irclog2html for #asterisk on 20051104

00:00.01ManxPowerOne of our internal "standard patches" is to set it to 350
00:00.15blitz[laptop]ManxPower: yo buddy... wuz up?
00:00.36file[laptop]blitz[laptop]: yay
00:00.46blitz[laptop]file[laptop]: zup
00:00.57file[laptop]blitz[laptop]: oh nothin'
00:01.08iCEBrkrDon't make me go get on my laptop
00:01.12file[laptop]oh oh oh
00:01.20ManxPowerblitz[laptop], Other than the fact that god hates me?  Not much.
00:01.21file[laptop]the soup nazi episode of Seinfeld is on
00:01.22blitz[laptop]iCEBrkr: TSN!
00:01.24BentleyManxPower: so if this was a problem using a sip-pstn gateway, and not zap card, then I am prolly S.O.L., eh?
00:01.34ManxPowerBentley, Yup.
00:01.36rene-A quick question, is realtime supported in sub 1.2 versions of asterisk ( in 1.0.9 ? )
00:01.40file[laptop]rene-: no
00:01.41*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
00:01.44blitz[laptop]nope
00:01.46rene-:(
00:01.48ManxPowerrene-, no and it never will be
00:01.53blitz[laptop]screw 1.0.9
00:02.05file[laptop]screw stable! up with 1.2.0!
00:02.15BoRiSstable....pfffffff
00:02.34BoRiSlol :-p
00:02.41ManxPowerIf you need Realtime wait a year or two until 1.2 is released.
00:02.59blitz[laptop]I'm getting ready to move things from 1.0 to 1.2 beta soon
00:03.07blitz[laptop]and then keep up with head
00:03.12iCEBrkrHrrm.
00:03.18blitz[laptop]1.0.9 is just giving me too many problems
00:03.21*** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com)
00:03.22iCEBrkrI should upgrade my 1.0.9 stuff
00:03.28rene-I know, but my boss is having the  idea of downgrading to 1.0.9 to see if that will help in our intermitent one way audio issue our realtime based GUI notwithstanding
00:03.37ManxPowerI upgraded to CVS-HEAD a while ago.  What a nightmare.
00:04.06ManxPowerEventually I found a CVS date that worked with only minor oddities.
00:04.12asteriskmonkeyim tryign to creat a small context that lets me dial a 10 digit number out once dialed in the asterisk :P anyone know how i go about that?
00:04.27MacRohardHey I wrote a 4 line patch to asterisk (res/res_agi.c - the end of the setup_env function) that exports variables to your AGI scripts.. someone might want to consider adding this to the main asterisk tree (I can't even believe it's not there already)
00:04.27MacRohard<PROTECTED>
00:04.27MacRohard<PROTECTED>
00:04.27MacRohard<PROTECTED>
00:04.28MacRohard<PROTECTED>
00:04.28iCEBrkrasteriskmonkey: IT'S IN THE WIKI
00:04.50blitz[laptop]~docs
00:04.52jbotsomebody said docs was 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
00:04.52blitz[laptop]~thebook
00:04.53jbotsomebody said thebook was Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk for purchase, or FREELY AVAILABLE under the Creative Commons license at http://www.asteriskdocs.org
00:05.28mog_workwow blitz we get it
00:05.31mog_workyou wrote a book
00:05.33mog_workit rocks
00:05.36iCEBrkrhaha
00:05.37mog_workno need for spammination
00:05.38ManxPowerMacRohard, submit it on bugs.digium.com  You might have to submit a disclaimer.
00:05.49e-miliohello all
00:06.19ManxPowerblitz[laptop], I read the first chapter or so.  It's much better than I expected it to be.
00:06.46e-milioHave anybody made spandsp work stable ?
00:07.03e-milioI mean with 100% fax sending success ?
00:08.00blitz[laptop]mog_work: sorry... I wasn't really trying to spam for the book... I was really just trying to direct someone to documentation
00:08.10e-milioWe are having probs with txfax. Page random cuts in halp
00:08.10blitz[laptop]I should really just change that link to show a link
00:09.39mog_workyeah... ^)^
00:09.57file[laptop]you will never get 100% success
00:09.59file[laptop]nevar!
00:10.10e-milioi will set for 80%
00:10.15alephcomNot that you get 100% over the pstn anyways. :-)
00:10.24file[laptop]you can make it happen on digital stuff... a PRI
00:10.31e-miliowe have a pri
00:10.52e-milioone fax is fine
00:10.58e-milioto not
00:11.05e-miliotwo not
00:11.20blitz[laptop]I'm not going to make any money on it... so I'm not spamming for that reason.. just trying to give the link out so people can use it
00:11.47mog_worki hear you blitz
00:11.55mog_workim not trying to reign on your parade
00:12.06file[laptop]Matttttt
00:12.24e-miliofile[laptop]: Have you had success with spandsp
00:12.25e-milio?
00:12.29blitz[laptop]mog_work: I know :)
00:12.30file[laptop]e-milio: yes.
00:12.35file[laptop]only for receiving though
00:12.38e-miliommm
00:12.43e-miliono we need to send
00:12.44mog_workfileeeee
00:12.45file[laptop]sending use hardware.
00:12.50blitz[laptop]just changes the jbot tag... should be a lot shorter now
00:12.58file[laptop]mog_work: how are you?
00:13.11mog_workim actually a little iritated
00:13.14blitz[laptop]going back to watch the leafs game
00:13.16mog_workhad a hell of a call
00:13.25e-miliofile[laptop]: which hardware would you recommend ?
00:13.35file[laptop]brooktrout
00:13.41file[laptop]mog_work: :(
00:14.35mog_workit happens
00:14.37MacRohardManxPower, ok. done.
00:14.46e-miliofile[laptop]: with * or other pkg ?
00:15.13file[laptop]hylafax
00:15.14mog_workwith asterisk get a t1 card
00:15.26mog_workwith hylafaxits only that brooktrout right?
00:15.42e-miliowe are trying with t406p and pri
00:15.46e-milionot good
00:16.12e-miliodont see much cpu usage neither irq activity but
00:16.23mog_workwhat seems to be the problem with that emillio
00:16.24e-miliosimultaneous faxes just cut
00:16.41mog_workouch
00:16.45e-milioramdonly the faxes just get cut in the middle
00:16.54mog_workhmm using latest spandsp?
00:16.59e-milio0.0.2
00:17.32e-milioi tried on two dif machines
00:17.36e-miliowith 2 dif cards
00:17.40e-miliowith 2 dif pris
00:17.47e-miliowith 2 dif providers
00:17.56e-miliosame results
00:18.23mog_workyuck
00:18.34mog_workhylafax is probably way to go
00:19.05e-miliois hylafax and two ext modem stable ?
00:20.16file[laptop]very very hungry
00:20.23*** join/#asterisk zotz (n=zotz@24.231.47.168)
00:23.34alephcome-milio:  I have nothing but good to say about hylafax when it comes to stability.
00:24.07*** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
00:25.02e-milioalephcom: using any hw in particular ?
00:25.09*** join/#asterisk gbrowley (n=gbrowley@221.127.107.205)
00:26.02*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
00:26.50*** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net)
00:27.48ManxPowerMacRohard, Be prepared to defend your patch to the death.
00:28.48alephcome-milio: We used just a USR sportster.
00:29.01gbrowleyI'm having a small problem with running my asterisk server on a dynamic IP address, when the IP address changes although asterisk reports them as registered when a converstaion takes place they can hear me but I can not hear the remote side. Anyone have any ideas?
00:29.19alephcome-milio: we quite using hylafax because we needed a nicer windows interface than was available but that was the only problem.
00:30.10e-milioalephcom: i just need to send pre-prepared tiff files, 2 o 3 simultaneoulsly
00:34.02*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
00:34.03MacRohardManxPower, why do you say that ?
00:34.27ManxPowerMacRohard, you'll understand soon, grasshopper.
00:34.32MacRohard(i don't .. i don't really give a fuck.. I already patched *my* asterisk)
00:34.50*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
00:35.08ManxPowerMacRohard, you can also submit your patch to the openpbx people
00:35.14*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
00:35.25MacRohardis that like some asterisk fork?
00:35.44sahafeezhow do you set the outbound callerid for everyline. i want every call out the pri to say the same thing
00:36.09ManxPowersahafeez, Set it before your outbound Dial statement
00:36.26sahafeezah, thanks.
00:38.36ManxPowerMacRohard, Yes, it's an "fully GPL" fork of Asterisk.
00:40.20asteriskmonkeyI can seem to get asterisk to do a callthrough thing properly
00:40.34asteriskmonkeyive tired usign the example online and that didnt work
00:46.21*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
00:46.33e-milioplease help: How can I detect if a pri is having frame slips ?
00:47.46*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
00:48.41*** join/#asterisk juice (n=juice@mo-67-77-189-131.dyn.sprint-hsd.net)
00:48.42*** part/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
00:48.57e-miliocommunity: How can I detect if a pri is having frame slips ?
00:50.20ManxPowerLooks like I need to get a new USA passport before they start putting RFID chips in them
00:50.41Sedoroxyou'll proably have to go get a new one
00:50.41ManxPowere-milio, you will hear occasional clicks in the audio.
00:50.58*** part/#asterisk asteriskmonkey (n=phil@HSE-Windsor-ppp211407.sympatico.ca)
00:51.06ManxPoweryou want the 2nd field of your span= line to be 1 for the line you want to get your timing from.
00:51.13e-milioManxPower, any log anywhere ?
00:51.41ManxPowere-milio, not that I am aware of.
00:51.51ManxPowerHDLC abort errors are not frame slips
00:52.01e-milioManxPower: mm
00:52.53e-milioManxPower: any idea how to prevent them ?
00:53.05ManxPowere-milio, frame slips or HDLC abort errors?
00:53.15e-milioManxPower: HDLC
00:54.03ManxPowere-milio, Read the Wiki, search the mailing list archives, replace your motherboard with a different brand/model, don't use SATA drives, don't use RAID controllers, enable IDE unmask interrupts, enable IDE DMA
00:54.07ManxPower~docs
00:54.08jboti guess docs is 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
00:54.09ManxPower~mailinglist
00:54.10jbotsomebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
00:54.21tzangerbasically you want a good PCI subsystem
00:54.57ManxPowere-milio, HDLC Abort errors are basically cause by 2 things. 1) PCI IDE issues (the most common) and 2) line problems (uncommon)
00:55.12e-milioManxPower: I am using scsi
00:55.30e-miliowith raid
00:55.41e-milionot good 4 what you said
00:56.03ManxPowerResults 1 - 10 of about 250 from lists.digium.com for  HDLC error. (0.23 seconds)
00:56.28ManxPowere-milio, RAID, SATA, etc are all just things that commonly lock interrupts for so long that the zaptel card loses data
00:56.52ManxPowerother common things are graphics modes, etc
00:57.14e-milioManxPower: we are running very basic. only the raid might be
00:57.18ManxPowerResults 1 - 10 of about 791 from lists.digium.com for  HDLC. (0.14 seconds)
00:57.32ManxPowere-milio, HDLC Abort errors are some of the hardest things to fix.
00:57.39ManxPowerThose and Echo
00:59.14e-milioManxPower: do you think hdlc can be break fax trans with spandsp ?
00:59.24tzangere-milio: fuck yeah
00:59.30*** join/#asterisk PBXtech (i=nik@223.sub-70-213-183.myvzw.com)
00:59.41tzangerfax/modem calls are *extremely* sensitive to that kind of loss
00:59.55ManxPowere-milio, not can, WILL
01:02.40e-miliotzanger, ManxPower, thanks
01:07.23*** join/#asterisk froguz (n=froguz@2-43-112.adsl.terra.cl)
01:07.27froguzhi
01:08.41PBXtechis there a skype to asterisk solution?
01:08.58ManxPowerPBXtech, No.
01:09.11PBXtechnot even external hardware?
01:09.36ManxPowerIs there a POTS/Skype adapter that supports far end disconnect supervision?
01:10.13PBXtechgot me :)
01:10.26froguzsomebody can tell me why Asterisk@Home uses a macro called hangupcall wich first reset the CDR before doing a simple Hangup? is that necesary? (this macro is called from app-callforward)
01:10.44ManxPowerfroguz, that's really a question for the A@H channel
01:11.09mog_workyes yes it is ^_^
01:11.11PBXtechor #amportal channel
01:11.23froguzoh i'm sorry i didn't know that channel exist
01:12.10mog_workits all good froguz
01:12.18froguzthanks
01:12.21mog_workyou just wont find a lot of love for amp and a@h
01:12.22mog_workhere
01:12.59ManxPowerA@H is to Asterisk, as a McDonalds hamburger is to a quality steak.
01:13.12*** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
01:13.49froguzhahaha i can notice that, i don't love a@h neither, it's just i have to deal with it
01:14.24mog_workno you dont
01:14.28mog_workbreak free my friend
01:14.58froguzwell yes, i have. work, you know. i'm not my own boss
01:15.07froguz=(not yet
01:15.41*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
01:18.51*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
01:21.50DrukenHMEwhat is the best variable to use for the caller id number?
01:22.17file[laptop]calleridnum
01:23.33DrukenHMEfile[laptop]: in your understanding... this would do ?
01:23.37DrukenHMEGotoIf($[${LEN(${CALLERIDNUM})} = 7]?6:8)
01:23.58moverwhy app_rxfax work on one box fine and on another box same asterisk (head) but diffrent hardware no? I user ztdummy
01:28.04DrukenHMEok, never mind... i finally got it to work... at least i think so
01:29.31*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
01:35.54infinity1anyone use destar?
01:36.59infinity1yay. beta2 packages for debian!
01:38.49sahafeezhttp://www.voip-info.org/wiki-Asterisk+cmd+Directory
01:38.57sahafeezPressing "1" will exit setting up the channel to enter the extension selecte
01:39.10*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
01:39.15sahafeezwhat does this look like in extentions.conf. there is no example.
01:39.44sahafeezi have the directory working, but after hitting 1 i do not get transferd
01:40.04infinity1tzafrir_laptop: hey! i c you :) ...can i get an opinion on destar? thanks for the deb packages btw.
01:42.06*** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net)
01:47.05*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
01:47.28harryvvHas anyone seen Arial around?
01:48.43PBXtechdoes the sc430 have the same probs as the sc420?
01:48.57infinity1no one has
01:48.57infinity1ariel There was no such nickname
01:49.52*** join/#asterisk paravoid (i=paravoid@void.photonics.ece.ntua.gr)
01:49.55infinity1has anyone used amp ? does it work well with a custom extensions.conf?
01:49.57*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
01:50.42*** join/#asterisk dsmouse (n=mouse@prism.datastacks.com)
01:51.04*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:53.34*** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85)
01:54.23*** join/#asterisk seacode (n=anthm@adsl-69-215-147-152.dsl.milwwi.ameritech.net)
01:54.50*** join/#asterisk darkpixel (n=darkpixe@209.216.179.194)
01:56.57*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
01:57.24PBXtechis pot really legal in denver now days? damn radio station foolin or what
01:58.45[Airwolf]PBXtech, I don;t have to worry about that :P
01:59.58*** part/#asterisk gbrowley (n=gbrowley@221.127.107.205)
02:00.42*** join/#asterisk znoG (n=gs@200.115.217.183)
02:01.18e-milioManxPower: where ztclock.c can be found ?
02:03.04*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
02:03.05e-milionever mind
02:05.38*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
02:18.05*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
02:20.31justinuany polycom experts around?
02:21.20justinubootrom 2.6.2 support HTTP?
02:22.02*** join/#asterisk darkskiez (n=darkskie@host86-138-169-225.range86-138.btcentralplus.com)
02:22.23*** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net)
02:24.39BhaalWKAnyone here had experience with Alcatel OmniPCX Office units?  Just wanna know how the rack mount kit works?
02:26.53*** part/#asterisk CoaxD (i=coax@shell1.cornernet.com)
02:33.36sahafeezquestion: in extension the s,1 bla is a cache all right? or not?
02:34.51justinuwhat?
02:36.03sahafeezwell s = start and in all the examples then have exten => s,1,app
02:36.37justinuso what's the question?
02:36.41justinui didn't understand
02:37.06*** join/#asterisk axscode (i=axscode@210.213.106.189)
02:37.20axscode~fxo
02:37.22jbotforeign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo
02:38.07sahafeezhum. not sure i know. i know what my inbound number is...but what if i want to make a rule that says any inbound calls do this.
02:38.18justinuoh
02:38.25justinuyou mean a "catch all"
02:38.27*** part/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com)
02:38.29justinunot a cache all
02:38.31sahafeezyes
02:38.41sahafeezsorry. having brain issues for the past week.
02:38.45justinuno, i don't think s is a catch all
02:38.57justinuyou want _N. iirc
02:39.14axscodewhat do i need from my computer so that i can connect and dial to POTS or my existing TELCO?
02:39.34justinuexactly what the bot said
02:39.56blitz[laptop]_X.  << catch anything starting with 0-9
02:40.00blitz[laptop]N == 2-9
02:40.03justinuoh, right.
02:40.05justinuder
02:40.09sahafeezfxs = subscriber, fxo = office
02:40.11blitz[laptop]Z == 1-9
02:40.18*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
02:40.22blitz[laptop]fxs == station
02:40.47sahafeezblitz: depend on who you talk to and how long they have been doing telco;)
02:41.02blitz[laptop]:)
02:42.59sahafeezi have exten => 2004,1,Directory(default,incoming,f)
02:43.07axscodeso i need an fxs,fxo card? is there any PCI that is around aside from digium?
02:43.33blitz[laptop]nope
02:43.42sahafeezand i can spell the person, find it, hit the number 1 and get nothing. is there another line i am missing? do i need to defince and extension for 1
02:43.43blitz[laptop]unless you use an ATA or something
02:43.50axscodewhats an ATA?
02:43.52axscode~ATA
02:43.53jbothmm... ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info
02:43.59justinuaxscode: sip ata, or sip channel bank
02:44.14justinuwhy wouldn't you use an ATA?
02:44.20justinuwhy put something in the pbx?
02:44.23sahafeezits a hack
02:44.29justinuhow is it a hack?
02:44.35blitz[laptop]someone showed a wicked mini-channel bank that connects via ethernet that looks very sweet
02:44.39mog_homeyou could put something in the pbx
02:44.41blitz[laptop]a hack? I disagree
02:44.44mog_homeaka 2400p
02:44.48mog_homeand be dominate
02:44.51sahafeezjust my 2 cents. depends on what you are trying to do i guess
02:44.52blitz[laptop]aye
02:45.10justinui think the ATAs are quite elegant, you don't have to restart the server to add them
02:45.11blitz[laptop]mog_home: do you remember what that mini channel bank thing was at astricon?
02:45.15sahafeezi just got a PRI and one TDM400 for the fax machine
02:45.22mog_homeyeah the 8 porter
02:45.23justinuif computers stop having PCI busses, you can still use your ATAs
02:45.28blitz[laptop]mog_home: yah... that thing
02:45.30mog_homexorcoms
02:45.34blitz[laptop]ahhh right
02:45.35blitz[laptop]thanks
02:45.37axscodeanyone have tried using "PANASONIC D1323 16SLC Digital Super hybrid system"
02:45.38justinui'd go with the redphone TDMoE boxes for PRI
02:45.42sahafeezso anyone on the directory stuff? i read the faq, no examples
02:45.59blitz[laptop]back later -- going to try focusing on programming for a bit
02:46.04justinugood luck
02:46.10blitz[laptop]thx -- I'll need it :)
02:46.15blitz[laptop]E911 portal :)
02:46.21justinuyay! :)
02:47.43axscodeso ATA = ADAPtER for POTS to ETHERNET?
02:47.48justinubasically
02:47.58justinucheck out the SPA-3000
02:48.04justinuone FXS port, one FXO port.
02:48.13justinuvery slick
02:48.51axscodeim here at philippines.. mostly i cant find one
02:49.05justinuhmm... asia is very hot on voip
02:49.07axscodeunless ill order from outside.. which i need one today.
02:49.14justinui'm sure you can get one from singapore or hong kong
02:49.19axscodeasia = singapore for voip..
02:49.26justinuyeah
02:49.46*** join/#asterisk SplasPood (n=sp@160.79.255.3)
02:49.49justinudoes air asia fly to to manilla?
02:49.54justinutheir flights are really cheap
02:50.01justinuyou could almost afford to go to singapore just to pick some up
02:50.12axscodeoh... and im poor.. so i doubt i can afford one.
02:50.23justinuthey cost about 100USD
02:50.28justinusome are as cheap as 45USD
02:50.31axscodeand i dont have that..
02:50.58axscodethats why im looking for a clone brand for PCI card to connect to pots
02:52.02axscodejustinu: ASTERISK ===> SPA3000 ===> TELCO
02:52.22axscodeis that right? how will the exten => looks like?
02:52.23justinuyeah... and you can plug a regular analog phone into the SPA3000 also
02:52.48*** join/#asterisk mwgbc (n=mwallace@adsl-71-132-220-182.dsl.pltn13.pacbell.net)
02:52.56justinuextension for what, dialing out?
02:53.15axscodedialing out.. and how about recieving a call from outside? is that posible?
02:53.19justinusure
02:53.24orlokwhy not just start off with a performance car? :)
02:53.29mwgbcWhat kind of problem would cause the error:
02:53.29mwgbcNov  3 09:37:58 NOTICE[15327]: pbx_spool.c:243 attempt_thread: Call failed to go through, reason 1
02:53.37orlokDoh, wrong channel.
02:53.43justinuhaha
02:54.23axscodejustinu: can u gimme example how the dialing out will look like? do i need a zapata for that?
02:54.27mwgbcorlok: This error started spontaneously without anyone changing any conf files
02:54.29justinuno zapata
02:55.06mwgbcIt was working previously
02:55.23justinusomething like exten => _X.,1,Dial(SIP/${EXTEN}@spa3000);
02:56.10justinuaxscode: you own't need any hardware in your asterisk box to make that work, besides an ethernet card
02:56.30axscodeoh. nice... so i can dial outside just having an ATA?
02:56.34axscodehehe. sweet..
02:56.35justinucorrect.
02:56.37sahafeezok then what is the S extension they have in every example
02:56.52justinuit's just the starting point for macros and stuff, i guess
02:57.08axscodethis is cool.. ok. so i dont need digium modems..
02:57.16justinunot really
02:58.00axscodejustinu:  how about when they dial-in to a specific asterisk sip user? is that stated also in extensions?
02:58.08justinuyeah
02:58.29justinuif you have caller ID, you can even route based on that.
02:59.35axscodeok so it will be. ..  exten=> 4324343,1,(ask-for-new-sip-number) <---where 4324343 is the number of the POTS line?
02:59.57justinuwell, yeah
03:00.03axscodeok.. got that..
03:00.10justinusorta
03:00.13axscodeok so ATA then..
03:00.23axscodehmm. problem is.. where can i find that.. haha..
03:00.24justinuthats what i'd do
03:01.16axscodehey justin.. any ATA right?
03:01.23justinuit needs an FXO port
03:01.26axscodeor some ATA dont work for asterisk.
03:01.35justinuas long as it's SIP
03:01.41justinuor iax
03:01.48justinudigium sells some cool iax ATAs
03:01.49axscodeso ATA with SIP support? or iax support...
03:01.51justinuvery small
03:01.59justinuyeah, and FXO port
03:02.02justinu~fxo
03:02.03jbotforeign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo
03:02.08axscodeFXO, ETHER port
03:03.54*** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com)
03:04.11justinuaxscode: http://www.sipura.com/products/spa3000.htm
03:04.37froguzis there a variable wich tell me the context for the calleridnum?
03:05.10froguzor any certain extension?
03:05.42axscodejustinu: yups ive seen that one..
03:06.38*** join/#asterisk PresuntoRJ (n=Presunto@200141086252.user.veloxzone.com.br)
03:07.02axscodethanks.. gud help..
03:07.08justinuno prob
03:07.17axscodeotherwise. if thers none exist.. have to check for digium pci modems
03:07.36*** join/#asterisk Splas (n=sp@160.79.255.3)
03:12.03*** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET)
03:12.36*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) [NETSPLIT VICTIM]
03:12.43*** part/#asterisk PresuntoRJ (n=Presunto@200141086252.user.veloxzone.com.br)
03:12.44*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM]
03:12.57*** join/#asterisk VxJasonxV` (n=jason@unaffiliated/VxJasonxV) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk riksta (n=rick@62.6.163.90) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk b0xii (i=b0xii@pool-68-238-104-195.dfw.dsl-w.verizon.net) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk mutilator (n=animenod@65.111.201.79) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk Druken (i=Druken@67.69.139.226) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk mmmToop (n=michael@196.31.11.194) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk zigman (i=zigman@irc.zigman.de) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk sambal (n=sambal@213.148.236.189) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk Sipingal (n=Sipingal@210.21.206.75) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-14-134.cybersurf.com) [NETSPLIT VICTIM]
03:13.40*** join/#asterisk prh (n=paul@212.13.203.80)
03:13.41*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk syle (n=blah@unaffiliated/syle) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk Emrah (n=user@adslgva0491.worldcom.ch) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk mog_work (n=mogorman@gateway.digium.com) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk cp5 (n=samy@69.106.105.125) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk oogle (n=jart@justin.ctlinc.com) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk pc2 (n=pc@209.151.52.81) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk loud (n=ariel@cypher.punk.net) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk brookshire (n=nubb@gateway.digium.com) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk adtech (n=asdf@199.185.139.89) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk javo (n=nemitzj@68-187-84-135.dhcp.eucl.wi.charter.com) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) [NETSPLIT VICTIM]
03:13.41*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk izo (n=izo@193.202.114.43) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk sivana (n=sivana@mixdown.ca) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk bon (n=bon@217.172.148.5) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk trelane (i=trelane@66.93.203.199) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk R3DB0x (i=nobody@66.142.28.36) [NETSPLIT VICTIM]
03:13.42*** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk Hmmhesays (n=Neg@24-117-213-113.cpe.cableone.net) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk thuper (n=thuper@gateway.digium.com) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk mattHelm (n=root@adsl-68-90-140-81.dsl.fyvlar.swbell.net) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk advorak (n=advorak@atypedigital.ischool.washington.edu) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk lunk (n=lunk@negative-influence.com) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk hohum (i=corbe@snoop.burghcom.com) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk JonR800 (i=jon@p1mp.org) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk Champi (i=Champi@damn.e-leet.be) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk cypromis (n=michael@83.149.70.59) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk orlok (n=jwr@202-44-174-4.nexnet.net.au) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk Beave (n=beave@vistech.org) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM]
03:13.43*** join/#asterisk ]data[ (n=data@69.56.182.146) [NETSPLIT VICTIM]
03:13.44*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM]
03:13.44*** join/#asterisk jarrod (i=jarrod@juniperyour.net) [NETSPLIT VICTIM]
03:13.44*** join/#asterisk Corydon-w (i=brown@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM]
03:14.18file[laptop]to all those who died in the netsplit, we will miss you!
03:14.19*** join/#asterisk toddf (n=toddf@ns0.fries.net)
03:14.27file[laptop]ah who am I kidding, no we won't
03:14.46file[laptop]oh looksee they're back from the dead
03:16.25justinuyou were the ones who died
03:20.00*** join/#asterisk CoaxD (i=coax@shell1.cornernet.com)
03:20.15*** join/#asterisk lunk (n=lunk@negative-influence.com) [NETSPLIT VICTIM]
03:21.31Lostfrogwow.. that's one hell of a split.
03:21.53*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
03:23.47axscodeasl
03:24.52*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
03:24.52*** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) [NETSPLIT VICTIM]
03:24.56Lostfroglol.. asl on freenode.
03:25.36iCEBrkrwow and lilo isn't spamming us with a reason why and how we should hang in there?
03:25.36iCEBrkrUNFUCKINGBELIEVEABLE
03:25.45QwelliCEBrkr: He's with them
03:25.54iCEBrkrhuh?
03:26.35Qwellhe's spamming the people that split
03:26.36*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) [NETSPLIT VICTIM]
03:26.36*** join/#asterisk snitt (i=snitt@snitt.info) [NETSPLIT VICTIM]
03:26.36*** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) [NETSPLIT VICTIM]
03:26.36*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
03:26.36*** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) [NETSPLIT VICTIM]
03:27.03iCEBrkrhahaha
03:28.26Qwelltook long enough
03:30.22LostfrogI like getting packages. :)
03:31.22DrukenHMEi personally suffer from sometimers
03:31.48iCEBrkrI dunno if I'm in a sober nuff state of mind to upgrade to 1.2.0 beta2
03:31.48LostfrogMmm.. 50pk Verbatim 8x DVD-R
03:31.48*** join/#asterisk marten_ (n=marten@83.227.150.77) [NETSPLIT VICTIM]
03:31.48*** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com)
03:32.10LostfrogGumbles?
03:32.35nomazdahmm
03:34.07DrukenHMEjessus christ... 22C in the afternoon, 5C overnight.... it's just not right
03:34.39*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk znoG (n=gs@200.115.217.183) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk distortion (i=distorti@junipero.3sheep.com) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk test34 (n=test34@unaffiliated/test34) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk blop (i=blop@VoIP-with-Asterisk.mgcp.h323.sccp.sip.iax.be) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk mrtwister (n=mrtwiste@cable-9-42.cgates.lt) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk file (n=jcolp@mctnnbsa31w-142166116148.nb.aliant.net) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk CMike (i=daemon@c-fe4171d5.116-1-64736c10.cust.bredbandsbolaget.se) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk MatsK (n=mk@55.80-203-80.nextgentel.com) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk Mw3 (i=mw3@national.t-error.hu) [NETSPLIT VICTIM]
03:34.39*** join/#asterisk bancus (n=treed@athena.crschmidt.net) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk Himeko (n=himeko@S01060040ca128fc3.ed.shawcable.net) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk doug (i=doug@zaxxon.telerama.com) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk many (n=many@daheim.ukeer.de) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk ender (n=me@fedora/ender) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk TMirage (n=mirage@cust.12.229.adsl.cistron.nl) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk JunK-Y (n=junky@67.71.156.36) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk kshumard (n=root@gateway.digium.com) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk thinker- (n=thinker@host254-190.pool15199.interbusiness.it) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk h3x (n=h3xor@64.192.116.16) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk Falle (i=falstaf@voip-forum.se) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk ApEtc (i=apetc@ip68-3-225-51.ph.ph.cox.net) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk welby (n=welby@tollcross.wheely-bin.co.uk) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net) [NETSPLIT VICTIM]
03:34.40*** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk eldu (i=damajor@tuxmania.org) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk inspired (i=mikael@213.197.167.61) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk Linx (n=linx@debian.geek.nz) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk cryzeck (n=cryzeck@www2.gathering.org) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk locksy (n=nlocksy@mrtg.sisgroup.com.au) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk wildcard0 (n=generic@S0106006097e16040.vc.shawcable.net) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk msw (n=msw@rdu-nat.rpath.com) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk __Elf (n=irc@rivendell.glassfish.net) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk encode (n=encode@li8-128.members.linode.com) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk wibblewobble (n=tim@sophie.wobb1e.co.uk) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk JamesDotCom (n=james@sweep.bur.st) [NETSPLIT VICTIM]
03:34.41*** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) [NETSPLIT VICTIM]
03:35.03*** join/#asterisk JonR800 (i=jon@p1mp.org) [NETSPLIT VICTIM]
03:35.37Qwellfunny
03:35.38kuku5anyone max out the broadvoice line
03:35.38kuku5like what is the limit
03:35.47*** join/#asterisk prh (n=paul@212.13.203.80)
03:35.51*** join/#asterisk cypromis (n=michael@83.149.70.59) [NETSPLIT VICTIM]
03:35.54*** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com)
03:36.04*** join/#asterisk flashnet (n=flashnet@62.79.123.79.adsl.hvi.tiscali.dk)
03:36.27LostfrogYou are only supposed to use one incoming or one outgoing.
03:36.44kuku5for the unlimited plan
03:37.00LostfrogAt least, that's how I read the service agreement.
03:37.03LostfrogOhh.. minutes.
03:37.17LostfrogI heard if you use too many minutes, they bump you up to business use.
03:37.21*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
03:38.22LostfrogI won't ever get a residential plan from them..
03:38.25blitz[laptop]iCEBrkr: :D
03:38.25blitz[laptop]rm -f /usr/include/asterisk/* and /usr/lib/modules/asterisk/*
03:38.25blitz[laptop]seems to help cause less problems once you do your make install
03:38.27iCEBrkrCoaxD: ??
03:38.27tzangerCoolAcid: ?
03:38.27tzangerer CoaxD ?
03:38.27CoaxDgee. I'll leave my fucking toll-free DID customers hanging for 3 days without even posting so much as a news item on their website describing the outage
03:38.37LostfrogI've read too my bad stories.
03:38.54iCEBrkrblitz[laptop]: I DID! I DID!
03:39.08iCEBrkrblitz[laptop]: Oh, it built just fine.
03:40.00*** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com)
03:40.04blitz[laptop]now that I got java working in my browser... I forget why I needed it working in the first place...
03:40.04iCEBrkrblitz[laptop]: Just gotta mod-up extensions.conf
03:40.04iCEBrkrtzanger: Hah, I did the same, but corrected it
03:40.05LostfrogCould someone explain todays Dr. Fun to me?
03:41.02blitz[laptop]nice
03:41.02iCEBrkrI'm a bit buzzed.
03:41.05blitz[laptop]can't wait till I get back to my house... I think I'll have a bunch of packages waiting for me
03:41.07blitz[laptop]some money too me thinks!
03:41.29iCEBrkrweeeeeeeeeeeeeeeeeewhooooooooooooooooooooooooo
03:42.01*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
03:42.05blitz[laptop]mog_home: lol
03:42.05blitz[laptop]mog_home: come on over!
03:42.21mog_homei dont think i have enough gas to do it
03:42.46*** join/#asterisk riksta (n=rick@62.6.163.90)
03:42.46*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
03:42.46*** join/#asterisk mutilator (n=animenod@65.111.201.79)
03:42.47*** join/#asterisk Druken (i=Druken@67.69.139.226)
03:42.47*** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl)
03:42.47*** join/#asterisk mmmToop (n=michael@196.31.11.194)
03:42.47*** join/#asterisk sambal (n=sambal@213.148.236.189)
03:42.47*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-14-134.cybersurf.com)
03:42.47*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
03:42.48*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
03:42.48*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
03:42.48*** join/#asterisk cp5 (n=samy@69.106.105.125)
03:42.48*** join/#asterisk oogle (n=jart@justin.ctlinc.com)
03:42.49*** join/#asterisk pc2 (n=pc@209.151.52.81)
03:42.49*** join/#asterisk loud (n=ariel@cypher.punk.net)
03:42.49*** join/#asterisk brookshire (n=nubb@gateway.digium.com)
03:42.49*** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net)
03:42.50*** join/#asterisk javo (n=nemitzj@68-187-84-135.dhcp.eucl.wi.charter.com)
03:42.51*** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net)
03:42.51*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
03:42.51*** join/#asterisk izo (n=izo@193.202.114.43)
03:42.51*** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net)
03:42.51*** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk)
03:42.51*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
03:42.51*** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net)
03:42.51*** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net)
03:42.52*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
03:42.52*** join/#asterisk trelane (i=trelane@66.93.203.199)
03:42.52*** join/#asterisk R3DB0x (i=nobody@66.142.28.36)
03:42.52*** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com)
03:42.52*** join/#asterisk Hmmhesays (n=Neg@24-117-213-113.cpe.cableone.net)
03:42.52*** join/#asterisk thuper (n=thuper@gateway.digium.com)
03:42.52*** join/#asterisk mattHelm (n=root@adsl-68-90-140-81.dsl.fyvlar.swbell.net)
03:42.52*** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss)
03:42.52*** join/#asterisk advorak (n=advorak@atypedigital.ischool.washington.edu)
03:42.53*** join/#asterisk hohum (i=corbe@snoop.burghcom.com)
03:42.53*** join/#asterisk Champi (i=Champi@damn.e-leet.be)
03:42.53*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net)
03:42.53*** join/#asterisk orlok (n=jwr@202-44-174-4.nexnet.net.au)
03:42.53*** join/#asterisk Beave (n=beave@vistech.org)
03:42.53*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it)
03:42.53*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
03:42.53*** join/#asterisk ]data[ (n=data@69.56.182.146)
03:42.53*** join/#asterisk Corydon-w (i=brown@pdpc/supporter/sustaining/Corydon76-home)
03:42.54blitz[laptop]mog_home: how far of a drive is KC to Huntsville?
03:42.54tzangerI'll meet ya there
03:42.54kuku5i have the biz plan
03:44.08iCEBrkrdude, did System() change in 1.2.0 beta2
03:44.12CoaxDif i wanna buy..say...a used car engine.. aside from ebay, where does one go?
03:44.12iCEBrkr?
03:44.33QwellCoaxD: pick a part?
03:44.57mog_homejunk yard coaxd
03:46.44MacRohardCoaxD, a scrapyard?
03:47.30*** join/#asterisk snitt (i=snitt@snitt.info) [NETSPLIT VICTIM]
03:47.30*** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) [NETSPLIT VICTIM]
03:49.22*** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net)
03:50.39tzangerCoaxD: scrapyard is my answer too
03:51.09LostfrogIMHE, used engines are bad news.
03:51.29iCEBrkroh WTF is this IAX / to - shit?!
03:51.49ManxPoweriCEBrkr, I think it's to make it consistent with the other channels
03:52.00mog_homenot always
03:52.06*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
03:52.10ManxPoweri.e. so you don't have to have one regex for IAX and another for any other channel name when parsing channel instances
03:52.11mog_homei have a friend that is good friends with a junkyard owner
03:52.18iCEBrkrAhh
03:52.18mog_homehe gets loads of cool stuff
03:52.21mog_homeeveryweekend
03:52.32harryvvwho does
03:52.36mog_homelike he recently got a 2000 dollar projector for 100
03:53.01harryvvCoaxD, uses engine for what reason
03:53.04iCEBrkrdoh
03:53.07iCEBrkrit's hang'n up on me.
03:53.18*** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi)
03:53.19harryvvmog, who does?
03:54.03mog_homewho does what
03:54.03harryvvget those prices
03:54.03*** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net) [NETSPLIT VICTIM]
03:54.03*** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) [NETSPLIT VICTIM]
03:54.03mog_homejunkyard
03:54.03harryvvat deep discounts
03:54.03*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) [NETSPLIT VICTIM]
03:54.04*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-14-134.cybersurf.com) [NETSPLIT VICTIM]
03:54.04Lostfrogok.. that's annoying.
03:54.08*** join/#asterisk Corydon-w (i=goldenro@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM]
03:54.08*** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) [NETSPLIT VICTIM]
03:54.08mog_homehad bad powersupply i think
03:54.08*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM]
03:54.19*** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) [NETSPLIT VICTIM]
03:54.19*** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) [NETSPLIT VICTIM]
03:54.21*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net) [NETSPLIT VICTIM]
03:54.21harryvvmog, a friend of yours got a projector at a junk yard for 100 dollars?
03:54.29harryvvmakes no sence.
03:54.30mog_homeyeah
03:54.32QwellThey have projectors at junkyards?
03:54.39mog_homethey have everything
03:54.44mog_homeit depends what kind it is
03:54.48Vcodead hookers, you name it..
03:54.54mog_homehe is outside of newjersey
03:54.56mog_homeso maybe
03:54.57LostfrogDead hooker? where?
03:54.57harryvvyou must be confused with electronic reclyclers right?
03:55.04Lostfroglol
03:55.16LostfrogI guess noone wants to explain Dr. Fun to me.
03:56.33iCEBrkrWow.. DBPut/DBGet got ugly
03:56.33harryvvthis h5n1 virus is very serios. Recent investigation dicovered that the 1918 flue pandemic was a bird flue.
03:57.08Vcotelecommuters of the world rejoice
03:57.28LostfrogWhy so?
03:57.40harryvvLostfrog, why so what?
03:57.59Lostfrog<Vco> telecommuters of the world rejoice
03:58.11*** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
03:59.13harryvvI was just watching a hour long special on the bird flue that is hitting south east asia. this is far more serios then I ever though. The h5n1 virus is constantly mutating and eventually, it will mutate into a human to human form.
03:59.32Vcoit'll mutate into human form?
03:59.41ManxPowerI hope tomorrow will be better than today.
03:59.45Vcolike walking and talking and shit?
03:59.46harryvvhuman transferable form of a virus
03:59.51Vcoi know i know
03:59.54iCEBrkrxten => s,1,Set(CIDNUM=${DB(callerid/number)})
03:59.57iCEBrkrThat's FUCKING UGLY
04:00.01Vcoit'd be cooler the other way
04:00.14ManxPowerToday started off with a BAT (Big Ass Truck) backing into the borrowed car, which we call the Death Mobile because it's brakes don't work well.
04:00.20harryvvit worrys me because we have a large chinese population here.
04:00.35Vco<in china>
04:00.41harryvvin vancouver bc
04:00.46ManxPowerThen the person who's car we were supposed to borrow had a family emergency so we could not borrow their car to go to the bank and go to the dealer and get a damn decent car.
04:00.47Vcoyou'r ein richmond right?
04:00.50harryvvno
04:00.53Vcooh...
04:01.09Vcowait...no ...that was soeone at the office today...
04:02.56ManxPoweriCEBrkr, no more ugly than exten => _XXXX,8,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?12:9)
04:03.31ManxPowerOr my fave: exten => _XXXX,13,GotoIf($[${LEN(${DIAL_DEST[${INDEX}]})} = 0]?14:5)
04:04.16*** join/#asterisk cripito (n=ncripito@c-67-161-130-59.hsd1.co.comcast.net)
04:04.18cripitohi
04:04.31ManxPowerI must admit using "subscripts" with variables in extensions.conf can rock.
04:04.48cripitodid anyone ever connect an oxygen pbx to asterisk?
04:05.05mog_homeoxygen pbx?
04:05.30cripitoyup 1 sec
04:05.50*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
04:06.11cripitoOOH323/AltiServ VoIP Line
04:06.11BrijnGood evening all
04:06.32iCEBrkrManxPower: It's just unfortunate that syntax kinda sucks ass.
04:06.32cripitoany way to known what codec is coming in h323?
04:06.35cripitosorry in ooh323
04:06.40iCEBrkrManxPower: But ya get used to it.. I guess
04:06.48mog_homewhere is oxygen pbx
04:06.57Kattyhihi
04:06.59ManxPoweriCEBrkr, soon you will see beauty in the chaos, grasshopper.
04:07.26cripitosnif snif
04:07.29cripitothat all what i got..
04:07.42*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM]
04:07.42BrijnKatty: You have Polycom SP 501's? Do you happen to have the Admin PDF? They are not avail for download anymore (none that Google turned up at least)
04:08.19cripitothis is what my asterisk is saying
04:08.21cripito23:38:06:774  ERROR:Incompatible audio capabilities - Can not open audio channel
04:08.21cripito<PROTECTED>
04:08.33KattyBrijn: what makes you think i have that information?
04:08.43iCEBrkrhehe
04:08.43KattyBrijn: google.
04:09.03BrijnI thought that you had 501's (or at least one), and was hoping you have the PDF
04:09.15KattyBrijn: we have 500s.
04:09.16BrijnThe lniks that Google turns up give 404's
04:09.36cripitoi think the pdf for 500
04:09.44cripitoi have that here
04:09.48Brijncripito: that would be nice!
04:10.12cripitoyeap i got userguide and adminguide
04:10.15cripitowhere u want it?
04:10.37Brijnmail, or if you have website/ftp close by, also ok
04:10.38ManxPowerBrijn, You mean like the ones here: http://www.freedomphones.net/polycom/files
04:10.59ManxPowerThe Wiki is your friend.  Love it, hold it, buy it flowers.
04:11.04BrijnUrmmmmmm
04:11.06cripito:D
04:11.07BrijnOoops
04:11.13BrijnThanx :)
04:11.15*** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net)
04:11.35cripitojust in case..;) i don't plan to delete those ones ;)
04:11.48LostfrogCan someone find me that question for 42? lol
04:11.59Lostfrogsix times nine..
04:12.00LostfrogI know.
04:12.03LostfrogSheesh.
04:12.05cripitook i need a h323 guru
04:12.08cripito;)
04:12.32cripitosnif snif this f&*&*# pbx don't connect
04:12.41iCEBrkrb00000000
04:12.42iCEBrkr<PROTECTED>
04:12.56Lostfrognice.
04:13.05LostfrogGuess they need a new module for 1.2.0.
04:13.09iCEBrkr:(
04:13.21iCEBrkrsize does matter! ;'(
04:13.25ManxPowerLostfrog, Should not need one.
04:13.27cripitoice did u put the g729 codec in the washing machine?
04:13.34iCEBrkrcripito: drier
04:13.37cripito:D
04:13.39LostfrogYeah.. it must of shrunk.
04:13.47ManxPoweriCEBrkr, everyone knows that!
04:14.10cripitowell manx it depends...
04:14.27cripitodepends who hold the short one ;)
04:16.12ManxPowerI downloaded and installed the G729 codec from Digium the other day into a CVS-HEAD of a month or so ago
04:16.15Kattyright. so when's hte next conference?
04:16.22LostfrogAnd where?
04:16.28cripitonext astricon?
04:16.32cripitoyeap...
04:16.36cripitogood to known
04:16.45Kattyfile: pester bkw about conference.
04:16.48cripitoplease denver.. please please
04:16.58LostfrogPlease East coast.
04:16.59Lostfrog:)
04:17.08cripitoand if we get another hurricane?
04:17.11cripitonaaaaaa
04:17.19cripitobetter center or montain ;)
04:17.30LostfrogI would like to eventually attend one, cripito.
04:17.32*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) [NETSPLIT VICTIM]
04:17.36Corydon76-homeYeah, next conference in Huntsville!
04:17.36iCEBrkrwelp, ulaw will have to do for now
04:17.47ManxPowerI suggest Birmingham or Atlanta
04:18.00FuriousGeorgehey all
04:18.01LostfrogGeorgia would be cool.
04:18.07cripitoyeap..
04:18.17cripitostill prefer colorado to georgia...
04:18.22iCEBrkrGeorgia and Alabama are both in driving range for me.. :)
04:18.35LostfrogIt was in Atlanta in 2004
04:18.40hypa7iatoronto!
04:18.42cripito:) well driver range.... is also from me
04:18.51hypa7iaastricon 2006 is going to be in baaaahston
04:18.52cripitojust 900 miles away
04:18.53Kattyit will probably be in chicago or vegas guys.
04:19.00hypa7iai think i recall oej saying
04:19.07cripitoVEGAS!!!
04:19.24harryvvOr vancouver BC the most livable city in the world.
04:19.42LostfrogBoston would be cool too.
04:19.48Juggieummmmm
04:19.52Juggieastricon 2006 is in dallas
04:19.54harryvvunfortunaly that is driving our realestate cost though the roof.
04:19.58Juggieit was announced already
04:20.00cripitoi am in the next even if i have to walk to the conferece
04:20.18iCEBrkrshow g729
04:20.18iCEBrkr0/0 encoders/decoders of 2 licensed channels are currently in use
04:20.19iCEBrkrw00t
04:20.26KattyJuggie: i'm talking about cluecon :)
04:20.27cripitonice ice
04:20.31ManxPowerI just got an e-mail that I can't decide if it's spam or a comment on one of my web pages.
04:20.43cripito:D
04:20.45LostfrogIt can't be both?
04:20.53FuriousGeorgeManxPower: probably just clever spam
04:21.04harryvvim off
04:21.06cripitohi george...
04:21.14cripitommm that's was fast.
04:21.25FuriousGeorgecripito: wadup
04:21.31Kattytwisted[asteria]: can i drive to your house and then you drive to dallas for astricon next year?
04:21.40cripito:D
04:21.49ManxPowerFuriousGeorge, I'm not sure.
04:22.27Kattytwisted[asteria]: are you still at work?
04:22.35Kattytwisted[asteria]: don't make me get my nextel and beepbeep you
04:22.42Lostfrogwow.. /me beats his boss till he pays his $3000 + dCAP fee and plane ticket and hotel to Miami.
04:23.00blitz[laptop]hypa7ia: Astricon 2006 is in Dallas, TX
04:23.00FuriousGeorgeManxPower: my mom got a bug in her laptop somehow that read her mail in ms office and sent her chopped up pieces of mail meant to look like actual correspondence.  was eerie
04:23.04LostfrogHe'd probably die before he paid.
04:23.09hypa7iaooh
04:23.13cripitofuriosgeorge: no much.. trying to figure out how i can delete 3 querys that are the same in a realtime extension call
04:23.16hypa7iai stand corrected :)
04:23.28iCEBrkrshit, I forget how to tell which codec a call is using
04:23.28blitz[laptop]I wish it was in Boston than Dallas
04:23.28FuriousGeorgecripito: I give up :)
04:23.32cripitosnif snif 3 mega bucks
04:23.38blitz[laptop]I'm not such a fan of the red states :)
04:23.40Lostfrogblitz[laptop]: me too.
04:23.46cripitoi think can be done...
04:23.50file[laptop]CANADA ASTRICON!
04:23.56blitz[laptop]file[laptop]: no kidding
04:24.07FuriousGeorgeis it in that part of canada with the legal pot?
04:24.14ManxPowerGads, no.  If Astricon was in Canada I'd not spend any time at the conference.
04:24.17blitz[laptop]there is no legal pot part
04:24.35FuriousGeorgeblitz[laptop]: err, strongly decriminalized
04:24.37blitz[laptop]although BC is very liberal about it
04:24.41Juggiepot isnt lega
04:24.43Juggieer, legal
04:24.46Juggiebut it isnt illegal eiher
04:24.49cripitoby the way anyone known what is the max g729 channels that a soekris card can handle?
04:24.51blitz[laptop]FuriousGeorge: its not decriminalized either
04:24.55file[laptop]blitz[laptop] is an expert on this stuff
04:25.01ManxPowerFuriousGeorge, Canada can't legalize pot because of international treaties, they CAN simply not enforce the laws that make it illegal.
04:25.09FuriousGeorgeblitz[laptop]: is it even encouraged?
04:25.15blitz[laptop]FuriousGeorge: I'd not say encouraged :)
04:25.22ManxPowerWhich is annoying, because when the political climate changes they can start enforcing those laws again.
04:25.27FuriousGeorgeor is its illegality merely a suggestion
04:25.27blitz[laptop]but no one really seems to care... depending on who you hang out with
04:25.29Juggieyou will not go to jail for smoking pot
04:25.38Juggieselling 50,000$ worth
04:25.39Juggieyes
04:25.47blitz[laptop]you go to jail for dealing... not possession
04:26.02Juggieits not really, illegal to smoke it, but its illegal to buy it
04:26.04Juggiego figure :)
04:26.07blitz[laptop]technically you COULD go to jail for possession... but no cop is going to both you unless you're smoking it on the street and blow it in his face
04:26.07FuriousGeorgeso i was installing ntp-client today and noticed its been on version 4.2.0 for 18 months
04:26.26hypa7iablitz[laptop]: even then, you'd have to get spittle on him while blowing it
04:26.30hypa7ia-_^
04:26.31blitz[laptop]lets just say its low priority for cops unless you own a grow house or something
04:26.32hypa7iathat sounded bad
04:26.33FuriousGeorgecoincidence?  I think not
04:26.45blitz[laptop]lol
04:26.55blitz[laptop]I filed bug 420 in the bug tracker :)
04:27.03hypa7iahaha awesome
04:27.03cripitosheez i am lost here.. pot and ntp-client don't mix well
04:27.08FuriousGeorgeblitz[laptop]: stoner
04:27.14blitz[laptop]FuriousGeorge: I've never smoked it
04:27.22blitz[laptop]I just know lots of people who do
04:27.33FuriousGeorgeblitz[laptop]: not that you remember anyway
04:27.39blitz[laptop]well... i tried it once, but I didn't inhale
04:27.42Juggietheres alot of other things cops are worried about
04:27.44Juggiebesides pot
04:27.46blitz[laptop]smells too much like skunk
04:27.49FuriousGeorgeblitz[laptop]: brownies?
04:28.03blitz[laptop]yah... brownies rock
04:28.12blitz[laptop]never had special brownies though
04:28.16Juggiehah
04:28.23FuriousGeorgeand they can't be inhaled.  trust me, ive tried
04:28.24Juggieblitz[laptop], only smokes pole.
04:28.33Juggie:P
04:28.35blitz[laptop]Juggie: ass
04:28.35ManxPowerAny sane person knows how stupid the drug laws are and any sane person knows the "War on (Some) Drugs" has been a miserable failure.
04:28.36blitz[laptop]:)
04:28.55blitz[laptop]ManxPower: the war on drugs keeps people employed... and helps keep people in fear
04:28.59FuriousGeorgepots a gateway drug so blitz[laptop] went straight to crank
04:29.08blitz[laptop]hehehe
04:29.09ManxPowerblitz[laptop], In that respect it's a success.
04:29.29JuggieManxPower, what ammuses me more is 'the war on terror'
04:29.42blitz[laptop]"the war on f00"
04:29.46ManxPowerJuggie, that just scares me.
04:29.47Juggiecanada is all up in arms about a gov scandal which fucked around with 200million
04:29.51blitz[laptop]substitute f00 for whatever you want
04:29.56Juggieyet, bush blows TRILLIONS on iraq
04:30.13blitz[laptop]Juggie: that doesn't make the 200 million right though :)
04:30.24Juggieblitz[laptop], no, but it puts things in perspective
04:30.29blitz[laptop]it can
04:30.37Juggiewe are cool up here ;P
04:30.45blitz[laptop]I'm a liberal though... and I'll continue to vote liberal
04:30.54blitz[laptop]Juggie: damn straight -- we rock :)
04:30.55blitz[laptop]lol
04:30.56Juggieconservative ;)
04:30.59Juggiebut i don tvote
04:31.01Juggie*vote
04:31.04blitz[laptop]Juggie: and I thought we could be friend
04:31.07blitz[laptop]s*
04:31.11blitz[laptop]Juggie: good -- keep it that way
04:31.14cripitoAltiServ from altigen... did anyone connects that to asterisk
04:31.25Juggietheres no point
04:31.34Juggiegoverment is fucked no matter who they are
04:31.38hypa7iatrue :)
04:31.48cripito:)))))
04:31.48Juggieas long as its not the bloc, its fine by me :P
04:31.52hypa7iahaha yes
04:32.05cripitoi can't vote .. :P so anyway....
04:32.12hypa7iaeven then... if they want to go that bad, they're the ones who fuck themselves over
04:32.14Juggiehypa7ia, i finally got a pika board working in *
04:32.28Juggieit works ok i guess, for analog
04:32.34hypa7iacool
04:32.35Juggiei'm used to pri now, so i'm spoiled.
04:32.42hypa7iahehe
04:32.54cripitoany driver accesible for a dialogic card?
04:32.56Juggiei like instant gratification ;)
04:33.06hypa7iai may do a key system -> asterisk -> voip termination setup for my dad's law firm
04:33.06Juggiecripito, yes, if you buy * business edition
04:33.12cripitonaaaaaaaaa
04:33.14Juggieit contains a dialogic driver
04:33.19cripitosnif snif
04:33.24Juggieand this is due to intel licensing requirements
04:33.32cripito995 bucks only b/c a driver?
04:33.39ManxPowerYou can, in theory, but the Dialogic driver from Digium
04:33.39cripitoi need to find another wya
04:33.41cripitoway
04:33.42digimemost stable veresion of asterisk now? 1.0.9???
04:33.44ManxPower$10/channel
04:33.51Juggieyah, you can try and buy it without the business eddition
04:34.00Juggiei might try and buy an unlimited license
04:34.06Juggieif they wil sell me one
04:34.09digimewhat version of * are people running in production?
04:34.14blitz[laptop]hypa7ia: I'm actually a socialist -- but unless I know the NDP can win my area, then I vote liberal
04:34.16cripitoso a quad card apart of the 7K from intel is more or less 960 extras
04:34.23blitz[laptop]as long the conservatives don't win, thats all that I really care about
04:34.26Vcoanything but harper
04:34.27blitz[laptop]but I'd like the NDP to win.
04:34.31hypa7iablitz[laptop]: okay, i can like you again :-p
04:34.36iCEBrkrWell, shit, I can't dial out of Voicepulse.
04:34.43blitz[laptop]hypa7ia: haha... oh yah... I'm totally a socialist :)
04:34.47Juggiecripito, i dont understand the question
04:34.50hypa7iai've been a dipper since i was like 16
04:35.02hypa7iaworked on like 5 campaigns
04:35.03Vcoo.O
04:35.04cripitowas a comment more than anything juggy
04:35.20Juggiehypa7ia, i'm glad i know your a girl
04:35.32Juggiebecause if a guy had said "i've been a dipper since i was 16"
04:35.35Juggiei'd be concerned :)
04:35.41blitz[laptop]LOL
04:35.43hypa7iahehe actually i'm going to try and leverage my political connections to get another user group some space
04:35.53hypa7iai'm not sure what else "dipper" means
04:35.58hypa7iadon't think i wanna know :-p
04:36.04iCEBrkrNov  3 23:35:54 WARNING[11951]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses!
04:36.06Juggie:P
04:36.08iCEBrkrWHAT THE MOTHERFUCK
04:36.09Juggiebum dipper :P
04:36.12blitz[laptop]hypa7ia: I'm sure you could figure it out if you really wanted to :)
04:36.35hypa7iahahaha
04:36.39blitz[laptop]hypa7ia: the one thing I like about the minority gov't is that it gives the NDP power
04:36.44Vcoanyone use DID's from Voxbone ?
04:36.51blitz[laptop]hypa7ia: so the NDP actually gets shit done
04:36.51Vcostill a terrible name if you ask me....
04:36.53hypa7iablitz[laptop]: me too :)
04:37.00Juggieblitz[laptop], did you hear layton's latest thing
04:37.07blitz[laptop]Juggie: no.. what did he say?
04:37.14Juggiehe wants some shit done to health care
04:37.15blitz[laptop]wicked :)
04:37.26Vcohe was actuallly pretty snappy on that
04:37.30blitz[laptop]yah, he wants a law put in that forbids privitazation of health care
04:37.31Vcototally out of character
04:37.40blitz[laptop]if I understand correctly
04:37.40hypa7iaoh man speaking of
04:37.41Juggieif the liberals dont comply, he'll partner with conservative to topple the liberal minority
04:37.51blitz[laptop]oh yah... he has a lot of power right now
04:37.52Vcowell..not 'snappy' , just super asseritve
04:37.52Juggiewhich should really be done anyway
04:37.54Vcohaha
04:37.54Juggieto clean house
04:37.55hypa7ianow that is strong-arming :)
04:38.03hypa7iarox0rs
04:38.09cripitochan_ooh323 is version h323v1 or h323v2 ?
04:38.12blitz[laptop]Juggie: guh.... the conservatives scare the hell outta me
04:38.29hypa7iamy parents' friend went to a private clinic in montreal for some experimental back surgery... no waiting lists
04:38.33hypa7iamakes me sad :(
04:38.45blitz[laptop]yah... its too bad people with money can do whatever they want
04:38.46Vcoremember when that liberal dude tossed him that insult when he was at parliament..
04:39.00Vcobasically that he might be the leader of the party , but he doens't even have a seat there yet..
04:39.46blitz[laptop]private clinics should be taxed heavily and that money put DIRECTLY back into public health care
04:40.25hypa7iayeah
04:40.29Vcojust like tobacco tax should ACTUALLY go back and (i think) go to smoking cessation stuff
04:40.37hypa7iaor we should just fix the public system
04:40.43hypa7iaand/or that is :)
04:41.03hypa7iainstead of spending $18 jillion on cisco callmanagers at the hospitals
04:41.08Vcoheh..heh..
04:41.08blitz[laptop]duh
04:41.09hypa7iawhich is what they're doing :(
04:41.19hypa7iaasterisk needs to talk to nurse-call systems
04:41.30iCEBrkrThis g729 shit has been pissing me off. I have YET to make the shit work and I bought a second license just to make sure I had all my bases covered.
04:41.33Vconevermind the ludicrous money they spent at UBC and VGH just a couple of years back
04:41.51file[laptop]I think I'm going to go to sleep
04:41.55Vcodoign deployment work, they bought a pile HP 4050n
04:41.55hypa7iaVco: university health system in toronto was $5M of cisco gear
04:42.00blitz[laptop]I think I'm going to visit the garage
04:42.02blitz[laptop]then go to sleep :)
04:42.09hypa7iathe "garage"
04:42.17Vcothere was 1 $3000 laser printer for every 2 patients in the hospital
04:42.24hypa7iacripes.
04:42.28blitz[laptop]hypa7ia: I'm actually going out to the garage :)
04:42.32Vcoevern the nurses were complaing they didn'tneed these things
04:42.33blitz[laptop]"what do you call it?"
04:42.39blitz[laptop]"a car hold"
04:42.43Vcohahaha
04:43.25hypa7iacarport
04:43.39Juggiehah
04:43.40hypa7iaVco: the cisco wifi stuff actually gets well used
04:43.43Juggiereminds me of the simpsons
04:43.52Juggiehypa7ia, sure i have 1.5 million sitting on a lab floor
04:43.59cripitoyes Altigen
04:44.00blitz[laptop]Juggie: uhhh... yah... thats kind of what I was trying to imply :)
04:44.03Juggiei have a 24port poe switch on my desk :)
04:44.10Juggieso i can power my phones :)
04:44.12Vcoi have a dead hooker on my desk...
04:44.23Vcoanyone have a van and some carpet they want to get rid of?
04:44.30JuggieVco, your approaching it all wrong, what you want is a live hooker under your desk
04:44.40Vcowell..
04:44.42cripito:)))))))))))))))))))))))))))))))))))))
04:44.47Vcoi've figurd that out now
04:44.51*** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it)
04:44.59cripitook altigen.... and asterisk
04:45.06cripitoi can call to altigen
04:45.13cripitobut those F*&*#$@#$@#)($)@#$ can't call to me
04:45.27QwellSo, sccp is actually kinda nice
04:45.28Vcois this with ooh323?
04:45.33Vcocripto
04:45.37Vcocripito
04:45.37cripitoyeap
04:45.39Vcorather
04:46.10cripitoi am thinking to install h323 and openh323 just to see changes anything
04:46.10Vcocall to your * box, or to a 323 client?
04:46.11cripitobut i doubt it
04:46.28*** join/#asterisk XTR (n=xtr@staff-nat.netnation.com)
04:46.37cripitoi can call from my sip phones in * to the phones in altigen
04:46.43cripitobut altigen can't send a call to me
04:46.56PazzoHi guys! I've had a self-packaged 1.2.0-Beta1 working fine (together with ser, mediaproxy etc). With 1.2.0-Beta2 (package from debian experimental) I run into trouble: everything seems to work fine, but I cannot hear any sound produced by asterisk!? (echotest etc)
04:47.06Pazzoany idea? similar problems?
04:47.07Vcoyea, i can't ring any h323 clients but they can call out fine..
04:47.08cripito23:38:06:774  ERROR:Incompatible audio capabilities - Can not open audio channel
04:47.08cripito<PROTECTED>
04:47.18Vcogoes straight to unavailable.
04:47.26Vcoeww....totally different issue than me
04:47.28Pazzobtw: no errors, even with -rvvvvvv
04:48.15cripitoso vco they can call your *?
04:48.25cripitocan we check versions , settings etc?
04:48.28Vcook, so not a lot of feedback for people that use voxbone, anyone knwo where else to find Japan DID's?
04:48.57cripitothere was something from didx.org offering that...
04:49.10Vcodidx bough the japan ones from voxbone
04:49.13Vco:/
04:49.15*** join/#asterisk nobell (n=jdegraff@70.103.228.158)
04:49.19cripitobut didx don't have a precisally good critic
04:49.26Vcothe h323 clients can call my * box, or route out over IAX etcetc....
04:50.06cripitoyeap man.. that's exactly what i need.. but if i connect the phone h323 to my server words fine...
04:50.14Vcobut i can't  Dial(H323/soemuser)
04:50.14cripitomagic...
04:50.27cripitousing chan_ooh323?
04:50.30Vcoya
04:50.40Vco"oo"h323
04:50.44cripitodid u try dial(OOH323/soeuser)?
04:50.49Vcoyea..
04:50.53Vcochannel unavailable.
04:50.55iCEBrkrmake my g729 go!!!
04:51.05Vcoshow channeltypes
04:51.11Vcolists it as H323
04:52.10Pazzohmmm... no one running 1.2.0 beta2??
04:52.15cripitoFeature     Feature Proxy Channel Driver   no           yes          no
04:52.15cripitoOOH323      Objective Systems H323 Channel no           yes          no
04:52.31Vcothis japan DID i still haven't figured out if it's potentianlly something on my end or what....calls come in, rings a sip phone, i answer the phne i hear busy ..but the calling party is still ringing..
04:52.46Vcohang up phone, rings right away again..
04:52.53Vcoanswer, it's busy..
04:53.27Vcothing it , it used to work, anbd i have other DID's with the same config and it all works fine..
04:53.49Vcoi'm jsut the provider for the .jp one is fubar
04:54.00Vcoi'm just guessting that the....
04:54.06Vcoguessing even...
04:54.09Vcodamn
04:54.24Vcoi need to start using keyboards with the same layout..
04:54.36*** join/#asterisk kvit (n=kvit@203.209.56.65)
05:00.17*** join/#asterisk Baph (n=Dave@dirobertson.plus.com)
05:02.51*** part/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET)
05:03.04iCEBrkr*CLI> Nov  4 00:02:43 NOTICE[12528]: chan_iax2.c:7746 iax2_poke_noanswer: Peer 'vpconnect-t01' is now UNREACHABLE! Time: 0
05:03.08iCEBrkrANything else?!
05:03.24VcoTime: 0   ?
05:03.50iCEBrkrI've never had issues with VP like this.
05:03.57iCEBrkrUnless there were hurricanes :P
05:05.54iCEBrkrvoicepulse-in-0  66.234.228.170  (S)  255.255.255.255  4569          OK (49 ms)
05:05.57iCEBrkrvpconnect-t02    66.234.228.166  (S)  255.255.255.255  4569          OK (46 ms)
05:06.00iCEBrkrvpconnect-t01    66.234.228.160  (S)  255.255.255.255  4569          UNREACHABLE
05:06.03iCEBrkrI love it
05:06.19*** join/#asterisk Eyecon (n=matt@jumala.plus.com)
05:11.19*** part/#asterisk bancus (n=treed@athena.crschmidt.net)
05:13.08*** join/#asterisk syle (n=blah@unaffiliated/syle)
05:27.52*** join/#asterisk infinity1 (i=brendon@solara.netcal.com)
05:28.00infinity1err ..anyone use the TORTURE feature?
05:28.16*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net)
05:29.19*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
05:29.33shmaltzhow does robbed bit t1 signal DID?
05:31.43infinity1how does the TORTURE status for the Dial application work?
05:33.46*** join/#asterisk t0p (i=t0p@202.8.86.162)
05:38.39Corydon76-homeshmaltz:  depends upon the signalling
05:40.46*** join/#asterisk Splas (n=sp@brooklyn.paravolve.net)
05:46.50*** part/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
05:58.41*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:01.27*** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com)
06:07.07*** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
06:07.16Qwellheadline: 78 year old woman talking on her cellphone while driving makes an illegal right turn at a red light, and gets hit by a bus.
06:07.22QwellAnybody see anything wrong with this picture?
06:07.39Qwell(or, more specifically, 4 things?)
06:08.34Netgeekshrm, 1. H in headline not capitalized
06:08.59hypa7iawrong side of the road unless it's english
06:09.01Netgeeksbut I don't think you wanted to know the 4 things wrong in your typing...
06:09.50_m_hypa7ia: or australian (or some other leftdriving place)
06:10.00Qwellnah, US
06:13.45*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
06:23.31*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
06:25.24*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:27.25*** join/#asterisk kaldemar (n=kalde@130.233.228.9)
06:33.26FuriousGeorgei use eyebeam and tbh i dont really like it.  i like how firefly loosk but afaik it can only handle one call at a time
06:33.48FuriousGeorgecan someone recommend a softphone for me to try with the little things, like support of multiple connections and perhaps a xfer button
06:37.58infinity1FuriousGeorge: i use eyebeam. seems okay.
06:37.58*** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
06:38.56infinity1its quite here lately
06:39.06infinity1whats the deal? everyone busy deploying beta2?
06:42.27*** join/#asterisk lehel (n=lehel@82.79.20.17)
06:42.59lehelmornin'
06:44.30FuriousGeorgeinfinity1: you know what bothers me about eyebeam?  it uses that simple messaging and i want to eventually use XMPP for messaging, so it annoys me that the "send an instant message" option wont go away
06:44.39*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
06:45.28*** join/#asterisk Splas (n=sp@brooklyn.paravolve.net)
06:46.05FuriousGeorgeid love to get my hands on a copy of X-Pro
06:47.06infinity1FuriousGeorge: url?
06:47.51infinity1simple vs XMPP? duno ...i just use the sip functinoality.
06:50.24FuriousGeorgeinfinity1: there is astjab.org and there is asterisk-im from jivesoftware.com
06:50.33FuriousGeorgexmpp=jabberd
06:50.55infinity1FuriousGeorge: i think i'm using X-Pro actually. you mean the commercial x-lite?
06:51.19FuriousGeorgeinfinity1: right, but eyebeam supports video (which could be cool) and messaging (which i cant disable)
06:51.42Vcoplus g729 ..no?
06:51.43FuriousGeorgeit just bothers me that even though i disable presence and dont implement it on the server it still gives the user the option
06:52.26FuriousGeorgeg'night all
06:52.33infinity1l8
06:53.13infinity1FuriousGeorge: it appears the version of x-pro i'm using doesn't support messaging. i can't find an option fori t
06:57.50*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
07:15.38*** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de)
07:22.41hardwiredrumkilla: pinch poke
07:26.36NetgeeksI don't think he's alive, I tried him an hour ago
07:26.53hardwiremofo made a completely funky comment on one of my bug reports
07:26.57*** join/#asterisk Poincare (n=jefffnod@dD577A88C.access.telenet.be)
07:27.06hardwireI strive to understand it.. but I don't think I am supposed to know its real meaning :)
07:27.52Netgeeksoooooh, which one!  I want to read it!
07:28.18*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
07:28.21hardwirehttp://bugs.digium.com/view.php?id=5338
07:29.26Netgeeksheh
07:29.45NetgeeksThis option is not present in 1.0
07:29.46hardwirewhat does that even mean?
07:30.03hardwireis it some sort of crazy statement I should spend the rest of my life studying?
07:31.07NetgeeksI think he's trying to say that the option 's' for chanIsAvailable doesn't exist in the 1.0 branch
07:31.50hardwireso I am not insane
07:32.01NetgeeksI know this because I slept in a Holiday Inn express last night
07:32.08hardwireheh
07:32.20hardwireI hate those commercials
07:32.34Netgeeksand, no, you are incorrect, we are all insane...
07:33.08hardwireso.. why he is posting that about a 1.2.x fixup?
07:33.19hardwireif s doesn't exist in 1.0.x whats the whoop?
07:33.22NetgeeksI'm beating myself to pieces trying to figure out why I'm getting this:
07:33.27NetgeeksNov  3 19:13:34 WARNING[7850]: channel.c:713 channel_find_locked: Avoided deadlock for '0x6000000000154840', 10 retries!
07:33.33Netgeeksand it's got me very insane
07:34.00NetgeeksI dunno what the whoop is... maybe whardier or someone asked him about it on irc and he added a note to the patch so they would read it
07:34.22hardwirewhardier == hardwire
07:34.46Netgeekshrm, well now, I guess you would know if you asked in irc then
07:34.51hardwirepossibly
07:34.55hardwireok I am good now
07:34.59hardwireits meant to be confusing
07:35.01Netgeeksunless you are of the multiple personalities type of insane
07:35.07hardwireyeh. no
07:35.10Netgeeksthen you may not know if you asked
07:35.12Qwellhardwire: It's because twisted set it to Pending Stable
07:35.18Netgeeksunless you talk to your selves
07:35.32QwellSo, drumkilla says "Not in 1.0.  Status => Closed."
07:35.34hardwireQwell: so pending 1.2.0 stable has to address 1.0 stable changes?
07:36.31Qwellwas just a simple mistake... drumkilla wanted to clarify that when he closed it
07:37.00hardwireit was good enough to cause me a wonderin.
07:37.02hardwireso.. oh well
07:37.15kuku5Anyone here storing credit card numbers?
07:37.30hardwireif they are they should be shot :0
07:39.11NetgeeksI store mine in my wallet on pieces of plastic
07:39.30kuku5:)
07:39.33*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
07:39.42*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
07:39.48NetgeeksI doubt thats what you were looking for, however  :)
07:40.15Netgeeksif you find my locked channel, let me know, hardwire
07:42.26*** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-50.cst.dccnet.com)
07:42.46kb1_kanobeevening all.
07:44.53steffhi all
07:47.50*** join/#asterisk bon (n=bon@217.172.148.5)
07:47.50bonhey
07:47.55bonanyone using realtime aroung?
07:47.56bonaround
07:47.57boneven
07:47.59bon:)
07:48.08bonUnable to load module cdr_addon_mysql.so
07:48.17bonNov  4 08:47:43 WARNING[24690]: loader.c:314 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directory
07:48.20bon:(
07:48.29bonrerun ldconfig, mysql lib dir in ld.so.conf
07:48.35bondon't know what else to do
07:49.59QwellDid you change mysql versions or something?
07:50.09Qwelllooks like it's linked against 4.1.15
07:52.21Qwelloff to bed
07:52.37*** join/#asterisk grimse (n=grimse@p5481DDC5.dip.t-dialin.net)
07:52.52bonQwell: well fresh install mysql5
07:53.07bonthere has been no mysql4 before on that machine
07:54.57steffsomebody using AVM C2 controller or other dual bri card?
07:56.56steffanyone?
07:57.08kb1_kanobeBehaviours you don't want to see at 11:30pm - playing a freshly recorded wav file oopses the Asterisk server... :-/
07:57.54Dr_Ray<PROTECTED>
07:58.20kb1_kanobeThat's true, but like every third person I meet in here at this time of night, I just want to go to bed... :-)
08:00.28Dr_RayI'm one of the other 2 :)
08:00.45Dr_RayI hate having to wait for calls to stop before I can reload the asterisk server
08:01.12kb1_kanobeAgreed.
08:02.05kb1_kanobeAlas, my issue appears to have been filesystem corruption, not asterisk... and now I have a mess on my hands.
08:04.14*** join/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net)
08:05.05*** join/#asterisk tryfoss (n=morten@80.239.93.22)
08:10.19steffso... anyone use chan_capi ?
08:11.44steffnobody :-(
08:11.49lehelsteff: me
08:11.58steff:-D
08:12.28lehel? ;)
08:12.37stefflehel: my * box work with only 2 bchan, i can't get the 4 bchan working
08:13.17lehelme neither steff
08:15.06zoasteff
08:15.11zoaim using a c4
08:15.23steff(i'm french, my english is poorly), i need some time to understood
08:15.31steffi have a C2
08:15.42zoaah mais je parle aussi le francais
08:15.44zoapas de probleme
08:15.52steffsuper
08:15.55skyenmois nonplus
08:15.57skyen;)
08:16.02zoala meilleure chose a prendre, c'est chan_misdn pour l'instant
08:16.24zoaou bristuff (si les drivers marchent pour votre carte)
08:17.21steffje vaistester avec chan_misdn,
08:17.44steffcela proviens de chan_capi le blocage a 2 bchan alors?
08:17.46skyenwhat does vaistester mean?
08:17.56skyeni'm getting the rest
08:17.56zoaoui je pense
08:18.10zoaits "je vais tester"
08:18.12zoai will test
08:18.20stefftring to test -je vais tester)
08:20.32steffthere was a version of chan_misdn know to work with avm ?
08:21.07steffi try 0.2.0
08:21.52zoayes
08:21.54zoait works
08:22.09zoatake chan_misdn from asteriskbeta2
08:22.11zoaits included
08:22.14zoano need for patches
08:30.07*** join/#asterisk stoffell (n=stoffell@59.46-201-80.adsl.skynet.be)
08:30.08kb1_kanobeany sox wizards around? I need to convert from prerecorded gsm files to asterisk-compatible 'wav' files.
08:30.36kb1_kanobeplain old 'sox ...gsm ...wav' results in 'not a wav49' in the console.
08:31.01kb1_kanobeBut I was expecting * to be looking for a 16bit linear, not a wav49?
08:32.15*** join/#asterisk BleedingMe (n=Bleeding@adsl-69-227-208-235.dsl.scrm01.pacbell.net)
08:32.34kb1_kanobehuh - asterisk-users is showing up in Google Groups.
08:33.29lehellong time kb1_kanobe
08:33.42kb1_kanobeI'm just a little behind the curve is all. :-)
08:33.50lehel:P
08:34.16*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:34.22*** join/#asterisk vinko (n=zic@208.5.87.254)
08:34.26lehelkb1_kanobe, u've dealing with fax?
08:34.43kb1_kanobenot really, I do some detect and passthrough.
08:36.09af_what dows the "n" priotity in the cvs asterisk?
08:36.14af_does, even :)
08:36.49*** join/#asterisk vinko (n=zic@208.5.87.254)
08:37.14Dr_Rayit's N for next, it replaces 1,2,3,4 in the extensions.conf
08:37.37af_oh I see: like go to the next statement in this extension?
08:37.49skyenwhat about the +101-feature?
08:38.18Dr_RayI think it is more for picking oreder
08:38.23Dr_Raynot teh goto
08:38.26Dr_Raybut maybe
08:38.30*** join/#asterisk many (n=many@daheim.ukeer.de)
08:38.48*** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net)
08:41.09tryfossis it normal that the cdr from a call going out from a queue to an agent is logged with billsec=0 ?
08:54.21*** join/#asterisk shido6 (i=shido6@d221-68-216.commercial.cgocable.net)
08:57.35*** join/#asterisk shanky (n=shanky@82.159.214.9)
08:58.25shankygood morning
09:05.57pooh_morning
09:06.39*** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee)
09:10.50*** join/#asterisk Ariek1 (n=ariek@84-245-28-221.dsl.cambrium.nl)
09:17.59*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
09:20.41lehelphonenr=faxnr, could anyone look at my CLI, why am i unable to send fax?
09:20.43lehelhttp://pastebin.ca/27580
09:21.35*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
09:23.26pooh_lehel: Sending 58740 on 2/3 to ipaddress:4569
09:23.33pooh_lehel: where is the REAL address ?
09:29.58Delvarquick question do digium g729 licances expire after a period of time/
09:30.31zoano they dont
09:31.33shido6:)
09:33.28kb1_kanobeTime to refrest the asterisk/festival interface - http://cslu.cse.ogi.edu/tts/flinger/
09:34.39kb1_kanobes/refrest/refresh
09:34.57mazzanetis it possible to play say an mp3 down a current open sip channel?
09:36.00lehelpooh_: the real address? i replaced the address with "ipaddress"
09:37.13Delvarcol thanks
09:38.12*** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com)
09:39.18*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.171.106)
09:39.24Delvarweird this box, been working fine for months, now g729 doesnt seem to work at all
09:39.46MuppetMasterHelllo
09:39.49*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
09:40.11MuppetMasterI have just started receiving a new error after doing some upgrades to v1.2beta2:  http://pastebin.ca/27581
09:40.19MuppetMasterAnyone else seen this, know how to resolve?
09:41.06Dr_Raywhich codec?
09:41.33MuppetMasterG711 beteween an Asterisk and a Sipura
09:42.36shankyanyone know the pinout schema for a E1 connector RJ45-BNC ? (I'm trying to connect a TE410P to BNC males connectors)
09:42.43shido6i used grubinstall
09:42.46shido6oops
09:43.18Ariek1does anybody know some info about the mixmanager
09:43.26Ariek1or where i can find it
09:49.52*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:50.29*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
09:50.45*** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-169-66.44-151.net24.it)
09:51.28BladeRunner05Hi all anyone has used Controller ISDN AVM B1 v.4.0
09:55.23lehelNov  4 12:56:26 DEBUG[25577] chan_zap.c: Monitor doohicky got event Event 160 on channel 2
09:55.38lehel<doohicky ? what this supposed to mean?
09:55.44*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
09:55.59lehelBladeRunner05, i'm using fritz
09:57.30*** join/#asterisk ful|work (n=fulgas@209.8.233.67)
09:57.43*** join/#asterisk jojo (n=jojo@c83-253-39-20.bredband.comhem.se)
09:57.52ful|workhey
09:58.47BladeRunner05lehel: the avm b1 card ?
10:05.16*** join/#asterisk Red5 (n=Red5@sodor5.ethertech.com.au)
10:07.26*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
10:08.49BladeRunner05lehel: r u there ?
10:09.39*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
10:10.07lehelBladeRunner05: AVM FRITZ!Card PCI < this one
10:12.09Red5hey all.. quick question. how can i place a direct internet call from my sipura to a friends phone via IP address ?
10:12.47Red5its not possible is it ? unless he's registered with someone..
10:15.01RaYmAn-Bxit is possible :) Assuming he has a SIP soft or hardware phone listening on that ip address...(check your sipura manual on how to dial ip's directly)
10:15.11RaYmAn-Bxobviously you can't dial his normal phoneline by ip address
10:15.46Red5i understand that... but if my sipura is registered with asterisk dont i need a dialplan for ip based calls too ?
10:16.03BladeRunner05Lehel: which kernel r using ?
10:16.51RaYmAn-BxRed5: it depends..Sipura can (if setup correctly) dial directly to an ip address..So it simply just doesn't let the call setup go through the register (=asterisk)
10:17.16Red5what about a 7940 ?
10:17.44RaYmAn-Bxnever used that so I have no idea..I'd assume it would be possible though..Check documentation :)
10:17.50Red5ok.. ta..
10:19.53*** join/#asterisk RGi_- (i=RGi@62.97.247.44)
10:20.33RGi_-anyone here using asterisk@home ?
10:20.49shido6used to
10:20.51shido6:)
10:21.01*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
10:21.30RGi_-I have problem making Digital Receptionist recordings... getting an error when it tries to copy the wave file...
10:22.39puzzledanyone seen this error when compiling ztdummy and know how to solve it?  error: #error ztdummy requires 1000 hz jiffies
10:23.38lehelpuzzled: my ztdummy works fine now;) and not seen that err
10:24.03puzzledgood for you :)
10:28.44heroinehi
10:35.59lehelahh, i can't manage to send a fax;(
10:37.57ful|workRGi_-: check your permissions...
10:39.30nfi|ermeshi all
10:39.47nfi|ermesanyone never used bristuff-CVS ?
10:40.21*** part/#asterisk shanky (n=shanky@82.159.214.9)
10:42.08RoyKnfi|ermes: yep. works for me (tm)
10:42.16MuppetMasterIs Asterisk Thread Safe?
10:42.37MuppetMasterSo, for example, if I am using a SetGlobalVar to increment a counter, what happens if two calls come in at the same time and try to increment that counter?
10:42.44nfi|ermesRoyK, havve you hfc based isdn caards ?
10:43.03RoyKnfi|ermes: one hfc-pci. just a sec, and I'll paste the lspci entry
10:43.15RoyK0000:03:05.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02)
10:43.21nfi|ermesok
10:43.29RoyKworks nicely
10:43.30nfi|ermesi have some problem to compile zaptel
10:45.16*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
10:48.52nfi|ermesRoyK, have u used bristuf patches or other patches ?
10:49.11RoyKnfi|ermes: no. just head
10:51.31*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
10:51.55zoaroyk ?
10:51.56zoayou here ?
10:57.41*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
10:59.08*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus)
11:00.29*** join/#asterisk sambal (n=sambal@213.148.236.189)
11:01.15*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
11:06.37*** join/#asterisk JmGV (n=jgomez@217.11.102.178)
11:09.34*** join/#asterisk mutilator (n=animenod@65.111.201.79)
11:09.58*** join/#asterisk bon (n=bon@217.172.148.5)
11:10.10bonany res_mysql.so user around please?
11:17.16InfraRedthat's not in standard asterisk compile
11:19.06BladeRunner05lehel: can u tell me wich version of kernel you are running on ?
11:19.43bonInfraRed: sure i know
11:19.53bonbut is there another way to have data stored in mysql?
11:20.14InfraRedyes!
11:20.24bonthat is?
11:22.14lehelBladeRunner05: 2.6.8-2-686
11:23.53BladeRunner05that kernel support your fritz card or u install the driver given by avm ?
11:25.21InfraRedcdr_addon_mysql.so
11:27.06InfraRedactually
11:27.08InfraRedignore me
11:27.18InfraRedi just foundout what res_mysql does
11:29.58lehelBladeRunner05: i install the driver
11:31.10bonInfraRed: heh:) i have cdr_addon already
11:31.13bonthat works flawlessly
11:31.19bonall the cdr is written to a mysql db
11:31.30bonbut i need to have peers/users and sip.conf in a db as well
11:31.33boncan't figure out how
11:43.35InfraRedthere is a page on voip-info
11:43.43InfraRedhttp://www.voip-info.org/wiki-Asterisk+RealTime
11:43.57InfraRedtells you the db scema
11:45.27tryfossI wonder... is it normal that the cdr from a call going out from a queue to an agent is logged with billsec=0 ? (the cdr is written once the call is answered by the agent)
11:47.27bonInfraRed: i already have those dbs
11:47.52bonlong long sip_buddies table
11:52.36*** join/#asterisk RoyK (n=roy@80.239.107.70)
11:53.22*** join/#asterisk potsboy (n=chrisg@196.34.241.242)
12:00.48jalsothi
12:01.00jalsotdoes anybody use atxfer with success?
12:01.56X-Robjalsot - put ',tT' on the end of your dial string.
12:02.08X-Robboth lower case and upper case.
12:02.46jalsotthe transfer itself works, however I get zero-way audio
12:03.25jalsotsip1->sip2-atxfer->sip3, sip1 and sip3 are in connection, however both sides are quiet
12:06.00*** join/#asterisk zotz (n=zotz@24.231.47.168)
12:15.58jalsotany idea?
12:21.01*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
12:22.56*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
12:24.55*** join/#asterisk emdub (n=mike@graffiti.corp.ef.net)
12:25.01emdubanyone around that can answer a (hopefully) simple macro question?
12:26.53emdubfor some reason a macro i have isn't falling over to the offset priority when a dial out via iax fails
12:27.02emdubnot sure why
12:28.24*** join/#asterisk neuwald (n=neuwald@weber.anpa.org.br)
12:28.48neuwaldhi folks. does anybody knows where can I found some call-back examples?
12:30.39*** join/#asterisk bmg505 (n=leon@rndf-146-60-157.telkomadsl.co.za)
12:31.31*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
12:31.52*** join/#asterisk fugitivo (n=ajf@209.13.242.109)
12:32.30lehelneuwald: voip-info.. google [?]
12:39.58DrukenHMEemdub: pastebin the macro...
12:41.19emdubpastebin?
12:41.25DrukenHME~pastebin
12:41.26jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/
12:41.30emduboic
12:42.42emdubhmm for some reason i can't connect to that url from home
12:42.56emdubweird
12:44.14emdubah here http://pastebin.ca/27589
12:44.48emdubthats just the specific part, im trying to get the macro to use the other $TRUNK_IAX_VP2 if the first dial fails
12:45.20emdubbut it never actually tries the other priorities... i tried doing a setvar and setting MACRO_OFFSET to 100 but that didnt work either
12:46.46neuwaldlehel yes, but I only found things using AGI
12:47.20*** join/#asterisk clive- (n=pirch@ndn-165-131-97.telkomadsl.co.za)
12:47.36neuwaldlet learn AGI...
12:48.24DrukenHMEhehehehe
12:48.36DrukenHMEembub it's +101 not +100....
12:48.51emdubdoh
12:49.10emdubdo i need to setenv MACRO_OFFSET or does it default?
12:49.18DrukenHMEalso, you'll need a g in the dialstring
12:49.37emdubnot sure i follow
12:50.00DrukenHMEin the first dialstring, you need to have the dialplan continue even if it fails...
12:50.01clive-can someone help me with a silly newbie perl question...how do I include the words "user@account" in the dialstr= statement
12:51.53BladeRunner05....anyone has used Controller ISDN AVM B1 v.4.0
12:52.22DrukenHMEemdub : http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
12:53.46*** join/#asterisk Madounet (n=Madounet@AMontpellier-206-1-2-97.w80-11.abo.wanadoo.fr)
12:58.09emdubhmm, seems as though adding the g option just makes it go back to the context that included it... i also made the priorities offset by 101 and that didn't fix it either... i updated the pastebin --- http://pastebin.ca/27590 (not sure why pastebin made a new url heh)
12:59.16DrukenHMEemdub: got the cli output?
12:59.23emdubyeah
12:59.25emdubone second
13:00.26*** join/#asterisk newbien (n=e@119.241.33.65.cfl.res.rr.com)
13:02.01newbienhi, need info on setting up asterisk server for linux user group to provide free outbout calls to members
13:04.05fugitivo~docs
13:04.06jbotextra, extra, read all about it, docs is 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
13:04.18emdubhttp://pastebin.ca/27595 -- drukenhmw
13:04.24emdubdrukenhme, rather
13:06.12newbienknow of any linux user groups that have an asterisk server providing outbound calls to lug members?
13:07.36DrukenHMEhmm.... looks like the first dial (to voicepulse) isn't erroring...
13:07.57emdubsomething i've done or?
13:08.28*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
13:08.48DrukenHMEWARNING: chan_iax2.c:1473 attempt_transmit: Max retries exceeded to host 66.234.228.160 on IAX2/66.234.228.160:4569/1 (type = 6, subclass = 1, ts=12, seqno=0)
13:09.05emdubnod
13:09.22emdubi assumed that was just a warning msg that was normal if the other side was pretty much dead :)
13:09.48DrukenHMEi guess you don't have voicepulse watched?
13:10.19DrukenHMEqualify= ?
13:10.24emdubnot sure i understand what you mean by "watched"
13:10.57lesouvageI'm trying to write my first agi script, reading the first 10 characters stored in a file and return it to asterisk. Can somebody please help me or point me to some examples of python agi scripts on the internet.
13:11.22emdubdruken - that looks to be for sip specifically
13:11.25emdubi connect to vp via iax
13:11.57DrukenHMEwell, all my iax connections have it... so i would assume it's not sip only :)
13:12.48emdubah i see let me give it a whirl
13:16.47lesouvageI have the agi script that isn't working yet pasted to http://pastebin.com/417099 . Any help will be apriciated.
13:17.02*** join/#asterisk Corndawg_ (i=whoisit@c-67-191-22-79.hsd1.fl.comcast.net)
13:17.19*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus)
13:20.11emdubyeah qualify doesnt seem to work
13:20.13emdubbummer
13:21.19ManxPowerlesouvage, check the docs directory in the asterisk source code.
13:23.44*** join/#asterisk lilwookie (n=root@modemcable005.89-81-70.mc.videotron.ca)
13:27.34synthetiqis there way to limit asterisk log files?
13:28.25ManxPowersynthetiq, Not really.  Have the system rotate the log files, then issue an asterisk -rv "logger reload"
13:28.37*** part/#asterisk clive- (n=pirch@ndn-165-131-97.telkomadsl.co.za)
13:28.40ManxPowersorry
13:28.43ManxPowersynthetiq, Not really.  Have the system rotate the log files, then issue an asterisk -rx "logger reload"
13:29.13lesouvageManxPower: thanks.
13:29.17emdubcmd dial
13:29.21emduboops
13:30.53*** join/#asterisk sivana (n=sivana@mixdown.ca)
13:31.59*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
13:33.00test34Can you have speech recognition with asterisk ?
13:34.03mog_homeyes
13:34.06mog_homevia sphinx
13:34.19ManxPowertest34, That question was asked on the mailing lists yesterday, as well as an answer was posted.
13:34.59test34ok thanks
13:35.26test34ManxPower, which mailing list ?
13:35.56test34asterisk-users@lists.digium.com ?
13:36.02ManxPowertest34, I don't recall
13:36.14ManxPowerThe subject didn't have the word "speech" in it.
13:36.47test34ok
13:37.43mutilator.
13:38.17*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:38.37*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:39.54*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
13:41.12test34http://lists.digium.com/pipermail/asterisk-users/2005-November/132160.html
13:43.55*** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com)
13:43.55docelmowhadup!!!!!!!!
13:45.11neuwalddoes anybody knows some link that explain how to implement a callback service?
13:45.46Ariek1exit
13:48.04test34neuwald, here's one http://lists.digium.com/pipermail/asterisk-users/2004-October/065406.html
13:53.50ExionDoes anyone know if this should work with SIP channel (not yet established): SetVar(PRI_CAUSE=17); Hangup();
13:54.05Exionthe packet comming back from asterisk include 503 Service Unavailable + X-Asterisk-HangupCause:: User busy
13:54.18ExionX-Asterisk-HangupCause seems to change but not the status type
13:55.46Exionidealy it should be status 486...
14:01.25*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
14:01.44Kattyhihi
14:05.10Kattytoo quiet.
14:05.18LostfrogSi..
14:05.20LostfrogToo quiet
14:06.01*** join/#asterisk Eight (n=blake@12-227-165-54.client.mchsi.com)
14:07.23LostfrogAnyone familiar with Avaya ACS systems?
14:10.16*** join/#asterisk potsboy (n=chrisg@196.34.241.242)
14:11.22*** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca)
14:13.23jalsotdoes anybody use atxfer with success? the transfer itself works, however no get no audio.
14:19.20*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
14:22.09*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
14:24.14*** join/#asterisk darkskiez (n=darkskie@host86-138-169-225.range86-138.btcentralplus.com)
14:25.13*** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com)
14:25.29Ferrarigood day all
14:25.38LostfrogGood morning, Ferrari.
14:26.17Ferrarianyone know the MAX length of a variable (using setvar/getvar) in asterisk 1.0X i am too lazy today to read source so i figured i would ask in here first
14:26.34Ferrarilooking for contents of the variable not variable name
14:26.45Ferrarii recall something about maybe 80 chars
14:27.21*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85)
14:27.30MuppetMasterHello.
14:27.51file[laptop]Hello has been replaced by, "Blurgblah"
14:28.24MuppetMasterI have compiled and installed ooh323 in the CVS HEAD (as of minutes ago) from within the /usr/src/asterisk-addons/..  And, the modules loads, based on the settings in /etc/h323.conf, but when I do a show channel types it does not show up.  http://pastebin.ca/27600
14:28.41MuppetMasterI had this working in v1.2beta1 with no problem.  Has something changed in v1.2-beta2 on this front?
14:29.55*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
14:30.12LostfrogBlurgblah, file[laptop].
14:32.15*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
14:32.22*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85)
14:32.44swm_Ok got a problem with asterisk meetme confrence.
14:33.03swm_Zap calls moved into a confrence work fine
14:33.12swm_sip calls from the outside into the confrence dont work
14:33.35swm_841 conf_run: Error getting conference
14:33.52swm_could it be a codec issue?
14:34.22swm_brb gotta chain smoke on the issue
14:38.17*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
14:43.01*** join/#asterisk mkrufky (n=mk@68.160.103.77)
14:45.20swm_anyone able to help?
14:46.29docelmoWhats new in life?
14:46.40docelmoSSDD?
14:46.48swm_SSDD?
14:46.59docelmoSame Shit Different Day
14:47.11*** join/#asterisk coppice (n=chatzill@5.162.17.210.dyn.pacific.net.hk)
14:47.26swm_I cant transfer sip callers/callouts into a meetme room... driving me nuts
14:50.41LostfrogMake them drive/fly to the same locale and sit down and talk.. lol
14:52.33*** join/#asterisk santiago (n=santiago@208.195.215.124)
14:53.49docelmoHas anyone has problems with keeping the GXP2000 registered?
14:55.42ManxPowerswm_, what codec are the SIP phones using?
14:56.01ManxPowerswm_, and can the SIP phones call voicemail or other things that require asterisk to play audio to the caller?
14:57.32swm_It's a outbound call I cannot transfer into the conference
14:57.53swm_say I call 5551212@asterlink and try to move it into the meetme room, it just dumps the call
14:58.19swm_I'm moving to CVS-HEAD current and see if that changes anything. Other than that It might be a codec issue
15:00.01Lostfrogthat is going to be a large bundle of paper.
15:00.14*** join/#asterisk peletiah2234 (n=benke@81.223.28.196)
15:00.34peletiah2234good afternoon everyone
15:00.50swm_Actually printed it once for the hell of it, had like 80 reams of paper and free toner for a month, so... Yeah, it took about 3 hours to print it all
15:01.58*** part/#asterisk JmGV (n=jgomez@217.11.102.178)
15:02.02LostfrogI've printed most of a 1.0 linux kernel before.
15:02.11tzangerI did that way back in the day
15:02.18peletiah2234does someone know if its possible to specify channel distinct signalling types, like pri_cpe for pri1 and pri_net for pri2(for making a stress test)?
15:02.52swm_I wonder how long it would take to print the linux 2.6 kernel.... heh
15:02.59jontowabout 6 years.. ;)
15:03.01jontowuntil 2.8?
15:03.05jontowone of the two.. :D
15:03.25jontowi'll bet on the former, given the quick turnaround of the linux folks
15:03.37swm_anyone get a plantronics bluetooth headset to work with asterisk yet ?
15:03.47LostfrogI doubt it would take more that a week.,
15:04.15LostfrogI would be dead before I finished..
15:04.20LostfrogMy boss would kill me.
15:04.29jontowthats assuming you've got a fast printer.. but we were back at dot matrix above ;)
15:04.51swm_I miss old dot matrix impact printers
15:04.52LostfrogYou notice, he said 'toner'
15:05.01LostfrogI want a line printer.. :)
15:05.10jontowhehe
15:05.11LostfrogDrive the neighbors crazy.
15:05.13swm_I want a serial based laser printer heh
15:05.20jontowi miss my old panasonic KX-24
15:05.20jontow:(
15:05.28swm_or even a serial based line printer
15:05.35newlThey used to be good printers.
15:05.40swm_Neeeeerrrrrrrrrrrrrr Neeerrrrrrrrrrrrrr Neeeerrrrrrrrrrrrrr
15:05.44jontowthat thing rocked.. never had a jam, never misprinted.. just worked until it ran out of paper or ink, fix, repeat
15:05.45*** join/#asterisk wolfson (n=ggggg@65.174.122.198)
15:06.56swm_I remember my IBM lab at school we had some Panasonic KX-somethin printers ... oh they were so loud without the noise hood on them
15:06.56*** join/#asterisk anti (n=russ@gentoo/developer/anti)
15:08.03*** join/#asterisk Inkubot (n=inkubot@200.75.4.7)
15:08.07znoGhow does one raise the volume on a Zap channel?
15:08.16swm_Wonder if they have a tatoo gun that works like a line printer but on a 2D axis ... changes ink and all that... that could be fun...
15:08.17znoGmost people ask me to talk louder, even though I'm talking prettttty loud!
15:08.50LostfrogRead the wiki?
15:08.53Lostfrogtxgain?
15:09.00swm_txgain works
15:09.10swm_but you can get echo tweaking it to hard
15:09.27LostfrogWhoever invented those printer stands had to have made a fortune.
15:09.40swm_printer stands?
15:09.54LostfrogThe ones with foam padding inside.
15:10.13LostfrogFor line printers.
15:10.34peletiah2234does someone know if its possible to specify channel distinct signalling types, like pri_cpe for pri1 and pri_net for pri2(for making a stress test)?
15:11.00swm_peletiah2234: IF YOU CODE IT YOURSELF
15:11.03Inkubotwhen i make a call with a analoge phone, asterisk wait 10 seconds aprox before make the call
15:11.15LostfrogInkubot: digitimeout?
15:11.23Lostfrogdigittimeout?
15:11.23Inkuboti really don't know
15:11.33swm_what are you using to call out with lnkubot
15:11.38Inkubota POTS
15:11.44swm_pots what?
15:11.50swm_Sipura? Cisco ATA?
15:11.53Inkubotaaa
15:11.56Inkubota cisco ata
15:11.57Inkubot182
15:12.14swm_Is your dialplan setup properly in the cisco? Maybe it's waiting for a timeout?
15:12.16Inkubotand also a IMC AccessLinx
15:12.20*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:12.26newbienknow of any linux user groups that have an asterisk server providing outbound calls to lug members?
15:12.28swm_Try dialing the number then pressing # after it, and see if it dials quicker :)
15:12.35Inkubotyeps
15:12.40Inkubotthat's works fine
15:12.43Inkubotwith #
15:12.58Kattyhmm.
15:13.07LostfrogHmm.?
15:13.14Inkubotbut here at work they don't want the # key
15:13.29docelmoDoes anyone know of a Radius billing suite?
15:13.41Inkubotgbill ?
15:14.03swm_lnkubot: your cisco ATA is not setup properly then for a dialplan.
15:14.03peletiah2234ok, thank you. didn't mean to be annoying... i was not sure if its possible to set it distinctively for a channel group(and i'm still not sure how to interpret your answer)
15:14.05Inkubotdocelmo: gbill, i think
15:14.14Inkubotok swm_ let me see
15:14.36swm_lnkubot: I could show you what mine looks like
15:14.42docelmonice thank ink:
15:14.47docelmothanks
15:15.18Inkubotthanks swm_
15:15.24Inkubotlet me see yours
15:15.34Inkubot(sorry for my english)
15:15.41swm_9..........|8...|7..........|11.|18.........|3..|4..|5...........|6>#35......................
15:16.05Inkubotdamn!
15:16.25Inkubothow you configurate the cisco ata ?
15:16.39Inkubotthrough the web page ?
15:16.42swm_http://ipaddr/dev
15:16.46Inkubota ok..
15:16.49Inkubotme too
15:16.57Inkubotand where you put that!
15:17.03Inkubotthere's nothing like that here
15:17.10*** join/#asterisk sung (n=sung@216.106.191.1)
15:17.24swm_section called "Dialplan"
15:17.31Inkubotaa
15:17.38Inkubotok..
15:17.44Inkubot0 is default
15:18.01swm_You have to customize it to your way of dialing for your country tho.
15:18.17Inkubota ok
15:18.30Inkuboti think i'm understanding how it's work
15:18.50swm_If you need more help refrence the cisco ata manual at cisco.com
15:18.50Inkubotthe first is 9+ 10 numbers ?
15:19.04znoGso the idea behind rxgain/txgain is to get both around 50% ? (in ztmonitor)
15:19.20Inkubotswm_: thanks men
15:19.21Inkubot:)
15:19.31swm_For america, I crop the 1, so it would be 9 503 312 1234 10 digits after the 9 to get out
15:19.42znoGLostfrog: is that right?
15:20.06LostfrogznoG: I don't know.. I just tweak it little by little 'till it sounds ok.
15:20.21*** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com)
15:20.52LostfrogI haven't had to do in a while.. I use a T1 now.
15:21.15LostfrogVoice level seems to be perfect.
15:21.40*** join/#asterisk oogle (n=jart@justin.ctlinc.com)
15:25.35znoGi can't get zero echo to work even tweaking [tx,rx]gain settings
15:25.49znoGi tried to raise/lower them to varying levels and.. nope
15:26.13ManxPowerechotraining=900
15:29.00tzangereww
15:29.02tzangerechotraining is nasty
15:29.25tzangerI realize it works but ewwwwwwwwwwwwwwwwwww
15:29.30tzangerit works like cod liver oil works
15:29.36tzangerthe cure is sometimes worse than the disease
15:30.30Inkubotswm_: thanks men again
15:30.35Inkubotworks fine
15:30.39Inkubot:D
15:31.19fugitivocvs [login aborted]: unrecognized auth response from cvs.digium.com:
15:31.31fugitivoit's me?
15:31.53fugitivoyes it's me
15:31.55fugitivo:)
15:32.22*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-66-83.hsd1.fl.comcast.net)
15:32.57fugitivoit works from one server, it doesn't from another one
15:33.38*** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss)
15:33.41Madkisshi folks.
15:33.45MadkissI need some asterisk developer.
15:35.27Beirdohave a call come in via IAX, go out via IAX to my cellphone...  the caller can hear me fine, I can't hear squat
15:35.38Beirdosomething's being a PITA
15:37.02znoGtzanger: why you say that? is it that ugly?
15:37.22*** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au)
15:37.34tzangerznoG: echotraining makes me wait a second before I can talk
15:37.36tzangerdrives me nuts
15:37.45tzangerif you have to pick the phone up to your head to answer that's cool
15:37.59tzangerbut 99% of the time when I place a call the phone is at my head already
15:38.05tzangeror I pick up the phone and hit the line t answer
15:38.49coppicetzanger: so, how much can you be trained to do in just one second? :-)
15:39.05tzanger:-)
15:39.56znoGah so I should call with speakerphone, and once answered pick up :)
15:40.03znoGi just have echotraining=yes
15:40.06fugitivotzanger: 1 second? i have to wait like 15 seconds with my x100p
15:40.16znoGi want to raise txgain so people can hear me a bit better, but the echo increases with txgain
15:40.22tzangerznoG: yep it does
15:40.28tzangerproperly balance the gains :-)
15:40.48fugitivoznoG: what hardware?
15:41.34znoGjust a std Zap card
15:41.47fugitivowhat's that? x100p?
15:41.55znoGyup
15:42.50filetzanger: ANDREW
15:43.52tzangerfile: what
15:43.59*** join/#asterisk marrandy (n=marrandy@209.216.76.1)
15:44.05filetzanger: hi
15:44.08tzangerheh
15:44.08tzangerhello
15:44.09*** join/#asterisk tryfoss (n=morten@ti131310a080-14221.bb.online.no)
15:44.16filetzanger: banana?
15:44.20*** join/#asterisk dsmouse (n=nnnnnnnm@prism.datastacks.com)
15:44.27marrandyhello - nice long list of people
15:44.31tzangerfile: no thanks, I ate
15:44.41*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85)
15:44.43dsmousehello, marrandy
15:44.44filetzanger: fine then!
15:44.44MuppetMasterHello
15:44.46marrandyanyone here using a fanless setup ?
15:44.53MuppetMasterUpgraded to CVS HEAD and now getting this:  chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 20 scheduled tasks all at once
15:44.55MuppetMasterAny ideas?
15:44.56tzangerfile: at least quote the entire message
15:44.59MuppetMasterA problem??
15:45.05develis anybody here using the alarmreceiver module/app?
15:45.21tzangerI have an unbelievably thick skull.
15:45.23tzangerMuppetMaster: yeah I know
15:45.35tzangerMuppetMaster: do me a favor and disable the jitter buffer... does it go away?
15:45.41MuppetMasterWill check
15:46.07coppicemarrandy: nobody is a fan of my setup
15:46.12marrandy:-)
15:48.06marrandyI'm looking to find a  system that can do 12 extensions, 4 or which are pots phones, the other 8 sip phones, probably 5 or 5 in use at any one time but I'm looking for a low power, fanless system if possible
15:48.45fugitivomini-itx
15:49.20develmarrandy, i just started some testing with the via mini-itx platform as well (but not the lower power fanless ones yet)
15:49.38marrandyyes, I've been looking at the via eden 533MHz, 800MHz and 1GHz but keep reading about issues, so was wonering of anyone here had a system
15:49.40fugitivomarrandy: via mini-itx model C3
15:50.07coppicemarrandy: what kind of issues?
15:50.17MuppetMastertzanger disabled, no errors yet
15:50.25fugitivomarrandy: i think you have to make some changes to the code, the via mini-itx is not 100% x86
15:50.32marrandyI could use a tdm400 with one pots line module and 3 pots phones if the m/b only had one pci slot
15:51.13zoai used it before
15:51.14zoait works
15:51.25marrandydevel:  would like to know what m/b you asre trying anf the results
15:51.31marrandyasre = are
15:51.58coppicefugitivo: you just need to treat the processor as an i586
15:52.04MuppetMasterI am also using the ooh323 driver, it appears that it now references itself to the config file /etc/asterisk/ooh323.conf instead of h323.conf as indicated in the readme.
15:52.16fugitivomarrandy: http://www.voip-info.org/wiki/view/Asterisk+hardware+mini-itx
15:52.17marrandycoppice: transcoding issues apparently
15:52.29develmarrandy, i've got the mII-10000, and if you check back in a few days i should have some detailed results
15:52.35*** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net)
15:52.39*** join/#asterisk paryl (n=paryl@209.236.78.59)
15:53.22coppicemarrandy: well, they have limited processing power for things like transcoding. FC4 won't install on one, because they screwed up the installer. generally, as long as you treat them as i586 machines they are fine.
15:54.02paryli'm trying to understand something about faxing... i keep reading "Right now Asterisk does not have very good fax support", but then they go on to describe what seems to be something similar to trying to fax over a VoIP connection
15:54.26filedid someone subscribe the asterisk-users list to the internic.at support e-mail?
15:54.38Rav1974hi guys, I have a TE110p with an ADIT 600. I am getting terrible echo. I tried RX & TX gain in asterisk & ADIT 600.
15:54.40marrandyfugitivo:  yes, thanks, I've read those.  was hoping somone here might have some working experience on those units
15:54.42synthetiqif you delete the logfiles will they be written to anymore?
15:55.46marrandydevel: do you have my email address? can you email me when you have some reults
15:55.48Rav1974digium wasn't a big help either.  I looked in google etc. but still no good.
15:56.14MuppetMasterIf I do an IAX -> IAX call from Asterisk A to Asterisk B, where Asterisk B simply returns a Busy() app, I simply get an originate fail over the Manager API.  How do I emulate a busy from a signaling perspective as one might expect from the PSTN?
15:56.26develmarrandy, /msg me your email addr and i'll keep track of it.
15:57.01marrandyat the moment, I'm running it on a dell poweredge 2400 with raid 5 which is a bit of a veast
15:57.09marrandyveast = beast
15:58.14paryldoes faxing work well enough if you have the fax plugged into a tdm400p and bridge it directly to a T1?
15:58.38develparyl, we do faxing over g711u sip
15:58.40coppiceparyl: sometimes
15:58.47snittparyl: that's ok
15:58.53fugitivoparyl: it should work most of the times, but it depends
15:59.13*** join/#asterisk marrandy (n=marrandy@209.216.76.1)
15:59.18parylsometimes?  it depends?
15:59.27fugitivoparyl: sometimes the tdm400 fails to get the fax, at least with my experience
15:59.33*** join/#asterisk morale (n=russell@S010600111155e117.cg.shawcable.net)
15:59.38coppiceparyl: for a long time faxing would not work with the tdm400, but someone told me they finally fixed the driver
15:59.56fugitivocoppice: really?
15:59.57*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85)
16:00.02kuku5tdm400 is the pri card?
16:00.06moraleis asterisk beta2 cvs-head?
16:00.13fugitivokuku5: no, analog
16:00.13marrandyaack..
16:00.22kuku5ah
16:00.29kuku5is the pri card cool for faxing?
16:00.50marrandydevel:  did you get it ?
16:00.51parylthe issue here is... we have 8 fax machines in the building
16:01.15paryland all voice/fax traffic goes over the t1
16:01.26fugitivoparyl: sending fax always worked for me, no fails
16:02.18znoGthey fixed the zaptel driver for faxing over the FXS port?
16:04.35develyep, marrandy, got it.  i'll get my results to you in a few days.
16:05.15parylfugitivo: k, thanks.  i hope it works out
16:08.53*** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk)
16:09.13fourcheezecan anyone recommend a BRI card for use with * ?
16:09.38*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85)
16:09.41*** join/#asterisk RussCC (n=face@216.157.205.211)
16:10.23MuppetMasterWith ooh323, I need to have port 1720 open for the signalling.  What about for the audio stream?
16:10.44*** join/#asterisk R3DB0x (i=nobody@66.142.28.36)
16:11.37fourcheezeare there BRI cards out there with open source drivers?
16:11.40*** join/#asterisk BleedingMe (n=Bleeding@ppp-69-237-68-225.dsl.scrm01.pacbell.net)
16:11.45fourcheezeI keep finding closed source ones
16:12.00RussCCwhen I attempt to make an outgoing call I get it squaking about localprefixes.conf, could some one please help me?
16:14.58*** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk)
16:16.18*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85)
16:17.15*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
16:17.32*** join/#asterisk Domiplus (n=domiplus@66-202-165-66.rev.knet.ca)
16:18.11Rav1974help me please guys, I am having terrible echo problems
16:18.20*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
16:18.24Rav1974I have a TE110p with an ADIT 600. I am getting terrible echo. I tried RX & TX gain in asterisk & ADIT 600.
16:19.08*** join/#asterisk Madounet (n=Madounet@AMontpellier-206-1-2-64.w80-11.abo.wanadoo.fr)
16:19.26Rav1974I'm praying it can be solved, I spent over $2,000 on equipment already...
16:19.46nfi|ermesproblem compiling zaphfc and other bristuff drivers
16:19.49nfi|ermeshttp://pastebin.com/417275
16:19.59malverian[work]Rav1974, It won't be ;)
16:20.00filehey Mr. DJ jam all night long
16:20.06filehey Mr. DJ play that song for me
16:20.44malverian[work]Rav1974, I have the same, and no matter what I've tried, echo persists on some calls.
16:20.49Rav1974malverian[work]: you are a real comic.  I'm still praying though...
16:20.52Rav1974:)
16:21.04MadounetHi, is there someone using zaprtc for IAX trunking ?
16:21.10malverian[work]Rav1974, I suppose you could try the new MG2 echo canceller.
16:21.18Rav1974malverian[work] so its the te110p thats the problem?
16:21.49*** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net)
16:22.27Rav1974does Sangoma have the same problems with echo?
16:24.34tzangerRav1974: it can
16:24.40*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
16:24.44tzangerRav1974: have you tried the MG2 echo canceller
16:24.46tzanger(CVS HEAD)
16:26.33*** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com)
16:29.07*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
16:29.57ooglei'd like to write a voicechanger version of Dial()... but i'm relatively new to asterisk hacking.  can anyone give me some tips on where to start?
16:31.04tzangeroogle: I'd start with echo()
16:31.21tzangerand what you read in, connect to your voice changer, and write hte result out
16:31.56malverian[work]tzanger, I just said that ;)
16:32.08tzangermalverian[work]: heh
16:32.28malverian[work]Oddly enough though, echo became _worse_ here when I used MG2 over KB1
16:34.22*** join/#asterisk ful|work (n=fulgas@209.8.233.129)
16:34.23Rav1974ya that doesn't help.  I was on the phone with digium & they did all sorts of echo cancelation tricks
16:34.39Rav1974I only get echo from fxo to fxs ports
16:34.50swm_cheap fxo card?
16:34.52Rav1974sip/iax to fxs doesn't have any echo
16:35.00Rav1974nope I'm on an ADIT 600
16:35.04tzangerRav1974: that doesn't make any sense
16:35.09Rav1974maybe I should try the cheap one :)
16:35.10tzangerRav1974: have you tuned your Adit600?
16:35.14tzangerwhat do you have the gains set to on there
16:35.16tzangerI have an Adit600
16:35.19*** join/#asterisk timscott (n=war@d198-166-226-205.abhsia.telus.net)
16:35.32*** join/#asterisk paryl (n=paryl@209.236.78.59)
16:35.37Rav1974tzanger: I tried, but i reset it to defaults as per the manual
16:35.41timscottIs there anyone here who could help me with a problem with Asterisk and the DISA function?
16:35.43Rav1974it helped with the volume problem
16:36.08tzangerRav1974: I find I have zero echo with the adit600 fxs or fxo... it's got very nice interfaces
16:36.33Rav1974tzanger: tell me your gain & impedance settings on adit please
16:36.46tzangerit's all default
16:36.53tzangerRav1974: what processor are you running on?
16:37.05Rav1974pentium 4 2ghz I think
16:37.17Rav1974i'm not sure if its 2ghz or 2.4ghz
16:37.22Rav1974with 256mb ram
16:37.35oogletzanger: thank you for the voice change tip
16:37.36sylecat /proc/cpuinfo
16:37.46tzangerok
16:37.48tzangerthat's fine
16:37.51tzangernow do this
16:37.56tzangeruse CVS HEAD
16:37.59tzangerif you're not already
16:38.01tzangerjust for zaptel
16:38.08Rav1974Intel(R) Pentium(R) 4 CPU 2.40GHz
16:38.09tzangeryou don't need to go to CVS HEAD for asterisk if you don't want to
16:38.17tzangerin zconfig.h
16:38.22tzangerenable MMX optimizations
16:38.27tzangerseelct the MG2 echo canceller
16:38.33tzanger(make sure all the ones above it are commented out)
16:38.43tzangerin the zaptel Makefile
16:38.52tzangerbelow the reference to "all the settings are in zconfig.h"
16:38.54tzangeradd
16:39.00Rav1974ya, the digium tech support did as you said
16:39.02tzangerCFLAGS+=-march=pentium4
16:39.09tzangerKFLAGS+=-march=pentium4
16:39.16tzangermake clean && make && make install
16:39.31*** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
16:39.35oogleok
16:39.36Rav1974i don't think he did "CFLAGS"
16:39.40Rav1974i'll try that
16:39.52timscottIs there anyone here willing to field a couple Asterisk questions about the DISA function?
16:40.09tzangertimscott: I can try
16:40.10Qwelltimscott: only if you ask them
16:40.15tzangergenerally though you just ask :-)
16:40.17Qwelloff to work :D
16:40.18timscottOh.
16:40.31*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:40.44timscottI'm trying to use the DISA functions in Asterisk, but my DTMF is not being passed from PSTN -> Asterisk.
16:40.46timscottWhat should I be doing?
16:40.56tzangertimscott: have you played with your zapata gains?
16:40.58timscottI've already tried like a million things.
16:41.01timscottErr
16:41.04timscottI'm using VoIP.
16:41.09tzangertimscott: you just said from PSTN
16:41.12timscottBut I can check, I can't remember what they are set to at the moment.
16:41.14timscottWell, yeah.
16:41.15tzangeris this a SIP provider?
16:41.18timscottIAX.
16:41.22timscottVoipjet.
16:41.26tzangerbleh, voipjet
16:41.29tzangertry this
16:41.30tzangeriax2 debug
16:41.33tzangerand try to call in
16:41.43tzangeryou should see messages about "DTMF event received: 5" or whatever digit was pressed
16:41.51tzangerif you do not see that then VoipJet is not sending it to you
16:41.57tzangerIAX2 DTMF is not a problem as it is with SIP
16:42.16tzangerbut if VoipJet's upstream is SIP and isn't passing it properly to VoipJet you'll not see it in IAX
16:42.17implicittimscott, and only a problem with sip if you are stupid as a log
16:42.32tzangerimplicit: not really
16:42.37implicitand don't understand the different dtmf modes
16:42.43implicitand what they are useful for
16:42.54tzangerthere are lots of providers who won't give you RFC2833 but insist you can get inband DTMF with compressed voice codecs
16:43.06fileugh
16:43.08filedon't remind me
16:43.12file<cough>Level3</cough>
16:43.21fugitivotzanger: which providers do that?
16:43.23implicitand there are lots who will suck your dick and not complain
16:43.24timscottI'll check.
16:43.53*** join/#asterisk lehel (n=mey@86.125.98.100)
16:43.55implicitit's not the problem of SIP, its the stupidity of people using it
16:44.04sivanawhere is * looking for sendmail?
16:44.05implicitand it's not the problem of RTP either
16:44.18fourcheezefile: level3 do that - I thought they were supposed to have some clue at least
16:44.21parylis there a way to make the polycom 501 'dial instantly'... so that as soon as what i'm dialing matches a pattern asterisk recognizes...
16:44.27filefourcheeze: I know
16:44.31mutilatorwell implicit
16:44.36Rav1974please recommend a reliable VOIP provider (only for outgoing)
16:44.37mutilatora phone isn't an advanced user interface
16:44.39swm_Why can you not move a parked call into a queue? "Channel 'Zap/1-1' sent into invalid extension '350' in context 'parkedcalls', but no invalid handler ... Hungup 'Zap/1-1' ... no channel type registered for 'PARK' ... Unable to request channel PARK/350
16:44.48mutilatorit works or it doesn't
16:45.04Chujiparyl : Try selecting the line first, then dialing
16:45.05implicitDoes anyone know how to or if you can enable SRTP on Cisco as5400?
16:45.09implicitfor SIP calls?
16:45.12shmaltzswm_ what are you trying to do?
16:45.21paryltzanger: you said "seelct the MG2 echo canceller"... i don't see an option for MG2
16:45.23swm_Move a parked call into a queue
16:45.23Chujiparyl : If I do that, and dial 10 digits, it dials
16:45.30tzangerparyl: CVS HEAD
16:45.44parylChuji: i want it to dial without waiting
16:45.46swm_there appears to be no bridge between queue's and parked calls
16:45.52paryltzanger: that what i'm in
16:46.03parylit should be in zconfig.h?
16:46.28tzanger#define ECHO_CAN_MG2
16:46.30tzangerin zconfig.h
16:46.45[TK]D-FenderHey File, care to pass your contact info for while you're in montreal so we can coordinate?
16:46.48timscotttzanger: I'm not seeing it.
16:47.00file[TK]D-Fender: sure
16:47.04shmaltzwwm_, you will have to pick it up, or make the timeout extension goto that queue using parkandannounce
16:47.15tzangertimscott: well you've successfully diagnosed your first Asterisk IAX2 problem :-)  Now go after VoipJet asking them why you can't see DTMF when you call your DID
16:47.45coppice#define ECHO_CAN_MG2 in not more than 25 words
16:47.52swm_shmaltz. I'm using FOP (flash operator panel) Beeing both defined, It should be able to be dragged from a parked state to a queue with no problem tho.
16:47.56timscotttzanger: will do.
16:47.59timscottGracias.
16:48.03tzangerno problem.
16:48.05tzangercoppice: ?
16:48.35shmaltzswm_, when you drag from a sip/zap channel to that queue does it work?
16:48.58swm_shmaltz. Drops the call gives the errors stated earlier
16:49.07implicitcoppice, do you know anything about srtp on ciscos or some sort of b2bua  that can also do srtp and proxys media?
16:49.14swm_it's because asterisk has no channel defined named "PARK"
16:49.15shmaltzswm_ even when doing it from sip/zap?
16:49.29shmaltzswm_ patebin your extensions.conf please
16:49.30swm_shmaltz. Affirmative
16:49.33shmaltzare you using AAH?
16:49.37coppiceimplicit: ciscos have SRTP. very few other things seem to right now
16:49.40swm_AAH ? WTF?
16:49.48implicitcoppice, for SIP calls do you know how to get support on the as5400?
16:49.59*** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com)
16:50.42shmaltz~aah
16:50.43jbotaah is probably Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324
16:50.44coppiceimplicit: I haven't used it. I just know its there
16:50.44shmaltz~aah is Asteirsk At Home
16:50.45jbot...but aah is already something else...
16:50.45impliciti know it's there but i don't know if it has support for negotiating keys over SIP
16:50.45implicitthey say a lot of crap about mgcp
16:50.45swm_What the question basically is. Why does asterisk specifically channel.c line 2525 ast_request have No channel type registered for 'PARK'
16:51.06coppiceimplicit: it wouldn't appear to be of much use with the key management part
16:51.07swm_~beat asterisk
16:51.15implicitcoppice, exactly
16:51.20impliciti don't know if they support it with SIP
16:51.24shmaltzswm_ please let me see ur extensions.conf
16:52.02Rav1974tzanger: please tell me where you added CFLAGS+=-march=pentium4
16:52.20swm_I dont think extensions.conf has anything to do with this issue, I can do everything but move a parked call to a queue
16:52.41tzangerin Makefile, under the comments about zconfig.h
16:52.42Rav1974tzanger: at the bottom of zconfig.h?
16:52.44tzangerno
16:52.48coppiceimplicit: the srtp.sourceforge.net project is from cisco, and that is only just springing to life to become more polished and complete. perhaps cicsco's implementation in their own kit still has bits missing
16:53.01impliciti am sick of cisco
16:53.09*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
16:53.16impliciti've had so many problems with REFER's on their gw's and shit too
16:53.49implicitthey are still smokin the sccp pot
16:54.00implicitwho the FUCK would use it, i don't know
16:54.19coppicecorporate users of CCM use it
16:54.30shmaltzswm_, OK, I know why
16:54.32implicitobviously the only ones
16:54.50coppicethat's all they need. they pay a lot :-)
16:54.51*** join/#asterisk js_birdboy (n=jselleck@60.68-224-177.reverse.smoothfusion.com)
16:55.13shmaltzswm_ what if you make a button in FOP for the other leg of the call (like zap or sip) that is parked and drag that?
16:55.39shmaltzswm_ there is a problem in the manager with redirects on parked calls
16:55.42js_birdboydoes anyone know a good way of keeping track of peak trunk connection usage?
16:57.32shmaltzyou can see that for yourself, fireup astman and park a call then try to redirect it
17:00.46*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
17:00.46*** join/#asterisk gst (i=gst@eris.sysfrog.org)
17:02.52ms345If I get the callerid number sent to * on my BellSouth PRI what do I do to get the CallerID name? Just ask the telco to send a name too?
17:03.10shmaltzms345, you can ask, but they wont do that
17:03.22shmaltzyou generaly don't get cname on PRIs
17:03.22ms345how are you supposed to get it?
17:03.33shmaltzms345, you don't
17:03.40shmaltznobody does on PRI
17:03.48ms345darn...
17:04.14shmaltzjs_birdboy, I remember something like MRTG on the wiki
17:04.15ms345I wonder if I can look it up on Google quick enough....
17:04.19shmaltzsearch for it
17:04.40shmaltzms345, sure, search the list there are some scripts on how to do it
17:04.50ms345thx!
17:05.06shmaltzms345, a quick hack would be
17:05.30ms345I already send the CID # via jabber, but looking for a name...
17:06.00shmaltzms345, you call a POTS that has callerID service with the callerID set to what you got on the incoming call, then make the dial statement to the final queue extension, you can then cache the name
17:06.13christoI get 'Could not create socket' whenever I attempt to connect to the manager interface on a * box. Netstat on the box shows nothing listening on 5038. How can I get the manager API to start accepting connections?
17:06.37shmaltzchristo, /etc/asterisk/manager.conf
17:07.27christoshmaltz - I have my manager.conf all set up.
17:07.40shmaltzchristo, then restart asterisk
17:07.56shmaltzchristo, can you pastebin your manager.conf
17:08.19ms345interesting... that is quite a hack - I'll dig further - thanks for the insight.
17:08.19christoshmaltz - yes, I've tried restarting *. My manager.conf is here: use Topdates::DB;
17:09.04shmaltzchristo, WTF?
17:09.28christodoh - bad paste... http://pastebin.ca/27616
17:09.46christoI nearly posted you my personal calendar :)
17:10.15christoyou could've seen which night I'm having dinner with my mum next week heh
17:10.25brettnemI bet you've got a lot of juicy stuff on that!
17:10.58js_birdboythanks shmaltz
17:11.15shmaltzjs_birdby, for what?
17:11.35*** part/#asterisk timscott (n=war@d198-166-226-205.abhsia.telus.net)
17:11.42shmaltzjs_birdboy, for what?
17:11.43js_birdboythe MRTG bit about tracking peak trunk connections
17:11.51shmaltzoh I c
17:11.52shmaltznp
17:12.21shmaltzchristo, you have enable=no, change that to enable=yes
17:12.23*** join/#asterisk CoderCR (n=creyna@dsl093-157-131.phx1.dsl.speakeasy.net)
17:12.37*** part/#asterisk CoderCR (n=creyna@dsl093-157-131.phx1.dsl.speakeasy.net)
17:12.48Rav1974tzanger: how do you think it will sound without an echo canceler?
17:12.51shmaltzerr, enabled=yes
17:13.36tzangerRav1974: it will have very loud echo.  the time between your voice going out and the echo coming back may be very short, or it may be a hundred ms or more depending on various factors
17:14.21christoshmaltz - I have played with the 'enabled' setting under the 'general' section in manager.conf and it doesn't make any difference..
17:14.29christoI just hadn't set it back to 'yes'
17:14.31Rav1974tzanger: where is the makefile (i'm new to this)
17:14.42tzangerzaptel source directory
17:14.53*** join/#asterisk delmar (i=delmar@203.114.178.231)
17:14.55shmaltzchristo, set it to yes, and restart
17:15.05christoI have
17:15.09christoa few times :S
17:16.36christohowever, I can see it running now when I do a netstat -tpl
17:17.00*** join/#asterisk rob314 (n=root@207.58.194.55)
17:17.09delmarsip --> asterisk -- iax2 -- asterisk ---> sip  ... having some minor echo problems. locally at each asterisk ie sip ---> asterisk -----> sip  .. no echo problems at all. what * config bits might I look into to help resolve this?
17:17.23fugitivograt
17:17.24fugitivogreat
17:17.35fugitivoi'm getting Illegal instruction running cvs version
17:17.42*** part/#asterisk rob314 (n=root@207.58.194.55)
17:17.43lehel:P
17:18.49*** join/#asterisk potsboy (n=chrisg@196.34.241.242)
17:21.37*** join/#asterisk Defraz (n=t0tal@72.24.26.210)
17:21.43shmaltzdelmar, try not using iax2
17:22.10delmarshmaltz, oh?
17:22.12Rav1974tzanger: sorry for sounding dumb, but there is no "makefile" in the zaptel source directory. only makdfw.c
17:22.21delmarshmaltz why not just make it work right?
17:22.33lehelRav1974: ) u r serious?
17:22.38shmaltzdelmar, because iax2 is not right
17:22.39christoRav1974 - run ./configure first?
17:22.52shmaltzI wish it work as good as sip but it dosnt
17:22.54delmarshmaltz, how so
17:23.07delmarshmaltz, so iax2 has known echo problems?
17:23.17shmaltzdelmar, one sec
17:23.20shmaltzim on the phone
17:23.20drumkillathat makes absolutely no sense
17:23.21fugitivoiax2 works ok for me
17:23.22fugitivono echo
17:23.30drumkillaiax2 has *NOTHING* to do with echo
17:23.32delmarshmaltz, i have found it working great in other solutions...
17:23.44delmarright
17:23.47drumkillanor does any other VoIP protocol
17:23.47delmarso where is my echo :P
17:23.53filedrumkilla: !!!!!!!!!!!
17:24.13delmarok so here is the solution that has echo.....
17:24.13drumkilladelmar: at some analog interface
17:24.19fugitivodelmar: using softphone?
17:24.40drumkillayay
17:24.49Chujidrumkilla : what's the original bug that the wW options went in
17:24.51delmarsip --> asterisk -- iax2 -- asterisk ---> sip ... codecs are ulaw  to  g729 over iax... back to ulaw
17:25.00delmarhardphones
17:25.07drumkillaChuji: 12312
17:25.11delmarno analog interfaces involved in that
17:25.11drumkillajk, I have no clue
17:25.15fugitivocodec_ilbc is broken on today's cvs?
17:25.26drumkilladelmar: there is still an analog interface at the endpoints
17:25.30drumkillayour sip phones suck :)
17:25.31tzangerRav1974: there is a file called "Makefile" there or you are unable to build it.
17:25.33*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
17:25.54fugitivoi have to noload => codec_ilbc.so
17:26.03delmardrumkilla, no, there is no analog interface dude. unless you count the mic/speaker at each hardphone
17:26.10fugitivoi'm getting Illegal instruction
17:26.16drumkilladelmar: yes, i do :)
17:26.19drumkillabecause that's what it is
17:26.45delmardrumkilla, well if thats your best effort in here... im just gonna laugh :P
17:27.04drumkillabest effort?
17:27.15lesouvageWill somebody please look at http://pastebin.com/417327 to see what I'm doing wrong with my first AGI script.
17:27.38fugitivoanyone else having problems with codec_ilbc.so?
17:28.07tzangerCorydon-w: re: 3360
17:28.22tzangerCorydon-w: I understand what you're doing but now this requires a change with all VOIP providers I use to be able to set this
17:28.22mutilatorhaha
17:28.25docelmo•lesouvage• its written in python ???
17:28.30mutilatorthis guy in our techsupport chatroom is a riot
17:28.38mutilatorhe's bashin on the supporrt dude hardcore
17:28.50mutilatorand support guy is trying to be nice as possible
17:29.08mutilator<dyoderindustries-Net4Kids> no just nevermind...I'll get Carla to help me, she's nicer than you
17:29.12mutilatorheh
17:29.31mutilatorguess ya gotta see it and know who in support hes talkin to to find it as funny as i do
17:30.54Corydon-wtzanger: it would require all service providers you use to update to the latest code after a fix goes in ANY way
17:31.09lesouvagedocelmo: from what I understand from the o'reilly book that should be working. I'm open for every advice and direction to info how read a first line of a flat file.
17:31.24delmarsiphardphone(ulaw)  ---> Asterisk ----> siphardphone(ulaw) works mint... siphardphone(ulaw) ----> Asterisk--->PSTN(TDM) .. works mint... siphardphone(ulaw)----->Asterisk ---->VoIPProvider(g729/IAX) .. works mint... siphardphone(ulaw) ---->Asterisk----->VoIPProvider(IAX2/g729) ... works mint.... so you see... most things are working fantastic... just not working so great when it comes to siphardphone(ulaw) --> Asterisk -->iax2/g729---->Asterisk----->sipha
17:31.25tzangerCorydon-w: nonsense
17:31.58tzangeron MY asterisk system if a peer's qualify has expired they become UNREACHABLE (hell it even says UNREACHABLE!) -- Dial should fail immediately with CHANUNAVAIL
17:32.01tzangernot CONGESTION
17:32.03Corydon-wtzanger: well, then, you don't need a change in the Asterisk source code
17:32.13tzangerCorydon-w: ??
17:32.28Corydon-wtzanger: you either need a change in the source code or you don't
17:32.32tzangerI do
17:32.45tzangerI need the damn Dial() to return CHANUNAVAIL if the qualify shows them as unreachable
17:32.46Corydon-wIf you need a change in the source, you're going to need your providers to update to that change
17:32.48tzangerI can do it
17:32.50tzangerno
17:32.52*** join/#asterisk KranZ (n=user@sme.bestline.net)
17:32.54tzangerit's on my system
17:32.54*** part/#asterisk KranZ (n=user@sme.bestline.net)
17:32.56*** join/#asterisk KranZ (n=user@sme.bestline.net)
17:33.10Corydon-wThen why not just change your system and be done with it?
17:33.26Rav1974tzanger: I feel silly to say it but, the "Makefile" not makefile was in the zaptel source directory. I forgot the case *sorry*
17:33.31fugitivoshould i post a bug for codec_ilbc? anyone else with the illegal instruction error?
17:33.42tzangerRav1974: no worries
17:33.50KranZilbc is crappy anyways
17:33.59delmarso back to my original question... what are the main things to look at with Asterisk when it comes to echo problems in a solution like  siphardphone(ulaw) --> Asterisk -->iax2/g729---->Asterisk----->siphardphone(ulaw).. .anyone?
17:34.07fugitivoyes, but it's not working for me at all, asterisk is crashing
17:34.24KranZthe quality is not worth it
17:34.33docelmo•lesouvage• I dont know python nor do I want to learn it. I was more or less making a crack at you for using python. In theory yes.. Python should work.. But I dont know how to develop in it.
17:35.26KranZdelmar: latency and if echo cancellation is enabled at both phones
17:35.41potsboydelmar, check you have ztdummy compiled if you dont have a card to provide timing
17:35.53docelmo•lesouvage• Do you know PHP? That I do know..
17:36.00Rav1974Features are now configured in zconfig.h
17:36.09KranZdocelmo: do u use php in the dialplan?
17:36.15docelmoYes
17:36.18docelmoin a round about way
17:36.19delmarpotsboy, im not using trunking.. thought u only had to worry about that if u are trunking and need a timing source?
17:36.22Rav1974tzanger: below that line I add the C&K flags
17:36.27tzangeryes
17:36.38KranZdocelmo: what mod did u install for that?
17:36.44docelmoabout 75% of the time I tell the dialpaln to dump to a phpAGI file and continue running the dialplan
17:37.04lesouvagedocelmo: What I try to do is run an agi script that reads the first line of a file (this is a phonenumber) and return it to asterisk for further use.  A suggestion on how to this in php is more then welcome.
17:37.14KranZso you dont do any call directing with php?
17:37.41docelmoI do everything in php.. I can get more accomplished 90% of the time
17:37.53docelmo•lesouvage• hold on I will make a quick agi for it.
17:38.44lesouvagedocelmo: thanks a lot. I spend hours on it without any result.
17:39.02docelmoYou want just 1 line from a text file?
17:39.03*** join/#asterisk nitram (i=foo@superblob.com)
17:39.15shmaltzdelmar, I'm back to that iax2 argument
17:39.17lesouvagedocelmo: yes
17:39.23docelmoThe first line?
17:39.30delmarshmaltz oh?
17:39.43shmaltzdelmar, iax2 does not work very nicely
17:39.51shmaltzit has terrible sound quality
17:40.08shmaltzincluding ticks and distorted sound
17:40.17KranZtiming
17:40.28delmarshmaltz, dude.. sound quality is really a codec thing. i have no problems except this one solution and I believe its a case of it not working because its not setup right and it can be fixed.
17:40.45shmaltzyour echo is however more related to something else and not iax2
17:40.45fugitivothe new echocan is worse than the old one
17:41.07shmaltzdelmar, codec, so which codec you recommend that I havnt tried yet with iax?
17:41.10delmarso in regards to this timer....
17:41.16lesouvagedocelmo: yes the first line. It's a file that holds a number that wasn't picked up. I will use it for a retry by asterisk.
17:41.33shmaltzI have a digium t110 with that iax problem and still
17:41.33delmarpotsboy, there is a X100 clone card in one box and a TDM card in the other... do I still need ztdummy?
17:41.54shmaltzdelmar, timing will not change echo problems
17:41.57delmarshmaltz, im using g729 atm.
17:42.25shmaltzdelmar I have tried alaw, ulaw, gsm, g729, g723 and no luck
17:42.37shmaltzanyhow, back to ur echo problem
17:42.48shmaltzwhat sip devices you using when you have echo?
17:43.21delmarso as I said above...
17:43.22delmarsiphardphone(ulaw)  ---> Asterisk ----> siphardphone(ulaw) works mint... siphardphone(ulaw) ----> Asterisk--->PSTN(TDM)
17:43.22delmar<PROTECTED>
17:43.24delmar<PROTECTED>
17:43.30delmarall working...
17:43.36delmarand working really nicely
17:43.52delmarsiphardphone(ulaw) --> Asterisk -->iax2/g729---->Asterisk----->siphardphone(ulaw) .. has minor echo
17:44.20delmarand i mean.. its not nasty loud stuff.. but its there.. and its at a level where it's kinda tolerable but .. can be a pain
17:44.39*** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
17:45.31*** join/#asterisk zoa (n=kkk@pirus.securax.be)
17:45.59delmarshmaltz, so in answer to your question.. sip devices are hardphones.
17:45.59docelmoSo now KranZ what did you want to know about AGI and PHP?
17:47.14shmaltzdelmar, try using ulaw or g729 all the way, with no transcoding, you are doing double transcoding which be the source of the echo
17:47.26docelmodelmar have you tried SIP->SIP for your IAX connection?
17:47.39docelmoThat too
17:48.22fugitivodamn, echo is higher with the new echocan
17:48.24docelmoI had the problem with echo on sip.. and it was all the codecs.. I cleared it up by g711u in the office to TDM out Cisco or PSTN PRI
17:48.27delmarshmaltz, yeah i had thought of this but the transcoding time is very low on each * box, ok so something to try when someone is at the remote office.. 7am here so a little early.. what else...
17:49.00docelmo7am.. Where are you?
17:49.09delmarok so SIP between * boxes rather than IAX2...
17:49.20docelmoI use sip unless I am trunking
17:49.23delmarNew Zealand
17:49.27docelmoahh
17:49.49shmaltzdelmar, try that transcoding before you don't eliminate the obvious you cant complain
17:50.04delmar:P
17:50.45delmarim interested in this timing thing anyway... whats the story... if there is ANY Zaptel hardware, there is a timer? ie X100 clone, or TDM card?
17:51.16docelmoIf you have Zaptel card timing is slaved from the card..
17:51.22docelmoNo card use clone or Ztdummy
17:51.40docelmoHave to watch with Ztdummy.. Have to have right USB chipset for it to work
17:51.43vetoAnyone played or have a SPA-941?
17:51.50delmarok
17:52.31delmarso on a system that has no Zaptel hardware whatsoever, I should setup ztdummy etc huh?
17:52.47delmaror is it not necessary unless trunking on IAX?
17:53.05fugitivoyou only want ztdummy for things like meetme where you need a timer
17:53.10shmaltzdelmar, you need for lots of other things as well
17:53.12fugitivoyou don't need if for iax
17:53.23shmaltzfugitivo, moh also works better with timing
17:53.47fugitivoshmaltz: using native moh too?
17:54.00shmaltzfugitivo, yep
17:54.07delmarwell I have a system up and running that is just * with no Zaptel hardware.. and it works great.. so since it aint broke.. i wont fix it :P
17:54.13fugitivodidn't know that :)
17:55.08shmaltzdelmar, I agree with you, my mother always says about my littel baby girl, if she is happy don't try to make her happier ;)
17:55.09delmarmoh?
17:55.15fugitivomusic on hold
17:55.16potsboycorrect me if i am wrong but ztdummy only comes into play with multiple calls
17:55.16shmaltz~moh
17:55.23*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
17:55.40shmaltzpotsboy, on iax trunking yeah, but scrollup to read more
17:55.48*** join/#asterisk oej (n=Olle@apollo.webway.se)
17:55.49potsboyoops..k
17:55.53delmarpotsboy, ok so u suggested timing issue.. but there is Zaptel hardware in both systems.
17:56.08shmaltz~moh
17:56.09jbot[moh] Music On Hold
17:56.34potsboyk... i though you said was only a sip network with no break out...why have cards then?
17:56.43delmari guess im wondering if there could be anything wrong with my IAX config between the * boxen... i dunno.. jitterbuffer on/off? dropcount? what is all that shit good for? :P
17:57.03delmarpotsboy, the cards are used.. and when they are used things are fine....
17:57.16delmarpotsboy, scroll up... i detail whats working mint.
17:57.45delmarand whats not...
17:59.35FuriousGeorgeim looking to  use something other than eyebeam
17:59.37delmarok so something else.. has anyone played with PA1688 based phones.. specifically any experience with IPPH203 (TigerNetCom aka TigernetCrap) because for the life of me I just cannot get these phones to do a simple call transfer.. got a couple of Grandstream BT101/102 phoens here and they are just awesome.. hold, transfer, hook-flash etc all works nice...
17:59.42FuriousGeorgeany suggestions
18:00.57iCEBrkrWhutup!
18:01.13docelmoShutup!
18:01.18docelmo:)
18:01.53docelmohay ice.. When do you want that box?
18:02.07FuriousGeorgei dont care if its isax or sip, but im wanna try something different than the xten/counterpath clients ive been using
18:02.15iCEBrkrdocelmo: Hang a sec.
18:04.01iCEBrkrdocelmo: Crap.  I'm gonna have to hold off on that box.
18:04.16*** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net)
18:04.46Inv_arpw00t I GOT POWER!!!
18:05.15Inv_arponly south fl residents would understand
18:05.21FuriousGeorgeive tried firefly but it doesnt seem to implement a transfer which kinda bothers me, nor does it seem to have the ability to take or make multiple calls
18:05.26docelmoINV ya ya ya
18:05.38docelmoI didnt have power last year.. I dont feel bad for you
18:05.39docelmo:)
18:05.50Inv_arp:)
18:05.57docelmoIm in Tampa BTW
18:06.29Inv_arpahh k you guys werent hit too hard?
18:06.50kuku5<PROTECTED>
18:07.10docelmome me!
18:07.15docelmoI terminate A-Z
18:07.20kuku5i need poland
18:07.24kuku5whast your fee
18:07.56docelmolemme go check my rate deck
18:07.59docelmohow much traffic
18:08.07kuku5v. little now
18:08.30docelmo0.0234
18:09.08docelmohttp://www.plainvoip.com/?page=showrates
18:11.43kuku5ouch
18:11.46kuku5too high
18:11.53*** join/#asterisk demetrio (n=demetrio@62.173.180.182)
18:11.56kuku5i get it for 1.9 now - high quality
18:11.58demetriohi everyone
18:12.06Inv_arpdemetrio: sup
18:13.24demetrioI just spent 2 hours trying to find out how blind transfer works in asterisk, and found nothing. does anybody here have a clue?
18:14.11kuku5ergh
18:14.14kuku5it just transfers
18:14.40js_birdboyas with many things in asterisk... it just works
18:14.43demetriosure, but how do they work? the only thing I know is that the # key is involed, that's all the documentation says.
18:14.49delmardemetrio, i have a couple of grandstream phones here attached to * and can xfer either hook-flash or use the transfer button
18:14.51docelmo•Inv_arp• Your happy to have your power arent you.. :)
18:15.14demetriolet's put it this way: what do I have to do to transfer a call?
18:15.16*** join/#asterisk Jake1932-mobile (n=Jake1932@18.sub-70-202-114.myvzw.com)
18:15.16delmardemetrio, sounds like you are not using the native transfer etc.
18:15.22docelmoflash
18:15.27demetrionope, I'm using SIP blind transfer
18:15.37js_birdboylike #extension does a blind transfer... is that what you are talking about?
18:15.41delmardemetrio, what device?
18:15.59demetriojs_birdboy: maybe, I don't know how it works, since it's not documented in any way
18:16.08FuriousGeorgei gotta run, anyone wanna recommend an alternative to eyeBeam (with a xfer button) for me?
18:16.44demetrioso if I set the "t" option in dial, the dialled phone will be able to transfer a call to exten simply pressing #exten ?
18:16.50js_birdboyon your sip device make sure your tones are set to allow DTMF (I think I got the abbreviation right)
18:17.41js_birdboyare you talkig about the t extension?  That is used for response/digit timeout
18:17.45demetriook, but can you please confirm that what I've just said is correct?
18:18.01delmaractually there is T and there is t.
18:18.05demetriowell.. the t option shoud be "allow the called person to transfer the call"
18:19.06demetrionevermind, I have to go now
18:19.12delmardoh
18:19.17*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)
18:20.32*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
18:22.37*** join/#asterisk diazonin (i=diazonin@114.222.8.67.cfl.res.rr.com)
18:22.40diazoninhey folsk
18:22.51diazoninI like asterisk, buts kind of hard to configure
18:23.01diazoningot it running on a Debian dual ppro 200,
18:23.07diazoninwonder if that is fast enough or not
18:23.36Jake1932-mobileOne sure way to find out
18:24.09diazoninwhat kind of Asterisk configuration tools do folks use?
18:24.17diazonina perl script or what
18:24.23Jake1932-mobileVi and nano
18:24.28IronHelixnano and notepad
18:24.30diazoninI figured as such
18:24.49diazoninyou folks use it actively?
18:24.57Jake1932-mobileYes
18:25.04diazoninwhat do you think of it
18:25.13Jake1932-mobileLove it
18:25.18IronHelixasterisk?
18:25.24diazoninyeah, asterisk
18:25.34IronHelixasterisk is quite possibly the best thing since sliced bread.
18:25.38IronHelixactually scratch that
18:25.41diazoninyou guys use softphones or hardphones and who do you all call with it
18:25.43IronHelixasterisk is better than sliced bread
18:25.52Jake1932-mobileHardphones
18:25.53Lostfrognano? eww.
18:25.59docelmoNANO RULES!
18:26.02diazoninwhat is your preferred hardphone
18:26.12LostfrogMight as well use dosemu and edit
18:26.12docelmoIf one can afford it.. Cisco
18:26.13diazoninlike one of those good looking ciscos or a cheaper brand
18:26.16Jake1932-mobile7960 cisco
18:26.18IronHelixSNOM phones are nice, polycom is nice too but good luck getting support
18:26.32diazoninwe have those ciscos at my school, NICE
18:26.36IronHelixpersonally i use a grandstream 2000, its cheap, it works, no major complaints
18:26.45docelmoIron!
18:26.51IronHelixdocelmo!
18:26.52diazoninwhat protocol you all use, SIP?
18:27.03docelmoDude.. Have you had any issues with them loosing registration after 1 day or something?
18:27.11IronHelixSIP for connecting to phones and providers if the use it, otherwise i try to use IAX whenever possible
18:27.13LostfrogThat is a religious war, diazonin. Mostly SIP or IAX
18:27.19*** join/#asterisk Tobias76 (n=tk@01230.info)
18:27.20IronHelixyeah
18:27.26diazoninmy bad, just wondering what I should use
18:27.27Tobias76hi
18:27.33IronHelixdocelmo- never had any reg problems, but they are all on a LAN
18:27.42IronHelixno worries diaz
18:27.47docelmoAre you DHCP or Static?
18:27.56IronHelixstatic mapped dhcp
18:28.00diazoninhow do you like using the phones over the internet, any problems with WAN use?
18:28.21diazoninI had vonage for a while, but just using an ATA they provided not using Asterisk, it was ok, but had some problems
18:28.27IronHelixconfig issues can take some working otu, especially when there is NAT involved but overall once configured correctly they are very reliable
18:28.36Jake1932-mobileI use it everyday for hours at a time over the internet - works fine
18:28.46IronHelixdiaz try broadvoice or quantumvoice both will support asterisk
18:28.57diazoninwhat are the rates like?
18:29.12IronHelixboth sell 'lines', about $20/mo for unlimited us/canada
18:29.14docelmoIron what are your peers setup like? Can I see one to compair? I am loosing my after about 24/48 hours then they will not reregister
18:29.24diazoninthanks for the advice folks
18:29.26IronHelixdocelmo- same LAN?
18:29.30docelmoyep
18:29.38*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
18:29.48IronHelixdiaz- if you want to buy minutes wholesale, there are a great number of companies that will sell you minutes for 1-3 us cents each
18:30.02Jake1932-mobile2c a minute with asterlink
18:30.30IronHelixalso diaz check out http://www.voip-info.org  anything and everything about voip can be found there
18:30.36IronHelixdoc- sure stand by
18:31.58*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:31.58kuku5was anyone able to change the ring on a gxp 2000?
18:32.11IronHelixyeah
18:32.12docelmono
18:32.13IronHelixits easy
18:32.19kuku5ergh
18:32.23IronHelixuse their crappy ringtone generator tool
18:32.25kuku5can you send me a tone thats ready ?
18:32.36IronHelixif you have the latest firmware 1.0.1.9 or 1.0.1.12 it will work
18:33.23IronHelixsure, stand by
18:38.33*** join/#asterisk justinu (n=j2@72.18.13.48)
18:38.40IronHelixkuku check dcc send
18:40.36*** join/#asterisk ronn (n=ronn@62-249-247-240.no-dns-yet.enta.net)
18:42.13*** join/#asterisk rubble (n=rubble@wbs-146-186-81.telkomadsl.co.za)
18:43.49ronnis there anyway to stop asterisk from generating ringing tone? even when not using the r optin in Dial() ?
18:44.14docelmoNo.. if no r is present then your getting progress from the endpoint
18:44.20ronni have the following Dial(Zap/g1/${EXTEN}|60|g||300)  but still * generates ringing
18:45.20IronHelixm?  makes music on hold
18:45.20ronndocelmo: with the above command i'm getting ringing both from * and the end point
18:45.36docelmoNo.. Just the PSTN
18:45.49hardwireok
18:46.00hardwirewould I be retarted to just lock down production to 1.2.0-beta2
18:46.11hardwireand gradually phase it up as betas are released.. vs CVS weekly?
18:46.21ronncan u force asterisk from generating the ringing tone?
18:46.28docelmoyes
18:46.29docelmoadd the r
18:46.51hardwireronn: from using it.. play a silence MoH?
18:47.32ronnhardwire: how do i do that?
18:48.05ronndocelmo: i'm getting the ringing with and without the r option.
18:48.30docelmothe PSTN PSTN PSTN is giving you progress which is RING in telco terms
18:48.31hardwireronn: read up on the musiconhold.conf
18:48.34hardwireand cmd dial
18:48.58hardwiredocelmo: taking the teletubbies learning approach?
18:49.09ronnthanks
18:49.18docelmoif it helps him understand..
18:49.31IronHelixlol
18:49.39hardwirethats so mean
18:49.51IronHelixouch
18:50.00hardwireugh
18:50.10hardwireI hate it when I become a brown noser due to the hole at the top of coffee lids
18:50.26IronHelixeasy to fix
18:50.35IronHelixdrink your coffee thru one of those stirrer straws
18:50.38IronHelixthe tiny thin ones
18:50.43IronHelixcools it too on the way up
18:50.54hardwireIronHelix: Ah.. thats how I get it into my eye.
18:50.59hardwireI avoid if at all possible
18:51.33IronHelixhehehe
18:51.34hardwirethe elastic suction vortex I create when wolfing down coffee tends to keep the flow going well after I have released the straw from my lops
18:51.46hardwirelops/lips
18:52.01Beirdostraw?
18:52.09IronHelixyeah stirrer is not recommended if you drink coffee fast
18:52.11Beirdowow
18:52.26hardwireok
18:52.27hardwireI don't care
18:52.30BeirdoI punch a second hole in the lid to get airflow
18:52.35hardwireI am using digitally imported sky.fm streams on my music on hold
18:52.39*** join/#asterisk svenna_ (n=svenna@p548D2D1E.dip0.t-ipconnect.de)
18:52.45Beirdoa venti latte will be drained in about 1 min
18:52.59hardwireBeirdo: thats my original problem.. the tilt required to wolf at such speeds quickly sends coffee out the upper hole onto my nose
18:53.07Beirdoheheh
18:53.11Beirdotrue enough
18:53.13Beirdohmmm
18:53.18hardwirethere must be a better way
18:53.29Beirdocould drink it like college students drink beer
18:53.31hardwireto avoid such coffee injuries as in the eye via straw.. and embarassing brown noser moments.
18:53.35hardwireBeirdo: hahahaha
18:53.36Beirdocoffee bong time
18:53.43hardwireJUST MIX IT AS IT GOES DOWN!
18:54.28hardwirea new coffee transport system will still however never address the coffe out of the nose at a humorous slashdot post delimma
18:54.36Beirdotrue
18:54.39BeirdoI got it.
18:54.40BeirdoIV
18:54.47IronHelixhehehe
18:54.50*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
18:54.54hardwirewho needs coffee then.. extract caffeine and plug in
18:55.01*** join/#asterisk moles (n=martin@85.194.2.201)
18:55.57hardwireok.. this channel is putting me to sleep
18:56.28Beirdohehe
18:56.59hardwireI really want to get some radio stations as the MoH here
18:57.01hardwirenot a problem
18:57.08*** join/#asterisk juanjoc (n=jcomella@OL48-53.fibertel.com.ar)
18:57.10hardwireI just need to get my radioshark a linux driver
18:57.29hardwireor find a rackmount tuner :)
18:57.49*** part/#asterisk moles (n=martin@85.194.2.201)
18:59.57hardwireso.. I need to use rec to go to sox to resample the souund from say 22khz to 8khz with its quadratic resample filter
19:01.54*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
19:01.56*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
19:04.18*** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
19:04.22MuppetMasterHello.
19:04.30hardwireherro
19:04.30Flautohello
19:04.48MuppetMasterWhen using the Async setting in the Manager API Originate action, how does one tie the call results to the original action requests?
19:05.22MuppetMasterDoes the ActionID return for every response/event to an originate action?
19:06.44jalsothi
19:07.06*** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
19:07.18jalsothow can I use res_features options [like atxfer] on a channel called from queue to an agent?
19:08.08jalsotwhen I set mapping for #1, I receive them, however they are ignored, no atxfer applied
19:08.13jalsotany idea?
19:09.18*** join/#asterisk santiago (n=santiago@208.195.215.124)
19:17.06justinuis manx here?
19:17.52hardwirerawr
19:18.22justinumuppetmsater: try asking on #asterisk-dev
19:18.39justinuthey're like smart and stuff
19:21.08LostfrogHas anyone had a problem with the hookswitch on snom 3x0s?
19:21.37LostfrogI just plugged in two of them in a row in which the hookswitch doesn't work.
19:21.54LostfrogThey're from two different boxes from the factory.
19:22.46*** join/#asterisk algorithmn (n=na@ool-44c29ac5.dyn.optonline.net)
19:23.55*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
19:24.53iCEBrkrHey, if I put a callfile in outgoing and I'm using the GroupCount() stuff-- and the group is 'full' will that callfile get deleted or will it 'queue' it for late when some ports are free??
19:25.09algorithmnmy current asterisk/realtime/mysql setup crashes when i have an extension of 900 registered with the system.. it crashes on any outgoing calls..
19:25.30iCEBrkrI'm assuming maxretry= should 'retry' the file if no ports available.
19:26.40Damintwisted?
19:26.48fileis a twisted individual
19:27.14iCEBrkrfile: I think we're all a bit twisted.
19:27.22LostfrogA bit?
19:27.23Lostfroglol
19:27.24iCEBrkrfile: Especially that Damin guy.
19:27.56iCEBrkrhttp://www.cyberdyne.org/~icebrkr/files/NACS_NET-Fake_Jerky_Boys.mp3
19:29.09LostfrogNo help on broken hookswitches on Snom 320s?
19:29.26*** join/#asterisk javo (i=1000@144.13.53.210)
19:31.02*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:31.19iCEBrkrW00t!
19:31.34iCEBrkrI'll just put some insane amount of retry times in here to make sure the call is made.
19:32.09*** join/#asterisk FuRR_ (n=meeps@bko29.chapman.edu)
19:32.46javoI have asterisk 1.0.9, and it will run just fine.  Problems arise when I try to use zaptel.  If chan_zap exists, asterisk will not load.  when i try to modprobe wcfxo, i get Unable to open master device '/dev/zap/ctl'
19:33.00javoi have udev, and there is no /dev/zap at all
19:33.11Lostfrogninety-nine million, nine hundred and ninety-nine thousand, nine hundred and ninety-nine retries.
19:33.32Lostfrogmodprobe zaptel?
19:33.41javonope, zaptel is loaded
19:33.59KranZjavo, what version of kernel?
19:34.05javothey both show as loaded, actually
19:34.06iCEBrkrLostfrog: Yes.
19:34.06javo2.6.11
19:34.32KranZyou got the latest cvs?
19:34.44javoi read about needing rules in /etc/udev/rules.d but they are already there
19:35.05javolatest cvs of *?
19:35.08KranZwhat does dmsg say when u try to load it
19:35.10Lostfrogok.. I'm not running udev anywhere.. can't help.
19:35.10KranZyeah
19:35.18javono i have stable 1.0.9
19:35.34javowcfxo: DAA mode is 'FCC'
19:35.34javoFound a Wildcard FXO: Generic Clone
19:35.38KranZmake sure you have 1.0.9.2 zaptel
19:35.43javoi do
19:35.44KranZthey fixed a compile bug
19:35.47*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
19:36.02ms345can anyone suggest why this doesn't set my outbound callerID?  from pri debug span:    Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) '4044257828' ]
19:36.29*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
19:36.29javozaptel-1.0.9.2
19:36.40ms345but it shows the base number of my pri, not 7828
19:37.18KranZyour telco might not allow the number you're trying to set
19:37.28javoit seems that the module loads fine, but it doesn't give me anything in /dev
19:37.37*** join/#asterisk makhtar (n=ageller@mail.bulletinnews.com)
19:37.38ms345i was able to set it with my nec pbx (now behind *)
19:37.58*** join/#asterisk aetg_bob (n=aetg_bob@70-35-141-64.ironoh.adelphia.net)
19:37.59KranZjavo, u googled?
19:38.02javoya
19:38.06ms345i tried 11 digits, 10 digits, 7 digits and 4 digits - all the same
19:38.09KranZms345: what method u using to set the callerid?
19:38.23javoall i can find is stuff about /etc/udev/rules.d
19:38.31javowhich appears to be correct on my system
19:38.40ms345Executing SetCallerID("Zap/92-1", "4044257828") in new stack
19:38.59*** part/#asterisk aetg_bob (n=aetg_bob@70-35-141-64.ironoh.adelphia.net)
19:39.08KranZset the ani
19:39.17KranZwith the ,a flag
19:39.39KranZor set(callerid(ani) = 4044257828)
19:39.50*** join/#asterisk Corydon-w (i=red@pdpc/supporter/sustaining/Corydon76-home)
19:40.15ms345KranZ - will try it - thx a bunch!
19:40.38*** join/#asterisk Math` (n=math@modemcable157.3-81-70.mc.videotron.ca)
19:42.21KranZms345: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid
19:42.59brettnemHey anyone know why I can't get asterisk to STOP sending 407s to SER? It looks like asterisk is trying to authenticate the original caller who placed the call TO ser. I'm using insecure=yes
19:43.03Math`I got * setted up on a router with multiple internet connections, is there any way to tell asterisk to bind to a certain ip for *OUTGOING* connections?
19:43.13*** part/#asterisk makhtar (n=ageller@mail.bulletinnews.com)
19:43.53Math`that is for sip, bindaddr= listens on that ip (and I still need to listen on the lan, even if Im using an external interface to register to an sip server
19:44.36Math`(or would the best way to do it be iptables?)
19:44.46*** join/#asterisk lmergen (n=Leon@grib.btcnet.nl)
19:44.56brettnemugh.. i can't get rid of this 407 :(
19:45.00lmergenhello, i have a question
19:45.27KranZbrettnem: are calls coming through * to ser, or ser to *
19:46.05lmergenat http://www.voip-info.org/wiki-Asterisk+auto-dial+out , it says i can use the "Data: " parameter in /var/spool/asterisk/outgoing files to pass data to the application - first of all, is this "application" Asterisk, and if so, how can I read that variable ? At the "Asterisk variables" page, nothing can be found about a "Data: " parameter
19:47.19Math`lmergen: I usually pass data as an extension to a context, tho this depends on what you want to pass
19:48.41*** join/#asterisk pun` (n=jeod20@adsl-067-034-127-246.sip.mia.bellsouth.net)
19:49.06pun`hi
19:49.26filebrettnem: I bet it's matching a user entry in sip.conf and asterisk is challenging :)
19:49.57*** join/#asterisk Darwin35 (n=richard@208.139.193.178)
19:50.22pun`hey guys, any wifi iax phone available yet?
19:50.22Darwin35ok who here has asterisk on fbsd and has a /usr/local/etc/rc.d asterisk.sh they can show
19:50.42Darwin35I need to see what I am missing
19:51.31Darwin35anyone ?
19:51.33Darwin35pls
19:51.38Math`pun`: not that I know
19:51.47Darwin35I know someone here is using fbsd
19:51.48Math`Darwin35: whats the problem your having
19:51.54javook, i just read in README.Linux26 that I need to have  CRC-CCITT enabled in the kernel.  However, when I go to enable it in make menuconfig, it does not give me an option to select it.  It only gives me ---
19:51.55drumkillaDarwin35: keep spamming the channel, and it might change my mind
19:51.57lmergenMath`, but that wouldn't allow for variable data, right ? I mean, i have to make dialplans for specific extensions, and can't do a "catch all" dialplan for all extensions, right ?
19:51.59Math`Im not using fbsd but maybe I can help
19:52.07*** join/#asterisk brookshire (n=nubb@gateway.digium.com)
19:52.13javodoes anyone know what has to be enabled to be able to enable  CRC-CCITTZ
19:52.17Darwin35drum
19:52.23Darwin35I have not been on in awhile
19:52.35Darwin35I just need to look at a working asterisk.sh
19:52.46Math`u tried googled?
19:52.48Math`google*
19:53.07Darwin35you get the asterisk.sh.in file
19:53.21*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
19:53.22Darwin35I need to see the changes made to the file
19:54.18Lostfrogjava: modprobe crc32c before loading zaptel and wcfxo
19:54.53Lostfrogjavo, even.
19:55.07LostfrogI put crc32c in /etc/modules
19:55.30javoLostfrog, i do not have the crc32c module because it will not let me select it in make menuconfig
19:55.49javoit shows --- instead of < >
19:56.00sahafeezwhat should i use in order to record custom greetings, etc..
19:56.30javoLostfrog, do you know what i need to enable before i can enable crc32c?
19:56.38LostfrogOne moment.
19:57.31*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
19:57.59brettnemHey is there anyway to get asterisk to NOT look in the From: header for authentication?!
19:58.09LostfrogDid you enable crc32c in Cryptographic options.
19:58.10Lostfrog?
19:58.14*** join/#asterisk jonarild (n=vircuser@195.28.172.4)
19:59.44jonarildsomeone with experience in zaphfc?
20:01.11LostfrogI don't know javo.
20:01.17LostfrogI have to run.
20:01.50jonarildinsmod: error inserting 'zaphfc.ko': -1 Invalid module format
20:01.50jonarildIm getting ths error: insmod: error inserting 'zaphfc.ko': -1 Invalid module format
20:02.08pun`Math, thanlks for your reply
20:02.18javook thanks anyways
20:02.23pun`anybody that knows about a wifi iax phone?
20:03.41jonarildor someone that knows about a SIP wifi phone that supports WPA encryption
20:03.46brettnemwhy does asterisk assume I want to authenticate if the from sip uri matches a valid peer? that's insane!!
20:06.20*** join/#asterisk jonarild (n=vircuser@195.28.172.4)
20:07.50azziewhat's the code to pick up an extension which is ringing? using AMP...
20:12.26MadkissUhm, I still need an asterisk developer.
20:14.45*** join/#asterisk mxmasster (n=mxmasste@ppp-71-138-119-74.dsl.irvnca.pacbell.net)
20:14.47mxmassterhi all
20:17.00mxmassteri need to configure some time based rules in asterisk - i.e. if it is between 8-6 m-f ring phone a, otherwise ring phone b
20:17.09mxmassterwhere can i find an example of how to do that
20:17.24delmarwell that was easy... #1 and *2 for blind and attended xfers works fine. ill just use that and stop fighting with the silly tigernetcrap transfer button :P
20:18.02GXTimxmasster: iirc the include directive does that
20:21.22mxmassterGXTi: you don't happen to have an example i can look at?
20:22.06iCEBrkrblah blah blah
20:22.16pun`someone here mentione some time ago that converted a sip wifi phone to a iax
20:22.20pun`mentioned..
20:22.37docelmoso ice you want that server? I now have 30Mb down and 5Mb UP!?!?!?!? YES!!!!!!!!!!!
20:22.45iCEBrkrLOL
20:22.52iCEBrkrdocelmo: Talk to me next week :P
20:23.14iCEBrkrdocelmo: I'd hate to give you half now, half later.. That's just shady on my part.
20:23.32iCEBrkrLike I said, I gotta make sure I have $$$ incase something else comes up.
20:24.24docelmothis is true.. I am moving the boxes down now.. Also I noticed the guy who did the code for X10 on linux is giving up his development position.. I emailed him about it. I am gonna make app_x10.so for asterisk out of it.
20:24.37iCEBrkrhahah
20:24.53docelmoWhat?
20:25.03docelmoI think its a good idea.. Then you can control it from the dialplan
20:25.05docelmo:P
20:25.07iCEBrkrexten s,1,x10(N)
20:25.26iCEBrkrturn on all devices in housecode
20:25.36tzangeriCEBrkr: sweet
20:25.37iCEBrkrI guess that'd be more like..
20:25.43iCEBrkrexten s,1,x10(A,N)
20:25.50docelmotz, Im the one going it.. Not ice.. haha
20:25.52iCEBrkrTurn on ALL 'A' devices.
20:26.04iCEBrkrdocelmo: Pff, I gave ya the idea.
20:26.07lmergenhey... say i have "Channel: Zap/g1/1234567890" in a /var/spool/asterisk/outgoing/file , how can i retrieve the number being dialed inside my extensions.conf ?
20:26.09iCEBrkr:-|
20:26.19tzangerlmergen: ?
20:26.20iCEBrkrdocelmo: Make it happen! Make it happen!
20:26.59docelmoOnce he says I can run with it. I am gonna start beta development on it. 2 trees 1 for asterisk and 1 for cLI
20:27.16iCEBrkrdocelmo: You using the bottlerocket code?
20:27.18[TK]D-Fenderdocelmo: I have an CM11A package I use for * to control my stuff at home...
20:27.21docelmoyes
20:27.24iCEBrkrsweet
20:27.31lmergentzanger, i want to fix an auto-dialing script, but in order to execute the Dial() command in dialplan, I need the number to be dialed... how can I retrieve that number ?
20:27.36[TK]D-Fenderif youwant I can send it to youl later
20:27.38docelmoI also checked.. The bottlerocket will control more than that..
20:27.41iCEBrkrdocelmo: I should make that a PHP module :P
20:27.44docelmoya would like to check it out
20:27.47tzangerlmergen: you should be able to pick up ${EXTEN}
20:27.58docelmoI am gonna make it internal to asterisk
20:28.10iCEBrkrmod_x10
20:28.21iCEBrkrNo more system() calls!!!
20:28.40[TK]D-Fenderdocelmo: It installs a single bin file really and I use System(heyu2 a1 on) for example from my dial plan.  is that sufficient for your needs?
20:28.58lmergentzanger, but in my dialplan I define "exten => s,1,Dial(Zap/g1/<<number>>,10)" for example... then my $(EXTEN) always is "s" ... or is there some way to define a dialplan for a "catch-all", all numbers ?
20:29.19iCEBrkrdocelmo: Start the Home Automation with Asterisk project!
20:29.22docelmoI need the source code which is what ice and I are talking about. I am gonna make a app_x10 to control all devices on yoru network from the asterisk dialplan
20:29.35docelmoHell ya dude.. Wanna help?
20:29.38iCEBrkrFor sure!
20:29.43iCEBrkrI still have all my X10 shit
20:29.49iCEBrkrI know C.
20:29.56iCEBrkrok, I can hack away at C
20:29.58docelmoIm gonna buy some.. I wanna setup my house.. I know of it. Not fluently yet..
20:30.33tzangerlmergen: then set a channel var for the call file to pick up
20:30.55lmergenhmmm k
20:31.22[TK]D-FenderWhat we really need for X10 is to help the guys working on the USB one.  Thats where the focus should be.  I'm still using my old serial one for the time being and won't pay for new stuff until they get it working decent
20:31.31*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
20:31.40lmergenbut say I set a "Channel: Zap/g1/1234", how can I pick up that "1234" in my extensions.conf then ?
20:31.49*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
20:32.07iCEBrkr[TK]D-Fender: Is there any effort for the USB version?
20:32.18docelmoWe can start on that.. USB is treated almost the same as serial
20:32.26*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)
20:32.33[TK]D-FenderiCEBrkr: Yeah, but its rusty right now.  they are reverse engineering it since X10 isn't helping much (while not hurting)
20:32.54iCEBrkr[TK]D-Fender: It should be the same shit as the serial version
20:32.57[TK]D-FenderLazy asses.....
20:33.01iCEBrkrOh, I guess you have to talk to the USB device.
20:33.05iCEBrkrhrrm.
20:33.16[TK]D-FenderiCEBrkr: No they changed quite a bit since it accepts all house/unit codes, and the extended versions.
20:33.26iCEBrkrand on the serial device you blip the terminal ready line and shit
20:33.36[TK]D-FenderThey should use a model like SIP/HTTP :D
20:33.55BlackthornHello. My * is on a real ip. I have a sipura which is behind a linksys nat router. * is set nat=yes and notify=yes.  On the console I get error message that one of the two lines on the sipra won't register. and the ip is 192.168.1.2
20:33.57iCEBrkrHrrm. Mine accepts all house/unit codes.
20:34.08[TK]D-Fenderthat way you could telnet it.... in fact, thats the next thing they should do!  A networked version!
20:34.20[TK]D-FenderiCEBrkr: Your serial version?
20:34.26iCEBrkr[TK]D-Fender: Yeah
20:34.43iCEBrkrbr a on
20:34.47iCEBrkrbr b on
20:34.49docelmoWell initially we will work with the serial version then move to usb down the road
20:34.53[TK]D-FenderWell you serial one should only work on one housecode at a time.  Thats last I recall from the manual dial on it.
20:35.31docelmoWell how many house codes would you need for 1 house?
20:35.36iCEBrkr[TK]D-Fender: I'm using bottlerocket. I specify the house code/unit number on the command line.
20:35.40[TK]D-FenderLook for the HEYU2 homepage and DL it from there.  Works dead simple and should be enough for you to make an app_x10 out of.\
20:35.46iCEBrkrdocelmo: Um, you can have a lot!!
20:35.48[TK]D-FenderGoogle = good
20:36.08iCEBrkrA-F, 0-9
20:36.10docelmoHEYU2 is USB?
20:36.28docelmoohh.. Well if 1 at a time then serial should be fine.. But the interface is interesting..
20:36.40[TK]D-FenderiCEBrkr:  yeah I need a whole whack since I intend to virtualize my codes so I can use the slim-line switches for custom controls remapped by * for me.
20:37.08[TK]D-FenderHEYU2 is serial for the older controller.  I don't have any links for the USB one yet since I don't own it yet.
20:37.27delmarok i have had little sleep so i might not be thinking streight.. anyway.. looking at attended and blind xfers using Dial(Tt) ...I want to have extensions to have both T and t, so that a call received OR made can be transferred etc...but doing this seems to make it that say.... an incoming PSTN caller could transfer themselves... what can i do about that.
20:37.32iCEBrkr[TK]D-Fender: You have that little serial port 'pass-thru' thing?
20:37.40docelmobottlerocket is fine for that then.. I have looked at the code its not very hard to understand.
20:38.10iCEBrkrdocelmo: yeah, it's pretty simple looking
20:39.06iCEBrkr${VARIABLES} are available to [macros] right?
20:39.55*** join/#asterisk falz (n=falz@proxy.supranet.net)
20:40.18docelmoFound linux drivers for the CM15A computer interface
20:40.21docelmofor USB
20:40.29docelmoCan you say USB app_x10.so?
20:40.34iCEBrkr:)
20:40.47falzanyone with cisco 7960's know if there's a way to force the caller ID on transferred calls to be the ORIGINATOR instead of the TRANSFERING PARTY?
20:40.55*** join/#asterisk Flauto (n=zhao@71.194.39.175)
20:40.56*** join/#asterisk test34 (n=test34@unaffiliated/test34)
20:41.04[TK]D-Fenderdocelmo: I found that source too, but never actually tried it.  They have some of the basics right now, but a lot is still missing.
20:41.05falz(as 7960's do with blind xfer)
20:41.24docelmook going to smoke.. brb
20:41.44NuggetI quit smoking 8 years, 5 months, 4 weeks, 2 days, 1 hour, 41 minutes, and 44 seconds ago.  During that time, I would have smoked 68,247 cigarettes. (That's like smoking a 3.23 mile-long cigarette)  By quitting, I've saved $11,943.23!  I've avoided inhaling 1.77 kg of tar, 109 grams of nicotine, and 1.09 kg of carbon monoxide.
20:42.31[TK]D-FenderAnd now have a lot of free time to calculate your savings!
20:42.50[TK]D-FenderYou should run a ticker like the Jerry Lewis Telethon!
20:43.21iCEBrkr[TK]D-Fender: he has that in a frame hung on his cubical wall.
20:43.50*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
20:43.56BlackthornHello. My * is on a real ip. I have a sipura which is behind a linksys nat router. * is set nat=yes and notify=yes.  On the console I get error message that one of the two lines on the sipra won't register. and the ip is 192.168.1.2
20:44.02[TK]D-FenderCould have saved a few bucks on the frame and increased his savings ;) Use recycled paper and Scotch tape next time :D
20:44.12*** join/#asterisk vitamintwo (n=vitaminm@wza.us)
20:44.45iCEBrkrBlackthorn: um ok?
20:44.50vitamintwoI've got a snom 320, and the programmable keys' status lights are behaving erratically. Anyone have experience with this?
20:45.25Blackthornheheh - oops.. I'm guessint it won't register because it is coming up with 192 address. and i'd like to know what I need to to "fix" this isse
20:45.31falzBlackthorn: if asterisk is on a public IP, why is NAT=yes?
20:45.44falzso your asterisk box is dual homed?
20:45.54iCEBrkrfalz: I think he meant the sip entries are set to yes.
20:45.54Blackthornnat=yes in the respective sip config for that user only.
20:46.07Blackthornno not dualhomed and the *box is not nat'ed
20:46.34shido6nat=yes should be in the user/peer/friend
20:47.27delmaruh wtf shido6 ?
20:47.27iCEBrkrBlackthorn: Your 'Proxy' setting in the Sipura is set to the IP of your asterisk box, right?
20:47.27falzso is 192.168.1.2 the "outside" ip of the NAT device?
20:47.27Blackthorncorrect ice.
20:47.37docelmook Im back
20:47.50iCEBrkrBlackthorn: UserID is equal to [whatever] in your sip.conf?
20:47.51delmardocelmo ...
20:48.05docelmoyes?
20:48.08delmardocelmo, do u know anything about using T or t with Dial command etc.. ?
20:48.18docelmoWhat do you want to know?
20:48.23docelmoIts in the wiki..
20:48.25delmarok here is whats happening...
20:48.32shido6careful with T and t , you give the caller more access to your dialplan
20:48.40BlackthornSee what is wierd is that line2 registers using the ip address of the linksys router.  line1 fails registration from 192.168.1.2 which is the sipura unit behind the linksys.
20:48.43iCEBrkrdelmar: T and t don't work the way you're thinking..
20:48.43delmarsure it is.. but im tired and this might be easily fixed or its just tough shit.. im not sure...
20:48.55delmarok sec.. here is whats happening...
20:49.25falzBlackthorn: asterisk shouldnt know 192.168.1.2 exists if it really is NAT'd though
20:49.32delmarExtensions are setup with T and t .. so they can transfer the call made.. OR received...
20:49.45*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:49.49iCEBrkrBlackthorn: Proxy should equal the public IP of your asterisk box.
20:50.11iCEBrkrdelmar: Bzzzt
20:50.14Blackthornfalz: I agree. Icebbkr: yes proxy on both lines are set to the * box.
20:50.20delmarI have a line to take care of incomign calls... with just t which calls other extensions.... when one answers... it can transfer.. and the caller on the PSTN.. cannot transfer...
20:50.31delmariCEBrkr let me finish :P
20:50.46delmarbut when the extension transfers the PSTN caller to another extension....
20:51.09delmarat that point... the PSTN caller OR the new extension.. can make a transfer.. with *2 or #1
20:51.17iCEBrkrdelmar: T allows any caller transfer the call, which is something you really don't want.
20:51.21delmarnow using Tt on extensions works perfectly...
20:51.23*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
20:51.30delmariCEBrkr, exactly...
20:51.52delmariCEBrkr i want this to be available to the extensions.. but not an outside caller... mind you...
20:51.54iCEBrkrdelmar: So you want anyone to be able to hit * and arbitrally transfer themselves??
20:52.03delmarwhat can they do... make a transfer and get disconnected.. LOL
20:52.21delmariCEBrkr, err ok it works like this....
20:52.25iCEBrkrdelmar: or accidentally hit * and got WTF?
20:52.33iCEBrkrs/got/go
20:52.50delmariCEBrkr, no its the default of *2 attended or #1 blind right now... so not simply *
20:53.04*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
20:53.09lesouvagedocelmo: I have some strange problem downloading phpagi version 2.14 from sourceforge. It's not the full size (21 kb instead of 35kb) and I can't  untar the archive. s there anither url where I can download phpagi?
20:53.09iCEBrkrdelmar: Is that something new?
20:53.15iCEBrkrdelmar: cuz it's always been *
20:53.32docelmoDownload from CVS
20:53.59iCEBrkrphpagi sucks! :D
20:54.09vitamintwoSo is anyone experienced with snom 320 wonkiness?
20:54.12delmariCEBrkr, the default in the features.conf is *2 and #1 i believe...
20:54.26delmariCEBrkr, ok just checked the sample.. and there isnt even anything in that.. so its whatever u like :P
20:54.33delmaranyway... that asside...
20:54.39docelmoice eat me
20:54.42iCEBrkrdocelmo: hehe
20:54.59iCEBrkrdocelmo: Don't make me go through that thing and clean it ALL up!
20:55.08iCEBrkrSince I'm a PHP nut, I just might have to.
20:55.18docelmoDo what you want.. I dont care.. I only did a few parts in there
20:55.26iCEBrkrSweet
20:56.03delmariCEBrkr, lets say extension1 makes a call via the PSTN, then that called party wants to speak to the boss on extension2... :P
20:56.09*** join/#asterisk Nix (n=Nix@81.213.125.220)
20:56.24Darwin35exit
20:56.44delmariCEBrkr having t only means u can transfer the call that is received... having T means u can also transfer the call you placed.... so extensions need to do both...
20:56.56delmariCEBrkr, or am i looking at it all wrong? how can i overcome this?
20:57.02*** join/#asterisk jarrod (i=jarrod@juniperyour.net)
20:57.16delmar<--- not enuf sleep
20:57.21jarrodwhere is cripito
21:04.45brettnemhey anyone know how to add a sip header on a call that originates on a PRI and goes OUT SIP?
21:05.23brettnemaparently sip_addheader only works on calls that originate SIP
21:07.13mxmassterwhat the dial string to ring multiple lines at the same time?
21:07.53falzDial(SIP/103&SIP/203,8)
21:08.01falzthat'll dial x 103 and 203 for 8 seconds
21:08.18mxmassterthanks
21:10.21*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
21:11.34mxmassterhow do i answer a call, and play music while i ring extensions?
21:12.10file[laptop]mxmasster: show application dial, type that on the CLI and read all the available options
21:12.10falzwhile on hold?
21:12.47mxmassterwell, maybe hold isn't the right word
21:13.04mxmassteri want to play music to the caller while the phones are ringing
21:13.13file[laptop]mxmasster: do what I said, there's an option to do it
21:13.16*** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-11-9.w81-248.abo.wanadoo.fr)
21:13.56file[laptop]'m[(class)]' -- provide hold music to the calling party until answered (optionally with the specified class.)
21:14.14mxmassterthanks
21:14.17mxmassterjust reading that
21:22.42*** join/#asterisk mflorell (n=mattf@rrcs-24-173-158-34.se.biz.rr.com)
21:24.08*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
21:24.35*** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3)
21:24.50iCEBrkrdelmar: I'd make sure your inbound and outbound contexts are separate so you can put T on outbound from internal extensions.
21:25.15iCEBrkrdelmar: ...and I've never gotten attended transfers to work.
21:25.42delmariCEBrkr, ok ill take a look at that and think about it...
21:25.51delmariCEBrkr, working mint here.
21:26.12*** join/#asterisk daddy_23 (n=daddy@200.93.234.1)
21:26.17iCEBrkrdelmar: Which version are you using again?
21:26.46delmarto be honest.. its been a while since i updated.... and i have no idea :P
21:26.48delmarsec.
21:26.52iCEBrkrshow version
21:27.21*** join/#asterisk _DAW (n=bob@adsl-156-91-219.msy.bellsouth.net)
21:27.22delmarAsterisk CVS-HEAD-04/14/05-17:26:18
21:27.41_DAWHello
21:27.52delmartime for an update for sure.. but nothing much is b0rked atm.
21:28.02delmarso no real need.
21:28.22delmari might update if i get too frustrated with that * to * via IAX echo issue
21:29.05_DAWDoes anyone know if it is possible to pass args to a macro via the M option in the Dial command?
21:30.44*** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca)
21:30.59sahafeezquestion: what app would people recommend to record greetings (not for voice mail but the for the menus). can I use some function in asterisk?
21:31.14iCEBrkrdelmar: F Bleeding-edge.
21:31.19_DAWsahafeez - record command
21:31.23iCEBrkrdelmar: My 'production' machine was 1.0.9
21:31.30sahafeez_DAW where is that?
21:31.54iCEBrkr_DAW: um.. what??
21:31.55_DAWhttp://www.voip-info.org/wiki-Asterisk+cmd+record
21:32.22iCEBrkr_DAW: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
21:32.25sahafeez_DAW: thank you. reading now.
21:32.33delmariCEBrkr yeah, u gotta stay a bit behidn the times sometimes.. to stay stable and reliable etc.
21:32.49iCEBrkrdelmar: I JUST upgraded to 1.2.0 Beta2 last night
21:33.03iCEBrkrIt wasn't as smooth as I thought it'd be.
21:33.06*** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet)
21:33.11_DAWiCEBrkr - for the M option you mean?  I read up on it there but there is no mention of it passing args like the Macro command.
21:33.12iCEBrkrBut whatever, I got it all working
21:33.14delmariCEBrkr, i was gonna do that on my mess about box here that i just showed u the version...
21:33.22iCEBrkr_DAW: Re-read, it's there.
21:33.22delmariCEBrkr, hows the beta running.. any good?
21:33.39delmariCEBrkr, does it have rx_fax / tx_fax support?
21:33.45iCEBrkrdelmar: Works like a charm.  I don't do faxing, so I dunno
21:34.00delmariCEBrkr, yeah i wanna play with receiving fax's next
21:34.10iCEBrkr_DAW: You can now add args to the macro by using a '^' char
21:34.20delmariCEBrkr, last time i tried to setup a macro and stuff it was a pain and didnt work so i gave up but ill have another crak
21:34.36iCEBrkrdelmar: <3 Macros
21:34.49_DAWiCEBrkr - just saw it.  should be with the M command documentation not all over the page..
21:35.15iCEBrkr_DAW: Um, RTFM, K/THX/BYE
21:35.21*** join/#asterisk Corydon-w (i=grey@pdpc/supporter/sustaining/Corydon76-home)
21:35.38iCEBrkrAll I gotta say is.. *I* found it.. and I'm half-retarded.
21:35.40mxmassteris there a place i can post my dialplan for help - i am trying to do a time of day include and then dial extensions based on that but it is not happy
21:35.51iCEBrkrmxmasster: http://pastebin.ca
21:35.54*** join/#asterisk marc324 (n=marc3234@206-248-157-247.dsl.teksavvy.com)
21:36.58mxmassterhttp://pastebin.ca/27641
21:37.08mxmassteri am missing something simple with this i am sure
21:38.15jarrodis cripi on?
21:39.21iCEBrkrmxmasster: Where'd you find the syntax for including by date/time?
21:39.29mxmasstervoip-info
21:39.31iCEBrkrmxmasster: I don't see it in the wiki
21:39.38mxmassterhold on
21:39.40mxmassteri will give link
21:40.04iCEBrkrPersonally, I'd do a GotoIfTime() and set up the different 'includes' in Contexts
21:40.04mxmassterhttp://www.voip-info.org/wiki/index.php?page=Asterisk%20tips%20openhours
21:40.29mxmassteriCEBrkr: the time works fine
21:40.36brookshireprank calls #1
21:40.38mxmassterthe problem is that it won't dial my internal extensions
21:40.39brookshire:(
21:41.04iCEBrkrmxmasster: adding those includes breaks your internal dialing?
21:41.20mxmassterno, when the user presses 1 for support
21:41.31mxmassterit works correctly to the date/time include
21:41.43*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:41.43mxmassterbut the first command has the dial sip/3874&...
21:41.49mxmassterit cannot connect to the SIP extensions
21:41.57mxmassterso it fails and starts calling the external numbers
21:42.11mxmassterbut those extensions work properly off of the main context
21:42.21jarrodhow can i collect a key sequence in the middle of a call in order to have asterisk start recording a conversation?
21:43.04*** part/#asterisk oej (n=Olle@apollo.webway.se)
21:43.45pooh_jarrod: you can not
21:44.47iCEBrkrjarrod: supposedly you can use *1 with the 'w' command in Dial()
21:45.26mxmassteriCEBrkr: what do you think?
21:46.05iCEBrkrmxmasster: I'm really not sure, the logic is a bit confusing to me and I haven't tinkered with that stuff at all.
21:46.17iCEBrkrSo unless I actually try it. I Dunno
21:46.21docelmoyay!
21:46.27mxmassteriCEBrkr: you said using gotoiftime
21:46.32mxmasstercan you show me an example
21:46.33iCEBrkrdocelmo: it's not 5 yet, you can't be YAYing
21:46.54iCEBrkrmxmasster: GotoIfTime(<times>|<weekdays>|<mdays>|<months>?[context|]extension|]pri)
21:47.05docelmoI am leaving now.. I can yay all I want.
21:47.10iCEBrkrYea, me too
21:47.15iCEBrkrI just printed my timesheet.
21:47.18iCEBrkrBEEER O'CLOCK!
21:47.31iCEBrkr`(_8_(|)===> MmmmmmmmmmmBeeeeeeeeeeerrrrrrr!
21:48.47docelmoeww..
21:48.54docelmook on that note.. I am going home.. CYA!
21:50.15delmariCEBrkr, ok i got sidetracked... so i need to look at incoming and outgoing contexts for the Tt thing... ok...
21:51.25brettnemhmmm.. nice homer thre
21:51.51delmariCEBrkr, i do have separate contexts for as much as possible.. ie incomingfxo outgoingfxo blah blah
21:52.51*** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net)
21:53.07mxmassterhow do i echo from te dialplan to the console
21:53.36Corydon-wmxmasster: show application verbose
21:55.45mxmassterinstead of having gotoiftime switch to a different context, is it possible to have it change to a different priority?
21:58.31falzyea, drop the context
22:00.32ChujiIs there any reason that my Sip conversations would get jittery when monitor is on, but fine when it's off?
22:00.39ChujiSip to PSTN
22:00.51ChujiCPU load is low
22:02.26kuku5how do i  do a blind transfer
22:03.20DaminAnyone used AudioCodes Gateways w/ Asterisk?
22:04.06Inv_arpChuji: prob a cheap power supply
22:04.42ChujiInv_arp : Power Supply?
22:04.45ChujiHuh?
22:05.18Inv_arpChuji: are you saying your pc monitor?
22:05.54ChujiInv_arp : No app_monitor
22:06.01Inv_arplol
22:06.03Chujiie when I'm monitoring calls
22:06.34Chujitechnically, res_monitor
22:07.10justinulol, wtf?
22:07.10*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk)
22:07.14ChujiMaybe I should try Muxmon
22:11.44kuku5how do i  do a blind transfer
22:11.48kuku5ona  gxp 2000
22:12.03sahafeezqueston about dial plans
22:12.40sahafeezif i have exten => 2004,1,Backgound(closed) ; play msg saiding enter extension
22:13.14sahafeezwhats the next line. all the examples have the next stuff with a list of exten => dial()
22:13.30lesouvagedocelmo: It's not working yet. Will you please take a look at http://pastebin.com/417591
22:13.37sahafeezi do not want a list, i just someone to beable to type the exten that they know and go there
22:15.07sahafeezwhat am i missing?
22:15.50justinunext line would be exten => <digitsentered>,1,Dial(SIP/whoever)
22:16.11Augheyhe wants  exten => 2004,2,WaitExten
22:16.24sahafeezah, thats it. and then?
22:16.33sahafeezwill it transfer?
22:16.47Augheyif you dial an extension, it will go to whatever extension you had dialed
22:16.59sahafeezok. can I add a time out?
22:17.18Augheyyes
22:17.24Augheyshow application WaitExten or read the docs
22:17.26justinuah
22:17.45sahafeezAughey: thanks that is what i was looking for. i will go and read now..
22:18.08Augheyits in the sample extensions.conf too
22:18.10sahafeezAughey: all the examples so what justinu said so...
22:18.39sahafeezAughey: I should look at that. i started with blank files for each *.conf and went from the book, wiki.
22:20.12*** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
22:20.27Augheyyou'd usually do a WaitExten(15) and then Goto(whatever step plays the extern-ext-of-person line) to have it loop the prompt
22:20.52sahafeezAughey: i just want it to dail the exten. thats it
22:21.33Augheyright, but if they don't dial within 15 seconds, you might want to play the prompt again
22:22.03sahafeezok. one sec. let me work out the seq..
22:23.27marc324can someone help with SIPP?
22:24.15sahafeezexten => 2004,1,Background(afterhoursmsg)   ;  1 for dir by 1st name, 2 for last
22:24.18sahafeezexten => s,n,WaitExten(15)
22:24.21sahafeezexten => 1,1,Goto(2005,1)
22:24.21AugheySurvey of Income and Program Participation?
22:24.23sahafeezexten => 2,1,Goto(2006,2)
22:25.27Augheydoesn't work, does it?
22:25.40sahafeezwell besided from the typos
22:26.21sahafeezno
22:27.12justinu~sipp
22:27.14jbotSingle In-Line Pin Package: The last "standard" PC RAM configuration before they started making SIMMsA lot like SIMMs, but they have little pins instead of contacts. SIPPs are to VLB what SIMMs are to PCI..  A suicide tool for geeks
22:27.32justinulol
22:27.42sahafeezAughey: got that out of the examples
22:27.47*** join/#asterisk mxmasster (n=mxmasste@ppp-71-138-119-74.dsl.irvnca.pacbell.net)
22:28.15marc324why is ./sipp -ap returns "invalid argument -ap"
22:28.36kuku5how do i pickup a parked call?'
22:28.45Augheydial the extension
22:29.12sahafeezkuku5: for the parking lot you setup
22:29.23sahafeezwhen you part a call if should tell you
22:29.30sahafeezhttp://www.voip-info.org/wiki-Asterisk+call+parking
22:30.57Augheysahafeez: if you need more help, pastebin your extensions.conf
22:32.03kuku5it doesnt
22:32.09sahafeezok. i am going to try to see if i can figure it out.
22:32.12kuku5i do a blind transfer to extensiton 7000
22:32.13kuku5700
22:32.20jontowdon't do a blind transfer.
22:32.26kuku5then what should i do
22:32.32jontowa 'warm transfer'
22:32.35kuku5how do i do that
22:32.38kuku5on a gxp2000
22:33.02*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)
22:34.02kuku5...
22:35.32*** part/#asterisk cypromis (n=michael@83.149.70.59)
22:35.47delmarkuku5, i have just been setting up blind/attended/parked calling etc... whats wrong?
22:36.37Augheys,1,Background(file)
22:36.43Augheys,1,WaitExten
22:36.51Augheyer, s,2,WaitExten
22:36.57Aughey1,1,Goto(whatever)
22:37.02Aughey2,1,Goto(whatever)
22:42.39*** join/#asterisk riksta (n=rick@62.6.163.90)
22:48.41Augheywhat context does it start in?
22:48.45Augheyincoming I assume?
22:49.12sahafeezyes
22:49.39Augheyand I assume you're dialing 3600 from some sip phone to enter into this?
22:49.42sahafeezno
22:49.53sahafeezdailing from my celling inbound on my pri
22:50.50Augheyand what do you dial after it plays closed-msg?
22:50.56sahafeezthe DNIS is the _3600. i have the routing in place for during hours, etc..
22:51.07*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
22:51.12bweschkedamin: contact netxusa for the tdm2400 cards
22:51.24bweschkethey're a distributor. they're supposed to ship the 3rd week in november
22:51.35sahafeezwell nothing. the closed msg say, hit one 1 for dir, 2 for dir last name or enter and extension. if they screw up i want pbx-invalid and place closed-msg again
22:53.27sahafeezmake ssense?
22:53.58Augheyso it does play the message right?
22:54.28sahafeezyes. and transfers to the 4xxx right. the dir trasfer does not work.
22:54.39sahafeezalso the jump back for invalid does not work right
22:54.45sahafeezexten => _3600,1,Answer
22:54.45sahafeezexten => _3600,2,Background(closed-msg)
22:54.45sahafeezexten => _3600,3,WaitExten(15)
22:54.47sahafeezexten => i,1,Playback(pbx-invalid)
22:54.47sahafeezexten => i,2,Goto(incoming,3600,1)
22:54.49Augheyit plays the message and then goes to 4xxx?
22:54.59Augheyyou're not making sense
22:55.07sahafeezno, plays and if you enter 4201 or such at anytime it works.
22:55.37AugheyI don't think it should do anything else because internal isn't included in the incoming extension
22:56.04sahafeeztrue, but the 4xxx transfer works and it is in interal
22:56.54Augheyif that's truely your full extensions.conf file, it shouldn't do that.  There should be no other valid extensions in the incoming context
22:57.52sahafeezah, i do have that in my inbound
22:58.10Augheyand take off the _ on the extensions that don't actually use patterns
22:59.16sahafeezon all?
22:59.43Augheyplease post your actual extensions.conf file, otherwise we're just going to go round in circles
23:00.10sahafeezok. but it is not quite complete as i am trying to test
23:00.12Augheyyou don't need it.  Shouldn't REALLY matter, but it implies there some sort of pattern for that extension
23:00.12sahafeezwill do it now
23:01.16sahafeezhttp://pastebin.com/417663
23:01.17Augheybut you have other things in the incoming context that you didn't show.  So it's behaving in ways that it shouldn't if you actually used what you posted, but you didn't, so me telling you what it should do doesn't mean anything if I'm not actually looking at your actual configuration
23:02.22*** join/#asterisk alvariux (n=unky@201.138.135.221)
23:02.24alvariuxhi
23:02.40alvariuxdoes somebody has build iaxclient on mac?
23:03.00alvariuxim getting this error ld: Undefined symbols:
23:03.00alvariux_fprintf$LDBLStub
23:03.01alvariux_vsnprintf$LDBLStub
23:03.01alvariux_printf$LDBLStub
23:03.01alvariux_snprintf$LDBLStub
23:03.02alvariux_sprintf$LDBLStub
23:03.04alvariuxlibtool: internal link edit command failed
23:03.06alvariuxmake: *** [shared] Error 1
23:04.08Augheysahafeez: well your [internal] extensions aren't part of your [inbound] context.  Move [internal] up above [inbound] and then include => internal in your inbound context
23:04.51sahafeezAughey: i thought that was bad.
23:05.24Augheywell, the way it is now, anyone could dial 5616 and get your _5615 extension
23:05.30mxmassteris there a way to make the dialplan timezone aware - i.e. the server is in utc, but i want to make gotoiftime decisions based on the local timezone
23:06.14Augheyyou need to partition your dialplan into more contexts so that you get what you want
23:06.14jontowmxmasster; any reason why you can't just set the server to the local timezone?
23:06.38jontowits a quick fix.. rm /etc/localtime ; ln -s /usr/share/zoneinfo/.../... /etc/localtime
23:06.41jontow?
23:06.50mxmassterexcept i have multiple locations in different time zones
23:07.14Augheyso put the extensions tha tyou want available in their own context, and include it in the incoming context.  Actually, you probably want an incoming_night context to only have the extensions you want there and have it go into that context
23:07.46jontowok, so there is no 'local timezone' anyway.. its on a per-location basis
23:07.55jontowjust translate all your times before you put 'em in the dialplan
23:07.59sahafeezAughey: ok. that seems to make sense
23:08.55*** join/#asterisk msw (n=msw@rdu-nat.rpath.com)
23:10.18mxmassterjontow: that's what i need to know, how do i translate the time?
23:10.24mxmassteroh you mean by hand
23:10.30mxmassterany automated way?
23:11.12jontowwriting scripts?
23:11.16jontowperl, C, shell?
23:11.22jontow"automated" is relative :)
23:11.49mxmassterautomated as in internal to asterisk
23:11.54jontowno
23:11.56*** join/#asterisk Flauto (n=zhao@71.194.39.175)
23:12.09jontowthere isn't anything like that to my knowledge, currently
23:12.13jontowit will involve code-writing
23:12.23jontowi haven't seen everything thats out there, but.. to my knowledge.
23:12.43*** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net)
23:13.05*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166116148.nb.aliant.net)
23:15.34DrukenHME~seen sivana
23:15.38jbotsivana is currently on #asterisk (9h 44m 45s).  Has said a total of 1 messages.  Is idling for 6h 31m 34s
23:17.38sahafeezdoes the exen => i,1 get jumped to in each contect that it is defined if something does not work right
23:20.25justinu~seen blitz[laptop]
23:20.30jbotblitz[laptop] is currently on #asterisk-doc (2d 5h 25m 49s) #asterisk (2d 5h 25m 49s).  Has said a total of 357 messages.  Is idling for 1h 47m
23:21.30Augheyyes
23:23.01*** part/#asterisk alvariux (n=unky@201.138.135.221)
23:23.47sahafeezAughey: ok i have it working
23:24.07sahafeezAughey: one thing left. i have the directories at ext2004/2005
23:24.42sahafeezbut in my msg i want to have them hit the exten, or 1 for dir1 or 2 for dir2 and have it jump to 2004/2005. can i do that
23:25.32Augheysure, just create a 1 extension and Goto(2004)
23:25.49Aughey1,1,Goto(2004)
23:26.02sahafeezok. makes senses. thanks. each time i get something else working the whole picture makes more sense
23:26.59sahafeezOK, I  Know what the exten =. i,1 is. but what is it when it is exten=> t,1 is t timeout?
23:27.39*** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com)
23:29.14*** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net)
23:29.16cripitohi
23:29.34cripitoi ws in the wrong one :D
23:30.32justinusahafeez: t is when they don't enter enough digits, I think
23:31.51Augheyhttp://www.voip-info.org/wiki/view/Asterisk+standard+extensions
23:32.16sahafeezthanks all. ready for my cut over.
23:32.20*** join/#asterisk BigJoe1 (n=BigJoe1@dsl-20-177.cofs.net)
23:32.23sahafeezheres hoping xo gets it right
23:32.27BigJoe1hello everyone
23:33.00syleJOE!
23:34.13syledon't know who you are but sounded good , like norm from "cheers"
23:34.46Flautois there a way i can setup the server to give the internet trafic to voice calls first and then others?
23:36.00Augheyiptables can probably help.  It won't be necessarly easy
23:36.43*** join/#asterisk Starmaker (n=magnus@85.8.2.169)
23:37.01Flautoaughey, it is not easy for setting up trafic?
23:37.29cripitoflauto in the wiki u can find a few things about qos
23:37.37cripitosome easy to do .. somes not so easy
23:39.00*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
23:39.01StarmakerI'm using alaw/ulaw right now for audio coding, but I'm thinking about trying G.729 instead, are there ANY advantages with it except it using less bandwidth?
23:39.32Flautocripito, what topic it unders?
23:39.35Augheynot compaired to u/alaw
23:39.39ooglei wrote a hack to channel.c that manipulates voice frame data and spent an hour trying to figure out why it wasn't working.  i loaded up gdb, froze the process and was like, huh?  how come the phones are still bridged?? DUH they're sip phones!
23:40.35Starmakerah, guess i'll stick to alaw/ulaw then, bandwidth isn't really a a problem
23:42.07cripitoflauto: http://www.voip-info.org/wiki/view/VOIP+Routers
23:42.26cripitostarmaker: me too
23:42.58cripitospecially if i can make tricks with the invite etc etc for save bandwith too..
23:43.32Starmakernow i just need to set up CDR with mysql :)
23:44.55cripitopiece of cake ;)
23:45.00*** join/#asterisk grimse (n=grimse@p5481DDC5.dip.t-dialin.net)
23:45.01Starmakerheh
23:45.03ooglei recommend using ODBC
23:45.06ooglemakes life easier
23:45.09Starmakeri'm trying :)
23:45.21cripitocd_addon_mysql is not soo dif to install
23:45.22Starmakerit still keeps logging to csv files :)
23:45.59Starmakercripito, well, I havn't compiled asterisk from source, i used debians asterisk-package
23:46.22cripitommm...
23:46.29oogleheresy!
23:46.33justinulol
23:46.34Starmakerheh
23:46.36cripitoi prefer the source code.. ejem...
23:47.04Starmakeri like apt-get's dependency solving :)
23:47.29cripitowell must me asterisk-addons-pkg or similar there
23:47.41Starmakercripito, no, actually there isn't
23:48.01Starmakerbut I should be able to set it up with odbc
23:49.41ooglei'm writing a voice changer, can anyone give me tips on ways i can change ast_frame data to accomplish this?
23:50.05justinuoogle, you might want to ask on #asterisk-dev
23:50.14ooglejustinu: right-o
23:50.16Starmakeri mean, it's really strange, if i do odbc show, it shows that it is connected to one of my dsns
23:50.46*** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il)
23:51.51voipjoyanybody can advice why i can't compile zaptel driver http://pastebin.ca/27578 ?
23:53.12Starmakeryes! i got it working :)
23:55.36Starmakerjust found a really strange thing
23:57.06Starmakerskype -> sip-calls are actually voip all the way
23:57.26Starmakerbut skype doesn't allow it's users to use SIP
23:57.42Starmakerthat's just plain evil :)
23:58.04ManxPowervoipjoy, You either have a bad CVS checkout or beta version, or you don't have the linux kernel source and headers installed.

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