that :p
02:37.58Qwellharry8: * technically supports video
02:38.06xhelioxperlmonkee: that's my purpose in life.
02:38.20jyukesdoes anyone know how to do Verizon-style ringback music?  So when i call an extension instead of ringing, it owuld play a mp3?
02:38.32jyukesthis would need to be a SIP 183 message with 1way audio
02:39.06harry8i've seen that it supports video, i was just wondering if there are any plans to do the hard phone + soft video client integration
02:39.19harry8that would really bring good desktop video conferencing i think
02:39.34harry8so you don't have to have a phone that has a video camera
02:39.38Qwellharry8: * would have to split the audio and video packets, and send them to different devices
02:39.55Qwelland good luck syncing outgoing audio/video
02:40.05harry8Couldn't it be setup
02:40.20perlmonkeeharry8: What he's saying is "Nice idea, but more trouble than its worth for obvious technical reasons."
02:40.29Qwellno, not even that
02:40.33harry8hmm
02:40.40Qwellnice idea, doable, may look like crap
02:40.46harry8Cisco has done it with their VT Advantage product
02:40.58harry8works very well
02:41.15Qwellharry8: the ip phone probably calls the PC (in fact, I know it does), which then calls the PBX
02:41.31QwellSo, all traffic would be done through the PC, including audio
02:41.39harry8imagine every hard phone you have could technically become a video phone :)
02:41.40Qwellthe client splits the audio from the video, and sends it to the phone
02:41.51Qwellhowever, the model you want, is different
02:42.01harry8yes i know the cisco VT has the phone call the PC
02:42.01Qwellthe PBX would need to split the audio and video, and also combine them
02:42.27Qwelland also...if done at a purely RTP level...
02:42.34Qwellyou would almost instantly support sccp and mgcp
02:42.42harry8hmm
02:42.48Qwell(big bonus there)
02:42.57harry8sounds hard :)
02:43.15Qwellharry8: I'd ask on the asterisk-dev mailing list.  Lay out your plans, ask if they have any advice/input
02:43.35harry8good idea you think?
02:43.50Qwellyep, I think so
02:44.00harry8well great that's one other person heheh :0
02:44.01QwellI mean...in reality, all softphones should just support it
02:44.22harry8i just think that a feature like that
02:44.26Qwellthe phone gets rtp (which contains video, and perhaps empty audio), and proceeds happily
02:44.34harry8would really bring video and desktop conferencing to a reality
02:44.55Qwellbut, like I said...good luck syncing the two
02:45.01*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
02:45.22harry8the problem with the cisco VT advantage is that you have to have the pc connected to the IP Phone
02:45.26QwellI don't know much about SIP or RTP, but, you may have issues if one client lags a little, or...
02:45.41harry8agreed, but it would be a good start though heheh
02:45.56harry8so post this on the asterisk-dev mailing list?
02:45.58Qwellharry8: The easiest way, would obviously be to do it how cisco is doing it, and write it as a seperate app, but...
02:46.11Qwellit's definitely not as cool that way
02:46.17perlmonkeeQwell: that would only handle outgoing calls.
02:46.25Qwellincoming too
02:46.35perlmonkeeUnless you have the incoming calls routed to the PC, and then split the audio to the phone.
02:46.37Qwellinstead of sending calls to the phone, you send them to the soft client
02:46.39harry8hmm, maybe i can pay someone to do it
02:46.41perlmonkeeright
02:46.43harry8Ii don't think i could write this
02:46.43Qwellperlmonkee: yes, that's exactly what cisco does
02:46.48*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)
02:47.07Qwellbut, that way is cheesy
02:47.10*** join/#asterisk techie (n=gus@antibala.com)
02:47.15harry8that is a good idea though
02:47.25harry8you definately have more control with the PC
02:47.31perlmonkeeHarry8: I don't see how your plan supports outgoing Soft video client + hard phone situations
02:47.31harry8than with an IP phone
02:47.48Qwellperlmonkee: the PBX would do the work that the cisco client does
02:47.53harry8the pbx would still handle the setup
02:48.02perlmonkeeQwell: I understand how it could do that for incoming
02:48.04perlmonkeebut not for outgoing.
02:48.29*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)
02:48.32Qwellperlmonkee: [bob] \n videopartner=10.1.1.15
02:48.36perlmonkeeYou place an outgoing call from your hardphone and...? * rings your softphone to setup the video channel?
02:48.44Qwell"Ahh, okay bob, I see you have a video partner...I'll send the video to him."
02:48.50harry8yup
02:48.52perlmonkeenot SEND
02:48.55harry8that's what I was thinking
02:49.05perlmonkeeQwell: I'm talking about *you* setting up an *outgoing* video stream.
02:49.07Qwellperlmonkee: and it would also know to receive rtp packets from that
02:49.26Qwellperlmonkee: okay, it's like this
02:49.28perlmonkeeif your initiating the call from the hard phone, the soft phone would have no idea.
02:49.39Qwell[bob] and [bobvideo]
02:49.40harry8the pbx
02:49.46harry8would tell the softphone
02:49.49Qwellbob calls *
02:49.54perlmonkeeunless * rings the soft phone and says "start giving me video"
02:50.01Qwell* says "hi bob, oh, okay, hold on, let me dial bobvideo
02:50.07harry8or you can configure the pbx
02:50.14Qwellbobvideo rings.  answer bobvideo, and now you have video
02:50.19harry8so that it already knows ahead of time that the phone and client support video
02:50.28perlmonkeethis is all wonderful, but I still get silence when using automon =(
02:50.34harry8Qwell: could you write this :)
02:50.44Qwellperlmonkee: for the third time, try playing the file from asterisk
02:50.47harry8and add the video recording funcationality too heheh
02:50.47Qwellharry8: it'll cost you a bit. :p
02:50.48Qwellbut, no
02:50.55harry8how much ?
02:51.04QwellI think there was work done recently to record, to do video better, or something
02:51.14Qwellharry8: I don't think I could write that
02:51.16harry8i guess I'll email the guys at Digium
02:51.20perlmonkeeQwell: I'm sorry, I never saw you say that before.
02:51.27perlmonkeeHow would I play a file from inside asterisk?
02:51.35QwellPlay() from the dialplan
02:51.40perlmonkeehurrr
02:51.54harry8qwell: thanks for the help with FC4 btw got it working
02:51.56perlmonkeePlay(/full/path/to/file.wav) ?
02:52.08harry8qwell: where's the link for asterisk-dev list?
02:52.36Qwellharry8: lists.diguim.com
02:52.45Qwellperlmonkee: relativepath/to/file
02:52.47Qwellno extension
02:53.07Qwelland steals his laptop!
02:54.02file[laptop]nooooooooo
02:54.18harry8thanks guys
02:54.23harry8for all the help
02:54.36perlmonkeeQwell: relative? starting from where?
02:54.39QwellI bet file could write all that! :D
02:54.49Qwellperlmonkee: /var/lib/asterisk/sounds, or something
02:55.06file[laptop]what am I writing now?
02:55.19Qwellfile[laptop]: something like Cisco's video stuff
02:55.27Qwellwhere it has a seperate video client, from the IP phone
02:55.34file[laptop]interesting
02:55.38Qwellvery
02:56.06Qwellbut, cisco's way is kinda hackish...it has to be a seperate client.  Would be VERY cool to have * do it
02:56.24*** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net)
02:56.47moraleanyone know why whenever i call my home phone number, the status code it returns after the call is over is FAILED?
02:56.50Qwellnot only would you have near instant support for sip, sccp, and mgcp, but also for just about any client, AND...you could like...
02:57.01Qwellyou could totally have an SCCP hardphone, and a SIP video client
02:57.40Qwellwhich is exactly what * is good at doing, is interprotocol communication
02:57.43rob0morale: Asterisk doesn't think the call went well. Try calling for pizza delivery. ;)
02:58.01Qwellmorale: there is a bug on the tracker.
02:58.03rob0seriously, I wondered about that too.
02:58.04moralecan i make Zaptel return a nicer return code?
02:58.07moraleah ok
02:58.17QwellI think all of them are relating to zaptel too...odd
02:58.28Qwellmorale: could you read the notes, and post your findings?
02:58.49QwellI'm starting to think it's directly related to zaptel.  I think everybody has said it only happens with incoming zap chans...
02:59.06Qwellzaptel/chan_zap rather
02:59.10moraleQwell: yeah, i will have a look
02:59.54perlmonkeeQwell: I'm now playing it from inside asterisk
02:59.55perlmonkeeI get silence.
03:00.05perlmonkee(also it was Playback(), not Play() )
03:00.13Qwellmorale: 5918
03:00.20Qwellperlmonkee: right, yeah
03:00.45*** join/#asterisk zotz (n=zotz@24.231.47.175)
03:01.08harry8there i've sent it off
03:01.09*** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net)
03:01.10harry8:0
03:01.34perlmonkeeso... I get huge valid .wav files with nothing but silence.
03:01.37perlmonkeeany more ideas?
03:02.04Qwellperlmonkee: nope...
03:02.14perlmonkee=(
03:02.17Qwellautomon works great for me...
03:02.24perlmonkeeThen I just hate you.
03:02.24[hC]me too
03:02.26Qwellperlmonkee: does it leave the old files around?
03:02.27perlmonkeeAnd you.
03:02.32perlmonkeethe split channels?
03:02.33perlmonkeeno.
03:02.33[hC]try upgrading to an i686 compiled kernel
03:02.34Qwellyes
03:02.35[hC]:)
03:02.37Qwellheh
03:02.52perlmonkee?
03:03.31moraleis there a function to end the dialplan? like.. s-ANSWERED,2,Exit ?
03:04.15[hC]Hangup ends the dialplan
03:05.58*** join/#asterisk jef_ (i=fischer@p54847C7F.dip.t-dialin.net)
03:08.00perlmonkeeOkay...
03:08.13perlmonkeeI just started a new recording and copied the wav files out before they could be merged
03:08.18perlmonkeethey contain the proper audio.
03:08.47perlmonkeeso when they are merged they become no good.
03:10.11moraleQwell: i've added onto that ticket anything i can think is relevant.
03:11.17*** join/#asterisk dorphalsig (n=dorphals@200.106.223.5)
03:12.02harry8have you guys had any problems with choppy audio?
03:12.07harry8i just finished my install
03:12.17harry8when i access voicemail, the prompt is choppy
03:12.22harry8but the phone conversations are fine
03:12.35moraleslow machine?
03:12.40dorphalsighey, I finally installed R2 support for *, but when I try to run ...
03:12.41harry8uh no
03:12.43dorphalsigI get this
03:12.56dorphalsigJan  7 21:49:57 WARNING[5896] loader.c: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging
03:12.56dorphalsigJan  7 21:49:57 WARNING[5896] loader.c: Loading module chan_unicall.so failed!
03:12.59harry8i am running it in vmware on a 2x dual core operton
03:13.14harry8with 16 gig of ECC ram
03:13.15harry8:p
03:13.25SkramXharry8: shiiit
03:13.26harry8it's not a speed issue
03:13.31SkramXwhy must it run vmware?
03:13.33harry8I didn't have this problem with FC1 though
03:13.37marcus2yeah the ecc makes it *fast*
03:13.46Qwellin vmware?  eww
03:13.50dorphalsig16 gig?
03:13.52harry8i am running the pbx in VMWare
03:13.57harry8yes 16 gig
03:14.00dorphalsigshit
03:14.09harry8yeah it's one of our vm hosts
03:14.12dorphalsigwhat you running, hotmail? :P
03:14.14coppicedorphalsig: chan_unicall cannot find the unicall library. If you installed it in /usr/local, is /usr/local in your /etc/ld.so.conf file?
03:14.26SkramXlol
03:14.29harry8hehhe
03:14.37perlmonkeeharry8: You sound like a UnixShell# user
03:14.52perlmonkeeexcept they don't use vmware because they don't SUCK as much as you do apparently.
03:14.52harry8pretty cool actully with vmware and virtual server
03:15.04harry8unixshell?
03:15.12perlmonkeeharry8: Look into XEN - it will make all the difference.
03:15.27harry8XEN?
03:15.41perlmonkeehttp://www.cl.cam.ac.uk/Research/SRG/netos/xen/
03:15.45SkramXLINUX-VSERVER.ORG
03:15.45harry8why do I suck? what's your problem?
03:15.47perlmonkeemany many times faster than VMWare
03:15.52*** join/#asterisk GD_ (n=GD@ppp20-adsl-165.ath.forthnet.gr)
03:15.56perlmonkeeHarry8: becuase you use VMWare instead of XEN
03:16.21harry8thanks, how old are you? 10?
03:16.49dorphalsigcoppice --> yes, just added. tried again and the same mistake
03:17.04Qwellharry8: posted a reply, with more detail
03:17.08X-Robooh, there's coppice. Heya steve.
03:17.12coppicewell, it cannot find the library for some reason
03:17.16perlmonkeeharry8: Youre just wasting so much of your hardware by using vmware.
03:17.23coppiceX-Rob: you've been quiet lately
03:17.26X-Robdorphalsig, try running 'ldconfig'
03:17.35perlmonkeethe overhead to run VMWare is absurd and disgusting.
03:17.45Qwellno it isn't
03:17.45dorphalsigcoppice --> may I priv ya?
03:17.53X-Robcoppice, I've been running around australia.. Heading off to Darwin on tuesday morning, then back, then up to cairns, then down to melbourne.
03:18.03harry8fine then just give a recommendation don't direct your personal remarks at me I don't know you
03:18.05Qwellyou aren't using vmware workstation, are you?
03:18.11harry8no GSX Server
03:18.20Qwellyeah, with a machine like that, I'd hope to
03:18.21coppicedorphalsig: you'll only get the same answer in private :-)
03:18.23harry8but i don't think it's a problem with the VM
03:18.23Qwellso*
03:18.24SkramXharry8: what is the base-os?
03:18.26SkramXWindows?
03:18.29harry8Yes :)
03:18.35SkramXharry8: LINUX
03:18.36dorphalsigcoppice --> yeah man, but ldconfig returns nothing
03:18.37Qwellokay, NOW it is a problem :P
03:18.43harry8I have a confession, I am a Microsoft Junkie :0
03:18.46X-Robdorphalsig, it's mean to return nothing.
03:18.46SkramXQwell: Indeed!
03:18.50X-Robthat means 'it worked'
03:18.51harry8but i do use linux :)
03:18.53coppiceX-Rob: I'm off to Delhi in a could of hours. travelling sucks
03:19.02X-RobOooh! Delhi is fun!
03:19.04SkramXharry8: come over to my place.. we will break that windows-pussy in :)
03:19.06X-RobLucky bugger!
03:19.21coppiceX-Rob: yeah. right.
03:19.25X-RobNext time I'm there, I'm _so_ going to make time to go to agra.
03:19.26harry8yes I am 80% windows 20% Linux :p
03:19.27X-RobWell
03:19.30X-Robfun if you don't have to work
03:19.34harry8but i do like Linux
03:19.55[hC]Im such a nerd, i get a kick out of reading new pages and new page modifications on voip-info. :( If their RSS feeds worked better, i'd subscribe to the page mods :S
03:20.41X-Rob[hC], keep that up and you'll become uberclued at asterisk.
03:20.52perlmonkeeharry8: You run windows on a dual core dual processor opteron system?
03:20.53dorphalsigX-Rob --> Yet, I still get the same darn message
03:21.03GD_hello.. could you tell me whether all HFC-S PCI A 2BDS0 cards are expected to work in NT mode?
03:21.09perlmonkeeharry8: you do suck =/
03:21.20SkramX:|
03:21.20[hC]X-Rob: yeah, i'm well on my way.
03:21.21coppiceX-Rob: I'm a kind of anti-tourist.
03:21.22harry8perlmonkee: yes one of our servers is a 2 x 2 dual core AMD
03:21.40X-Robhurm.
03:21.41SkramXfor what? a fucking asterisk pbx
03:21.45SkramXthats way overkill, bro
03:21.47harry8no
03:21.51harry8it's not only for the pbx
03:21.53perlmonkeeno, its a VServer box.
03:21.58harry8it is a virtual machine host
03:21.59SkramXOkay.
03:22.01harry8:)
03:22.02perlmonkeea windows VServer box.
03:22.08SkramXYeah, perlmonkee we sell Vservers.
03:22.16X-Robcoppice, dorphalsig -- maybe he's botched the compile somehow? Howabout LD_PRELOAD=/usr/wherever/libunicall.so
03:22.22perlmonkeethats nice, SkramX.
03:22.40SkramXMy company uses http;//vserver-linux.org, it has much less overhead than virtuozzo and even less than Xen
03:23.18*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
03:23.30dorphalsigX-Rob coppice --> I did a find / -name libunicall.so
03:23.42dorphalsigand it returned /usr/lib/asterisk/modules/chan_unicall.so
03:24.00*** part/#asterisk Naturalblue (n=Kay@195.26.12.229)
03:25.03perlmonkeeSkramX: I don't understand the joke.
03:25.10SkramXnothing;
03:25.16coppicedorphalsig: that does make sense. you search for libunicall.so, and get chan_unicall.so back?
03:25.16SkramXTootles, 'Will be back later
03:25.38dorphalsigooops
03:25.42perlmonkeeSkramxX: the URL you provided, as I assume you know, forwards to microsoft.com
03:25.45dorphalsigsearched for chan_unicall
03:25.47dorphalsig:$:$:$:$:$:$:$
03:25.47X-Robcoppice, he's obviously using find --with-esp
03:26.17coppicedorphalsig "whereis libunicall.so" would be more interesting
03:26.23GD_the bristuff packages is referred to in the asterisk's gentoo ebuild changelog. Does this mean that if I emerge asterisk with the bri use flag turned on, bristuff will be compiled as well?
03:28.36SkramXperlmonkee: oops. linux-vserver.org lol
03:28.56SkramXperlmonkee: it doesnt do that for me.
03:29.23SkramXperlmonkee: http://www.linux-vserver.org/ <-- thats what I /meant/
03:29.24perlmonkeeSxramX: yeah, I realized after I said what I did that I directly followed your link - except not - it was improper (http; <- semicolon) and thus FireFox annoyingly went to google, got the first hit for http (microsoft.com) and sent me on my way.
03:29.40perlmonkeeso... we're both idiots =/
03:29.42SkramXperlmonkee: oh, oops.. sorrry.
03:29.44SkramX:)
03:30.18perlmonkeeinteresting to know that microsoft.com is the first hit for http
03:31.07SkramXthat is weird; indeed
03:32.25dorphalsigX-Rob coppice --> I did a find / -name libunicall.so --> just added /usr/local/lib to my ld.so.conf
03:32.34dorphalsigyet I still have the problem :s
03:32.42*** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net)
03:32.58kuku5What do I need to do to have 1 port PoE for a 7940g ?
03:33.38X-Robdorphalsig, did you run ldconfig after you added it?
03:34.40*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
03:39.25_Sam--how would i configure cdr_mysql.conf to log to more than sql database?
03:39.33_Sam--or more than host
03:39.36_Sam--more than 1 host
03:39.47_Sam--sorry i cant speak english tonite
03:39.53_Sam--log to more than 1 host or database
03:40.26_Sam--just add another context for the second one?
03:40.59X-Robdon't think you can.
03:41.09*** join/#asterisk justinu (n=justinu@207.181.0.86)
03:42.18_Sam--i guess i could do the second one via odbc
03:44.20mcquaidanyone have experience with siproxd?
03:46.51*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
03:48.57*** join/#asterisk _Vile (n=vile@90.b160.bendtel.net)
03:50.05_VileHI
03:51.57*** join/#asterisk bmg505_ (n=leon@dsl-146-28-210.telkomadsl.co.za)
03:56.42*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
04:01.47*** join/#asterisk Defraz (n=t0tal@72.24.220.144)
04:03.41*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
04:08.56*** join/#asterisk anon-troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
04:09.10[hC]can i use particular expressions in my dial plan like
04:09.19[hC]10[23] to represent 102 or 103?
04:09.23[hC]like in bash? if so, how?
04:09.44mog_workexactly like that hc
04:09.45[hC]I guess thats technically a valid regular expression
04:09.49[hC]hm.
04:09.54*** join/#asterisk Defraz_ (n=t0tal@72.24.220.144)
04:10.01mog_workyou can do ranges
04:10.05mog_workas well
04:10.10anon-troybi have setup my cisco 7940 with the SIP image however i want to be able to dial URL's
04:10.24[hC]I wonder why this isnt working then.
04:10.28[hC]exten => _1NXXNXXXXXX,/_10[12],Macro(voip-out,${EXTEN})
04:10.33[hC]oops
04:10.36[hC]friggin comma
04:10.42mog_workheh
04:11.34[hC]odd. still didnt like it.
04:12.23mog_worki can double check
04:12.28mog_workbut that is valid regex
04:12.32[hC]yeah..
04:13.15mog_workyeah thats right
04:13.18mog_worklooking at example
04:13.20mog_workconfig
04:13.23[hC]im stupid
04:13.24mog_workbut thats for cid
04:13.33[hC]i removed the comma
04:13.34mog_workbecause its after /
04:13.36[hC]notice anything else missing?
04:13.38[hC]the ,1,
04:13.56[hC]butterfingers in vi tonight.
04:14.32_Vilevi can do that to you...
04:14.34[hC]works now, thanks for humoring me :)
04:15.33mog_workno problem we are here to help
04:16.20coppiceI never knew that
04:16.39justinuheh
04:17.19_Vileshit, I thought this was an idling channel?
04:17.25troybanyone here experienced with cisco ip phone setp?
04:17.27troyb*setup
04:17.39justinumost people are idling here
04:17.46justinua small fraction are active
04:17.51troybyeah makes sense
04:17.57*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
04:18.14troybwork on a saturday night.. thats depressing eh? mog_work
04:18.18justinubigtime
04:18.23mog_worknope
04:18.27mog_workits actually pretty nice
04:18.32mog_worki get to do real work
04:18.38znoGguys, any of you sell X100P cards in .au?
04:18.50_Vilenoone to bother you when noone's around..
04:18.50troybmog_work as long as you enjoy what you do :)
04:18.55*** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net)
04:18.58mog_workyou probably want a real hw in australlia as the lines suck
04:19.22znoGmog_work: actually i need to buy it in AU but it's for another country
04:19.38mog_worki see
04:19.41znoGargentina, where the lines would probably suck a lot more :)
04:19.44mog_workwhich country?
04:19.53mog_worknah argentina isnt as bad...
04:20.03troybanyone know how to setup URL SIP dialing?
04:20.38znoGmog_work: distinctive ring detection doesn't work with argentinian style of distinctive ring... which doesn't make the lines bad, but anyway.. :)
04:21.06*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
04:21.10mog_workif you are doing that troyb why go through asteirsk
04:21.15kuku5What do I need to do to have 1 port PoE for a 7940g ?
04:21.16justinuwhat is the argentinian style?
04:21.40znoGjustinu: no idea, but asterisk doesn't detect it.
04:21.59znoGjustinu: if i ring number 1, or number 2, both come in with cadence 0,0,0
04:22.08justinuhmm
04:22.40troybmog_work im not :) but this is one of the few channels where there are people who are knowledgable
04:23.06mog_workasterisk is not meant to be a sip proxy
04:23.11justinuyou can uri dial thru any sip proxy, check out ser
04:23.19mog_workbut you could probably allow asterisk to route stuff like that for you
04:23.23mog_workits just pointeless
04:23.32justinuor i think you can uri dial off of fwd's proxies
04:23.36troybmog_work yeah i just dont have a production asterisk server at my disposal
04:23.37morale<PROTECTED>
04:23.38moralels
04:23.48mog_workyou dont have a linux box?
04:23.49QwellYou can dial URIs right from you phone, can't you?
04:24.10mog_workyes but most people need proxy to get out of lan qwell
04:24.45Qwelloh, right
04:24.58mog_workthat whole nat issue....
04:24.58troybthe LAN is mine if i need to port forward or anything thats fine
04:25.11troybmog_work there is no NAT on this network :)
04:25.28mog_workoh well than just tell your phone to allow uri dialing
04:25.31mog_workit wont touch asterisk
04:25.32justinucan you do reinvites with iax?
04:25.33mog_workand it will work
04:25.38mog_workyes justinu
04:25.43justinucool, didn't know that
04:25.44troybmog_work i dont mean to sound like an idiot but how do i do that :P
04:25.45mog_workwell its an iax native transfer
04:25.50mog_workthe path leaves the middle point
04:25.53justinuyeah
04:25.57mog_workbut you also lose signalling
04:25.59justinuoh
04:26.01mog_workas iax is all together
04:26.02justinunot so good
04:26.07mog_workso it wouldnt work if you were charging
04:26.10justinuyeah
04:26.13mog_workbut if its internal
04:26.15mog_workits great
04:26.18mog_worklike inter office
04:26.29QwellI wonder if sccp can reinvite...probably not
04:26.29_VileznoG, you should have logs
04:26.30justinuany plans to change that, or would you just say use SIP?
04:26.34_Vileabout not being able to set cadence
04:26.39_Vileon distincitive ring
04:26.49justinuqwell: sccp uses rtp, right? so it should
04:27.02Qwellsccp is stupid
04:27.32mog_workit has been discussed justinu
04:27.32_Vileit's set as a LOG_WARNING, so if you're logging warnings, look for "Unable to set distinctive ring cadence"
04:27.36justinui like sccp
04:27.38mog_worklike cloaning signalling
04:27.42QwellI love sccp
04:27.50troybmog_work how do i allow uri dialing?
04:27.52QwellI want it to have my children
04:27.56mog_workits a phone issue
04:27.56justinubut I like SIP too, they're just two completely opposite ideas
04:28.09_Vilethough, maybe it's deeper than that -- and it's not even getting to that ast_log. hm
04:28.27mog_worksip is good if there is no nat
04:28.27znoG_Vile: i start asterisk with -vvvvvv and where it should wait a couple of seconds to detect the type of ring, it just goes straight in with cadence 0,0,0
04:28.32mog_workotherwise pttttttttttth on sip
04:28.42justinuthere's solutions for that
04:28.46justinusession border controllers
04:28.57QwellI need an sccp proxy
04:28.58mog_worki prefer iax instead
04:28.59znoG_Vile: asterisk should wait a couple of seconds when it gets a ring, AFAIK, and it's not. It just goes straight in, and usedistinctiveringdetection is definately on.
04:29.28QwellIf sccp, sip, and iax were in a cage match...you know which one would win?
04:29.37justinusip would sit on sccp and iax, and win
04:29.50*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)
04:29.51trixtersounds about right
04:29.52justinu10,000 pages of RFCs and drafts
04:29.56Qwellpfft, sip has to stick to the RFC.  No dirty fighting
04:30.01troybmog_work how do i allow uri dialing?
04:30.02*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
04:30.08mog_workits a feature of the phone
04:30.16mog_workit would be different for each phone
04:30.19troybi see though i cant find it in the system menu
04:30.22wunderkinmog_work are we there yet?
04:30.28troyb(cisco 7940)
04:30.28_Vileinteresting
04:30.51mog_workare we where yet
04:30.51justinuwherever you go, there you are
04:31.02mog_workunless its the event horizon
04:31.23QwellThe matter has left the event horizon
04:31.35QwellThat doesn't work, does it?
04:34.08_Vile7940?
04:34.27_Vileyou dial, and switch to alpha I think
04:35.02justinuin the cisco world, how do you setup voip trunks? what protocol?
04:36.22*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
04:36.37_Vilemgcp or sip
04:37.03justinuic
04:37.10_Vileoh ur kidding
04:37.15justinuno
04:37.42_Vileok, what gateway
04:38.00justinui dunno anything about cisco gateways, so i was curious
04:38.16justinui need to find something that speaks SIP-T
04:38.26justinusomething cheap
04:38.29mog_workyou can use asterisk
04:38.29justinuor free
04:38.33mog_workgo grab that patch
04:38.36justinui coudln't get * to work with sip-t
04:38.39mog_workyou mean sip tcp
04:38.41mog_workright
04:38.42justinunope
04:38.49mog_workwhats sip-t
04:39.29_Vileno
04:39.30_Vilehe means:
04:39.38_Vilehttp://www.voip-info.org/wiki/view/RFC3372
04:39.41justinuSIP-T is like SS7 over sip, SIGTRAN
04:39.44*** join/#asterisk Shakhruz (i=Shakhruz@83.221.169.216)
04:40.01_Vilethat would be isup over sip right?
04:40.04justinuyes
04:40.08justinui need it
04:40.10_Vilethat's done....
04:40.15_Viletmk
04:40.30_Vileshit, let me pull my notes
04:40.32justinutmk?
04:40.37_Vileto my knowledge
04:40.42justinuah
04:40.42_Vilethere is ISUP code over SIP
04:40.47_Vilealready in CVS
04:40.49justinusweet
04:40.53_Vileor it's an add-on
04:40.55_Vileor something
04:40.55_Vilesec
04:41.24justinulevel3 wants to give me SIP-T
04:41.28justinuwhich seems nice
04:41.47_Vilewell
04:41.49_Vileits new code
04:41.56_Vileless than a few months
04:41.58justinuit's ok, all their shit is beta too :)
04:42.01_Vileok
04:42.05justinui'll be doing interop tests with it
04:42.12justinuso I'll probably work on the code myself
04:44.39_VileI could be wrong here, I may be thinking of chan_ss7 ---- give me a min
04:44.49_VileI remember isup over sip being in *
04:45.01_Vilesomething to do w/ verisign's sip-7 service
04:45.07justinui tried it, and * chocked on the multi-part mime type that l3 was sending me
04:45.08sivana*sigh*
04:48.11znoGdoes anyone actually use distinctive ring detection in Asterisk? i would *love* to see a working zapata.conf
04:49.26mog_worki use the ring codes
04:50.06troybdamn it now my phone lines went missing off the display lol
04:50.19JunK-Yyo guys
04:50.25mog_workyo JunK-Y
04:50.28JunK-Ywhats up?
04:50.35fileyo yo you crazy french person :P
04:50.37QwellJunK-Y: y0
04:50.39*** join/#asterisk sim5im (n=root@71.196.10.2)
04:50.50JunK-Yhey hey bus driver 202!
04:50.59JunK-Ylo Qwell
04:54.02QwellJunK-Y: If you hear anything more about ETel, you should msg me, so we can work out details if we need to.  Cool?
04:54.16JunK-Yall right houston.
04:54.25JunK-Ysame thing for ya.
04:54.28Qwellyep
04:54.58Qwelland this time...
04:55.04QwellI'm gonna practice playing pool...
04:55.09Qwelland I'm gonna kick yer ass! :P
04:55.19JunK-Yya ya keep dreaming.
04:55.19JunK-Y:)
04:55.21Qwellheh
04:55.23sivanaheh
04:55.34JunK-Ysivana: are ya comin' too?
04:55.38QwellIt was the lighting in that bar, I tell ya...
04:55.45sivanawhere we going?
04:55.49Qwellsivana: ETel
04:55.50JunK-Yetel!
04:55.56sivanain SF?
04:55.58Qwellyeah
04:56.02sivanaI wish... no can do :)
04:56.10sivanajust got back from 12 days in AZ/CA :)
04:56.25*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
04:56.34troybany reason a phone would all of a sudden stop recognizing the lines and say loads error?
04:56.46harryvvnope
04:56.53troybironically it's still loading my .cnf because it says my custom headername.. yet my lines disappeared
04:58.29filemmm melted M&Ms
04:58.48JunK-Ytime for read. see you tomorrow guys.
04:59.38troybdamn it says no load specified
05:02.04troybalso will unplugging the phone too much lead to problems?
05:02.29brockj49464How do I debug if I am getting CIDNAME from my voip provider?
05:02.42*** part/#asterisk Defraz (n=t0tal@72.24.220.144)
05:03.49_Vilejustin, I don't think SIP-T has been implemented.
05:04.06justinuok
05:04.18_Vilebut
05:04.19JunK-Ydumpchan()
05:04.19_Vilecheck out:
05:04.27_Vilehttp://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+faq
05:04.35_Vilehttp://www.voip-info.org/wiki-Asterisk+SS7
05:04.44_Vileand http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+commands
05:05.01_Vilethere is code, just needs to be worked
05:05.56dorphalsigHello, I'm trying (without much success) to install R2 Support for *, I dont know if anybody can give me a hand
05:06.13_Vilealso, search around a bit about asterisk and the Verisign SIP-7 service
05:06.38dorphalsigSo far, I downloaded and compiled all the libs the wiki instructed
05:06.46dorphalsigand patched the channel makefile accordingly
05:07.23justinuvile: thanks for the pointers
05:07.23dorphalsigI'm 90% thru unicall.conf
05:07.39justinutalk to coppice about that, i think
05:07.56dorphalsigCoppice ... you there?
05:08.21coppice<PROTECTED>
05:08.27dorphalsigok can ya help me out?
05:08.53coppiceok
05:09.08dorphalsigI am unsure which group should I put on my unicall.conf
05:09.22troybis there a way to restart a cisco phone gracefully as opposed to pulling the cord?
05:09.28_Vileyes
05:09.34_Vile*,6,settings
05:09.39_Vileat the same time
05:10.00troybthanks :)
05:10.05dorphalsigI am really unsure what to put there, to be quite sincere
05:10.08troybi just get the feeling unplugging = bad
05:10.12coppicedorphalsig: groups are not a unicall issue. they are a general * issue
05:10.27dorphalsigI have some groups defined in my zapata.conf
05:10.35_Vileyou can also telnet into it
05:10.36_Vileand reboot
05:10.45_Vileunplugging is ok, but *shrug*
05:10.58coppicedorphalsig: do you understand what a group is?
05:11.02_VileI use the * command or reboot from the telnet prompt
05:11.04troybfor some reason _Vile the phone lost it's calling functions
05:11.16_Vilejust one phone?
05:11.21_Vileor multiple phones?
05:11.23troybyeah i only have one
05:11.26_Vileok
05:11.34_Vileinteresting, describe
05:11.36troybi didnt make any changes which is weird
05:11.48dorphalsignot quite:s
05:11.55_Viledescribe what you did prior to it losing its call function
05:11.55troybit is now grabbing my load file from TFTP after a reboot (it never used to do that)
05:12.01troyboh,
05:12.03dorphalsigI think its groups of channels with the same priviledges
05:12.04dorphalsigright?
05:12.11troybi made changes to the proxy settings because im not behind nat
05:12.14dorphalsigor rather destined to the same thing
05:12.18dorphalsigIe. outbound calls
05:12.18_Vileon the phone?
05:12.21troybyup
05:12.32troybbecause at that point settings wern't updating from the config file to the phone
05:12.36troybso i made the changes directly
05:12.43_Vileyep
05:12.48_Vilenow you have to make the changes in the config
05:12.57_Vileand let it tftp grab those changes
05:13.12troybso im having these problems because the tftp config file isnt the same as the one on the phone?
05:13.31_Vileno, the phone will align itself to the config that you're letting it grab
05:13.34coppicedorphalsig: a group is a bunch of trunks which are interchangable. all the channels in a n E1 typically for a group. however, you might have an E1 where the telco has dedicated some channels to be incoming and some to be outgoing.
05:13.42_Vileyou have to reconfigure the tftp config file
05:13.44_Vilefor what you want
05:13.48troyboh i agree
05:13.49_Vileand let the phone grab it
05:13.59troybthe file is sitting in the TFTP root nothing is stopping it :)
05:14.05troyband the file is being grabbed without hesitation
05:14.08_Vileok
05:14.14_Vilewhat does the phone screen show
05:14.18troybbut its now saying phone unprovisioned
05:14.20_Vile7940 right?
05:14.21_Vileok
05:14.22troybyup
05:14.27_Vileso, your config file needs to be fixed.
05:14.33_Vileunprovisioned in all caps?
05:14.35troybi wanted a 7960 but it was a little out of my reach :)
05:14.38troybnope
05:14.41_Vileok
05:14.42troybPhone Unprovisioned
05:14.45_Vilecan you view the tftp
05:14.49_Vileerr
05:14.50troybyes i can :)
05:14.50_Vileconfig file
05:14.54troybyup i can
05:15.01troybdo you want me to do a pastebin?
05:15.05_Vilecan you pastebin it, removing the password crap
05:15.06_Vileyep
05:15.06_Vilethx
05:15.11troybno thank you :)
05:15.13troyberr
05:15.16troybthank you
05:15.29troybexpressions dont come out right on irc :P
05:15.43_Vileand do you have TFTP logs?
05:15.46_Vileand pastebin those
05:15.50_Vilemight be looking for dialplan etc
05:15.58troybsure
05:16.02_Vilethx
05:16.04_Vilebrb in 2 min
05:16.53troybhttp://pastebin.com/495996
05:18.35_Vileok
05:18.40_Vilesec
05:19.02*** part/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net)
05:19.12troybnp
05:19.17_Vilewhere part of the screen does it say "Phone Unprovisioned"?
05:19.23troybbottom center
05:20.00_Vileok
05:20.17_Viledo you have TFTP url?
05:20.27_Vilepastebin'd TFTP url?
05:20.34troybi dont understand
05:20.45troybi have a tftp server yes.. it's internal
05:20.48_Vileyou should have TFTP logs in your /var/log directory
05:20.51_VileI need to see those
05:20.52_Vilesec
05:21.04troybohh i can turn logging on and restart the phone sec
05:21.18_Vilein your messages file
05:21.30_Vileok thx
05:21.47troyb*6settings is a lot easier then reaching for the power brick
05:21.52_Vileyep ;)
05:22.00Qwellpsh, SIP
05:22.06Qwellreal men need to do **#**
05:22.19troyb_Vile it also worries me that if i screw that brick up another one is an arm + leg
05:22.34_Vileno worries
05:22.40_Vilerelatively cheap on ebay
05:22.56troybi hate ebay :) i have only bought one thing on it and its under dispute right now bastards
05:22.58_VileI think we got 10 for $200 or $300
05:23.29troybheh
05:23.29troybhttp://pastebin.com/496000
05:23.33_Vileyou can use PoE too, but more expensive
05:23.34_Vileok brb
05:23.37troybyeah :)
05:30.18brockj49464_Vile: Any idea on figuring out if VOIP provider is sending CIDNAME in sip?
05:30.33*** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
05:33.46_VileDo a NoOp,${CALLERIDNAME} right after your Answer
05:33.56_Vileif empty, no.
05:34.11troyb_Vile i replied to your pm
05:34.45_Viledidnt get it
05:35.03*** join/#asterisk Cyon (n=cyon@cyons.net)
05:35.03troybohh
05:35.06*** part/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
05:35.33*** join/#asterisk ManxPowe (i=ewieling@24.sub-70-210-168.myvzw.com)
05:36.47brockj49464I have done a dumpvar when it sets the CallerID and CallerIDName is not set.  Would it set it AFTER answer?
05:37.10_Vileit gets set after answer
05:37.58*** join/#asterisk sim5im (n=root@71.196.10.2)
05:38.01_Vileif you're using an x100p, it gets set after the first ring, if it's pri or sip it's available after answer
05:38.26brockj49464Let me check...
05:38.33_Vileor iax.. I'm leaving that out because a real man's protocol is sip ;)
05:38.48_Vileactually, I like iax
05:39.10_Viletroyb, still no messages?
05:39.59Cyonlol
05:40.18troybi sent you a pm again
05:40.22_Vilenothing
05:40.22Cyonwhat if I were a woman, which is the appropriate feminine protocol to use?  :-P
05:40.26_Viletroy
05:40.26troyboh my..
05:40.28_Vilein here is fine
05:40.31troybokay :)
05:40.35_Vilejust not multi-line
05:40.42troyb<troyb> but it says at the top Troy's IP Phone SIP
05:40.42troyb<troyb> it would of had to read the config file to get that
05:40.46troybohh :P
05:40.52_Viletwo lines is ok
05:40.55_Vilesec
05:40.56Cyonlol
05:40.59*** join/#asterisk GD_ (n=GD@ppp68-adsl-215.ath.forthnet.gr)
05:41.10troybi read the error status of the phone and it said:
05:41.16troyb<troyb> <troyb> W351 unprovisioned proxy emergency
05:41.17troyb<troyb> <troyb> W350 unprovisioned proxy_ backup
05:41.17_Viletroy
05:41.23troybyup
05:41.24_Viledid it say Troy's IP Phone before?
05:41.40troybyes but i edited and rebooted just to see if it would change and it did
05:41.50GD_hello.. has anybody come up with a solution to the unresolved symbols error which occurs after modprobing zaptel and zaphfc?
05:42.03_Vileinteresting
05:42.52_Vileit reset to "Troy's IP Phone"
05:42.56_Vileor "Troy's IP Phone SIP"
05:43.21CyonGD_:  as in where if you have asterisk running, then taking either of those actions, asterisk essentially freezes?
05:43.22troybwell it says SIP in the right hand corner regardless
05:43.30troybright now it says "Troy's IP Phone"
05:43.31_Vileoh, yes
05:43.45brockj49464I have sip debug ip for my provider and I still don't see anything after being answered.  WHat would I be looking for? Are you sure sip sends it after answer?
05:43.48CyonGD_:  That's all I know of, and it's still happening in 1.2.1 for me.
05:43.53_Vileon the right top, I expected that not to be included ok
05:44.03troyb;)
05:44.13_Vilechange it in the config, to say "blah"
05:44.18_Vileand *6settings it
05:44.21troybokay
05:44.23GD_no i can't get it to run.. I tried asterisk -vvvvgc and I get fatal errors.. so I thought that it might be due to my not having any drivers loaded... but loading modprobing the modules fails...
05:45.07GD_oh well... I found somewhere of the net that some of these symbols have been deprecated as of kernel 2.6.13... do you think that might be the reason? I'm using 2.6.15 at the moment..
05:45.17troybits rebooting
05:45.27troybuniversal application loader.. sounds cool :P
05:45.39_Vileheh
05:45.42_VileI may have found it
05:45.46CyonGD_:  well since you have a kernel newer than the deprecation, no.  ;-)  Unless that was a typo
05:45.55_Vilehttp://pastebin.com/495996
05:46.02_Vile#
05:46.02_Vile# Outbound Proxy info
05:46.02_Vile#
05:46.02_Vileoutbound_proxy: ""
05:46.02_Vile#
05:46.03_Vileoutbound_proxy_port: "5060"
05:46.07_Vilewell, shit
05:46.15_Vileok anyway, outbound_proxy needs to be set
05:46.21GD_no I'm using 2.6.15... according to that I need an OLDER kernel...
05:46.21troyboh really?
05:46.24CyonGD_:  And asterisk should run; test it without zaptel.conf; see if it will work when not getting any zap channels
05:46.37GD_Jan  8 08:46:56 WARNING[19556]: chan_zap.c:924 zt_open: Unable to open '/dev/zap/channel': No such file or directory
05:46.38GD_Jan  8 08:46:57 ERROR[19556]: chan_zap.c:6473 mkintf: Unable to open channel 1: No such file or directory
05:46.38troybokay it says Blah on the screen
05:46.40GD_is this normal?
05:46.54QwellGD_: you using udev?
05:46.59QwellGD_: README.udev
05:47.04GD_yeap.. gentoo does by default
05:47.06sim5imoutbound proxy dont need to be set unless you're using it
05:47.07*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
05:47.09Qwell(assuming the modules are loaded successfully)
05:47.20troyb_Vile what should i set the address too? my router?
05:47.21_Viletroy, set outbound_proxy
05:47.26troybsure ^^
05:47.31GD_what modules? zaptel and zaphfc? well they dont!!!
05:47.34_Vileno, set it to 10.0.69.100
05:47.37QwellGD_: well, why not?
05:48.00GD_because I get an unresolved symbols error after modprobing them... and they won't load
05:48.03sim5imtroyb:  also you need a port
05:48.04_Vilewhich should be your * box, or voip provider or whatever
05:48.20_Vilehe's got one
05:48.22_Vile5060
05:48.23_Vileoutbound_proxy_port: "5060"
05:48.29troyb:_
05:48.31troyb* :)
05:48.31sim5improxy1_port
05:48.42troybyup thats already set to 10.0.69.100
05:48.43_Vilebouncing sec
05:48.44sim5imor was it in the proxy1_address ip:port
05:48.52sim5imi just setup a 7960 on friday
05:48.59troybcoolness
05:49.07_Viledon't need a port on proxy1_address
05:49.10_Vilejust need outbound
05:49.15sim5imbut i dont have axs to the tftp at the moment , to look at the conf
05:49.23sim5imI never use outbound proxy
05:49.35_VileI actually have proxy_emergency, proxy_backup, and outbound_proxy used
05:49.49sim5imyeah redundancy
05:49.55troyb_Vile when i look at phone status it says error proxy emergency and proxy backup not set
05:49.56_Vilebut none the less
05:50.02troyband it still says Phone Unprovisioned
05:50.03_Viletroyb
05:50.04_Vileok
05:50.11sim5imim on a VMware emulated box right now, instaling openH323
05:50.11_Vilenow, sec
05:50.20_Vilesee ur msg's
05:50.36sim5imand gonna try run asterisk on this vmware virtual linux box
05:50.49troyb_Vile got it.. does it matter that i dont have a box with that ip assigned?
05:50.56_Vileyes
05:50.59_Vilesee the next msg
05:51.01troybsure
05:51.07troybgot it
05:51.16troybi dont have a registered nickname which is why you dont get my pm's
05:51.36_Vileahhhhhh
05:51.40_Vileyou should register
05:51.41_Vile;)
05:51.42troybsure :)
05:51.51troybit seems troyb is taken. jerk, its actually my name :P
05:52.15_Vileand vile was my original nick on EF
05:52.28_Vile*shrug*
05:52.37troyb1alright im regsiered :)
05:52.41Cyonlol
05:52.49CyonI got Cyon.  ;-)   :-P
05:53.08_VileI figured noone would ever use vile, go figure... do a whois
05:53.09_Vile;)
05:55.09CyonYeah...I've had mine over 10 years...now it's a cell phone brand..  :-(  http://en.wikipedia.org/wiki/Cyon
05:56.15brockj49464vile:  So should I be able to see it in a "sip debug ip (provider ip)" or not?
06:00.44znoGwikipedia is like hotmail, every dictionary word is taken
06:01.20justinuSIP-T isn't taken
06:01.35justinui guess that's not in any dictionary but newton's tho
06:02.57trixtereven all the words from the pr0n dictionaries?
06:03.13trixterstuff like snowball, dirty sanchez, hot karl
06:03.23trixterI am sure there are dictionaries you cna still add to wikipedia with
06:03.24justinusome of those cyon phones look pretty cool
06:03.47Cyonjustinu:  lol  That makes me feel so much better  :-P
06:04.13justinubetter than the junk we get shoved down our throats here
06:04.25Cyonlol
06:04.44CyonVery true; we think the US is so advanced; but we get hand-me-down technology in everything
06:05.05justinuthe only reason people think we're advanced is the've never been anywhere really advanced
06:05.16CyonExactly
06:05.51CyonHell, I didn't know much better, even while reading all sorts of tech news..I just assumed I'd never bothered looking for it here.  When I was in Japan, omfg.  lol
06:06.01*** join/#asterisk PJMattF (n=matt@cpc2-fare3-3-0-cust79.cos2.cable.ntl.com)
06:06.07justinuyeah, i'm headed there in a few months
06:06.11CyonSweet  :)
06:06.17CyonSpeak the language?
06:06.21justinunope
06:06.37justinuwish I could
06:06.41coppiceI loved the US ads someone did for GSM roaming phones, before the US had much GSM - "works anywhere outside the western hemisphere" weird :-)
06:06.52CyonIt isn't needed at all; but just picking up some basic words helps a lot
06:07.12justinui carry a quad band gsm phone
06:07.21justinuso wherever I go, I can just buy prepaid cards
06:07.39coppicenobody seems to have a 5 band phone yet :-)
06:07.49justinuare there 5 gsm bands yet?
06:07.56CyonHmmm, be careful; almost no phones are compatible with Japan's network, even if the technolgy could be, they just don't let it work.
06:08.01justinuoh really?
06:08.04justinuno GSM there?
06:08.13trixterjustinu: beer and coffee are almost the same so at least you can get breakfast and afternoon beverages without too much of a problem :)
06:08.16justinui gotta ask benjk about that, he'll know
06:08.24trixterjustinu: there is gsm400 that is coming out
06:08.24justinutrixter: lol
06:08.43trixterso yeah there would be 4+ that are techniclaly available although I dont know if there are any phones that do all 4
06:08.43PJMattFI'm having problems installing * 1.2.1 onto my new Ubuntu Breezy install - I've got it apparently building OK, but can't get the ztdummy module to work ("FATAL: Module ztdummy not found" when trying to modprobe it)
06:09.00Cyon"Note however, that due to different technologies used, mobile phones from your home country, including GSM phones, are likely not to work in Japan."
06:09.05Cyonhttp://www.japan-guide.com/e/e2223.html
06:09.07coppiceour local CDMA rand UMTS phones roam to Japan
06:09.09Nugget"not likely"  heh.
06:09.27justinui don't want to roam, i want to be onnet to whatever is there
06:09.29*** join/#asterisk [hC] (i=turnerd@66.199.130.40)
06:09.40justinumy phone issn't subsidy locked either
06:09.54[hC]anyone have a regex to match NXXNXXXXXX OR 1NXXNXXXXXX without having to explicitly specify both?
06:09.58Nuggetjust call your provider and rent a phone from them for while you're there.
06:10.03[hC]maybe do a [1|] or something?
06:10.08Nuggetor rent a phone from your hotel and forward your number to it.
06:10.11justinui hate my provider, they can kiss my ass
06:10.16coppiceonly a few countries do the loocking thing with GSM phones
06:10.18Corydon76-home[hC]: no such regex
06:10.20Nuggetit'll be about $100 a day and $2 a minute.
06:10.24CyonI rented...got it in the airport, and dropped it off when leaving
06:10.34justinuthey must have prepaid service in japan
06:10.42Nuggetnot for gaijin.
06:10.45justinuseriously?
06:10.52CyonNugget:  Policies have changed...now you can
06:10.55[hC]Corydon76-home: suck. :(
06:11.03Corydon76-homeOr just do 1NXXNXXXXXX,1,Goto(${EXTEN:1},1)
06:11.18Nuggetcould be, I haven't been to japan in about 10 months.
06:11.20[hC]That would work, depending... i suppose.
06:11.29trixterI want the 7.2Mbps data that tmobile is doing in germany
06:11.34[hC]Ive got to maintain a list of every NPA in our area codes
06:11.38trixterthat much bandwidth portable and wireless ...  Mmmm
06:11.50justinuyou know what they charge for that service?
06:11.52Corydon76-homeGee, wow, gratitude for fixing your problem.  Thanks.
06:11.54[hC]because my telco plays the 'you have dialed a long distance number' instead of simply rejecting the call
06:12.01CyonNugget:  I couldn't say when, but I walked into narita, asked for the cell phone rental at the information booth, anmd they pointed around the corner, 5 minutes later I had my cell
06:12.02NuggetI always just rent a phone from american express' vendor, which ain't cheap but is painless and hassle-free.
06:12.03justinunone of my german friends can afford wireless data
06:12.16Cyonjustinu:  I had it 8 days, and it was < $100 total; one second
06:12.24Nuggetnot bad at all.
06:12.36NuggetI recommend that, then.  :)
06:12.42justinuare there wifi hotspots in japan?
06:12.59Cyonhttp://www.narita-airport.jp/en/guide/service/list/svc_19.html
06:13.36CyonIt seemed to me that all their rates were close enough I just didn't care who it was.
06:13.36trixterjustinu: nope dont know, saw the PR release not any thing else
06:13.36justinuhttp://www.telecomsquare.co.jp/en/index.html
06:13.53justinuabout 7 bucks a day
06:13.58justinui guess i can deal with that
06:14.59justinu4.2 yen per second?
06:15.02CyonYeah, sounds about right
06:15.07justinuisn't that like 4 cents a second?
06:15.18justinuthey must mean minute
06:15.42NuggetI wouldn't be too sure of that.
06:15.46Cyonlol
06:15.48[hC]yen != cent, is it?
06:15.52Nuggetjapan (well, tokyo) is ridiculously expensive.
06:15.53justinuthat's fucking crazy
06:15.59CyonInternational is extremely expensive
06:16.01Nuggetit's a little less than a cent, but it's easiest to just pretend it is.
06:16.17[hC]Im worried to see what my cell bill is going to be after roaming and placing calls in costa rica for the last 12 days
06:16.17Nuggetthe place I stay for NTT Data is 28,000 yen a night, and it's nothing special.
06:16.22justinui'll be using my ip phone as much as I can
06:16.39Cyon1 USD = 114.665 JPY
06:17.02Cyon1 JPY = 0.00872106 USD
06:17.07[hC]exten => s,n,GotoIf($["${CALLERIDNUM}" = ""]?SetUnknownCID)
06:17.08[hC]exten => s,n(SetUnknownCID),Set(CALLERID(num)=Unknown)
06:17.13[hC]Eaaargh
06:17.16[hC]I hate windows
06:17.18[hC]Sorry.
06:17.33Cyonlol  We all do....I hope  *peers around*
06:17.48[hC]Im puttying in a remote office right now and hit the right mouse button by accident.
06:17.48NuggetI hate windows.  I hate Linux too.
06:17.55[hC]Amen nugget.
06:18.03CyonNugget:  Hmmm, it's possible to be cheap, just takes work.
06:18.05justinuall computer operating systems suck
06:18.11NuggetCyon: yeah
06:18.22Corydon76-home[hC]: Ugh, don't use ${CALLERIDNUM}
06:18.31Nuggetit's hard if you have work to do and barely have conversational japanese skills, though.  :)
06:18.31Corydon76-homeUse ${CALLERID(num)}
06:18.44justinuyeah, it's 2 bucks a minute
06:18.46[hC]Corydon76-home: i was actually replacing that..
06:18.50justinuincoming is free
06:18.50CyonNugget:  I found the absolute best prices were when I went to a travel stand thingy and got a room for the night on the best deal they had...they always had super deals to use if it was the same day
06:19.02justinui'll just use my * box as an intl callback
06:19.03justinufuck it
06:19.15CyonNugget:  True; but I have no japanese skills at all...Maybe they pitied me.  ;-)
06:19.17Corydon76-homeSo, do you find it difficult to conceptualize setting a function to a value?
06:19.21NuggetI can't tell you how many times I've walked into a restaurant and started to order, gotten 30 seconds into the conversation at which point they realize I barely speak japanese, and then had my menu switched out for the gaijin menu with pictures and doubled prices.  :)
06:19.23Cyonjustinu:  lol
06:19.26justinuthey're charging in 1second increments
06:19.34justinui can make it dial the digits pretty quick
06:19.45justinumake the phone dial, that is
06:19.53CyonNugget:  lol  My favorite places were where I get a ticket for food.  :-x
06:19.57Nuggetheh
06:20.23CyonNugget:  The food was good, it was cheap, and didn't need to say much at all.  lol
06:20.33CyonNugget:  omg yes.  Yuki...*sigh*
06:20.37justinuhow about the prostitutes, are they as expensive as everything else?
06:20.49Corydon76-homeHmmm, I wonder if anybody has attempted to patent international callback yet... that would be a tidy little racket...
06:21.15Cyonjustinu:  I don't know...:-P
06:21.27[hC]Corydon76-home: Yeah, the variable-functions are... not the most intuitive at first..
06:21.55justinucorydon: i could swear i remember hearing that someone did
06:21.58Corydon76-homeGood, because I'm the one who got that functionality added
06:22.13[hC]Nugget: Dell is selling the 30" Cinema HD display for $1999 now in their case. You probably paid almost double that for yours, huh? :x
06:22.36[hC]Corydon76-home: why, just to be unique? :)
06:22.44dorphalsighey, can anybody help me with this error msg?
06:22.45dorphalsig[chan_unicall.so]Jan  8 01:10:52 WARNING[6376]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging
06:22.58Cyonjustinu:  But just go to here http://www.japan-guide.com/e/e3011.html  and you'll be all set  ;-)
06:23.15Corydon76-homeNo, to be an asshole.  I wanted to make sure that it was quite possibly the most difficult system to conceptualize, so people would continue to pay me the big bucks
06:23.28CyonCoaxD
06:23.30CyonBah
06:23.56justinulol
06:23.58[hC]Corydon76-home: You must work at microsoft?
06:24.04[hC]:)
06:25.04Corydon76-homeActually, I hate seeing code that polls for a change in a variable.  I'd much rather have code that is triggered upon a value change
06:25.06dorphalsigAnybody?
06:25.18PJMattFI'm having problems getting (yes you guessed it) ztdummy going, on my new install of ubuntu breezy - it *looks* like it's building properly (I've overcome the obvious "not got ..." issues) but it won't modprobe. can anyone help?
06:25.23justinupolling is usually never a good idea
06:25.40Corydon76-homewhich is exactly what setting a function does... act as a trigger...
06:25.43*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)
06:25.51justinuyou mean a callback?
06:26.00[hC]PJMattF: i presume you did 'make install' and 'depmod -a' before trying a modprobe?
06:26.03Corydon76-homeNo, a trigger
06:26.09[hC]Corydo76: Makes sense.
06:26.12[hC]doh.
06:26.14justinusounds like a callback to me
06:26.21[hC]exten => s,n,GotoIf($["${CALLERIDNUM}" = ""]?SetUnknownCID)
06:26.21[hC]exten => s,n(SetUnknownCID),Set(CALLERID(num)=Unknown)
06:26.24[hC]son of a.
06:26.28Cyonrofl
06:26.41[hC]there. whitespace in my copy buffer now.
06:26.45Corydon76-homeIt's not a callback, because you aren't passing around a pointer to the function
06:26.50justinuok
06:27.02PJMattF[hC]: yep, still gives me "module ztdummy not found" and the matching "error running install command for ztdummy" failures.
06:27.11Qwell[hC]: umm...
06:27.19QwellWhy are you setting cidnum if cidname is empty?
06:27.21Qwell:)
06:27.32QwellYou probably meant to set cidname...
06:27.47PJMattFI picked up Asterisk fresh from svn earlier today, it appears to have done the UDEV changes (I recall doing that manually before - I rebuilt my machine today with different distro)
06:28.04dorphalsigHi, I'm trying to get R2 to work with *, I already added /usr/local/lib to ld.conf.so, but I keep getting this when trying to start *. [chan_unicall.so]Jan  8 01:10:52 WARNING[6376]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging
06:28.08[hC]Qwell: er? That checks callerid num, and sets caller id num.. to unknown.
06:28.16dorphalsigcan anybody help me?
06:29.11CyonQwell:  Yeah...It looks logically right to me, what am I missing?
06:29.18[hC]Qwell is high.
06:29.22Cyondorphalsig:  Sorry, not a clue
06:29.31[hC]Dont mind him.
06:29.37justinulol
06:29.45Cyonhaha
06:29.57[hC]have to get incoming caller id turned on on these lines yet. Was sick of seeing 'asterisk'
06:30.19justinuthe idea that you have to pay extra for caller ID is a fucking joke
06:30.48Corydon76-homeWow, you're using a module not included in Asterisk, yet asking the Asterisk folks to help.
06:31.23Corydon76-homeExcuse me, #debian, but could you help me out with a problem I've been having on my RedHat 9 system?
06:32.00PJMattFand the little light in my microwave is broken
06:32.26justinuthen openss7.org website sucks
06:32.29justinunone of the links work
06:33.08[hC]Anyone have any ideas when the Linksys SPA-942 will be released?
06:33.13[hC]I really want dual ethernet and PoE
06:33.25[hC]that would really make this 941 much better.
06:33.51Cyondual ethernet definitely PoE; don't overly care
06:33.51[hC]Heh, found a funny bug with it though, if you set a user password on the web interface, trying to use the 'redial' function on the phone asks you for the user password.
06:34.14Cyon[hC]:  Hmmm, only 941, not the 841s?
06:34.20[hC]Cyon: not the end of the world no, but it would be nice. its about the only thing its missing.
06:34.30[hC]Cyon: not sure, havent played with an 841.
06:35.01Qwell[hC]: hmm, okay
06:35.10Cyon[hC]:  Yeah, I can see nice
06:36.28Qwell[hC]: well...be consistant with your variable/function use! :P
06:36.44QwellI read that like 4 times too, kept reading it as name
06:36.48[hC]Qwell: haha. I've already switched it.
06:37.12[hC]I wonder what linksys is using to power their spa-9000's pbx functions.. I doubt they used asterisk
06:37.22jbroomelinksysone!
06:38.57SkramXI have asterisk questions galore!
06:39.20SkramXlater.
06:39.21SkramXlol
06:39.21rajivi wonder when we will see pics of the new linksys spa phones
06:40.05SkramXrajiv: which model?
06:40.49rajivspa901 spa921 spa922 spa942
06:40.55rajivhttp://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_News_C2%26cid%3D1136499819516&pagename=Linksys%2FCommon%2FVisitorWrapper
06:41.04[hC]Hmm. I guess they did use asterisk.
06:41.06[hC]Linksys has made a special version of Asterisk that integrates with the SPA-9000. The changes Linksys has made to Asterisk are getting integrated into future Asterisk releases.
06:41.17Qwell[hC]: oh?
06:41.24QwellWhat changes?
06:41.41[hC]Not sure, thats all the article says.
06:42.00[hC]Thats on the SPA9000 PBX/Router/ATA/SIP server thing they released
06:42.10[hC]looks alright for a very very basic ~5 phone small office install
06:42.18[hC]very cheap, 399
06:42.19marcus2the iax.conf that comes with 1.2.1 mentions some compile-time jitterbuffer options
06:42.26rajivbut only 1 pots port
06:42.32marcus2but i cant figure out where they would be controlled.. the makefile doesnt have anything obvious
06:42.39_Vilehc, what features?
06:43.05_Vileonly 1 pots out/inbound?
06:43.59[hC]_Vile check out the article on voxilla.com
06:44.04SkramXhey... anyone ever used asterisk to like keep track of a call, and make sure the call doesnt go over -X- ammount of minutes
06:44.11_Vilethx
06:45.17SkramX??
06:45.29[hC]SkramX: yes, lots ... One option that comes to mind is AbsoluteTimeout
06:45.36_Vileskram, yes
06:45.42_Vileskram /|\
06:46.01SkramXwell, i want to announce right before it hangs up
06:46.03*** join/#asterisk dancing_ninja (n=chatzill@61.246.9.140)
06:46.13dancing_ninjahello friends
06:46.31_Vileskram, is it AGI?
06:46.36[hC]SkramX: im sure if you search for absoultetimeout on voip-info.org you will find some relevant examples.
06:46.40SkramXI dont have anything set up yet
06:46.47dancing_ninjai was wondering if we can use asterisk in buig call center setting?
06:47.02_Vilethis is for a pre-paid type app?
06:47.14SkramX_Vile: yeah
06:48.20_Vilehttp://www.voip-info.org/wiki-Asterisk+Prepaid+Applications
06:48.28SkramX_Vile: I need to code my own
06:48.28_Vilethere is a lot of code there
06:48.30SkramXany agi scripts?
06:48.31_Vileahh
06:48.33_Vileyes
06:48.43SkramXPHP AGI?
06:48.48troyb1_Vile im surprised you are not ready to hibernate after you finished with me?
06:49.21_Vileskram, not that I know of
06:49.32_Vilebut the syntax should be easy to move from perl to php
06:49.40_Viletroy, ;)
06:50.31troyb1are you EST?
06:50.35_Vilepst
06:50.49_Vilethough, I'll be up 'til 1 or 2 tonight
06:50.51justinupst? where you at?
06:50.51_Vilepst :)
06:50.57_Vileoregon
06:51.00_Vileyou're in cali
06:51.00justinuah
06:51.02justinuyes
06:51.08troyb1only 7 am for you then?
06:51.08_Vile:)
06:51.16_Vile10:51pm
06:51.17_Vilehere
06:51.24troyb1so much for that idea
06:51.26_Vilehaha
06:51.27troyb1naw im in Toronto
06:51.34_VileI have a friend in toronto
06:51.39troyb1your Northern neighbor :)
06:51.42_Vileformer business partner
06:51.52_Vileyeah, you guys quietly come down and steal our cheese..
06:52.05troyb1haha yeah.. i just came back from Florida actually
06:52.11_Vilewhat part?
06:52.14troyb1ironically i did buy cheese today
06:52.16troyb1Naples
06:52.21_Vilenice place
06:52.32troyb1i call it god's waiting gate
06:52.40_Vilehahaha
06:52.46_VileI call it, quiet
06:53.02troyb1yeah it's slow paced.. damn good miniature golf course there ;)
06:53.46dancing_ninjai was wondering if we can use asterisk in big call center setting?
06:53.56troyb1i went to Miami for a few days but i mean it costs a lot more to stay there then naples for sure
06:54.06rajiv[hC]: i was mistaken. 2 pots ports not 1
06:54.42trixterI love my 1 litre insulated coffee mug
06:54.48trixterer my 2 litre
06:54.52justinulol
06:55.03justinuthat's a huge mug!
06:55.05dancing_ninjaam i invisible
06:55.19justinuninja's are pretty stealthy
06:56.20rajivdancing_ninja: you can use asterisk where ever you like
06:56.22troyb1damn it would suck working for time.gov if you were even 10 minutes late there would be no arguing
06:56.22trixterjustinu: yeah..  I have a couple cups of something a day
06:56.50dancing_ninjayeah they sure are ......but can some body pl guide oe provide a link where I can find links  to setup asterisk in a big call ceter (1000 seater)
06:57.01trixterbeer is nice when I am out and stuff
06:57.21justinui doubt anyone here has info on a call center that big
06:57.27dancing_ninjai need to get ademo up and convince my ceo otherwise we buy avaya :-(
06:57.30justinuthat's serious stuff, and serious money
06:57.38SkramXis there like a way to make an announcement 10 minutes into a call?
06:57.44troyb1dancing_ninja hire a professional team to do that for oyu
06:57.46troyb1*you
06:57.46SkramXwell, after a Dial() command
06:57.50justinuskramx: show application dial
06:57.55trixterSkramX: technically yes, there is app_bridge
06:57.56justinuit does all that shit
06:57.59dancing_ninjafrom where ?
06:58.02troyb1i dont know about you but i wouldnt want to unpack 1000 IP Phones
06:58.08trixteryou can bridge to something that plays an anouncement if you really really wanted
06:58.08justinutroyb1: lol
06:58.18trixteror you can whisper into it
06:58.20trixteror something
06:58.28troyb1dancing_ninja are you in the states?
06:58.34dancing_ninjai need to get a demo up ...some kind of feasibility study before that
06:58.50troyb1most corporations can provide that data to you
06:58.52_Viledancing, 1000 wouldn't be too difficult
06:59.01justinuskramx: http://www.voip-info.org/wiki-Asterisk+cmd+Dial
06:59.02troyb1_Vile i have seen bigger :)
06:59.02dancing_ninjano i am in India and mlaysia
06:59.07_Vileme too
06:59.23_VileIndia, call center for mainly us traffic?
06:59.29justinuskramx: look at open L()
06:59.33justinus/open/option/
06:59.38dancing_ninjayup US and UK
06:59.54_Vilewhat do you have in place now?
07:00.06trixterI am doing a US call center for a US company..  same state even but only 10M minutes a month pending the 600k minute test goes well
07:00.19dancing_ninjawe got cisco and avaya stuff
07:00.21trixterwhy export jobs when its so cheap to do it here?
07:00.33SkramXjustinu: i am looking
07:00.39justinueveryone hates telemarketers
07:00.42_Viletrixter, expanding the wealth, is what I've come to the conclusion of
07:00.42troyb1trixter the US economy cant support it is my guess
07:00.47SkramXso I would do Dial(who,L(x,y,z))?
07:00.59justinuyeah
07:01.00dancing_ninjaweneed to open a new center so looking for alternatives
07:01.15troyb1if your really going to try and setup a thousand cisco IP Phones then call up a few companies have meetings with them
07:01.30troyb1i have a feeling none of them would think twice about inviting you out to lunch :)
07:01.40_Viledancing, what software pops up on the screen -- or is it a simple call agent pool?
07:01.50_Vileyeah
07:02.09trixterSkramX: Hmm..  just thought of another ghetto way too.  With dial you can goto a macro on connect, now I havent  tried this but I would guess you can idle in that macro for 10 minutes then play some tone or something
07:02.10_Vileand dinner, a few brunches...
07:02.22troyb1_Vile no questions asked.
07:03.06SkramXmacro == agi?
07:03.19CyonSkramX:  No, macro == macro
07:03.30SkramXalright
07:03.33SkramXnever written a macro
07:03.34troyb1it would definetely have to be a 6 figure budget just for the phones alone
07:03.35SkramXhmm
07:03.35dancing_ninjadoes asterisk support skype ?
07:03.57justinutroyb1: i bet he wants to use softphones :P
07:03.59dancing_ninjaor only sip based stuff like vonage
07:05.47SkramXhow do i know how long a call has been in progress?
07:06.04SkramXdancing_ninja: NO
07:06.50dancing_ninjawhat abt a sip based stuff ?
07:07.12rajivsip is no problem
07:08.55dancing_ninjagrt ...so  i can use any sip based network like vonnage or wengo and and use softphones ....?
07:09.17trixterSkramX: macros are kinda like contexts in the dialplan although they are slightly different ...  they are defined within the dialplan though
07:09.18Cyondancing_ninja:  Sure
07:09.28SkramXmeh
07:09.44CyonSkramX:  They are really simple once you see an easy example
07:10.04trixterthey arent that hard..  it would be like      [macro-waitfor10]  exten => s,1,wait(600)     exten => s,2,playback(beep)
07:10.08SkramXany examples :?
07:10.35trixterfor what you want anyway..  then call it like   Macro(waitfor10)   or something..  the M option in dial is what you need to look at for your use
07:10.43trixterits totally ghetto but it should work
07:10.52SkramXwell it would need to be a variable it waits for.
07:11.02OloBolaHello. When trying to view my arguments passed from dialplan to php: $agi->verbose($argv[1]);, PHP hangs, never prints anything. I can get the first argument fine ($agi->verbose($argv[0]);) so I know my code is ok
07:11.56trixterok..   [macro-waittone]  exten => s,1,wait(${arg1})   exten => s,2,playback(beep)
07:12.21trixterthen Dial(SIP/somewhere/something,M(waittone,600))
07:12.26trixteror something
07:12.40trixterno error checking but you get  the idea
07:12.45trixteror should
07:13.16trixterthat should work with a lot of versions of asterisk regardless of extra addons
07:15.34dancing_ninjais there java based interface or api with asterisks ?
07:18.56rajivyou can call java programs from asterisk if you want. there is the AGI api which works with anything that can read from STDIN and write to STDOUT
07:19.54trixterdancing_ninja: how exactly do you mean?
07:20.02trixterthere are programs that do call management and other functions in java
07:20.20trixterthere are other java based programs that interface with java directly...  just dont know what the end goal is
07:21.22dancing_ninjai mean to say ......if there is java wrapper to asterisk api so that i call in using java program rather than jni
07:21.37trixterwhat specifically are you going toi call though
07:21.58trixtergeneraelly when people make reference to asterisk they mean the program not the libraries
07:22.11trixteryour comments make me think you want libraries basically
07:23.24dancing_ninjayeah ...i play to integrate asterisk/voip intoour crm and groupware
07:23.57dancing_ninjaand make a custom control panel and quatility dashboard
07:24.37dancing_ninjawhich hopefully will be released under lgpl eventually
07:27.17trixterthere are a few that already exist
07:27.33trixterbet if you just wanted to mod one you could use their base class files to get the functionality
07:27.41trixterbtw you appear to want the "manager api"
07:27.57trixternot "asterisk api" -- so you have better terms to look for it
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07:29.08dancing_ninjaok ...
07:29.42dancing_ninjatrixter: can u pl provide some links
07:30.47trixterwww.google.com
07:30.57justinu~docs
07:30.58jbotextra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
07:31.05trixteroffhand I dont know where a java class would be for the asterisk manager api
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07:31.23trixterjustinu: he doesnt want docs he wants a java class for the manager api
07:31.31Qwellhttp://www.voip-info.org/wiki/view/JAVA-AGI
07:31.37trixternot for agi
07:31.38trixtermanager api
07:31.46Qwelloh
07:31.49OloBolais this correct? exten => s,2,agi(test.php|'testing 1 2 3') trying to pass 'testing 123'
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07:44.05trixterI feel like a monkey, eating trail mix, nuts, raisins and M&Ms
07:44.23trixter"Red M&M, blue M&M, they all are the same color in the end" - homer simpson
07:46.14trixteris there any digium t1 card that does atm?
07:46.25trixteror do I have to use sangnoma?
07:46.32trixtercause I know they do it
07:50.54_Viletrixter, what are you looking to do?
07:51.11trixterT1 in atm mode
07:51.13trixternot channelized
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07:51.24_Vileinteresting
07:51.47_Viletmk, the digium code doesn't support this, but gimme a sec to confirm
07:51.52trixtera lot of people have done it a lot of times in  the past
07:54.19_Vilelooks like sangoma
07:54.24_Vilethough
07:54.32_Vile*asterisk developers*
07:54.38_Vilethis topic was brought up in late 2003!
07:55.09_Vilehttp://lists.digium.com/pipermail/asterisk-users/2003-September/021893.html
07:55.32_Vilelemme check code
07:56.10_Vilenuffin'
07:57.25_Viletrixter, you a coder?
07:57.58trixteronce or twice
07:58.48_Vilewell, if you decide to take the hard road and figure it out on asterisk, throw it back, seems useful
07:59.40_Viles/asterisk/digium
07:59.41trixterasterisk is gpl that makes it really hard for me
07:59.47trixterits against my religion to contribute to gpl products
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08:00.56_Vileim about out of steam, going to bed in a min
08:01.24_Vilequiet night
08:01.49SkramXoh
08:05.53_Vileok later
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08:24.09trixterHmm..  got confirmation today that with asterisk modules at least (big explanation of how they interface with the system) the modules themselves dont have to be gpl just gpl compatible which means that I might actually od something with regards to asterisk development becuase I can use a BSD lciense for my modules
08:24.20trixterwhich alieviates a lot of my problems with development for asterisk
08:27.00jahanihi
08:27.04jahaniwhat mean this error ?
08:27.04jahaniERROR[2354]: chan_sip.c:10856 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete.
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08:29.54mogormanno trixter
08:29.59mogormanone simple thing
08:30.06mogormanyou have to load a module with gpl key
08:30.16mogormanas well as libraries that are gpl
08:30.29mogormanyou can no use gpl libs in non gpl code
08:30.32mogormanperio
08:30.32mogormand
08:30.42trixteraccording to the FSF I can do that
08:30.45trixterI will take their word over yours
08:30.49mogormani know
08:30.55mogormani talked with benjk for an hour
08:30.56mogormanabout it
08:31.01mogormanit will be resolved this week
08:31.03mogormani hope
08:31.09benjkit has already been resolved
08:31.14mogormanand i imagine there will be a statement of some kind
08:31.28trixterwell regardless licensing@fsf.org staes that I can, so I will if you as a digium spokesperson want to make this a legal issue go ahead I can give you my lawyers name and address
08:31.45trixterI am doing what the FSF states I can do under section 7 of the gpl you cant impose extra terms on me
08:31.50mogormani am taking it up with fsf
08:31.57mogormani know exactly what you are implying
08:32.00benjkasking a second time, the FSF insisted that LGPL is a proper license for Asterisk modules
08:32.04trixterso if you want to violate the gpl to tell me I cant go ahead I will do it anyway, the only way to stop me is to file a lawsuit, is that your intention?
08:32.06mogormanbut it is a simple fact
08:32.12mogormangpl is a 1 way street
08:32.17trixterits a fact that I have documentation from the FSF on my side
08:32.48mogormanugh im tired of having argument again
08:33.01trixterso if you want to file a lawsuit lemme know I will forward you the name and contact information of my lawyer
08:33.04mogormanlike i told benjk
08:33.11mogormani am going to discuss this with fsf
08:33.17trixterothwrise I will side with the FSF on the issues relating to the gpl in this matter
08:33.19mogormanand disclose any contact i have
08:33.19benjkmog: why do you think legal issues are black and white and simplistic if it takes hard work and study to go through law school and take the bar exam
08:33.23mogormanif this issue is resolved
08:33.28mogormanin any way
08:33.42mogormani will make steps ness. to have lgpl and bsd keys added
08:33.49mogormanagreed
08:33.53mogormanmy brother is a lawyer
08:33.58mogormanas are both my uncles
08:34.10trixterwell anyway, I am going to release something tonight just to do it, if you want to attempt to file a restraining order to prevent me go ahead, if you want to file against me for licensing infringement go ahead I will give you the name of my lawyer
08:34.25benjkand they specialise in intellectual property with regards to software and publishing?
08:34.26mogormanas im not the co holder
08:34.52mogormanactually yes
08:34.53mogormanone is uber geek
08:34.57trixterwell if its so clear cut why does the general counsel to the open source initiative agree with me on this issue
08:35.01trixterper a 2003 linuxworld article
08:35.06trixteror was it linux journal?
08:35.08mogormani know trixter
08:35.10trixterone of those two
08:35.13tzafrir_laptopbasically you can not use gpl-conflicting modules: res_crypto (openssl) and chan_[o]h323
08:35.14mogormanim not going to make desc.
08:35.17mogormani will let fsf
08:35.17benjkand they told you that Asterisk modules cannot be LGPL licensed?
08:35.18mogormando it
08:35.22trixterhowever the FSF is on my side in this so feel free
08:35.29tzafrir_laptopOh, and codec_g729a
08:35.29mogormanbut i want a formal statement
08:35.32mogormannot you telling me
08:35.40benjkso how do you explain that the FSF licensing department states otherwise
08:35.47mogormanthat too tzafrir_lapto
08:35.51trixterI will follow the FSF directions in one other matter
08:35.53tzafrir_laptopAnybody wants to rewrite res_crypto with gnutls?
08:35.56mogormani am going to fsf
08:35.59trixterYes, taking into account all the information you have provided,
08:35.59trixterthe LGPL is an acceptable license for the modules. Please refer the
08:35.59trixterDigium licensing team to:
08:35.59trixter<PROTECTED>
08:36.01mogormanlike i said
08:36.03benjkthe licensing specialists of the organisation that released the GPL and LPGL licenses
08:36.05trixterunless of course you arent part of hte licensing team
08:36.13mogormani know
08:36.14trixterthat is from zak, licensing@fsf.org
08:36.19mogormani realize
08:36.31tzafrir_laptopwhat's the FSF here for?
08:36.33mogormanyou guys obviously want it cleared up
08:36.39mogormanthats what i want to have happen
08:36.52mogormanbut right now its obviously a  grey area with us in a dispute
08:36.53trixteremail zak, its quite simple
08:36.56trixteranyone can actually do that
08:36.58mogormancan you please let me get a formal answer
08:36.59trixterand get the same information
08:37.02mogormanand make the changes ness.
08:37.05benjkas far as I am concerned the FSF has cleared it up for us
08:37.15trixterI will be releasing something, not quite sure what yet, but something tonight
08:37.17mogormanthen why dont i have said email
08:37.18benjkbut you did insist that we were doing something wrong
08:37.22mogormanwhy isnt it public anywhere
08:37.33mogormani want to do this in public arena
08:37.37trixterif you want my lawyer info to serve with a restrianing order to prevent me from licensing with a compatible but not gpl license lemme know
08:38.14tzafrir_laptopmogorman, again, is there anything that isn't clear? you can't use real-time mysql together with dundi, as things stand now
08:38.14trixterthat email is from tonight btw, so as of about 1 hour or so ago zak was answering emails
08:38.17trixterhe may still be
08:38.33mogormanthats true tzanger
08:38.37mogormanerr tzafrir_laptop
08:38.43mogormanunless they issue us an exception
08:39.06tzafrir_laptopthey won't. Unless mysql decide to change their license
08:39.08benjkwhether MySQL's policies are in line with the GPL is an entirely different question
08:39.22tzafrir_laptopOr unless somebody bothers rewirting res_crypto
08:39.25mogormanthey can issue an exception
08:39.28benjkbut since that question doesn't concern me, I am happy not to spend my time on it
08:39.30mogormanwithout changing license
08:39.47mogormanas co holder you can do anything you want
08:40.17benjkno you can't
08:40.24mogormanactually you can
08:40.33mogormanif you are copy right holder of a work
08:40.35tzafrir_laptopAs for me, I'd be hapier to have something that is stricktly GPL, so I won't have any problem using other GPL libs
08:40.40mogormanyou can do anything you want with said work
08:40.53benjkfor example, if you release code under the GPL, you cannot tell people that they cannot use or redistribute that GPL licensed code in ways that the GPL allows
08:40.53trixterso digium owns the copyright on the netbsd code in asterisk?
08:41.00trixterwhat about the libc code that is part of it too?
08:41.22trixterbenjk: per section 7 of the gpl no extra restrictions :)
08:41.27benjkwhat you license under the GPL grants recipients the rights the GPL grants
08:41.33trixterwhich includes removing the ability to use gpl compatible licenses
08:41.40benjkyou cannot take those rights away with EULAs etc
08:41.54Qwellexception != restriction
08:41.57tzafrir_laptopbenjk, well?
08:42.06mogormanits like religious debate at this point
08:42.09benjkand MySQL AB are quite likely overstepping this line
08:42.17tzafrir_laptopbenjk, mysqlab sell you mysql under a different license
08:42.30tzafrir_laptopthat is, not you specifically, I figure
08:42.38benjkbut as I said, I don't care to spend my time on that issue because I don't use MySQL
08:42.40trixtermogorman: I dont see it as religious, the FSF has spoken digium is infringing on the gpl rights of the people that get the gpl version
08:42.48trixterits quite simple, or at least a digium representative is trying to
08:42.54mogorman?
08:43.03mogorman?
08:43.11benjktzafrir: I am talking about GPL MySQL
08:43.16mogormanone i dont rep. digium in this argument
08:43.36trixteroh so saying I cant is a moot point then
08:43.37mogormanand two i never in my time at digium have seen any debate between us and fsf if anything we are good friends
08:43.40trixterat least that is cleared up
08:43.46tzafrir_laptoptrixter, where has the FSF spoken so, could you please give me a link?
08:43.47mogormani disagree
08:44.04Qwelltrixter: I too would like to see the communication back and forth between them regarding this
08:44.05benjkaccording to MySQL AB's terms they claim several restrictions on what you can do with GPL MySQL which go beyond what the GPL says
08:44.08mogormanwhich is why when i speak with fsf i will disclose any info i recieve
08:44.09trixtertzafrir_laptop: I copied in the email, and even gave zak's email address (licensing@fsf.org) if anyone wanted independant confirmation
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08:44.20trixterthey suggested that digium licensing team read their faq and pointed to a specific entry
08:44.41benjkand the GPL says that for code you release under GPL terms you must not impose any such additional restrictions
08:44.43Qwellbenjk: With MySQL, using it with a NON GPL product, WOULD VIOLATE the GPL.  MySQL is doing "The Right Thing" by selling it under a diff license
08:44.50trixterzak was there 1.5 hours ago, prolly still is (I just checked the timestamp)
08:45.10benjkconsequently, I consider MySQL AB in violation of the GPL in respect of their GPL-MySQL release
08:45.20trixterusing gpl code with a commercial product isnt necessarily violating the license, becuase of how mysql defines that
08:45.23Qwellbenjk: If you're at ETel, I'll GLADLY talk to my contact at MySQL with you
08:45.37trixtermysql states that if you store data in their database with a commercial app or was handled by a commercial app you have to get a commercial license
08:45.51Qwellwe can let him clear it up.  You *KNOW* they have lawyers, and you *KNOW* they've talked with the FSF on multiple occasions.
08:45.53trixterthey FURTHER state that if you use mysql commercially you have to get a commercial license, even if the software itself isnt commercial
08:46.00benjkespecially the claus that says that you cannot make your customer to install MySQL with an installer for example
08:46.01Qwelltrixter: no, they don't. :)
08:46.13tzafrir_laptoptrixter, is this in reference to plugins from asterisk-addons?
08:46.31trixternow this is over mysql
08:46.35trixtergetting a link hold on
08:47.20OloBolahow do I go about sending arguments to an agi script? I've seen four different ways so far, can't get any of them to work. exten => s,2,agi(test.php,10001) Trying to send 10001 to test.php
08:47.25benjkno it is a general assessment of how MySQL AB claim restrictions for the GPL release of their software which go beyond the restrictions of the GPL itself
08:47.44Qwellbenjk: come to ETel
08:47.55Qwellbenjk: I'll let you talk to a MySQL rep in person.  I'll even introduce you.
08:48.25tzafrir_laptopThat said, practically any file in the source tree calims to be GPL. And actually don't refer to a modified license or any excptions.
08:48.31benjkI don't have the resources to fight this through, but I do have the right to state that there are such additional restrictions
08:48.42benjkconsequently I don't use MySQL
08:48.48X-Robtrixter, the important bit is 'if the program you're using it with isn't GPL'ed'
08:48.56QwellX-Rob: exactly
08:49.02Qwellwhich I've stated repeatedly
08:49.03benjkso I don't risk to release an installer which MySQL then take issue with
08:49.09X-Robif the program you're using it with _IS_ GPL'ed then it's fine and no-one cares.
08:49.38X-Robwups
08:49.38mogormanX-Rob!
08:49.42X-Rob*eek*
08:49.44mogormani havent seen you in ages
08:49.49X-RobI've been busy.
08:49.51X-Rob8-\
08:49.56mogormanheh
08:49.59mogormanhow you been
08:50.05benjkthose companies have the right to try an be as restrictive as they can or as they can get away with
08:50.18benjkand I have the right to avoid their products as a result
08:50.34benjkand to recommend to all my customers never to touch those products
08:51.10X-Robhttp://aussievoip.com.au/salute1.wmv
08:51.26mogorman?
08:51.30X-Rob8-(
08:51.31benjkPostgreSQL is available as an alternative, I can use that without even having to worry about what may or what may not upset MySQL AB's legal folks
08:51.53X-RobFuner.
08:52.32X-RobFus all hell coz my PC is  day.
08:52.36X-RobGah. I'm laggy as all hell coz my PC is rendering
08:52.41X-Robyay it's finished.
08:52.49mogormanX-Rob, hows the amp side of force?
08:53.05X-Robmogorman, I've been busy as crap recently.
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08:58.17X-RobHurm. Anyone know anything about quicktime, and the best codec to use whilst encoding?
08:58.27X-Rob(I realise this is a totally inappropriate channel for this questions)
08:58.36mogormanfor what kinda video
08:58.37benjkQuicktime uses a variety of codecs
08:58.42mogormanwith quicktime you can use h264
08:58.47mogormanproccessor intensive
08:58.49mogormanbut tiny as hell
08:58.55benjkthe latest version even includes H.264 now
08:59.11benjkbut that runs only on G5 IIRC
08:59.12X-Robmogorman, to re-render that link I posted above. It's crappy for a 5mb file.
08:59.30mogormanhrm?
08:59.52elcuco"when you are looking out the window for rain, put your glases on... otherwise you will not see the rain."
09:00.28mogormani see
09:01.33OloBolahow do I go about sending arguments to an agi script? I've seen four different ways so far, can't get any of them to work. exten => s,2,agi(test.php,10001) Trying to send 10001 to test.php
09:02.12OloBolaI tried this to: exten => s,2,agi,test.php|10001|10001|10001
09:02.19OloBolatoo
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09:02.52OloBolaI can get the first argument, not the second.
09:03.04OloBolausing agiphp
09:05.51OloBolafrom php: $agi->verbose($argv[1]);
09:06.03OloBolait just hangs
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09:09.08jebbaanyone using a utstarcom f1000? :)
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09:46.43Qwellbed time...I think
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10:18.25GordoHi Guys... I need Help!!! Big Time.... I have been at this for days....
10:18.44GordoI can make calls to test extensions @ the * server, but not between phones
10:18.51GordoIts killing me!!
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10:47.02trixtermy bsd licensed asterisk app that does really nothing at all..   http://www.trxtel.com/app_bsdapp.c -- basically written becuase I was told I couldnt
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10:47.53MT`AwAyhello~
10:48.39MT`AwAygot a little problem : when someone uses a service where I use WaitExten to get the code they type, I get the last key they entered in the CDR records as DST field
10:49.42MT`AwAyanyone has an idea ?
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11:09.51GordoAny chance of some help from the * guru's?
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11:41.39tzafrir_laptopGordo, try detailing your problem. Maybe one of the non-gurus lurking here could help
11:42.59tzafrir_laptopPlease explain exactly what does work, and what doesn't. Also state version of Asterisk
11:44.49GordoI have a fresh installation of * on a gentoo box (sparc), I have an Aastra 480i sip phone and a couple of soft phones... All my phones are registered, and they can call test extensions with recorded messages
11:44.57Gordobut hte phones cannot call each other
11:45.13iDunnohave you got extensions set up for the phones?
11:45.18Gordoyeah
11:45.32iDunnowhat happens when you dial the extension?
11:45.52GordoI can see that the call is setup in *, but then it puches me straight to voicemail
11:46.04Gordoretrans_pkt: Maximum retries exceeded on call
11:46.22GordoNo one is available to answer at this time
11:46.35iDunnophones are SIP?
11:46.38Gordoyep
11:46.46iDunnoasterisk can talk out on port 5060 to the phones?
11:47.09iDunnophones on the same lan segment as the asterisk box?
11:47.19iDunnofirewall anywhere in the middle?
11:47.34Gordophones are on a different subnet, but only a routehr in hte middle
11:47.40Gordono firewall or ACL's
11:48.05iDunnotry putting one of them on the same subnet.
11:48.14iDunnosee if that then works.
11:48.20Gordook, wont take long...
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11:53.52aksisif I have a static ip, do I need to use FWD iax setup?
11:54.17aksisim having a little trouble grasping the big picture of voip
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11:55.18GordoiDunno: Ok, changed ip of my * box and configured sip phones... but exactely the same problem
11:55.39GordoWhat gets me is that I can call a test extension and hear a recorded message fine
11:58.02iDunnosuggests that the phones can talk to * but * can't talk to the phones to me, so a networking or router problem.
11:58.06tzafrir_laptopGordo, seems like it's time for you to pastebin extensions.conf , sip.conf and a trace from the CLI in verbose mode
11:58.33Gordook.. all phones are on the same VLAN/ subnet now
11:58.36Gordopasting now
11:58.47tzafrir_laptop~pastebin
11:58.49jbotwell, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
12:00.54Gordook, the two conf files @ http://pastebin.com/496229
12:01.45Gordocli trace @ http://pastebin.com/496230
12:02.17rabelaisdo I have to compile in MeetMe support?
12:05.51*** join/#asterisk Gr1NcHeuX (n=devine@AStDenis-105-1-22-116.w81-248.abo.wanadoo.fr)
12:06.06Gr1NcHeuXsalut ici et bo nene a tous ^^
12:06.11Gr1NcHeuXune kestion
12:06.50Gordoany luck guys?
12:07.28Gr1NcHeuXkkun sait  quoi ca correspond ?
12:07.31Gr1NcHeuX-rw-r--r--   1 root   root         163 2006-01-02 11:23 mamie-a-la-plage.avi
12:07.46Gr1NcHeuX-rw-r--r--   2 root   root         163 2006-01-02 11:23 mamie-au-resto.avi
12:08.02Gr1NcHeuXla diff entre le 1 et le 2 ?
12:09.09*** join/#asterisk Oryn (i=oryn@falcore.fsck.tv)
12:09.24*** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no)
12:09.55rabelaisGr1NcHeuX: I'm not sure what you're asking, but is seems like you're asking if there is a difference between the two files
12:10.15Orynanyone know why my music-on-hold seems to be playing very very slowly? (sounds like a special effect from a star trek movie)
12:10.38rabelaisGr1NcHeuX: there doesn't seem to be one, but you can compare using diff, enter: diff mamie-a-la-plage.avi mamie-au-resto.avi           if there is any output, then they're not the same
12:11.03Orynbye the way it does work great for scaring away sales reps :P
12:11.14Gr1NcHeuXno it between 1 and 2
12:11.22Gr1NcHeuXwhat's the differcne ?
12:11.40Gr1NcHeuXerf sorry for my por english i'm a french guy :)
12:12.48Mothergreetings
12:13.12*** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se)
12:13.13MotherI have a couple of boxes running HEAD from march 2005, and want to roll them up to 1.2.1, any particular caveats?
12:13.29MotherI remember reading someplace that the extensions format has been changed a lot
12:13.41*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
12:16.13rabelaisGr1NcHeuX: I told you, there doesn't look like there's a difference, if you want to be sure, you can use the "diff" program to check the file content and it will report any differences between the two files
12:16.46rabelaisbut when I say there doesn't "appear" to be any differences, all that means is that the files have the same size and were modified at the same time
12:16.56rabelaisthat's no guarantee that the files have the some content
12:16.59Gr1NcHeuXok rabelais thx
12:17.06GordoiDunno: What would prevent * from talking to the phones if hte phones can talk to *
12:17.53iDunnofirewall, blocked UDP ports, something like that.
12:18.24iDunnoGordo: I'd put * and a phone on the same switch.
12:18.43GordoThey are
12:18.50GordoSame Subnet, same VLAN, same switch
12:19.50MT`AwAyGr1NcHeuX> tu peux aussi utiliser md5sun
12:19.51MT`AwAyerf
12:19.53MT`AwAymd5sum
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12:20.29Gr1NcHeuXdeja fais MT`AwAy le probleme c ke les fichiers sont differents en taille
12:20.40Gr1NcHeuXdonc le md5sum differe
12:21.08MT`AwAybah donc les fichiers sont differents
12:21.21Gr1NcHeuX1 root root
12:21.25Gr1NcHeuX2 root root
12:21.29*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)
12:21.44Gr1NcHeuXen dehors de leur taille c le chiffre ki me pose un soucis
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12:22.07MT`AwAyy'aurais pas un repertoire dans le lot ?
12:22.17Gr1NcHeuXnop c deux videos
12:22.52MT`AwAyaussi loin que je me souvienne le numero juste apres les permissions correspond au nombre de fichiers visibles dans le repertoire
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12:23.25MT`AwAyou bien
12:23.59mrtwisterhello. i currently use oh323, but there i cannot set fast start and h245 options per gateway (peer). cna i do it in other channels (h323 and ooh323 )?
12:24.02Gordotzarfir: Any luck with those configs?
12:26.40Gr1NcHeuXMT`AwAy, c aussi c que g savais mais y a un soucis dans le sens ou ce sont des videos donc un seul fichier logiquement
12:27.22MT`AwAybah oui
12:27.41MT`AwAysinon il me semble que pour les fichiers ca peut egalement representer le nombre de 'hard link'
12:28.05MT`AwAy-rw-r--r--  1 magicaltux magicaltux 0 2006-01-08 14:27 test
12:28.11MT`AwAyet apres un hardlink :
12:28.12MT`AwAy-rw-r--r--  2 magicaltux magicaltux 0 2006-01-08 14:27 test
12:28.13MT`AwAy-rw-r--r--  2 magicaltux magicaltux 0 2006-01-08 14:27 test2
12:28.36MT`AwAyen gros t'as une autre mami au resto sur ton disque
12:28.43elcuco... i think my irc client is broken...
12:28.46Gr1NcHeuXok cool
12:29.02Gr1NcHeuXg'avais pige merci pour ca :)
12:29.13MT`AwAypas de soucis~
12:29.19MT`AwAysinon je vois pas trop le rapport avec asterisk
12:29.22GordoBRB guys
12:29.23Gr1NcHeuXc sympa 2 mamie au resto +tot k'une :p
12:29.36Gr1NcHeuXMT`AwAy, g me suis juste plante de chan pour demander
12:29.44MT`AwAy'kay
12:29.52Gr1NcHeuXet g un asterisk chez moi c pour ca ke chui la de tps zen tps
12:30.01MT`AwAyok
12:30.11MT`AwAyj'utilise aussi asterisk, mais pour le public ;)
12:30.18MT`AwAy(petit serv public en france, pour des francais)
12:31.34Gr1NcHeuXta vue ke chez un fournisseur de telephonie ils ont une offre a 0.01e la min de portable en france ?
12:32.39Gr1NcHeuXcf budget-telecom
12:32.56Gr1NcHeuXstu mets ca sur ton serv tu vas gagner des sous ;)
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12:34.20GordoIdunno: you have me thinking that the windows firewall is blocking my softphone from working actually...
12:34.38GordoI have one desktopn sip phon and two softphones for testing atm
12:34.39trixterdamn firewalls
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12:34.45iDunnooh, that's always possible. the windows firewall is a pain
12:34.47trixterthe world would be an easier place without em
12:34.53Gordohrm yeah... and as if it would be as simple as turning it off
12:35.05Gordonoooo, group policy prohibits it
12:35.31trixterdoesnt the windows firewall like to let programs open up ports?  like fairly easily?  I know that microsoft messenger uses upnp to disable firewall settings on any MS inspired network firewall (like their access points)
12:35.51MT`AwAyGr1NcHeuX> je pense pas qu'ils autorisent la revente de leur offre
12:35.55Gordohah, really
12:35.59Gordothat would be right
12:36.24Gr1NcHeuXben t'es pas oblige de la revendre hein
12:36.38trixterwe have one here and there is no way to disable that via any of the standard tools (obviously you can rewrite the firmware but ...)
12:37.10trixterso users can trump the admin, and its basically soap or soap style at least data so its easy to manipulate to open ports and whatnot..  very bad mojo
12:37.29p0g0__trixter:  use un pnp from here to nail the messenger http://www.grc.com/freepopular.htm
12:37.41MT`AwAyGr1NcHeuX> dans ce cas je vois pas trop comment je vais gagner des sous~
12:37.56p0g0__gibson has been around for 30 years & is very competant
12:37.57Gr1NcHeuXMT`AwAy, sur l'offre globale
12:38.05trixterp0g0__: its not a big problem, just one that would have been nice if it were docuented
12:38.10*** join/#asterisk backblue (n=moo@87-196-6-89.net.novis.pt)
12:38.27trixterand right now nothing is even connected to that AP its just for testing
12:39.39MT`AwAyGr1NcHeuX> sur mon offre globale j'apelle deja plus de 30 pays gratuitement (pour les fixes, et les mobiles pour 3 de ces pays)
12:40.33Gr1NcHeuXgratuitement ?
12:40.38Gr1NcHeuXsans abonnement ?
12:41.09MT`AwAyGr1NcHeuX> juste un petit abonnement de 25 euro/mois
12:41.30Gr1NcHeuXok abo valable pour des comm entreprises ?
12:41.40MT`AwAyvi
12:41.47Gr1NcHeuXcool chez ki ?
12:41.50MT`AwAyenfin pour les entreprises c'est 30 euro au lieu de 25
12:41.54MT`AwAypar contre pas de revente
12:42.04Gr1NcHeuXouep du direct
12:42.10Gr1NcHeuXc chez ki ?
12:42.29MT`AwAybroadvoice
12:42.52MT`AwAyje teste le service actuellement~ ca marche pas mal
12:43.16*** join/#asterisk svenna_ (n=svenna@p548D1EE2.dip0.t-ipconnect.de)
12:43.43Gr1NcHeuXg deja essaye chez eux mais la ou g suis g trop de ping avec eux
12:43.57MT`AwAyok
12:44.05GordoBesides 5060 what ports should I be allowing on hte windows firewall
12:44.14trixteryour rtp ports
12:44.24tzafrir_laptopbasically, all of the RTP ports, see rtp.conf
12:44.25trixterrtp.conf - default is 10000-20000
12:44.44trixterudp
12:44.55Gordothere is no way to open a port range on XP firewall is there...?
12:45.03trixterdont use it dont know
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12:46.44GordoThanks for all your help guys, but I need sleep....
12:47.24Gr1NcHeuXGordo, the better solution is to put a debian on ure machine ;)
12:47.43trixterI say no firewall no service packs
12:47.46trixterthat will fix it right up!
12:48.38*** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it)
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13:20.11OloBolatrixter: should I be able to grab these as well:  exten => s,2,agi(dtmf.php,10001|11569|friday|1|22|2006)  $argv[2], $argv[3] etc?
13:20.54*** join/#asterisk Guest3456 (n=root@200-233-134-246.xd-dynamic.ctbcnetsuper.com.br)
13:23.45*** join/#asterisk Skymarshal (n=Skymarsc@p54AF67BB.dip.t-dialin.net)
13:24.06OloBolaI got it.
13:24.21SkymarshalHi,
13:24.23Skymarshaldoes anybody know what the difference between VM_CALLERID and VM_CIDNUM is?
13:25.08blitzrageSkymarshal: CALLERID has both the name and number, CIDNUM is just the number
13:25.24Skymarshalaaaaah!!!! Thanks! :-)
13:25.35blitzragetry using them in a NoOp() statement next time to see the actual output of what they are
13:26.01blitzrageor check your asterisk source code under the ./doc/ directory in README.variables <----- your friend!!!
13:26.02zoahey ho blitzrage
13:26.08blitzragezoa: ho hey!
13:26.14Guest3456i found a bug in res_config_mysql.c, when the sip user unregister the file res_config_mysql.c put on mysql a lot of garbage on the tables sip_buddies in the fields port and ipaddr...anyone know how to fix it?
13:26.40blitzragezoa: do you know of that application that lets you share the screen between multiple computers like a KVM kinda thing, but its software for Linux/Windows? Its on sourceforge, but I can't seem to remember the name of it
13:26.49zoavnc
13:26.55blitzragezoa: nope, thats not it :)
13:27.10*** join/#asterisk BladeRunner05 (n=gianni@adsl-24-214.38-151.net24.it)
13:27.12zoayou mean like one big screen ?
13:27.12blitzragezoa: it lets you use a single KB/mouse combo and move across screens
13:27.26blitzragezoa: yah, kinda. Its like a KVM, but you run the software on all the computers
13:27.27tzangerblitzrage: x2vnc
13:27.29tzangerx2x
13:27.34blitzragetzanger: don't think thats it either
13:27.39p0g0__blitzrage: tightvnc, realvnc , etc
13:27.40tzangerblitzrage: that's what I use at work
13:27.44blitzrageits not a VNC :)
13:27.52tzangertightvnc or RDP if I need to actually see a remote screen on my laptop
13:28.06Skymarshalbye guys
13:28.09tzangerbut at work I have a 20" monitor beside the laptop so I use x2vnc so I can just use my laptop mouse/kb on the PC
13:28.10blitzragewhen I find it I'll tell you what it is because its awesome :)
13:28.11zoahmm, blitzrage: i didnt know that existed, so if you find it, let me know
13:28.12zoa:)
13:28.22blitzragezoa: yah, its vicked c00
13:29.46dpryoblitzrage: synergy lets you share keyboard and mouse, but not display
13:29.56blitzragedpryo: thats the one I wanted :)
13:30.04blitzragedpryo: it lets you move the mouse/KB across multiple screens
13:30.12dpryo:)
13:30.29tzangerdpryo: yeah that's what I use x2vnc and x2x for
13:30.31dpryoI'm using it between my linux, mac and windows
13:30.49blitzragedpryo: yah, I have 3 comps here at the school and odn't want to move my hands to another KB/mouse
13:31.00tzangeryeah that's exactly what these programs do
13:31.03blitzrageand I'd used that software before and it works great
13:31.23blitzragevnc is a bitch since you have to control the remote computer on the same screen
13:31.29Guest3456i found a bug in res_config_mysql.c, when the sip user unregister the file res_config_mysql.c put on mysql a lot of garbage on the tables sip_buddies in the fields port and ipaddr...somebody know how to fix it?
13:31.31blitzrageI want the same kb/mouse across multiple screens
13:31.32tzangerblitzrage: huh?
13:31.38blitzrageGuest3456: we heard you the first time
13:31.52blitzrageGuest3456: and we still don't know
13:31.54zoablitzrage: tell me when you found it
13:32.04tzangerblitzrage: my laptop runs linux.  I run x2vnc -east compiler.ca.benshaw.com:0.
13:32.16tzangerthen I move my linux mouse over to the right of hte screen and now the kb/mouse runs the windows machine beside it
13:32.17blitzragezoa: http://synergy2.sourceforge.net/
13:32.42tzangeryou could run a second one -west or -north or whatever and get a second/third/fourth system in the fray
13:32.59blitzragetzanger: using all 3 monitors?
13:33.11tzangerblitzrage: YES
13:33.12blitzragetzanger: I still picture VNC as using remote computers on the same screen
13:33.15blitzragetzanger: I don't believe you
13:33.20blitzrage:D
13:33.23tzangerblitzrage: that is what it typically does, but x2vnc does not
13:33.28blitzrageoh I see
13:33.34blitzrageyou must be lieing though
13:34.05tzangersynergy does the exact same thing, but why run two daemons?  I already have VNC on every machine so I can hit them remotely... x2vnc does it without placing the screen
13:34.23tzangerx2x does the same thing but connects to another X server, not a VNC one
13:34.32tzangerthere's likely a win2vnc too IIRC
13:34.36tzangerI know there is in fact
13:35.40dpryotzanger: Can one move xclients from one server to another?
13:35.58dpryoThat would rock.
13:36.11dpryoxinerama between computers :)
13:36.14tzangerdpryo: not that I'm aware of.  There's no way to "split" the ownership of an app across two X servers
13:36.26dpryox.org-people are working on it though.
13:36.27tzangeryou'd need a third x server and have it actually handle it :-)
13:36.35dpryoYeah, some proxy.
13:37.03tzangerthat has caught me a few times.. >I'm so used to x2vnc I sometimes take an app and try to move it ot the monitor  to get it off the laptop and hten I wonder for a moment as to why it didn't owrk :-)
13:37.10dpryohehe
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13:40.23tzangerheh
13:40.35tzangersynergy can't handle the three-finger salute but it owrks great on x2vnc :-)
13:40.54tzangerit does have some nice features, scrolllock locking would be handy
13:42.35fugitivowhat's synergy?
13:42.37fugitivo(morning)
13:42.57tzangersynergy is the coming together of disparate ideas or concepts in unrelated fields
13:43.15blitzragesmart ass
13:43.25tzangerno, I have a sweet ass
13:43.29fugitivonot at this time
13:43.38blitzrageI guess I'll try this x2vnc then I guess since tzanger thinks its so great and he is all knowing and such...
13:44.01tzangerblitzrage: your apology and beer will be accepted at the next torastricon meeting
13:44.08blitzragetzanger: :D
13:44.29tzangerI need to get cracking on this telebridge thing, jimvanm will be upset if I don't have something wild to show him
13:44.41*** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
13:44.44blitzragetzanger: yah... I'm working on a redundency solution with DUNDi today
13:44.51tzanger:-)
13:44.58blitzragetzanger: I don't know anything about your telebridge app :)
13:45.00fugitivodid you try the "Desktop sharing" feature of kde?
13:45.02fugitivoit's great
13:45.05tzangerblitzrage: I know you don't
13:45.09tzangerit's top secret shit
13:45.18blitzragetzanger: good thing we're talking about it in this public forum :D
13:45.22tzangeryep
13:45.28tzangergotta create a buzz
13:45.38blitzragefugitivo: I prefer gnome now (I used to use KDE -- its sloooooow)
13:45.44*** join/#asterisk _4d4m_ (n=adam@44-8-101-159.adsl.legend.co.uk)
13:45.46fugitivoslow?
13:45.48tzangerblitzrage: you're on crack
13:45.52fugitivowith todays cpu? nahh
13:45.53blitzrageits slower than gnome I've found
13:45.53tzangeror using Debian
13:46.03fugitivolol
13:46.11*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
13:46.13fugitivomy sister uses gnome
13:46.14blitzrage...and don't even get my started on debian
13:46.22blitzragefugitivo: smart girl. She hot?
13:46.28tzangerhahahahha
13:46.34fugitivoshe's not smart
13:46.43fugitivoi don't know if she's hot, it's my sister
13:46.44blitzragesmarter than you in regards to Xwindows :D
13:46.55blitzragefugitivo: oh come on... I'm sure you know whether she's attractive or not
13:47.03fugitivono, i don't
13:47.14tzangerkde's integration is what keeps me there.  Gnome is messy and "big" and don't even get me started on the build environment and dependency hell there
13:47.23fugitivoi second that
13:47.36fugitivoif you install kde you don't need anything else
13:47.37iDunnoyou're all broken, ion3 is the true way forwards, it stays out of the way, it's light, and it shows you just how many apps are completely fucking broken ;)
13:47.50fugitivoi don't care if it's light
13:47.54fugitivoi have a lot of cpu and ram
13:48.02tzangernah, I used to be a big windowmaker fan
13:48.07tzangerstill am, just can't do without the integration
13:48.13blitzragetzanger: know anyone who needs 2x1GB Registered ECC CL2.5 RAM?
13:48.14fugitivoi have 64bits and 1gb ram
13:48.17tzangerdcop and ioslaves rock your world
13:48.24tzangerblitzrage: I have some of that myself
13:48.31fugitivowhy should i use a light desktop? :)
13:48.36blitzragetzanger: I have some that I'm trying to sell <-- got it for $100.
13:48.43tzangerheh
13:48.50blitzragetzanger: doesn't work in my AMD machine :)
13:48.59tzangerfirst rule of sales: never reveal your margins
13:49.07*** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net)
13:49.10blitzragetzanger: sales suck
13:49.20fugitivosales keep us alive
13:49.20blitzragetzanger: I think I might just eBay it
13:49.37blitzrageI just let all the crooks deal with sales
13:52.09*** join/#asterisk Xandor (n=astar@ANantes-154-1-4-15.w81-53.abo.wanadoo.fr)
13:52.14*** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se)
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13:52.56Xandorhello
13:53.47Xandori want to do a shortcut ex : on my soft phone i type 5555 & it call the number 0243435458
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13:54.41blitzrageXandor: exten => 5555,1,DoSomething()
13:54.58fugitivoi love postgresql
13:55.02blitzragefugitivo: me too
13:55.09tzangerpostgres rox yer sox
13:55.22blitzrageand pgAdmin makes it bareable to use :)
13:55.27fugitivoyes
13:55.29tzangerhahah
13:55.31tzangeryes pgadmin is nice
13:55.40tzangeranyway gotta get the kids to their mom and spend the day with alina.  :_0
13:55.40benjkyeah, psgql is the way to go
13:55.41tzangerer :-)
13:55.42tzangerlater
13:55.45Xandorexten => 5555,1,Dial(0243435458) i tried this but it doesnt work
13:55.48blitzragetzanger: peas
13:55.58blitzrageXandor: you have no technology
13:55.59fugitivoso those mysql lovers can kiss my ass when they say that postgresql is difficult to use
13:56.08blitzrageXandor: how does Asterisk know how to connect the call?
13:56.09benjktry Dial(ZAP/1/number,...
13:56.11tzangerfugitivo: amen; I always found mysql more difficult to install and use
13:56.16Xandorexten => 5555,1,Dial(SIP/0243435458) i tried this but it doesnt work
13:56.17fugitivoi agree
13:56.17Xandoridem
13:56.18*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
13:56.23benjkor Dial(SIP/number@yourprovider, ...
13:56.33blitzrageXandor: you still don't have the right format
13:56.35fugitivoXandor: you want to use SIP or ZAP?
13:56.41XandorSIP
13:56.51benjkSIP/number@provider
13:56.57fugitivoyeah, read what benjk said
13:56.58blitzrageDial(SIP/name_in_square_brackets_in_sip.conf/number_to_dial)
13:57.07MT`AwAyoh btw anyone knows if there's a way to use meetme without loading a module ? Currently my kernel does not have support for loadable modules
13:57.10blitzrageor number @ provider :)
13:57.26fugitivoMT`AwAy: you need zaptel or ztdummy
13:57.26blitzrageMT`AwAy: asterisk loads the app_meetme.so module, not the kernel.
13:57.33blitzrageand that  :)
13:57.38MT`AwAyyup
13:57.41MT`AwAymeetme requires a timer
13:57.51MT`AwAybut there's only timers as kernel modules
13:57.56blitzrageyep
13:57.56fugitivoyes
13:57.57benjkapp_conference works without timers
13:58.01MT`AwAyand I can't load a module on my current kernel
13:58.07fugitivothen recompile
13:58.08MT`AwAyapp_conference ?
13:58.14benjkyep
13:58.19MT`AwAyfugitivo> I can't reboot this server at all
13:58.33blitzragebenjk: didn't know that didn't need a timer. How does it work then? (just curious)
13:58.49tzafrir_laptopyou don't need to boot to install zaptel
13:58.51blitzrageMT`AwAy: sounds like a server that shouldn't be running Asterisk then :)
13:58.55MT`AwAybenjk> I'll look that on google ;)
13:58.59_4d4m_hi all. i have an * server on my home lan, connectivity provided via adsl. my router forwards all packets from my public IP to my * box. i connect to FWD via iax. internal phones are connected to * using SIP. I have one way audio when recieving calls from FWD numbers (phone rings and i hear them, they dont hear me).  i've tried all the nat options and been through the docs but nothing seems to work. can anyone help?
13:59.33benjkapp_conference has less features than app_meetme
13:59.34blitzragealright, I'm disconnecting from screen now so I can get some work done. Lates!
13:59.34tzafrir_laptopMT`AwAy, why can't you load modules?
13:59.44*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
13:59.53fugitivo_4d4m_: too long for this day and hour
14:00.26_4d4m_fugitivo: heh.. you and me both
14:00.54_4d4m_been messing with it and searching docs for the last few hours
14:01.10*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
14:01.14MT`AwAytzafrir_laptop> the kernel does not have support for loading modules, and I can't reboot on a new kernel for now
14:01.37fugitivoMT`AwAy: sometimes you need to sacrifice something to get another thing :)
14:01.54tzafrir_laptopanybody here with bristuff 0.3.0 installed?
14:02.03MT`AwAyfugitivo> yup, but I'll first try app_conference
14:02.39Xandor[14:54:50] <blitzrage> Dial(SIP/name_in_square_brackets_in_sip.conf/number_to_dial)
14:02.46Xandorthanks its works
14:03.15Xandordidn't find this on google
14:03.41tzafrir_laptopLooking at asterisk.patch there I notitced it seems to change the version of the manager interface from 1.0 to 1.2 . Is  that right?
14:04.13elcucotzafrir_laptop, yes, that right. you really found that. :)
14:05.57tzafrir_laptopI would be bothered if noone else had
14:07.51*** join/#asterisk javar (n=javar@69.79.51.8)
14:08.11*** join/#asterisk shawn (n=welles@219.145.57.45)
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14:09.56johan_hammy*poke*
14:10.20*** join/#asterisk p0g0 (n=p0g0@mrtc-mm-590160.mis.net)
14:10.44johan_hammyCan anyone here assist me with a zaptel make install error?
14:15.00tzafrir_laptopjohan_hammy, you posted to asterisk-users?
14:15.50tzafrir_laptopnote the '[ -f ]' . That is a syntax error, I believe
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14:15.54johan_hammy*nods*
14:16.05tzafrir_laptopWhat version of asterisk?
14:16.22johan_hammyI'm in digest mode (I already get hundreds of e-mails a day), so I wouldn't see any replies for a few hours.
14:16.25johan_hammy1.2.1
14:16.54SkymarshalHi, if I use
14:16.56Skymarshalmailbox = password,name,email,operator=yes
14:16.57Skymarshalhow does the system know who is operator? Where is that defined?
14:19.24johan_hammyI do see a "/bin/sh: -c: line 0: `if [ -n "" ]; then  if [ -f  ]; then mv -f  .bak ; fi;"...  in the previous command executed.
14:19.32Simon-Skymarshal:   -= Info about application 'VoiceMail' =-
14:19.42Simon-Skymarshal: If the caller presses '0' (zero) during the prompt, the call jumps to extension 'o' in the current context.
14:19.58tzafrir_laptopjohan_hammy, MODCONF is not defined...
14:20.10SkymarshalSimon: Thanks! :-)
14:20.45tzafrir_laptopjohan_hammy, what distro do you use?
14:21.20johan_hammyCentOS 4.2 is the host OS, but I'm running OpenVZ for some virtual servers.
14:21.37tzafrir_laptopwhat kernel version?
14:21.55johan_hammy2.6.8-022stab050.1
14:23.00tzafrir_laptopdo you have ther /etc/modprobe.conf and/or /etc/modprobe.d ?
14:23.14*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
14:24.30tzafrir_laptopanyway, the workaround is to rem-out the line in the install: target that checks for $(MODCONF) .
14:24.42tzafrir_laptopYou don't really need that file.
14:24.49johan_hammyWhen I locate modprobe, I get /sbin/modprobe.  WHen I check the host OS, I see some other modprobe stuff.
14:25.24johan_hammyokay.  THanks.  Maybe what I'll do is talk to OpenVZ about modprobe, but if they're no help, I'll just comment it out.
14:25.25tzafrir_laptopSimply make sure you run ztcfgsometime  before asterisk is started and after the modules are loaded
14:26.05tzafrir_laptopAnyway, please report a bug in the mantis. It's a buggy part in the makefile
14:26.55johan_hammyThanks much
14:27.44johan_hammyI've only done a couple Asterisk systems.  THis is my first 1.2.1 build and I decided to go at it on a Virtual Server.  I've had quite a few bumps along the way.
14:27.47Xandorsome one knwo disa?
14:29.32Xandorwhen i call ma asterisk & i recall another number if i stop the call , it continues ringing on the other side
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14:47.20dorphalsigHi, I've recompiled asterisk with R2 Support, but when I try to load it, I get the following error  [chan_unicall.so]Jan  8 09:34:24 WARNING[18213]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging
14:47.20dorphalsigJan  8 09:34:24 WARNING[18213]: loader.c:554 load_modules: Loading module chan_unicall.so failed!
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14:51.10dorphalsiganybody=
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15:03.04SkymarshalI am writing a documentation about the voicemailsystem of asterisk. Is there a plan of the voicemail menu? Where can I find it?
15:03.47rob0~docs
15:03.50jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
15:04.12rob0particularly http://www.voip-info.org/wiki-Asterisk is best IMO
15:04.32SkymarshalBut there is no plan of the menu.
15:04.56rob0hmmm, Asterisk-cmd-Voicemail or something like that?
15:05.54svenna_hi all
15:06.08svenna_hey, voicemail is my problem at the moment :-)
15:06.14svenna_im a bid stuck here
15:06.23svenna_i got it working
15:06.47ruud_orgSkymarshal: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
15:07.15Skymarshalruud_org: Thanks! :-)
15:07.38svenna_but, when i add new busy message for a user and dial in, i get to here the users message AND the default one.
15:07.54svenna_where can i configure to nit here the default one?
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15:15.16*** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)
15:16.31dorphalsigHey
15:16.42dorphalsigthis is the 4th time I recompile libunicall
15:16.58dorphalsigand I still get WARNING[2183]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging
15:16.59*** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)
15:17.13dorphalsiganybody knows anything about R2 support?
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15:34.37ManxPowerdorphalsig, that sort of message usually means a mismatch between the version of the module and the version of Astersk
15:35.02ManxPowerAre you SURE that Unicall has been ported to the version of Asterisk you are using?
15:35.41*** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)
15:39.12ManxPowerMy internet connection sucks.  I can either listen to streaming audio OR do something else that uses the internet, but not both.
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15:47.46dorphalsigManxPower --> No, it hasnt, but ppl have installed it without problems ...
15:48.33dorphalsigI've had reports... of it working w/o problems...
15:49.40svenna_how can i keep asterisk from playing: Playing 'vm-intro' (language 'en') and make it playing 'de'-files which are stored under /de?
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15:51.50ManxPowersvenna_, "show application setlanguage"
15:52.20svenna_do i have to type that in asterisk console?
15:52.28svenna_sorry, im a noob...
15:53.32rob0svenna_: yes
15:53.44svenna_ok i see
15:53.46svenna_thx
15:55.58svenna_but where do i SET the language? in extension.conf?
15:56.57inv_Arpsvenna_: you got to start looking at  www.voip-info.com
15:57.26svenna_ok, i found it
15:57.31svenna_sorry 4 that!
15:58.05ManxPowersvenna_, you can set the language in several ways, as part of the extension in extensions.conf or for the device in zapata.conf, sip.conf, etc
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15:58.35svenna_ah, ok thanks a lot!
15:58.57svenna_i will look four it in capi.conf in my case
15:59.25ManxPowerBTW, does anyone here know of 1u or 2u cases that can take a standard ATX motherboard and support at least 1 PCI card (with maybe a vert to horiz riser card)?
16:00.17ManxPowerI'm looking at colocating at a local ISP.  My current case is like 4U since until now it's always been in racks that are not charged by space used.
16:01.42*** join/#asterisk kiwnix (n=egarcia@229.red-82-158-154.user.auna.net)
16:02.21*** join/#asterisk puzzled (n=yeahrigh@53533C13.cable.casema.nl)
16:05.39svenna_and it works :-)
16:12.39_Sam--a 2u one shouldnt be hard to find
16:13.05_Sam--the 1u will probably be hard to find a standard ATX compatible
16:14.31SparkzI have a couple of http://www.chieftec.de/?page=products_big&id=95&k_id=3&language=uk
16:14.41SparkzWorks well
16:16.15Motherhas anyone compiled chan_bluetooth for 1.2.1?
16:16.46*** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)
16:16.51MotherI've been trying various fixes found googling around but nothing cures a bunch of undeclared's
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16:31.19ManxPowerSparkz, thanks
16:31.39ManxPowerSparkz, I figured 2u would be the smallest I could get that would work with a standard motherboard
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16:32.51MotherManxPower: do you know if chan_bluetooth can be compiled into 1.2.1?
16:33.02ManxPowerMother, I do not know.
16:33.07MotherOK thanks
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16:37.04PakiPenguinhello everyone
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16:56.45*** part/#asterisk Naturalblue (n=Kay@195.26.12.229)
17:07.39*** join/#asterisk cyberd0g (n=blah@ip70-177-41-173.br.br.cox.net)
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17:10.34*** join/#asterisk acehunky (n=chat_jok@59.184.5.167)
17:10.45acehunkyhi ppl
17:10.46mjmacnot too much in the way of SIP-based soft videophones, eh?
17:12.04*** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net)
17:12.08acehunkyhi, i am looking for some document which explains what is context, dial plan, extension etc .. i have been surfing asteriskguru and voip-info .. but its kinda confusing
17:12.58benjk~docs
17:12.59jbotfrom memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
17:13.15*** join/#asterisk Assid (n=assid@203.115.64.5)
17:13.33*** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)
17:13.50ManxPower~mailinglist
17:13.52jbotfrom memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html
17:14.02Skymarshalacehunky: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
17:16.44*** join/#asterisk Tili (i=Tili@202-133-65-187-dialup.sat.net.pk)
17:17.32acehunkyumm lemme see the urls .. thanks jbot, Skymarshal
17:18.33*** join/#asterisk roulduke_ (i=612j9l3n@p508D42CF.dip0.t-ipconnect.de)
17:19.24acehunkyaah and one more thing that i wanted to ask ... how can i simulate a call via cli ?
17:19.28acehunkyis there any way
17:19.29acehunky?
17:22.21benjkyou drop a call file into /var/spool/asterisk/outgoing
17:22.36benjkbtw, 'jbot' is a bot
17:23.40benjkif you do ~docs or ~mailinglists etc etc, it will spit out some info it has been fed before
17:23.53wunderkinwhy when i do a svn checkout of asterisk-1.2 it says revision 7866 but show version says 7847M?
17:24.09wunderkindo they go off of different things?
17:24.35Corydon76-home7847 was the latest change to that branch... 7866 is the latest revision to the tree
17:25.01wunderkinok...
17:25.16acehunkyoh thanks benjk ... good to know that .. do you know of any hacks with this bot ?
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17:27.48*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
17:27.57wunderkinso i probably gave the wrong revision on my bug.. oh well
17:31.00*** join/#asterisk Seggy (i=rbutler@tsss.org)
17:31.47benjkhacks?
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17:51.57lrizzoping... any autoconf guru around here ?
17:52.50svenna_what does trunkmsg realy mean? or: "MSD digits ti strip" what are MSD digits?
17:53.20tzafrir_laptopacehunky, I use ! to run a script that generates a call file...
17:54.21swm_!kill fags
17:54.24swm_file dies..
17:54.35filehow rude.
17:54.39bkw__swm_, you're rude
17:54.55swm_LOL
17:55.05Corydon76-homeFunny, he doesn't look like a cigarette
17:55.18Corydon76-homehow do you kill a cigarette, anyway?
17:56.07*** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net)
17:56.09theblueHi all.
17:56.28theblueI'm trying to connect to FWD using asterisk, and I'm wondering how to use an outbound proxy, if its possible.
17:57.40*** join/#asterisk Skymarshal (n=Skymarsc@p54AF67BB.dip.t-dialin.net)
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17:59.12swm_bkw_ is a child molestor
18:01.19Corydon76-homeI'd strongly advise you to retract that
18:02.39rob0Maybe swm_ is Fred Phelps? :)
18:03.27Corydon76-homeIt's still slander
18:04.06rob0what brought all this on anyway? (/me scrolls up)
18:04.23rob0hmmm
18:04.30jbroomeslander is spoken
18:04.35jbroomelibel is written
18:04.37thebluelibel is written.
18:04.38thebluejinx.
18:04.42jbroome:)
18:04.50dogtaniannope
18:04.53*** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com)
18:04.55theblueBut does anyone know how I can use an outbound proxy with SIP?
18:04.57dogtanianliable is published
18:05.05sivanahehe
18:05.12jbroometheblue: on topic questions will not be tolerated
18:05.24thebluejbroome: OH MYE.
18:05.29rob0file is a Fine Human Being (for a canuck kid, that is ;) )
18:05.31jbroome:)
18:05.39xachenhe is?
18:05.46rob0definitely
18:05.52rob0right file?
18:05.53xachena Candian, eh?
18:06.05xachenCanadian*
18:06.09xachengasp
18:06.13xachenNew Brunswick
18:06.17xachenbloody NBers :P
18:06.30rob0At least he's not a Newfie.
18:06.32fileyup, all my fault
18:06.36sivanaheh
18:08.07xachenNewfies make me laugh
18:08.10xachen<< Albertan here
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18:09.05svenna_is there a way to skipp vm-intro in voicemail?
18:09.09*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
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18:10.45xachenactually New Brunswick people make me laugh
18:10.51xachenthey all seem funny for some reason
18:10.55xachengoofy is the word actually :)
18:10.56alexhopperWe are
18:11.02alexhopperCompletely
18:11.43alexhopperAsk file about how I run band practices, he'll tell you the definition of goofy :p
18:12.09fileor the definition of failure :P
18:15.52swm_yawn
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18:24.27cnet2hi, i just bought a polycom 501, but i wanted to update the firmware.  I'm not sure how it's done. I installed tftp server on my machine, and put the boot and sip files for the firmware on the root dir.  How do I access the phone, or how do I make the phone go to my boot server?
18:28.59thebluecnet2: Usually there's a way to force a TFTP update on your phone.
18:29.09cnet2got it.. jeje
18:29.13thebluecnet2: Do you have your tftp server running and ready to update at the moment?
18:29.28*** join/#asterisk brookshire[home] (i=matt@69.247.184.46)
18:29.37cnet2yes.. i the thing is i hadn't seen the tftp option on the phone.. lol :S,
18:29.42theblueOh.
18:36.15CoaxDfuck. why do these stupid people crack a useful app, only to fucking put a virus in it when they release?
18:37.05rob0for fun and profit?
18:38.14CoaxDrob0: I spose. But. c'mon. why not just release the crack?
18:42.35*** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com)
18:45.05tzafrir_laptopWhy do other _____ people use software from an untrusted source?
18:45.54dogtanianto make the rest of us look clever
18:48.12tzafrir_laptopAs I wrote to -users: now that 1.2.2 has been released, I can safely announce my 1.2.1 debs :-(
18:49.42tzafrir_laptophmm, why did I think 1.2.2 was released?
18:49.54*** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com)
18:50.27theblueCan anyone tell me how to make it so that after I dial "393" and then any string of numbers, it only dials that string of numbers (of any length) to SIP/fwd?
18:51.24wunderkinfrom the dev topic
18:51.48benjkdial(SIP/${EXTEN:3}@fwd ...
18:52.05theblueBut what extension do I do?
18:52.15theblue_393.?
18:52.19benjk_393X.
18:52.25fndudeHi all. I just installed asterisk. I am trying to test it, but currently I have no voip provider. I would like to connect via my local network/xp client. From what I gather I should have port 4569 open. Is this a tcp/ip port?
18:52.32theblueBut that first number could also be a *.
18:52.55benjk_393. will do as well
18:52.56thebluefndude: No, UDP.
18:53.10thebluebenjk: Ok, then why in hell does X-Lite keep giving me a 404 not found error?
18:53.33theblue(Actually, I'm doing this in IAX2)
18:54.18fndudetheblue: ok very good. Thank you.
18:54.23thebluefndude: No prob.
18:56.46*** join/#asterisk SwK_ (n=SwK@12-219-151-128.client.mchsi.com)
18:57.32*** join/#asterisk matt_ (n=matt@cpe-66-108-191-40.nyc.res.rr.com)
18:59.34benjksip debug is your friend
19:00.32theblueD'oh.
19:01.12theblueWhat I'm doing could be entirely done in SIP, you think SER would be easier to use for this/
19:02.54theblueNo, it wouldn't, apparently.
19:03.03*** join/#asterisk jannic (n=jan@potemkin.hitnet.RWTH-Aachen.DE)
19:06.55*** join/#asterisk ast_freak|Laptop (n=jesse@12.28.106.2)
19:08.56dorphalsighey
19:09.19dorphalsigcan anybody help me build unicall? I've been trying since this morning
19:09.30dorphalsigand all I get is a warning telling me spandsp.h is unusable
19:10.51*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)
19:12.57*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
19:15.09cnet2anyone had any problems with the digium  TDM2400?, my xorcom asterisk installation doesn't seem to recognize it.  any special drivers i should install?
19:16.20*** join/#asterisk zotz (n=zotz@24.231.47.175)
19:19.13*** join/#asterisk T42X (n=T42X@193.219.62.88)
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19:27.52brookshire[home]cnet2: what version of zaptel are you running?
19:27.54brookshire[home]do you know?
19:27.55T42Xmy asterisk is behind a nat, forwarded to outside. everyone can connect and call, but there are some problems though. if someone from outside calls someone who's inside the network - he only hears his voice, but the one who's inside cannot hear the callers voice. and when both callers are from the inside network - none of them can hear each other
19:28.00cnet2nop..
19:28.04cnet2let me find out
19:28.56*** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
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19:38.20*** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
19:40.34fndudeIs the correct way to build your architecture under the extensions.conf file?
19:41.11T42Xarchitecture?
19:41.24fndudeThe way your calls are recived, extensions, etc.
19:42.30rob0dialplan is probably the term you want
19:45.37mjmacanyone else noticing a problem with VPC origination today?
19:46.04filebrookshire[home]: !!!!!!!!!!!!
19:49.56brookshire[home]josh!!!!!!
19:50.06fileomg omg omg Mattttttttttt
19:50.09*** join/#asterisk O3OBBITHA (n=O3OBBITH@213.150.185.18)
19:50.20filebrookshire[home]: fyi, you're H-O-T
19:50.28brookshire[home]oh?
19:50.39filea pixie told me so
19:50.42fileso it MUST be true
19:51.00brookshire[home]heh..
19:51.11*** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
19:51.34*** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
19:51.52filebrookshire[home]: how are you feeling?
19:52.03brookshire[home]much better!
19:52.08brookshire[home]my mom is sick now :/
19:52.12file:(
19:52.22brookshire[home]everyone around me is getting sick
19:52.39brookshire[home]i called it black death
19:52.40brookshire[home]hah
19:53.26*** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net)
19:53.46fileatleast I'm not getting sick
19:53.53fileif so I would be very unhappy with you
19:54.48brookshire[home]i would be very unhappy with me too
19:55.12brookshire[home]i have a memory leak in java
19:55.19brookshire[home]i didn't think that was possible
19:55.31brookshire[home]but apparently it is!
19:55.48filethat's because you're silly
19:56.05brookshire[home]i have no idea where to find it either
19:56.06brookshire[home]lol
19:56.13*** join/#asterisk brif8 (n=The_Bear@lazyjtrainingcenter.com)
19:56.19fileturn around three times, and say "memory leak be gone!"
19:56.24brookshire[home]i feel like doing some simple loop tests to see if it is just java
19:56.37brookshire[home]no clicking the heals?
19:57.03filenope
19:57.15filethe system has been upgraded.
19:58.02brif8curious question monitor() records the conversation to a sound file to be played back later. But is it possible to have phone 1 making a converation and phones 2&3 listening in. NO not NDA spying. But for training purposes to help train new help desk staff. We handle a medium helpdesk and constantly have a staff change over. I would like to use actual phone calls as part of the training ex.
19:58.45*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
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20:00.38*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
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20:06.12*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
20:07.35*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)
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20:17.34*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167048221.nb.aliant.net)
20:18.25ruud_orgis there any way to pass voice mail status (mwi) from one asterisk box to another (let's say you have sip phones registered to box A, but their voicemail lives on box B)
20:19.21rob0hey brif8 that reminds me, I never figured out my MixMonitor answer. What do you have to specify for sample rate, size and encoding for the resulting .raw file?
20:19.43*** join/#asterisk john867530 (n=john8675@12-218-49-92.client.mchsi.com)
20:20.48john867530I am trying to get zapras to work I installed and patch pppd and when I try to connect I get "LCP: timeout sending Config-Requests" also when I dial the number direct it does not make modem tones
20:31.33*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
20:42.25*** join/#asterisk Defraz_ (n=t0tal@72.24.220.144)
20:42.49_4d4m_hi all.. asterisk is refusing to accept calls from a user (over iax), claiming there is no context available to handle the call. the error message states it is looking for a context in the format s@<my-context-label>
20:42.59*** join/#asterisk anon-troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
20:43.15_4d4m_and whatever i change the context label to, the error message reflects it.. any ideas?
20:51.53*** join/#asterisk Defraz_ (n=t0tal@72.24.220.144)
20:56.24*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
20:57.30De_Mon_4d4m_ the context needs to exist in extensions.conf
20:58.17*** join/#asterisk dorphalsig (n=dorphals@200.106.223.5)
20:58.20dorphalsigHi
20:58.31De_Mondorphalsig sup
20:58.52dorphalsigAnybody has any experience building spandsp?
20:59.03dorphalsigI build spandsp
20:59.10dorphalsigand then I try to build libsupertone
20:59.12_4d4m_De_Mon: the context does.. it is listed in iax.conf and extensions.conf as [contextname], but whatever it is, the call is refused saying s@[contextname] does not exist
20:59.45dorphalsigand I get a funky message saying "spandsp.h is there but can not be compiled, make sure you have no missing headers bla bla bla"
21:00.06dorphalsigbut I get no warnings or anything building spandsp
21:00.11dorphalsigand I need that to add r2 support
21:00.56De_Monhrm
21:01.41dorphalsigchecking spandsp.h usability... no
21:01.41dorphalsigchecking spandsp.h presence... yes
21:01.54dorphalsigconfigure: WARNING: spandsp.h: present but cannot be compiled
21:01.54dorphalsigconfigure: WARNING: spandsp.h:     check for missing prerequisite headers?
21:02.32*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
21:02.53dorphalsigideas?
21:02.57dorphalsigsuggestions?
21:03.03dorphalsigmiraculous solutions?
21:04.31*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
21:05.29dorphalsigok I'll settle for a not so miraculous solutions
21:05.33dorphalsigas long as it works :P
21:05.54QbYI've got a queue, with agents that aren't actually on the phone system..  (They are in another office we connect via pstn)  the Dial command has the t option so that they can transfer back in..  However, today for some reason, its saying     -- Playing 'pbx-transfer' (language 'en')
21:05.54QbY<PROTECTED>
21:05.54QbY<PROTECTED>
21:06.42QbYits like it doesn't even listen to the entire extension (ie. 303)
21:16.54*** join/#asterisk _mistral (i=mistral@jstevenson.plus.com)
21:23.06*** join/#asterisk shacha (n=shacha@32.179.8.67.cfl.res.rr.com)
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21:27.05*** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
21:27.27mrtwistertesting sip under DMZ.. can someone call me?
21:28.03dogtanianwhoch country?
21:28.10dogtanian*i
21:28.55mrtwisterdogtanian: over sip
21:29.02mrtwisterdogtanian: not intl call
21:29.17dogtanianah
21:29.26dogtaniani only do iax to external :)
21:29.40mrtwisteri see
21:29.41dogtanianfiddles about with sip for ages but it ended up pissing me off
21:40.33Blankmandogtanian: what are you using for your ata's?
21:41.00Blankmandogtanian: I do the same ... SIP is only for internal networks that are wired :-)
21:41.44BlankmanQwell, you on?
21:41.48QwellBlankman: hey, I am
21:41.52Qwellnever got an email from you
21:42.02BlankmanQwell: sorry I forgot to send you the stuff :)
21:42.14Qwellnp, was wondering what happened
21:42.34BlankmanI will get it to you later ... you around in a few hours?
21:42.37*** join/#asterisk aksis (i=aksis@republicofarizona.net)
21:42.41QwellBlankman: probably, yeah
21:42.45BlankmanI have some ? to ask you about festival if your game?
21:43.06SERGEUSHow can i get UNIQUEID inside my app?
21:43.07QwellI've done festival twice...it only went well once
21:43.12QwellSERGEUS: cdr?
21:43.19Qwell${CDR(uniqueid)}
21:43.38SERGEUSchar *uniqueid;
21:43.38SERGEUS<PROTECTED>
21:43.39SERGEUS?
21:44.14Qwellno
21:44.19QwellI don't think that'll work
21:44.29QwellYou'll want to pull it from the CDR
21:44.35SERGEUSyes :) that doesn't work - why?
21:44.48Qwellbecause uniqueid isn't a variable
21:45.11SERGEUSi thought it is ... http://www.voip-info.org/wiki-Asterisk+variables
21:45.42*** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net)
21:47.21Qwellokay, so it is
21:48.22SERGEUSi'll try ast_cdr_getvar(chan->cdr, "uniqueid",&tmp, workspace,sizeof(workspace), 0); now
21:50.50*** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net)
21:54.31*** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
21:55.36SERGEUSQwell, thank you very much
21:56.28*** join/#asterisk miguellinux (n=miguel@64.76.202.18)
21:58.40SERGEUSBTW, simple dump showing that there are only 4 variables defined for my SIP channel (SIPCALLID, SIPUSERAGENT, SIPDOMAIN, SIPURI) - and what about all other Variables described in WIKI? I'm doing something wrong, or eiki is outdated?
22:00.11QwellSERGEUS: for the most updated variables, look in doc/README.variables
22:00.31*** part/#asterisk shacha (n=shacha@32.179.8.67.cfl.res.rr.com)
22:00.45blitzrageQwell: !
22:00.49Qwellblitzrage: !
22:00.51SERGEUSQwell, thanks :)
22:00.57blitzrageQwell: how goes?
22:00.59Qwellblitzrage: Shame you aren't going to ETel...
22:01.03*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
22:01.08blitzrageQwell: I might actually
22:01.12Qwelloh yeah?
22:01.17blitzrageQwell: its still a possibility
22:01.22QwellWhat about the one in FL?
22:01.29blitzrageQwell: hey, you can program in C right?
22:01.33Qwellyeah
22:01.36blitzrageQwell: yah, don't think that's happening anymore
22:01.49blitzragefile: there you are :)
22:01.51Qwellahh, sucks
22:02.04Qwellblitzrage: well, if you're at ETel...umm...you can buy the beer this time! :D
22:02.28blitzrageanyways... I'm looking for someone who could maybe make it so that when you have multiple regexten's for a peer, it adds them all, and not just the last one
22:02.32blitzrageQwell: no problem
22:02.41fileblitzrage: oh... that wouldn't be too hard
22:03.00blitzragefile: yah, I figured as much. Should simply be a loop checking for all regexten's
22:03.02Qwellyeah, could probably just copy the code from codecs or contexts
22:03.05blitzragelike... a while loop :)
22:03.17filecodecs are different...
22:03.21blitzrageI need to go and look at C code one of these days and be able to implement this small stuff myself :)
22:03.44Qwellfile: well, the adding to the flag part, yeah, but it still accepts multiple "allow=" lines
22:03.46blitzrageanyways. With it I will be able to dynamically find a peer and the DID associated with that peer (a peer may have multiple DIDs)
22:04.08blitzragewas surprised that wasn't how it worked anyways :)
22:04.21*** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au)
22:04.31fileah crap I don't have my certificate on this box
22:04.47blitzrageQwell: you going to Etel?
22:05.15Qwellblitzrage: very very likely
22:05.16blitzrageQwell: Damin and KK and I are trying to hook up a hotel room so we can go
22:05.29blitzrageQwell: I'll be in K.C. the week of Jan 16th anyways
22:05.30Qwellgonna share a room with JunK-Y if he goes
22:05.34QwellKC?
22:05.39blitzrageQwell: Kansas City
22:05.43Qwellright
22:05.46blitzrageQwell: nice! he's cool
22:05.50QwellWhat's KC have to do with SF?
22:06.05blitzrageQwell: Astricon training the week of Jan. 16th
22:06.09Qwellahh
22:06.14QwellSo, not in Canadialand
22:06.18blitzrageno sir
22:06.22QwellI see
22:06.37QwellIf I can get off work, I might go to Spring VON too
22:06.44blitzrageVON... ewwww ;)
22:06.51Qwellfile is making me
22:06.52Qwellheh
22:06.54blitzragelol
22:07.22fileI'm cheating
22:07.39fileI'm flying out in March for a job interview, and cheating and doing a cheap US flight to San Jose :P
22:07.50Qwells/cheating/commiting a felony/
22:07.51Qwell:D
22:07.55blitzragenice
22:08.13QwellCan anybody hit battle.net? ;/
22:08.15*** join/#asterisk ys76 (i=ys76@linoa.etherkiller.de)
22:08.15fileto quote the company, "we don't care when you leave here, as long as your destination is your home"
22:08.18QwellWas wondering why wine wouldn't let me update, heh
22:08.29Qwellfile: VON = home
22:08.36Qwellhome is where your friends are
22:08.36Qwellheh
22:08.36blitzrageanyways... I'm outta here for a bit. I'll look up someone here if I can't figure out my regexten stuff myself and maybe throw some cash around
22:08.40fileQwell: haha
22:08.44fileblitzrage: kk
22:08.55*** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
22:08.59blitzragefile / Qwell: peas in the middle east!
22:09.09filepeace!
22:12.39*** join/#asterisk Jestre (n=ack@dargo.trilug.org)
22:12.45*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
22:12.59JestreAnyone having trouble with voicepulse connect today?
22:13.31mogormanhey file
22:13.35mogormanhows it going
22:13.40filepeachy, you?
22:13.43mogormanlots of bugs on the bug tracker....
22:20.07*** join/#asterisk harryk (n=me@harry.org.ua)
22:20.46*** join/#asterisk zebedee1 (n=zebedee1@allthebrooks.plus.com)
22:25.14fugitivowhy i need to discuss search engine optimization if I can have google?
22:25.34*** join/#asterisk SugarGuest142 (n=SugarGue@BSN-95-217-164.dial-up.dsl.siol.net)
22:26.34*** join/#asterisk aksis (i=aksis@idea-anvil.net)
22:28.06SugarGuest142hi! i'm new to asterisk
22:28.16fugitivocongratulations
22:28.19SugarGuest142:)
22:28.31*** join/#asterisk BladeRunner05 (n=gianni@adsl-146-214.38-151.net24.it)
22:28.49SugarGuest142and I am looking for a install guide - aka For dummies - does it exist?
22:29.00fugitivo~docs
22:29.01jbotwell, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
22:29.05*** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com)
22:29.08fugitivocheck the handbook
22:29.12*** join/#asterisk a1fa|64 (n=a1fa@24.144.51.82)
22:29.14a1fa|64i am so 31337
22:29.21a1fa|64teach me how to haXXX :p
22:29.31a1fa|64god man! some kids tried to prank call me
22:29.34SugarGuest142could you tell me where to start?
22:29.42a1fa|64i had a blast sanding them with my asterix pbx :P
22:29.44fugitivoSugarGuest142: http://www.digium.com/handbook-draft.pdf
22:30.18warthawgis there a handy guide to explain how to config my bt-101 so i can use it with asterisk to talk to my iax provider?
22:30.55dorphalsigSugarGuest142 --> You speak spanish¿
22:31.01*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
22:31.30mogormanasterix?!?! asterisk.....
22:31.39mogormanthe best place is
22:31.42a1fa|64hahaha :P
22:31.43mogorman~thebook
22:31.45jbotrumour has it, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
22:31.46a1fa|64i need to setup a trap for them
22:31.55a1fa|64so they get lost in the world of IVR menus
22:32.13warthawgyo no soy marinero
22:32.15fugitivoalfa|64: there's something on the wiki for telemarketers
22:32.17aksishi, i have asterisk setup and running with an iax connect to fwd. I am failing to grasp how to setup the exts and the sip so that I can be reached at sip:aksis@idea-anvil.net and call out with kphone. Here is my config http://www.idea-anvil.net/~aksis/
22:32.25a1fa|64fugitivo: awesome
22:32.25a1fa|64hehee
22:33.01Qwellmogorman: umm...I need help
22:33.08Qwell...I forgot what I was going to emerge. :(
22:33.16mogormanemerge -e world
22:33.23mogormanwill keep you busy for a bit
22:33.25Qwellk
22:34.06mogormanqwell go kill some bugs
22:34.22aksisI have no problem dialing from the asterisk cli to 500 and interacting with FWD...
22:34.39Qwellwow that's a big list of packages
22:34.53aksisI am not quite clear on the relation between my pbx and FWD... do I even need FWD?
22:35.03mogormanno
22:35.17aksisMother: is that no to me?
22:35.29aksiserrr mogor: is that no to me?
22:35.29warthawgthere are several basic things i am not clear on, this telelphony business, doodle erlang erlang
22:36.04a1fa|64omfg
22:36.06a1fa|64this is awesome
22:36.07a1fa|64:p
22:36.19a1fa|64talk about prank calling somebody
22:36.19a1fa|64dial-out
22:36.20a1fa|64and force them to use the menu
22:36.22a1fa|64this is awesome
22:36.29mogormanalfa?!
22:37.09rob0haha
22:37.48a1fa|64this sucks
22:37.52a1fa|64i cant play the .gsm in winampo
22:38.32mogormaninstall sox
22:38.39a1fa|64sox?
22:38.53a1fa|64sox@sourceforge?
22:39.10*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
22:39.17*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
22:39.48dorphalsigthere is a winamp gsm plugin
22:39.50mogormani dont know where sox is stored
22:40.32warthawgin the top drawer, your shorts go in the drawer below
22:41.45a1fa|64omg
22:41.48a1fa|64this guy i sannoying
22:41.59a1fa|64http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture
22:43.03*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
22:43.49a1fa|64this is great
22:43.54a1fa|64i am making those people very mad
22:44.15mogormanbye
22:44.31*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
22:44.31[TK]D-Fendera1fa|64 : So hows you're system working now?
22:44.41*** join/#asterisk Flauto (n=zhao@71.194.194.48)
22:44.49a1fa|64its working great
22:45.10rob0bye mogorman
22:45.37[TK]D-FenderGood to hear.
22:46.13a1fa|64yeah man
22:46.18a1fa|64i had some people prank call my cellphone
22:46.27a1fa|64so i strike back : http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture
22:50.44a1fa|64.aif is much better quality
22:50.50a1fa|64i am guessing, astierisk supports aif
22:53.50a1fa|64anybody?
22:54.07a1fa|64are there any voice filters for asterisk?
22:55.19Qwella1fa|64: There is a thirdparty voicechanger app
22:55.55Qwellhttp://www.lobstertech.com/voicechanger/
22:56.17a1fa|64i am there
22:56.21QwellJunK-Y's patch is very cool.  I'm not sure if they've integrated it yet though
22:56.21a1fa|64have you tried this patch>?
22:56.37Qwellyep, works nicely
22:56.59*** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk)
22:57.23a1fa|64mind trying it out?
22:57.31a1fa|64record a demo
22:57.33a1fa|64or something
22:58.23a1fa|64http://sky.prohosting.com/oparviai/soundtouch/soundstretch.html#examples
22:58.36a1fa|64404
23:01.14MotherI managed to get chan_bluetooth working!
23:01.23*** join/#asterisk litage (n=nick@203.220.55.70)
23:01.37*** join/#asterisk dflow (i=pch@yennefer.sisco.pl)
23:01.40Motherpairing it with my BT headset and works like a charm
23:02.25*** part/#asterisk dflow (i=pch@yennefer.sisco.pl)
23:02.52[TK]D-FenderMother : How do you dial with it?
23:05.09benjk[TK]D-Fender: you don't
23:05.28benjkyou use a desktop dialer
23:06.27Mother[TK]D-Fender: well - this headset (SouthWing SH305) can either initiate voice-dial with +BVRA=1 or dial a pre-stored number
23:06.48Motherit can store the last received call's CallerID and dial it with a long press of the Push4 button
23:07.53Motherother than that as benjk says, or using an exten to dial using the headset
23:11.29zebedee1Hi all. I'm trying to get my new asterisk box to play nice behind a masquerading firewall, and am having a few problems with Asterisk reaching the outside world.
23:12.22[TK]D-Fenderzebedee1 : You need to set either "EXTERNHOST" or "EXTERNIP".  Than you need to define "LOCALNET".  This is all in the [general] section of SIP.conf
23:13.24a1fa|64they need to merge voicechanger with 1.2.2 tree
23:13.25a1fa|64:p
23:13.48zebedee1At present, I've set 'externalip' to 84.92.168.173 (my external ip), and 'localnet' to 192.168.1.0/255.255.255.0
23:13.57*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
23:14.37zebedee1...but no joy. Registration to 3rdP voip services keep timing out
23:14.39RaYmAn-Bxzebedee1: "externalip" or "externip"?
23:14.44[TK]D-Fenderzebedee1 : And you'll need to forward 5060,10000-20000 all UDP to your * box
23:15.03[TK]D-FenderThats supposed to be EXTERNIP.
23:15.59zebedee1Ah! currently externALip. I'll change it over, and give it a go.
23:16.06[TK]D-Fender:P
23:18.28file[TK]D-Fender: !!!
23:18.30*** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net)
23:18.39[TK]D-Fenderfile: ~~~
23:18.43[TK]D-Fenderer....
23:18.46[TK]D-Fenderfile: !!!
23:18.47fileyeekz, almost stained my nice new white shirt
23:19.00zebedee1Nice one - Asterisk is dialing out! Thanks a lot.
23:19.07[TK]D-Fendernp
23:20.15*** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net)
23:20.28*** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
23:21.39mrtwisterhello... loose all evening... i setup DMZ, asterisk under nat, but incoming SIP calls are one-way audio... i map 5060 and ports 10000 to 25000 too, and calls have to go :) how to set asterisk for NAT?
23:24.57[TK]D-Fendermrt : You need to set either "EXTERNHOST" or "EXTERNIP".  Than you need to define "LOCALNET".  This is all in the [general] section of SIP.conf
23:26.02wunderkinecho..
23:26.20aksiswhen someone calles into my box sip:aksis@idea-anvil.net its going to come in on 5060 right? So I have that forwared to the asterisk server. From there how is asterisk treating it? is "aksis" going to be the ext? how do I map aksis@idea-anvil.net to kphone? can kphone and asterisk be on the same box?
23:26.21[TK]D-FenderI prefer the term "cut&paste" ;)
23:26.47*** join/#asterisk Defraz (n=t0tal@103-16.69-92-cpe.cableone.net)
23:27.14infinity1realistically, how are you suppose to dial an email address from a phone?
23:27.19*** join/#asterisk chiardon (n=dorphals@200.106.223.5)
23:27.37[TK]D-Fenderaksis : You'll need to have "exten => aksis,1,dosomethingoranotherhere" in your incoming context in defined in sip.conf's [general] section
23:27.45aksisIm setting this up more for ip phones.
23:28.13[TK]D-Fenderinfinity1 : well you wouldn't dial an *E-MAIL* address...
23:28.59[TK]D-Fenderaksis : what does that imply?
23:29.08infinity1err ...i guess dialing an email address is for dialing with software?
23:29.17dorphalsighey
23:29.20QwellYou don't dial email addresses
23:29.20SkramXj #XORG
23:29.22SkramXoops
23:29.26QwellYou dial SIP addresses, or something
23:29.32dorphalsigis there much difference between asterisk 1.1 and 1.2?
23:29.42Qwelldorphalsig: yes
23:29.44Qwell1.2 actually exists
23:29.45aksisI am under the impression that you can dial an ip phone with  a domain name and a prefix?
23:29.56infinity1whats the difference between an email address and a sip address? aren't they generally the same?
23:30.01aksiserrr prefix=ext
23:30.22iDunnoinfinity1: erm, they 'look' the same, they're handled differently.
23:30.28aksisinfinity1: the domain translates to an ip and the user translates to an ext...
23:30.36[TK]D-Fenderaksis : Sort of.  Only if the hostname IS the IP in question.
23:30.39iDunnoinfinity1: email tends to use an MX record...
23:30.52iDunnothere's a different chunk of evil magic DNS for SIP
23:30.58iDunno(which I can't remember :)
23:31.02infinity1iDunno: and sip uses SRV
23:31.10iDunnoinfinity1: that's the one :)
23:31.26infinity1but its still looks like an email address. which i can't type in using a hardphone
23:31.38iDunnoyou've got a crap hardphone, then.
23:31.42Qwellinfinity1: Get a good one
23:31.51[TK]D-Fenderinfinity1 : Depends on the hardphone.  I can from mine (SPA-941), and many others suppotr it as well.
23:31.52iDunnocertainly you can type then on the Snom190
23:31.53aksishmmm so aslong as the A == MX then its all good?
23:32.18infinity1i have a polycom. it doesn't have a qwerty keyboard. i guess i could type it in somehow, but it doesn't seem practical
23:32.20aksisinf I am maping aksis to 1001
23:32.28[TK]D-Fenderaksis : if you're already using *, all SIP calls hit it be default anyways.
23:32.35iDunnoinfinity1: it's got characters on the numbers, surely...
23:32.36aksisso either way the extention can be dialed...
23:32.43[TK]D-Fenderif your want it to dial your phone right away, jsut do that in the dial-plan
23:32.44iDunnoinfinity1: you've used a mobile, right? ;)
23:33.00aksishehe... this is where I am getting lost...
23:33.08infinity1iDunno: my point is, it takes much more effort dial dail characters than #s
23:33.28iDunnoinfinity1: so set up extensions in your * setup to dial sip addresses ;)
23:33.31infinity1does anyone even dial a sip address?
23:33.44[TK]D-Fenderaksis : just add "exten => asksis,1,Dial(SIP/myphone,20)" or whatever into the context you dfeined in sip.conf
23:34.07aksisexten => aksis,1,Goto(999)                    ; alias aksis to 999
23:34.08aksisexten => 999,2,Dial(${CONSOLE})         ; this will dial ${AKSIS}
23:34.10benjkinfinity: most modern phones support some form of phonebook
23:34.26aksisthats what I was going to use, but CONSOLE isn't correct is it?
23:34.34benjkyou put the sip address in the phone book, then dial by selecting from the phone book
23:34.35infinity1benjk: what does that mean  for dialing a sip addR?
23:34.38[TK]D-Fenderaksis : that won't work.  The goto tries to jump to PRIORITY 999.
23:34.50infinity1benjk: so its like speed dial.
23:34.52infinity1?
23:34.56[TK]D-Fenderand your exten 999 doesn't have a "1" priority
23:35.00benjklike on a mobile phone
23:35.24benjkwhere you have a menu of some kind and you can scroll through all the entries you entered
23:35.36infinity1does anyone here ever dial sip addresses?
23:35.44benjkthen just press the dial button when you found the desired entry
23:35.54benjknot manually
23:35.57[TK]D-Fenderinfinity1 : I have.  A pin-in-the-ass, but can be done.
23:36.03[TK]D-Fenderpain*
23:36.11benjkouch
23:36.13infinity1there must be a better way
23:36.24benjkpin in the a$$ - sounds painful indeed
23:36.33[TK]D-Fenderinfinity1 : there is.  Put it in your dial-plan
23:36.49infinity1[TK]D-Fender: what do you mean?
23:37.30[TK]D-Fenderinfinity1 : exten => 666,1,Dial(SIP/satan@hell.com,20)
23:37.38infinity1ahh ..i see.
23:37.52infinity1thats not a bad idea. but it still requires more work.
23:38.22mrtwisterwhat is it? asterisk under nat
23:38.24mrtwister---
23:38.24mrtwisterset_destination: Parsing <sip:202.59.90.178:1000> for address/port to send to
23:38.24mrtwisterset_destination: set destination to 202.59.90.178, port 1000
23:38.24mrtwisterReliably Transmitting (NAT) to 202.59.90.178:1000:
23:38.24mrtwisterBYE sip:202.59.90.178:1000 SIP/2.0
23:38.43infinity1is there some kind of directory * can subscribe to which looks up a phone # and dials it via sip instead of using pots if possible?
23:38.55benjkyes, enum
23:38.56*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
23:39.11infinity1benjk: are any companies using it?
23:39.15benjkE164.org, E164.arpa end a bunch of others
23:39.29*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
23:39.45infinity1benjk: is there some kind of standard?
23:39.54benjkthere are already about 70-80 million numbers in there or so I seem to remember
23:40.02mrtwisterinfinity1: abou enum - ipkall.com use 100%
23:40.03*** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk)
23:40.30infinity1mrtwister: does vonage support dialing their customers using SIP?
23:40.35justinuno
23:40.36benjkinfinity: we have a howto for enum
23:40.54mrtwisterinfinity1: to enum? think, not
23:40.55benjkhttp://www.astmasters.net/howtos.html
23:41.11benjkscroll down a bit to the Enum article
23:41.30infinity1benjk: cool.
23:41.41infinity1looks like something i should probably already know :)
23:45.29*** join/#asterisk Dovid (i=dovi5988@17.sub-70-193-186.myvzw.com)
23:47.52infinity1has anyone implemented enumlookup with their standard dialing with success? or is it better to have the user manually do an enum dial?
23:48.58*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
23:49.16Dovidhello there shmaltz
23:49.41shmaltzhi Dovid
23:49.57SkramXoy veu!
23:50.00SkramX*Vey!
23:50.02SkramXlol
23:50.08shmaltzlol
23:52.29*** join/#asterisk rappu_noxep (n=me@harry.org.ua)
23:53.25aksis[TK]D-Fender: thanks, I am able to ring my phone now...
23:54.51troyb1hey guys i have a cisco 7940 SIP 7.3 and im having trouble dialing sip email addresses
23:55.59QwellThey aren't email addresses
23:56.14troyb1sorry for my mis-statement :)
23:56.22Qwellwe just went over this :P
23:56.35troyb1is it in my best interest to scroll up?
23:56.36*** join/#asterisk PJMattF (n=matt@cpc2-fare3-3-0-cust79.cos2.cable.ntl.com)
23:56.52*** join/#asterisk Vijay (i=Vijay@203.122.28.109)
23:56.54mrtwistersomeone, please call me at 500@80.240.10.68 over sip, i tesing NAT...
23:57.01*** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
23:57.12troyb1mrtwister i'd volunteer but i cant even get mine working *grin*
23:57.27DrukenHMEvolunteer for what ?
23:57.29Vijayhello mrtwister
23:57.38mrtwisterhi
23:57.42troyb1calling his sip address
23:57.54PJMattFdoes anyone have any suggestions for a good (i.e. fairly easy-to-use) linux distro for experimenting with Asterisk - I'm sick of the amount of obvious stuff that's absent from Ubuntu.
23:57.58Vijayi need to configure my asterisk run to run behind the nat
23:57.59DrukenHMEmy system will do it
23:58.08troyb1PJMattF ubuntu
23:58.09mrtwisterPJMattF: ubuntu is very good
23:58.14Vijaygentoo is the best i would say
23:58.15troyb1mrtwister i beat ya too it
23:58.17DrukenHMEgoto www.abss.ca and enter in your sip URL into the callme box
23:58.19PJMattFit's missing tons of damn packages though
23:58.33rob0what is missing from Ubuntu?
23:58.33PJMattFand i have no idea which ones are stopping infernal ztdummy from building
23:58.34mrtwisterPJMattF: @home and rapid
23:59.05Dovidlol
23:59.06PJMattFwell it may not be "missing" but there's assloads which isn't in the default ubuntu install
23:59.09mrtwisterPJMattF: but for live system ubuntu server, debian server is very good.... gentoo too, but i not like to wast time to compile whole distro
23:59.12rob0most of the "easy" distros do not install what you need to compile things
23:59.14DovidI like CENT OS works great for me
23:59.16DrukenHMEtroyb1: who needs a SIP url called?
23:59.33mrtwistersomeone, please call me at 500@80.240.10.68 over sip, i tesing NAT...
23:59.34mrtwisterme
23:59.36mrtwister:)
23:59.51Qwellmrtwister: <DrukenHME> goto www.abss.ca and enter in your sip URL into the callme box

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