irclog2html for #asterisk on 20050705

00:00.15SpaceBassnot i
00:02.00Hmmhesayslooks alright
00:02.05Hmmhesaysand AMP'ish
00:02.06lowridesespacebass: do you have that screen name to send?
00:02.22lowrideseno one is answering there with them right now
00:03.57*** join/#asterisk file[laptop] (~file[lapt@mctn1-3634.nb.aliant.net)
00:04.26SpaceBassback
00:04.39SpaceBasslowridese:  no I sure don't
00:05.04lowridesebut they had someone that helped you in the end?
00:05.36SpaceBassyeah, he didnt work for BV, but he helped
00:05.54SpaceBassand then I scrapped *  when I needed help again Ariel_ was a big help
00:07.36SpaceBassok, evoloution on os x... slow
00:11.38*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
00:11.45infinity1can someone spend a few cycles with me on this CHANUNAVAIL problem i'm having with my dialplan?
00:11.47shido6what are u runnin os x on?
00:12.07SpaceBasspowerbook 12:
00:12.09SpaceBass12"
00:12.13SpaceBass1 gb ram
00:13.25SpaceBassnot that impressed with evolution... uses OWA to talk to exchange
00:13.50Hmmhesaysinfinity1 might help if you post some cli output on pastebin
00:14.55infinity1Hmmhesays: k. i'll do that.
00:15.24Hmmhesaysthen pat your head, rub your tummy and whistle
00:15.47tzafrirdoes: asterisk -rx 'restart now' work (restarts, doesn't crash) for anybody here that runs asterisk as non-root?
00:15.50*** join/#asterisk yoink (~yoink@MTL-HSE-ppp165512.qc.sympatico.ca)
00:16.13tzafrirSeems broken in 1.0 . I wonder if it got fixed in HEAD
00:17.20jmannQuestion for all-- is there a way that I can have a dns name in the externip= option in the sip.conf file
00:17.40jmannthis is due to having DHCP from my provider
00:17.41SpaceBassjmann: sure
00:18.12jmannwhat is the syntax
00:18.34HmmhesaysI would try externip=www.myrclean.com
00:18.46Hmmhesaysheh, once again a good joke foiled by bad typing
00:18.58infinity1Hmmhesays: http://pastebin.ca/16893
00:18.59jmannI am getting an Invalid address to externip
00:19.34Hmmhesaysok infinity set verbose to something high make a call and paste the cli output in there
00:20.48infinity1Hmmhesays: http://pastebin.ca/16895
00:21.23infinity1i'm trying to get it to dialout on nufone if the zap interface is busy.
00:23.16Hmmhesays${DIALSTATUS} = CHANUNAVAIL looks like that is returning false
00:23.19infinity1Hmmhesays: it seems to me that dialstatus ins't being set to unavail.
00:24.02infinity1Hmmhesays: any idea what my problem is? i think this is suppose to work
00:28.42Hmmhesayswell as i'm sure you noticed the call is not making it to s,22
00:29.35infinity1yea. its jumping to 122
00:30.11infinity1hm. on second though. that should be a 123 i think.
00:30.31Hmmhesaysdoesn't dial return n+101 on failure?
00:30.38infinity1yea
00:30.55infinity1oh. wait.
00:30.59infinity1122 is right :)
00:31.20HmmhesaysIf all the called channels are busy, Dial will exit with a return code of 0 and will continue in the current context at priority n+101, if it exists, where n is the priority of the Dial command.
00:31.46infinity1http://lists.digium.com/pipermail/asterisk-users/2005-February/089017.html
00:31.48infinity1same problem
00:32.33Hmmhesayscomment out 122 and reload and try your call
00:32.44infinity1Hmmhesays: i tried that. it works.
00:32.53Hmmhesaysdid you see what I said above?
00:33.31infinity1so that way it is working is correct?
00:33.33SarahEmmhihi
00:33.35SarahEmmbackies :o)
00:33.38Hmmhesaysinfinity1: yes
00:33.42infinity1it retruns busy instead of CHANUNAVAIL
00:33.50infinity1argh.
00:33.59infinity1so how do i use CHANUNAVAIL? :)
00:34.05Hmmhesaysleave that busy commented out
00:34.12Hmmhesaysor...
00:34.33Hmmhesaysset s,122  to goto s,22
00:35.14*** join/#asterisk iq (~IQ@70-59-162-85.omah.qwest.net)
00:37.06SarahEmmshido6: should i resend my request to the main email address, or is it okay as-is?
00:37.10infinity1so whoever i stole the code from is an idiot ...and i'm prolly a bigger one . hah
00:37.49infinity1Hmmhesays: thanks for the help. i'll play with it some more.
00:38.41file[laptop]stealing?!?
00:39.04Hmmhesaysheh, stole? that's an amp generated config i'm sure
00:39.04infinity1file[laptop]: borrowed :)
00:39.16infinity1Hmmhesays: whats amp?
00:39.28Hmmhesayswhere you borrowed that dialplan from
00:39.35Hmmhesays~amp
00:39.35jbotit has been said that amp is an Audio MPEG Player.  [non-free], or http://amp.coalescentsystems.ca/
00:39.43litagehow exactly does asterisk work with e164.org?
00:39.48Hmmhesaysheh, that's not quite right
00:40.19infinity1do most people use something like amp to create dialplans? or do they do it by hand?
00:40.27Hmmhesaysamp is a gui for asterisk
00:41.05file[laptop]by hand.
00:41.24Hmmhesaysyeah, i'd agree with that
00:41.51infinity1aite. i guess i'll keep banging my head learning this
00:41.52infinity1heh
00:42.22Hmmhesayshelps if you start off with your own dialplan... you'll learn more banging your head that way
00:42.53infinity1so it works like you said.
00:42.54infinity1<PROTECTED>
00:43.13Hmmhesaysi suppose it does
00:45.30Hmmhesaysnow if your iax connection returns busy it should still play the busy prompt
00:47.16SarahEmmhihi litage
00:48.28*** join/#asterisk NetSkier (~ns@ca-redbch-cuda1-c9b-a-152.stmnca.adelphia.net)
00:49.53litagehowdy there SarahEmm
00:49.53litagewhat's cookin
00:50.15*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
01:01.28SarahEmmnot too much
01:01.28SarahEmmbrain is full
01:01.28SarahEmmrelaxing
01:01.28sylelooking for area code 204 DID's
01:01.28SarahEmmsyle: have you talked to blitzrage?
01:01.28syleyeah he told me about a site, i emailed them and never got a response back
01:01.28syleso still looking
01:01.28SarahEmmwhich site?
01:01.28sylehttp://www.mixnetworks.com/
01:01.28SarahEmmahh
01:01.28SarahEmmerr
01:01.28SarahEmmblitzrage works with them
01:01.28SarahEmmhe can set you up himself afaik
01:01.28Hmmhesayshow many incoming simultaneous calls are you looking for to one did?
01:01.28SarahEmmHmmhesays: me? one
01:01.28SarahEmmoh, duh
01:01.28SarahEmmnevermind, not me :P
01:01.28syleright now just 1 hmmhesays, but if i ever get a company contract more
01:01.28SarahEmmthanks Hmm-home
01:01.28SarahEmmHmmhesays
01:01.28Hmmhesaysvonage has a business plan you can use with asterisk
01:01.29Hmmhesaysomg I have a clone
01:01.29wunderkindamn why doesnt anything ever work right for me ;/
01:01.29Hmmhesaysvatsamatta
01:03.09wunderkini have ztdummy installed, when i try to make a conference, it says invalid conference number
01:03.22SarahEmmthat's not a ztdummy issue anyway...
01:03.24Hmmhesayslsmod
01:03.54wunderkinyeah it shows
01:04.11HmmhesaysSarahEmm rmmod ztdummy and see what your asterisk says
01:04.33SarahEmmHmmhesays: *blinks* you're serious? 'no timing source' shows 'invalid conference number'???
01:04.48Hmmhesays*that is not a valid conference number please try again*
01:04.58SarahEmm*blinks*
01:04.58SarahEmmgah!
01:05.01SarahEmmthat's intuitive :P
01:05.12SarahEmmdo you at least get something on the console?
01:05.32Hmmhesaysgood question, I don't remember
01:05.48PatrickDKsounds like meetme.conf isn't setup right
01:06.34infinity1i don't suppose there is a way to check if a zap interface has dialtone before using it
01:06.42wunderkinyou cant do.. like meetme(1|d) and it will create #1?
01:07.03SarahEmminfinity1: well, x100p's throw a RED alarm if there isn't a line hooked up.. does that help you?
01:07.14PatrickDKhmm, dunno, I haven't done dynamic conferences
01:07.18PatrickDKnever had a reson to
01:07.40*** join/#asterisk PBXtech (~nik@001-740-536.area1.spcsdns.net)
01:07.44infinity1SarahEmm: whats the command to see this alarm?
01:07.50wunderkinwell i dont really want to, but just trying to figure out a workaround for another problem
01:08.27PBXtechwhy am i getting permission errors on call files when moved to the outbound dir? directory and file are chmod777 and user is asterisk..
01:08.45litagehow exactly does asterisk work with e164.org? can you tell your viop device to dial john@smith.com and asterisk will query e164.org for the corresponding voip/phone #?
01:08.51*** join/#asterisk redG ([U2FsdGVkX@67.51.185.15)
01:08.51redG<PROTECTED>
01:09.09SarahEmminfinity1: zttool will show it. asterisk will throw a msg on the console when it goes into RED. i'm not sure how to get it from code
01:09.22PatrickDKlitage, no, you query the phonenumber at e164, and it tells you if there is a match
01:09.44PatrickDKread up on e164
01:09.57litagePatrickDK: i have been, but they're a bit fuzzy on details
01:10.08PatrickDKfuzzy?
01:10.14litageunspecific
01:10.19PatrickDKit very simple, does phone number xxx-xxx-xxxx exist
01:10.23infinity1SarahEmm: hm., i was trying to see if there is a way to not use a zap interface if the phone line is broke for dialing out. guess it requires some serious scripting
01:10.28PatrickDKyes it does, it's at sip/user@ip
01:10.49*** join/#asterisk rnovotny22 (~rnovonty2@c-66-41-170-125.hsd1.mn.comcast.net)
01:10.55litagePatrickDK: yes but e164.org says that you can call someone via their email address
01:10.57PatrickDKinfinity, I never found a way for asterisk to do that, my cisco box's do though
01:11.10PatrickDKlitage, that must be a custom extention
01:11.14SarahEmminfinity1: well, you should be able to
01:11.24SarahEmminfinity1: you'd need to detect if there's an alarm on the channel before dialing out
01:11.40SarahEmminfinity1: i'm going to be Super Busy until saturday, i'll try to investigate then and if there's no way to do it now, make one
01:11.45SarahEmmthere prolly is tho
01:12.32litagePatrickDK: so the user has to query e164.org manually, find a match, and put that phone # into their voip device?
01:13.01PatrickDKna, normally the voip device does that
01:13.14syleanyone tried the iaxy's?
01:13.25infinity1i see. zttool changes the active flag.
01:13.32DarthCluelitage: or you write some code to have * do it via res_js or agi or something of that nature.
01:13.34infinity1too bad that info isn't availabe as a dialplan command
01:13.46litagethanks DarthClue, i'll look into res_js and agi
01:13.54PatrickDKdarthclue, asterisk already does it
01:14.01SarahEmminfinity1: it's not in any variable?
01:14.59DarthCluePatrickDK: using what?
01:15.37PatrickDKe164
01:15.37infinity1SarahEmm: i'm still looking . i haven't found one
01:15.37PatrickDKenum.conf
01:15.37SarahEmminfinity1: 'kay. i'll look into it on saturday and look at fixing it if i don't find a way to do it.
01:15.37SarahEmminfinity1: what id you try dialing out on an alarmed channel? it doesn't just jump to priority+100 or somesuch?
01:15.37DarthCluelitage: there ya go.
01:15.44sylei got a question: when i am dialing out my analog phone provider line on my fxo port , *67 in asterisk doesn;t execute *67 on my phone providers line, how can i fix this?
01:16.05PatrickDKsyle, make asterisk support *67
01:16.09PatrickDKthat is alot of fun :)
01:16.13infinity1SarahEmm: well, the problem is that you cant jump to 101 because it also means that the chanels are ful
01:17.20SarahEmmPatrick^: make it support? huh?
01:17.20infinity1SarahEmm: so if you use zap + iax to dialout, 101 wouldnt' work. according to the docs +101 is for busy/no avail chanels
01:17.20SarahEmmPatrick^: isn't it just normal DTMF?
01:17.20sylewell we are just talking about digit pass-through on analog lines
01:17.20PatrickDKsarahemm, yes, his dialplan doesn't support it though
01:17.20litageDarthClue: what do you mean?
01:17.20SarahEmmPatrick^: oh. well that's a dialplan issue then :)
01:17.20infinity1is zttool only ncurses? its really annoying.
01:17.38sylecan i setup a *67 extension in dialplan to override asterisk's default *67?
01:17.41SarahEmminfinity1: okay, but if the channel is in alarm, it's unavaialble.. do you need to seperate the two?
01:17.41PatrickDKsyle, it's all in extentions.conf
01:17.45DarthCluelitage: take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+E164+Call+Routing and http://www.voip-info.org/tiki-index.php?page=Asterisk+config+enum.conf
01:17.55PatrickDKsyle, asterisk doesn't have a default *67
01:18.34sylesure it does
01:18.35PatrickDKunless your using zaptel fxs ports
01:18.35infinity1SarahEmm: yea. because unavail also means it is in use
01:18.35syleyeah i was using zaptel fxs ports for testing
01:18.35infinity1SarahEmm: wait.
01:18.35PatrickDKah, I only use sip phones
01:18.41infinity1SarahEmm: hmm
01:18.48syleit don;t work on sip?
01:19.25sylehmmm, so you do *67, and in that extension do like setCID caller unknown kind of thing?
01:19.47infinity1SarahEmm: i think it would work if the dial returned control to the dial plan and returned CHANUNAVAIL
01:20.10SarahEmminfinity1: it doesn't? what happens?
01:20.23infinity1SarahEmm: it just sits there
01:20.37sylepatrick what ip phone do you use?
01:20.52SarahEmminfinity1: and you've verified the channel is showing RED right now?
01:21.50infinity1SarahEmm: when i removed the conection to the zap interface, it still tries to use it and doesn't fail fast enough
01:21.50PatrickDKsyle, yuou don't unset cid, unless your using a sip/iax voip provider
01:21.50fileDarthClue: I'm awake, or atleast I think I am!
01:21.50PatrickDKif you using fxo out, you have to dial *67 before your number
01:21.50infinity1SarahEmm: i've never seen this RED thing. i loaded zttool and i only see active chaning
01:21.50infinity1er s/chaning/changing/
01:21.54DarthCluefile: it is taking you way too long to respond, you are absolutely asleep.
01:22.08fileam not
01:22.10fileI'm working
01:22.19infinity1SarahEmm: active changes when i unplug and plug the zap interface in.
01:22.52DarthCluefile: everything you are doing is just a dream, therefore you must be asleep.
01:23.49sylepatrick your just horrid to talk to lol
01:23.49sylebbiab
01:23.49infinity1SarahEmm: i'm using a tdm400p. maybe becuase it only has one interface it doesn't go red?
01:23.49fileDarthClue: wow - your logic is great!
01:23.49infinity1SarahEmm: er i mean because it has multiple interfaces
01:23.49PatrickDKsyle, I will help yuou, I will not do it for you
01:23.49SarahEmminfinity1: ahh, i don't have a tdm400 so i have no idea =-/
01:23.49SarahEmmx100p here
01:24.23infinity1SarahEmm: when you unplug the pstn from the x100p it goes red? mine just changes activity to 0 instead of 1
01:24.23wunderkinvaletparking doesnt work right for me either ;/
01:24.39DarthCluewunderkin : which valet parking are you using?
01:24.59wunderkinDarthClue, umm.. app_valetparking from bkw i think
01:25.09SarahEmminfinity1: mine goes red in zttool
01:25.19infinity1SarahEmm: interesting.
01:25.21DarthCluewunderkin: verify that and then tell us what it is doing.
01:25.23SarahEmmusing CVS HEAD zap stuff
01:25.37infinity1SarahEmm: i'm using 1.0.9 debian unstable
01:25.40SarahEmmahh
01:25.43SarahEmmmaybe it's a HEAD thing
01:25.48wunderkinDarthClue, well.. maybe not.. in the header it says anthm?
01:26.07DarthCluewunderkin: that's the right one.  what is it doing?
01:27.55wunderkinDarthClue, i have exten => 7,1,ValetParkCall(1|mylot|600|1|2|dial)   / exten => 8,1,ValetUnParkCall(1|mylot); i park it ok and it keeps saying the lot number.. when i pick it up it works but keeps saying the lot number..then when i hangup it crashes asterisk.. or i think it did something else weird the time before that
01:28.22DarthCluewunderkin: stable or head?
01:28.28wunderkinhead from 6/30
01:29.50DarthCluewunderkin: i don't think anthm or bkw_ are here at the moment.  they would be the ones to provide the best assistance.  drop them an email or wait till morning (12 hours) and they will be around.
01:30.19filegrab a backtrace and see where it's crashing
01:30.26wunderkinok.. are there other versions i can try?
01:30.27DarthClueunless file wants to volunteer.
01:30.42filewell there's super valet parking too
01:30.45wunderkinyeah
01:31.05DarthCluevalet_parking supercedes super_valet_paerking.
01:31.08filebut uh backtrace!
01:31.12DarthClueer...parking even.
01:32.33wunderkinill try
01:33.03*** join/#asterisk likwid-- (~likwid@nc-65-41-255-12.dyn.sprint-hsd.net)
01:33.11wunderkinthere it went with the weirdness this time.. didnt crash
01:34.31wunderkinok well theres no core so i guess i have to attach to it
01:35.05wunderkinnice gdb isnt installed
01:39.45tzafrirmaybe you have strace?
01:40.02tzafrirSometimes it is good enough
01:40.19wunderkinwell i just installed gdb .. i havent used it that often so.. ill try :D
01:40.31SarahEmmstrace rocks
01:41.41wunderkinok i don tknow what im doing
01:43.12tzafrirwunderkin, do you have a core dump?
01:43.20wunderkinnot for this issue
01:43.29tzafrir(did you run asterisk with -g to get one?)
01:43.42wunderkinit looks like asterisk gets killed
01:43.45wunderkinyes
01:44.00tzafriryou can attach gdb/strace to a runnig asterisk process
01:44.31wunderkinwell, its attached to something but i dont know where to go from there
01:44.52wunderkini did gdb program then the pid of asterisk.. at least one of them
01:44.53tzafrirwunderkin, what is the issue you try to isolate?
01:45.19wunderkina problem with valetparking.. it is doing weird things and asterisk ends up getting killed
01:45.33SarahEmmwhat makes it 'valet' call parking?
01:45.49SarahEmmdoes a little asterisk process come and move your call to the parking lot instead of manually having to drive it there yourself?
01:46.18wunderkinfile wanted me to backtrace it
01:48.03wunderkinok, well i did an strace on it
01:48.11wunderkinfile, is that good enough?
01:48.48DarthClueSarahEmm: yeah, something like that.
01:52.14wunderkini dont know how to use gdb on a running program
01:52.50wunderkinstrace says its getting a sigkill
01:56.13*** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net)
01:56.13milkyflavahello
01:56.13milkyflavaI have finally got my TDM400P and I am going through the setup at asteriskdocs.org
01:56.13milkyflavaDo I need to use the IAX or SIP config files or just the zaptel.conf file? Do I setup the zaptel.conf then use either IAX or SIP?
01:56.17milkyflavaIs zaptel for setting up the card then the IAX or SIP is for making calls?
01:56.40milkyflavaAm I even making sense?
01:57.28wunderkinwell.. you use zap to access the card... if you have other phones off of asterisk that arent connected to your card, then you would use sip or iax
01:58.12milkyflavawould thise be like soft phones and IP phones?
01:58.18wunderkinyes
01:58.18milkyflavathise=those
01:59.50milkyflavaThanks, this stuff is confusing!
01:59.50wunderkinif you have a phone on a fxs port, it would use zap
01:59.50milkyflavaBut I can use them together also, correct?
01:59.50infinity1milkyflava: hah. you haven't even touched the surface.
01:59.50milkyflavalol, I know
01:59.50wunderkinto dial out the fxo port it would use zap too
01:59.50wunderkinwell you could have a sip phone dial out on your card
01:59.50milkyflavaI have a dev card with 1 fxo and 1 fxs port on it the 11B I think it was called
02:01.54*** join/#asterisk SarahEmm (~sarahemm_@MTL-HSE-ppp168389.qc.sympatico.ca)
02:01.54SarahEmmhihi
02:01.54SarahEmmagain
02:01.54SarahEmmWAP crashed :P
02:01.54milkyflavawunderkin: Thank you.
02:01.54DarthCluewunderkin: are you using super valet parking or valet parking?
02:01.54wunderkinDarthClue, just regular right now..
02:02.08wunderkini was just trying to figure out what the difference was with super
02:02.12DarthClueok, super is old so i just wanted to make sure.
02:02.18wunderkinoh..
02:02.28DarthCluesuper was an alpha that anthm did as a whim.  valet is what has been officially released.
02:02.36wunderkinok
02:03.12*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
02:04.15wunderkinthis is really just trying to hack around attended transfer.. which doesnt work right the way im doing things
02:04.16wunderkinunless it just doesnt work the way im wanting it to
02:06.01*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
02:06.01bjohnsontotally unrelated to asterisk but I can't seem to bring up a web site.  does somebody know how to use a public http proxy?
02:07.42*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
02:09.02*** join/#asterisk chonlada (chonlada@202.29.6.19)
02:09.39chonladaHi all, can i find NEC Neax 7400 ICS manual ?
02:10.35Pete_Largotry google?
02:10.54chonladayes, i am try. but cannot found.
02:11.18Pete_Largohttp://www.sundance-communications.com/cgi-bin/forumdisplay.cgi?action=topics&forum=NEC&number=14&DaysPrune=1000&LastLogin=
02:14.35Pete_Largosomeone there may be able to help
02:14.36SarahEmmnini all
02:19.24Pete_Largonight Sarah
02:19.24wunderkinWOOOOOHOOOO
02:19.24wunderkin(sorta)
02:19.24Pete_Largogood news wunderkin?
02:19.24wunderkinyeah i finally figured out why atxfer wasnt working right for me :D
02:19.24wunderkinbut now i gotta figure out a caller id problem.. i think that one is a source problem
02:19.24Pete_Largogood for you :)
02:19.24chonladaPete_Largo thank you.
02:19.24*** join/#asterisk CpuID (~nathan@dsl-202-173-176-82.qld.westnet.com.au)
02:19.24wunderkini was using # for atxfer which i guess # is reserved for the regular transfer.. i was wondering why it was doing blind :)
02:19.24CpuIDok ppls, question, is asterisk HEAD these days kinda like a 1.1 branch or something?
02:19.24JunK-Y1.2
02:19.25DarthClue1.2
02:19.37CpuIDah k
02:19.59CpuIDso will 1.0.x lead up to 1.1, and then HEAD as you said will stay 1.2 right?
02:24.46CpuIDjust confirming, so i know whats what when reading through src :)
02:24.46CpuIDor will 1.1 get skipped altogether?
02:24.46rabelaisdo I need "insecure = very" to get incoming calls from fwd?
02:24.46Pete_Largofwd has a great setup guide on their web site
02:24.46loudthat parameter should be changed or something, people think that they can get owned with that.
02:24.46rabelaisI can't get calls otherwise, so is it not a security hole to allow that?
02:24.48loudnope
02:25.31loudits to allow reged hosts to call without re doing auth.
02:26.39rabelaisok, I read that on the wiki, I was a little confused, thanks for clarifying it     :)
02:36.46CpuIDi think its good having insecure=very as is :)
02:37.05CpuIDgives hosts a good reason to change their shit, so people wont think their gonna get owned all the time lol :)
02:37.14CpuIDthe name suits nicely
02:39.59loudme too, unless it brings a security flaw, i dont care, but people think its like a doyouwantogethacked = yes
02:39.59CpuIDlol
02:39.59*** join/#asterisk darylp (~daryl@63-208-162-60.digitalrealm.net)
02:43.35Ariel_Hello everyone
02:48.18litageif my voip provider only uses H323 and my network uses NAT, how can i setup asterisk to allow multiple simulataneous incoming/outgoing calls?
02:51.37*** join/#asterisk spoot_nick (~julio@CPE-147-10-171-214.nsw.bigpond.net.au)
02:51.37Ariel_h323 is not very friendly to nat.
02:51.37spoot_nickdoes anybody know why asterisk (using the samples conf files) would get calls, but not output any sound at all?
02:51.37spoot_nickerror msg: WARNING[15566]: file.c:550 ast_readaudio_callback: Failed to write frame
02:51.37litageAriel_: yeah that's what i've read, so i'm curious for any solutions that might exist
02:51.59Ariel_litage, everyone I know that uses h323 is on an static IP address.
02:52.21Ariel_spoot_nick, is this sip your not getting sound on?
02:52.27spoot_nickyes
02:52.48Ariel_spoot_nick, is the server on the same network as the sip device?
02:52.58spoot_nickAriel_: yep
02:53.06spoot_nickno NAT between them, local area network
02:53.11spoot_nickfresh install according to the "10 minute guide to asterisk", http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart
02:53.18spoot_nickslackware 10.1
02:53.55spoot_nicki'm using a sjphone to test it. i dialed "1000" for the congrats msg, and no sound gets out of it. same with the echo service
02:54.24litageAriel_: can those people you know using h323 do multiple, simultaneous in/out calls?
02:54.27Ariel_spoot_nick, is the sip phone registered?
02:54.35Ariel_litage, yes
02:54.53spoot_nickAriel_: yes it is. example console output: Registered SIP 'julio' at 192.168.0.102 port 5060 expires 120
02:54.59litageAriel_: how did they accomplish that?
02:55.50Ariel_litage, I have not used h323 in over a year. It's a bear to get configured actually to get it to install on the asterisk box. But it does work once you get pass that.
02:56.13spoot_nickweird thing is i don't get any error msgs. except for that one 'ast_readaudio_callback: Failed to write frame'
02:56.27spoot_nickwhich makes sense, since it's just audio that seems to be failing me
02:56.46Ariel_spoot_nick, I don't know the 10 minute setup let me read what it does. But first I would check your settings as it not configured correctly.
02:56.59litageAriel_: all the info and docs needed to get h323 configured on asterisk is available on voip-info.org?
02:57.26Ariel_spoot_nick, for your sip.conf account do this disallow=all allow=ulaw canreinvite=no
02:57.49Ariel_litage, most of it. yes and there is readme's as well.
02:58.12litageAriel_: as in Asterisk's readmes, right?
02:58.49Ariel_litage, there is an h323 channel drivers that comes with asterisk there is a readme there.
03:02.17litageAriel_: what exactly is a channel?
03:02.17spoot_nickAriel_: my current conf files, http://202.92.94.93/~julio/ast_confs.tar.gz
03:02.17tzangerlitage: a channel is something that audio comes in and out of
03:02.17tzangera resource is something that can manipulate a call  (might provide audio, mangle audio, detect audio, etc.)
03:02.17Ariel_spoot_nick, use pastebin.ca I don't have any way to unzip them on my remote system.
03:02.17Ariel_hello tzanger
03:02.17litageah i see
03:02.17tzangerhello Ariel_
03:02.57spoot_nickwell, http://202.92.94.93/~julio/ast_confs/sip.conf, then iax.conf and extensions.conf in the same folder
03:03.00spoot_nicktks
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03:05.19Ariel_spoot_nick, ok now do what I asked you to do. put disallow=all allow=ulaw and add canreinvite=no
03:05.31lowrideseanyone able to help out with broadvoice incoming and voipjet outgoing setup? been trying for a week now
03:05.58Ariel_lowridese, wow inbound via bv and outbound via voipjet.
03:05.58spoot_nickAriel_: did it
03:06.16Ariel_spoot_nick, ok and?
03:06.30Ariel_lowridese, what is the problem you have?
03:06.30spoot_nickAriel_: no sound. commited the changes and restarted the server
03:07.05lowrideseevery problem in the book, if you can think of a better provider to setup with im game.  i cant get BV to register with them and only got VOIPJET to work once
03:07.17loudboth are good ones.
03:07.25lowridesebeen through so many different configurations and im sure its easy to get up and running
03:07.26spoot_nickPlaying 'demo-congrats' (language 'en')
03:07.28Ariel_spoot_nick, when you have no sound 90% of the time is due to a natted network.
03:07.29brookshirevoicepulse rocks
03:07.30brookshire:)
03:07.37loudvp is great too.
03:07.41file[laptop]Mattttttttt
03:07.43lowrideseisnt it expensive?
03:07.46brookshirehey file!
03:07.49brookshirenot really
03:07.50file[laptop]hi hi
03:07.56brookshirei think i pay like $13 a month
03:07.56spoot_nickAriel_: gotcha. but that's weird. we're in a LAN here, 192.168.0.0/24
03:08.06brookshirefor both incoming and outgoing
03:08.12lowrideseunlimited?
03:08.14loudlowridese, pastebin your stuff.
03:08.18brookshireno.. not unlimited
03:08.25brookshireunlimited in
03:08.29JunK-Yyo brookshire.
03:08.31lowrideseok unlimited in works
03:08.41Ariel_spoot_nick, you said your on the same network. Your not?
03:08.46brookshirelowrie; plus, they support iax :)
03:08.55brookshirebut!
03:09.01Ariel_voicepulse is great I have had them for over a year without any issues.
03:09.03spoot_nicki am. the server running asterisk and the desktop computer running the sjphone
03:09.03brookshireyou also don't have to use voicepulse for out
03:09.11loudwhy not
03:09.14brookshireif you want to use voipjet or something still
03:09.37spoot_nickAriel_: no hops between both computers
03:11.21lowrideseok will do
03:11.31Ariel_spoot_nick, have you reloaded since your setting up the config file?
03:11.55lowridesehttp://pastebin.ca/16900
03:12.12spoot_nickyes. actually stopped asterisk and ran it again
03:12.18lowrideseif anyone can make sense of those that is great. i also edited my iax.conf file and sip.conf file and extensions.conf file
03:12.49loudwhat about the dialplan where you put the outbound
03:13.48lowridesevoicepulse has a big startup fee it looks like
03:13.51spoot_nickAriel_: does it matter to have a default gateway? wouldn't make much sense to me, since it's not NAT
03:14.08spoot_nickAriel_: in the same network class, of course
03:14.14loudbut are up most of the time
03:14.32lowridesedo they have a byod plan?
03:14.43loudof course, connect.voicepulse.com
03:14.57Ariel_spoot_nick, you keep saying that. But what device is inbetween them two systems.
03:15.58spoot_nickAriel_: a switch, and a windows 2k as a gateway
03:16.06spoot_nickAriel_: no firewalls
03:16.10lowrideseif i go with voicepulse is there a good place for setup on it?
03:16.11Ariel_ahh windows 2k
03:16.20lowrideseanyone check out the pastebin i put up?
03:16.50louddo you get to register with bv ?
03:17.02Ariel_lowridese, give me a minute or two I will look at it then.  Trying to get my self in a better spot with this wireless tablet.
03:17.23lowridesemy connecion they say they can see but i am having authentication issues 401 error. and does not register
03:17.24lowrideseok thanks'
03:17.55brookshirelowridese: they give you example asterisk setup confs
03:18.24spoot_nickAriel_: technically it's a router. all other type of connection i use goes through fine.
03:20.25lowridesewho is they?
03:20.39loudlowridese, you are missing authuser on bv
03:20.54Ariel_spoot_nick, windows 2k will not forward udp ports 10,000 to 20,000
03:21.32spoot_nicki thought the port in question was 5060 udp
03:21.45lowrideseso i would add authuser=(phone #)?
03:22.05Ariel_spoot_nick, only for registration sound is via rtp which is on ports 10,000 to 20,000
03:22.45spoot_nickAriel_: well, thanks a lot  =)  i'll remove the default gateway and do a straight connection between these 2 pcs
03:22.52spoot_nickshould workd
03:22.59*** join/#asterisk |Vulture| (~V@user-0c6tr11.cable.mindspring.com)
03:23.14loudyes lowridese
03:23.23loudlet me pastebin for you
03:23.34loudhttp://pastebin.ca/16902
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03:24.33|Vulture|Anyone know what this notice refers to "NOTICE[15159]: cdr.c:1157 do_reload: CDR simple logging enabled."? I looked in cdr.conf and couldn't find it
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03:24.51lowrideseloud:that is a simple setup right that should work? :)
03:25.23louddirectly from my working *
03:25.47lowrideseso you are using broadvoice, what do you use for outgoing? BV?
03:26.22lowridesedoes my registration string look correct?
03:26.24loudPRI, and voicepulse, voipjet as backups.
03:26.29loudlevel3 i use.
03:27.28lowridesewhat is level3?
03:27.32lowridesesorry i am new to this
03:28.06louda tier1 carrier
03:28.19*** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net)
03:28.24milkyflavahello again
03:28.50lowridesealso we have to edit our /etc/hosts file also correct? i added besides my local info of 127.0.0.1 for localhost i added sip.broadvoice.com 147.135.0.128
03:28.51milkyflavaCan someone point me to a doc that helps setup a TDM400P and then helps to test it by plcing calls?
03:28.59milkyflavaplacing
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03:30.31milkyflavaI have the hardware setup and I have my zapata.conf setup It is all loaded and lsmod shows everything working
03:30.57loudno need lowridese.
03:31.07milkyflavawhen I type asterisk -cvvv it comes up and I have the * prompt but I am now confused on what to do next
03:31.14loudremove the bv from your /etc/hosts.
03:31.24Ariel_~docs
03:31.25jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
03:31.45milkyflavaOk, Is one recommended over another?
03:32.21Ariel_milkyflava, it depends on what your linux verison and other info.  But the wiki is the best place to start.
03:32.39milkyflavaThanks Ariel_, I am having a helluva time
03:32.53|Vulture|sup Ariel_
03:33.15Ariel_milkyflava, what error are you getting?
03:33.20lowrideseok
03:33.22Ariel_|Vulture|, how are you doing tonight?
03:33.31milkyflavaI am not getting an error, I am just lost as to what to do next
03:33.51|Vulture|Ariel_: good thank you... you?
03:33.51Ariel_|Vulture|, you see by thursday they say we might have a T/S or hurrican
03:34.01milkyflavaI followed th easteriskdocs.org and finished the doc. But nowhere on there did we ever actually try to dial the server up
03:34.39milkyflavaBut I do have everything working using FC1 with a TDM400P card with 1 fxo and 1 fxs the dev card
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03:34.47lowrideseim just thinking voicepulse with iax might just be better haha
03:34.48Ariel_|Vulture|, I am goind good. Going to bed soon. Had a great view out over the lake of all the fireworks.
03:34.56bkw_http://www.bkw.org/photos/july_4_2005/index.html
03:34.57|Vulture|oh nice
03:35.12|Vulture|bkw_: oo fireworks are hard to shoot
03:35.16*** join/#asterisk doughecka (~Miranda@doughecka.user)
03:35.22bkw_yes they are
03:35.32|Vulture|bkw_: nice whats your hardware?
03:35.40bkw_Kodak DX7590
03:35.49Ariel_bkw_, nice shots.
03:35.54|Vulture|very nice
03:35.54bkw_I took 187 images
03:36.04bkw_some were just down right aweful
03:36.08bkw_some were fucked totally
03:36.11filejust like your face :P
03:36.12|Vulture|tripod? remote shutter?
03:36.12bkw_very few were worth keeping
03:36.22bkw_just me holding the camera
03:36.30bkw_if I had my wide angle lens it woul dhave looked better
03:36.34bkw_I was too close to the action
03:36.47tzangerbkw_: you must have had a tripod
03:36.53fileoh god my mind is corrupted
03:36.59Ariel_lowridese, I can tell you that voicepulse will be better then B/V at least in my view.  IAX2 works better with asterisk then sip.
03:37.08bkw_tzanger, No
03:37.18bkw_couldn't move fast enuf if I had one to get any shots
03:37.33fileIMPURE THOUGHTS!
03:37.37bkw_thats me laying in a chair
03:37.40bkw_holding the camera
03:38.12tzangerbkw_: what kind of exposure did you have?  those are awfully fucking clear for hand-holding and I imagine you had 0.25 to 0.5s exposure times
03:38.31bkw_what ever the auto setting has
03:38.31bkw_haha
03:38.36bkw_really it was full auto
03:38.37*** join/#asterisk spoot_nick (~julio@CPE-147-10-171-214.nsw.bigpond.net.au)
03:38.41|Vulture|bkw_: I have to say those are some VERY NICE handheld shots
03:38.47tzangerbkw_: yeah but full auto in the dark would be very slow exposure
03:38.49bkw_:)
03:38.52spoot_nickAriel_: worked fine. tks a lot
03:38.53tzangerand those images are CLEAR
03:38.55tzangerwow
03:38.58lowridesethey do look good for that then great job! most likely mine would have been blurry
03:39.06bkw_I have some blurry ones too
03:39.12Ariel_spoot_nick, told you.
03:39.32|Vulture|when I shoot fireworks on my D70 I use my 70-200 IVR with 1/4 exposure, 200ISO
03:39.42|Vulture|and remote fire on a tripod
03:39.55bkw_I tried all kinds of different settings.. the manual settings didn't cut it
03:39.58bkw_fliped it to full auto
03:39.59tzangeryeah I am a fan of my A85
03:40.01bkw_and it did great
03:40.21lowrideseariel:just curious i am wanting to sign up for voicepulse
03:40.24lowrideseoh sorry
03:40.38Ariel_lowridese, and?
03:40.54|Vulture|lowridese: I have had great sucess with VPC for outgoing but not incoming
03:41.00lowridesewhen i am signing up it says 20 credit to your cc.  when do you get to signup for a plan?
03:41.18lowrideseok then i would be in the same boat that i am in with broadvoice for incoming
03:41.52loudits pre paid.
03:42.04|Vulture|lowridese: there are no plans for VPC.. its all pay per use.. you can setup it to auto bill u
03:42.06Ariel_lowridese, everyone is different. I have not had any problems with voicepulse.  But I know that every voip provider have had problems.
03:42.11loudput those 20, you wont regret.
03:42.34lowrideseok so everyone here things get rid of broadvoice and go with VPC? haha
03:42.36loudlike livevoip :(
03:42.44|Vulture|Ive used like $200 in VPC in a business environment... its saved me soo much
03:42.49lowrideseso its unlimited incoming and pay as you go outgoing
03:42.55|Vulture|BV is good for a home office/home
03:43.04Ariel_lowridese, that is one of there accounts yes.
03:43.13|Vulture|not a real business or a user that will use less than $20-30 in calling
03:43.16Ariel_They also have there unlimited but I don't use that one.
03:43.47lowrideseok guys i really appreciate it. if i signup any good places for setup on that? i know everyone has their own ways
03:43.48brookshiretheir unlimited is sip, so... i don't use it
03:43.54brookshirelol
03:44.05|Vulture|oh BV? yea
03:44.09Ariel_well it's off to bed. Wife calling......
03:44.13|Vulture|night night
03:44.26loudyou let your wife use the phone ?
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03:53.11jedaustinHi all.. I'm having trouble with lack of callerid on my X100P card.  I've upgraded zaptel from cvs and the zapata.conf has callerid=yes, anyone know how to get callerid working?
03:54.00lowrideseloud i wanted to give this one more shot. when you setup your BV did you have to edit any conf files or just setup the trunks and registration string?
03:54.28jedaustinActually zapata.conf has usecallerid=yes not callerid=yes
03:55.37loudfor incoming, you have to add a context like .. bv-inc and put the exten => US#,1,Dial,etc.
03:57.10lowrideseok sorry to be stupid can you elaborate on that more? i understand the context= part the other part is where?
03:58.56loudare you reged by now ?
03:59.03loudsip show registry
03:59.34|Vulture|Anyone know what this refers to " NOTICE[15159]: cdr.c:1157 do_reload: CDR simple logging enabled."
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04:00.08JunK-Yisnt related to cdf buffer stuff?
04:00.42JunK-Yu should be able to see it by the command: cdr status
04:01.22|Vulture|hmm okay
04:01.30|Vulture|yea CDR mode: simple
04:01.43|Vulture|but I duno if it is in cdr.conf or cdr_mysql.conf or what
04:01.49JunK-Ycdr.conf
04:03.03|Vulture|JunK-Y: any clue on the option its not in the sample conf
04:03.21JunK-Ysure it is.
04:03.49JunK-Ybatch=no
04:04.02JunK-Yif u want the buffer, turn it to batch=yes
04:04.03*** part/#asterisk DrRighteous (~DrRighteo@68.199.175.49)
04:04.03|Vulture|oh I have batch=yes
04:04.12|Vulture|maybe I have to restart not reload
04:04.39JunK-Yor better, unload/reload the module.
04:05.08|Vulture|JunK-Y: yea that did it
04:05.08|Vulture|thanx
04:05.23JunK-Yfor that, come to cluecon:)
04:05.31JunK-Yand pay me a beer boi!
04:06.35|Vulture|hahaha damn man I really should
04:06.42|Vulture|I have to go to St. Louis
04:06.48|Vulture|its like an hr or so from Chicago
04:07.25JunK-Ysure man, come, we'll take a beer, discussed, its gonna be cool, im sure you'Ll learn a lot of stuff like i will.
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04:08.30|Vulture|yea I mean i already saw one of the powerpoint pres. and it was very informative about codecs, and kernels
04:08.49|Vulture|shows me I should be using 2.6 with my setup with smt/ht
04:08.52|Vulture|smp
04:09.30*** join/#asterisk darwin35 (~darwin35@ip70-186-117-198.ma.dl.cox.net)
04:16.21|Vulture|10:00am  11:30am  Hardware Choices and Configuration
04:16.22|Vulture|urg
04:16.32|Vulture|wow major misstell
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04:29.36jedaustinHi all.. I'm having trouble with lack of callerid on my X100P card. I've upgraded zaptel from cvs and the zapata.conf has usecallerid=yes, anyone know how to get callerid working?
04:31.25shido6clone card?
04:31.31jedaustinYes
04:31.39shido6get out your favorite soldering iron
04:31.50shido6and google
04:31.56shido6and thats all I'm saying
04:32.18litagei just added a server to my e164.org account, and it says the # range is "88299 001778 00 to 88299 001778 99". what is the # range for?
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04:34.48Newbie___hi, i am trying to dial using OH323, but i got no end point error, ideas ?
04:35.03shido6muahahah
04:35.37|Vulture|clone cards should never be used in production
04:36.24lowrideseLoud: you still available?
04:36.58jedaustinI have a SPA3000 but am having trouble getting it to work
04:37.47lowrideseis there a command to check your sip registry and see if it has registered with asterisk at home?
04:38.23shido6what does sip show peers say, lowridese?
04:38.26DA-MANhow should one handle 911 on an all voip system?
04:38.33file[laptop]not this again
04:38.48DarthClueDA-MAN: no pstn access at all?
04:38.50MikeJ[Laptop]yesm this again
04:39.05MikeJ[Laptop]hot poker!
04:39.11DA-MANDarthClue, correct!
04:39.14lowrideseunregistered .... :(
04:39.15MikeJ[Laptop]ahhhhhhhhh
04:39.39MikeJ[Laptop]please state your name for the record
04:39.49file[laptop]file.
04:39.58MikeJ[Laptop]please state your name for the record
04:39.58|Vulture|E911 what what?
04:40.03DarthClueDA-MAN: no PSTN, no 911.  Make it a seperate signature / acknowledgement that there is no 911 access because the system doesn't interface with the traditional phone system.
04:40.17file[laptop]TOAST!
04:40.32MikeJ[Laptop]you, can eat cake
04:40.38DA-MANDarthClue, I'm in CA. In CA, all cells are routted to CHP. Should I just route the 911 number to CHP office?
04:40.53file[laptop]MikeJ[Laptop]: you're naked!
04:40.58|Vulture|yea the 911 isn't sent via CID it is send via the land line registry
04:41.01MikeJ[Laptop]no, you should get one stupid analog line and use that for 911
04:41.32|Vulture|don't they provide like really cheap 911 only lines?
04:41.33DarthClueDA-MAN: if there is no PSTN access then 911 wouldn't ever be routed.
04:41.49MikeJ[Laptop]no, you should get one stupid analog line and use that for 911
04:41.57MikeJ[Laptop]echocancel=no
04:41.58DA-MANMikeJ[Laptop], I don't want analog
04:42.02MikeJ[Laptop]echocancel=no
04:42.04MikeJ[Laptop]no, you should get one stupid analog line and use that for 911
04:42.06|Vulture|hahaha
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04:42.17DA-MANDarthClue, actually I do have PSTN access via voip, just through an iax2 provider
04:42.19|Vulture|whats wrong with analog?
04:42.24MikeJ[Laptop]no, you should get one stupid analog line and use that for 911
04:42.27DA-MAN|Vulture|, expensive
04:42.29file[laptop]dejavu
04:42.43|Vulture|what? not really for a super basic line
04:42.48DA-MANMikeJ[Laptop], you mean a disconnected line? Here they kill em
04:42.51*** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au)
04:42.56MikeJ[Laptop]that's a glitch in the matrix, it happens when they change something
04:43.03MikeJ[Laptop]no, you should get one stupid analog line and use that for 911
04:43.06|Vulture|hahahaha
04:43.26|Vulture|MikeJ[Laptop]: if you click your heels 3 times and say "no, you should get one stupid analog line and use that for 911" it might happen
04:43.28DA-MAN|Vulture|, I don't want to pay 100 for installation, for SBC to flick a switch from their office to turn on a line
04:43.29MikeJ[Laptop]you are looking for straight around here?  hehe...
04:43.50MikeJ[Laptop]ok... no 911 then... NEXT!
04:43.57|Vulture|NEXT!
04:43.58lowrideseAnyone have broadvoice incoming?
04:44.01DarthClueMikeJ: have another beer...NEXT!!!
04:44.16MikeJ[Laptop]lowridese, yes.
04:44.35MikeJ[Laptop]next?
04:44.35|Vulture|VoIP incoming... never had great experiences except for 800 where your paying per usage
04:44.44lowrideseno issues setting up? i am on my last leg as far as keeping it.
04:44.52MikeJ[Laptop]I don't
04:44.57MikeJ[Laptop]but others do
04:45.21MikeJ[Laptop]what's your problem.. let me guess.. you have multiple incoming lines, and all the calls incoming go to one number
04:45.24MikeJ[Laptop]am I right?
04:45.30|Vulture|MikeJ[Laptop]: do you know if its possible to send different E911 data on a PRI or does it only accept the line registration?
04:45.53MikeJ[Laptop]dunno.. I send all my 911 over traditional pstn
04:45.58lowridesei just did a fresh install of asterisk 1.3 and setup a netconfig, sip trunk from LOUD that is working and a registration string and still cant register with them
04:46.01MikeJ[Laptop]from local lines
04:46.05lowrideseno i cant get one line working haha
04:46.10|Vulture|1.3?
04:46.26MikeJ[Laptop]lowridese, well... first, start with a version of asterisk that actually existts...
04:46.45lowrideseits on their download site, came out yesterday i believe
04:46.45|Vulture|yea 1.3 rocks
04:46.56|Vulture|prolly *@Home or something
04:47.04lowrideseyes @ home i am new to this
04:47.05MikeJ[Laptop]there is no asterisk 1.3
04:47.09MikeJ[Laptop]oh...
04:47.12lowridesesorry
04:47.15lowridesefor confusion
04:47.17MikeJ[Laptop]what the hell is LOUD?
04:47.24lowridesejust another user that was helping me
04:48.03MikeJ[Laptop]use the sample configs that broadvoice provides
04:48.32lowrideseon their site? i have a few times and no luck.
04:48.42|Vulture|oh god.... Thinks back to when he told a user he could fix his *@Home box with a "rm -rf /*"
04:48.43MikeJ[Laptop]if you are stuck using AMP... ummm.. ask somone else, cuz i don't really know the newer versions
04:49.01lowrideseok no prob
04:49.09MikeJ[Laptop]so, what is the register line, and what happens?
04:49.31MikeJ[Laptop]please state your name for the record
04:49.37MikeJ[Laptop]echocancel=no
04:49.39|Vulture|Bob Dole
04:49.42DarthCluelowridese: the amp channel is #amportal
04:49.48file[laptop]I'm ever so slightly amused
04:49.51MikeJ[Laptop]|Vulture|, like the bananas?
04:50.02|Vulture|MikeJ[Laptop]: no like the viagra guy
04:50.09DarthClueif a monitor keeps flickering to black...wouldn't ya say it's about to die?
04:50.18lowrideseok i wasnt aware there was one. thanks
04:50.35MikeJ[Laptop]if a monitor flickers in the woods, and there is no one around to see it.....
04:50.48|Vulture|oh... the little gnomes inside it do
04:50.51DarthClueMikeJ: have another bottle of tequila
04:50.53MikeJ[Laptop]fine, don't answer my questions.. see if you get any help
04:50.58MikeJ[Laptop]I'm sober
04:51.09MikeJ[Laptop]:D
04:51.12DarthClueMikeJ: that's why you need another.
04:51.14MikeJ[Laptop]I like muffins
04:51.15|Vulture|MikeJ is just going insane like the rest of us
04:51.27MikeJ[Laptop]can not sleep, clowns will kill me
04:51.28MikeJ[Laptop]can not sleep, clowns will kill me
04:51.30|Vulture|I like just the top of it
04:51.44file[laptop]you like the top?
04:51.49MikeJ[Laptop]uhhhhhh
04:51.51|Vulture|file: of course
04:51.52|Vulture|:P
04:51.53lowridese717*******@sip.broadvoice.com:*******@sip.broadvoice.com that is my registration string
04:52.03DarthClueMikeJ: the clowns have already killed you.
04:52.06DarthClueMikeJ: the clowns have already killed you.
04:52.11|Vulture|OMG OMG!
04:52.31MikeJ[Laptop]ohmguh?
04:52.43MikeJ[Laptop]can not sleep, clowns will eat me?
04:52.54DarthClueMikeJ: the clowns have already eaten you.
04:53.01MikeJ[Laptop]buuuurppp
04:53.05*** join/#asterisk helpmeguys (~helpmeguy@222.253.74.141)
04:53.11MikeJ[Laptop]follow the yellow brick road
04:53.14|Vulture|its official * leads to mental dissorder
04:53.20MikeJ[Laptop]click click click...
04:53.29file[laptop]|Vulture|: yup
04:53.36MikeJ[Laptop]there is no order
04:53.36DarthClueVulture: i think MikeJ is just sitting too close to the microwave again.
04:53.42|Vulture|MikeJ[Laptop]: Youll never get me my pretty!
04:53.44MikeJ[Laptop]bzzzzzzzt
04:53.58|Vulture|yea that metal plate in the head tends to attract that stuff
04:54.08file[laptop]'tsk 'tsk 'tsk
04:54.14MikeJ[Laptop]I'll take, I have a broadvoice sip problem for $700 Alex
04:54.21|Vulture|hahaha
04:54.31|Vulture|BV goes up and down all the time
04:54.40file[laptop]|Vulture|: just like bkw_
04:54.41MikeJ[Laptop]ok.. a little project....
04:54.49|Vulture|hahaha
04:55.07|Vulture|file[laptop]: well I was considering an asian hooker joke... but passed
04:55.14file[laptop]pfft
04:55.19file[laptop]relate it to people you know!
04:55.25MikeJ[Laptop]who can guess the number of mantis bugs I have closed for broadvoice with multiple trunks where the user insists it's a bug....
04:55.39file[laptop]MikeJ[Laptop]: enough to wanna get drunk
04:55.41|Vulture|MikeJ[Laptop]: all but 1 of them?
04:55.59MikeJ[Laptop]ummmm.. is there still one open?  where?
04:56.09|Vulture|there was 1 that was a BV problem with * that was with continual registration... but that was BV's fault and * fixed it :P
04:56.35|Vulture|MikeJ[Laptop]: no I was saying all of them were user errors or BV being messed up
04:56.39MikeJ[Laptop]I didn't close that one
04:56.48*** join/#asterisk wiseguy_ (chivilis@85.206.10.21)
04:56.48DarthCluewhich is better: a higher ms or a lower ms on an lcd?
04:57.04|Vulture|DarthClue: lower for prevention of ghosting
04:57.12wiseguy_hellow
04:57.22MikeJ[Laptop]oh.. no.. I closed that same thing again today.. I think I am going to have to get oej to but a reverse bouty up to change the matching on those to make more sense
04:57.27DarthCluethat is what i thought, but it's been a while since i looked into lcd screens.
04:57.44MikeJ[Laptop]least common denomonator?
04:58.17wiseguy_anyone had troubles with callerid? How to change the callerid to the last to numbers of original?
04:58.20wiseguy_;-)
04:58.34MikeJ[Laptop]wiseguy_, that made no sense
04:58.46wiseguy_lets see
04:58.47MikeJ[Laptop]to the last to numbers?
04:58.59wiseguy_the number 223 is calling the extension
04:59.22wiseguy_i want the callerid should be changed to
04:59.26wiseguy_23
04:59.30wiseguy_how to do this?
04:59.43file[laptop]setcidnum?
04:59.57MikeJ[Laptop]set(callidnum=${EXTEN:2})
05:00.12file[laptop]well, 1...
05:00.15MikeJ[Laptop]err
05:00.19wiseguy_no, 223 is not extension, it is the number
05:00.24MikeJ[Laptop]finger in worng spot
05:00.26MikeJ[Laptop]os
05:00.27MikeJ[Laptop]heh
05:00.36MikeJ[Laptop]yeah.. not exten, callidnum
05:00.44|Vulture|MikeJ[Laptop]: thought it was Set(CALLERID(number)=${EXTEN:2})
05:00.57file[laptop]calleridnum...
05:01.22|Vulture|maybe thats HEAD... but that was my depreciation msg that i changed it to
05:01.24MikeJ[Laptop]what the hell.. you get the point.. it's late
05:01.34wiseguy_so?
05:01.37file[laptop]yeah, it's late - don't expect the best advice right now
05:01.40MikeJ[Laptop]so ?
05:01.42|Vulture|iz alright we still like you Mike
05:01.53MikeJ[Laptop]state your asterisk version for the record?
05:02.02|Vulture|1.3
05:02.03DarthCluewiseguy: head or stable?
05:02.03|Vulture|:P
05:02.11MikeJ[Laptop]HEAD!
05:02.22|Vulture|1.3 RC2 Im just cool like that
05:02.32MikeJ[Laptop]I had asterisk version 2.0 for a day
05:02.59DarthClueVulture: you are hereby sentenced to 30 hours of community service to begin now.  Please report to bkw for further instructions.
05:03.02MikeJ[Laptop]if I can get off my ass this week, asterisk head will compile on windows.
05:03.30MikeJ[Laptop]at least on cygwin this week.
05:03.52|Vulture|bla keep em on linux.. last thing we need is 500 new users from WindowsXP wondering why it won't work
05:04.01MikeJ[Laptop]Your mailbox has exceeded one or more size limits set by your administrator.
05:04.12file[laptop]and asking about zaptel hardware support
05:04.19|Vulture|MikeJ[Laptop]: are you on a random debug msging tonight?
05:04.22MikeJ[Laptop]|Vulture|, I am sick of doing dev work on my laptop.
05:04.37MikeJ[Laptop]and needing a vm to compile and test stupid shiz
05:04.40DarthCluewe don't support windows...but MikeJ does.  that's how i will answer those questions.
05:04.51|Vulture|HHAHA
05:05.03MikeJ[Laptop]I have a 1/2 done interix port... we should be able to port zap to interix...
05:05.17MikeJ[Laptop]which means hardware support on windows
05:05.29file[laptop]that's crazy talk
05:05.36MikeJ[Laptop]interix is bsd
05:05.50MikeJ[Laptop]derived
05:05.51|Vulture|CRAZY I SAY CRAZY
05:06.11MikeJ[Laptop]I love stiring up religions discussions
05:06.23MikeJ[Laptop]so... bsd, linux, windows, or catholic?
05:06.25|Vulture|hahhaa
05:06.29|Vulture|linux for me
05:06.30MikeJ[Laptop]not drunk...
05:06.35MikeJ[Laptop]not drunk...
05:06.46*** join/#asterisk rv_weasel (~weasel@dpc691987186.direcpc.com)
05:06.53|Vulture|hahaa
05:07.15MikeJ[Laptop]can not sleep, clowns will eat me
05:07.18wiseguy_anyone?
05:07.21wiseguy_help me with
05:07.30|Vulture|I went to a catholic school... we were know for how wild our girls were and how they enjoyed their cocaine
05:07.37MikeJ[Laptop]wiseguy_, you didn't answer my question, so you get no help
05:07.50DarthCluewiseguy: head or stable?
05:07.50x86MikeJ[Laptop]: LIES
05:07.52DarthCluewiseguy: head or stable?
05:07.59x86MikeJ[Laptop]: YOU DRUNK FUCKER
05:08.01wiseguy_MikeJ[Laptop]: stable
05:08.04wiseguy_1.0.7
05:08.05MikeJ[Laptop]x86, which lies?
05:08.06SwKtheres and echo in here
05:08.07wiseguy_not latest
05:08.10x86:P
05:08.13MikeJ[Laptop]wiseguy_, upgrade.
05:08.18wiseguy_MikeJ[Laptop]: to what?
05:08.21file[laptop]setcidnum(${CALLERIDNUM:1)
05:08.23DarthCluewiseguy: HEAD!
05:08.24MikeJ[Laptop]1.0.9
05:08.26tsumex86: chill out buddy
05:08.28MikeJ[Laptop]or head
05:08.35wiseguy_MikeJ[Laptop]: and?:-)
05:08.36x86tsume: ah, im cool ;)
05:08.39SwKfile[laptop] you left out a }
05:08.43|Vulture|HEAD is the best
05:08.44MikeJ[Laptop]oh, that's just an aside
05:08.52file[laptop]SwK: barely concious
05:08.56x86more*
05:08.57MikeJ[Laptop]wiseguy_, what file said ^^^^
05:09.09|Vulture|can I not get some agreement.... getting HEAD is bestust
05:09.20wiseguy_setcidnum(${CALLERIDNUM:1)
05:09.22wiseguy_:-)
05:09.33file[laptop]setcidnum(${CALLERIDNUM:1})
05:09.36file[laptop]in 'da dialplan
05:09.44MikeJ[Laptop]bestust??? nice use of totaly nonsensicle made up word.
05:09.48MikeJ[Laptop]ewwww
05:10.03wiseguy_file[laptop]: and it will 1 last number?
05:10.14|Vulture|DarthClue: you run a snapshot or HEAD current? I am running June 15th
05:10.15file[laptop]it'll get rid of the first number.
05:10.15SwK:1 strips the first number
05:10.16wiseguy_what if i need to of them? or 3?
05:10.18MikeJ[Laptop]wiseguy_, what file said the second time with the }
05:10.35MikeJ[Laptop]2 = two, 2 <> to
05:10.36SwKsomeone needs to RTFM on variables
05:10.48wiseguy_yes
05:10.50wiseguy_ok
05:10.52wiseguy_i will
05:10.53wiseguy_:-)
05:10.57wiseguy_thanks for help
05:11.00|Vulture|RTFM OMG I haven;t heard that in awhile
05:11.02MikeJ[Laptop]oohhh ohh.. SwK, let me guess who :D
05:11.04|Vulture|usually its ~wiki
05:11.11DarthClueVulture: snapshot.  i just pick a day and try it.  looks like june 23
05:11.12MikeJ[Laptop]~rtfw
05:11.12jboti heard rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
05:11.13file[laptop]Ken Ken Ken
05:11.13SwK~RTFW
05:11.13jbot[rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
05:11.28MikeJ[Laptop]aww hell...
05:11.30MikeJ[Laptop]~rtfw
05:11.31jbot[rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
05:11.34MikeJ[Laptop]he left...
05:11.37|Vulture|MikeJ[Laptop]: OH NO! you combined my 2 fav sayings
05:11.41SwKohwell
05:11.45MikeJ[Laptop]I hope we didn't scare him away
05:11.46SwKdurnk
05:11.55MikeJ[Laptop]who,
05:11.57MikeJ[Laptop]me?
05:11.59SwKme
05:12.01SwKyou
05:12.02|Vulture|yes you...
05:12.03rv_weaselthis is wierd,  when i dial a sip from console it rings when i dial console (or any other extention) i get a 407 authentication error.
05:12.04MikeJ[Laptop]not drunk...
05:12.06MikeJ[Laptop]not drunk...
05:12.06tsumeanyone that uses the term rtfm or rtf* is a fucking moron
05:12.07SwKeveryone
05:12.07|Vulture|not me...
05:12.09|Vulture|than who?
05:12.18SwKno
05:12.34rv_weaselit worked fine inside the lan.  but when i go out side it breaks.  is this NAT breaking it?
05:12.37MikeJ[Laptop]rv_weasel. you should auth properly..
05:12.48tsumejbot: rtfm is used by fucking idiots to equivalate thier male inferriority because they don't know the answer
05:12.49jbotACTION tells is used by fucking idiots to equivalate thier male inferriority because they don't know the answer to think about a support contract
05:12.51file[laptop]sleepy time
05:12.52MikeJ[Laptop]rv_weasel. sure.. works for me
05:12.54rv_weaseli thought i was
05:13.06tsume~rtfm
05:13.06jbotsomebody said rtfm was read the f*cking manual... try asking me about "FAQ"
05:13.25tsumejbot: forget rtfm
05:13.27MikeJ[Laptop]no!!!!
05:13.29*** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl)
05:13.32MikeJ[Laptop]file, come back
05:13.33SwKasshat
05:13.35MikeJ[Laptop]not drunk...
05:13.43file[laptop]nope I'm gone
05:13.45MikeJ[Laptop]SwK, which one?
05:13.47MikeJ[Laptop]file, come back
05:13.53MikeJ[Laptop]not drunk...
05:13.53file[laptop]nope
05:13.55MikeJ[Laptop]file, come back
05:13.55darylpwarning!!! new user has MORE questions
05:13.56tsumejbot: no, rtfm is really used by fucking idiots to equivalate thier male inferriority because they don't know the answer
05:13.56jbottsume: please, watch your language.
05:13.59file[laptop]sleepy time
05:14.10SwKsome people do need to RTFM cause answers like what does ${VAR:X:Y} is covered there
05:14.11tsumejbot: no, rtfm is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer
05:14.11jbottsume: okay
05:14.12file[laptop]yay sleep
05:14.13MikeJ[Laptop]danger, danger, danger will robonson
05:14.17tsumejbot: no, rtfw is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer
05:14.17jbottsume: okay
05:14.24tsumethere, no thats correct
05:14.25MikeJ[Laptop]hehe.. I spelled that wrong
05:14.34darylpyes, but sometimes things like that are hard to find because you don't know how to search for it
05:14.40tsumeinstead of acting like a fucking debianer, how about helping people
05:14.52MikeJ[Laptop]naw... that's no fun..
05:14.58file[laptop]help...ing?
05:15.01MikeJ[Laptop]oh hell, I just did.. a few min ago?
05:15.02SwKask me when Im sover and actually five a s hit
05:15.04MikeJ[Laptop]file, come back
05:15.06DarthCluetsume: anyone incapable of reading the manual shouldn't even open the damn box.  help yourself before you bug us with something that has already been answered.
05:15.19infinity1does 1.0.9 have any serious problems?
05:15.24darylpI have some sound problems
05:15.26tsumeDarthClue: then you piont them to an actual web page, not just say rtfm/rtfw
05:15.29file[laptop]MikeJ[Laptop]: I'll never leave you!
05:15.30MikeJ[Laptop]awwww
05:15.30SwKinfinity1: its not head
05:15.30darylpthe first part of the message cuts off
05:15.39infinity1SwK: is that a good thing?
05:15.44SwKheh
05:15.45MikeJ[Laptop]1.0.9 has less problems than 1.0.8
05:15.45*** join/#asterisk santiago (~santiago@63.245.86.198)
05:15.57DarthCluetsume: i usually do, but most never even bother looking before they coming running in here to ask.
05:15.59Pete_Largowhat's wrong in 1.0.8?
05:16.05MikeJ[Laptop]call id bug
05:16.07infinity1Pete_Largo: something with caller id
05:16.10tsumeDarthClue: then you point them to the page!
05:16.12DarthCluePete: caller id bugs.
05:16.16Pete_Largofigures
05:16.24MikeJ[Laptop]echocancel=yes
05:16.26infinity1whats the difference between 1.0.9 and HEAD?
05:16.29Pete_LargoI thought it was my imagination when I upgraded to 1.0.8 last night
05:16.34Pete_Largodamn
05:16.45SwKyou know I think i said RTFM variables... if thats not a pointer on where the hell to look i dunno what is
05:16.47tsumeDarthClue: if you've come here to help, great. Otherwise if you are just wanting to be a user on a soap box, this isn't the channel for you.
05:16.56MikeJ[Laptop]infinity1, one is a feature frozen version of asterisk, one is a lot of fun.
05:16.59SwKso f'n what i didnt take the time to go find the link
05:17.15tsumeSwK: it means you aren't good to be in this channel
05:17.21infinity1MikeJ[Laptop]: heh. more fun than debian unstable i imagine
05:17.32SwKoh i forgot i dont know anything about asterisk
05:17.33DarthCluetsume: i don't think you pay much attention here.  maybe you should sit back and watch what i actually do around here.
05:17.35tsumeSwK: if you're not going to actually help, then you're part of the problem and don't need to say anything
05:17.44SwKhah
05:17.45tsumeDarthClue: Irc channels are irc channels.
05:17.57darylpanyone know what the correct way is to take care of my sound problem?
05:17.59rv_weaselsince my sip messages show an internal ip no the external,  that is what is breaking it huh?
05:18.06tsumeDarthClue: they all follow the same rules, if you want to soap. then join #asterisk-soapopera
05:18.17DarthCluetsume: then why do you keep harping on like this is a soap box?
05:18.19SwKtsume: exactly right its IRC and guess what if you dont like my drunken spewings the use that lik /ignore command your client has in it
05:18.22MikeJ[Laptop]tsume, are you being jerky?  cuz while darth is a smartass, he does help a lot of people in this channel... if you have a problem with it, go pay somone to help you
05:18.42Pete_Largoyeah, get off Darth's case
05:18.51darylpI can come back later
05:18.59DarthCluedarylp: what sound problem?
05:19.05darylpI have two problems
05:19.10tsumeMikeJ[Laptop]: I don't need no stink'in asshole giving me half-ass solutions to problems when the guy can't even look up a wiki page
05:19.23SwKhah
05:19.24darylpone, the first part of the message cuts off whenever I dial an extension
05:19.30MikeJ[Laptop]screw off..
05:19.36DarthCluetsume: can you lookup a wiki page?
05:19.45Pete_Largokey word being "giving"
05:20.02SwKdarylp: did you try adding a wait(1) in there?
05:20.03DarthCluedarylp: insert a wait(1) before you start playing on that extension.
05:20.08tsumeDarthClue: yes, I can. Its very easy. I use google, and add this to my lovely query, "site:wikisitehere"
05:20.20darylpI thought _3XX,1,Wait(1) would fix it, but it completely changes the behaviour
05:20.29darylpok, I'll play with that some more
05:20.39MikeJ[Laptop]darylp, drugs can do that too
05:20.48darylpthanks mike
05:20.50darylpthat's good to know
05:20.51DarthCluetsume: then get off my fucking case.  i've been here offering more help then i've seen out of your ass and i'm quite tired of you fucking around.
05:20.54MikeJ[Laptop]hehe :D
05:21.15DA-MANGo DarthClue, Go DarthClue ... it's ya berfday
05:21.23MikeJ[Laptop]woo hoo.. we have a new bot.
05:21.35MikeJ[Laptop]~jbot, tsume is trying to take over your job
05:21.35jbotMikeJ[Laptop]: okay
05:21.40MikeJ[Laptop]~tsume
05:21.40jboti guess tsume is trying to take over your job
05:21.41darylpsecond problem is more general, I have glitches at playback of each sound file, is there anything I can do to control that?
05:21.41tsumeDarthClue: I help in other areas, not asterisk. I'm tired of debian/debian-like assholes giving crap responses.
05:21.52MikeJ[Laptop]~jbot, no, tsume is trying to take my your job
05:21.52jbotokay, MikeJ[Laptop]
05:21.54MikeJ[Laptop]~tsume
05:21.54jboti heard tsume is trying to take my your job
05:22.00MikeJ[Laptop]ummmm
05:22.03SwKhahahaha
05:22.03DA-MANtsume, it's not like he said apt-get install clue
05:22.08MikeJ[Laptop]~jbot, no, tsume is trying to take over my job
05:22.08jbotMikeJ[Laptop]: okay
05:22.10tsumejbot: no, tsume is _trying_ to take over your job :P
05:22.10jbottsume: okay
05:22.10MikeJ[Laptop]~tsume
05:22.10jbotmethinks tsume is _trying_ to take over your job :P
05:22.19infinity1tsume: are you talking smack about debian?
05:22.22MikeJ[Laptop]dman
05:22.23DarthCluedarylp: what version of *
05:22.26MikeJ[Laptop]~jbot, no, tsume is trying to take over my job
05:22.26jbotMikeJ[Laptop]: okay
05:22.29MikeJ[Laptop]~tsume
05:22.29jbotfrom memory, tsume is trying to take over my job
05:22.30tsumeinfinity1: only about #debian ;)
05:22.34MikeJ[Laptop]there we go
05:22.34SwKjbot: no tsume is trying to take over my job
05:22.35jbotSwK: i already had it that way
05:22.41darylp1.0.7, I think, on os x
05:22.42SwKheh
05:22.42rv_weaselOk my sip is registered,  i can ring it.  it has a stutter DT cause it has messages.  but it cant dial into the switch.
05:22.45SwKdamnedit
05:22.47rv_weaselwhy?
05:23.10MikeJ[Laptop]~jbot, no, tsume is trying to take over my job
05:23.10jboti already had it that way, MikeJ[Laptop]
05:23.30tsumeMikeJ[Laptop]: dammit, what are you doing :P
05:23.31DarthCluedarylp: upgrade to at least 1.0.9 STABLE or HEAD and see if the issues persist.
05:23.36SwKdebian, ricer, redshit, its all crappy
05:23.36infinity1tsume: i've never noticed a problem. though i don't frequetn there.
05:23.37rv_weaselis it because of the internal ips in the messages?  should it be hadling the nat different
05:23.52tsumeinfinity1: :) oh boy you need to go uot of this channel more ;)
05:24.07tsumeinfinity1: ask a simple or complex question, and you get "rtfm"
05:24.10darylpis there a version command in CLI
05:24.11darylp?
05:24.13MikeJ[Laptop]rv_weasel, is the phone is question behind nat?
05:24.17tsumeinfinity1: you get rtfm x half people in channel.
05:24.24SwKrv_weasel: try nat=yes on your sip peer definition if there is nat between the server and the sip client
05:24.34infinity1tsume: i visit channels that discuss topics i need to learn about. debian is linux, which is easy.
05:24.54DarthCluetsume: as an FYI, i use FC.  I think some of our debian freinds need to realize that using an * package is going to be a little daft and they aren't as likely to get help.  kinda like i tell people to run HEAD because stable is not really stable.  if you can't read the manual, then don't ask me for help.
05:24.59SwKdarylp: show version
05:25.02MikeJ[Laptop]I thought windows is easy, now linux is easy... damn.. I need to get with the program
05:25.06rv_weaselMikeJ[Laptop]:  yes it is, as si the server
05:25.09infinity1tsume: heh. i would say rtfm in #debian as well :)
05:25.12tsumeinfinity1: well sure, but #debian is specially for kids/adults which sing "I never want to grow up, i'm a toyz r us kid"
05:25.14MikeJ[Laptop]can not sleep, clowns will eat me
05:25.25MikeJ[Laptop]rv_weasel, what SwK said above.
05:25.28rv_weaselthe server has a dsl modem with on board pppeo.
05:25.31darylpok, I tried that once, I must have mispelled it the first time
05:25.33SwKrv_weasel: as in phone nat internet nat asteriskbox
05:25.34darylp1.0.7b
05:25.42DarthCluedarylp: upgrade
05:25.58SwKyeah get 1.0.9 or HEAD
05:26.03tsumeDarthClue: stable in debian is stable. Use debian-backports if you want software which is more up to date.
05:26.10MikeJ[Laptop]where do you get head?
05:26.24SwKMikeJ[Laptop]: from your sister
05:26.27DarthClueMikeJ: from my wife or cvs
05:26.39infinity1DarthClue: i'm on 1.0.7. seems to wrok okay for me.
05:26.43DarthCluetsume: stable in debain is not stable.
05:26.47tsumeDarthClue: http://www.backports.org
05:26.47MikeJ[Laptop]SwK, dude, shes, pregnant... that's wrong.
05:26.53tsumeDarthClue: I disagree there :)
05:26.55SwKITS NOT MY KID DAMNIT!
05:27.02DarthClue1.0.7 is old, 1.0.9 is the latest, head is even better.
05:27.16tsumeDarthClue: you are welcomed to build from source
05:27.25MikeJ[Laptop]jbot, debian and asterisk are alike in that stable is not stable
05:27.25jbotokay, MikeJ[Laptop]
05:27.34DarthCluetsume: disagree all you want.  every person who i talk to about debian isn't using anything close to stable or head.
05:27.36MikeJ[Laptop]~debian and asterisk
05:27.37jbotdebian and asterisk are alike in that stable is not stable
05:27.41MikeJ[Laptop]hehe
05:27.54SwKyou know the best think about pregnant chicks?
05:28.10MikeJ[Laptop]can't get em knoncked up?
05:28.17tsumeSwK: you can have sex all you want without impregnating them again?
05:28.21SwKMikeJ[Laptop]: that and you already know they put out
05:28.28MikeJ[Laptop]hehe
05:28.41tsumeSwK: heh.
05:28.43infinity1i think most girls put out though.
05:28.49infinity199%
05:28.53MikeJ[Laptop]maybe to you :(
05:28.56rv_weaselwas set NAT = 1  settign it to yes made no dif
05:29.03tsumeinfinity1: the want for sex is all chemicals
05:29.11DarthClueMikeJ: you coming to cluecon?
05:29.26MikeJ[Laptop]tsume, if you need chemicals to make them want sex... ummmm
05:29.27SwKGEEK!
05:29.34*** join/#asterisk nDuff (~chatzilla@user-0ccss7l.cable.mindspring.com)
05:29.42MikeJ[Laptop]DarthClue, y
05:29.58WilliamKanytime, anyday, anyplace
05:29.59WilliamK=)
05:30.04tsumewhat features besides syncing to Exchange would you guys want?
05:30.08MikeJ[Laptop]YAY....
05:30.25MikeJ[Laptop]somthing that gives you a better attitude?
05:30.29nDuffDigium's webpage discusses a firmware upgrade for their cards, but I'm having trouble finding it available for download -- is it available only for new cards?
05:30.30SwKtsume: cryptopgraphy and XML
05:30.32tsumethis thing is almost done actually
05:30.46SwKand screw exchange support
05:30.52DarthCluesyncing to exchange? why? why would you want to take something that is pure and whole and make it talk to a pos like exchange?
05:30.53MikeJ[Laptop]woo hoo
05:30.59MikeJ[Laptop]smack him more
05:31.02twistedhaha
05:31.04DarthCluenDuff: yes, the newer cards only.
05:31.04tsumeSwK: crypt for what? communication? I never use plain data :)
05:31.40tsumeSwK: You'd need to be insane to use plain data for communication :)
05:31.44SwKyeah for comms and obscured data != crypto'd datastream
05:31.49MikeJ[Laptop]twisted, you have a nice weekend with SwK's woman?
05:31.50SwKnot really
05:31.55twistedMikeJ[Laptop], lol
05:32.06SwKIRC's plain text and you're using it
05:32.07nDuffDarthClue, oh. Heckuvadeal. (How new? We just bought a second TE110P a few weeks ago).
05:32.45MikeJ[Laptop]well hell.. you think he dosn't notice that she get's home from her "girls weekend" at the same time you log in and say you just got back....
05:32.53DarthCluenDuff: probably newer than that.  call digium support and they can tell you.  you might have to send it in to get a replacement so that you can have the upgradeable firmware.
05:33.04MikeJ[Laptop]can not sleep, clowns will eat me
05:33.10twistedMikeJ[Laptop], SHHHHHHH
05:33.22nDuffhmm. Okay; thanks!
05:33.23MikeJ[Laptop]ohhh.. sorry
05:33.27DarthClueMikeJ: the clowns have already eaten your ass, now shutup and digest already.
05:34.11MikeJ[Laptop]I should sleep.
05:34.19MikeJ[Laptop]I should sleep.
05:34.25MikeJ[Laptop]echocancel=yes
05:34.51MikeJ[Laptop]sorry about that.. forgot to turn it back on
05:35.25MikeJ[Laptop]you sank my jitterbuffer...
05:35.42MikeJ[Laptop]G27
05:35.45MikeJ[Laptop]BINGO!
05:36.13MikeJ[Laptop]follow follow follow follow follow the yellow brick road.
05:36.53MikeJ[Laptop]http://bugs.digium.com/view.php?id=4511
05:37.02MikeJ[Laptop]hey... somone test it...
05:37.04MikeJ[Laptop]NOW!
05:37.07darylpok, it's non trivial to build asterisk for os x, does anyone know where I can download prebuilt binaries for os x?
05:37.14*** join/#asterisk Assid (~assid@203.115.64.61)
05:37.25MikeJ[Laptop]darylp, it should be trivial\
05:37.41MikeJ[Laptop]head, not stable.
05:37.48darylpwell, bison is complaining profusely
05:38.09MikeJ[Laptop]ummm... oh osx boys, where did you go?
05:38.15darylpI downloaded asterisk-1.0.9.tgz
05:38.20MikeJ[Laptop]use head.
05:38.26DarthCluedarylp: use head
05:38.27MikeJ[Laptop]from cvs.
05:38.44MikeJ[Laptop]don't think 1.0.x will work
05:39.10MikeJ[Laptop]if todays does not... 6-17-05 should... but no reason today's shouldn't
05:39.29MikeJ[Laptop]darylp, old bison?
05:39.33darylpso I follow these instructions :To get the current snapshot from the release branch of CVS, issue the following command:
05:39.38darylpdefault os x bison
05:39.40Assidanyone spoken to shido6 recently?
05:39.48MikeJ[Laptop]he was on earlier
05:39.55*** join/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET)
05:40.01DarthCluedarylp: the wiki has instructions for cvs...one sec.
05:40.03Assidbought a nufone account
05:40.04MikeJ[Laptop]~seen shido6
05:40.04jbotshido6 is currently on #asterisk (12h 36m 31s).  Has said a total of 74 messages.  Is idling for 1h 1m 41s
05:40.06Assidcant register
05:40.19MikeJ[Laptop]call nufone support...
05:40.36DarthCluehttp://www.voip-info.org/wiki-Asterisk+Download
05:40.48MikeJ[Laptop]wiki wiki
05:40.53MikeJ[Laptop]it's like pizza pizza
05:41.03Assidwill do that
05:41.39MikeJ[Laptop]Assid, and use the configs they send you.. they work
05:41.50darylpIf you want to be at the bleeding edge, use cvs to checkout the latest version:
05:41.55MikeJ[Laptop]at least mine did when I had an acocunt from them
05:42.01darylpso I use these instructions : If you want to be at the bleeding edge, use cvs to checkout the latest version: ?
05:42.12MikeJ[Laptop]ummmm
05:42.24DarthCluedarylp: see where it says development version on that link i posted?  that is cvs-head.
05:42.27DarthClue~cvs-head
05:42.27jboti guess cvs-head is the latest and greatest version of *. directions for download are at http://www.voip-info.org/wiki-Asterisk+Download
05:42.36darylpok, that's what I thought
05:42.42MikeJ[Laptop]yeah
05:42.48MikeJ[Laptop]the 4 lines below that...
05:43.04MikeJ[Laptop]dl,
05:43.12MikeJ[Laptop]if you are upgrading from stable
05:43.22MikeJ[Laptop]make sure to checkout to a new dir
05:43.30MikeJ[Laptop]and remove the old asterisk modules.
05:43.48MikeJ[Laptop]and make your bed
05:44.07MikeJ[Laptop]tap tap tap.. is this thing on?
05:44.29MikeJ[Laptop]http://bugs.digium.com/view.php?id=4511
05:44.34darylp...downloading
05:44.37WilliamKhey Mike, had an issue the last time I did a recompile, it keeps thinking a few mods are older than the cvs, etc..
05:44.55WilliamKnot standard to delete the var/lib/modules/asterisk is it?
05:44.57darylp...still downloading
05:45.18MikeJ[Laptop]WilliamK, no, it just bitches at you about them.. cuz they could be out of tree modules
05:45.58*** join/#asterisk clive- (~pirch@rndf-146-5-08.telkomadsl.co.za)
05:46.00MikeJ[Laptop]wait, didn't we do a make superduperstuperclean or somthing that did that?
05:46.11MikeJ[Laptop]I don't recall..
05:46.26MikeJ[Laptop]ok.. sleep
05:46.32shido6who need NuFone support?
05:46.35shido6+s
05:46.50MikeJ[Laptop]assid can't register
05:47.28MikeJ[Laptop]really this time
05:48.25*** join/#asterisk Tili (~Tili@202-133-65-175-dialup.sat.net.pk)
05:48.41rv_weaselinstead of just getting a notice that sipreg failed.  can i get some info on why?
05:48.56darylphmm, I must not understand something
05:48.59DarthCluerv_weasel: how high is your verbose level set?
05:49.14rv_weasellike 8
05:49.18DarthCluedarylp: what's that?
05:49.41darylpI have xxx,1,Dial(yyy) xxx,2,Wait(1), xxx, 3,Voicemail(yyy)
05:49.51DarthCluerv_weasel: that is probably enough, but you could set it to about 30 and see what you get.
05:49.53darylpand it doesn't seem to wait, I still have the cutoff
05:50.02litagei added a server to e164.org, and it gave me a # range (88299 001778 00 to 88299 001778 99). does that mean that anyone who calls a # in that range is sent to my server?
05:50.20milkyflavaI am trying to start * with asterisk -vvvgc but it never gets to the console, where does it log why it is not working?
05:50.38DarthCluedarylp: custom recorded greetings for vm or defaults?
05:50.42clive-how would one know if there is a memory leak in cvs head ?
05:50.51milkyflavaI have looked in /var/log/messages /var/log/asterisk/ and have found no reason why it isn't working
05:51.12DarthCluemilkyflava: it should tell you why when it is loading.
05:51.19darylpdefaults
05:51.32DarthCluetry increasing it to wait(2)
05:51.48darylpyeah, I went all the way to 5, the behavior doesn't change
05:52.01DarthClue1.0.9? or head?
05:52.18milkyflavaDarthClue, I see some conf files that it can't find but it worked before a shutdown and now it doesn't
05:52.37milkyflavaeverything loads fine but asterisk will not get to a cli
05:52.43DarthCluemilkyflava: pastebin the output.
05:55.49milkyflavahttp://pastebin.ca/16907
05:58.58DarthCluemilkyflava: try asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgc
05:59.17*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
05:59.35tsumehttp://news.bbc.co.uk/2/hi/europe/4649987.stm <-- what a joke.
06:00.11tsumethe person sueing is obviously an anti-american
06:00.44milkyflavaDarthClue, paste that also or should that be painfully obvious?
06:01.06`SauronAnd what's wrong with being anti-american?
06:01.44Assidshido6??????
06:01.56DarthCluemilkyflava: if you don't see something with that command, there is something really, really wrong.
06:02.18*** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net)
06:02.22DarthCluetsume: no, they are just plain stupid and are trying to make a quick buck.
06:02.42milkyflava:) I am really, really new. I just got my card and am just trying to do my first install.
06:03.20DarthCluemilkyflava: not a problem, what results do you see with that new command?
06:04.39milkyflavaIt looks the same
06:04.58DarthCluemilkyflava: what kind of card?
06:05.16milkyflavaTDM11B the dev card with 1 fxo and 1 fxs
06:05.17clive-ayone else using CVS head as of yesterday?
06:05.48DarthCluemilkyflava: did you setup your zaptel.conf and zapata.conf
06:05.57DarthClueclive: why?
06:06.15milkyflavaDarkClue, yes I have them both set up
06:06.24darylpif anyone cares os x bison must be upgraded before you can build asterisk, the version in darwin ports looks like it's working (it didn't error out on bison)
06:06.33clive-Darth my free memory is plummetting, and looks very worrying to me
06:07.38DarthClueclive: don't know.  mine is from the 23rd.  if no one else responds, you might check back later and see if anyone else is using it or just back up a few days.
06:07.41darylpif I make install will it override my conf files?
06:08.12DarthCluedarylp: no, but make samples will and it is a good idea to back up the conf files and then copy / paste what you need into the new ones.
06:08.27darylpthanks
06:08.39DarthCluemilkyflava: did you modprobe zaptel and wctdm?
06:09.07milkyflavaDarkClue, yes, they both loaded without a problem
06:09.57DarthCluemilkyflava: the only thing i can see is that it looks like it isn't happy that so many conf files are missing.  maybe you need to re-install.
06:10.00darylpdo I need to make zaptel?
06:10.05darylpcause it's erroring like a sob
06:10.30milkyflavaDarkClue, Ok, I don't mind installing 6-7 times I will get better with each one
06:10.50milkyflavabut is there an install using FC* with this card that you could recommend?
06:10.55DarthCluedarylp: zaptel is required for ztdummy.  if you want meetme or anything that requires timing then you don't need zaptel
06:11.36DarthCluemilkyflava: i use fc3 with that card and it works just fine.  did you download from cvs?
06:12.20darylpok
06:12.25darylpI dont' think I have a choice
06:12.26milkyflavaDarkClue, I d/led it tonight from CVS
06:12.33darylpzaptel seems very linux specific
06:12.40milkyflavausing FC1 and the asteriskdocs website
06:12.57darylplibpri won't build either
06:13.59helpmeguysI need help: http://bugs.digium.com/view.php?id=2266    how do I get the 24 bytes key to send reboot to Grandstream phone?
06:14.08loudi had no problems, fc4_64+t410p+2.6.12
06:15.09loudhelpmeguys
06:15.13darylpwell, it built, and installed
06:15.14DarthCluemilkyflava: i use FC3 so some things may be different in FC1.  but i use the docs on voip-info and it works flawlessly.  get rid of the existing files / directories in etc/asterisk, var/lib/asterisk and /usr/src that pertain to asterisk, then go to the site i am about to post the link for and follow the directions for cvs-head and then scroll down to where it says upgrading asterisk and follow the directions for compile and
06:15.22darylpthe sound is improved, but I still have glitches
06:15.25loudcurl -c cookies.txt -d"P2=PASSWORD&Login=Login&gnkey=0b82" http://*.*.*.*/dologin.htm
06:15.29loudcurl -b cookies.txt http://*.*.*.*/rs.htm
06:16.37helpmeguysloud: thanks for the tip, but I want to be able to send SIP NOTIFY message, since the phone might be behind NAT and stuffs
06:16.43DarthCluedarylp: not sure, could be an osx issue.  bkw might be able to help you, but i don't recall if he runs it on osx or not.  you'll have to wait a few hours for the rest of the U.S. to wake up.
06:17.47darylpok, well, I can deal with it for now, it's not like I don't have a hundred other issues to work out
06:18.19milkyflavaDarthClue, It was my undrstanding that using this card I wouldn't need SIP or IAX is that correct?
06:18.34milkyflavaI could just use my POTS line
06:19.37DarthCluemilkyflava: if you want just standard pots communication and don't want any voip then yes.
06:20.38milkyflavaDarthClue, but I could still get the PBX stuff working and use IP phones inside my network?
06:21.04DarthCluemilkyflava: you would then need to use SIP or IAX in your network for the IP Phones.
06:21.34milkyflavaDarthClue, Ok, thanks, I have been going at this for about 8 hours and am getting more confused as I go.
06:21.58*** join/#asterisk tessier_ (~treed@222.253.74.141)
06:22.06DarthCluemilkyflava: just take it slow.  it will all be clear in the end.
06:23.16milkyflavaDarthClue, I hope so, it's hard just finding one good doc that will walk a newb through from beginning to end
06:23.49DarthCluemilkyflava: the wiki is pretty good, if you can navigate it.  otherwise, my only answer is that it is being worked on.
06:24.19DarthClue~cluecon
06:24.19jbotcluecon is probably http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
06:24.19tessier_Where in the realtime config do I tell it what db to connect to? ip, user, etc?
06:24.31milkyflavaDarthClue, The wiki is pretty hard to navigate through, I will keep at it and eventually get it...I hope.
06:24.43tessier_The extconfig.conf just tells it mysql, db name, table name
06:25.29DarthCluetessier: i don't understand your question?
06:25.59*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
06:26.09tessier_DarthClue: I am setting up realtime for asterisk. I want to set up a voicemail system. I want the voicemail configs to be in my mysql db on the same host.
06:26.25blitzrageDarthClue: lol - no more of those :)
06:26.30milkyflavalol
06:26.31tessier_I added this to extconfig.conf: voicemail => mysql,asterisk,voicemail_users
06:26.42blitzrageDarthClue: theres a really cute chick who waitresses at vball. She's hot.
06:26.52tessier_That is all the wiki says I have to do.
06:27.07DarthClueok, and what happens?
06:27.10tessier_I am wondering where I can tell it what my db ip is, what username/pass etc?
06:27.22tessier_I don't see how it can possibly connect with just that line.
06:28.29DarthCluetessier: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime ... there is a section that says 'How to configure Realtime - MySQL Method' ... i think that's what you want.
06:31.32helpmeguysI need help: http://bugs.digium.com/view.php?id=2266    how do I get the 24 bytes key to send NOTIFY reboot to Grandstream phone? I don't want to use curl because the phone might be behind NAT
06:31.37*** join/#asterisk iceyp (~icepick@firewall.unix.co.nz)
06:32.01milkyflavaDarthClue, Can you post that link to the wiki doc?
06:32.16blitzrageDarthClue: don't think I'm going to make it to cluecon
06:32.16iceyphey guys, anyone upgraded a 7940 phone here? I can get P0S3-05-3-00 onto the phone, but it wont accept anything newer
06:32.21DarthCluehmg: are you trying to find out if the bug was applied or what?
06:32.27iceypI'm guessing i need to upgrade the boot loader somehow?
06:33.03DarthCluemilkyflava: which one?
06:33.20iceypthe phone keeps saying "Booting DSP" when i go into menu's etc
06:33.43milkyflavaDarthClue, you said you would post one but you never did. :)
06:34.00DarthCluei did?
06:34.05milkyflavalol, yeah
06:34.29helpmeguysDarthClue: yes, was it applied? If it was not, I can apply it for my case, but I will need the grandstream key anyway :(
06:35.20milkyflavaDarthClue, you said "go to the site I am about to post the link..." but you never did
06:36.08DarthCluedoh...http://www.voip-info.org/wiki-Asterisk+Download
06:36.18DarthCluesorry, it's late.
06:37.01*** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it)
06:38.25milkyflavaDarthClue, Thanks, don't worry, I am the one asking for help. :)
06:39.56DarthCluehmg: i would say it wasn't applied. let me on something though.
06:41.13DarthCluehmg: there is a sip notify command...not sure if it can do what you want or not.
06:41.53iceypcan one upgrade to POS3-07-4-00 when using boot load ID PC030301
06:42.00iceypon a cisco 7940 phone
06:42.23DarthClueiceyp: i don't think any of the cisco guys are here right now.  you might have to wait a few hours for them to wake up.
06:42.31iceypsweet
06:42.33helpmeguysDarthClue, the bug says "You also need grandstreams administration toolkit to produce the small key file that we attach to the NOTIFY" ... that is what I'm trying to find
06:43.34helpmeguysDarthClue, I already tried "sip notify" and it worked for CISCO phones, however Grandstreams returns with 415 response .. I need an additional key for grandstream
06:45.04Pete_Largoquestion: does the 's' extension work for calls that have a 'DID' number?
06:45.24Pete_Largoor rather, is it supposed to?
06:46.14DarthCluePete: i believe it depends on the digits being sent.  usually, it doesn't go to s, but to the matching digits as sent.  eg, if the number is 555-1212 and 1212 gets sent, then 1212 is the extension that gets matched.
06:47.19Pete_Largowell, let's say that I want all incoming calls from a certain context to go to a certain phone, regardless of the DID number.  can I use 's' or do I need to list out each DID (or match it with X's)?
06:49.06DarthCluePete: i use _X. to match all incoming
06:49.19Pete_Largoinstead of s ?
06:49.21Pete_Largowhy is that?
06:49.29Pete_Largojust out of curiosity
06:50.10Pete_LargoI mean, really, what would the difference be between 's' and _X. ?
06:50.17DarthCluePete: i also have s.  not everything goes to s, so _X. picks up what doesn't.  it may not be the right way to do it, this box is just a test.
06:50.28DarthClue_X. will match everything if i remember right.
06:50.33Pete_Largosame here Darth, just testing, and trying to understand...
06:50.49Pete_LargoOK, so if _X. will match EVERYTHING, then why have 's'?
06:51.23Pete_Largook, not true that _X. will match _EVERYTHING_, it requires at least one digit
06:51.29Pete_Largohmm
06:51.56Pete_Largoso is 's' for calls that come in with ZERO digits but won't match calls with ONE or MORE digits?
06:53.33DarthCluePete: the wiki page on s...http://www.voip-info.org/tiki-index.php?page=Asterisk+s+extension
06:54.00Pete_Largoyeah, I've made it to here... http://www.voip-info.org/tiki-index.php?page=Asterisk+standard+extensions
06:54.11Pete_LargoI guess I'm just bored
06:54.56Pete_Largothanks (yet again) DarthClue
06:55.17DarthCluePete: not a problem.
07:01.05iceyphmmm, for some reason my cisco phone is looking for a non existent file off the tftp called P0S3-07-4-00.sbn , one i dont have, anyone got any ideas?
07:02.45*** join/#asterisk wiseguy_ (chivilis@85.206.10.21)
07:03.04wiseguy_help me again, how to get last two numbers of callerid, in extensions.conf?
07:03.05wiseguy_:_)
07:03.17wiseguy_${CALLERID}:1, do not work
07:03.31Assidhow the hell do i stop this lady from saying "welcome"!!!
07:03.31wiseguy_anyone
07:03.39wiseguy_:-))
07:04.09DarthCluewiseguy: what version of *?
07:04.26wiseguy_DarthClue: 1.0.9
07:04.50DarthCluewhat is the exact line you have in your dialplan?
07:05.06litagei added a server to e164.org, and it gave me a # range (88299 001778 00 to 88299 001778 99). does that mean that anyone who calls a # in that range is sent to my server?
07:05.14wiseguy_exten => _86XXXXXXX,1,SetCallerID(${CALLERID}:1);
07:05.14wiseguy_exten => _86XXXXXXX,2,Dial,Zap/g1/${EXTEN}
07:05.42DarthCluelitage: we don't know.  maybe you should ask e164.org
07:06.03DarthCluewiseguy: change it to CALLERIDNUM
07:06.44wiseguy_DarthClue: :1 it means what?
07:08.15Assidhrmm
07:08.24DarthCluewiseguy: go here http://www.voip-info.org/tiki-index.php?page=Asterisk+variables
07:08.28Assidhow do i know whether or not my card supports callerid?
07:08.30DarthClueAssid: what lady?
07:08.40Assid?
07:08.40DarthClueAssid: is it a clone?
07:09.15Assidyes
07:09.29Assid0000:00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
07:09.32DarthCluei don't think the clones support callerid.
07:09.39Assidhow do i find out?
07:09.45twistedtry it?
07:09.46twistedmaybe?
07:10.41Assid<PROTECTED>
07:10.42Assid<PROTECTED>
07:10.42Assid?
07:11.53DarthClueAssid: try CALLERIDNUM
07:12.03DarthClueor CALLERID
07:12.14DarthCluei am pretty sure they are case-sensitive
07:12.27darylpugh
07:12.54darylpon debian (ubuntu-hoary) the build processing is complaining I don't have termcap support
07:13.23nDuffIf I don't have an action defined for when fax detection kicks in, will having faxdetect=incoming have any effect? (I want fax detection to take effect in a particular menu where I have "exten => fax,1" defined, but not alter behaviour elsewhere; is this general behaviour?).
07:13.58DarthCluenDuff: it will look for a fax extension everywhere i think.
07:14.05DarthCluedarylp: i don't know.
07:14.15*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
07:14.30nDuffDarthClue: erm. And if it doesn't find one, does it terminate the call, or pretend it didn't happen?
07:14.40*** join/#asterisk epablo (~epablo@WLL-24-pppoe197.t-net.net.ve)
07:14.57epabloHi people.. How's it going?
07:14.57AssidJul  5 12:44:30 WARNING[3954]: pbx.c:5754 pbx_builtin_setvar_old: SetVar is deprecated, please use Set instead.
07:14.58DarthClueit will terminate the call based on invalid-extension
07:15.34ZeeeknDuff you are testing spandsp?
07:15.34nDuffOh. That's what I was afraid of.
07:15.41DarthClueepablo: run!
07:15.53darylpone needs the libcurses dev files
07:15.58nDuffZeeek: no, I just want to route fax calls to the fax machine even if they're made to the primary number.
07:15.59DarthClueAssid: yes, it is deprecated, but it won't kill you just yet.
07:16.10ZeeeknDuff makes sense
07:16.14DarthCluedarylp: yeah, i think so.
07:16.33epabloI'm using 1.0.7.  Earlier today I got chan_h323 to leave me cdrs on my cdr_odbc.  But it broke with no reason.  Can anyone helpme out?
07:16.44ZeeeknDuff so you detect the fax and send it to an extension?
07:16.55nDuffZeeek: I've played with spandsp a bit, but app_rxfax's library dependencies made Asterisk die an awful, awful death, and I'm not going to go back there until getting the more important (redirect to the *real* fax machine) functionality working first.
07:17.02Assidnope
07:17.04DarthClueepablo: step one, upgrade to at least 1.0.9 or HEAD.
07:17.09Assiddont think callerid works on this clone
07:17.10nDuffZeeek: yup. Only trick is that we have several extensions with fax machines, and I don't want to redirect any of them.
07:17.14*** join/#asterisk Myshenka (~gunde@217.9.101.85)
07:17.17epablosorry chan_oh323.  I hace the amaFlags=billing
07:17.21nDufferm, redirect calls initially intended for any of them.
07:17.28ZeeeknDuff just yesterday another fax machine kept trying and didn't make it through
07:17.28epablook.. I'll try that
07:18.03epabloDarthClue: where there any bugs found in 1.0.7 related to cdrs?
07:18.05DarthClueepablo: that's step one, once you've done that, try again, and then come back if it still doesn't work.
07:18.29nDuffand the other problem is not having a second fax (modem|machine) handy and consequently being unable to test. That's probably a big enough issue that I'm going to put things back and call it a night shortly.
07:18.30DarthClueepablo: i doubt it, but 1.0.9 is the latest stable and most of us prefer HEAD
07:19.09nDuffDarthClue: is HEAD actually sanely usable in a production environment?
07:19.24DarthCluenDuff: yes
07:19.26epabloDarthClue: I used to use HEAD.  But since I'm on a production enviroment..
07:19.50DarthClueHEAD is used in all sorts of production environments
07:20.21epabloDarthClue: We'll I'll give 1.0.9 a try.. be back with you in a while to see how it went
07:21.15ZeeeknDuff you need a test call?
07:22.11nDuffZeeek: I'd need a bunch of them -- I don't anticipate this working out-of-the-box. The offer's appreciated, but I'll just hold off 'till I'm able to handle it on my own.
07:22.13ZeeekCALLING EVERYONE IN PARIS: lunch with Mark Spencer (who?) this Friday; be there or be there
07:22.36ZeeeknDuff ok, I have jfax for now so I could easily help you out at some point
07:23.43Assidhrmm.. this thing keeps saying that welcome
07:23.44Assidand i dont want it
07:23.51*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
07:24.01Zeeekyou don't want to be welcomed?
07:24.06ZeeekGet Out!
07:24.07Assideeks
07:24.08Assidsorry.. mine..
07:24.15Zeeek(that's how it feels not to be welocmed)
07:24.55DarthClueAssid: what thing keeps saying welcome?
07:25.05ZeeekWELCOME
07:26.01DarthClueDarthClue knows that format c: is possible when using /u
07:26.09AssidDarthClue: my bad.. i added a background of welcome
07:26.15Assidto play the "welcome" from the operator
07:26.30Assidand what happens on pstn.. people dont hear that.. so it sounds like CUM
07:26.38Assidwhich is kind of embarassing
07:26.48Zeeekyes, that would be in many cases
07:26.48Assidcoz my aunt just missed the pbx
07:26.59ZeeekAssid add a wait(1)
07:27.11Assidnah.. just knocked it off
07:27.58Assidbut when i dial a known extension.. shouldnt it automatically go to the DialStatus ?
07:28.05Assidi mean if nonswer and busy etc
07:28.08Assidcoz its not doing it
07:28.19Assidor do i have to define it every context?
07:30.45Assidhrmm.. time for another shower
07:30.49Assidtoo damn hot
07:32.38*** join/#asterisk mbranca (~matteo@81.208.92.210)
07:36.03Zeeekactually cooled down a lot here and floods in the north
07:36.12Zeeekit's a nice 18°C now
07:36.25*** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
07:37.23nDuffZeeek: actually, if you wouldn't mind sending a fax to 512 874 7601 and seeing if it gets redirected to a fax machine, I'd rather appreciate that.
07:38.29DarthCluedown to 71F here...too bad the humidity is still sitting at 87%!
07:38.44ZeeeknDuff sure just a sec (need to login and remember password etc)
07:38.45MyshenkaAre there any known issues with the asterisk users mailing list? With addresses from a certain domain I cannot subscribe at all, and being subscribed with a different address my mails dont seem to get published when I write to the list.
07:39.21MyshenkaAnd yes, concerning the first issue I did write to the mailing list owners, but didnt get a reply
07:40.01Pete_Largois anything broken in 1.0.9?
07:41.09DarthCluePete: shouldn't be, why?
07:41.56wiseguy_am, i have 12 e1 channels, howto describe to asterisk, that he should use one of available channels from these 12?
07:41.59wiseguy_:-)
07:42.15DarthCluewiseguy?
07:42.35wiseguy_ammm
07:42.39ZeeeknDuff I'm having a small problem... but it'll happen RSN
07:42.42wiseguy_Dial,Zap/g1
07:42.49wiseguy_it will do it?
07:44.15Pete_Largono reason
07:44.35ZeeeknDuff ok fax sent
07:45.38nDuffZeeek: excellent, it detected the fax and redirected to the appropriate line. Do you have any feedback on whether your side thinks it was correctly sent? (I'm not in-office to grab the physical piece of paper).
07:45.50ZeeekI'll see
07:46.23Zeeekyep they report sucessful transmission
07:46.28nDuffsweet!
07:46.29nDuffmany thanks.
07:46.37Zeeeknp
07:47.13nDuff(there's a nonprofit we're going to be giving some building space and access to our phone system to -- this means I'll be able to put them behind just one extension, rather than needing a second one for their phone line as well).
07:47.36epabloI just upgraded to 1.0.9 and now I'm getting this: usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: ast_smoother_feed
07:47.42epabloAny ideas?
07:48.09nDuffs/for their phone line/for their fax machine/
07:48.12nDuffgah.
07:48.36Myshenkaepablo, before you upgraded, did you delete the contents of the asterisk modules directory?
07:49.16epabloMyshenka: No.  But I built my own rpm and installed it with -Uvh
07:50.42*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
07:51.07epabloMyshenka:  all my modules are from today..
07:51.57Myshenkaepablo: I am a newbie here, just the error message looked familiar (-:
07:52.29*** join/#asterisk jansaell (~jan@192.71.194.38)
07:52.54epabloMyshenka: Ok.. thanks.. all ideas are well recived ;)
07:55.15*** join/#asterisk truescot (~ts@84.119.222.91)
07:56.47truescotcan anyone help me please, i have installed free world dialup and can make outgoing calls and everythign else works fine except i get no incoming calls, it doest even show anything in the asterisk console when an incoming call is started
07:57.10truescoti am guessing i have not forwarded all correct ports to the machine on my router, has anyone got any info?
08:00.19Zeeektruescot with sip debug ?
08:06.09Assidshido6?
08:06.32*** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net)
08:08.42infinity1i have a weird problem. i can dial my sip phone's extension from my zap interface but i can't dial the zap from the sip phone.
08:09.06Assiddid u set up a extension?
08:09.07infinity1is there somewhere else to define an extension for a phone other than extensions.conf
08:09.12Assidto handle through zap
08:09.26Assidshow me your dialplan for calling through zap
08:09.27infinity1exten => 110,1,Macro(stdexten,110,${PHONE3})
08:09.48infinity1i have the sip fone right above it
08:10.58wiseguy_help me
08:11.06wiseguy_with channels
08:11.14wiseguy_i have e1 card with 12 channels
08:11.15Assidi dont see a Dial(ZAP/1...
08:11.36Assidtherefore you arent making any call through zap
08:11.51Assidall your dial() would be SIP/IAX2 if you look at it
08:12.38Assidwiseguy_: sorry.. dunno how.. just started using 1 extension last night on x100p clone
08:13.05infinity1Assid: how weird.
08:13.30infinity1Assid: for some reason you don't have to have sip phone extensions listed in extensions.conf. i didn't have the context included.
08:13.34*** join/#asterisk truescot (~ts@84.119.222.91)
08:13.41infinity1Assid: but the sip phones still worked. weird.
08:13.47Assidinfinity1: right..
08:13.53Assidexten => _1.,1,Dial(Zap/1/${EXTEN})
08:13.56Assidtry that
08:14.01Assidand call any number starting with 1
08:14.10Assidit should dial through your zap interface
08:14.53infinity1yea. that works. for some reason you don't need it for sip. only zap.
08:15.14Assidas i said.. you need to tell zap when to be called
08:15.29infinity1Assid: you don't for sip?
08:15.36Assidalso.. your dial plan must include a ZAP interface calling.. hence Dial(Zap/1..
08:15.52Assidsip as well
08:15.58Assidhence why i said starting with 1
08:16.07infinity1Assid: what i'm saying is you don't need it for sip. it works without it
08:16.09Assidwhat you could do is have * for pstn.. and 1 for sip
08:16.23Assidyes.. cause you are using Dial(SIP
08:16.29Assidnot Dial(Zap
08:16.37Assidit doesnt know where to route the calls
08:16.40infinity1i didn't have dial anything
08:17.27Assidyou have to tell * which interface to use to make the call..
08:17.32Assiddo you want zap/sip protocol
08:18.02infinity1Assid: thats what i though. i removed the lines and it works for sip
08:18.24Assidif you read the stdexten context.. you will notice that all of them dial through sip
08:18.55infinity1Assid: mine doesn't
08:19.05Assideh?
08:19.14Assiddont you have a Dial() ?
08:19.19infinity1no
08:19.22infinity1not for sip.
08:19.29infinity1it must be using the context name in sip.conf
08:19.52Assidprolly.. i have it. but then mine isnt the best way to go
08:20.56infinity1very strange. well, i learned something strange/new.
08:21.30Assidi like having to tell * what to do.. lets me send things down IAX when i want to
08:21.40Assidand Zap and SIP when it suits me
08:22.46Assidshido6 you there?
08:24.32*** join/#asterisk Vlasis (user@194.219.121.194)
08:25.55Vlasishello all, does anyone else have problems with chan_capi 0.3.5 after upgrading to * 1.0.9?
08:26.03*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
08:27.16VlasisI keep on getting :"error sending DATA_B3_REQ"
08:27.44Assiddoes anyone have a nufone account>
08:28.32CpuIDAssid, ya
08:28.37Assidsup CpuID
08:28.38CpuIDaccess to 3 actually
08:28.40CpuIDnada u
08:28.42tzafrirVlasis, what 1.0.9 exactly?
08:28.48Assidgoin on
08:28.50Assidu on dal?
08:28.55tzafrirfrom debian unstable?
08:28.57CpuIDnah not on dal
08:29.01CpuIDhavent been on dal much recently
08:29.05Assidhehe
08:29.10CpuIDcbf connecting to  multiple networks all the time
08:29.34Assidi cant seem to get my nufone connected
08:29.50Assiddidnt know u were into * man
08:30.00CpuIDlol yea
08:30.00*** join/#asterisk Newbie___ (me@211.24.146.11)
08:30.04skrustymorning
08:30.04*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
08:30.09CpuIDive got an * box at work, and one at home and one at my colleagues house
08:30.16Assidaaah
08:30.24Assidhavent seen u on for quite some time
08:30.31CpuID5 cisco 7960s at work, a yuxin YWH100 iax handset at home, and an IAXy for my gf's place shortly
08:30.33CpuIDjust gotta configure it
08:30.42Assidhehe
08:30.46CpuIDheh just cos i aint in a channel dont mean i aint on ;)
08:30.50Vlasistzafrir: I upgraded to Asterisk version 1.0.9 yesterday, and it seems that there are problems with chan_capi
08:30.54CpuIDusually always on FN here
08:30.58*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
08:31.03CpuIDso whats up with your nufone?
08:31.07Assidyeah... ive barely logged in myself
08:31.07CpuIDare you doing 1way or 2way?
08:31.08Assidhrmm
08:31.10Assidcant get it to work
08:31.25Assidkeeps saying no authority found
08:31.49Vlasistzafrir: i used chan_capi 0.3.5 up to yesterday with Asterisk 1.0-RC2 and all worked fine, but after upgrading to Asterisk 1.0.9 the isdn channels give "error sending DATA_B3_REQ"
08:31.49CpuIDok, this for outgoing calls?
08:31.50*** join/#asterisk Jas_Williams (~Jason@host86-130-10-146.range86-130.btcentralplus.com)
08:32.03Assidyep
08:32.14Assidput in $10 for testing..
08:32.50Assidsince last night i cant get it up
08:33.29CpuIDsec gonna pn you
08:33.34CpuIDpm*
08:33.38Jas_WilliamsAssid: bad choice of words
08:33.41skrustyyeah ;)
08:34.31CpuIDlol Jas_Williams
08:34.32CpuID:)
08:34.44CpuIDthat almost deserves to be quoted somewhere ;)
08:35.38skrustylike the topic? ;)
08:35.52CpuIDer...yes sounds good :)
08:37.26CpuIDdamn i love nufone CID
08:37.31CpuIDso fun to prank people :)
08:37.35skrustyhehe
08:37.36truescotbtw i managed to get iax to work on FWD, i just forgot to put a did in :)
08:41.07CpuIDAssid, get my msgs?
08:42.03*** part/#asterisk Vlasis (user@194.219.121.194)
08:43.18*** join/#asterisk loick (~loick@81.255.80.161)
08:48.13*** join/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg)
08:56.29tzafrirCpuID, so now you can blackmail Assid with a bash.org submission?
08:56.44*** part/#asterisk Myshenka (~gunde@217.9.101.85)
08:57.01CpuIDlol :)
09:00.42*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
09:02.03*** join/#asterisk Assid (~assid@203.115.64.61)
09:03.21AssidCpuID: u there?
09:03.25CpuIDya
09:03.27CpuIDget my msgs?
09:03.33Assidnope
09:03.34Assidi got cut
09:03.35CpuIDarr
09:03.40CpuIDdamnit i closed all my shit
09:03.43Assidstupid isp's antennae died
09:12.29*** join/#asterisk jeroko (~root@220.Red-83-38-229.pooles.rima-tde.net)
09:13.58CpuIDwho was it here who runs nufone again?
09:14.03CpuIDi know the guy used to hang around here
09:14.10Assidchido
09:14.34CpuIDah
09:15.03DarthClueJerJer runs nufone i believe
09:15.33CpuIDhmm, yea that nick is more familiar to me
09:15.43Assidhe aint around either
09:16.16DarthClueit's too early in the U.S. for anybody to be around.  I have to get up in less than 2 hours and I have not yet even gone to bed.
09:16.31Assiddont sleep
09:16.36Assidit will give you a headache
09:16.50Assidhrmm i need to start increasing th number of hours i sleep
09:17.02Assid5-6 is starting to make me feel really odd
09:17.33DarthCluei usually get maybe 2 a night during the week then crash on the weekends for 12 to 16 hours at a time.
09:17.44Assidthats the thing
09:17.48Assidi cant get more than 6
09:17.51Assidno matter what
09:17.57Assidsunday.. maybe 8. if im lucky
09:18.09Assidfriday night.. 4
09:18.12Assideven 3
09:18.13DarthCluego a week without and it will be easy to get hours and hours
09:18.21Assidhrmm
09:18.24Assidalready 3 months
09:20.23jerokothis channel is for making questions about asterisk?
09:20.55Assidnah.. its for answering them
09:20.58Assidhehe
09:21.27jerokothen, here is one..
09:21.36DarthCluejeroko: no one is awake right now, you'll have to make your question in 4 or 5 hours when the rest of the world wakes up.
09:22.01AssidWe are unavailable right now, please leave a message after the tone..                         BEEEEEEEEEEP
09:23.33jerokocan asterisk creat CDR's in realtime (with live calls)?
09:23.58jerokofeel free to answer those who are not aslept
09:25.16Jas_Williamsjeroko: Yes when a call finishes cdr is written
09:28.32jerokoyes, i have that already working, but i need a crd at the beginning of the call and then to fill it with complete information after the call has ended
09:28.47jerokojust to keep track of current calls
09:29.05DarthCluejeroko: to keep track of current calls?
09:32.10jerokomy english is a little poor, so i don't know if i am been well understood
09:34.26tzafrirjeroko, asterisk does create CDR in real time by default.
09:34.43tzafrirtail -f /var/log/asterisk/cdr-csv/Master.csv
09:35.07tzafriroh, that kind of real-time...
09:35.11tzafrirno
09:35.24ZeeekI think he means looking at it during the call
09:35.35Zeeekthere is no CDR record until the call completes AFAIK
09:35.37jerokoyes, that is
09:35.57Zeeekjeroko you need to use the manager interface to get the info you want
09:36.20*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
09:36.21Zeeekif it is just to look at, Flash Operator Panel does a good job
09:36.28tzafrirSome applications try to do the math on their own, assuming they know about all the calls the user makes
09:37.29jerokoi've been reading about the manager interface, but i haven't find much documentation
09:39.36jerokothere has to be some already-built aplication for manager interface
09:47.02Jas_Williamsjeroko: what type of application are you trying to write ?
09:47.38jerokowhat i am looking for is a way to obtain running calls information( source, destiny, current time...) from a DB that is wroten everytime asterisk begin or ends a call
09:48.39*** join/#asterisk key2 (~tree@gob75-2-81-56-64-17.fbx.proxad.net)
09:49.44jerokothe idea is to make a web where someone can watch what calls are going on
09:50.23Jas_Williamsjeroko: Then the manager interface is what you require http://www.voip-info.org/wiki-Asterisk+manager+API checkout the perl modules
09:50.26key2jeroko: yeah but u don't need the manager for that
09:50.31key2just connect to asterisk and do a show
09:50.37key2it will give you all the details
09:53.19Jas_Williamsjeroko: you could also do asterisk -rx "show channels"
09:53.32Jas_Williamsfrom your application and parse the output
09:54.52jerokobut there isn't enough information, after that, i should make a "show channel <channel name>" for each one
09:55.15*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
09:55.26shidoZzzZz
09:55.39shidoturned my box off , now it wont turn back on because its too hot
09:56.47jerokodo u know if there is any script doing that?, if not i'm affraid of learning perl from scratch
09:56.49Assidwhich box?
09:57.01Assidshido: i had CpuID test my account.. he has the same issue
09:58.10shidook, its back on, brb
09:58.17CpuIDya
09:58.33CpuIDdang, gf had a little prang in her car
09:58.38CpuIDnothing major, just car damage
09:58.51CpuIDwill soon find out how bad lol
09:59.12Assidoh boy
09:59.24Assidwas it "parked"?
09:59.25*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
09:59.32Assidshido6:!!
09:59.36shido6sup?
09:59.39Assiddude.. its def. the account
09:59.46shido6really?
09:59.46Assidi had CpuID check it as well
09:59.51Assidyep
10:00.00Assidand he already has an account
10:00.01Assidhis works
10:00.02Assidmine doesnt
10:00.13Assidyou dont like me do you?
10:00.29CpuIDyep
10:00.36CpuIDi tested my configs, just changed the user/secret
10:00.39CpuIDno authority found
10:00.47CpuIDswitched straight back to my own, worked perfect
10:01.15*** join/#asterisk The_Duke (~the_duke@80.92.64.103)
10:01.46Assidshido6?
10:02.23shido6I need a moment to check the switch again
10:02.29Assidokie
10:02.33shido6thumbscans at 6 am are confusing
10:02.56shido6make a phone call, Assid
10:03.20Assidnope
10:03.42Assidbtw: there is no facility to change password?
10:05.37*** join/#asterisk ZeeLax (~zeelax@217.22.162.234)
10:05.46*** part/#asterisk jeroko (~root@220.Red-83-38-229.pooles.rima-tde.net)
10:05.59shido6you're right
10:06.05shido6assid takes a dump
10:06.07shido6no authority found
10:06.08shido6odd
10:06.36Assidsee
10:06.41Assidso why dont you delete my account
10:06.44Assidand recreate it
10:06.50Assidwould be easier to debug
10:07.05Assidmake it manually
10:07.06*** join/#asterisk meppl (mephisto@p54AAC178.dip.t-dialin.net)
10:08.17CpuIDmanually...err, lol
10:09.25*** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr)
10:09.25Assidbah
10:09.42CpuIDpfft
10:09.51shido6make a call
10:10.12Assidnope
10:10.14Assiddidnt work
10:10.24Assidi have no authority
10:10.43*** join/#asterisk helpmeguys (~helpmeguy@203.210.213.52)
10:10.46*** join/#asterisk meppl (mephisto@p54AABE80.dip.t-dialin.net)
10:12.21CpuIDstfu lol
10:13.02Assidim planning to add thumb print scanning to my pc..
10:13.07Assidto "protect" my files
10:13.08Assidhehe
10:13.16Assidpasswords dont cut it
10:13.25*** part/#asterisk toresbe (tsb@developer.skolelinux.no)
10:13.56Assidonly certificates and thumb scan
10:14.05Assidmaybe do both together
10:14.19Assidhell.. add a retina scan in there
10:15.40truescotu know the new ibm laptops come with a fingerprint scanner
10:16.01Assidyeah
10:16.02Assidi know
10:16.08Assidi aint buying a new laptop coz of it
10:16.22Assidjust the fingerprint scanner.. for the hell of it
10:16.23Assidhehehe
10:16.25truescotwe have a few at work and a pile of fingerprint scanners for accessing bloomberg
10:16.29Assidi wonder how well it works on linux
10:16.39truescoti put it on for logon as well but i got bored of it
10:16.42Assidfingerprint scanners to access the news?
10:17.09*** part/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg)
10:17.38truescotblomberg is more than news, and costs a fortune to run and bloomberg is damn tight on the licenses and insist everyone logs in with fingerprints
10:17.59Assidoh.. you work @bloomberg?
10:18.03truescotnope
10:18.12truescoti work at a private bank with bloomberg terminals
10:19.06*** join/#asterisk florinm (~florin@212.240.100.208)
10:19.21florinmhi boys and girls ;)
10:19.29florinma small problem with a TE110 :(
10:19.36shido6whats wrong now, florinm ?
10:19.36truescotgo for it
10:19.42florinmi can receive ok calls via all the channes
10:19.48florinmbut i can't dial out :(
10:20.07truescotare u using asterisk or asterisk@home?
10:20.08florinm-- Executing Dial("SIP/200-f089", "ZAP/g0/907951000003") in new stack
10:20.08florinm<PROTECTED>
10:20.12florinmasterisk
10:20.39Assidshido6: any luck?
10:21.00shido6no, I have not received your password change request yet
10:21.11Assidno no..to get it working
10:21.17truescoti had this problem when i had not set the outgoing trunk properly
10:21.21Assidcan change the password later..
10:21.37bublboblHi all ! What is the 1st sound file played when VoiceMailMain() is called ?
10:21.55florinmand how u fixed it ?
10:22.08shido6yes, change the password
10:23.01Assidokay fine
10:23.04Assidemail sent
10:23.12Assidbut i really just care to get it working first..
10:25.25truescoti had to sort out the correct channels in group 0 in zaptel.conf
10:25.54florinmwhat u mean ?
10:26.03florinmi have the TE110 card
10:26.14florinmE1 with 30 voice chanels
10:26.19truescotsorry i ment zapata.conf
10:26.26truescotsame as me
10:28.45florinmcan u show me ur conf ?
10:28.57truescoti am not at work atm so i don't have it
10:29.03florinmk
10:29.10truescotgive me a sec and i will see if i can get it
10:29.18florinmcheers ;)
10:29.25*** join/#asterisk postel (~el@53f944c513317627.session.tor)
10:34.17Jas_Williamsflorinm: post you zapata.conf to pastebin :)
10:34.42florinm2 sec
10:35.30Jas_Williamsflorinm: also you are sending the 9 out over the PSTN change the {EXTEN} in the dial command to to {EXTEN:1}
10:35.31florinmhttp://pastebin.ca/16922
10:35.39*** join/#asterisk jonathh (~asd@host81-136-131-42.in-addr.btopenworld.com)
10:35.40florinmyes is ok :P
10:35.48florinmit need a 9 in front!
10:36.09florinmthe setup is a bit dodgy here
10:36.21bublboblSorry all, i don't have my asterisk nearby, does anyone know the sound file played when you invoke VoiceMailMain() ? :-$
10:36.26jonathhhello, anyone here setup or maintain asteriisk in a production environment? Are there any tried and tested ways to harden the install? read-only partitions UPS etc? etc?
10:36.28florinmisdn <-> asphire<->asterisk
10:36.35shido6"Comedian Mail"
10:36.39shido6"Mailbox?"
10:36.55bublboblshido6>  yes, comdeian mail
10:37.32bublboblwhen you invoke VoiceMail(), I think it is vm-intro, but I forgot for VoiceMailMain()
10:37.41shido6<PROTECTED>
10:37.49bublboblshido6>
10:37.54bublbobltahnk you
10:37.57shido6:)
10:38.01florinmJas_Williams : any idea ?
10:39.33Jas_Williamsflorinm: looks fine try more debug 'debug span 1' in the cli when you make the call attempt
10:40.12florinmk
10:40.25florinmlinux*CLI> debug span 1
10:40.26florinmNo such command 'debug span' (type 'help' for help)
10:40.43Jas_Williamsoh
10:40.55Jas_Williamszap debug span 1
10:41.04florinmlinux*CLI> zap debug span 1
10:41.05florinmNo such command 'zap debug' (type 'help' for help)
10:41.34Pete_Largohelp zap debug
10:41.35Jas_WilliamsI should really remember these things :)
10:41.36florinmdebug channel  Enable debugging on a channel
10:41.36florinm<PROTECTED>
10:41.41Jas_Williamspri debug span 1
10:42.06*** part/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
10:42.10florinmk :P
10:42.27Jas_WilliamsYou will now get a lot of output paste the results to pastebin
10:42.37florinmno extra info!!!!
10:43.44florinmseems that debug didn't done any good? as i didn't got any extra info ??
10:44.04Jas_WilliamsThen the call is not going out PRI
10:44.17florinm<PROTECTED>
10:44.17florinm<PROTECTED>
10:44.34*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
10:44.37Jas_WilliamsThere are no availiable channels in PRI
10:44.47Jas_Williamsshow channels
10:44.58florinmChannel  (Context    Extension    Pri )   State Appl.         Data
10:44.58florinm0 active channel(s)
10:44.58florinm0 active call(s)
10:45.20Jas_Williamszap show channels
10:45.31PoWeRKiLLhello !
10:45.50florinmhttp://pastebin.ca/16924
10:47.49Jas_Williamsflorinm: try selecting an individual channel in your dial eg daial(ZAP/30/${EXTEN})
10:48.10florinmk
10:48.59bublboblbye all
10:49.07*** part/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr)
10:49.42florinm<PROTECTED>
10:49.42florinm<PROTECTED>
10:50.38Jas_Williamsflorinm: you say incomming calls work
10:50.42florinmyes
10:50.56florinmi can dial in no problem
10:51.00*** join/#asterisk darylp (~darylp@63-208-162-60.digitalrealm.net)
10:51.15Jas_WilliamsThis is a TE110P single port card ?
10:51.17*** join/#asterisk moonaddict (~moonaddic@213.129.253.62)
10:51.33moonaddicthey all. any idea where I could by a digium card in austria?
10:51.48florinmyes
10:52.43Jas_Williamsmoonaddict: Australian Technology Partners in Melbourne
10:52.55moonaddictno sorry austria/europe
10:53.01Jas_Williamsopps
10:53.04Delvarlol
10:54.04moonaddict:)
11:00.43darylpany osx experts?
11:03.59*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
11:05.54moonaddicthave I understood this correctly? to connect a bunch of analog extension (i.e. phones, faxes) I need e.g. A DIGIUM TDM400P + FXS modules
11:06.15moonaddictand then another BRI (zap) interface that can connect to my telco's NTBA
11:06.17moonaddictright?
11:06.21Assidhrmm i am using DISA.. however.. when i put off the phone from the remote location.. it didnt hangup
11:06.28moonaddictmy mouse is hovering above the "buy now" button :)
11:06.41Assidthe call was yet on between the asterisk box .. and sip provider
11:06.47Assidanyone know how to stop this?
11:14.01*** join/#asterisk kakazz (~Krab@203.81.232.176)
11:14.31*** join/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc)
11:14.57kakazzhey
11:15.49kakazzneed little help on PBX
11:16.44kakazzhello
11:16.47kakazzany one out there
11:17.25Assidjust ask your question
11:17.31Assidif anyone can help you they will
11:17.55*** join/#asterisk The_Duke2 (~the_duke@80.92.64.103)
11:18.03The_Duke2hello
11:18.07Assidhrmm.. how much bandwith does GSM take?
11:18.09The_Duke2can messages like     -- parse_srv: SRV mapped to host sipgate.de, port 5060
11:18.14The_Duke2sorry...
11:18.26Assidi mean the bitrate is 13.2 .. what about the bandwith?
11:18.46The_Duke2are messages like "Got SIP Response 404 Not Found...." stored in a variable???
11:19.02AssidThe_Duke: not found extension?
11:19.05PatrickDKbandwidth for gsm is like 27kbps
11:19.10The_Duke2the DIALSTATUS variable does have NOT FOUND?
11:19.27The_Duke2Assid: not found extension, never heard of that.....
11:19.48Assidwell
11:20.00Assidthe number/extension your trying to reach is not there?
11:20.47Assidkakazz: did you try to message me? it just auto banned you
11:21.06Assiderr.. ignore
11:21.23kakazzwt eve
11:21.32kakazz<PROTECTED>
11:21.44kakazzfor 30 people
11:24.14The_Duke2Assid: OK what I need to do is tell my * to something special when i get's a message 404 not found back via SIP...
11:24.22kakazzany know what hardware card should i use for interconnecting 30 people
11:24.38AssidThe_Duke2: there is a good chance its not even reaching your provider
11:24.40Assidso..
11:24.42The_Duke2Assid: this could be a voice message, the number you dialled has not been allocated... etc etc etc....
11:24.43Assidcheck for that first
11:25.08AssidThe_Duke2: dont know how to do that.. google?
11:25.20Assidkakazz: ip phones?
11:25.34kakazzsoftphone
11:25.38Assidwell
11:25.43Assidthen you dont need a hardware card
11:25.48The_Duke2I tried. but everybody talks about ${DIALSTATUS} which cannot have a value of NOTFOUND (or anything similar)....
11:25.59kakazzand if i be using a IP phones
11:26.04Assidhardware cards are when you want hardware phones --analog/ip
11:26.17Assidhrmm
11:26.19Assidyeah
11:26.26Assidi guess dialstatus may do the trick
11:26.50kakazzif i m using Outbound calls on Soft phone ,, and internally i m using Analog phone wht will be your advice
11:27.13*** part/#asterisk moonaddict (~moonaddic@213.129.253.62)
11:27.25AssidFXO cards
11:27.41kakazzthanks assid
11:27.42kswailmoonaddict - yes you need FXS modules for your internal phones
11:27.42Assiddepending on the number of extensions
11:27.59Assidactually
11:28.02AssidFXS cards
11:28.04kakazzhow much i can expand in that matter
11:28.07Assidnot FXO
11:28.11Assidwell
11:28.18Assidyou could buy a T1 card
11:28.27Assidthat comes iwth 24/32 lines
11:28.35kakazzthanks
11:29.03kakazzdo you ref this Digital CTI Cards : Digium TE110P
11:29.16Assidnever used it.. cant recommend it
11:29.21Assidbut digium makes good hardware
11:29.23kakazzok
11:29.33Assidspecially since its asterisk compliant
11:29.43kakazzi m just new to it ..
11:29.55Assidhey till last night.. i didnt do a FXO interface
11:30.08Assidtoday i am already rerouting PSTN->iax calls
11:30.21The_Duke2assid: I would really use DIALSTATUS, but it returns BUSY for calls which return a 404 Not Found.... which is incorrect...
11:30.22Assidthe guys in this channel ROCK
11:30.38Assidyou may wanna look for NOTFOUND
11:30.53Assidinstead of busy
11:32.25The_Duke2Assid: I would like to, but DIALSTATUS can only have the following states:
11:32.33The_Duke2CHANUNAVAIL | CONGESTION | BUSY | NOANSWER | ANSWER | CANCEL | HANGUP
11:45.12ZeeekThe_Duke_of_Oil?
11:49.24*** join/#asterisk Thumann (~Thumann@217.157.30.66)
11:52.32*** join/#asterisk Bhaal (bhaal@bhaal.staff.freenode)
11:52.53BhaalHey guys, just got a quick question..
11:53.17Zeeekgo go go
11:53.25*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
11:53.48BhaalIm using asterisk to allow multiple sip fones use 1 account with a sip provider...  Just wondering if the callerID number will translate through asterisk to the handsets?
11:54.16Zeeekshould
11:54.41BhaalFor an incoming call I mean, forgot to clarify that...
11:54.57ZeeekI think it shoud work
11:55.29BhaalCool, not able to test yet, provider isnt able to supply me with a pstn number as they dont have any allocated for my area...
11:56.00Thumannhey peeps! ;>
11:56.36ZeeekBhaal interestingly FWD allows multiple clients on th same account
11:57.03djinDid anyone have any luck setting up modem dial-up (Banking) throught SIP?
11:57.25BhaalIm in australia .. Im not sure if the provider Im using allows the same thing, although I dont really want it setup that way...  If the line is being used, its being used...  Like a normal fone...
11:57.46djinTrying Draytek Router icw. G.711a and no luck so far.
11:58.22Zeeekoh it still would be busy if busy. Anyway the multiple clients would all ring at once where with asterisk you could ring different ones at different time of day, etc
11:59.10BhaalZeeek: Yeah...  I actually wanna setup a 'who do you want to call' type stuff as there are 4 housemates living here...
11:59.47Zeeek"to speak to the pimply teenager on the second floor, please press 1"
11:59.56Thumannlol
12:00.05Zeeek"for information on coke pricing..."
12:00.26Thumann"if you are an ignorant phonesalesman who deserves to die press 2"
12:00.42Zeeek"to hear the most recent insult about the person calling from: three three one four two three one... please..."
12:00.59*** join/#asterisk Samoied (~Samoied@popeye.opens.com.br)
12:01.22Zeeek"please do not touch any buttons on your phone while we adjust the voltage"
12:01.35Zeeek"ok, now, do you feel lucky today?"
12:02.05Wonka"please do not touch that button again"
12:02.23Thumannhah!
12:02.32Zeeek"I SAID PRESS THREE! "
12:02.35Thumann:) i nned that..
12:02.47Zeeek"oh oh oh!"
12:02.53Thumannbut i dont think 48v is enough to actually boil someone.. :-/
12:03.12Zeeekbut maybe a loud tone could be a powerful message
12:03.17robl^48V DC can kill
12:03.30Thumanntrue.. with enough amps...
12:03.43ZeeekHow about a general filter: "please type 123 to continue"
12:04.03Zeeekthat kills any auto call machines
12:04.11robl^there are many cases where people have used a phone while in a bathrtub and then droped it in the water.. it was enuff to stop their heart
12:04.25Zeeekthat was ground differential thoug
12:04.36Thumannwhat about a dice roll kinda thing.. # to start stop roll.. 1-9 is hangup.. 0 is try again.. :D
12:05.16*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
12:06.03Zeeekstill a few seats left for lunch with kram this friday - c'mon, we'll pay for your ticket to Paris
12:06.14robl^Zeeek, those "feneral filters" also filter out mst granmothers.  Mine can barely dial a phone number correctly.. let alone use an IVR.
12:06.26Zeeeklunch will cost just $2,299
12:06.42*** join/#asterisk mountie (~mountie@trb229.travel-net.com)
12:06.51Zeeekyeah I know, IVR suck anyway, but they're fun to play with for a while
12:07.06ZeeekI hate getting them as a caller though
12:12.06ManxPowera ring is about 90V in the USA.
12:12.30ZeeekManxPower you are soooobehind
12:15.29*** join/#asterisk virterm (~virterm@207.107.229.2)
12:17.32ManxPowerZeeek: I'm still 1/2 asleep.
12:17.41ManxPowerAnyway, I'm about 7 hours behind 8-)
12:17.41Zeeekget some cafeine in you man!
12:18.07ManxPowerworking on it.
12:18.12ZeeekI'm actually sleepy after being online for 7 hours
12:20.23cochiI'm actually sleepy after being asleep for 7 hours ;)
12:21.26Assidis there a way to increase the number of rings before it picks up pots?
12:22.03ZeeekAssid who?
12:26.01Assidpstn
12:27.49ManxPoweri use CLID matching to automatically forward calls from my grandmother to my phone.
12:31.00*** join/#asterisk salvini_fs (~felipesal@200164050004.user.veloxzone.com.br)
12:31.27*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
12:32.59*** join/#asterisk jhava (~icechat5@200.58.26.21)
12:33.18Jas_WilliamsAssid: yes add a wait before an onward dial in extensions.conf
12:33.35Assidyeah.. just added a wait(5)
12:37.18*** join/#asterisk durex (~ironman@weber.anpa.org.br)
12:45.07*** join/#asterisk DarthClue (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net)
12:47.29*** join/#asterisk evangelion (~manzy_zet@ip-185-118.sn1.eutelia.it)
12:47.43evangelionhi all
12:52.15*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
12:54.19*** join/#asterisk lodeon (~not4u@as1-6-4.ld.bonet.se)
12:54.38lodeoncorydon: there?
12:55.01*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
12:59.41*** join/#asterisk rv_weasel (~no@adsl-68-91-83-3.dsl.ksc2mo.swbell.net)
13:01.47rv_weaseli had everything working.  i needed to change some network configuration, so i reboot.  now it wont come back up.  look at my pastebin http://pastebin.com/307775
13:02.52*** join/#asterisk DarthClue (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net)
13:04.19rv_weaselwait,  i have to run ztcfg at startup?
13:04.53jontowyes
13:05.04*** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk)
13:06.30rv_weaseli think i found the problem.  i needed to install wcfxo and run ztcfg a second time in my modprobe.conf
13:06.54rv_weaseli guess i shoule reboot to be sure huh?
13:07.00ManxPowerif /etc/modules.conf is set up correctly, then ztcfg will be called after every driver load.
13:07.26Assidhrmm... how do you get the logging and ocnfiguration of * on pgsql?
13:07.31ManxPowerThat's done by the post-install line in modules.conf.
13:07.46evangelionhave you ever run ser on a mosix cluster?
13:07.53ManxPowerIf you want zaptel to start automagically, do a make config in the zaptel dir, that will install the init script.
13:09.09tzafriron debian: simply add it to /etc/modules
13:09.10rv_weaselrun twice in modprobe.preload:  install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
13:09.12rv_weaselright?
13:09.26rv_weaseltwo cards
13:09.37ManxPowerrv_weasel: I have 8 asterisk servers and never needed to do that.
13:10.10tzafrirThere is basically no reason it should be run in the post-install there if you can guarantee it would be run later (e.g: init.d script)
13:10.21ManxPowerI just do a "chkconfig zaptel on".
13:10.36rv_weaselok
13:10.40rv_weaseli follow ya
13:10.59ManxPowertzafrir: not everyone uses the init scripts.
13:11.30tzafrirManxPower, I said that an init.d is *one way* to verify that.
13:11.42tzafrirrunning the ztcfg there automatically is configusing
13:11.50ManxPower*nod*
13:12.09tzafrirconfigusing is a confusing way to write confusing
13:12.37*** part/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM0012c999e6a0.cpe.net.cable.rogers.com)
13:12.51darylphas anyone used galaxyvoice?
13:13.10Jas_Williams2
13:13.26*** join/#asterisk mithro (~tim@212.129.242.85)
13:13.36MRH2anyone know why I don't get echo on a PSTN->voip provider->LAN  but do get an echo over the LAN? is it likely echo cancellation with the provider?
13:14.11tzangerMRH2: you don't get it over the pstn but you do over the lAN... how in the hell would the provider be responsible for that?
13:14.29rv_weaselok,   all set rebooting to be sure
13:15.01MRH2wondering why I don't get it with the provider (although this is a good thing)
13:16.03tzangerMRH2: is your earpiece or mic volume cranked?
13:16.12Jas_WilliamsMRH2: what are you using as phones to get echo on the LAN ?
13:17.04MRH2yep cranked to the max -  this is why i get the echo on the LAN (Poly IP500)
13:17.51MRH2but how is the voip provider getting rid of this echo?
13:17.59MRH2do u think?
13:18.38crash3mthe echo cancellation patch?
13:19.35MRH2do all PSTN gateways normally use some sort of echo cancellation
13:20.49ManxPowerIf they do PSTN/VoIP, yes.
13:21.02darylpnobody is using galaxy?
13:21.39MRH2cool that explains it
13:23.09*** join/#asterisk \usr\sbin (~squirrelv@202.57.81.146)
13:24.47*** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it)
13:26.14*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
13:26.49*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
13:26.56*** join/#asterisk zoa (~zoa@pirus.securax.be)
13:27.50*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
13:28.40*** join/#asterisk Katty (~angela@68.112.15.110)
13:28.43*** join/#asterisk indra (~indra_wat@microinfo.rain.fr)
13:28.59indraanyone up?
13:29.10darylpI am
13:29.35indragood, u might be my saviour :)
13:29.51darylpnot likely, I'm here cause I'm stuck
13:30.01indraah :(
13:30.24darylp*sigh* and it's not like EVERY day I get a chance to be a savior
13:30.39*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
13:30.42indrau might still try to give me a clue no?
13:30.58darylpI'm really new, but others might answer ifyou ask your question
13:31.22darylpsavior=saviour
13:31.37indrathanks daryl :)
13:31.37SplasPoodholy crap.. first time ENUM has actually worked in a real world scenario for me!
13:32.06indraguess living in France these past 5 years have completely dulled my english :(
13:32.29darylpno, you were right, I was correcting my mistake
13:32.30darylphaha
13:32.40indraI have never done any telephony stuffs in my life (am in critical software stuffs)
13:33.10indrabouhahahahah :D well GLAD to know my english is not THAT bad yet :D
13:33.45indraso, i am doing my summer internship and my boss has decided to put me on a cheap project using Asterisk
13:34.23darylpsounds like a match made in heaven
13:34.30darylpor perhaps some other place
13:35.03indrawell i won't bother everyone here with the story of my life, just want to know if there is any way to treat call waiting with Asterisk
13:35.27indrathe features mention that it is possible, but i haven't found any line in the doc about it
13:35.35ManxPowerindra: What SPECIFIC problem are you having with Call Waiting?
13:35.45indrawell, how to configure the call waiting?
13:36.01ManxPowerWith Zap, SIP, MGCP, SCCP, or H323?
13:36.04indrai mean to swith back and forth between several calls on a terminal
13:36.06indraSIP
13:36.16ManxPowerindra: that is TOTALLY controlled by your SIP device.
13:36.29ManxPowerMost things in SIP are totally controlled by your SIP device.
13:36.40indrawell, that is bad news for me  but thanks manxpower
13:37.27ManxPowerindra: Asterisk will always send a second (or more) call to a SIP device.  What happens at that point is up to the SIP device.
13:37.50indrabut i thought asterisk does the job of the call queueing?
13:38.06ManxPowerindra: "queueing" is the wrong term.
13:38.40ManxPowerAsterisk will handle call waiting internally on ZAP devices, as well as (I think) MGCP and SCCP devices.
13:38.40indraah?
13:39.17ManxPowerBut with SIP, asterisk will just send the call.  The device will usually send back a BUSY HERE message to Asterisk if the device can't handle (or is configured to not handle) another call.
13:39.39ManxPowerSIP expects most of the smarts to be in the phone, not in the PBX.
13:39.52indraManxPower : okay, i got the image
13:40.31indraManxPower : any clue for the internal document for the SIP device - Asterisk communication?
13:42.30ManxPowerindra: Huh?  The SIP RFC, but that's not going to help you.  You need to look at the docs for the SIP device you are using.
13:42.35*** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it)
13:43.23ManxPowerWhat SIP device ARE you using, indra?
13:43.46indraManxPower : great, just what I need. This company of mine has made their own modem and I am supposed to be making an application bridging their modem and asterisk application
13:45.22ManxPowerA SIP device is NOT a modem.
13:45.43mishehuperhaps he means ATA ?
13:46.08indramishehu : they call this modem as RETA but it is something they build on their own
13:46.24indramishehu : a specific solution they offer to particular clients
13:46.52*** join/#asterisk mutilator (~animenodv@65.111.201.79)
13:46.55mishehuindra: stop talking in marketspeak, and explain what *exactly* you mean.
13:47.23indramishehu : i have an asterisk application in one end
13:47.32mishehuyou sound like you're reading an advertising brochure.
13:48.23indramishehu : worse than a brochure, i only have a vague description!! :(
13:48.35darylpmy problem is easier, I'm sure of it
13:48.44mishehuthen you can't do anything with a vague description, can you?
13:48.52*** join/#asterisk rittwage[work] (~peter@intsys.net)
13:49.12darylpwhat does RETA mean?
13:49.20ManxPowerindra: you can't do your job if you don't have the technical info.
13:49.38rittwage[work]Hi guys-  I know this has been asked 1,000 times, but is there an easy to use softphone-based operator console?  FOP does funny things for me and isn't a one-click solution.
13:49.40cypromisreverse engineer it
13:49.41cypromishehe
13:49.44mishehudarylp: my guess is that RETA is market speak
13:49.51indramishehu : well i need to gather whatever information i can get on asterisk to make this bridge (an application) between asterisk and the modem
13:50.20mishehuindra: you need to know what the hell it is you are trying to get to work with asterisk before you start messing with asterisk.
13:50.21indradarylp : mishehu was right, this RETA stuff is just a name invented by the sales dept
13:50.26darylpwhat exactly must the bridge do, surely you have that information
13:50.39*** join/#asterisk mogorman (~mogorman@207.111.174.1)
13:50.44mishehuit's like trying to find a cure when you don't even know the disease.
13:51.01darylpthe modem answers a line and then....
13:51.11*** join/#asterisk pif (ldm@zenon.apartia.fr)
13:51.12darylpa fax comes in and....
13:51.19*** join/#asterisk drumkilla (~russell@drumkilla.developer.and.stable.maintainer.asterisk)
13:51.20*** mode/#asterisk [+o drumkilla] by ChanServ
13:51.22ManxPowerA modem takes serial data and converts it to a modulated audio tone.
13:51.28darylpwell yes
13:51.33ManxPowerThat has nothing to do with VoIP or SIP.
13:51.39indradarylp : simply for calls, no fax included
13:52.17darylpok, so the modem is a voice/data product?
13:52.54*** join/#asterisk fantomax1 (~fanto@81.208.114.250)
13:52.56indrayes
13:52.57fantomax1hi all
13:53.09darylpI guess modem is the wrong term when you aren't really doing any modulating or demodulating
13:53.13darylpnevertheless
13:53.20fantomax1can anyone suggest me a tool to control the performances of an * server ?
13:53.29darylpmoney
13:53.39fantomax1in term of resources .. etc
13:53.45SplasPoodmore money
13:53.57ManxPowerfantomax1: It's called "ulimit".
13:53.59SplasPoodand a whip
13:54.02fantomax1i know
13:54.10fantomax1ulimit is a command
13:54.16ManxPowerAnd?
13:54.25fantomax1can i use it as testing?
13:54.31ManxPowerIt's not like you can control Asterisk's resource usage in any other way.
13:54.34darylpsince there's so much explaining going on right now perhaps someone would like to help with my simple problem
13:55.05Hmmhesaysheh... always love questions that start off like that
13:55.07florinmany unlimited VOIP provider inclusive mobile (uk) out there so i can use with asterisk ??
13:55.29*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
13:55.34darylpxlite->galaxyvoice, behind nat, works fine, can make and receive calls
13:55.36Nuggetunlimited voip == punch the monkey for the free ipod
13:55.47florinm???
13:55.52rittwage[work]nobody is ever asked for an operator function that FOP doesn't handle or did I just miss a particular phone model or softphone that does it...
13:55.54indramanxpower: if u could give me a hint as where to look for the documentation explaining the interaction between asterisk and SIP devices ?
13:55.57rittwage[work]?
13:56.03darylpasterisk-> galaxy voice, behind same nat, sends request but never gets a reply, can't receive calls
13:57.59Hmmhesaysprobably because your invite contains the internal address of the asterisk box, if I had to guess
13:58.32darylpso how do I fix that?
13:58.49skefflingIs there a script around that imports a .csv file in to the cidname database? (no point re-inventing the wheel!)
13:58.56florinmexternalip=xxx.xxx.xxx.xx something like so
13:59.02darylpboth xlite and asterisk are on the same internal network
13:59.07florinmin sip.conf
13:59.07darylpok, I will try taht
13:59.10*** join/#asterisk nylon1234 (~tonton123@203.131.137.76)
13:59.54florinmand in the galaxy do u have something like "Use NAT IP    " ?
14:00.03Hmmhesayshttp://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+externip
14:00.12florinmi had same problem with a grandstream 486
14:00.16nylon1234hi guys......anyone had used vicidial here....i have a hardtime making the auto dial work...any help pls?
14:00.18darylpnat=yes
14:00.35Kattyalive, but terribly sleepy
14:00.47florinmand i had to put there the enternal ip of the internet where the grandsteam was on
14:01.08florinm:D
14:02.25Hmmhesaysit has indeed been a party weekend
14:02.28KattyDarthClue: i have an infection in the uretha
14:02.42KattyDarthClue: on antibodics....but still alive ;)
14:02.43Hmmhesayspainful
14:03.00ManxPowerI wonder who my IRC client automagically blocked when they tried to /msg me.
14:03.07KattyHmmhesays: quite.
14:03.08darylphmm, it has no effect
14:03.16darylpaffect?
14:03.20ManxPowerI have it set up that way because so many assholes try to get private support for free.
14:03.27Hmmhesayshmmm, try setting externip
14:03.34Hmmhesaysinstead of nat=yes
14:03.37florinmand ??
14:03.40Hmmhesays:D
14:03.47Kattyyes. i'm SICK
14:03.49florinmno, need to have nat=yes too ;)
14:03.49Kattysniffle.
14:04.15mutilatoro_O
14:04.15Hmmhesayslast time I tried that I got a pen in the gut
14:04.27Hmmhesays;(
14:04.35tzangerHmmhesays: you got a pen in the gut for what
14:04.35Kattyha.
14:04.59Hmmhesayshugging Katty, lol
14:05.03tzangerhahaha
14:05.06Hmmhesayswatch out for the sunburn
14:05.56Kattyyou goofball
14:05.57`SauronI think I got smacked last time I tried to do that to Katty
14:06.02Kattydidn't you wear sunblock?
14:06.11tzangersunblock's for the weak
14:06.12`SauronTrue to her name, she is. :p
14:06.26mishehu`Sauron: like yoda, speak you.
14:06.29Kattytzanger: and for those who don't want to burn
14:06.35Hmmhesayssunblock? not soon enough... 5 hours on the river is a long time
14:06.38tzangermy goal is to be mistaken for a puerto rican (spelled wrong I'm sure) by the end of hte summer
14:06.46tzangerHmmhesays: I did that may 24 weekend
14:06.50tzangerlonger actually I think
14:06.51KattyHmmhesays: :<<<
14:06.53tzangerthe big float :-)
14:07.14Hmmhesayswe have a river near here, it's just a giant party, you pay 5 bucks and float down it
14:09.39indramanxpower: got to go, but thanks for the info on SIP devices
14:09.39Hmmhesaysi'm suprised people don't drown actually
14:10.34`Saurondrunk people float by default.
14:10.45Hmmhesaysbut do they float right side up
14:10.57indrabye all, have a nice swim for those who decide to jump into the river :)
14:11.13Hmmhesays<chuckle>
14:12.15tzangerHmmhesays: sounds like what we did
14:12.39*** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr)
14:12.52jhiverAnybody from NuFone about?
14:12.58tzangerwe went up to paisley, ontario (about 1.5hrs from where I live) and partied on a private campground, then the owners of the campground threw us in a truck, drove us 10mi upriver and dropped us in a bunch of canoes
14:13.35*** join/#asterisk daryl_just_daryl (~daryl_jus@63-208-162-62.digitalrealm.net)
14:14.08Hmmhesaystzanger: sounds like a blast,  we didn't have canoes though, just tubes
14:14.21tzangerHmmhesays: tubes?  good lord man where do you keep the booze?
14:14.52Hmmhesaystzanger: in coolers tied to the inside of the tubes
14:15.02tzangerHmmhesays: sounds like a lot of work :-)
14:15.36Damin_Floating coolers, with really good latches! :)
14:15.43Hmmhesaystakes like five minutes to tie up a cooler tube, it's one larger tube with a smaller tube tied in the middle, then you set the cooler in there and strap it down
14:15.50Damin_Specially designed for high performance drunken tubing!
14:17.01Hmmhesaysone group of people had one of those huge 70 somethine quart coolers, in a big huge tractor tube
14:17.11tzangerHmmhesays: sounds like an easy way to crush your nuts if you hit a rapid or a shallow spot
14:17.21tzangerstill sounds like fun though
14:17.39Hmmhesayshaha, no.. you aren't in the same tube.. you tie it off to the tube you ride in
14:18.03Damin_tzanger: Tubing is a blast.. You hang your ass in the tube, sit back and go for a ride.. When you get hot, you just jump into the water..
14:18.25Hmmhesaysor fall off trying to surf the cooler
14:18.25Damin_tzanger: You don't have to even let your ass touch the water if you don't want..
14:18.46Damin_That too...
14:18.55tzangerDamin_: hehe yes I've been tubing many times
14:19.15Damin_I used to be a Summer Camp Counselor in my teen years, and one of the activities was tubing.. I'd do it twice a week for like 10 weeks in a row.. :)
14:19.21*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
14:19.23SpaceBassmorning
14:19.35Damin_tzanger: I got to be an expert at it. ;)
14:19.45Hmmhesayspassing out beer to the teens?
14:19.46Hmmhesayslol
14:19.52HmmhesaysMorning SpaceBass
14:19.54tzangerDamin_: :-)
14:19.56Damin_tzanger: But the best trips where when you headed about 30 miles up river and made a day out of it. :)
14:20.19tzangerDamin_: yup, I think we only went about 10 miles upriver to canoe but nobody was paddling so it took all day :-)
14:20.36Hmmhesayssome guys had a party barge this time around... a bunch of tubes with a piece of plywood and a picnic table on top.. with tunes
14:20.42tzangerhttp://www.mixdown.ca/~andrew/photos/PaisleyMay24-2005/
14:20.43Damin_tzanger: And when they were paddling, they were pointed the wrong way..
14:20.51tzangerDamin_: that too
14:21.03tzangerthe canoe trip starts about 1/2way down there
14:21.19Hmmhesayswhich one is you?
14:21.30tzangerHmmhesays: hahaha  I ain't telling
14:21.52Hmmhesaysif I had to guess
14:21.54Hmmhesayshttp://www.mixdown.ca/~andrew/photos/PaisleyMay24-2005/tn/img_3040.jpg.html
14:22.05tzangerhaha
14:22.08tzangerno that is not me
14:22.20tzangerthere are no pics of me in teh canoe
14:22.29tzangerI wouldn't give the camera to anyone else on the water for fear of them dropping it in the river
14:22.52Hmmhesaysthere's one guy in there that looks like this guy named 'goose' i know
14:22.59tzangerwho's that
14:23.04DaminYou guys didn't go canoeing.. You went drinking, and brought canoes along to sit in...
14:23.13tzangerDamin: well yes
14:23.17Hmmhesayshttp://www.mixdown.ca/~andrew/photos/PaisleyMay24-2005/tn/img_3048.jpg.html on the right
14:23.18daryl_just_darylgreat I've left, I can get my nick back
14:23.28clive-any jitter buffer experts about
14:23.30tzangerhaha yeah that's mike
14:23.33tzangerclive-: what's the question
14:23.34Hmmhesaysdarylp: ghost
14:23.34DaminLook at image 3040..
14:23.43darylpconnection issues
14:23.45*** join/#asterisk thal (~thalunil@walledcity.de)
14:23.49thalsalut
14:23.52darylpseems to be my story today
14:23.54DaminGot his feet kicked up, beer about to be chugged..
14:23.59Hmmhesaysdarylp you can ghost your nick any time you want
14:24.02tzangerDamin: yup
14:24.08DaminAlright.. I need to go to work..
14:24.11tzangerwe did some pretty intense paddling a couple of times... but that's about it
14:24.17clive-tzanger for some strange reason the RTT value goes very high very quick, causing teh jitter buffer to go wacky on me
14:24.25darylpI don't know how to do that
14:24.30tzangerclive-: what version
14:24.37clive-cvs-head
14:24.57tzangerclive-: cvs head shouldn't have that problem it should chop off the top 5% of jitter values
14:25.00ManxPowerclive-: What DATE of CVS-HEAD?
14:25.10thali have problems getting chan-capi-0.35 getting compiled.
14:25.11tzangerManxPower: it should be fine from like april even I think
14:25.16clive-yesterdays date
14:25.19thalwhat can /usr/lib/gcc-lib/i486-linux/3.3.5/include/stddef.h:213: error: syntax error befo
14:25.19thalre "typedef"
14:25.19thal<PROTECTED>
14:25.35ManxPowerclive-: both sides are the same CVS-HEAD date?
14:25.41clive-tzanger the wierd ting is that my RTT is never over 1000
14:26.09tzangerclive-: define "goes wacky"
14:26.14clive-Manx the other side.....I really dunno which version they are running...is theer a big version skew issue?
14:26.24tzangerclive-: who are you terminating to
14:27.07clive-tzanger, well the RTT jumps from the normal 350ms value up to 1200ms..stays there for a bit and eventualy goes back down, in the meanwhle the conversation is basicallly a mess
14:27.10clive-magrathea
14:27.19tzangerCVS HEAD before about may (I think) had an issue where the jitter buffer'd take a big shit if you sent DTMF
14:27.30tzangerit's actually a zaptel issue and hte far end needs to fix it
14:27.41jhiver~seen shido6
14:27.41jbotshido6 is currently on #asterisk (4h 28m 16s).  Has said a total of 17 messages.  Is idling for 3h 49m 44s
14:27.56clive-zaptel?....
14:27.59HmmhesaysI need to read a perl book
14:28.28tzangerclive-: I'm not sure what the new command is but you might want to try jb debug and see what it looks like when that happens
14:28.33*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:28.33*** mode/#asterisk [+o anthm] by ChanServ
14:28.36tzangeryou shoudl normally see vvvvvvvvvvvvvvvvvvvvvvvvvv
14:28.41clive-tzanger thanks ill try that
14:30.04tzangerif the far end stops sending packets you'll see LLLLLL (loss) which would happen with a jump in lag
14:30.19*** join/#asterisk mbranca (~matteo@81.208.92.210)
14:30.29tzangeryou'll see GGGG too which is "grow" (jitter buffer growth) -- followed by some 'l' (late)
14:30.31*** join/#asterisk wek (leroy@yoyoyo.net)
14:30.32tzangerI have to go over this again
14:31.05clive-thousands of these vclamping target from 206 to 180
14:31.06clive-vclamping target from 206 to 180
14:31.11clive-vclamping target from 184 to 180
14:31.11clive-vclamping target from 181 to 180
14:31.15clive-vclamping target from 181 to 180
14:31.17clive-oops,,,sorry
14:31.23tzangerclive-: that's a very very small jitter buffer
14:31.27tzangerwhat are your jitter buffer settings?
14:31.34*** join/#asterisk jeffik (~Jeff@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com)
14:31.36faaanthm Hi
14:31.52clive-tzanger 180 is small?
14:31.57tzanger180 is tiny
14:32.06tzangeryou're clamping to 180 which is useless
14:32.09tzangermay as well not have one at all
14:32.11clive-tzanger what do you recommend
14:32.12clive-?
14:32.16tzangerclive-: what are your settings?
14:32.27clive-180 ms
14:32.32clive-for jitter buffering
14:32.34tzangerclive-: WHAT ARE YOUR SETTINGS
14:32.42tzangeras in pastebin.ca your jitter buffer settings
14:32.47tzanger180 tells me nothing
14:33.08clive-in iax.conf?
14:33.18tzangeryes of course
14:33.50clive-jitterbuffer=yes
14:33.51clive-qos=lowdelay
14:33.52clive-;dropcount=1
14:33.53clive-maxjitterbuffer=180
14:33.54clive-;maxexcessbuffer=250
14:33.59tzangererase that maxjitterbuffer line
14:34.01tzangerlet it default
14:34.21MikeJ[Laptop]wow...
14:34.24MikeJ[Laptop]~pastebin
14:34.24jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca
14:34.29MikeJ[Laptop]clive-, use it
14:34.29clive-with the newjb I commented out teh otehr stuff
14:34.34*** join/#asterisk jmacz (~jmacz@63.245.86.244)
14:34.34*** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca)
14:34.36clive-sorry MIke
14:34.39MikeJ[Laptop]:P
14:34.59clive-tzanger will it find its own jiter buffer settings?
14:35.22tzangeryes that's the whole point of hte new jitter buffer
14:35.29tzangerI have jitterbuffer=yes and forcejitterbuffer=no
14:35.30tzangerthat's it
14:36.36[TK]D-FenderI can't seem to remember the Polycom web-admin default passwords, anyone care to remind me?  (doesn't appear to be "polycom" / "456")
14:36.47clive-forcejitterbuffer?...tas a new one
14:36.59*** join/#asterisk inNEEDofHELP (~jordan@rrba-146-118-79.telkomadsl.co.za)
14:37.06clive-let em wait for this person to get off the call so I can restart asterisk
14:37.19*** join/#asterisk virterm (~virterm@shiva.kanatek.com)
14:37.35clive-tzanger what value do you get on your line?
14:38.33tzangerclive-: it's dynamic
14:38.36tzangerthat's the whole point
14:39.15inNEEDofHELPhello,i just need to know if my panasonic pbx d500 e1 connection can pass calls to0 asterisk server? any help would be greatly appreciated.
14:41.00clive-inneedofhelp, howzit,,,
14:41.04thalinNeedofHelp: ist it able to speak SIP?
14:41.36clive-tzanger, ok, I have removed the maxjitterbuffer,,,we will see if thatworks,,still confused why the RTT jumps from 350 to 1200
14:41.48inNEEDofHELPthal,no i dont think so :)
14:42.04inNEEDofHELPclive-howzit..any ideas?
14:42.23*** join/#asterisk brookshire (~matt@207.111.174.1)
14:43.21vaewynBOOYAHH!   errr... morning all
14:44.30tzangerclive-: tha'ts your network, it has nothing to do with the jb, the jb is trying to compensate
14:44.50eKo1inNEEDofHELP: does it speak ISDN?
14:44.55vaewynanyone played with TCP SMDI and * yet? Or other methods of doing MWI from/to Norhell switches?
14:46.02SpaceBassis there an easy way to disable music on hold on one specific extension?
14:46.24*** join/#asterisk hubbaba (~me@68.114.5.160)
14:46.31ChkDigitSpaceBass: Put that extension into another context.
14:46.41clive-thanks for your help
14:46.56SpaceBassChkDigit thats what I was trying to avoid
14:47.38*** join/#asterisk da_monumental_1 (~da_monume@rrcs-24-172-102-198.midsouth.biz.rr.com)
14:48.53SpaceBasshow about for the outbound routing, can I disable it at that point?
14:49.22*** part/#asterisk clive- (~pirch@rndf-146-5-08.telkomadsl.co.za)
14:50.12*** join/#asterisk Myshenka (~gunde@217.9.101.85)
14:50.23hubbabaAnyone had any luck setting up an AdTran TSU 600 with 24 FXS ports.  I have a TE410P which has span 1 configured and working with a PRI.  Span 2 goes to the CSU and is using a T1 crossover cable.  Both ports light up green on the card when modprobed and ztcfg shows the channels properly.  My problem is that when  I plug in an analog phone, I don't get a dial tone.
14:51.22ChkDigitSpaceBass: I suppose you could add a SetMusicOnHold to any outbound extension, and make it go to dead air.
14:52.09SpaceBassdead air is great
14:52.11MyshenkaCan anybody recommend a page where I can read more about placing the contents of extensions.conf in a mysql table when I want to use Realtime? I know how to deal with the static bits and with exten=>... but what to do with include=> and switch=> ?
14:52.54SpaceBassChkDigit have two trunks one personal one business... dont want bruce springsteen or dave mathews playing when I put work calls on hold :)
14:53.30ChkDigitGotcha.  You should have a context for each trunk, selecting the right music on hold.
14:53.53SpaceBassChkDigit that will work
14:55.00Hmmhesaysopenvpn just rocks some days
14:55.11ChkDigitopenvpn rocks everyday.
14:57.35jhiverI can't get openvpn to always give machines the same private IP address, no matter how hard I try
14:57.42jhiverother than that, it works great
14:58.20Hmmhesaysassign them statically
14:58.40*** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
14:59.25NuxiDefinitely Asteirsk related => http://nytimes.com/2005/07/05/health/05sex.html?hp&ex=1120622400&en=98912eace83692a8&ei=5094&partner=homepage
14:59.43Nuxilol
14:59.55tzafrirNuxi, and for those of us unwilling to subscribe?
15:00.43Nuximake up a name, etc.  I use a new one every time the cookie expires.
15:01.39*** part/#asterisk hubbaba (~me@68.114.5.160)
15:02.25*** join/#asterisk santiago (~santiago@63.245.86.165)
15:03.36Nuxitzafrir, found a "mirror" http://www.contracostatimes.com/mld/cctimes/living/science/12056017.htm
15:05.34*** join/#asterisk pigpen (~mark@fw.seamans.cc)
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15:17.08eKo1Hmm... i have 'exten => 399,1,voicemailmain(@myvoicemailcontext)' and it still asks me for the mailbox. What am I doing wrong?
15:20.37*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
15:21.28shido6yess
15:21.36shido6it doesnt know what your voicemail mailbox is
15:21.40shido6you're not passing that to it
15:22.47vaewyn${EXTEN}@myvoicemailcontext   :}
15:24.47wektrafrir- bugmenot.com
15:28.15eKo1hmm...but it says in the documentation that the mailbox is an optional argument.
15:28.37Jas_WilliamseKo1: it is without it it prompts you to enter you mailbox :)
15:28.43fileyeah, so if it doesn't get it... it has to ask you
15:28.46fileit's not psychic
15:28.53*** join/#asterisk doolph (doolph@201.226.146.178)
15:29.20eKo1Oh, I see. I thought it 'figured it out' automagically.
15:30.08*** join/#asterisk mhnoyes (~mhnoyes@user-38lc0ml.dialup.mindspring.com)
15:30.31Jas_WilliamseKo1: it will do if you put ${CALLERIDNUM}@myvoicemailcontext
15:30.57eKo1yes, i was about to do that
15:31.06Jas_Williamsif the callerID is not a voicemial box then it will prompt for one
15:32.52eKo1worked like a charm. thanks
15:34.33wunderkinwhat are some codecs that can survive packet loss better than ulaw.. other than g729
15:34.35*** join/#asterisk Assid (~assid@203.115.64.61)
15:34.37Assidheya
15:35.02wunderkinout of 123 packets, i have 6% loss now
15:35.16vaewynjust NEVER use ${CALLERID}  and the 's' extension unless you control ALL callerid sources and can prove it :P
15:35.28vaewyns/extension/option
15:35.41vaewynwunderkin: ilbc
15:35.50vaewynhandles packet loss the 'best'  audibly
15:36.10*** join/#asterisk funxion (~funxion@63.214.236.140)
15:36.12wunderkinok
15:36.35wunderkinthanks, my cable modem used to be perfect before the holiday.. now its starting to suck.. :(
15:36.45vaewynif it is packet loss due to bandwidth restrictions then try GSM also if you feel like it
15:36.59wunderkinna i have plenty of bandwidth
15:39.00*** join/#asterisk Silik0n (~krice@newrso.suspicious.org)
15:40.57Assidhell yeah
15:46.03tzangerwunderkin: ulaw probably "survives" packet loss better than most others since hte least amount of voice information is sent per packet
15:46.12tzangerthrow a good PLC jitterbuffer on it (asterisk has one) and it works well
15:46.35tzangerilbc and g729 and perhaps speex too all have packet loss concealment algorithms to try and improve lost packet sound
15:47.09wunderkinso try turning on the jitter buffer? ilbc still is a little shaky
15:47.53*** join/#asterisk |Vulture| (~V@user-0c6tr11.cable.mindspring.com)
15:47.53wunderkini dont need it for all connections, only the one to my house.. which right now goes to a sip phone
15:48.53tzangerwunderkin: ilbc sounds like ass I've found, and I can't tell why, it's supposed to have excellent MOS
15:49.00wunderkinthe box is in a data center, the problem is just the connection to my house..
15:49.02wunderkinyeah
15:49.07tzangerI use gsm or g729, that's it.  (ulaw for faxes)
15:49.15*** part/#asterisk evangelion (~manzy_zet@ip-185-118.sn1.eutelia.it)
15:49.27Assidi thought gsm is "as clear" as ilbc
15:49.28|Vulture|tzanger: you don't like ilbc?
15:49.33doolphcant g729 support fax?
15:49.39tzanger|Vulture|: I want to like it (open source, open protocol) but it sounds like ass
15:49.40|Vulture|no ulaw for fax ONLY
15:49.40eKo1no
15:49.45tzangerdoolph: no compressed voice codec can do fax
15:49.45Nivexthe few times I've used ilbc, it's been nice, but I tend to stick with ulaw and gsm for compat purposes
15:49.57doolphmmm
15:49.58|Vulture|tzanger: I thought it sounds better than gsm
15:50.02tzangerevery time I turn on ilbc I get complaints about quality
15:50.04doolphdan
15:50.06tzangeruse gsm and nobody complains
15:50.12DaminOh shit..
15:50.12*** join/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca)
15:50.14Optichey yo
15:50.14ManxPowerI use Speex mostly.
15:50.16|Vulture|strange
15:50.45|Vulture|Ill have to try GSM again
15:50.47Assidisnt that supposed to be robotic?
15:50.52Nivexfor some reason I had Speex enable itself on a link this weekend and the other end said I was really choppy
15:51.08NivexAssid: gsm codec doesn't sound any worse than a cellphone
15:51.15ManxPowerNivex: That would be when you don't use disallow=all and then allow=thecodecyouwant
15:51.17Assidsame thing?
15:51.17|Vulture|cell phones sound bad
15:51.18doolphfor me its same
15:51.19Optici've got a little system with 10 spa-841's and 5 polycom ip500's... when a spa-841 calls a polycom, the spa user's voice is very quiet to the polycom user... polycom to polycom or spa to spa calls are fine
15:51.22Assidhrmm
15:51.27Assidthats weird
15:51.33Assidcellphones sound good here
15:51.35NivexManxPower: yeah.  I went back and got rather specific with it.
15:51.38doolphbecause polycom is better
15:51.46tzangerAssid: lpc10 is robotic (and very funny)
15:51.46[TK]D-FenderHey, anyone know where I can get documentation on Polycom's Micro-Browser XML format?  Or at least some samples...
15:51.53|Vulture|I notice a big diff on gsm and ulaw
15:51.58wunderkinis gsm known as something else? in a number form? i dont see it on the list on my gs101
15:52.02tzanger|Vulture|: well of course you will.  :-)
15:52.10Opticthe polycom phones are nice :)
15:52.16vaewyndomo ari gato mr roboto
15:52.27NivexWhat I really couldn't figure out is why my friend's asterisk box would allow a ulaw call but then couldn't hear the audio.  I switched to gsm and it worked just fine.
15:52.29[TK]D-Fenderwunderkin : GS101 doesn't support GSM AFAIK
15:52.30|Vulture|Optic: agreed just about to get some new 501s
15:52.35wunderkindamn
15:52.43Opticit's a bit of a problem in our system though
15:52.51Opticthey don't play well with the sipuras
15:53.18Jas_Williams2
15:59.12*** join/#asterisk santiago (~santiago@63.245.86.244)
15:59.41*** join/#asterisk vmlinuz (~nabudocon@red-corp-200.76.231.29.telnor.net)
15:59.57|Vulture|g729 sounds good.. but too expensive to use other than on external phones linked to * for me
16:00.02*** join/#asterisk blazint (~blazin@cm99.epsilon201.maxonline.com.sg)
16:00.18|Vulture|isn't there some GPL version of g729 in the works?
16:00.26tzanger|Vulture|: $10 is too expensive?
16:00.32loudthe intel one, but don't be cheap
16:00.34vaewynstill not legel in the US even if it is GPL
16:00.34tzanger|Vulture|: no, g729 cannot be GPLd
16:00.42loudoh
16:00.51|Vulture|tzanger: ah okay yea I have 4 licenses
16:01.12doolphwhat's the difference between the intel one
16:01.14|Vulture|tzanger: I am using ilbc between offices and for LD calls, then g729 to phones outside the LAN
16:01.39tzanger|Vulture|: you might want to try gsm -- smaller bandwidth footprint and (in my testing with a staff of 40) far better quality
16:01.58|Vulture|tzanger: I will try it tomorrow morning
16:02.02vaewynI was real tempted to go beat the #@!$@#$ outta the patent holders when I found they were at VON  :}
16:02.10*** join/#asterisk HDG-laws|sl33p (~yo@adsl-69-208-237-54.dsl.sfldmi.ameritech.net)
16:02.14|Vulture|lol
16:02.42*** part/#asterisk Myshenka (~gunde@217.9.101.85)
16:02.44wunderkinwell.. should i try the sip jitterbuffer
16:02.57ManxPowerwunderkin: That stuff is only in CVS-HEAD
16:03.03vaewyndo... or do not... there is no try   ;P
16:03.06HDG-laws|sl33pAnyone around who can help answer a few questions about Asterisk for use with Snom 360 + PSTN failover configuration?
16:03.14wunderkinyeah im using head
16:05.00*** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk)
16:05.09tzangervaewyn: what's wrong with the g729 patent?
16:05.11tzangerseriously?
16:05.39tzangerit's a novel, innovative invention.  While I dont' think that software patents should last for more than 5 years, it's a valid patent in my eyes
16:06.07|Vulture|its a good codec
16:06.29vaewynsoftware patents in general I have issue with...  but not even making it 'free for non-commercial use' is just being greedy
16:06.31|Vulture|I just wish I could buy like 50 licenses and have them be used dynamically on all my servers
16:06.58doolphwhy do you need 50 licenses?
16:07.07tzangerI have no issues with software patents, just as I have no issues with hardware patents.  it takes a lot of work and expense to do this kind of work and there should be protections in place to let me recoup my investment before the floodgates open
16:07.08doolphwill it runs within internet or just lan?
16:07.17tzangernow 17 years is a bit long, even for hardware
16:07.22vaewynI just have 2 servers with g729a landing points... rest of them don't talk it
16:07.27ManxPowerI don't have much of a problem with the G729 patent holders.  They will at least license it in small qtys.
16:07.35tzangerand the whole idea of frivilous and stupid, broad, non-novel patents sucks
16:07.36|Vulture|I have 9 * servers each making ~10 calls at a time
16:07.36cpmBut software patents DO last for more than 5 years, a lot more in fact. This whole software patent thing is bogus. It's ALL derived, everything software, and most "things" in general are inspired by something else. It's all bogus.
16:07.46tzangercpm: bullshit
16:07.47*** join/#asterisk trickyrick77 (~rsegrest@207.111.174.1)
16:08.02vaewynManxPower: umm... your definition of 'small quantities' must be different then mine
16:08.16ManxPowervaewyn: Um, 1 license at a time?
16:08.19|Vulture|lol
16:08.20tzangercpm: software patents are no more evil than hardware patents.  And yes things are based off other things but that doesn't mean that the approach or resolution of the technical issues was easy or obvious.
16:08.34ManxPowervaewyn: Maybe you are thinking of the greedy G723.1 bastards?
16:08.39tzangercpm: as I said, I feel that 17 years for software is insane.  But patenting software isn't insane.
16:08.52tzangercpm: just as I said that I feel that 17 years for hardware's insane.
16:08.59vaewynManxPower: go ahead...  call them up and try and buy one license...  they will refer you to a 'reseller' that has purchased a 10k block from them
16:09.13ManxPowervaewyn: I just call up digium.
16:09.26ManxPowerDigium issues me a license.  All done.
16:09.30vaewynexactly...  hence... the patent holder is not allowing small scale sales
16:09.35vaewynit's just digium being nice
16:09.47vaewynthey had to buy a block of them to start
16:09.54cpmWhat's the point of a patent?
16:09.57ManxPowervaewyn: Hmm?  Until about a year ago, digium got each license, individually from the patent holders.
16:10.00tzangervaewyn: the patent holder is allowing it.  how is it bad?
16:10.07*** join/#asterisk alephant (~cmd@c-24-3-52-93.hsd1.pa.comcast.net)
16:10.11alephantHi all...
16:10.22tzangeryou can't buy individual MAC addresses either, you have to buy a big block...  how is this a problem?
16:10.48Nuxiyou can make up a mac address with out a license.
16:10.56vaewynexactly :}
16:11.01vaewyngood call Nuxi
16:11.09tzangerNuxi: actually that's not supposed to be allowed
16:11.14alephant...if I connect * to an analog phone line, and I want to bounce incoming calls to my cell phone, how many phone lines and FXS's do I need?
16:11.21ManxPower<PROTECTED>
16:11.32tzangersame with PCI IDs... you can hijack someone else's (digium did this even) or make one up
16:11.42*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
16:11.51Nuxioops was that out loud.
16:13.26vaewynI think...  1-5 years...  let them patent something they can PROVE within reasonable doubt is unique (and not just because it is done on a computer) and rape the crud outta it... but after that let it be free
16:13.54cpmtzanger, dont' get me wrong, I don't blame digium in the least, that's the way of the world, but I do stand by the concept that "ideas" should be patented is bullshit.
16:14.39mutilatorwhats being patented?
16:14.53vaewynheck... even real 'hardware' patents are getting out of control these days...  I mean.. they let them patent a swing for @@#$@ sake
16:14.53shido6ideas shouldnt be patented?
16:15.09shido6wow
16:15.54vaewynI love it that I can't patent    1 + 1 = 2...   but I can patent    my $tot = 1 + 1; print $tot;
16:16.06shido6you ever had an idea taken from  you and someone else made millions because of it?
16:16.18mutilatordoesn't everyone
16:16.22Nuxiactually you can patent 1 + 1 = 2.  It's a method.
16:16.34mutilatorw00t
16:16.36shido6no
16:16.42mutilatorcharge elementary schools to use it
16:16.54NuxiIt's a method of constructing 2.
16:16.56tzangercpm: ideas shoudl not be patented, and indeed they can't be
16:17.00tzangerPROCESSES are patented
16:17.01*** join/#asterisk mkrufky (~mk@68.160.103.77)
16:17.07Assidright
16:17.14Assidhyperthreading.. hypertransport
16:17.16Nuxiideas can easily be wrapped in process.
16:17.23mutilatori love it when you talk dirty Assid
16:17.26Assidbottom line.. nothing gets done
16:17.45Assidmutilator: now unless your a girl.. you aint getting none
16:18.28Assidburn!
16:18.29Assidokay
16:18.30*** join/#asterisk cinix (~ax@24-52-166-190.lndnnh.adelphia.net)
16:18.37Assidi desperately need to get more sleep in
16:19.06eKo1ditto
16:19.07DarthClueAssid: sleep is waste of time.  I should know.
16:19.24cpmSure they can. How many software patents would like to review that are nothing more than ideas?
16:19.56*** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net)
16:20.10*** join/#asterisk damajor (~damajor@tuxmania.org)
16:20.12cpmsome history, http://www.vrijschrift.org/swpat/030508_1/
16:20.20NuxiIt used to be you had to have an implementation to patent.  now you can patent a "gee wouldn't it be nice if ..."
16:20.35DarthClueCome to cluecon.  The only voip conference to feature nearly every major VOIP developer.
16:20.47AssidDarthClue: tell me about it
16:20.55*** join/#asterisk rob (~rob@2001:4c20:0:2:211:43ff:fecd:8834)
16:21.05Lee__if there are any app_conference users here, I get no sound when connecting two SIP phones to the same room. Has someone else found this?
16:21.08Assid4-5/day.. everyday ..since months
16:21.12Assidstarting to really get to me
16:21.48vaewyneep!   prior art!
16:21.57AssidNuxi: you are in violation
16:22.04robHi, has anyone got Asterisk running properly on OS X 10.4.1?
16:22.05AssidDarthClue is already patented by me
16:22.14Assidyou now owe me 100 million USD
16:22.18*** join/#asterisk vmlinuz (~nabudocon@red-corp-200.76.231.14.telnor.net)
16:22.21vaewynNuxi: is that argv[0]  or argv[1]  ?  ;P
16:22.37Lee__rob, no.
16:22.54NuxiAssid, no, I patented a diferent way of using DarthClue. not the closet use you used.
16:22.55Assidi wanna patent the process of patenting
16:22.55robThere's a package floating round which is a 2003-10 CVS, which seems to be broken with dtmf on the 7960.
16:23.12Lee__have you contacted the package author?
16:23.12AssidNuxi: i patented DarthClue itself..
16:23.25Assidtherefore you are in violation
16:23.29NuxiAssid, I'll see you in court.
16:23.32Assidfine
16:23.39Nuxifine.
16:23.46robLee__: The developer's site on versiontracker takes you to Asterisk.org.
16:23.47Assidyou gotta travel 7000 miles to come and meet me now
16:24.15AssidDarthClue: that was already patented earlier..
16:24.20Lee__rob, see if you can track down an email address. there's also lists.digium.org.
16:24.22Assidyuo can add it to your tab
16:24.48Assidbut since you helped me earlier.. i give you 1 month free usage.. license free
16:25.29mkrufkythis patent stuff is all a big joke, right?  (just entered room)
16:25.43vaewynyes and no  :P
16:25.47mkrufkyheh
16:25.49vaewynSWEET!!!!
16:26.06Nuximkrufky, some people are grumpy about codec patents.
16:26.29mkrufkyi am grumpy about it too...
16:26.32mkrufky:-(
16:26.35robLee__: I will do, but is there any other way that I can deal with the problem other than having to chase someone down?
16:26.38vaewynthe other half shall be solved when you open the patents on killing politicians :P
16:27.12mutilatorthose are called hits ;P
16:27.30Lee__rob. have you tried setting dtmf=rfc2833 for the Cisco phone in sip.conf?
16:27.36mkrufkynuxi: you AGREE with these patents?!?
16:27.46vaewynbut I get to use openoffice for free ;P
16:27.46|Vulture|are we still on this?
16:27.50mkrufkylol
16:27.51vaewynyep
16:28.09mkrufkygotcha
16:28.20Lee__couldn't resist
16:28.23robLee__: Yes, I have, unfortunately it seems to be doing nothing else.
16:28.34tzangerLee__: :-)
16:28.43Lee__rob, you may not like my other suggestion.
16:28.44tzangerNuxi: you can't patent ideas
16:28.47tzangeryou patent processes
16:28.57mkrufkyi think that some patents are okay, but not the stupid little patents where m$ will patent the use of scrollbars or something basic like that
16:29.01Nuxiany idea can be wrapped loosely in a process.
16:29.04Lee__rob, use Linux.
16:29.12vaewyntzanger: go look at the recent patent filings...  find a line of code in them...  trust me... they ain't there
16:29.26tzangermkrufky: those are obvious and trivial patents and should never be allowed.  that's a failure of the patent system being greedy
16:29.33mkrufkyexactly
16:29.39tzangervaewyn: see my last comment to mkrufky
16:30.02mkrufkytzanger: it's like a new word for technological monopoly
16:30.10Lee__it would definitely benefit Asterisk if all the codecs used internally were non-patented.
16:30.32vaewynheck... we might as well move to...  anyone can patent anything that hasn't been patented before...  but you only get it for 1 year... you can't get a hold inthe market in that long...  screw you
16:30.43NuxiIt benefits Asterisk to be able to use all codecs.
16:30.50Nuxipatented or not.
16:31.16mkrufkyi havent looked into that... how does asterisk use a patented codec?
16:31.32mkrufkyis there a binary version of the codec that must be linked or something?
16:31.39Lee__mp3 for moh, g729 for SIP/IAX channels.
16:31.48Lee__yeah, binary.
16:32.02mkrufkywhat a PITA
16:32.02vaewynor... even better make the patents only patentable by a single person that is responsible for it...  no more companies holding patents and such
16:32.04Lee__or compiled outside of Asterisk in the case of mp3 support.
16:32.16shido6whew
16:32.39Nuximake mpg123
16:32.42Lee__mkrufky: tip your glass to the companies that patented the codec for that one.
16:32.53mkrufkylol
16:33.08mkrufkythe companies are taking this stuff too far if u ask me
16:33.29mkrufkyi cant believe the hurdles we have to go through just to get a datasheet to write a linux driver
16:33.45mkrufky<-- video4linux dev
16:33.56*** join/#asterisk shaonss (~shaonss@61.68.15.40)
16:34.21vaewynactually...  in the long run the best thing would be if all standards bodies required free-to-use patents grants for anything they ratified...  that would rock right there... and drive restrictive patent holders out of business
16:34.26robLee__: Ah, I found the developer's site and have now upgraded.
16:34.32robHowever, I still have the problem.
16:34.50|Vulture|Anyone know if it is possible to subtract or add to a ${TIMESTAMP}, I need to add 3 seconds to it
16:35.04mutilatori think they do that with medicine now don't they vaewyn?
16:35.14Lee__mkrufky: this month's LJ says the V4L driver is currently unmaintained. Is that true?
16:35.28vaewynmutilator: in some cases yes... I think more are running that way
16:35.53shaonsscalling card application need help!!!!!!!
16:36.04mkrufkyMauro Carvalho Chehab has just stepped up as the new maintainer
16:36.06mutilatori just heard something about that in the news a bit ago, they don't require royalties on exprimental use of stuff
16:36.18mkrufkyvaewyn: Mauro Carvalho Chehab has just stepped up as the new maintainer
16:36.39mkrufkyvaewyn: I have been helping him out a lot lately with patching to -mm
16:36.56shaonsswhich calling card application is the best for asterisk?
16:37.05mkrufkyvaewyn: and also helping to mediate between v4l and dvb trees
16:37.10Lee__cool. I don't use video much but it's good to know it's not abandoned in the kernel.
16:37.38Lee__I just compiled 2.6.12 and noticed a load of DVB additions since 2.6.8
16:37.48mkrufkyLee__: in fact, there has been a TON of activity in v4l lately
16:38.05mkrufkyLee__: DVB is a separate project, although there is some overlap with v4l
16:38.08|Vulture|Anyone know how to add seconds to a ${TIMESTAMP}?
16:38.13vaewynspeaking of which... I need my trusty friend to finish the DVB drivers so they will work with analog alos attached to the pcHDTV2000/3000
16:38.20Lee__Delta Song uses Linux for each seat's video display. I saw the FB penguin boot up  :)
16:38.28mkrufkyvaewyn: do u have a link to that article in lj?
16:38.31*** join/#asterisk Cheetah (~Snak@62.217.48.111)
16:38.49mkrufkyvaewyn: i believe that card has full analog support
16:38.55Cheetahhi folks
16:39.07Cheetahis there a way to redirect an incoming call in extensions.conf to the voicemail box after a certain amount of rings?
16:39.09shaonsscan anyone please help for calling card application?
16:39.12robLee__: How will Linux help me on one simple dtmf problem, surely the configuration issue could exist on anything?
16:39.20mkrufkyvaewyn:  If the analog part needs fixing, let me know and I can probably take care of that for you
16:39.20Cheetahlike, 10 seconds/rings and nobody picked up
16:39.30vaewynmkrufky: was Lee__ that saw that
16:39.39Lee__rob, because I am not using OS X and I don't have said DTMF problem and we have the same phones.
16:39.50mkrufkyoops
16:40.03mkrufkyLee__: do you have a link to that article in lj ?
16:40.08Lee__mkrufky: it wasn't an article, it was a one liner in the diff -u kernel notes section.
16:40.19mkrufkyoh, i gotcha
16:40.33mkrufkyu will notice that the MAINTAINER file has been updated in 2.6.13-rc1
16:40.37*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com)
16:40.43mkrufky:-)
16:40.43vaewynmkrufky: problem right now is that I can't seem to get it to run in analog and DVB... I have to toast the modules and reload the other set to switch between DVB/HDTV and analog
16:40.53Lee__I'm not that cutting edge with kernels. I stick to "releases" if there are such things.
16:40.55robLee__: What are your settings for SIPDefault: dtmf_inband:0 , dtmf_outofband: avt_always, dtmf_db_leve: 3, dtmf_avt_payload: 101?
16:40.56vaewynmkrufky: can't do that from in mythtv so :P
16:41.28*** join/#asterisk NightHawke (~NightHawk@66-190-111-175.static.sprn.tx.charter.com)
16:41.33NightHawkehi'lo
16:41.40*** join/#asterisk yoink (~yoink@MTL-HSE-ppp165512.qc.sympatico.ca)
16:41.42Lee__rob, I'll msg you. is that ok?
16:41.48*** join/#asterisk The_LightSide (~dial@tpr-165-247-230.telkomadsl.co.za)
16:41.57robLee__: That's fine thanks.
16:42.12mkrufkyvaewyn: interesting......  it is cx88 board #22, correct?
16:43.49NightHawkei got a set of TDM 400P FXO/FXS cards. what i want to do is hook up a set of analog phones into the FXO card.
16:43.59NightHawkehow do i register each extension in AMP?
16:44.12|Vulture|FXS for phones;FXO for lines
16:44.19NightHawkeFXS thanks
16:44.30NightHawkei got the ZAP extensions in there but when i try to get a outside line i get a busy
16:44.50|Vulture|NightHawke: I am not sure for amp, but what you want to do is edit zaptel.conf and zapata.conf to see the TDM cards
16:44.51mkrufkyvaewyn: its very busy in here..... meet me in #v4l and I can probably help you with your board
16:44.54mutilatorfricking rye chips
16:45.01mutilatorthats what i hate about chex mix
16:45.07mutilatorthe damn rye chips
16:45.08NightHawkeAMP has direct links to see the conf files
16:45.08Lee__so...anyone using app_conference successfully? When I connect to a conference I get no audio.
16:45.33|Vulture|NightHawke: then you will just say Dial(Zap/(Zap FXS Line),20(time in seconds to dial),r)
16:45.50vaewynmkrufky: :}  yeah... it gets that way..   not sure which chip...  not in front of me at the moment...  I have the dmesg | grep bttv   output for it
16:46.17|Vulture|mutilator: I don't like all the pretzles
16:46.18NightHawke|Vulture|, i see the extensions in zaptel.conf now
16:46.58vaewynmkrufky:  hahahahaha   I think it know's you...  it just worked without reloading  hahahaha
16:47.16mkrufkyvaewyn: problem solved already???
16:47.42*** join/#asterisk LeoB (~chatzilla@wireless-107.media.mit.edu)
16:47.57vaewynmkrufky: I think you scared it into submission...   that is wierd...  oh well... if it happens again I'll get some diag info on it
16:48.00mkrufkyvaewyn:  i just noticed that pchdtv3000 had two different boards, one is bttv-based and the other cx88-based
16:48.04mkrufkyhaha
16:48.08shido6oh lord
16:48.12shido6dont get me STARTED
16:48.14mkrufkywell i'll be happy to help you
16:48.14vaewynmine is the 2000 board so...
16:48.17shido6MSI TV@anywhere
16:48.17LeoBhello there, how can I prevent log messages from eating my entire hard drive?
16:48.19shido6BS
16:48.36NightHawkeok vulture, i'm lost on this... that and the effing phone is ringing
16:48.36shido6stinkin ati and linux hate each other, two tv cards
16:48.44NightHawkei'll ahve to deal with it later on
16:48.47shido6i cant get sound to work on my TV@nywhere
16:48.50shido6conexant based
16:49.00mkrufkyshido6: meet me in #v4l
16:49.03LeoBthey've eaten 13GB already...
16:49.07Lee__LeoB: logger.conf
16:49.09vaewynshido6: hehehe...  I am fairly sure I will never buy an ATI device for under linux ever agian :P
16:49.28mkrufkyvaewyn: maybe pchdtv3000 and pchdtv2000 use same chips?  i dont see direct support for 2000 board anywhere
16:50.35vaewynmkrufky: I know they use different firmware for one chip at least  'or51211' versus like 'or51221' or such
16:50.44vaewynnot sure which chip that is though
16:52.00Lee__nobody with app_conference? too specialized? a hack?
16:52.15*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
16:53.38NightHawkeDial(Zap/(Zap FXS Line),20(time in seconds to dial),r)  i assume that 20 is the extension # assigned to the zap extension
16:54.21|Vulture|NightHawke: I doubt its 20 since you only have 2 TDM cards so 1-4,5-8
16:54.25vaewynumm... no... it is the time in seconds to ring the zap interface before giving up and going on in the dialplan
16:54.36HDG-laws|sl33pAre any digium cards required for a pure SIP / * setup ?
16:54.47|Vulture|no
16:54.50shido6vaewyn NEVER
16:54.50vaewynHDG-laws|sl33p: nope
16:54.55HDG-laws|sl33pIf I want failover to PTSN, then I will
16:54.57shido6I love the AIW in xp but
16:54.57HDG-laws|sl33pCorrect?
16:54.58shido6not
16:55.00shido6in linux
16:55.38mkrufkyvaewyn:  i dont know... i havent seen any bttv/dvb boards yet.. so im not sure how they fall in v4l/dvb trees
16:56.00NightHawkeDial(Zap/(Zap 1-3),20(time in seconds to dial),r)
16:56.35vaewynshido6: if I'm gonna have that much $$ and hassle though I am going to have my hdtv ;P  so linux with the pchdtv  or air2pc cards is the way to go :P
16:56.56spyrouxdoes the version 1.0.9 of asterisk include Realtime configuration ?
16:57.42vaewynI think realtime is cvs head only
16:58.17shido6ooh
16:58.18shido6http://www2.ati.com/drivers/linux/linux_8.14.13-inst.html
16:58.24shido6but Ive done this manually
16:58.50ManxPowerRealtime is only in CVS-HEAD
17:00.13HDG-laws|sl33pWith a 100% SIP system + *, what would I need in order to have a POTS failover ?
17:00.17ManxPowerThrough the generosity of our Premier sponsor [for Cluecon], Sangoma Technologies
17:00.21ManxPowerIck.
17:00.25spyrouxManxPower: thanks
17:02.30|Vulture|ManxPower: any clue how I would add 3 seconds to a TIMESTAMP?
17:02.48mkrufkyvaewyn: as of yesterday, there is now support for fusionhdtv3 gold (t/q) in video4linux cvs for both dvb and analog
17:04.20vaewyn|Vulture|: I think you have to use a temp variable...  ${TIMESTAMP} is not writeable
17:05.01Lee__I'm going to call it on the wiki. app_conference is broken with SIP phones and CVS HEAD.
17:05.09|Vulture|vaewyn: yes but right now I am just using TEMP, I have to isolate the timestamp's second side, then subtract then readd it together
17:05.15|Vulture|file: any ideas?
17:05.48*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
17:06.05|Vulture|I have: Math(TEST,${TIMESTAMP}+3) but returns "TEST|20050705-130136+3" = 20050708.000000
17:06.41outtoluncVariables marked with a * are builtin functions and can't be set,
17:06.41outtolunconly read in the dialplan.  Writes to such variables are silently
17:06.41outtoluncignored.
17:06.49outtolunc${TIMESTAMP}            * Current date time in the format: YYYYMMDD-HHMMSS
17:07.13|Vulture|yea... so I need to isolate HHMMSS and then subtract 3, then recombine it with the YYYYMMDD
17:07.50|Vulture|but how do I isolate it?
17:07.53vaewyn|Vulture|:  SetVar(TEMP=$[${TIMESTAMP}+2])
17:08.00vaewynerr...  or 3... not 2 :}
17:08.11|Vulture|vaewyn: Ill try that
17:08.48*** join/#asterisk brenda (~b@c-67-182-205-227.hsd1.ut.comcast.net)
17:09.36*** join/#asterisk DrRighteous (~DrRighteo@68.199.175.49)
17:09.44ManxPowerput a space around the +
17:09.49|Vulture|kk
17:10.04DrRighteousanyone have experience with mutlple instances of asterisk on a box?
17:10.06vaewynergh... that's right also
17:10.17ManxPowerI use this: SetVar(LOOP=$[${LOOP} + 1])
17:10.46ManxPowerDrRighteous: Only when the version of "ps" on the box shows each thread as a seperate instance of Astersik
17:10.53vaewynsomeone needs to program a 'foreach' application :}
17:10.53|Vulture|ManxPower: when I put the space it set it to 3
17:11.10ManxPower|Vulture|: then try my method
17:11.25ManxPowervaewyn: there might be in AEL?
17:11.32bkw_ManxPower, so why the Ick?
17:11.38DrRighteousManxPower: my problem is that Im using the -C to allow for multiple asterisk directories for resources/pids... but I still getAsterisk already running on /var/run/asterisk/asterisk.ctl
17:11.53ManxPowerbkw_: Having a primary competitor sponsering an "asterisk developers conference"
17:12.08fileCluecon isn't an asterisk developers conference
17:12.12ManxPowerHas Sangnoma even contributed any code to the Asterisk CVS.
17:12.18bkw_yes
17:12.24ManxPowerfile: then someone should change the /topic
17:12.30|Vulture|nah that returned: WARNING[15159]: ast_expr.y:666 op_plus: non-numeric argument
17:12.36ManxPowerAh, PBX, not Asterisk.
17:12.38filethe topic is perfectly fine :P
17:12.39bkw_its about 33% asterisk related
17:12.39ManxPowerThat's not so bad then.
17:13.00enderanybody know what the default username/password is on Sipura phones?
17:13.03enderIP301 ?
17:13.05ManxPower|Vulture|: maybe you need a newer version of bison.  All I can say is that The sample I pasted is verbatum from my extensions.conf
17:13.20vaewynummm then if it is only 33% asterisk is it taking 50% of the /topic?  ;P
17:13.23*** join/#asterisk Mw3 (mw3@daisy.chains.ch)
17:13.31bkw_ManxPower, and since when is illegal to be a competitor in an open market?
17:13.40|Vulture|ManxPower: bison (GNU Bison) 1.875c what are you running?
17:13.52tsume*0.00000000001kb
17:14.07bkw_tsume, what software?
17:14.20tsumebkw_: its medical building software :)
17:14.24bkw_ah
17:14.26vaewynender: IP301 is polycom... not sipura
17:14.38enderoh shit.
17:14.38tsumebkw_: I didn't feel like charging 700 extra dollars to all customers
17:14.41enderI'm a dork.
17:14.43filebkw_: I want to KILL cube1
17:14.44vaewyn:}
17:14.49enderI just replaced my sipura w/ a polycom.
17:14.53Opticmooo
17:14.56enderanybody know what the ploycom default password is?  (:
17:14.59Opticender: which polycom?
17:15.02Opticender: 456
17:15.06vaewynpolycom/456
17:15.10Opticender: for the web it's Polycom/456
17:15.14tsumebkw_: they kb'ed me fast out the room like a car on nitros
17:15.15Opticyou need the cap P :)
17:15.36tsumebkw_: Its like channel of deliquent trolls :P
17:15.38enderOptic: 301 thakns.
17:15.43Opticpolycom's ftp provisioning is a bit shady
17:16.10enderhrm, 'polycom' user and '456' password is not working.
17:16.32Opticyou need the cap P :)
17:16.35OpticPolycom/456
17:16.50enderah
17:17.35Optic301 eh?  We have a bunch of 500's here
17:17.42enderyeah, 301.
17:17.48Opticsipura callers sound very quiet on them
17:17.53Opticwhat's the difference with the 1's?
17:17.56Optic301, 501, etc?
17:18.02enderseems like a decent value bump from the Sipura phones
17:18.08Opticyes
17:18.10endermore memory and something else I thought.
17:18.13vaewynmore mem
17:19.01*** part/#asterisk loick (~loick@81.255.80.161)
17:20.32vaewynI wish they would enable the microbrowser on the 500s as well...
17:20.42*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
17:21.41HDG-laws|sl33pAnyone using Snom 360, I'm preparing to buy 15 of them
17:24.05HDG-laws|sl33p:( I killed the channel
17:24.34vaewynsorry... snom's are technically very nice phones... but their user interface and feel are attrocious
17:24.46PBXtechwhat is the cisco directory that was included in the older Ast@home?
17:25.09*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
17:25.10HDG-laws|sl33pAny other suggestions then for 6 CO line capable SIP?
17:25.14SpaceBassPBXtech what is it? xml
17:25.17PBXtechya
17:25.53vaewynPolycom 600s
17:26.08SpaceBassPBXtech its just an xml file that you have to manually edit
17:26.20SpaceBassfor 6 lines, the 7960 isnt bad
17:26.21PBXtechthey had a web interface to it
17:26.33SpaceBassconfiguring 15 is just as easy as 1
17:27.12MikeJ[Laptop]~rtfw
17:27.12jbothmm... rtfw is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer
17:27.23bkw_jbot forget rtfw
17:27.36MikeJ[Laptop]jbot, remeber the old rtfw :)
17:27.38crash3mread the fuckin web?
17:27.39bkw_jbot no rtfw
17:27.50filejbot: forget rtfw
17:27.54SpaceBassalways thought it was stfw
17:27.57crash3mme to
17:28.00vaewynbkw_ forget jbot
17:28.01SpaceBassand pretty much a rude thing to say, to boot
17:28.02vaewyn:P
17:28.04filejbot: no rtfw
17:28.08*** join/#asterisk Assid (~assid@203.115.64.61)
17:28.18vaewyn~rtfm
17:28.18jboti heard rtfm is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer
17:28.27vaewynhehehe
17:28.36SpaceBass~stfw
17:28.37jbotmethinks stfw is Search The F*cking Web.  See http://justf*ckinggoogleit.com/
17:28.43SpaceBasslol
17:28.55*** join/#asterisk mrgoby (~mrgoby@64.208.211.80)
17:28.59*** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc)
17:28.59MikeJ[Laptop]jbot, forget rtfm
17:29.05*** join/#asterisk gtigene (~chatzilla@70.89.216.41)
17:29.27DarthClue~rtfw
17:29.27jbotfrom memory, rtfw is http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses.
17:29.37MikeJ[Laptop]jbot, no rtfm
17:29.38jbotsomebody said no rtfm was Read The Fucking Manual (TM)
17:29.52gtigeneMy asterisk uses SIP phones on a local area network but sometimes it drops calls. Another time I could hear my caller but he could not hear me. I am using CVS Head. What could be the cause, or where should I look..
17:30.11PBXtechi found it http://sf.net/projects/webaddressbook
17:30.13Opticsounds like it's not just a lan
17:30.18Optici've never had sip fail me on a regular basic lan
17:31.05gtigeneOptic, was that a response to my  question?
17:31.10Opticyes
17:31.19gtigeneOptic: thank you
17:31.28HDG-laws|sl33pSounds like NAT.
17:31.29Opticare you sure you don't have any routers, or nat, or anything like that between your asterisk and your phones?
17:31.39vaewyn~rtfm
17:31.39jbotfrom memory, rtfm is Read the Fine Manual (or in other words we are too busy for your question and the answer exists elsewhere)
17:31.40vaewyn:}
17:31.53gtigeneOptic, there is only an Ethernet switch
17:32.07Opticgti: that is a very strange thing then
17:32.16HDG-laws|sl33pOnly a switch? What is the switch connected to?
17:32.16Optici can't help you because I have never had that problem :)
17:32.27gtigeneOptic, OK
17:32.32Assidhrmm.. if an ip phone is on NAT but the asterisk server is on the net.. will the devices connect?
17:33.10gtigeneHDG-laws|sl33p: Switch is connected directly to phones and directly to Asterisk host.
17:33.58HDG-laws|sl33pWhere is the internet coming into play?
17:34.00enderis it still true that Polycom phones don't like type=friend ?
17:34.13ManxPowerender: that has NEVER been true
17:34.39vaewyntype=friend is just semi evil... and the people that first ran into that were using polycoms...
17:34.48enderhrm, ok.
17:34.51enderwiki says it is.
17:34.54vaewynbugs by association... not really a problem
17:34.58gtigeneHDG: There is one phone, hardly ever used, that is not on the LAN. They are using it to test something called a GRE tunnel. It talks to us through a router.
17:35.10ManxPowergood thing they are making peers and users going away soon and everyone will use friend
17:35.33vaewynManxPower: really?   is there a thread on that?
17:35.41ManxPowervaewyn: I'm sure there is. 8-)
17:35.46vaewynI hope they have fixed the friend issues :}
17:36.04ManxPowerFortunatly, type=friend isn't REALLY evil.  It just appears that way to people that don't understand their =friends
17:36.31*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
17:36.34vaewynthere are things you can do (correctly) with the user/peer that friend barfs on still
17:36.42ManxPowertype=friend assumes incoming and outgoing authentication username/secret are the SAME.  This is usually true for phones, but not always true with ITSPs or Gateways
17:36.44AssidHDG-laws|sl33p: just curious if that question was for me
17:36.59vaewynManxPower: exactly
17:36.59ManxPowervaewyn: I think all that stuff is/will be fixed.
17:37.37HDG-laws|sl33pAssis No, I don't believe so
17:37.38ManxPowerthey SHOULD have had type=incoming|outgoing|bidirectional
17:37.40HDG-laws|sl33pAssid*
17:37.47vaewynhehehe... now instead of seperate entries we'll have  'peersecret=blah   usersecret=blah2'  :P
17:38.07Assidi wanna link 2 * boxes!!!
17:38.11Assiddunno who with
17:38.12Assidhehe
17:38.21Assidhrmm
17:38.27vaewynManxPower: peer/user is much better for defining than 'incoming/outgoing'  cause those terms are relative
17:38.30Assidwhat about voip gateways?
17:39.31*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
17:39.39Assidshido6 my man
17:39.54vaewynWill be nice to not have to dupe some info in both entries though...  friend is a good idea...  just needs to be 'fixed'
17:40.09bkw_shido6, yo
17:40.12shido6?
17:40.14shido6bkw
17:40.18bkw_you gonna still speak?
17:40.20shido6you talked jeremy into using your sys
17:40.22shido6yeah
17:40.25shido6Fraud
17:40.32Assidfraud?
17:40.39shido6CC fraud
17:40.43Assidwho did it?
17:40.45shido6Rates Fraud
17:40.48bkw_can you go register and select the speaker option.. then I can get you into the head count?
17:40.59shido6Assid, its what took live voip out in addition to other things
17:41.05shido6ok
17:41.11Assidhrmm
17:41.11bkw_no livevoip was setup to be a fraud from day one
17:41.17bkw_this isn't the first time they have done this
17:41.35bkw_shido6, I have found some interesting info
17:41.42*** join/#asterisk point (1000@213.27.44.55)
17:42.05tzangeroh?
17:43.13shido6greg@cluecon, bkw
17:43.49*** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt)
17:44.33darwin35and the day drags on
17:46.12ManxPowerIf my damn customer hurries up and gets to the remote site, I MIGHT be able to see a movie this afternoon
17:46.52*** join/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc)
17:48.14*** join/#asterisk jfonsecausa (~jfonseca@12.42.141.34)
17:48.21eKo1what movie?
17:50.22BhaalHey guys, with Asterisk how easy is it to setup an automated operator so people can choose the extension they want to call once asterisk has answered the call?  and voice mail if the extension isnt answered?
17:50.46AssidBhaal: pretty easy
17:50.58Assidcomparatively
17:51.05vaewynBhaal: not too hard...   hey...  just out of curiosity... do I remember you from #fnr?
17:51.19Bhaalvaewyn: yes, you do
17:51.33vaewynBhaal: ;}  cool... long time :}
17:51.36AssidCpuID u up?
17:51.39BhaalAssid: So its reasonably well explained in the docs?
17:51.45mishehubah.
17:51.54Bhaalvaewyn: hehe indeed...  Hows things?
17:51.58mishehuBhaal: good source of info is at http://www.voip-info.org
17:52.09AssidBhaal: yeah.. you can find some info on t
17:52.10Bhaalmishehu: Ahhh thanks
17:52.40vaewynBhaal: it's not spelled out...  but everything is there... and once you get the hang of it... these are easy apps...  :}  and we are always here to help with problems
17:52.49vaewynBhaal: Things are good :P
17:52.53*** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt)
17:53.29AssidBhaal: just record your greetings and stuff and keep them ready
17:53.32*** part/#asterisk point (1000@213.27.44.55)
17:53.38Assidonce you start working on it.. things move rather fast
17:54.03SpaceBassAssid you ever get up and running?
17:54.13Bhaalsweet...  I saw a festival conf file, how easy is it to setup with that?
17:54.28AssidSpaceBass?
17:54.30vaewynBhaal: You know... the #fnr guys seem to get into asterisk a lot  :}  I've seen SynRG and Mysticone and now you :P
17:54.41Bhaalhaha
17:54.43SpaceBassfestival works fine, but it sounds like crap... think 1982 robot voice
17:54.45*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
17:55.09vaewynBhaal: festival is a bitfun...  but have someone with a good voice just record it...  way easier and cheaper on the CPU... and sounds better :P
17:56.00*** join/#asterisk darylp (~daryl_jus@63-208-162-62.digitalrealm.net)
17:56.08AssidSpaceBass: acutally.. i got it to "work" but 2/4 times.. it doesnt really do what i want and hangs up my zap
17:56.09Assidbrb
17:56.12Assidphone call
17:56.16BhaalIve bought a sip <-> regular fone adapter .. and have a sip provider (no incoming number yet coz they havent got any allocated) .. Ive got asterisk working as the proxy etc ... just waiting for the incoming number and I will set the rest up...
17:56.27*** join/#asterisk tris (~tristan@camel.ethereal.net)
17:56.57vaewynwow... #fnr is gone...  1 person...   :{
17:57.24Bhaalvaewyn: Its #freematrix or something like that now
17:57.25SpaceBasswhat was #fnr?
17:57.35BhaalSpaceBass: Freenode Radio
17:57.39SpaceBassahhh
17:57.42vaewynSpaceBass: freenode radio back in the day :P
17:57.44vaewynit rocked
17:58.02vaewynor in otherwords... geeks with mics :P
17:58.08Bhaalhahaha indeed
17:58.43*** join/#asterisk MattH (~matth@noc-wireless.chilitech.net)
17:58.44*** part/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca)
17:58.51MattHHi.. does anyone know what's up (or down) with digium's 877 number?
17:58.53MattHI get a fast busy signal
17:58.56Bhaalvaewyn: I have a radio show on Pulseradio (www.pulseradiogroup.com) its called Preparation Zero ... We have a different DJ each week etc...
17:59.13Bhaalvaewyn: Not restricted to free music or anything, its all dance related music
17:59.21vaewynBhaal: cool... I'll check it out
17:59.50vaewynBhaal: monkey still glowing?   ;P
18:00.04BhaalThat wasnt my show hahaha, that was Barbicane
18:00.11vaewynohh that's right
18:00.27BhaalI cant even remember what my show on FNR was called...
18:00.32vaewyngetting my nicks messed up
18:00.45Bhaalhehe
18:01.07BhaalAnyway Im off to bed, going to get a 3-4hrs sleep before heading into work...
18:01.18vaewynBhaal: have fun... nice seeing you again!
18:01.20HDG-laws|sl33pvaewyn So Polycom 600 over Snom 360?
18:01.26bjohnsonwoo hooo.  Intellicad is released for linux workstations
18:01.33Bhaalyup, I will be hanging around for a while..  chat to ya later
18:01.55vaewynHDG-laws|sl33p: For our application it made sense... especially with the microbrowser to play with
18:02.41HDG-laws|sl33pvaewyn This is for an insurance agency, I just hope I'm not stepping into quicksand doing this instead of having a PBX installed.
18:02.46HDG-laws|sl33p;)
18:03.30greg_workHDG-laws|sl33p: just watch those voip providers.. so many sketchy ones
18:03.43HDG-laws|sl33pI was looking at Broadvoice I believe it was
18:03.45HDG-laws|sl33pThey have BYOD
18:03.57Lee__only good for wholesale though.
18:04.32InfraRedrnk
18:04.39InfraRedurl ?
18:04.40HDG-laws|sl33pHow much of a pain is it going to be to setup a POTS failover with the SIP (whichever phone I end up with) and *
18:04.56greg_workwholesale is better than those packaged "unlimited" deal things
18:05.05vaewynHDG-laws|sl33p: Broadvoice isn't bad...  I recommend Nufone primarily though cause I have had less connection issues to them  (caveat: I have a short path to them so...  it may just be that)
18:05.07InfraRedand where are they based?
18:05.23greg_workie, you usually just pay a flat rate of somwhere between 1 and 2 cents/min .. if you don't use it, you don't pay anything
18:05.44HDG-laws|sl33pWould be nice if the IP Dialtone providers mentioned their connectivity so you would know if you had local peering or not
18:06.17greg_workif something happens and you get high latency to one provider, you switch to another. they also allow multiple concurrent calls, while those broadvoice-type services often don't
18:06.41wunderkinwell it looks like my audio problem was caused by my provider.. or their isp.. suckage.. :/
18:07.17greg_workyou're also not paying extra to get "features" like call waiting, caller id, voice mail etc..
18:07.28greg_workif you have *, it can do all that stuff. all you need is a raw pipe coming in
18:08.05Lee__rnk is $0.007/minute but you have to deposit $5000 up front.
18:08.26InfraRed5k
18:08.27InfraRedouch
18:08.37Lee__HDG-laws|sl33p: there is no dialtone with IP phones. It's a recording in the phone.
18:08.38jhiverwhat's 'rnk'?
18:08.48enderis there a page out there for using the web config of these Polycom phones for * use?
18:09.20mishehuender: it's pretty straight forward, as long as you're using the sip firmware...
18:09.28greg_worki use sixtel for some long distance.. i think i had to put down $10 and get 1.5c/min or so
18:09.29Lee__RNK will let you test their system for a whole day if you call them. it worked well here and had < 20ms latency.
18:10.26endermishehu: you'd think so, however there are some things which seem very ambiguous.
18:10.30greg_worki have a DID with another provider for $5 cdn /mo (and that's only because the number is not in a big city, where it would be $2.50/mo), and i pay 1.1c/min within their calling area
18:10.34mishehuender: such as?
18:10.40endermishehu: like sip configuration vs registration
18:11.02endermishehu: and Identification with a server address box as well as Server 1 w/ an address box
18:11.30greg_workto cmopare to broadvoice, i'd have to use over 2700 minutes of LD just to make it break even (more, and broadvoice is cheaper)
18:11.31endermishehu: and 'outbound proxy' with a server box and 'Server 1' with a server box under sip config
18:11.59greg_workof course, read the fine print with those 'unlimited' services ... they often consider "unlimited" usage to be ie, 3000 mins..
18:12.20mishehuender: if something is repeated, it is because you can set different line appearances to use a different config, but there is also a "central" config that will be used as default.
18:12.23greg_workand that's not even taking into account that they may not even support >1 concurrent call
18:12.42vaewynumm...  5000$/.007 is 714285 minutes... or 496 days...  egads that's a lot to prepay
18:13.20greg_workyeah, that seems more like actual wholesale targetted at a LD provider or something
18:13.28vaewyn*nods*
18:13.32greg_workwhereas the "wholesale" service I get is more like ... no-frills voip
18:13.33endermishehu: yeah, I'm having a hard time figuring out what is the 'central' one, I haven't got my phone to register correctly yet.
18:13.38vaewynwould work great for here at the univeristy :}
18:13.58endermishehu: is 'outbound proxy' the 'core' SIP server I should use (IE * )
18:14.19*** join/#asterisk fugitivo (~ajf@168.226.246.44)
18:14.19fugitivohello
18:14.26jhiverhi
18:14.54endermishehu: also there are a lot of radio buttons that have no label for each option.
18:14.59jhiverI agree with greg_work. 'Pay as you go' is as good as it gets with VoIP
18:15.15endermishehu: there is a Type: line w/ a radio button choice of 2, no label on what each of those two are, so how the heck do I know what I'm selecting?
18:15.39ManxPowerender: that's a well known bug in the polycoms.  No idea how to fix it, since I use config files for them.
18:16.30The_LightSidedoes anyone know where i can get a DSS console for * ?
18:16.35enderManxPower: ah ok.
18:16.38enderManxPower: I foudn it in the source.
18:16.42greg_workjhiver: well, there's just effectively two different classes of voip providers... the end-user targetting ones, and the *-user targetting ones.. services targetted at end-users who want to plug a sip phone into their router are just not the best fit with asterisk
18:16.45mishehuender: and I don't have a phone here that I can view at the moment...
18:16.49mishehuso I can't help really.
18:16.55mishehugot my hands full at teh moment.
18:17.04jhiveryep
18:17.40jhiveraltough if there was a kick ass IAX ATA this could change as you could start a end-user IAX VoIP provider service
18:18.15tclarkjhava: pa168V
18:18.17jhiverbut I think it would mean 'forget about unlimited plans' which isn't a bad thing anyways
18:18.20[TK]D-FenderHey, anyone know where I can get documentation on Polycom's Micro-Browser XML format?  Or at least some samples for em to play with?
18:18.45Lee__[TK]D-Fender: IIRC it's valid XHTML
18:18.46HDG-laws|sl33pgreg_work So you are saying using * for SIP isn't a good fit?
18:19.02greg_workHDG-laws|sl33p: no, not at all
18:19.07jhiverI guess the good thing about 'unlimited' plans when you're a provider is that it's pretty easy to bill
18:19.36jhiverbut then you have to keep an eye on your expenses anyways
18:19.38jhiversooo....
18:19.53greg_workHDG-laws|sl33p: but there's no point in getting a service that provides you with call waiting , voicemail, etc etc etc, when you're connecting it to *, which can provide voicemail, calll waiting, etc etc but on YOUR terms
18:20.12The_LightSidedoes anyone know where i can get a DSS console for asterisk@home??
18:20.14greg_workie-.. say you use a BV line for an incoming business line. if you use BV voicemail, your whole business gets one mailbox
18:20.22wunderkinwell.. the only reason you would want your provider to have voicemail is if you arent registered
18:20.34HDG-laws|sl33pgreg_work: Hmm, the more and more I read the more and more nervous I get ;O
18:20.43greg_workif you use *, instead of using BV you can have an autoattendant, and have individual mailboxes for every person
18:21.24greg_workwunderkin: true. but if not being registered is an issue (ie, you don't have a dedicated internet connection of some sort that has an SLA) then you shouldn't be relying on it for your primary phone line
18:21.43jhiverHDG-laws|sl33p: greg_work is telling you the right thing: you should probably worry about finding a solid, basic VoIP connection and leave all the fancy 'added value services' to asterisk
18:21.53jhiverso that you keep the 'added value' for yourself ;)
18:22.08HDG-laws|sl33pI'm not concerned about "value added services", I'm more concerned with providing my boss a "maximum cost per month"
18:22.16HDG-laws|sl33pGoing on a per usage leaves that "open-ended"
18:22.45jhivergarbage
18:22.57jhiverjust look at your phone invoices for the last X month
18:22.59greg_workHDG-laws|sl33p: i work at a small business, and here's what we do: we have 3 incoming POTS lines on a hunt group, that go into *. I also have a DID number that comes in over VoIP.. that DID is also in the hunt group (well, it will be shortly, i'm waiting for them to change it now)
18:23.40[TK]D-FenderLee__ : I didn't really give it a full XHTML header so I guess that could be it.  Do you have a sample file by any chance?
18:24.05greg_workHDG-laws|sl33p: outgoing LOCAL calls are made using two of our POTS lines (leaving one free for incoming), or if they're both busy, a VoIP provider. outgoing LD calls are done using a Voip provider first, and if that's busy, or our internet is down, it fails over to POTS lines
18:24.19*** join/#asterisk coldfeet (~cold@dsl-80-46-109-145.access.as9105.com)
18:24.22HDG-laws|sl33pjhiver: Charges are never the same, we have annual renewals so one month you will be talking to X client and X client's insurance company in X state and some months you won't. I could look at the most ever used and use that for a basis, but it is easier to tell them "look, $30/mo per line"
18:24.27Lee__[TK]D-Fender: nope, but I used google's XHTML page and it loaded.
18:24.44coldfeethi guys, anyone here using MYSQL in extensions, I have the following query, but it fails, cant see why
18:24.46Lee__hee hee. tiny  http://www.google.com/xhtml
18:24.47coldfeetselect\  a.reseller_id\ from\ subscriber\ s\,\ subscriber_add\ a\ where\ s.phplib_id\ =\ a.phplib_id\ and\ s.username\ =\ \'040600\'
18:24.50HDG-laws|sl33pgreg_work you sound like the guy I need to speak with, you are doing what I want to do
18:24.53Lee__ah!!!!
18:24.54greg_workeffectively, it means if our internet connection goes down, our call capacity just decreases to 3 conncurrent calls (on POTS).. we're not crippled, but people may get congested tones when trying to place calls
18:24.59HDG-laws|sl33p6 CO lines with POTS failover
18:25.19jhiverYeah but I don't see any business right in their minds doing 'all you can eat' for businesses
18:25.34jhiveralthough some of them do 'softcaps', I remember having seen one doing that
18:25.42greg_workimho, VoIP over the internet is just not reliable enough to depend on, ESPESSIALLY if you don't have a dedicated T1 or Fiber or better connection, with a tight SLA
18:26.00jhiverthey would look at the last average 3 month consumption and as long as it's below a certain level not overcharge
18:26.06Lee__yeah, get an SLA if your client can sue you.
18:26.26*** join/#asterisk WeezeyD (~ohno@206.210.111.115)
18:26.30greg_workLee__: well, i'm thinking more for customers.. if you can't make/get phone calls, it's pretty crippling to a business
18:26.42HDG-laws|sl33pWe aren't at capacity of requiring a T1, but we are given a five 9's SLA from Comcast I believe.
18:26.54HDG-laws|sl33pBut I wanted to have at least a 3 line POTS failover
18:27.00HDG-laws|sl33pIn the event of internet downtime
18:27.08HDG-laws|sl33pSo we could AT LEAST answer calls.
18:27.10*** join/#asterisk toot (~chris@212.20.250.186)
18:27.15greg_workVoIP for inter-office communications (between branch offices) is probably ok.. just make sure there's a backup way to communicate
18:27.42greg_workHDG-laws|sl33p: how many concurrent calls are you planning for?
18:27.46toothey anyone using festival ? is the quality good or any other recommendaions?
18:27.53WeezeyDI set up TDMoE, but callerid isn't being passed, does caller ID not work with TDMoE?
18:27.54[TK]D-FenderLee__ : I just saved Google's page over the one I was testing with and get another error
18:28.02HDG-laws|sl33p3-4 concurrent, we've maxed out the 6 lines quite often
18:28.06[TK]D-FenderLee__ : Could you pastebin your sample
18:28.16Lee__that /is/ my sample.
18:28.24Lee__Ploycom 600
18:28.34*** join/#asterisk Corydon76-home (three@Corydon76-home.sustaining.supporter.pdpc)
18:28.38[TK]D-Fendereek... I did google.ca, I'll try google.com
18:28.58jhiverhahaha, that rnkvoip 'phone for life' offer sounds just so bogus
18:29.00*** part/#asterisk vaewyn (freeman@mail.parrishmachine.com)
18:29.07greg_worktoot: i was mildly satisfied by one of the voices from festvox.org .. also tried cepstral, thought it sounded worse than the default festival voice
18:29.19jhiver'give us loads of money upfont, and you'll have phone for life until we're bust'
18:29.37greg_workHDG-laws|sl33p: in that case, i'd say use your POTS as your primary lines, and use VoIP for extra capacity
18:29.38mrgobytoot: festival is the best tts that is open source, imo...  quality is okay...  i had best results with mbrola voices
18:29.40*** join/#asterisk salvini_fs (~felipesal@200164050004.user.veloxzone.com.br)
18:30.01[TK]D-FenderLee__ : Dammit, all failures
18:30.02mrgobythe festival app though is unreliable in asterisk
18:30.09mrgobybest to use AGI workarounds
18:30.16HDG-laws|sl33pPots for both in/out or just for in?
18:30.26greg_workmrgoby: tried festvox.org? i use voice_cmu_us_bdl_arctic_hts .. didn't try or hear mbrola though
18:30.31enderhrm, I have a weird issue w/ my polycom.
18:30.40enderWe are using 4 digit extensions here, 100X for now.
18:30.47greg_worktoot: and I second the AGI thing.. i have a php script that does TTS for me, and it caches
18:30.53mrgobygreg_work, not sure if i tried that... will check though
18:31.05enderon my polycom, when I press '10' it immediately tries to do some sort of call, it fails, then I get my prompt back half a second later and I can continue putting digits in.
18:31.15endersomething is making the phone send something after 10 is hit.
18:31.19mrgobythe approache in php_agi is a good one
18:31.34HDG-laws|sl33pgreg_work: I don't follow
18:31.40mrgobyin regards to festival caching
18:31.47*** join/#asterisk vaewyn (freeman@mail.parrishmachine.com)
18:32.07greg_workHDG-laws|sl33p: for both .. but really, i've been fairly happy with my setup. it's a simle LCR (least cost routing) config for outbound
18:32.24vaewynBhaal: Southern Exposure was yours :}  thank goodness for irc logs :P
18:32.28greg_workLD goes over VoIP (where it's cheaper permin) while local calls go over POTS first (since it's 0/min)
18:33.09greg_workbut if all POTS are used up, it uses VoIP, if VoIP is down (for LD) it uses POTS, and if all POTS are used up and voip is down, they just get a busy signal
18:33.13HDG-laws|sl33pgreg_work How do I work the SIP in with the POTS, I've gotten confused ;o
18:34.20greg_workHDG-laws|sl33p: well if you use a config like AMPortal, then there's a whole section for configuring how calls are routed and over which trunks (disclaimer: i wrote the outbound routing stuff in amp ;) )
18:34.26*** join/#asterisk loick (~loick@ATuileries-151-1-19-219.w82-123.abo.wanadoo.fr)
18:35.04HDG-laws|sl33pgreg_work But there would be no issues having all phones in the new office be SIP ?
18:35.19greg_workBut basically you make a dial macro .. like  [macro-dial]  1,Dial(Zap/g0/${ARG1})   102,Dial(SIP/provider/${ARG1})
18:35.38HDG-laws|sl33pJust a second, let me get out my Greek dictionary
18:35.41greg_work.. if the first dial fails it moves to the next one ..
18:35.42greg_workheh
18:35.45*** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net)
18:35.49milkyflavahello
18:35.50HDG-laws|sl33pI understand the concept of the failover
18:36.08milkyflavaCan someone recommend a good softphone that uses IAX?
18:36.10HDG-laws|sl33pI'm just afraid I don't have the entire hardware list or know exaclty what I need to make this come together
18:36.13greg_workAMP (a web gui / config for *) does all that stuff.. but if you really want to do it by hand then you need to read up on dialplan stuff
18:36.19Lee__milkyflava: no
18:36.23milkyflavalol
18:36.41greg_workHDG-laws|sl33p: well, as far as in the office, it depends on what you want to do, and how your network is configured
18:37.22milkyflavaThanks Lee__, I'll get SIP through NAT then.
18:37.41The_LightSidepossibly iaxcomm... very basix though
18:38.05greg_workHDG-laws|sl33p: here, we use all SIP phones. I have at least two cat5's running to every desk, and the voip phones are on a physically different switch than the computers
18:38.07vaewyniaxcomm works well
18:38.19HDG-laws|sl33pSo you don't use the internal switching
18:38.35greg_work(though they are connected .. i don't really know if theres a huge benefit to that or not.. but at least the majority of the traffic is seperated.. i've never had an issue with not enough bandwith on the local network)
18:38.50greg_workno, my phones don't even have switches
18:38.55shido6poe switches on ups?
18:39.09HDG-laws|sl33pAh, the phones I'm looking at have internal switches to connect PC or networkable printer to
18:39.16HDG-laws|sl33pSo I can avoid extra runs of cat6
18:39.21HDG-laws|sl33por 5e
18:39.23Kattymew
18:39.29greg_workwe just moved in here 7 months ago, so i had the benefit of wiring it. did it all with multiple cat5e's
18:39.29JerJermooo
18:39.32*** join/#asterisk inspired (mikael@213.197.167.61)
18:39.41Kattysuddenly, i'm all inspired.
18:39.45greg_workwhen you wire a building, cat5e becomes dirt cheap
18:39.58HDG-laws|sl33pI can wire it all up, I just don't know what I'm doing as per running the wiring thru the ceilings and down the walls ;o
18:39.59greg_workprobably cheaper than the extra money for voip phones that have switches in them ;)
18:40.08shido6Lord
18:40.17HDG-laws|sl33pThe voip phones that have switches in them also seem to be the same phone with 6 lines ;)
18:40.20vaewynheck yeah...  'good' cat5e is 70$/1000ft
18:40.24shido6I did Detroit Public Schools , wiring buildings sucks
18:40.25vaewyndirt cheap
18:40.31shido6esp OLD buildings
18:40.32robl^cat5e!?!?  what?!?! no one still uses thinnet 10Base2 RG59 coax for networking?
18:40.37HDG-laws|sl33pshido6: Where are you? I am in Hamtramck!
18:40.46shido6Im in Windsor right now
18:40.47Lee__HDG-laws|sl33p: Snom and Cisco phones with switches are good.
18:40.48vaewyn~lart robl^
18:40.54greg_workshido6: it's ready for PoE, but i didn't use it because i have spa-841's and they removed the poe feature (grr)
18:40.57shido6HDG-laws|sl33p, NuFone is a michigan based company you know?
18:41.03The_LightSiderobl^: dont u mean RG58?
18:41.07HDG-laws|sl33pLee___ I was prepared to buy Snom 360's today and now I'm backing off that and looking to Poly 600's
18:41.10mutilatorisn't everything
18:41.15HDG-laws|sl33pshido6 I did not know that, interesting
18:41.16greg_workI do have an ATA in the server room on UPS though, that powers two analog phones (one cordless)
18:41.29The_LightSidemy bad!
18:41.29milkyflavaDarthClue, You awake?
18:41.32robl^The_LighterSide, whats a few RG's between geeks? :)
18:41.37The_LightSidelol
18:42.18greg_workisn't rg59 the really cheap crappy coax cable they used to use for tv?
18:42.24greg_worksadly people probably HAVE used that for thinnet
18:42.25SpaceBasspretty much
18:42.36The_LightSidecheers for now! catch u all later!
18:42.36SpaceBassrg6 is the only way to go for digital cable or satelite
18:42.55SpaceBasshey... 14mb tolken ring was much better than 10baseT
18:42.58vaewyn*nods* rg6 or rg11 on really long runs
18:43.13greg_workHDG-laws|sl33p: so are you wiring from scratch then?
18:43.23SpaceBassbut then again, I really liked working on as400s too :)
18:43.24robl^Tolkien Ring?  the ONE RING ethernet :)
18:43.28HDG-laws|sl33pgreg_work THere is wiring in place that is going BYE BYE.
18:43.38HDG-laws|sl33pOLD cat5
18:43.56SpaceBasswhats the typical rate for hireing someone for structured wireing these days?
18:44.20jhiverhey, what's asterisk packetization time when using SIP?
18:44.28greg_workhm. not a clue.. it's really not that hard though, probably not worth the money :p
18:44.33greg_worki'd imagine $50-70/hr
18:44.42greg_workthat's a lot to pay somenoe to pull cable
18:44.48SpaceBassyeah it is
18:44.53HDG-laws|sl33pI have someone willing to run the cable for $85/run
18:44.58greg_workhell, even pull the cable yourself and then get someone to come in and terminate it all, if you don't know how to do that
18:45.01Lee__get an intern.
18:45.15SpaceBassthinking about homes really... if you don't do it yourself its hard to find someone to wire cat5 in a house
18:45.25jhiverok I have my answer: 20ms
18:45.41greg_workbut really.. invest in a decent crimp tool ($50+), a bag of rj45's (you'll use 3-4 per cable when you first start) .. and its pretty simple
18:46.00greg_workHDG-laws|sl33p: i also highly recommend a patch panel
18:46.13greg_workHDG-laws|sl33p: this is mine: http://office.mwater.ca/files/office/patchfront.jpg
18:46.25tootwill give it a good - thanks for the comments
18:46.48SpaceBassI really needed a patch pannel... I ran everything stright to the switch
18:46.56HDG-laws|sl33pgreg_work Patch panel is a guarantee
18:47.33HDG-laws|sl33pgreg_work I know how to do all that stuff, it is the running of the wires down walls that I'm "uncertain of"
18:47.37greg_workwith RJ45, you can plug RJ11 in (ie, analog phone lines) .. since I have cat5e running everywhere, it's very simple to add an analog phone line (ie, for a fax/interac machine, or from my SIP ATA adapter to the analog cordless phone) anywhere
18:47.48HDG-laws|sl33pgreg_work As well as exactly what hardware I need to do this cross between VoIP and POTS
18:47.48SpaceBassHDG-laws|sl33p what are you wiring?
18:48.17mutilatorA BOMB!
18:48.22SpaceBassRUN!
18:48.31tsumethey set us up the bomb
18:48.35greg_workHDG-laws|sl33p: drop ceiling? most places are constructed with the tops of walls open .. ie, there will be studs that go ALL the way to the top of the roof (usually 1-4' above the drop ceiling)
18:48.40tsumeall your base are belong to us
18:48.58greg_workand the drywall will just go a few inches above the drop ceiling and stop, leaving an open gap .. easy to drop down wires
18:49.04Lee__has someone compiled CVS HEAD today?
18:49.14jhiverHey guys
18:49.16SpaceBassgreg_work too bad houses aren't built that way! wiring b/t floors is a pain!
18:49.18Lee__cause my update seems to be missing a directory.
18:49.22greg_workSpaceBass: yes :)
18:49.29jhiverAny ideas what UDP range is asterisk using when using SIP?
18:49.51Lee__outtolunc: yes, it is missing a directory?
18:49.51SpaceBassjhiver for rtp its configured in rtp.conf
18:50.20outtolunciirc there were a couple for gnophone and others
18:50.26greg_workHDG-laws|sl33p: to get POTS lines into your * box you need something with fxo ports on it .. like a TDM400 from digium (... avoid, imo) or an ATA ("analog telephony adapter" ... i think)
18:50.42jhiverCheers
18:50.54SpaceBassand ATA is an fxs not an fxo :)
18:50.57Lee__make: build_tools/mkdep: Command not found
18:50.58SpaceBassclear as mud?
18:50.58HDG-laws|sl33pgreg_work So I'd either have to run a digium card out of the * or use ATA's?
18:51.14greg_workSpaceBass: is it? i didnt know that
18:51.22*** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net)
18:51.27Lee__the Sipura has an FXS and FXO
18:51.36greg_workwellgate makes a 4-port fxo device that talks SIP
18:51.37Lee__as to some big media gateways
18:51.41greg_workyeah, i think it is an ATA still
18:51.44Lee__*do
18:52.07DaminWiiwiiwii...
18:52.09greg_workHDG-laws|sl33p: i have a TDM400P with 4 fxo ports, and personally, i wouldn't buy one again
18:52.18HDG-laws|sl33pgreg_work eek!
18:52.21greg_workit's crashed on me about 5 times now (in 6 months)
18:52.21obsidian-studiosgreetings, anyone got any suggestions for keeping CID/ANI info off a caller, that say calls, does not leave a message, and does not reach a person via an extension? Ideally a email report or something?
18:52.56outtolunclee: didn't see that, and my 5000 line buffer only has from preprocess to the end
18:53.01ManxPowergreg_work: Did you report the problem to Digium?
18:53.07outtolunc(i've been busy since then)
18:53.15greg_workManxPower: no, because i wasn't running the lastest driver at the time
18:53.32ManxPowerobsidian-studios: Perhaps if you reworded your question, you might get an answer.
18:53.35SpaceBassobsidian-studios you could use AMP and the reports it generates... export to PDF... im sure there is a way to automate it and use sendmail
18:53.38obsidian-studiosgreg_work: FYI, I have a TDM400P in a * deployment where a hurricane took out the old pbx system, and so far the TDM400 card has worked perfectly
18:53.53obsidian-studiosSpaceBass: cool
18:54.09greg_workit just suddenly scrolls "TDM PCI master abort" over and over on the local console, and otherwise totally freezes. cold reboot is the only fix
18:54.17*** join/#asterisk Skarmeth (~Skarmeth@201009024136.user.veloxzone.com.br)
18:54.30obsidian-studiosI seem to have people calling in and hanging up, trying to get info on them, any suggestions?
18:54.38obsidian-studiosManxPower: is that worded better :)
18:54.39greg_worki think it's the drivers and not the hardware.. but the bottom line is i don't totally trust it
18:54.44*** join/#asterisk durex (~ironman@weber.anpa.org.br)
18:54.52greg_workobsidian-studios: caller id
18:55.35obsidian-studiosgreg_work: it's the caller id info I need in an accessible place, as I am talking about someone who calls, listens say to the main menu and hangs up
18:55.54obsidian-studiosI guess I could keep a regular phone plugged in as well and look at it from time to time
18:56.13obsidian-studioswas inquiring on a more elegant solution, but I do not really need all the info in a cdr
18:56.14greg_workobsidian-studios: oh. well, use CDR
18:56.23SpaceBassobsidian-studios I still suggest amp and its reports
18:56.24Lee__I guess I just needed to checkout the CVS again. It got the missing directory.
18:56.29obsidian-studiosgreg_work: thought a cdr would have to much info
18:56.43obsidian-studiosSpaceBass: cool,  I will go research and check it out now
18:56.47greg_workobsidian-studios: or put  Noop(caller id is ${CALLERIDNUM}) in the dialplan .. it will get logged (if you have logging on)
18:56.59greg_workand you can   grep "caller id is" /var/log/asterisk/full
18:57.07obsidian-studiosgreg_work: yeah I know I can log it, but then I have to look at logs etc
18:57.26obsidian-studioswould be totally cool if I knew someone was in the system before they enter my extension or etc
18:57.28*** join/#asterisk pa (~Paolo@pa.user)
18:57.32paHello!
18:57.33pawow
18:57.38greg_workcrontab -e      59 23 * * * grep "caller id is" /var/log/asterisk/full
18:57.41paan IRC channel for Asterisk!
18:57.49greg_workthen you'll get an email every night (assuming you rotate that file nightly)
18:57.51obsidian-studiosgreg_work:  :) nice
18:58.02obsidian-studiosgreg_work:  any ideas for real time
18:58.21greg_workuse FOP
18:59.00greg_workor use CDR and write a two-second web app that does SELECT callerid, date FROM cdr ORDER BY date DESC LIMIT 10
18:59.07paI have a simple question (then i go to RTFM): Can i call someone by phone with Asterisk not being at home on the machine with ISDN card, but from remote over TCP?
19:00.02*** join/#asterisk meshuga (meshuga@c-24-20-154-158.hsd1.wa.comcast.net)
19:00.03pathat is: ME -- TCP --> HOME + Asterisk + TA ISDN -- PHONE NETWORK --> Called
19:00.35greg_workpa: yes, you just have to use a SIP or IAX phone and open the approate ports on the firewall to let it connect to asterisk
19:01.04meshugaso, i recently switched from ztdummy to a digium wildcard as a timing source, now i cant hear anything in VM. is there configuration changes I need to make to tell asterisk to use the new timing source?
19:01.06obsidian-studiosgreg_work:  why do you not like the TDM400 cards?
19:01.19meshugathe console acts like its playing
19:01.24meshugabut you cant hear anything thru my sip clients
19:01.33pagreg_work: i think they are VoIP phones.. there are linux applications that act like a such voip phone using microphone and speakers or earphones?
19:01.33Skarmethhi all, I've a simple ask about Asterisk and X100P cards... I'm searching for it about two days without sucess... I've a extension from my analog PBX connected to it and I want to make calls throung it using my SIP/AIX2 soft phones.
19:01.58SkarmethHow I can allow the soft phone dial the destination number?
19:02.04SpaceBasspa tons... iaxcomm and i think one for gnome are peopular
19:02.05meshugapa: iaxcomm woulda
19:02.06greg_workobsidian-studios: getting a call on my cell phone early in the morning "greg, the phones are not working! what do we do!!" and finding out that it's scrolling "TDM PCI master abort" on the console, not responding on the network, and requires a reboot
19:02.07meshugaworks
19:02.13meshugagnomephone doesnt work at all
19:02.24greg_workthose calls - espessially when they wake me up - put me in a bad mood
19:02.25pathanks a lot! going to RTFM :-)
19:02.29SpaceBassSkarmeth did you set up the cards on the box using the ztcfg tool?
19:02.35Skarmethusing a fixed extension, I can complete the call, but not allowing the shoft phone tell it
19:02.36pabut first add #Asterisk to list :-)
19:02.39greg_workand i've had about 3 of them since i started with *. not impressed at all
19:02.41obsidian-studiosgreg_work: what kernel and etc? I see really good uptime and etc with the TDM400 at a clients, with allot of abuse?
19:02.54obsidian-studiosgreg_work: wow
19:02.54greg_work2.6 kernel, * stable
19:03.02greg_work2.6.8 to be exact
19:03.04obsidian-studiosgreg_work: distro?
19:03.12greg_workdebian sarge
19:03.19obsidian-studioshmm, might be env?
19:03.33obsidian-studiosI run Gentoo, and bet I am using newer version of everything than you?
19:03.34meshugagreg_work: i havent had good luck with tdm400s either
19:03.38SkarmethSpaceBass, it's working, I can get call coming from it and transfer to SIP/IAX2 clients
19:03.45meshugagreg_work : sangomas work better, IMO
19:03.47Lee__newer != better
19:03.59obsidian-studiosgreg_work: running 2.6.11 or 2.6.12 kernels ATM
19:04.00meshugabut i'd just use a media gateway really
19:04.19meshugaonce i get to multiplexing in 4 t1s, i don't like trusting something that if it fails the whole pbx goes down to change.
19:04.23greg_workobsidian-studios: my ztcfg binary is dated 2005-06-09
19:04.26obsidian-studiosgreg_work: could be what you are seeing is weirdness, incompatibility, or instability due to other things?
19:04.32Skarmethmy /etc/zaptel.conf looks like: fxsks=1 newline defaultzone=us newline loadzone=us
19:04.41mutilatorit possible to start recording calls from the cli?
19:04.50meshugamutilator : no
19:04.56greg_worki WAS using a slightly older (december 2004) one until then though
19:05.08greg_worki'm actually not sure if it has crashed since then
19:05.09obsidian-studiosgreg_work: yes, but the gcc it's using, glibc, kernel calls, kernel headers, etc might be older? So it's more what your current stuff is compiled against
19:05.12mutilatorjust manager interface & extensions then
19:05.22meshugaexactly
19:05.30obsidian-studiosgreg_work: only time I have seen * stop working or fail, is when I have done something
19:05.32greg_workat the same time though, i had to resort to using a cron job to  unload and reload the wctdm driver every night
19:05.36Nuximanager scripts can be called from the cli.
19:05.38meshugawell, or you can ethereal and pick up the RTP stream ;)
19:05.47greg_workobsidian-studios: the kernel header is upto date
19:05.56greg_workgcc                   3.3.5-1
19:06.04obsidian-studiosgreg_work: something is not right, but I am not sure i would blame * or the drivers, or hardware without looking at all vars
19:06.18meshugathose tdm400p's are very static prone.
19:06.19*** join/#asterisk meppl (mephisto@p54AABE80.dip.t-dialin.net)
19:06.25greg_workii  libc6                 2.3.2.ds1-20          GNU C Library: Shared libraries and Timezone data
19:06.26meshugai've rma'd 2 or 3
19:06.40meshugai just stick with a maximum of 2pt sangomas these days
19:06.47meshugaand media gateways for anything higher
19:06.50meshugauh
19:06.57meshugaNuxi: none of that stuff has 'cvs heads'
19:07.15SkarmethSpaceBass, can you send me a example extension to make calls throung X100P?
19:07.16Nuxioh, well, the latest and greatest of each.
19:07.18meshugaand plus, that'd be not smart to use development gcc and glib and whatnot.
19:07.25meshugaNuxi: incorrect sir
19:07.26greg_workobsidian-studios: well, the actual system it's running on is the same as a couple other servers i have
19:07.28SpaceBassSkarmeth think there is one on the wiki
19:07.31SpaceBass~wiki
19:07.35obsidian-studioswell I am glibc 2.3.4.20041102-r1 and gcc 3.4.2-r2 on the * box
19:07.51meshugaSkarmeth : there is, look for 'development kit lite' instructions, it used sample configs for the x100p
19:07.58Nuxiwow, thanks meshuga, I've been knighted.
19:08.02greg_workwell, i dunno
19:08.03obsidian-studiosgreg_work: I would start looking to the * developers and the env they use
19:08.22greg_worki would rather just spend an extra $200 or whatever to get a media gateway
19:08.23*** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk)
19:08.28meshugaNuxi: the libc5 and glibc2 issues come to mind, plus of course gcc 2.95 and gcc 3.* as well.
19:08.40meshugagreg_work : not even that much more these days
19:08.59obsidian-studiosgreg_work: are you pulling * from CVS? Or using Debian binaries?
19:09.06greg_workcvs
19:09.17obsidian-studiosgreg_work: when I left RH I attempted to go Debian but had way to many problem on stable
19:09.18meshugai've been using octtel media gateways
19:09.20meshugato decent sucess
19:09.29meshugadebian stable == 3 years old
19:09.33obsidian-studiosgreg_work: kernel stuff on machines that ran the same kernel config and etc under RH for years
19:09.38greg_workmeshuga: well, not anymore ;)
19:09.39obsidian-studiosgreg_work: horrible
19:09.43meshugatrue, as of last week?
19:09.44greg_worki've only ever used sarge though
19:09.50greg_workmeshuga: nah, 3 or 4 weeks now
19:09.55MRH2anyone experince backgound noise fading in and out (when someone speaks) when you call noisy environments
19:09.55meshugaah, i was off abit
19:09.57obsidian-studiosgreg_work: you are working from current cvs I bet coming off developers machines using much more current stuff than you
19:10.03meshugai'm pretty done with debian
19:10.22brookshireasterisk compiles just fine on debian
19:10.23obsidian-studiosgreg_work: to pull from cvs, you need to make sure you are compiling against what the developers are, or you will get unexpected, unreliable, unstable results
19:10.26mishehuMRH2: that probably has to do with your phone or software.
19:10.26brookshiredid you use the wiki?
19:10.41greg_workmeshuga: its just a mind thing. debian stable (with woody, anyway) really was STABLE.. but it got old
19:10.48meshugabrookshire : we didn't say it didnt, hes having problems with tdm400p and debian.
19:10.53greg_workgreat for servers, where you get it working and don't want to screw with it (besides security updates)
19:10.55obsidian-studiosbrookshire: the problem is greg_work: is seeing problems with * and unreliability I have not see
19:11.02meshugagreg_work: slack is still better ;)
19:11.03greg_workbut effecively, you can use testing and it's quite safe
19:11.11brookshiremeshuga: did you call digium support? or email them?
19:11.18greg_workobsidian-studios: it's not *.. it's zaptel drivers
19:11.25MRH2I have tried different combos with the same result though
19:11.26meshugabrookshire : its not my problem. why would i do that?
19:11.28*** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl)
19:11.30obsidian-studiosgreg_work:  in my case it was all servers I had problems with Debian, BIND servers would kernel panic
19:11.43obsidian-studiosgreg_work: it's all related
19:11.48meshugahahaha bind would kernel panic?
19:11.51obsidian-studiosgreg_work: * the drivers, etc
19:11.52meshugayou got to be kidding.
19:11.54greg_worki have a lot of debian servers out there, and i trust it 100%
19:12.05robDoes anyone use OS X as their primary Asterisk server platform?
19:12.12greg_worki don't think i've had any of them kernel panic in production
19:12.16MRH2and am not using noise suppression, echo cancellation or neting like that
19:12.19vaewynOhh.. I hope not rob...
19:12.21obsidian-studiosmeshuga:  it's like this, I took a kernel config from RH, ran make oldconfig etc under debian, compiled the kernel, ran it, and after a few days I got a kernel panic
19:12.27meshugarob : my benchmarks indicated not being able to push out more then 50 calls
19:12.30meshugaon a g5
19:12.33meshugaconcurrently
19:12.34obsidian-studiosso I tossed Debian, and went Gentoo, will never look back
19:12.40vaewyn:} within an order of magnitude
19:12.47meshugarob: plus, it lagged in weird places. i wouldn't use it as a primary.
19:12.48robI'm just looking at it running locally for something for my home single phone.
19:12.56meshugarob: use a wrt54g?
19:12.57brookshirei've never had a problem with debian and zaptel either, oh well :/
19:13.02robI'm just having some crazy issues with getting my phone to work with it properly.
19:13.03greg_workoh yeah.. thats the other big thing. for the extra money a media gateway costs compared to a TDM400p, not having to screw around with echo canncelatoin settings would easily make up for it
19:13.24greg_workmine STILL echos from time to time, and i've spent hours now messing with it
19:13.26meshugagreg_work: plus, the ability to swap in a spare in <20 seconds
19:13.28meshugais ideal
19:13.30greg_workyes
19:13.30robmeshuga: Well, currently, I have an Asterisk server in a datacentre, which is using an IAX tunnel to get to me.
19:13.42obsidian-studiosgreg_work: allot of things in your env does not sound right
19:13.44meshugarob: wrt54g will be iax tunnels.
19:13.46*** join/#asterisk maik (~maik@bfs.cs.uni-sb.de)
19:13.52*** join/#asterisk rajo_ (~rajo@bfs.cs.uni-sb.de)
19:13.53meshugaer will do.
19:13.55robmeshuga: And then a local asterisk, which has my 7960 connected.
19:13.55greg_workobsidian-studios: specifics?
19:14.00obsidian-studiosgreg_work: I would recommend not using CVS unless you are running Debian unstable
19:14.13robmeshuga: Ah, I see. I'm just wondering whether I've found a specific bug in the OS X version with some phone models, but I'm unsure.
19:14.16greg_workobsidian-studios: i'm pulling from cvs, but i'm using *-STABLE.. not head
19:14.26brookshire:)
19:14.33JerJerobsidian-studios:  there is no need for anyone to run STABLE
19:14.37obsidian-studiosanything in CVS is going to be compiled against newer stuff I would bet
19:14.41meshugarob: some phone models? very doubtful. the sip implementation is solid. is your 7960 running latest sip?
19:14.42JerJerHEAD is far superior code
19:14.52brookshireyeah.. head is where it's at
19:15.02obsidian-studiosgreg_work: well it's like this, your problems are weird, and I would look else where
19:15.05greg_workJerJer: except its too much work to follow to be comfortable to use
19:15.11robmeshuga: Yep. it's just that at the moment, can't get audio source d at the 7960 to go anywhere else.
19:15.16JerJergreg_work: ?
19:15.18greg_workJerJer: ie, how do i know if i pull HEAD today that it will be suitable for production use?
19:15.19obsidian-studiosgreg_work: before you bash *, the drivers, or hardware you would have to rule it all out
19:15.19brookshiregreg_work: how so?
19:15.24brookshireit's one command
19:15.25JerJeryeah how so
19:15.26brookshiremake install
19:15.32obsidian-studiosgreg_work: I bet if it was Gentoo, most if not all would go away
19:15.38meshugahahha
19:15.39obsidian-studiosor if you were using newer stuff in your env
19:15.47JerJerobsidian-studios: asterisk doesn't give a shit about the distro you run
19:15.53meshugaobsidian is hilarious
19:15.59robmeshuga: I've had a friend around and we had it fine running on a desktop machine, however, it's a little broken using OS X.
19:16.06obsidian-studiosgreg_work: FYI, the * stuff in Gentoo is not as up to date as * in general, because the Gentoo guys like the test, get feedback, fix stuff etc
19:16.18greg_workare there any plans to release a 1.1 anytime soon anyways?
19:16.21JerJerdon't ever run a distro version of asterisk
19:16.25meshugarob: yes, its quite broken. the fact that apple keeps breaking backwards compatibility with every point release has alot to do with it
19:16.26jfonsecausaeasy to updte from debian -asteriskk 1.0.5 to debian Asterisk 1.0.9? Which best practice CVS-HEAD or Stable from Digium?
19:16.28JerJercheck it out of the cvs yourself
19:16.35brookshire1.2 is the next planned release
19:16.36obsidian-studiosJerJer: aything you compile cares about the version of gcc, glibc, kernel, kernel headers, give me a break, go program a gi
19:16.37meshugaobsidian-studios : the founder also is a microsoft employee.
19:16.53JerJerobsidian-studios:  all of which has nothing to do with the specific distro you run
19:16.56meshugaobsidian-studios : actually, i disagree.
19:16.57brookshireuse CVS-HEAD for now
19:17.00brookshireuntil 1.2
19:17.06brookshirethen after that, who knows :)
19:17.13meshugaobsidian-studios : i havent had a gcc problem since 3.* became mainstream
19:17.13obsidian-studiosI know I have had apps break across version of gcc, glibc and etc
19:17.16greg_workwell see, is HEAD working today?
19:17.18robmeshuga: Ah, so generally it's not too reliable? I figure I might try and route some IP space to my phone so that it's not behind NAT or anything then, and abandon the IAX tunnel.
19:17.20JerJerthere is nothing stopping you from upgrading gcc in a really old version of debian
19:17.21meshugasame with glibc2 hitting mainstream
19:17.25JerJeror the kernel
19:17.25brookshireHEAD usually works
19:17.29obsidian-studiosthere are differences in gcc 3.3, 3.4 and 3.5
19:17.31meshugarob: use a wrt54g over a os x box.
19:17.34greg_workbrookshire: so how do you know?
19:17.41obsidian-studiossuch that when using things like distcc things will majorly break bad
19:17.42brookshirebut then again.. let's not talk about 1.0.8 stable not working
19:17.45JerJerobsidian-studios:  and what does this have to do with what distro you run?
19:17.48robmeshuga: I'll have to source one, thanks for your help.
19:17.52meshugaobsidian-studios : and i havent personally witnessed one package that had a problem
19:17.53greg_worki mean ,say today there's some obscure bug?
19:17.54obsidian-studiosJerJer:  it;s all about your env
19:17.58meshugarob: they are <$70 on froogle
19:18.09obsidian-studiosJerJer: we are talking an open source app built on other open source dependencies
19:18.10JerJerobsidian-studios:  ok so why in hell do you think gentoo would run any different than debian?
19:18.12obsidian-studiosyou got to resolve all
19:18.16JerJerok and ?
19:18.16brookshiregreg_work: i just do ;)
19:18.27obsidian-studiosJerJer:  using much newer software, much more current with all project release
19:18.27JerJerthis has nothing to do with what distro you run
19:18.32JerJerasterisk does not care
19:18.48greg_workbrookshire: because you're a dev, or follow development/cvs updates?   i would love to be able to, but i simply don't have time to do it
19:18.49obsidian-studiosprograms are developed and compiled and tested in certain env
19:19.05obsidian-studiosif you are not using at least the same versions of everything or newer you will have problems
19:19.08obsidian-studiosand weird ones
19:19.17obsidian-studioswe are talking about differences on the asm levels at time
19:19.22greg_workobsidian-studios: not necessarily
19:19.26JerJeragain this has nothing to do with a specific distro
19:19.27greg_workand in fact, not likely
19:19.43obsidian-studiosgreg_work:  well I have never ever had the type of unreliability and etc you speak of
19:19.48obsidian-studiosgreg_work:  and I am in FL
19:19.56greg_workyes no one has, afaik
19:19.59obsidian-studiosgreg_work: it rains daily, we have all kinds of lighting, surges etc
19:20.04greg_workthuogh i don't know what location has to do with it
19:20.06obsidian-studiosgreg_work: so listen
19:20.17obsidian-studiosgreg_work: weidness like that will come from your env
19:20.33obsidian-studioswhen I use distcc and one machine is out of sync with gcc and glibc my binaries turn to crap
19:20.40JerJerdon't use distcc
19:20.57obsidian-studiosit's very very important when working with C and C++ stuff you use the same env or similar stuff, or newer
19:21.03obsidian-studiosheaders, libraries, etc
19:21.08JerJerno
19:21.20obsidian-studiosI am a programmer and deal with this stuff daily
19:21.35JerJeri've ran asterisk on Linux based every environment i can find it it simply works
19:21.37obsidian-studiosask other * programmers, I am not a * programmer but I bet it's no different for them
19:21.48paexcuse me, where can i find source for iaxcomm? i have linuxppc and there isn't binaries for linuxppc
19:21.58obsidian-studiosJerJer: ok so you explain why greg_work:  has problems no one else has? Or a resolution or etc?
19:22.04JerJerobsidian-studios:  i am an asterisk contributor
19:22.16obsidian-studiosJerJer: not contributor, developer
19:22.28obsidian-studiostalk to drumkilla and ask
19:22.30JerJerum i contribute code, dumbass
19:22.32greg_workbecause there's a difference?
19:22.58obsidian-studiosJerJer:  contribute, test, are responsible for, paid to develop etc is not all the same
19:23.09JerJerwhat?
19:23.13JerJergo anony someone els e
19:23.13paok found :)
19:23.18obsidian-studiosI have contributed to many projects, but do not know them like the orginal developers
19:23.34obsidian-studiosor people maintaining or responsible for tracking down bugs, fixing them etc
19:23.49shido6clueless
19:23.50greg_workout of curiousity, JerJer what's your opinion on using a TDM400 vs external media gateway?
19:23.55obsidian-studiosJerJer: bottom line I am trying to help greg_work:
19:23.56JerJerobsidian-studios:  ok and your point is?
19:24.03JerJerTDM400 any day
19:24.08obsidian-studios***Nuxi  JGenerator and others I did not request to be on the project
19:24.23obsidian-studios<PROTECTED>
19:24.31JerJerobsidian-studios:  find a clue
19:24.38The_LighterSideanyone know anything about a software DSS panel?
19:24.41obsidian-studiosJerJer: ask drumkilla
19:24.45file[laptop]break it up.
19:24.50greg_workThe_LighterSide: FOP (asternic.org i think)
19:24.51obsidian-studiosJerJer: drumkilla maintains the 1.* branch and etc
19:24.58The_LighterSidethanks :)
19:25.00MikeJ[Laptop]find a clue?
19:25.32file[laptop]at Cluecon!
19:25.34file[laptop]~cluecon
19:25.34jbotrumour has it, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
19:25.48JerJerobsidian-studios:  i don't run STABLE for obvious reasons
19:26.08obsidian-studiosbottom line I do not consider the TDM400P cards to be crap, and I chimed in because greg_work:  was mentioning it, and problems I have never ever ever heard of
19:26.36obsidian-studiosso I was trying to help out, and before I called Digium or said their cards, drivers or etc was junk, I would fully research, test and etc
19:27.25MRH2is there any recommendation enabling / disabling  /or no real difference  hyperthreading on a machine running  asterisk.
19:27.33greg_workobsidian-studios: i never said they were junk, i just said based on my experience i don't trust them, and wouldn't recommend them
19:27.38crash3mfragmentation of a SIP register packet would cause issues right?
19:27.49obsidian-studiosgreg_work: yes I know but you have to look at why or find a reason
19:28.06obsidian-studiosgreg_work: I would bet $ that if you say switched to Gentoo that problem would go away,
19:28.08crash3mMRH2: I thought I heard multithreading was found very insecure recently
19:28.25greg_workobsidian-studios: well, thats out of the question, and frankly, a rediculus fix
19:28.27obsidian-studiosgreg_work: I would love to be wrong and the problem exist regardless of env
19:28.46Assidhrmm.. do you guys suggest using CVS for production?
19:28.47obsidian-studiosgreg_work: well then go find out what the min gcc, glibc and etc the * developers recommend
19:28.49brookshirealways multithread
19:28.57greg_work1, i don't know gentoo.. 2, this is a production server that runs a critical system
19:28.57JerJeri have countless TDM400P based systems and have never had a single problem with FXS
19:29.03*** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
19:29.05obsidian-studiosgreg_work: as I bet there is something in your env specifically causing the weirdness
19:29.11greg_worki'm not switching OSs
19:29.19greg_workJerJer: this is 4-port fxo
19:29.31obsidian-studiosgreg_work: I run production critical * deployments on Gentoo with no problems
19:29.35coldfeetdoes nayone know howto use variables in a mYSQL query in asterisk...
19:29.48coldfeet...if ${name} =paul
19:30.02coldfeethow would you incorporate this in a where statement
19:30.04obsidian-studiosgreg_work: I have had a TDM400P in use for close to a year now without any problems, doing all kinds off stuff
19:30.17greg_workJerJer: i have intermittent echo issues, but the biggest issue is that i ocasionally get a message scrolling "TDM PCI master abort" and the machine effectively locks up
19:30.19brookshirewhere?
19:30.31coldfeetwhere fname=${name} or where fname=\${name}
19:30.34obsidian-studiosgreg_work: you do not have to switch, but research to see what the mins you need to get for your env are, and work on getting them
19:30.34MRH2hyperthreading as 1 cpu lookis like 2
19:30.38brookshireno second where
19:30.41coldfeettried all types of escpaes but no luck
19:30.45JerJergreg_work:  I have setup like 3 or 4 FXO systems and they are doing fine
19:30.54greg_workobsidian-studios: it would take me abuot 12 seconds to upgrade to newer gcc or whatever
19:30.55*** join/#asterisk darkskiez (~mhb@host-84-9-95-155.bulldogdsl.com)
19:31.06brookshiredon't forget the ''
19:31.07obsidian-studiosgreg_work:  at min you should use a newer kernel, since I bet the zaptel,  and etc drivers out of cvs are meant for a newer kernel?
19:31.10brookshirelearn mysql :)
19:31.12crash3mmrgoby: CORRECT
19:31.21funxioncoldfeet: maybe System(mysql -e select blah blah blah)
19:31.22JerJergreg_work:  are you running a known supported motherboard?
19:31.28JerJerfunxion: oh god no
19:31.29*** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
19:31.33obsidian-studiosgreg_work:  so mess around with it, I have a very strong feeling your weirdness is due to something in your env
19:31.49greg_workJerJer: well, i don't know what the issue is. it's hard for me, because it's production. i can't reliably reproduce it, it just happens after somewhere between 2 and 30 days
19:32.15mrgobycrash3m: about what ?
19:32.19greg_worki reload the wctdm drivers nightly, and it seems to help (it crashed this morning, but someone else rebooted, so i don't know if that was the issue .. probably though)
19:32.43Mw3hi, i have a tdm400p with 4 fxs in our office. sometimes the secretary is annoying me with this: when she made a mistake during dialing and she press down the hook button for a 1sec there will be no dial tone
19:32.47obsidian-studiosgreg_work: you can point and blame *, zaptel, and the drivers all you want, but they are not getting complied by them selves or in a env or world all to themselves. It is pulling stuff from your env, and depending on what it gets, might vary your output
19:33.00JerJerobsidian-studios:  bullshit
19:33.11greg_workJerJer: i'm not sure, where's the list of supported boards?
19:33.17mishehuI don't think I'll get any more of the tdm400 cards...
19:33.38JerJerhttp://www.digium.com/index.php?menu=compatibility
19:33.39obsidian-studiosgreg_work: or look at it like this, you bought a Digium card that comes with Digium support so call Digium
19:33.46crash3mmrgoby: misfire, sorry
19:33.52obsidian-studios<PROTECTED>
19:33.54crash3mMRH2: correct
19:34.18JerJerobsidian-studios:  everything came from cvs at some point
19:34.19JerJerso yes
19:34.25obsidian-studios<PROTECTED>
19:34.35JerJerand you are talking pure bullshit
19:34.44obsidian-studiosJerJer:  you get me an min * requirement sheet or compatibility for gcc, glibc , etc
19:34.51JerJerit doesn't care
19:35.20*** part/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc)
19:35.21JerJeri've ran asterisk on gcc 2.95 to 3.4 without a single issue
19:35.25brookshireobsidian-studios: at least not for gcc and glibc
19:35.33JerJerglibc and uclibc - same
19:35.37meshugahahah obsidian is still going off on how much gentoo is better
19:35.39brookshirenow there are a few necessary libraries
19:35.40JerJerit simply doesn't matter
19:36.14obsidian-studiosmeshuga: I am not talking about gentoo, If I develop an app with a version of glibc and compile it with gcc, and you do it with older versions you are going to have issues
19:36.16brookshireany distro that you have to compile 'nslookup' first in order to use it, is crap
19:36.19mutilatorwhats a good prog to merge in and out monitoring?
19:36.21brookshire:)
19:36.23mutilatorwavs
19:36.54JerJersoxmix
19:37.07obsidian-studiosGentoo just happens to be a source distro so you are compiling and running the latest version of software from each of the open source projects
19:37.23JerJerobsidian-studios:  then you are a shitty developer
19:37.27mutilator=\
19:37.29mutilatorno apt source
19:37.44obsidian-studiosJerJer:  great I am glad you feel that way, as you know what about me
19:37.55obsidian-studiosJerJer: did you get drumkilla's input?
19:38.08funxionmutilator there is apt source for sox
19:38.10obsidian-studiosJerJer:  or anyone at Digium to back up your statements?
19:38.20greg_workJerJer: nothing on that page says anything about my hardware.
19:38.30MRH2crash3m:  k hyperthreading enabled it is
19:38.35JerJeri'm speaking about my personal experiences with asterisk
19:38.37mutilator;P
19:38.39obsidian-studiosgreg_work: contact Digium you get paid support when you bought the card
19:38.51obsidian-studiosgreg_work: that's the horses mouth, so hear it there
19:39.04JerJeri have done thouands of installs in various environments
19:39.22greg_workyeah, the only thing is i don't have anything to tell them
19:39.23obsidian-studiosJerJer:  great, so explain and fix greg_work problem
19:39.29JerJerdon't run stable
19:39.33greg_work"my tdm400p sometimes dies"
19:39.51greg_workHEAD scares me
19:39.58obsidian-studiosgreg_work:  they will work with you and not leave you hanging, I bet they will start asking about the env and etc
19:40.04Assidhead scares you?
19:40.07Assidwhy?
19:40.09MRH2lol
19:40.11greg_workif i had time to follow the -dev list and cvs updates, i would run it
19:40.14obsidian-studiosAssid: teeth
19:40.16JerJeri love head - i wish i could get some more of it
19:40.17*** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com)
19:40.22Assidhehe
19:40.48JerJergreg_work: who says you have to follow anything?
19:40.56greg_workas is, i don't have any way of knowing that it's suitable for production if i check it out right  now
19:40.57brookshirejust update every month or so
19:40.59brookshirelol
19:41.05JerJergreg_work:  very simple
19:41.10JerJercd /usr/src
19:41.11greg_workother than if * crashes tomorrow, then i know it wasn't ;)
19:41.13JerJermkdir production
19:41.15JerJermkdir development
19:41.18JerJercd development
19:41.23JerJercvs co asterisk zaptel
19:41.31JerJerthen compile those
19:41.33AssidJerJer: what u suggest? head/production?
19:41.44JerJerthen make install everything
19:41.46brookshireHEAD
19:41.48brookshirelol
19:41.50CoaxDanyone have any idea how to unfuck a locked e-brake (drum!) on a 1997 plymouth breeze? :)
19:41.51JerJerthen if you find a problem in the development directory
19:41.52brookshireHEAD HEAD HEAD
19:41.56brookshireUSE HEAD
19:41.57brookshire:)
19:41.58Assidbah.. dev. / production
19:41.59JerJersimply cd /usr/src/production
19:42.01CoaxDthe cable lets down.. but the damn brake wont unlock! *blah*
19:42.07JerJerand make install those and reload the kernel module
19:42.11JerJerback to your old sysem
19:42.14JerJersystem
19:42.19greg_workJerJer: oh yeah, fair enough
19:42.20JerJer29 seconds
19:42.21tzangerwerd to the jer jer herd
19:42.24CoaxDGot the tire off. need to get the drum off, but the brakes are locked. blah.
19:42.24greg_workexcept "find a problem"
19:42.37JerJercome in a 2am and make those changes
19:42.40JerJerthen make a dozen calls
19:42.41greg_workthat means.. the phone system crashes? what if i'm not in the office? what if it's 3am?
19:42.48brookshireCoaxD: #1997plymouth :)
19:43.00brookshirei doubt anyone will know that answer in here
19:43.01brookshirelol
19:43.11CoaxDbrookshire: HEY! bah. i thought..  why not just spew my pain in an entirely off-topic manner, here in a group with a lot of people, and see what happens.. :)
19:43.11greg_workwhat if it's a bug (like whatever the problem is with my tdm400p) that will show up at some random point after days of running smoothly?
19:43.12funxionlol
19:43.16MRH2yay compile finished bbl
19:43.19ManxPowerOR you can just wait a few days after a 1.0.x release and then not have to do as much pre-production testing.
19:43.27SpaceBasswow... just noticed you can get allison to record voice prompts through digium's site
19:43.34CoaxDbrookshire: Hey, at least my asterisk has a 3 month uptime, and doesnt require fixing! *lol*
19:43.37brookshirelol
19:43.41JerJergreg_work:  then you know the problem still exists
19:43.44greg_workManxPower: like exactly what i do now. except everyone is telling me i shuoldn't run 1.0.x because HEAD is better :)
19:43.47obsidian-studiosSpaceBass: really for free or $?
19:43.59JerJerto which you can say "this problem exists in cvs head as of such and such date"
19:44.02greg_workJerJer: and i'd be totally willing to test that on a testing system
19:44.06ManxPowergreg_work: They are blinded by all the cool new features.
19:44.13greg_workbut i don't feel as adventerous with my production systems
19:44.16JerJerso then buy another TDM400P with one FXO and test
19:44.17tzangerManxPower: oh please.  :-)
19:44.32greg_workJerJer: if digium wants to give me a tdm400p and a system, i'd be enitrely willing to test
19:44.40greg_worki doubt that will happen though
19:44.44JerJerbuy one
19:44.44ManxPowerThey are prolly the same people that try to get their Longhorn and then deploy it in production
19:44.46JerJerthey are cheap
19:45.02obsidian-studiosgreg_work:  have you spoke to Digium at all or any of their developer/employees
19:45.02robl^SpaceBass, you can also  contact allison directly.  Digium's site is good for small volume recording (3-4 prompts), but Allison will work by the hour and it will be cheaper if you want LOTS of prompts
19:45.03*** join/#asterisk AlexCeli (~Alex@200.37.85.90)
19:45.12greg_workif i'm going to buy anything, it will likely be a media gateway ;)
19:45.17salvini_fscan someone help me setting up asterisk with mywebcalls?
19:45.24greg_workobsidian-studios: no
19:45.36obsidian-studiosgreg_work: give it a shot, you already paid for it when you purchased the card
19:45.37greg_workmaybe i'll just email them now
19:45.39JerJergreg_work: then you will find problems in chan_sip
19:45.42brookshirerobl^: sshhh! don't tell everyone our secert :)
19:45.45obsidian-studiosgreg_work: should get you somewhere faster than here
19:45.49greg_workJerJer: hehe, its possible ;)
19:45.59obsidian-studiosgreg_work: I would call, not email, but either way
19:46.04SpaceBassrobl^ somehow I cannot justify her for my home * box IVR
19:46.06robl^brookshire, ooops!  sorry!  you'll have to punish me!
19:46.15SpaceBassrobl^ but it would be fun
19:46.19obsidian-studiosgreg_work: I would be prepared to grant them access to the machine if possible
19:47.06*** join/#asterisk anti (russ@anti.developer.gentoo)
19:47.13SpaceBassLOL
19:47.27SpaceBassI've never been whipped with cat5... but imagine it might hurt... fiber would too
19:47.41brookshirenot cat5, cata5!
19:47.44brookshire:)
19:49.18pawhat is "predictive dialer"?
19:49.29ManxPowerWonka: Great idea!  I'll have to try making one of those, as well as a fiber version
19:49.34*** join/#asterisk johnm (~johnm@johnm.developer.gentoo)
19:49.43enderis there a page on the wiki that details the RFC for digitplan syntaxes?
19:49.50JerJergreg_work:  you check the standard shit?   interrupt sharing?
19:49.52MikeJ[Laptop]NERDS!
19:50.12eKo1cat5 is nothing, try rg-49 coaxial....
19:50.44greg_workJerJer: as much as i could, i don't really know enough about that anymore (ie, it said irq 207 or something)
19:50.55funxionpa: google it
19:50.58Wonkaone with two markings, to hang it on the wall with a vampire clamp
19:50.59SpaceBassI'm a functional nerd... quite similar to a functional alcholic... I keep it well hidden... I am an normal member of societ... I never wear my "got root" shirt out of the house
19:51.00JerJer?!  207
19:51.05JerJercat /proc/interrupts
19:51.10SpaceBassof course I'm also a functioning alcholic... but thats another room
19:51.12funxionlol
19:51.14*** join/#asterisk [Jedi] (iRRRRrrm@154.Red-217-127-168.pooles.rima-tde.net)
19:51.16[Jedi]Hello
19:51.17Wonkaand one with some good handle, to use as a LART
19:51.19greg_work201:   27529176   IO-APIC-level  wctdm
19:51.30JerJerwholy WTF batman
19:51.43greg_workthats what I thought ;)
19:51.55greg_worksec, i'll show you the whole thing
19:52.05JerJeruse pastebin
19:52.13greg_workhttp://pastebin.ca/16968
19:52.34ManxPowergreg_work: if you use APIC you can get things on interrupts 2 - 240 or so.
19:53.22greg_workManxPower: APIC is new to me.. i stopped messing with hardware a long time ago, that was the first time i've seen an irq > 15 :p
19:53.37JerJeryeah i don't know much about APIC
19:53.55JerJercan you try to turn that off in bios?
19:54.04greg_worksomeone suggested changing the PCI slot, i haven't had a chance to do that yet
19:54.05JerJerand turn off the parallel port and usb (if you are not using them)
19:54.07greg_workhm, not sure
19:54.12greg_workyeah, thats a good idea
19:54.19JerJerand eth1
19:54.23obsidian-studiosyou can turn off APIC at the kernel level on boot apic=no
19:54.24JerJeragain if not in use
19:54.26greg_worki use eth1
19:54.28JerJerok
19:54.38ManxPowergreg_work: APIC gives you LOTS of interrupts.
19:54.44obsidian-studioswhoops apic=off
19:54.50ManxPowerIF your motherboard uses it well.
19:54.55JerJerbut does it also give you LOTS of problems?
19:55.10greg_workManxPower: which frankly is probably a good thing :) 15 irqs always used to be the biggest PITA... :)
19:55.12obsidian-studiosgreg_work: make another entry in grub or what ever boot loader and append apic=off to turn off and see
19:55.49obsidian-studiosgreg_work: even if present in bios, using that command will disable in kernel, you can also not build APIC support in the kernel to go even further
19:56.02greg_workManxPower: well, being the only person who seems to know about APIC here :) .. do you think theres a benefit to turning it off?
19:56.11obsidian-studiosgreg_work: APIC not really a great idea for servers and etc
19:56.27obsidian-studiosgreg_work: unless you have a need you are aware of
19:56.58obsidian-studiosgreg_work: you will not hurt anything by turning off APIC
19:57.55greg_work(btw, i'm using an Asus A7N8X mobo, with an athlon XP 2000+ cpu)
19:58.45obsidian-studiosgreg_work: hardly a specific server board? Kill APIC and see if you problems go away
19:59.52greg_workalright, well, i'll try some stuff in about 1.5 hours, (5:30 EST) if you guys are still around
20:00.10greg_workpeople don't take kindly to taking the phone system offline in the middle of the day ;P
20:01.51ManxPowergreg_work: Not really.  In theory APIC might increase your interrupt latency a little bit, but that's the only issue I know of.
20:02.24[Jedi]Last week I bought a new TE405P
20:02.30ManxPowergreg_work: if you disable APIC, then you'll have to do all the work to confirm the card is not sharing interrupts, of course.
20:02.31[Jedi]and it says "Found a Wildcard: Wildcard TE410P/TE405P (1st Gen)"
20:02.35obsidian-studiosok I was totally off base with what I was looking for with regard to * call notification. Now I am leaning toward notification via some IM protocol. Anyone got info if one is better, more reliable than another. I currently can use IRC, or GAIM/AIM. I see ways for Jabber, but do not have a Jabber account etc?
20:02.37[Jedi]what does (1st Gen) mean?
20:02.47[Jedi]is my card an old model?
20:02.48file[laptop]first generation
20:02.53ManxPower[Jedi]: That means it's not a second generation card.
20:03.06file[laptop]imagine that!
20:03.06[Jedi]and what's a second generation card? what's the difference?
20:03.28file[laptop]http://www.digium.com/index.php?menu=press/pr_2gen_firm
20:03.33mutilatorwhen it has babies
20:03.42ManxPowerthank you, file
20:03.47[Jedi]thankx file
20:03.48file[laptop]go look at that URL
20:04.00file[laptop]then talk to Digium, cause that's the extent of our knowledge really in this area
20:04.37[Jedi]why have they sold me an old card if there're newer models available?
20:04.59[Jedi]well not they but their 'reseller' in spain... ://
20:05.02funxionobsidian-studios: what type of call notification are you doing?
20:05.03darkskiezas thats _just_ been released
20:05.09file[laptop]maybe the reseller didn't have it yet
20:05.25[Jedi]I think they should have told me
20:05.39[Jedi]it was my fault, in fact
20:05.43[Jedi]but they should have told me
20:05.45file[laptop]so talk to them...
20:05.48file[laptop]nothing we can do about it
20:05.55obsidian-studios<PROTECTED>
20:05.58ManxPower[Jedi]: for %90 of people there is no functional difference.
20:06.02darkskiez[Jedi]: you can post the card back and have it updated
20:06.04*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
20:06.18[Jedi]it's just a firmware updatE?
20:06.23darkskiezyes
20:06.28funxionAHH
20:06.31[Jedi]uhm and it can't be done by software?
20:06.38darkskiezone of the features of the new firmware is - Field upgradable firmware for future updates.
20:06.43[Jedi]hehehe
20:07.21enderwhich dtmf mode should I use for Polycom phones?
20:07.51[Jedi]well I'll keep my old "1st gen" card... and I'll buy next cards directly from digium
20:08.07darkskiez[Jedi]: whened you buy the card?
20:08.22[Jedi]when? last week
20:08.26Assidokay heres something weird..
20:08.40Assidi have to connect asterisk as a front end to a voip gateway
20:08.43darkskiez[Jedi]: dont think 2nd gen existed last week
20:08.48[Jedi]wow
20:08.54[Jedi]are they that new?
20:09.04Assidwhen a person presses an extension.. how do i send that to the gateways extension
20:09.06*** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
20:09.07[Jedi]then... bad luck for me :)
20:09.19JerJerAssid: by dialing
20:09.31AssidJerJer: a bit more complicated..
20:09.35JerJerno
20:09.37JerJernot at all
20:10.03darkskiezI concur
20:11.48[Jedi]which is the top number of TDM400 cards with FXO in a system?
20:11.58[Jedi]are these cards CPU-intensive like PRI cards?
20:12.03JerJeri wouldn't go any more than 1 maybe 2
20:12.12*** join/#asterisk wasim (~wasim@wasim.active.supporter.pdpc)
20:12.17JerJerbecause a T1 card+channel bank will be more effective
20:12.32[Jedi]but if it PRI wasn't in any way available
20:12.40[Jedi]and I really needed as many FXO's as possible
20:12.40brookshirehttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P
20:12.43*** join/#asterisk heka (~heka@82.114.68.124)
20:12.50[Jedi]how much FXO's could work in a single system?
20:12.59InfraRedone million
20:13.00AssidJerJer:  i dont knowhow to accept authorization from this device
20:13.02JerJer[Jedi]:  so then get a T100P and an Adtran TA750 with FXO cards
20:13.11JerJerAssid:  a type=user
20:13.17ManxPower[Jedi]: With a 4-port card and a channel bank you can put 4x24 ports
20:13.20Assidhrmm
20:13.24ManxPowerwell, 4 channel banks
20:13.32JerJeryp
20:13.33JerJeryep
20:13.34Assidlemme do this.. lemme set up the softphones.. and tell them to test on this..
20:13.41[Jedi]that would be for PRI T1/E1 cards
20:13.48[Jedi]what I need are standard phone-line interfaces
20:13.51JerJerno
20:13.57JerJera channel bank gives you FXO ports
20:13.59brookshireyou could use e1 and get more channels ;)
20:14.00[Jedi]uhm
20:14.05JerJerand aggregates them into a T-1
20:14.11[Jedi]oh I see
20:14.26[Jedi]and how much does such a thing cost?
20:14.33ManxPower[Jedi]: A channel bank converts each channel in a T-1/E-1 into an analog port.  (FXO or FXS, depending on the cards in the channel bank)
20:14.34JerJerdepends
20:14.42ManxPower[Jedi]: I just bought a channel bank off eBay for $180
20:14.44brookshireadtran is expensive, but high quality
20:14.59[Jedi]ManxPower: 180$ for how many E1s?
20:14.59JerJerand a T-1 card is like 500 $
20:15.05[Jedi]ando how many FXOs?
20:15.06ManxPowerAdtran Total Access 750 (Adtran TA is the only channel bank I'll buy)
20:15.08JerJeryou don't want E-1s
20:15.22brookshiree1 equip is more expensive
20:15.37ManxPower[Jedi]: 12 ports, these were FXS.  You'll have trouble finding FXOs.  Why do you want FXOs anyway?
20:15.46[Jedi]ManxPower: long history
20:15.59ManxPower[Jedi]: You will be VERY unhappy with analog FXO.
20:15.59brookshirecheaper than a t1?
20:16.16[Jedi]cheaper prices for fxo termination
20:16.18[Jedi]for some destinations
20:16.24brookshiremake since :)
20:16.35ManxPowerWe paid $300 for a 24 port FXS Adtran TA 750 a few months ago
20:16.37brookshiredigium is coming out with a 24port fxo card soon, maybe 2 months
20:16.44[Jedi]uhmmmm interesting
20:16.52ManxPowerbrookshire: We've been hearing that for the past 18 months.
20:17.01brookshireyeah.. prototype is in
20:17.01ManxPowerI'll believe it when it SHIPS.
20:17.03[Jedi]less interesting then
20:17.08[Jedi]:)
20:17.25[Jedi]so i could use an E1/T1 port of my TE405P and a channel bank
20:17.35brookshireyeah
20:17.44[Jedi]and the quality is that bad, brookshire?
20:17.49darkskiez24port FXO? that must be a beasty cable that attaches.
20:17.56ManxPower[Jedi]: Yes.  But remember since FXOs are much less common in channel banks, you'll have a harder time finding them on the used market.
20:17.56brookshireno.. what do you mean?
20:18.01*** join/#asterisk shido (~greg@d57-87-253.home.cgocable.net)
20:18.03ManxPowerdarkskiez: Amphenol, I'm sure.
20:18.12ManxPowerJust like channel banks use
20:18.20shidoyep
20:18.24brookshireamphenol connector
20:18.29brookshireyes!
20:18.43mutilatorJ1 = ?
20:18.54brookshirej1 = japanese t1
20:18.59darkskiezwould one of them fit on a pci?
20:19.04mutilator..o
20:19.16Silik0nwhen are the tXXXp's getting J1 part cert?
20:19.20brookshireyeah
20:19.33brookshireSilik0n: probably never
20:19.36brookshire:)
20:19.49Silik0nI know how to get that done ;)
20:19.55brookshirereally now
20:19.58Silik0nyeah ;)
20:20.02brookshirebecause that's impossible in the us
20:20.04brookshireLOL
20:20.08Silik0nnope
20:20.10Silik0nnot really
20:20.22ManxPowerYou bribe the certification authority, of course.
20:20.26brookshiretalk with malcolm please :)
20:20.46darkskiezSilik0n: they'll never know you've connected an uncertified card :)
20:20.49brookshirehit them with the cata5 pair
20:21.23Silik0ndarkskiez but most japanese purchasers of such equipment will not even consider it if its not cert'd
20:21.38darkskiezSilik0n: they wont even know there is certification.
20:21.53Silik0ndarkskiez are you japanese?
20:22.05*** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net)
20:22.08milkyflavahello
20:22.08darkskiezDont have a mirror handy
20:22.37Silik0nor should I say do you really understand how rigid their society is... especially their business society...
20:22.40milkyflavaI have a TDM11B dev card and I have * setup and my analog phone connected to it
20:22.42eKo1i though japan uses the US system...
20:22.49darkskiezSilik0n: are the Sangoma cards certified there?
20:22.49Silik0nhell they dont even fart without making sure their asses are covered
20:22.55brookshireno
20:22.55Silik0nnope
20:23.09Silik0nsangomas are not cert'd at this time
20:23.10eKo1US carrier system that is
20:23.18darkskiezSilik0n: get a business sponser and get them certified yourself
20:23.18milkyflavaI was wondering if a digital phone in the 2.4ghz range would not work plugged directly into an * box
20:23.25ManxPowereKo1: no.  Japan uses their own varient of T-1
20:23.43Silik0neKo1 actually the japanese telecom system is sorta like hte US system only bastardized to lock out non-japanese equipment and such
20:23.44brookshire??
20:23.46darkskieza DECT FXS base card would be nice.
20:23.53brookshire2.4ghz phone?
20:23.57brookshirewhy wouldn't it?
20:24.03milkyflavayes, from uniden
20:24.08ManxPowerall mine do.
20:24.10milkyflavaexcellent
20:24.11eKo1Silik0n: ah, that said it all
20:24.28*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
20:24.32milkyflavaI am getting a dial tone but as soon as I hit a number it hangs up so I just wanted to make sure
20:24.36Silik0neKo1 they dont even have GSM ... they has some whacky implementation of CDMA
20:24.46milkyflavathanks brookshire
20:24.49eKo1cdma2000?
20:24.58eKo1wait, that's qualcom...
20:25.03Silik0ni forget what it is, but its not "standard" cdma
20:25.18Silik0nits the Japanese version... heh cant use a japanese cell phone anywhere but in japan
20:25.30darkskiezSilik0n: could you get a certified convertor ?
20:25.42Silik0nthere are people working on such beasts
20:25.43Himekounless it's a vodafone phone
20:25.54Himekovodafone is running w-cdma 2100
20:27.03Himekothe other providers are using whacky stuff
20:27.14darkskiezSilik0n: http://www.voip-info.org/tiki-index.php?page=Sunrise+J1+PRI+solutions
20:27.24darkskiez<PROTECTED>
20:27.37Himekoouch
20:27.39Silik0ndarkskiez yeah
20:27.42[Jedi]wow
20:27.44[Jedi]that's a lot
20:27.46Silik0nthats only $500 USD
20:27.51Himekoliek is liek $4000
20:27.51[Jedi]500 $?
20:27.55Silik0nor is it $5000 USD
20:27.58[Jedi]this is like 3500EUR
20:28.11Himekoer that is like $4000
20:28.18darkskiez<PROTECTED>
20:28.23Silik0nI think its 105 jpy to 1USD
20:28.37Silik0nor something like that anyway
20:28.39shidoerrr
20:28.44[Jedi]498 JPY or 498000 JPY?
20:28.45shidowhy not use the quad t1
20:28.47shidoe1 in
20:28.48shidot1 out
20:28.48[Jedi]hehe
20:28.49shidodone
20:28.59darkskiez1 USD = 111 JPY
20:29.09Silik0n498.000 is japnese for 498,000 for you north americans
20:29.17Himeko.=, as a delimiter in manyplaces of the wor;d
20:29.33*** join/#asterisk astor (~chatzilla@135.80-203-85.nextgentel.com)
20:29.35[Jedi]almost everywhere excepting english, . is the delimiter
20:29.44milkyflavaWhen I hit a number button on my phone in the * cli it says Hung up 'Zap/1-1' why is that happening?
20:30.07darkskiezso, 4500 USD, nice.
20:30.44astorare there any regulatory issues with using a Uniden UIP1868 wireless handset which operates on 5.8GHz in europe?
20:31.33*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
20:32.00darkskiezNot sure what this is, but would you use two to do a J1/T1 convertor thing http://www.pcuniverse.com/464991.htm ?
20:32.35mogormandigium card will work on a j1 line
20:32.50Assidis incoming free on michigan did's for nufone?
20:33.10darkskiezmogorman: he needs a certified thing tho.
20:33.19mogormanouch
20:33.27mogormani dont know of anyone who has it cert.
20:33.45brookshiremogormon: you can use j1 internally in a company though
20:34.06milkyflavaI think it has to do with my zapata.conf file but am not sure
20:34.59*** part/#asterisk [Jedi] (iRRRRrrm@154.Red-217-127-168.pooles.rima-tde.net)
20:35.14*** join/#asterisk The_LightSide (~dt@wbs-196-2-121-52.wbs.co.za)
20:35.37*** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net)
20:35.56*** join/#asterisk loick (~loick@ATuileries-151-1-13-190.w82-123.abo.wanadoo.fr)
20:36.18*** join/#asterisk bankrupt (~bank@pcp08695394pcs.500ash01.tn.comcast.net)
20:36.46Ayanowhat's new in the asterisk world?
20:37.02InfraRednothing
20:37.13InfraRedi'l just having problems with 1.0.10
20:37.17InfraRedi'm
20:37.24Ayanowhat is it doing?
20:37.46InfraRednothing i want it to do
20:37.55AyanoI've been there befor
20:38.13darkskiezSilik0n: http://lists.digium.com/pipermail/asterisk-dev/2004-September/006090.html
20:38.51*** join/#asterisk nosc (~nosc@200.121.129.178)
20:39.38milkyflavadoes [trunkgroups] have to be in the zapata.conf?
20:40.08Cresl1nyou can use it publicly too
20:40.12Cresl1njust don't make a big deal about it :-)
20:40.39brookshireyou just can't use it to connect to the japanese telco
20:40.59Ayanoor india's pots lines
20:41.50Assid1,0.10 out ?
20:42.21InfraRed:)
20:42.23InfraRedfinally
20:42.30Assidits out???
20:42.40InfraRedsomeone who noticed my subtle lame joke
20:42.59AyanoOOPPS
20:43.16InfraRedwell done Assid
20:43.22InfraRedyou win a cup of tea
20:43.29InfraRedbit cold
20:43.33InfraRedbut it's nice
20:43.34InfraRed:)
20:44.30*** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net)
20:45.40*** join/#asterisk [Jedi] (iRRRRrrm@154.Red-217-127-168.pooles.rima-tde.net)
20:45.59Assidbrb
20:46.15milkyflavaCan someone look at this http://pastebin.ca/16972 and tell me where to look.
20:46.47*** join/#asterisk The_LightSide (~dt@wbs-196-2-123-224.wbs.co.za)
20:46.55milkyflavaI think it has to do with my zapata.conf file that I am probably missing something
20:47.36milkyflavaI have a TDM11B on FC1 trying to get my analog phone to work, I get a dial tone but when I try to call it from my cell I am getting that in the CLI of *
20:47.56milkyflavaWhen I try to dial a number it immediatley hangs up and I get a busy signal
20:49.18*** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca)
20:50.00*** join/#asterisk SarahEmm (~sarahemm_@MTL-HSE-ppp168389.qc.sympatico.ca)
20:50.01SarahEmmhihi
20:50.05Beirdoheya
20:50.12SarahEmmhihi Beirdo :o)
20:50.12newmedianwaves to SarahEmm
20:50.13SarahEmmyarrr.
20:50.17SarahEmmshido: *pokies*
20:50.18Beirdoyarr!
20:50.24*** join/#asterisk Assid (~assid@203.115.64.61)
20:51.25*** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net)
20:51.42BeirdoSarahEmm: that game is sooo evil :)
20:52.44funxionmilkyflava: do you have anything entered in the default context of your extensions.conf?
20:53.07SarahEmmBeirdo: yepyep ;)
20:53.11Beirdoheh
20:53.14Assidwheres shido???
20:53.20Assidi wanna buy a did man
20:53.34milkyflavaFunxion, I was just looking and no I don't even have [default] just [incoming] I have been trying multiple things to figure this out and must have deleted it
20:53.38AssidSarahEmm: which service?
20:53.45Beirdomine works great.
20:53.47SarahEmmAssid: trying to sign up with xetricom :)
20:53.47*** join/#asterisk _mwoodj_ (~mwoodj@hyper-eye.digium.sponsor.pdpc)
20:54.26funxionwell from what I can tell your zapata drops you into the default context which is missing
20:54.45funxionyou need to create [default] and add some exten's
20:55.04Assidwhere the hell is JerJer anyways
20:55.11Assidi want a DID
20:55.15milkyflavafunxion, you are exactly correct. I will do that and try again. Thanks so much, I am getting completely lost. :)
20:55.15CoaxDmichigan
20:55.19Assidis incoming free on michigan did's ?
20:55.26funxionnp
20:55.27CoaxDassid: Ya
20:55.28fileyes
20:55.31*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
20:55.31Assidokay
20:55.37BeirdoI can attest to that
20:55.40CoaxDassid: If yer a voip telco and not offering "free" incoming, you'
20:55.42loudnufone does did's ? didnt know that.
20:55.44CoaxDyou're not gonna last long
20:55.59CoaxDloud: Hell yea man. They're also the only voip telco that offers free 800 DIDs
20:56.06Assidthey do.. michigan/toll free only
20:56.07Assidnow..
20:56.09CoaxDloud: (You just pay for the incoming minutes you use)
20:56.19Assidso thats not free incoming
20:56.20loudhow cool, do you know if they have cali dids ?
20:56.28CoaxDassid: Of course not
20:56.32Assidfree incoming -- unlimited incoming
20:56.32BeirdoAssid: the MI ones are $0/min
20:56.43Assidokay..
20:56.50Assidi wanna purchase a number man..
20:56.54SarahEmmwell, a few
20:56.58SarahEmms'easier said than done ;)
20:57.01Assidi just wanna do some testing
20:57.08CoaxDassid: So get one!
20:57.09Assidwheres JerJer/chido anyways
20:57.09Assid?
20:57.19Beirdojust order it online, no?
20:57.22CoaxDassid: Your whole damn setup with any of these telcos, the most its ognna cost you is the first months' fee on a DID
20:57.30*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
20:57.41CoaxDassid: hell, JerJer doesnt even bill me for my MI did. I just use it for checking my voicemail, anyway
20:57.54Beirdoshh
20:57.57Beirdoheh
20:57.58SarahEmmBeirdo: depends which company
20:58.03SarahEmmBeirdo: not for anyone that offers 416 DIDs
20:58.04Assidactualy
20:58.09CoaxD(Then again, i do business with him on other areas, so..)
20:58.12BeirdoSarahEmm: yeah, sucky
20:58.12SarahEmmthey're all 'email person X'
20:58.15SarahEmmyeah...
21:00.19Beirdobut the only VoIP provider I've found in Puerto Rico so far is Verizon or one the other big ones...  not really asterisk-friendly.
21:05.38brookshiregot to love how the big telecos don't get it
21:06.08SarahEmmyeah...
21:06.25Beirdoyeah, sucketh
21:06.30opus__i'm getting call disconnects during calls, like 3 4 minutes into a call
21:06.38opus__i'm using alaw
21:06.44opus__should I try g729a instead?
21:07.48*** join/#asterisk alerios (~c81e471a@208.195.214.137)
21:10.52aleriosHi!,..  I've a strange problem (like all of them): most peers can make calls throug a Zap channel, but a couple of peers can't, and they are in the rigth context, and the channel is free. ¿any ideas?
21:15.15*** join/#asterisk harryvv (~noyb@S010600a0c93f6f7e.vs.shawcable.net)
21:16.31astorcan anyone recommend some wireless ip phones?  I need to be able to have 2-3 lines operational at the same time.  does this preclude the cheap non-wifi systems?
21:16.31*** part/#asterisk Moc (~mochouina@h66-201-214-109.gtconnect.net)
21:17.23SarahEmmast_freak: don't get a wifi system. :)
21:17.45SarahEmmi've heard of few that are good, and i don't know of any wifi ones... you can't use POTS ones with FXS?
21:17.46SarahEmm+ports
21:18.43astorSarahEmm: I can, but I thought it could be easier with wifi + I'd save having to buy the FXS stuff.
21:21.16aleriosastor: I  have a Zyxel p2000w and is not good
21:21.52aleriosis not much configurable
21:22.48astoralerios: what kind of configuration do you need?
21:23.57aleriosfor example, i couln't make it to send dtmf with rfc2833, icould only sent it inband
21:24.09astorI worry that some of these phones don't support anything else than WEP security.  But maybe that's no worse than what DECT or other digital wireless systems offer.  Anyone knows?
21:25.50LoRezDECT probably doesn't have ready-made tools to crack it easily available via the net with an off the shelf wireless card
21:25.57SarahEmmastor: you'd think (be easier and save having to buy FXS) but most of the wifi phones Suck :P
21:26.09LoRezthat's kinda run-onish, but I'm sure you get my meaning
21:28.09antiAnyone have any idea when I try to place calls to my asterisk at home via iax2, I get the message "No authority found" on the calling end, and the called end I get a connection attempt rejected.
21:28.48astorSarahEmm: so what's the current view on UIP1868 and similar proprietary wireless systems with voip gateways?
21:29.03SarahEmmastor: i dunno :)
21:29.11SarahEmmastor: i just relay what i hear here and reading review.
21:29.19SarahEmmi've never passed voice through * actually heh
21:29.20SarahEmmjust text
21:29.24astorSarahEmm: that's what I'm looking for :-)
21:29.42astorSarahEmm: gossip.. :-)
21:30.33astorthe UIP1868 could save me having to buy FXS-stuff, but I'm not sure if I can install two such systems in the same area.
21:30.34SarahEmmlol
21:30.38SarahEmmi've not heard any gossip on that yet
21:31.20*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
21:32.15*** join/#asterisk maruk (~maruk@i-194-106-46-242.freedom2surf.net)
21:32.23enderwhen using a polycom IP-30[01], and I want both line buttons to be forthe same extension, is it best to just register 2 lines to the same number or is there a better way?
21:33.00astoralerios: so rfc2833 should be on my list of required features..
21:35.47aleriosastor: for me, it is
21:36.02*** join/#asterisk Prion (~swschulz@cpe-024-211-202-206.nc.res.rr.com)
21:36.14astoralerios: I see that the F1000 is supposed to support rfc2833
21:43.07*** join/#asterisk znoG (~gs@200.115.216.109)
21:43.19pjzender: I've got a 500 and I just register the first button and leave the others blank
21:44.07enderpjz: yeah, I just tried that and that seems better.
21:44.16aleriosastor: gotta go. good luck
21:44.39pjzender: I use the other stuff for group-ring stuff so people can tell where they're getting the call from
21:44.55enderpjz: have you gotten sntp working?  My GMT offset isn't being used really.
21:47.13Assidhrmm how do i roll over a call to athe next free extension?
21:47.56*** join/#asterisk AlexCeli (~Alex@200.37.85.97)
21:48.20Assidsuppose i have 101, 102, 103 extensions.. and if i dial to 101.. it should roll to 102 if 101 is busy.. or 103 .. if 102 and 101 is busy
21:48.37*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
21:48.51Assidor even cycle like 103 if 102 is busy.. then at 103.. go back to 101.. if 103 is busy
21:49.01*** join/#asterisk cgcorea (~cgcorea@63.245.14.194)
21:49.24SarahEmmthat's just dialplan logic
21:49.34SarahEmmyou've rtfw?
21:49.43harryvvchecking out the *@home example seems that not all nick driver libraries are installed
21:49.59harryvvHello SarahEmm
21:50.06SarahEmmhihi
21:50.15harryvv:)
21:50.33harryvvender is port 123 open
21:51.00AssidChanIsAvail(SIP/2001&SIP/3001)
21:51.02Assidok
21:52.22abatistahello everyone
21:52.29SarahEmmhihi
21:52.54enderharryvv: open on what?
21:53.06enderharryvv: it seemed to get the right date, just the wrong time.
21:53.24enderalthough I don't recall if it had the right date before.  and I can't find a way to manually set the time.
21:55.05*** join/#asterisk |Vulture| (~V@user-0c6tr11.cable.mindspring.com)
21:55.13|Vulture|Anyone using PoE on the IP500/501?
21:55.42*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
21:55.48*** join/#asterisk DJ-Pyro (~DJ-Pyro@207.250.58.17)
21:55.57harryvvright now in the process of getting @home setup.
21:56.00pjzender: yeah, it looks to me like the sntp implementation won't traverse a gateway, though, so I just set up a server on my pbx
21:56.10harryvvVulture, not right now but need a switch
21:56.10ariel_quick question. I need to setup asterisk on a via-c3 the wiki does not give any real help on getting it working other then saying it needs a kernel-source select "CyrixIII/VIA-C3"
21:56.18pjzender: which is also my dhcp/ftp server
21:56.41ariel_is this on basic kernel build option or what. I don't seem to find it on CentOS 3.4/3.5
21:56.54|Vulture|harryvv: do they require an different cable? or just a regular RJ45 cable with a 802.3af switch?
21:56.54pjz|Vulture|: afaict, all IP500s use POE :)
21:57.10|Vulture|pjz: yea I didn't know if they could accept Netgear POE
21:57.20enderpjz: ah, that would be why.  Hrm, ok.
21:57.22pjz|Vulture|: oh, I think POE is pretty standard
21:57.31pjzender: I ran into the same thing :)
21:57.38|Vulture|I heard 79XX can't use Netgear's
21:57.44pjzoh, really?
21:57.45|Vulture|okay then looks like Ill be getting one
21:57.51harryvvvulture, I dont think thay use the same cable.
21:57.52pjz|Vulture|: hrm, well, don't take my word for it
21:58.21pjz|Vulture|: get a salesrep somewhere to promise that it works
21:58.23|Vulture|okay
21:58.29|Vulture|good idea
21:58.30DJ-Pyroso we're not getting any audio on our ds3 from global crossing, they had a tech note that went out last week specifically about asterisk and no audio being passed, I just upgraded to cvshead and we're still having this issue, anyone know anything about this or why it won't work?
21:58.36harryvvits pretty easy to seperate DC from the TCP/IP pulses from ether net cable so yes it should be the same.
21:58.39pjz|Vulture|: then if it breaks you can give it back to them and say 'you promised'
21:58.57|Vulture|lol
21:59.30SarahEmmDJ-Pyro: what are you using to interface a DS-3 with *?
21:59.55*** part/#asterisk jfonsecausa (~jfonseca@12.42.141.34)
22:00.48|Vulture|Sangoma makes a DS3 card
22:01.16mogormanthey have asterisk drivers?
22:01.21|Vulture|yes
22:01.24DJ-PyroSarahEmm: DS3 splits off through an adtran mx2800 into 7 servers, each with a 4 port card in it
22:01.33DJ-Pyrothe card is fine, channels don't alarm, dchannel is up
22:01.42ariel_|Vulture|, no it does not use the normal PoE polycom IP-500
22:01.48mogormanhow many channels can you actually drive in a machine?
22:01.49|Vulture|ariel_: thank you
22:01.51DJ-Pyrowe see the calls come in, play tt-monkey, get 2 digits, and read back 4
22:02.03DJ-Pyrodon't hear any audio and don't see the dtmf tones
22:02.08SarahEmmDJ-Pyro: nice
22:02.19SarahEmm(well except for that)
22:02.45DJ-PyroI know, GC put on a tap in chicago and didn't see any traffic
22:03.09DJ-Pyroit's also something to note that globalcrossing knows about asterisk, they put out an internal tech notice about issues customers have been having with it and no audio
22:03.11mogormani mean vulture, whats the point of ds3 card if your machine can only push 200 or so channels
22:04.11*** join/#asterisk JonUK (~me@81-178-16-14.dsl.pipex.com)
22:04.19|Vulture|mogorman: yea I know thats why DJ-Pyro's setup makes more sense
22:04.37JonUKhi everyone, anyone care to offer any advice on what ISDN BRI card works best with *
22:05.19JonUKI have been looking at the AVM and Eicon cards, but cannot see much difference between them
22:06.01*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
22:06.53JonUKno one??
22:06.55SarahEmmblitzrage: you here?
22:06.59SarahEmmJonUK: i have no idea, sorry
22:07.04SarahEmmshido: mew?
22:07.33milkyflavaI just got my first phone call to work into my * box to my analog phone!
22:07.50JonUKwell done milkyflava
22:08.07milkyflavaThanks JonUK!
22:08.55shmaltzanybody here knows what happens if A receives a call from B and B transfers A to C using the transfer button on their phone (attd xfer), what shows up in the CDR? under channel will it show B or A?
22:09.10shmaltzI know I can test it but if someone else did the test already why should I bother
22:09.32JonUKshmaltz, cant answer, sorry
22:10.10DJ-Pyroanyone have any insight?
22:11.28JonUKshmaltz, I would think that nothing would be shown in CDR, as the phone is handling the transfer to the Destination C
22:12.09shmaltzJonUK, well what you think and what * thinks are 2 different thigs
22:12.18shmaltzthigs = things
22:12.27JonUKok, was just a guess !
22:12.31DJ-Pyrohttp://pastebin.ca/16979
22:12.38DJ-Pyrothat's what the console says
22:12.43DJ-PyroI don't see what we're doing wrong
22:12.59Inv_arpariel_: should work fine since its x86
22:15.42ariel_Inv_arp, it's not working fine. I can't get asterisk to run on it.
22:15.47shmaltzJonUK, were u under the impression that my guessing talent is not as good as yours?
22:16.37*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
22:17.32ariel_Inv_arp, I found some emails that said you need to make sure you compile it for Cyrii/via-c3  But I can't seem to find the option.
22:20.10*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
22:20.49puzzledariel_: i think someone on the list mentioned compiling for i586 will work
22:21.47ariel_puzzled, what is the option for that? I have had one of my worst days today and I am not thinking correctly.
22:22.19pjzariel_: compile it for pentium architecture
22:22.23puzzledariel_: you are talking about compiling * right?
22:22.33*** join/#asterisk Dan_K (dan@c-24-8-35-8.hsd1.co.comcast.net)
22:22.56ariel_puzzled, yes or do I need to set the kernel up first differently?
22:23.24puzzledariel_: if the kernel runs fine on the box I guess there is no need to change the kernel
22:23.36puzzledariel_: try editing PROC in the asterisk Makefile
22:23.41ariel_puzzled, I keep getting asterisk unable to load.
22:24.41ariel_puzzled, I will do that. My head is just not in it.  I am going back to the hospital.. see you all.
22:24.48Dan_KHello, I am a new asterisk user.  I signed up with broadvoice and their minimalist byod plan so I could play with a real voip connection.  However when trying to dial out from asterisk, it seems to ignore my fromuser and fromdomain directives in the sip.conf for broadvoice
22:25.10Dan_KIs there anything I need to look at specifically?
22:25.16puzzledariel_: in my asterisk Makefile on line 81 it says: #PROC=i586. enable that and try again
22:25.20*** join/#asterisk gbdrbob (drbob@alltalk.demon.co.uk)
22:25.36*** part/#asterisk gbdrbob (drbob@alltalk.demon.co.uk)
22:29.14Dan_Kah, nevermind, I figured it out.  helps if I name the section correctly in sip.conf and extensions.conf
22:30.09SarahEmmshido? blitzrage?
22:31.11blitzrageSarahEmm: yes?
22:33.12pjzhow do I prepend something to the CID string in stable?
22:33.28pjzwhat's the function call in asterisk?
22:33.34blitzrage${CALLERID}this_is_the_string I believe
22:33.34pjzer, in extensions.conf
22:33.55pjzbut how do I set it?
22:34.07pjzsomething like SetCallerID() or something?
22:34.17blitzrageyes
22:34.42pjzso SetCallerID(FOO: ${CALLERID}) ?
22:35.37*** join/#asterisk SarahEmm (~sarahemm_@MTL-HSE-ppp168389.qc.sympatico.ca)
22:35.40SarahEmmwheee wifi--
22:36.41blitzragepjz: believe so - give it  shot
22:37.26tzangermmmmmmmmmm stale nonce... my favourite
22:37.50*** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il)
22:37.51blitzragetasty
22:42.00*** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
22:42.18dca[laptop]hey anthm, you around?
22:42.49anthmyes
22:43.18blitzrageSarahEmm: were you looking for me?
22:43.31tzangermy blackfin books came today :-)
22:43.53dca[laptop]anthm: has any else reported problems with ChanSpy cuasing seg faults?
22:44.00Assidshido ???
22:44.02Assiddid?
22:44.17tzangerAssid: does the channel really need to know about your issues? :-/
22:44.38Assidhehe
22:44.40Assidsorry
22:44.52anthmnot that I know of
22:45.47SarahEmmblitzrage: err.... yes, but now i can't remember why :) Sowwy
22:45.58SarahEmmAssid: shido seems to not be around.. i'm looking for him too heh
22:46.01SarahEmmAssid: wondering what's up with my DID :)
22:46.33SarahEmmooh right
22:46.37SarahEmmblitzrage: ETA on just a 416 DID?
22:48.15Assidyou bought one?
22:48.30SarahEmmAssid: from shido? i'm trying to
22:48.50SarahEmmAssid: i have no idea where things are tho =-/
22:49.07SarahEmmhe said 416 DIDs should be quick, but i'm not finding that heh
22:49.10blitzrageSarahEmm: ummm... like 1-2 weeks I think to order
22:49.27Assidhow the hell can i get a did of only 9 digits?
22:49.34SarahEmmAssid: huh?
22:49.34Assidare there 9 digit numbers?
22:49.36SarahEmm9??
22:49.39Assidyeah
22:49.42Assidthats what i got
22:49.43blitzrageseems wrong
22:49.45SarahEmmAssid: oh. where in the world are you?
22:49.51SarahEmmblitzrage: uhh... 1-2weeks for a 416?
22:50.00SarahEmmthat seems long
22:50.08SarahEmmAssid: if you're in NA that's wrong...
22:50.27dca[laptop]anthm: did you catch that last question?
22:50.40blitzrageSarahEmm: yah.. I think thats what I've been told, although I've found it seems to be closer to 4-5 days
22:51.43SarahEmmhrm.
22:51.45SarahEmmokay....
22:52.13opus__cool i got g729 to work
22:52.53Assidhrmm time to start packing up..
22:52.56Assid4.30 am
22:53.23pjzNo application 'SetCallerID(TOC: ${CALLERID}'
22:53.33opus__missing )
22:54.04Assidgnight folks
22:56.28pjzokay, so why didn't the CID change work?
22:57.57anthmyah my answer was, not that I know of
22:58.44*** part/#asterisk mkrufky (~mk@68.160.103.77)
22:58.54blitzragepjz: missing )
22:59.02pjzI see the SetCallerID() execute in the stack
22:59.05pjzblitzrage: fixed that
22:59.17pjzbut my phone doesn't show the modified CID
23:02.14dca[laptop]anthm, sorry didn't see it, tks guess something is up with my box then
23:03.04*** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt)
23:08.30*** join/#asterisk WeezeyD (~ohno@206.210.109.226)
23:08.42WeezeyDanyone have a TDMoE set up between two asterisk boxes?
23:08.50key2when someone calls me on my zap, is {exten} the callerid ?
23:08.52*** part/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
23:09.12*** join/#asterisk psywar (psywar@rasterburn.org)
23:09.26psywaranyone have problems with a SPA-2000 throwing the occasional bad UDP checksum?
23:09.42pjzohhh.. is there a max length to CID?
23:09.43psywarit terminates my calls, really annoying
23:10.15WeezeyDmy SPA-3000s always gives that checksum thing
23:10.51psywardoes it terminate your calls and annoy you greatly?
23:10.59WeezeyDno
23:11.03psywarhrm
23:11.08WeezeyDit just gives that error
23:11.19psywarI guess I need to do more correlation.
23:11.28psywarmaybe the two events aren't correlated.
23:11.30*** join/#asterisk pingywon (~mike@pcp0010034410pcs.reding01.pa.comcast.net)
23:12.35psywarI'm only doing VoIP over about 1m of cat5 cable, dunno why I'm getting so many abrupt terminations
23:12.35psywarseems to happen in bursts, and only with certain callers
23:12.35infinity1i see a volume voicemail patch. is that available in the main code?
23:12.35WeezeyDmaybe a bad switch or cable?
23:12.36psywarcould be cable, since I make them myself
23:12.42psywarI doubt it though
23:14.18*** join/#asterisk AgiNamu (~bob@200.6.217.202)
23:14.28AgiNamuHey, does SIP md5secret for users work with non-Asterisk users?
23:19.49*** join/#asterisk Chotaire (chotaire@chotaire.net)
23:19.56Chotairegood morning vietnam.
23:20.37SarahEmmhi Chotaire
23:21.20*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
23:22.30terrapenanybody use Cingular here?
23:22.42terrapengosh, i know nothing about mobile phones these days
23:23.01greg_workJerJer: still around?  disabled everything in bios, now my interrupts look like this: http://pastebin.ca/16992
23:24.10greg_workweird thing though.. i had been using only ports 1, 3 and 4 of my TDM400P (all fxo).. forget exactly why, but I was having a problem with port 2 not answering or something.. anyway, just tried to go back to using 1, 2 and 3, and port 2 seems to be permenantly offhook
23:25.43terrapencingular has this EDGE thing
23:25.49terrapensupposedly quite fast
23:26.04terrapendunno anybody who uses it
23:26.09WeezeyDg'damn.  CallerID doesn't seem to be working over TDMoE
23:28.22*** join/#asterisk josephcool (~asterisk@adsl-67-121-209-208.dsl.sndg02.pacbell.net)
23:30.03josephcoolI have a problem. When I do a Playback(custom/aa_1) I don't get playback, but if I do Playback(/var/lib/asterisk/sounds/custom/aa_1) it will work. How do I setup my default directory for this?
23:30.05niZonhas anyone used a handytone 488?
23:30.45josephcoolI haven't but I love this Sipura SPA-841...
23:30.52niZonhmm
23:30.59*** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com)
23:31.18josephcoolI don't know how to set the directory correct so asterisk will see the sound files.
23:31.31niZoni want a cisco 7940/60 I have some uses for the XML features...
23:31.46niZonbut right now I want to know some stuff about the ht488
23:32.10josephcoolDo you know anything about how asterisk directories works?
23:33.08niZonmy directory is /usr/share/asterisk/sounds/
23:33.16niZoni dunno how to change it
23:33.38josephcoolhmmm
23:33.40*** join/#asterisk JunK-Y (~junky@69.156.216.243)
23:33.40infinity1~voicemailvolume
23:34.18josephcoolI BET That is the problem...
23:34.20pjzjosephcool: afaik that's hardcoded somewhere
23:34.22josephcoolHow do you change it?
23:34.33josephcooloh... hardcoded? guh.
23:34.36josephcoolugh even.
23:34.39niZondo a "locate sounds" or "find /*|grep sounds"
23:34.41niZon*shrug*
23:34.54_DAWjosephcool - you could move it with sim link
23:35.35josephcoolno, I see that... holy smoke, I get it now... the problem is I installed it apt-get... that must make the /usr/share sounds directory, but AMP creates and tries to use /var/lib/asterisk/sounds...
23:36.02pjzjosephcool: I'd suggest a symlink
23:36.09josephcoolinstead of a recompile?
23:36.12pjzjosephcool: yeah
23:36.13*** join/#asterisk cfrank_ (~cfrank@bi01p1.co.us.ibm.com)
23:36.14WeezeyD~tdmoe
23:36.14jbothmm... tdmoe is for a situation where you need TDM reliability without traditional TDM hardware. TDMoE is useful because it allows the above familiarity, flexibility, and reliability of TDM but over inexpensive Ethernet instead of T1s or E1s.
23:36.17josephcoolTrue...
23:36.18infinity1anyone know anything about the voicemail volume problems? is this some kind of sick joke?
23:36.22*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
23:36.30josephcoolok, how do I create a simlink?
23:36.34SarahEmmwhat problem infinity1?
23:36.38WeezeyDln -s
23:36.41SarahEmm(i'm just curious, i have no idea what the issue is or a fix btw
23:36.44infinity1SarahEmm: http://lists.digium.com/pipermail/asterisk-users/2005-June/113872.html
23:36.44SarahEmmshido: mewmewmew?
23:37.10file[laptop]SarahEmm: KITRICH!
23:37.10infinity1SarahEmm: from what i'm reading voicemail/zap is not possible.
23:37.12WeezeyD~callerid
23:37.24*** join/#asterisk tinch0 (morey@84.77.99.140)
23:37.39WeezeyDinfinity1: sending Zap to voicemail?  works fine.
23:37.41SarahEmmhihi file[laptop]
23:37.51SarahEmminfinity1: i don't beleive that...
23:37.53tinch0hi
23:37.54SarahEmminfinity1: lots of people do that.
23:38.01infinity1SarahEmm: i feel betrayed. i spent many days on this * stuff and its broke.
23:38.51infinity1SarahEmm: ...and it seems no one cares. i don't get it. there must be more to this.
23:38.58tinch0how is the right syntax for Dial/SIP if I want to use a certein account at a proxy server?
23:38.58SarahEmminfinity1: well, i'm sure zap and voicemail is possible. a ton of people do it
23:39.05SarahEmmyou're getting low volume, infinity1? just in voicemail, no other * stuff?
23:39.07niZonI want * to ring certain voip extensions when a call comes in via PSTN on the HT488, but I want the HT-488 to keep the PSTN line on hook until I pick up a voip extension, anyone tried this?
23:39.20ManxPowerinfinity1: that voicemail volume problem only happens with one of the formats
23:39.45infinity1ManxPower: what format should voicemail be using?
23:39.52ManxPowerinfinity1: Try gsm
23:40.01infinity1ManxPower: i read the bug report on it. and has a LOT of comments
23:40.06tinch0similar to IAX syntax I think... anybody?
23:40.09ManxPowerinfinity1: Yup.
23:40.40WeezeyDinfinity1: mine works fine, I use wav49 and gsm, wav49 for email attachments.
23:41.01infinity1WeezeyD: using zap as a receiving interface?
23:41.06WeezeyDyep
23:41.21infinity1WeezeyD: i just got a VM in my email. i couldn't hear shit.
23:41.27infinity1WeezeyD: i'll try some other formats.
23:43.06josephcoolSo how do I create a symlink that will re-point the /usr/share/asterisk directory to /var/lib/asterisk like amp wants. and will that take everything recursive?
23:43.59ManxPowerinfinity1: What format ARE you using?  WAV has known problems
23:44.13pcmttvdy is a special protocol that does nothing
23:44.18pcm~ttvdy
23:44.47pcmhmmm ... though the bot was learning from the channel ?
23:45.09SarahEmmttvdy? what's that?
23:45.18SarahEmmpcm: it is, but you have to prefix it with jbot:
23:45.19ManxPowerjbot, ttvdy is a special protocol that does nothing
23:45.19jbotManxPower: okay
23:45.33infinity1ManxPower: mine was set to format=wav49|gsm|wav
23:45.40ManxPowerjbot, ttvdy is a special protocol invented by the CIA.  Now I have to kill you.
23:45.40jbot...but ttvdy is already something else...
23:45.45ManxPowerjbot, no, ttvdy is a special protocol invented by the CIA.  Now I have to kill you.
23:45.45jbotokay, ManxPower
23:45.51infinity1ManxPower: i changed it to wav, from reading the bug report, i thought that was best.
23:45.58ManxPowerinfinity1: And what do you use each of those formats for?
23:46.04pcm~ttvdy
23:46.04jbotrumour has it, ttvdy is a special protocol invented by the CIA.  Now I have to kill you.
23:46.05ManxPowerThere should be a REASON for each format.
23:46.09infinity1ManxPower: that was the default
23:46.28ManxPowerWe all know Asterisk's defaults suck.
23:46.36*** join/#asterisk outtolunc (outtolunc@adsl-69-110-52-142.dsl.pltn13.pacbell.net)
23:46.46infinity1ManxPower: i just started using this. i just got a call over one zap, and it bridged to my cordless phone (zap). there was echo on my end. weird.
23:47.05*** join/#asterisk rv_weasel (~no@adsl-68-93-10-206.dsl.ksc2mo.swbell.net)
23:47.06infinity1ManxPower: so you recommend gsm for the format?
23:47.19SarahEmmpcm: what *is* ttvdy?
23:47.33infinity1ManxPower: any idea whats up with the echo?
23:47.46rv_weaselok,   i need a good sip client for linux.  one with console features would rock too
23:47.54pcmsarahemm: it's nothing :)
23:48.04ManxPowerinfinity1: Yes.
23:48.15rv_weaselI got my system all setup though!!!  everything is workign great
23:48.25*** join/#asterisk eng1neer (1000@adsl-68-94-42-172.dsl.rcsntx.swbell.net)
23:48.26ManxPowerEcho is caused by using only 2 wires for tx and rx.
23:48.28*** join/#asterisk jpablo (~jpablo@201.138.154.114)
23:48.40infinity1ManxPower: so i need a special cable?
23:48.41ManxPowerinfinity1: Don't try to run before you learn to walk, grasshopper.
23:48.42jpablohi, anyone here has an tdm in production ?
23:48.44jpablowith 4 fxo modules.
23:48.45rv_weaselrouting all my calls out through the net at 1.1 cent/min
23:48.57ManxPowerinfinity1: no, you need to balance your volume levels and turn on echo can
23:48.59infinity1ManxPower: what do you mean? i just got a working dial plan
23:49.03josephcoolSo I can't see where I would change the directories from /usr/share/asterisk to /var/lib/asterisk   Anyone know how to change this?
23:49.20josephcoolI understand I can symlink it... but how does it normally setup?
23:49.23*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
23:49.25ManxPowerinfinity1: Exactly.  You just got a working dialplan and now you are asking about echo, one of the hardest things in telecom to get rid of.
23:49.41infinity1ManxPower: ouch.
23:49.42jpabloI'm having LOTS of problems, with it, the fxo lines just hang or something, it says they are Onhook, when they are idle and they don't work for anything.
23:49.48*** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
23:50.01infinity1ManxPower: i just dialed someone through the pbx (zap/zap) and there was no echo. weird.
23:50.23ManxPowerThat's because the audio path delay in zap<->zap is not enough for you to HEAR the echo.
23:50.31*** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-7-173.w81-248.abo.wanadoo.fr)
23:51.39josephcoolAnyone know how I switch directories for asterisk?
23:51.46ManxPowerall this stuff is covered over and over and over and over and over and over and over and over and over again in the mailing list archives.
23:51.53rv_weaseli was wondering about echo cancelation.  it seems the digital ends always have echo.  i need to balance the analog ends rx and tx gain to get rid of that dont i?
23:51.57ManxPowerjosephcool: "cd directory"
23:51.57infinity1ManxPower: err ..when someone called me zap/zap, i heard the echo. is that different then when i initiate the call
23:52.01josephcoollol manx...
23:52.06infinity1ManxPower: k
23:52.18infinity1ManxPower: i somehow got on the topic. i didn't do any research yet.
23:52.21josephcoolI mean to change from /usr/share/asterisk to /var/lib/asterisk like AMP wants.
23:52.24*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
23:52.33infinity1ManxPower: so i'll just leave the VM to gsm, see how it goes, and google on the echo thing.
23:52.35ManxPowerjosephcool: I'm sorry, I cannot help you with that.
23:52.39infinity1ManxPower: thanks :)
23:53.00MikeJ[Laptop]googleit
23:54.03rv_weaselnow here is a question.  if i got a DID account with IAX i could forward my single landline to that and make a cheap trunk
23:59.33jpablois they anyway to tell a tdm to use other irq ?

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