irclog2html for #asterisk on 20050610

00:00.01*** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net)
00:00.34SarahEmmhmmm. are there any commercial SIP/IAX paging devices out there?
00:01.56*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
00:02.04puppetapp_intercom aint used anymore
00:02.13tzafrir_laptopdoes IAX support sending text messages like SIP?
00:02.27tessierIf you are using asterisk for anything other than a very basic company PBX you need to get your head checked.
00:02.32tzafrir_laptopAnd does asterisk recognize those?
00:02.33tessierI've learned it the hard way.
00:02.51puppettessier: ehm why?
00:02.56puppettessier: asterisk can do ALOT of things
00:03.00SarahEmmtessier: whyfor?
00:03.04puppettessier: from basic to super advanced
00:03.33puppettessier: alot of commercial VoIP providers uses asterisk for customer gateways
00:03.38tessierpuppet: I am well aware of what it can do. And what it can do. It can do pbx'y things.
00:03.42tessierpuppet: No, a lot do not.
00:03.49puppettessier: a lot d yes
00:03.50SarahEmmthere's lots of stuff that * can do that most basic *or* advanced PBXs can't
00:03.51puppetdo*
00:03.54tessierA few do.
00:04.05puppettessier: ALOT VoIP providers uses asterisk yes
00:04.11puppetnot a few
00:04.15tessierpuppet: Proof by repeated assertion will not sway me.
00:04.24SarahEmmtessier: what's the argument against?
00:04.39puppettessier: why shouldnt people use asterisk for that?
00:05.18*** join/#asterisk jeffik (~Jeff@69.158.17.52)
00:05.42rristroph2hello
00:06.06tzafrir_laptophi
00:06.10rristroph2i am currently recording calls to .wav files - how do i record into an mu_law format?
00:06.14puppettessier got quiet :(
00:06.36tzafrir_laptoprristroph2, you can use wav. But it has to be a "proper" wav.
00:06.55tzafrir_laptoprristroph2, do you have sox installed?
00:07.01rristroph2yes, it is recording ok
00:07.05SarahEmmpuppet: *nods* :)
00:07.05rristroph2yes i do have sox
00:07.18rristroph2can i use sox to convert to mu_law?
00:07.38JunK-Yrristroph2: sure.
00:07.47blitzragerristroph2: what are you using to record?
00:07.50rristroph2hmmm, can you give me an example command line on that?
00:07.51JunK-Yand its u-law, pronounced mu-law P:)
00:07.56tzafrir_laptoprristroph2, no, but you may need to convert the wav file to mono, 8khz 16bits per sample
00:07.58rristroph2oh
00:08.02Nuggetmoo law!
00:08.09rristroph2yes thats what i need to do laptop
00:08.11blitzragemeow law
00:08.17JunK-Yive script for that, go read, there's info everywhere.
00:08.23tzafrir_laptopNugget, mu. micro
00:08.27puppettessier: died?
00:08.31Nuggetno, moo. cow
00:08.33JunK-YNugget: i propose to have a cow for the new asterisk book, im sure you supporting the idea huh? :)
00:08.33rristroph2so, like this  "sox abc.wav abc.ulaw"
00:08.35Nugget]:8)
00:08.40SarahEmmblitzrage: YAY! meow law!
00:08.45Nuggetyay cows
00:08.51blitzrageSarahEmm: hahaha.. I was wondering how long till I got a reaction :D
00:08.55blitzrageJunK-Y: I agree!
00:09.04rristroph2tzafrir_laptop - what you think on that sox format?
00:09.08SarahEmmblitzrage: hehee i'm multitasking between irc and tv :)
00:09.08blitzrageJunK-Y: do you know of any previous book with a cow on the cover?
00:09.15blitzrageSarahEmm: understandable :)
00:09.24JunK-Yin o'reilly series? nope.
00:09.32blitzrageSarahEmm: you coming to the next TAUG meeting?
00:09.48blitzrageor anyone in Toronto for that matter
00:09.48opus__does anybody use realtime asterisk? what do I put in extconfig?
00:09.58tzafrir_laptoprristroph2, sox can convert formats and do other useful manipulations.
00:10.11tzafrir_laptop~sox
00:10.14jboti heard sox is Sound Processing Tool. URL: http://sox.sourceforge.net/
00:10.17rristroph2tza - ok, i'm looking at that now
00:10.24rristroph2thanks!
00:10.27puppet~saraemm
00:10.42opus__~yo
00:10.43jbotYo wazzaaaap my brotha?!
00:10.43SarahEmmblitzrage: i dunno! when?
00:10.44tzafrir_laptopnot useful enough. I'm too tierd to search now. But there's a relevant page on voip-info
00:10.45puppetjbot: i heard saraemm is strange
00:10.47jbotpuppet: what are you talking about?
00:10.47SarahEmmpuppet: what?
00:10.51SarahEmmpuppet: what about me?
00:10.51puppet;D;D
00:10.53puppetdamn
00:11.01SarahEmm<-- is saraHemm not saraemm
00:11.07rristroph2tz-another question - I have a program that is "reading" a xxx.16K file and I want to get a .wav file convert to have it read that
00:11.07puppetdamn ;D
00:11.10puppeti missed the h ;P
00:11.14SarahEmmand yes, i'm very very strange ;)
00:11.14puppetsarahemm: im getting tired ;P
00:11.27opus__jbot: i heard loop is ~yo
00:11.28jbotopus__: what are you talking about?
00:11.29blitzrageSarahEmm: June 24th (last Friday of every month)
00:11.37SarahEmmblitzrage: ahh
00:11.44SarahEmmsomewhere up near me, no?
00:11.49tzafrir_laptop~puppet
00:11.52blitzrageSarahEmm: yah... I think its in North York
00:11.58SarahEmmwhich is where i am :)
00:12.02blitzrageSarahEmm: do you know Toby's is?
00:12.05SarahEmmblitzrage: i'll consider it :)
00:12.16SarahEmmblitzrage: yeah, few blocks from me
00:12.22blitzrageSarahEmm: thats where it is :)
00:12.37SarahEmmcoolies! that's Very Close then :)
00:12.46blitzrageSarahEmm: then you have no excuses then :)
00:12.48SarahEmmdang
00:12.51SarahEmm26th you said?
00:12.56blitzrageSarahEmm: 24th I think
00:13.00*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
00:13.00blitzrageSarahEmm: its on a Friday
00:13.01SarahEmmoh. working afternoon shift it looks like
00:13.21blitzrageSarahEmm: and here comes the excuses :)
00:13.23puppettessier: WAKE up ;P
00:13.40SarahEmmblitzrage: lol.
00:13.48*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
00:13.48SarahEmmblitzrage: i am, really! i'd like to meet up with local asteriskey peoples tho
00:14.15blitzrageSarahEmm: yah, stick it in your calander. We're getting a website going soon and will be scheduling events and such
00:14.16twisted[work]heh
00:14.18tessierpuppet: What?
00:14.24twisted[work]wheeee!!!!!
00:14.31tessierblitzrage: Welcome to our ool. Notice there is no p in it.
00:14.41blitzragetessier: I love that sign :)
00:14.51twisted[work]i personally like the one that says
00:14.53blitzragetessier: I don't swim in your toilet, so please don't piss in our pool.
00:15.00SarahEmmblitzrage: well, this month i'm working, in theory... we'll see :)
00:15.00twisted[work]"we don't swim in your toilet, please don't peen in our pool"
00:15.01twisted[work]haha
00:15.02twisted[work]yeah
00:15.08blitzragetwisted[work]: :D
00:15.12blitzrage<< better than twisted[work]
00:15.18twisted[work]sup blitzrage
00:15.18SarahEmmblitzrage: you coming?
00:15.21twisted[work]whoa
00:15.24twisted[work]did i walk in on cybersex?
00:15.24blitzragetwisted[work]: haha.. not too much man, wuz up?
00:15.32blitzrageLOL
00:15.40blitzrageSarahEmm: where might be coming? :)
00:15.41harryvvI feel like hanging xlite on a cross
00:15.43SarahEmmtwisted: LOL!
00:15.46twisted[work]hehe
00:15.48SarahEmmtwisted: err..... no. :P
00:15.54blitzrageSarahEmm: to the meetup?
00:15.56SarahEmmblitzrage: yeah :P
00:15.58twisted[work]SarahEmm, oh. damn.  it's been awhile since i've seen public cybersex
00:16.02twisted[work]:P
00:16.07SarahEmmgah! i'm not cybering with Blissex...
00:16.10SarahEmm... or blitzrage.
00:16.13twisted[work]rolf
00:16.18twisted[work]er
00:16.20twisted[work]rofl
00:16.20twisted[work]hahaha
00:16.26blitzrageSarahEmm: yah, hopefully I can finally make it this month. I'll have just gotten back from Spain and be heading to St. Louis that weekend, but I should be able to make it
00:16.37blitzrageSarahEmm: I've been out of the country for the last 2 meetups
00:16.37SarahEmmblitzrage: cool... not sure if i will. i'll ponder it some
00:16.42puppettessier: awake now?
00:16.42blitzrageSarahEmm: tis coo
00:16.46puppettessier: we asked you why
00:17.00blitzragetwisted[work]: rolling on the laughing floor?
00:17.02puppet(02:04:32) < SarahEmm> tessier: what's the argument against?
00:17.02puppet(02:04:47) < puppet> tessier: why shouldnt people use asterisk for that?
00:17.05twisted[work]so blitzrage, hows ze writing going?
00:17.22twisted[work]shift swapper!
00:17.28blitzragetwisted[work]: oh its going. We go to production tomorrow, so I have a bunch of work to do tonight.
00:17.33twisted[work]blitzrage, whoa
00:17.38twisted[work]to PRODUCTION?  or to editing?
00:17.47blitzragetwisted[work]: to editing (they call it production...)
00:17.51twisted[work]oh.
00:18.03twisted[work]i call production the actual end act that produces something :P
00:18.05blitzragetwisted[work]: so in 2 months or less from tomorrow, we should see a hard copy book :)
00:18.12blitzragetwisted[work]: yah, I agree
00:18.16twisted[work]blitzrage, yay!
00:18.24twisted[work]blitzrage, i still want a signed first edition
00:18.27blitzragetwisted[work]: yah... I'll be happy when its finally done :)
00:18.39blitzragetwisted[work]: LOL.. I still dont' get why people want my signature, makes no sense :)
00:18.43twisted[work]preferably one of the first 50 copies
00:18.44blitzragetwisted[work]: but hey, for you, of course.
00:18.47twisted[work]yay!
00:19.00blitzragetwisted[work]: go pre-order a copy right now then :D
00:19.06twisted[work]and where do i do that?
00:19.20blitzragetwisted[work]: go to amazon: ISBN - 0596009623
00:19.23twisted[work](hell, i don't even know the title of the isbn)
00:19.26blitzragetwisted[work]: or search for "Asterisk"
00:19.26twisted[work]oh
00:19.28puppettessier got quiet again :o
00:19.37blitzragetwisted[work]: Asterisk: The Future of Telephony
00:19.59twisted[work]so should I ship it to you first?
00:20.00blitzragetwisted[work]: its cheap right now too, only like $26 with shipping
00:20.13blitzragetwisted[work]: are you coming to Astricon USA in October?
00:20.27blitzragetwisted[work]: I could just sign it there - or feel free to ship it to me first and I'll send it back to you
00:20.29citatswhere is astricon usa?
00:20.30twisted[work]blitzrage, yea
00:20.31twisted[work]that works
00:20.35blitzragetwisted[work]: coo
00:20.45blitzragecitats: not sure yet (or not supposed to say yet :))
00:20.46SarahEmmooh, anaheim
00:20.54twisted[work]blitzrage, you got your name in the byline :)
00:20.54twisted[work]nice
00:21.04blitzragetwisted[work]: I'd sure hope so :)
00:21.34SarahEmmhmm. anotehr conference to consider :P
00:21.54blitzrageSarahEmm: Astricon is awesome - good times
00:22.04twisted[work]yes
00:22.07twisted[work]astricon rocks the socks
00:22.11blitzrageSarahEmm: plus you'll get a book with your admission :)
00:22.18twisted[work]where else can you freak out desk attendants with a wireless desk phone :P
00:22.34SarahEmmhmm. but people keep trying to convince me to go to cluecon ;)
00:22.36blitzragetwisted[work]: LOL - good times :)
00:22.40blitzragetwisted[work]: I think I have pictures of that
00:22.47blitzragetwisted[work]: they were dark, but I got a couple
00:22.50*** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net)
00:23.21blitzragetwisted[work]: I should go and find those pics on my computer somewhere and upload them to my photo gallery
00:23.48twisted[work]blitzrage, preordered ;)
00:23.53blitzragetwisted[work]: NICE!
00:24.03twisted[work]$26 with free shipping is cool
00:24.14blitzragetwisted[work]: yah... way better than that "other" book
00:24.16twisted[work]hehe
00:24.26twisted[work]wow, amazon reccomended two tool albums to go along with my book
00:24.28twisted[work]strange.
00:24.34blitzragetwisted[work]: really?  thats the BEST :)
00:24.38twisted[work]hehe
00:24.40twisted[work]i love tool
00:24.48blitzragetwisted[work]: don't say that too loudly :)
00:25.00blitzrageI also love the band Tool. :)
00:25.06citatsi've been suprised with some amazon recomendations.  things that are polar opposites but stuff i was into
00:25.08twisted[work]oh..ooops...hahahaha
00:25.14twisted[work]ROFL
00:25.35blitzragetwisted[work]: I think even you'll like this book.
00:25.41blitzrage:D
00:25.52twisted[work]possibly
00:25.53blitzrageat least I hope so... spent enough hours on it
00:25.53twisted[work]dude
00:26.03twisted[work]that OTHER book you can get it for 76.41
00:26.14blitzragetwisted[work]: yah I know - thats the funniest thing
00:26.20twisted[work]and it comes with an asterisk cd
00:26.21twisted[work]rofl
00:26.35blitzragetwisted[work]: yah... a burnt CD with Maxell on top :)
00:26.38blitzragetwisted[work]: or BenQ
00:26.46twisted[work]blitzrage, haha
00:26.59*** join/#asterisk cypromis (~michael@83.149.70.59)
00:27.05*** part/#asterisk makhtar (~ageller@mail.bulletinnews.com)
00:27.24blitzragefile[desk]'s name is in the book :)
00:27.29twisted[work]mine too?
00:27.44blitzragetwisted[work]: it will after I write it in? :)
00:27.48twisted[work]hehe
00:27.49twisted[work]yay
00:27.57SarahEmm*giggles*
00:28.05twisted[work]jesus
00:28.07twisted[work]504 pages?
00:28.08blitzragetwisted[work]: I had to put file's name in there - he answered ton of questions for me
00:28.11twisted[work]is that including the appendix?
00:28.16blitzragetwisted[work]: yah
00:28.20twisted[work]oh ok, good.
00:28.29blitzragetwisted[work]: 10 chapters, 5 appendices (one appendix is 100 pages)
00:28.36twisted[work]lol
00:28.42file[desk]good late night reading
00:28.43blitzrageyah... it was huge :)
00:29.05twisted[work]7 minutes
00:29.16blitzrageSarahEmm: hey!  no giggling, this is a VERY serious channel
00:29.19file[desk]I forget, I'm waking up at 5:30
00:29.22blitzrageSarahEmm: run by communists
00:29.25file[desk]I need to pack my Mac
00:29.29harryvvi give up 3 people and lots of help latter xlite on a remote end does not pass two way rtp conversation.
00:29.29SarahEmmmeep!
00:29.48puppettessier: ??
00:29.56twisted[work]whoa
00:29.57blitzrageSarahEmm: I have no idea what a kitrich is
00:30.05twisted[work]there's actually a 2nd OTHER asteriskbook
00:30.08twisted[work]that sells for 6.50
00:30.17twisted[work]haha
00:30.19blitzragetwisted[work]: that one is even WORSE than the other one :)
00:30.23SarahEmmblitzrage: http://www.sarahemm.net/glossary.php
00:30.26SarahEmm^-- SarahGlossary
00:30.31twisted[work]no way, not if it's 6.50!
00:30.43blitzrageSarahEmm: LOL
00:30.44twisted[work]that's like saying the 12.50 clones of the x101p aren't as good as the digium ones!
00:31.12SarahEmmblitzrage: i have some weird sayings and words *giggle*
00:31.18puppettwisted[work]: x101p clones ownes digium cards
00:31.21puppet;D;;D;D;D;D
00:31.22blitzrageSarahEmm: ditto
00:31.31twisted[work]puppet, *schmack*
00:31.34puppethaha
00:31.43puppetnight peps
00:31.56blitzrageSarahEmm: I like the en (suffix) :)
00:31.58*** join/#asterisk Legend (~legend@24.244.142.133)
00:31.59CunkI didn't realize Asterisk has been around since 1969...interesting
00:32.39twisted[work]rofl, hey kram ;)
00:32.44kramhi & bye :)
00:32.55twisted[work]!@
00:32.55twisted[work]no
00:32.57twisted[work]!!!
00:33.03blitzrageSarahEmm: LOL - nice desk :)
00:33.16SarahEmmblitzrage: i use it at work, i say stuff like 'pixen' to mean 'multiple Cisco PIX devices' and people get confused :P
00:33.26SarahEmm'cuz i say VAXen, and it's just a logical extension...
00:33.33SarahEmmblitzrage: it's tidier now.. same stuff there mostly hto :)
00:33.41twisted[work]heh
00:33.45twisted[work]pixen.
00:33.51twisted[work]boxen
00:33.53twisted[work]vaxen
00:33.53blitzrageSarahEmm: I love the Oscilloscope.... wish I had one
00:33.54SarahEmm*nods*
00:33.54twisted[work]faxen
00:34.01blitzragemacken :)
00:34.07twisted[work]ehh?
00:34.12blitzragei.e., to mack
00:34.20twisted[work]oh, that's mackin i though
00:34.21SarahEmmblitzrage: heh, s'an old one, still has nuvistors in it. works great up to like 20MHz which is fine for what i do, mostly microcontroller-based stuff
00:34.35SarahEmma lot of what i do is logic anyway, then the logic analyzer makes sense.
00:34.45blitzrageSarahEmm: yah for sure
00:34.59blitzrageSarahEmm: Basic Sign Communication - nice book
00:35.04SarahEmmblitzrage: hehee.
00:35.12SarahEmm*nods* ASL1 book :)
00:35.25SarahEmmblitzrage notices details....
00:35.28rayvdi have an iax link set up between two asterisk servers in iax.conf.  appears to be working.  i'm confused about the basic steps for getting my SIP clients on serverA to be able to make calls to SIP clients on serverB ...
00:35.29blitzrageSarahEmm: I used to have an ASL book, but don't know where it went
00:35.37rayvdanyone got a good link to an example config?  voip-info is down :(
00:35.43SarahEmmblitzrage: been working on learning since last year sometime...
00:35.50blitzrageSarahEmm: I'm all about the details - just ask oej when I proofread his training slides :)
00:36.09blitzrageSarahEmm: I haven't been around anyone whos signed for a few years now... so I forget a lot :(
00:36.12file[desk]you make everything groovy
00:36.13file[desk]wild thing
00:36.21SarahEmmblitzrage: aww...
00:36.22blitzragehaha.. SarahEmm in a Nutshell :)
00:36.37SarahEmmblitzrage: i was living with a partner who signed until about a month ago... lately i'm not around many people that sign anymore
00:36.40twisted[work]ooh word
00:36.41twisted[work]test done
00:36.56SarahEmmblitzrage: :o) that's the book about me *giggle*
00:37.01*** join/#asterisk Smi|k (~Ling@adsl-66-159-200-157.dslextreme.com)
00:37.05tzangerhahaha
00:37.19tzangerI can tell you all about my nuts in a nutshell... they're freakin boiling
00:37.21tzangerit's a bagsticker out there
00:37.35blitzrageSarahEmm: covers DeathLaser - LOL
00:37.41blitzragetzanger: LOL
00:37.52file[desk]ohhhhhh this opening for Metallica is awsome
00:37.53tzangerI drove from listowel to whitby and back today
00:37.54tzangerugh
00:37.56Smi|kso much knowledge here, I hope someone knows.... does anything exist that has a phone line plug on one side, and a network jack on the other side? it accepts faxes and saves them to a network folder
00:38.00tzangerbut I achieved the impossible
00:38.04Smi|ksimple little device, but I cant find it anywhere
00:38.06SarahEmmblitzrage: an ex put that book cover image together, and as i work in a high-security place i can never really talk about my job at all.. they figured i must work at a place building some kind of mass destruction laser device :)
00:38.13blitzragetzanger: you got that cement mixer working?
00:38.19tzangerSmi|k: yeah it's called either an iaxy+pc or a PC with TDM01B
00:38.23tzangerblitzrage: of course
00:38.26tzangerwas there any doubt?
00:38.33blitzragetzanger: no comment
00:38.42tzangeractually
00:38.43tzangerno cement
00:38.44yaaarword
00:38.44blitzrageSarahEmm: hahaha... cool :)
00:38.47Smi|kbut it requires a PC?
00:38.47tzangerthey started up with no load today
00:39.09Smi|kthere is no stand alone device that can just receive and save to a folder on the network?
00:39.26tzangerSmi|k: nope, make one an dmake millions on the missed market
00:39.28SarahEmmblitzrage: :)
00:39.34tzangeroh wait... there's no missed market, which is why there's no product
00:39.36file[desk]there's a lot more then simply that, it's a complex thing to do Smi|k :P
00:39.43Smi|kI am looking at efax
00:39.53SarahEmmblitzrage: the gallery just got put up a month or three ago, so there's not so much there yet.. i've been throwing new stuff on it now and then, a new pic of me, some new boarding pass piccies.. random stuff :)
00:39.56Smi|ksomething like 10-15 cents per page to receive them
00:40.02yaaarok so this should be fun. ubuntu apparently has asterisk packages, so I'm going to throw down a quick one-phone connection at my house on this box.
00:40.06Smi|kand they can raise their rates any time
00:40.11file[desk]Smi|k: http://www.faxmicro.com/
00:40.19Smi|kfile[desk], reliable?
00:40.22*** join/#asterisk xeet2 (~xeet3@bwi1-br1-gig3-1.jsci.net)
00:40.25file[desk]yup
00:40.27Smi|kfor companies, you dont want to have to change your fax number
00:40.31yaaaryou guys figure one phone will work ok alongside X on a p4-2.4 w/ 512M?
00:40.44SarahEmmyaaar: uhh. yeah.
00:40.48yaaarnifty
00:41.00tzangeryaaar: don't do it
00:41.07yaaarwhy not?
00:41.11xeet2anyone know how to make * not follow a sip "moved temporarily" redirect?
00:41.15tzangerbecause X wants top priority and so does *
00:41.21tzangeruse a dedicated PC, even if it's a cheapie
00:41.26yaaarit's not like it has to work or anything....i just brought the phone home from work, figured i'd try it out...
00:41.37tzangerI ran 1FXO+1FXS on a P90MMX without issue aside from not being able to use ilbc
00:42.01blitzrageSarahEmm: yah.. I started mine at leifmadsen.com/gallery/ a few months ago as well. I broke my camera screen about 2 months ago though unfortunately and just got it back 2 days ago. Luckily to took Futureshop over 60 days to get it back to me, so I got a brand new one for free instead of the refurbished one for $150 (since cracked screens aren't covered under warranty)
00:42.03yaaartzanger, how about i renice X?
00:42.16*** part/#asterisk Cunk (~chatzilla@64.222.148.45)
00:42.23SarahEmmblitzrage: ahh. my camera the screen flips around to protect it when not in use heh
00:42.26*** join/#asterisk SwK (~ken@12-219-156-206.client.mchsi.com)
00:42.30tzangeryaaar: do what you please, just don't be disappointed when it plays up
00:42.33tzangerblitzrage: nice
00:42.38SarahEmmblitzrage: i need to take more piccies so i can put them up heh
00:42.47tzangerI have a lot of cement plant pics on it today
00:42.52yaaartzanger, "play's up"?
00:42.52blitzrageSarahEmm: wish mine did that - but I bought a cover this time so it doesn't get broke :)
00:43.00blitzragetzanger: lol... how boring :)
00:43.09tzangeryaaar: just try it, when it breaks complain here and I'll say "I told you so"
00:43.13blitzrageSarahEmm: HP R707 <--- nice camera
00:43.16yaaaroh, ok, that's fair
00:43.18SarahEmmblitzrage: heh :)
00:43.26tzangeryaaar: but please stop asking questions about why when I've already said it's gonna act up when you least expect it :-)
00:43.33xeet2anyone know how to make * not follow a sip "moved temporarily" redirect?
00:43.44blitzrageSarahEmm: damn you - got me beat :)
00:44.01twisted[work]yay
00:44.02twisted[work]okay
00:44.03twisted[work]time to go home
00:44.05twisted[work]bbiaf.
00:44.10blitzragetwisted[work]: peas!
00:44.21file[desk]blitzrage: you're silly btw
00:44.31SarahEmmblitzrage: where do you live btw?
00:44.33blitzrageso I've heard :)
00:44.44blitzrageSarahEmm: Oakville - moving to downtown at the end of summer I think though
00:44.45*** join/#asterisk SuperMMan (~sales@d142-59-44-20.abhsia.telus.net)
00:44.52SarahEmmcoolies :)
00:44.58*** join/#asterisk Ahewes (~rsb@209.81.2.61)
00:45.04SuperMManHello all i know this isn`t the channel but does anyone here deal with GPS units?
00:45.06blitzragemust on this HD somewhere....
00:45.19blitzragemust be*
00:45.20SarahEmmSuperMMan: yes... but this isn't the channel ;)
00:45.25SarahEmmastricon sounds nifty
00:45.39SuperMManSarahEmm you deal with it. if so mind if i message you
00:45.41blitzrageSarahEmm: yah, its was great fun. I head to another one in Spain on Saturday
00:46.08SarahEmmSuperMMan: yes i do, and sure msg me
00:46.24SarahEmmblitzrage: wowie. company paid?
00:46.37AhewesAnyone know how to reverse the order in which voicemail messages are played?  i.e. newest first instead of oldest first
00:46.40blitzrageSarahEmm: well, paid by a company I work for - I'm self employed :)
00:46.56SarahEmmblitzrage: ahh, okay.
00:46.58SarahEmmnifty :)
00:47.07blitzragetzanger: which reminds me - you ever get a chance to talk to any accountants?
00:47.26tzangerblitzrage: he's out of the office, I needed to talk to him about my taxes too
00:47.32opus__Does anyone have asterisk working with realtime and odbc?
00:47.44blitzrageSarahEmm: yah, I might have to get the company I work for in the US to sponsor me for a greencard since most of the work I do is for US companies
00:47.50SarahEmmahh
00:47.51opus__I think my extconfig.conf is wacked, can somebody give me an example?
00:47.52blitzragetzanger: ok cool, just let me know when you get a hold of him
00:48.11SarahEmmblitzrage: i'm staying out of the US for now other than visiting.. tho i wish that wasn't the case.
00:48.46blitzrageSarahEmm: yah, not really my choice, just seems a lot of US companies want to work with me :)
00:48.58blitzrageSarahEmm: plus I don't mind converting US dollars to CAD :)
00:49.05SarahEmmblitzrage: heh, i wouldn't either :)
00:49.16tzangerblitzrage: of course
00:50.11opus__nobody uses realtime? :(
00:50.27blitzrageopus__: nope, looked at it once, and decided I didn't want to spend the time on it :)
00:51.13xeet2opus: hmm?  I do
00:52.59opus__xeet2 - i have all the sip.conf realtime caching turned on, still no users.  i added ast_log output in some functions in config.c...
00:53.11opus__but i think my extconfig.conf is messed up due to lack of documentation
00:53.16opus__i have, for sip; sipusers => SYBASE,asterisk,sip_buddies
00:53.44opus__theres like a billion examples, sipusers sippeers sipbuddies which is it?
00:53.48xeet2opus: users don't ever show up with realtime
00:53.59*** join/#asterisk edy2 (anonymous@202.134.128.82)
00:54.00opus__well, i can't dial them either..
00:54.11xeet2you don't dial a user, you dial a peer
00:54.15xeet2or a friend
00:54.37xeet2are you able to register?
00:54.49edy2hello everyone
00:55.06*** join/#asterisk outofjungle (~outofjung@61.17.134.218)
00:55.46opus__well, i haven't even tought of registering.
00:55.59opus__does a realtime sip users have to call only a realtime extension?
00:56.10xeet2no
00:56.24edy2I have a modem,a phone line, and a computer running linux.I want my linux to greet all calls on the phone line? is it possible thru modem without using x100p
00:57.06xeet2opus: realtime is only queried when it has something to look for.  doing a sip show peers or sip show users will not do a query for all entries
00:57.12opus__xeet2 - can you show me your extconfig.conf?
00:57.36opus__undertood
00:58.22opus__if I'm reading that right, users is one table correct?
00:58.42xeet2well, thats the way I did it, makes it simple
00:58.48xeet2and then every entry has a type=friend
00:59.19opus__even sippeers?
00:59.49xeet2yes
01:00.19xeet2at least in my case.  that might not work for you, don't know what your setup is
01:00.27edy2anythoughts on my question\
01:00.41edy2developing an answering machine using linux with a lucent modem only
01:01.11xeet2opus: no users ever show up under sip show users or iax2 show users, and only registered peers show up under show peers
01:01.30xeet2edy2: just get an x100p or use voip
01:01.40opus__xeet2 - there is a hack to do that,
01:02.04xeet2opus: if there is, feel free.  I don't need it =)
01:02.11opus__;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
01:02.14xeet2edy2: where are you located?
01:02.24*** join/#asterisk DaLion (~DaLion@Quebec-HSE-ppp225549.qc.sympatico.ca)
01:02.27DaLionhi all
01:02.38xeet2opus: that doesn't make users show up
01:02.40DaLioncan i ask what would be an autocomplete for 7 digit and send that to zap instead ?
01:02.46opus__oh. hmm
01:02.48DaLionwould
01:02.53DaLionexten => _514NXXXXXX,1,DIAL(Zap1/${EXTEN})
01:02.55DaLionwork
01:03.13opus___X
01:03.31xeet2_NXXXXXX,1,Dial(Zap/1/514${EXTEN})
01:03.38DaLionoh
01:03.44tzangerthey have 10-digit-dialing in montreal?
01:03.46DaLionso if i dial only 7 it would try that right ?
01:03.52DaLionyes
01:03.56DaLiono
01:03.57DaLionlol
01:04.00xeet2if you only dialed 7 digits it would dial it out with a 514
01:04.31DaLionok
01:04.36DaLionso i need my 10 digit also
01:04.47DaLionmeaning if 7 digit assume its 514 else use voip gateway
01:05.23DaLionseems to work
01:05.43DaLionyep works
01:05.45edy2xeet2:i am located in Pakistan
01:05.45DaLiongood
01:05.54DaLiondidnt want to use my zap for longdistances lol
01:06.33xeet2edy2: I don't know of any voip providers off hand that can do pakistan
01:06.34edy2fine x100p, but whats voip?
01:06.35DaLiondarn just got my CSCo plugged to asn asterisk for local .. plus answer incoming.. and 3 out voip lines plus internal voip if 10 dgiti
01:06.39DaLionok nough fun
01:06.49DaLionanyone know a good services xml makeer for csco ?
01:07.00edy2i just want to have an answering machine sort of thing
01:07.07xeet2dalion: what kind of app are you trying to make?
01:07.12DaLionhehe
01:07.13DaLionnone
01:07.18DaLionjust toying with it all
01:07.39opus__cscope?
01:07.49DaLionphone has 2 lines for business.. 1 for personal voip.. plus my 4th is local asterisk on telco line zapped.. with al sorts of goodies.. like DISA and telemarketer hell for all
01:08.24DaLionnow i need to get an extensions from  LA , CTU ;)
01:08.39xeet2ctu?
01:09.20tzangerhe's got a private link to cthulu
01:09.35SarahEmmlol
01:09.47DaLioncoounter terrorism unit ;) watch 24 ?
01:09.54DaLiongot ringtone
01:09.59DaLion;) wel itd default merlin 1
01:10.35DaLion!jbot bring voip-info up
01:10.54DaLionyehah..
01:11.21DaLionhad to get a pro9800 on a lcd 21" to hook sat link up.. now i got a tv all the time no need to take time off
01:12.04*** join/#asterisk likwid-- (~likwid@nc-65-40-166-134.dyn.sprint-hsd.net)
01:14.03opus__whats the difference from -r 1.1 and -r HEAD?
01:14.40SarahEmmyou just asked that in asterisk-dev opus
01:14.42SarahEmmand it was answered there
01:14.59SarahEmm1.1 is the... 1.1 tag. HEAD is the HEAD tag. HEAD is the latest stuff in CVS. 1.1 is revision 1.1.
01:15.44xeet2anyone know how to make * not follow a sip "moved temporarily" redirect?
01:15.46DaLionbasicaly head is experimentl not stable but all options
01:15.49opus__Well, I want a good answer:)
01:16.05DaLionis that nice  enough opus__
01:16.06DaLion;)
01:16.06opus__perhaps something in HEAD is messing up realtime odbc in 1.1
01:16.23DaLionmaybe it also not rewriting SDP packets in nat=yes
01:16.24DaLionalso
01:16.27opus__you know, you have to try every single combination until it works..
01:16.36PyroSteveanybody used a door lock/door phone to lock a door instead of unlocking it ?
01:16.39twistedannnnnnddd...
01:16.39DaLionme ?
01:16.39twistedi'm back
01:16.42SarahEmmopus: err.. you want a good answer?
01:16.54SarahEmmopus: i gave the only answer there is, really.. what are you looking for exactly
01:17.08opus__yeah, i need to subscribe to -cvs
01:17.14twisteder
01:17.22xeet2PyroSteve: shouldn't be too hard to do something like that
01:17.52PyroSteveI have the project of interfacing a samsung DCS phone system with the magnetic door lock
01:18.24LegendPyroSteve: errr, what?
01:18.35PyroStevethere is a door phone interface module that has the system plugs into but its designed to unlock the door for 500 ms
01:18.43edy2PyroSteve:thats interesting!
01:18.58LegendPyroSteve: right, a doorphone and an electric strike, what about it?
01:19.01opus__pyrosteve yes,
01:19.16twistedthere's a 1.1 tag?
01:19.21twistedthat's a new one on me :P
01:19.32twistedi'm familiar with HEAD and v1-0 tags, but not 1.1
01:19.49xeet2I'm assuming 1.1 will do realtime?
01:19.56twistedWHAT 1.1?
01:20.02SarahEmmerr, good point
01:20.10SarahEmmyou could grab revision 1.1 of each file, but that wouldn't work so good
01:20.11opus__pyrosteve :http://www.xplproject.org.uk/
01:20.27SarahEmmnini all!
01:20.35twistednite SarahEmm
01:20.52blitzrageSarahEmm: nite
01:21.11twistedblitzrage, d'oh
01:21.17*** join/#asterisk TESTER2 (~Cyber@modemcable219.42-81-70.mc.videotron.ca)
01:21.30TESTER2any way yo enable mwi notification on xten x-lite?
01:22.01Nuggetx-lite doesn't pose any unique challenges regarding MWI.
01:22.02opus__wwait, asterisk HEAD is 1.1
01:22.47mepplgute nacht  -  good night
01:22.59blitzragetwisted: w00t - bkw_ still has a copy :)
01:23.03X-RobWoo
01:23.05twistedblitzrage, yay
01:23.06TESTER2Nugget: how?
01:23.08X-RobGXP-2000 wiki updated
01:23.10opus__Asterisk CVS-NHEAD-06/09/05-21:18:13
01:23.15X-RobThese are _hr0n_ phones now.
01:23.17puppet<PROTECTED>
01:23.19puppet;D
01:23.19Nuggetthe same way you enable it with any other phone
01:23.21twistedyeah, that wasn't doctored
01:24.00*** join/#asterisk wfu (~wfu@203.131.175.66)
01:24.04TESTER2It's enable but event if I have a mssg the tone doesn't change or a special ligth flash...
01:26.06*** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com)
01:26.24twistedTESTER2, by chance, did you set the mailbox in sip.conf?
01:26.49TESTER2yes 1234@context
01:27.47TESTER2But what does x-lite suppose to do when the MWI is on?
01:28.31xeet2ask you to buy the full version?  =)
01:28.35Nuggethaha
01:28.57TESTER2On my vista-350 (on a spa-1001) the ligth flash and the tone change, but what x-lite is suppose to do?
01:32.37*** join/#asterisk file[desk] (~jcolp@mctn1-6383.nb.aliant.net)
01:34.02*** join/#asterisk diablopico (~russ@ip68-8-227-7.sd.sd.cox.net)
01:34.14diablopicohello
01:34.19*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
01:34.45diablopicoi have a question about using asterisk as a Voip gateway ..
01:34.58diablopicois it the same as pbx
01:35.56xeet2diablopico: you can do anything you want with *
01:36.19*** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net)
01:37.51Nuggetasterisk is like zombocom
01:37.57gambolputtyexcept solve hunger and world peace of course
01:41.14TESTER2Someone knows a sip softphone for windows with a MWI feature?
01:41.15Chujiwtf is a zombocom heh
01:41.22Chuji~zombocom
01:41.40Chuji~google zombocom
01:44.18bkw_opus__, your show version is a liar
01:44.58opus__hmm
01:45.33opus__cvs 1.1 is weird, 1.2 won't compile.
01:45.44Qwellcvs 1.1?
01:45.48opus__they both looks abandond
01:45.55opus__cvs 2.0 is the shit, j/k
01:46.37SwKs/is the/is/g
01:50.39*** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146)
01:52.10opus__Still can not resolve realtime
01:52.33*** join/#asterisk file (~jcolp@mctn1-6383.nb.aliant.net)
01:55.35SwKis there any good sip monitoring tools that I can add to nagios out there?
01:56.09MavvieSwK: what do you want to monitor on sip-level?
01:56.10SwK(something that'll poke a sip proxy every then and again and lemme know that its crapping itself?)
01:56.46SwKMavvie: make sure it'll still forward a call and make sure I dont get a 4XX or 5XX error
01:57.16SwKa little deeper then
01:57.24SwKjust "poke" "ack"
01:57.35MavviePORTNAME=       sipsak
01:57.35MavviePORTVERSION=    0.8.11
01:57.35MavvieMASTER_SITES=   http://download.berlios.de/sipsak/
01:57.38MavvieCOMMENT=        Small command line tool for SIP testing
01:58.00MavvieI think that is the best I can give you.
01:58.03SwKyeah then I gotta write something around it heh
01:58.06SwKi'm lazy
01:58.28Mavvieyou use nagios, then you're not allowed to be picky on the extra features you want.
01:59.15SwKyeah
01:59.39SwKi dont have to use nagios
01:59.55Mavvie"is there any good sip monitoring tools that I can add to nagios out there?"
02:00.02Mavviesounds like it was a prerequisite.
02:00.07*** join/#asterisk X-Rob (~rob@dsl-202-173-151-24.qld.westnet.com.au)
02:00.08SwKyeah I know I said nagios but it doesnt have to be that
02:00.18X-RobDoes anyone have a SIP trace of a BLF session?
02:00.29SwKI was thinking of nagios as I've used it before
02:00.34X-RobGrandstream are happy to implement it, but they can't find any documentation on it.
02:02.18blitzragecan anyone verify that asterisk.conf is generated based on ifdef's in the Makefile ?
02:04.10citatsblitzrage: its generated when you do make samples
02:04.37blitzragecitats: thanks - I just found it in the Makefile too. I thought I'd seen it in there :)
02:05.37citatsyou dont have to rub it in
02:05.40blitzragelol
02:05.56blitzragecitats: well, I've been dealing with 35 C weather all week with just a pair of fans :)
02:06.56citatsi've only got one fan.  :(
02:07.04blitzrage:(
02:07.14citatsbut i did borrow a window a/c unit for the bedroom i put in this morning
02:07.24blitzragecitats: well you can have my dual-fan window fan since I have A/C now :D
02:07.43blitzragecitats: you just built a bedroom this morning?
02:07.49citatsheh
02:07.56blitzrageneat-o
02:08.12citatsheat getting to my brain.  talk like yoda i do
02:08.25blitzragehehehe
02:08.28blitzragecitats: where j00 at?
02:09.04citatsmichigan for now
02:09.27citats89 F in here right now
02:09.49blitzragecitats: ahhhh, not too far away from me then (Toronto). Used to live in Sarnia, driving to Flint to switch the dashed on grey market trucks for dealerships in the US.
02:10.20blitzragecitats: yah... its like 28 C here too
02:10.41blitzrageI live on the 13th floor of an apt. luckily, so the air is a bit cooler up here
02:11.19citatsespecially when you have a/c :)
02:12.28blitzragecitats: haha... well, I didn't know I had it. The vent was closed in my room because of the winter (I can't stand how how my room gets, window open most of the winter), but then I was walking by my room mates room and felt some cold air. Then I noticed his vent was pumping out A/C, so I opened my vents back up, and voila :)
02:12.52blitzragecitats: I'm sorry, I'll stop bragging now (but I know what its like to not have A/C)
02:12.55blitzrage:)
02:14.34citatsi've got to get central air and a new furance put in soon
02:16.23citatsi should just move my tv and run coax up to the bedroom so i can watch the pistons game in the one room with a window a/c
02:17.12*** join/#asterisk Cunk (~chatzilla@pool-70-22-232-156.bos.east.verizon.net)
02:17.16*** join/#asterisk santiago (~santiago@63.245.86.198)
02:17.17citatslast year we had less than a week of days a bit cooler than this the whole summer.  if this is the beginning i'm screwed
02:18.01blitzragecitats: yah... like you said, we've got more warm days this past week than all last summer
02:18.14blitzragecitats: game 7 tonight?
02:18.19citatsgame 1 of finals
02:18.29blitzragecitats: holy crap, I've been out of the loop! :)
02:18.42blitzragecitats: oh right, game 7 against Miami was on Monday
02:18.55blitzragecitats: guess they won :D
02:19.37blitzragewow... 33/31 with 4 mins left
02:19.42blitzrage(of the first)
02:19.45citatsyep, i missed most of it because i was on a plane.  caught most of the 1st qtr at the airport though.  i'm not a huge basketball fan, but something has to make up for hockey
02:19.48blitzrageerrr.. 2nd :D
02:19.49blitzragelol
02:19.59blitzragecitats: totally agreed! :)
02:20.13blitzragecitats: Go Leafs Go :D
02:20.49citatsi just love the playoffs in hockey.  at the beginning, every day you can watch at least 2 games
02:20.57blitzragecitats: yah, thats great
02:21.34*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
02:21.59blitzragedive! :)
02:25.40*** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca)
02:26.17*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
02:34.17*** join/#asterisk anthm (~anthm@CPE-69-76-83-52.wi.res.rr.com)
02:34.17*** mode/#asterisk [+o anthm] by ChanServ
02:36.58blitzrage~seen Manxpower
02:36.59jbotmanxpower <~eric@32.199-78-194.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 7d 19h 2m 51s ago, saying: 'when you set the callerid you normally do not include the dialing prefix (0 or 00) or the country code.'.
02:39.27bjohnsondamn.  sixtel news email says they just started a 24 hour support service.  I've been trying to get them to fix my DIDs for close to 2 weeks
02:39.34*** join/#asterisk user1fn (~joe@joe.dsl.1fn.net)
02:39.34TESTER2always bad pin number for a conference, any idea? it's set in meetme.conf
02:40.11xeet2bjohnson: hmmm, odd
02:40.15user1fnman oh man... I'm getting HDLC errors like crazy on my PRI
02:40.26QwellTESTER2: got a timing device?
02:40.34user1fntelco says it's my equipment, digium says it's my telco
02:40.55Ahewesanyone using a sangoma card?
02:40.56TESTER2Qwell: x100p
02:41.22xeet2bjohnson: what area are your dids out of?
02:41.56*** join/#asterisk PMantis (~PMantis@cpe-66-66-114-3.rochester.res.rr.com)
02:42.34PMantisI'm on Gentoo, bunch of new updates emerged, now: ZT_CHANCONFIG failed on channel 1: No such device or address (6)
02:43.38user1fnanyone got theories on why this would happen?
02:43.39user1fnchan_zap.c:7398 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
02:43.47user1fnwhen there are no calls on my pri
02:44.06xeet2user1fn: is the service working?
02:44.16xeet2are you able to pass calls in and out?
02:44.17user1fndrops calls, but works, yes
02:44.27xeet2mmm, you have some line issues then
02:44.32user1fnsome calls will work fine while others are dropped
02:44.33xeet2did you call your carrier?
02:44.51user1fnyes... they took my line down for 30 minutes testing and told me their stuff is fine
02:45.02xeet2did they loop it to the smartjack?
02:45.16xeet2what zaptel card are you using?
02:45.24user1fnnot sure... they said they were running "intrusive" tests
02:45.27user1fnt100p
02:45.44xeet2what do you have set as the timing source?  is the circuit the primary source?
02:46.14user1fnspan=1,1,0,esf,b8zs
02:47.03xeet2user1fn: who's the carrier?
02:47.08user1fnEschelon
02:47.58*** join/#asterisk zimdog (~zimdog@c-67-164-190-201.hsd1.co.comcast.net)
02:48.00xeet2ni2? d channel on 24?
02:48.02user1fnthey were getting pretty short with me since their "tests showed no problems"
02:48.11user1fnactually, using ni1
02:48.20user1fnwe had dialplan issues with trying to use ni2
02:48.37user1fnd is 24
02:48.50user1fnit's a fractional T1, so I have 1-12
02:48.54TESTER2so no idea why I always get "bad pin" number in meetme?
02:49.09shidoheh
02:49.20shidoyour meetme.conf configured right?
02:49.26xeet2user1fn: what kind of dialing issues did you have with ni2?
02:49.31opus__tester2 - make sure you can send audio through the channel from the phone, and also check dtmfmode
02:49.37Qwellshido: of course it is. ;]
02:49.40user1fncouldn't dial anything more than 7 digits
02:49.47user1fn<PROTECTED>
02:50.25user1fnor maybe it was that I always had to dial 10...
02:50.36user1fnit was an issue with local vs. long distance
02:50.40xeet2user1fn: what is your resetinterval set to?
02:50.52user1fnuh... where is that set?
02:50.59xeet2zapata.conf
02:51.17user1fnit's not set
02:51.23user1fndefault, I suppose
02:51.24xeet2set it to never
02:51.31xeet2you're running head?
02:51.51TESTER2opuss__: dtmfmode=rfc2833 and yes I can send audio
02:51.52user1fnI was, but rolled it back to 1.0
02:52.21xeet2ok so, let me get this straight.  dialing 10 digits, your carrier can't decide whether thats local or ld?
02:53.06*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3884759.sympatico.ca)
02:53.23user1fnI think I remember them saying it was that all numbers were being passed as local
02:53.25user1fnnever national
02:53.27DaLionhey since voip down can anyone point me to a ( if you know your party's extensiosn enter it know .gsm  ) ?
02:53.40user1fnso it would fail when the number was actually long distance
02:53.43xeet2user1fn: sounds like they don't know what they're doing
02:53.56user1fnheh... makes it hard for me to figure out what they need
02:53.59xeet2do you know what kind of switch you're connected to?
02:54.12user1fnwhen I can't tell them some hardware they're used to using, they just get silent
02:54.17user1fnnope
02:54.22DaLionoh god... weird meteo lady was once main news reporter.. they downgraded her lol
02:54.24tzangerhahahaha
02:54.26DaLionwrong chanel
02:54.28tzangerin one of hte other channels I lurk in
02:54.32DaLion<PROTECTED>
02:54.33bkw_tell me what you want from me...
02:54.35tzanger"I just took a big shit"  "Who'd you take it from?"
02:54.38h3xuser1fn: you probably need 1+10D
02:54.40bkw_would you like to dance with me?
02:54.57tzangeroh dear
02:55.00tzangerthere's a skunk outside
02:55.04tzangerI wonder if the cat's gonna get sprayed
02:55.04xeet2h3x: the carrier should figure that out on their own though, on ni2 at least
02:55.16h3xni1 and ni2 has nothing to do with dial plan issues
02:55.26*** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com)
02:55.33user1fnI was passing the same number of digits you would normally pass
02:55.37user1fn7 digits for local
02:55.43h3xni2 adds some things like 3rd party transfer, network side hold, etc
02:55.44user1fn1+10 for ld
02:56.01user1fnI never changed anything and it worked on ni1
02:56.07h3xhowever, you need to set the flag for national stuff
02:56.09zimdogDoes anyone have any experience connecting * to a broadsoft switch?
02:56.18user1fn(except for the switchtype of course)
02:56.20h3xthat is just because ni1 is implemented differently in zaptel
02:56.22tzangeryou almost ALWAYS want 'unknown' for your dialplan
02:56.27xeet2h3x: hmmm, didn't know that
02:56.52h3xWhat did you set the pridialplan line to?
02:57.09user1fnwe tried it on national, local and unknown
02:57.12h3xand prilocaldialplan
02:57.25user1fnright now I have it commented out, so default again
02:57.36h3xI bet the real problem is you need overlapdial=no
02:57.47user1fnno idea what that is
02:57.59xeet2I've never had to pass 1+10D for ld calling on a pri, just 10 has always worked, so this is new to me =)
02:57.59h3xoverlapdial=yes sends one digit at a time, potentially
02:58.11h3xno overlap means it has to send all digits at once
02:58.13h3xto set up a call
02:58.22h3xI have no idea why it defaults to yes in extensions.conf
02:58.22user1fnah
02:58.34xeet2h3x: but he's getting *active* calls disconnected though, after the call set up
02:58.44h3xWhat?!
02:58.52user1fnyeah... dropped calls
02:58.56xeet2I was only asking about what his ni2 dialing problems were =)
02:59.00xeet2has nothing to do with the real problem
02:59.07h3xdoes it do that on ni1 as well?
02:59.14xeet2yeah
02:59.20user1fnit's only ever worked on ni1 with this telco
02:59.20xeet2fcs errors on d channel
02:59.21h3xthen your t1 is probably not clean
02:59.27user1fnused to work on ni2 with Qwest
02:59.38h3xloopback test that t1
02:59.41user1fnI get HDLC errors (8 and 6)
02:59.42xeet2his carrier did a loop and ran tests and said it was clean
02:59.49h3xloop it the other way
02:59.57user1fnthe other way?
03:00.00tzangerxeet2: clock slips (are you syncing to the line?)  overloaded CPU/system?
03:00.06h3xyeah send a loopback code towards trhem and test it
03:00.10TESTER2is there some issues with the meetme in v1-0?
03:00.15h3xalthough he probably dosent have the equipment to do that
03:00.15user1fnsystem load is at .03
03:00.26h3xOh Qwest eh. is this Qwest local or LD?
03:00.43tzangeruser1fn: run vmstat 1 for about a minute and note interrupts/context switches and CPU loading for spikes
03:00.45user1fnused to be qwest
03:00.48user1fnit's now Eschelon
03:00.49h3xlocal i take it since you were talking about local calls
03:01.17h3xthey sold the lec side?
03:02.23user1fnforgive my ignorance, but what denotes a context switch?
03:02.39xeet2user1fn: when you get a call disconnected, do *all* of the calls on the pri drop or just 1 or 2?
03:03.15user1fnI get a series of errors and then an error saying it can't find the D channel
03:03.20user1fnI'm assuming all of them
03:03.30user1fnbut I'm not absolutely positive
03:03.33h3xSo is eschelon reselling Qwest/USWest T1s?
03:04.01user1fnI think so... our cross connect was installed by Qwest techs
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03:04.19xeet2user1fn: you said this is a fractional t1...  is it really fractional or they' just aren't using 13-23?
03:04.38user1fnthey says it's fractional
03:04.49user1fnbut I'm not sure of their implimentation
03:04.50xeet2hmmm
03:04.57h3xI don't even think the LEC side of Qwest sells an integrated product.
03:04.58user1fnadding more lines is extremely cheap
03:05.10xeet2its really weird to have a true fractional circuit and still use channel 24
03:05.15h3xIs it eschelon T1 and using Qwest for the local loop maybe?
03:05.21h3xxeet2: No its not
03:05.30user1fnthat sounds more like it to me
03:05.33h3xThey do that because those stupid 5ESS and DMS switches dont get it if it aint on 24
03:05.37user1fnthey claim to use all of their own equipment
03:05.52h3xuser1fn: did they say what kind of switch they have
03:05.57user1fnno... sorry
03:06.04user1fnI can probably call back and ask, though
03:06.05xeet2h3x: well then its not a true fractional circuit.  fractional t1 ports just plain don't do all 24 channels
03:06.16h3xIf NI-2 really dosent work you should be using a DMS or 5ESS switch type, not NI-1
03:06.21h3xNI-1 is really designed for BRI
03:06.43xeet2alot of services are sold on "fractional" t1s but are really on full t1 ports
03:06.48h3xxeet2: its actually an integrated circuit if he is doing data on the other ds0's
03:07.10xeet2h3x: right
03:07.19h3xin any case, putting the D channel on 24 is a standard thing no matter how many DS0s are eaten up for B channels
03:07.25edy2is there a website that can tell me all about this wan network interconnects like whats a fractional T1,eschelon,qwest stuff
03:07.40h3xMostly because stupid ass PBXs like nortels and avayas and all that will not LET you put it on any other channel
03:07.46xeet2sure, I had never heard of a pri on a true fractional t1 port before, didn't think it was possible because of the whole channel 24 d channel thing
03:08.07h3xI've set up a bunch of fractional PRIs before with like only 4 B + D
03:08.08tzangerI have a PRI with only 15 Bs turned up
03:08.25h3xmost all ILECs will not let you set up anything but 23B+D
03:08.35h3xbecause they dont have a tariff for integrated or fractional
03:08.38tzangerh3x: I'm through Bell Canada.. definitely an ILEC
03:08.44xeet2h3x: yeah, but your t1 port itself is capable of 24 channels, so its a full t1 port, just not full t1 service, right?
03:08.46edy2are there pris on t1s too
03:08.55h3xxeet2: We're talking marketing terms here :)
03:09.00xeet2ah
03:09.09xeet2well then we might as well just throw everything out the window then =)
03:09.13h3xhaha
03:09.24h3xmarketing person> "all you need is a fat pipe and you are good to go"
03:09.27xeet2edy2: pri's are delivered on t1, e1, t3, etc...
03:09.42xeet2h3x: ugh, I really did hear that today too
03:09.51h3xuser1fn: by the way, where are you at anyway
03:09.55user1fnhe he... I'm actually calling to ask about their switch
03:10.01user1fnUtah
03:10.05h3xwhat city
03:10.11newmedianMega link. Micro link. let's rename everything.
03:10.19user1fnit's a DMS500
03:10.24xeet2mmmm
03:10.28user1fnSalt Lake
03:10.36h3xare you doing inbound and outbound local?
03:10.38user1fnour equipment is in a data center there
03:10.40edy2xeet2:PRI is supposed to be a lot of B channels and one D channel (it prolly has two variations like 30+2 or some other)
03:10.42TESTER2is there some issues with the meetme in v1-0? (always bad pin even if no pin or force pin in the extensions.conf)
03:11.03user1fnI believe so
03:11.08edy2xeet2:cant PRI run on ISDN
03:11.19xeet2edy2: why are you telling me this?  *you* asked the question
03:11.26xeet2edy2: PRI *is* isdn
03:11.29tzangeredy2: pri IS isdn
03:11.35tzangerBRI IS isdn
03:11.40xeet2haha
03:11.41tzangeryou have your terms all confused
03:11.52edy2yeah that i agree. bri is isdn
03:11.59edy2but i thought u cud do pri on isdn as well
03:12.09edy2like isdn had two flavors: bri as well as pri
03:12.17tzangeredy2: PRI is a signaling protocol (ISDN) on top of a phsycial link (T1/E1)
03:12.19h3xuser1fn: have you considered getting VoIP in/out and sell your T1 card on ebay :P
03:12.33tzangeredy2: BRI is a signaling protcol (ISDN) on top of a physical link (no idea what this is called)
03:12.35edy2and i thought T1/E1.... are just big pipes (we decide the type of signalling on em)
03:12.36Mavvietoo
03:12.36edy2right?
03:12.37user1fnlol.... I'm beginning to
03:12.41tzangerMavvie: yes it is
03:12.44tzangerall my DS3s are ISDN
03:12.46Mavvieaha.
03:12.58user1fnthrough whom, h3x?
03:12.58newmedianall your DS3s are belong to ISDN
03:13.01Mavvienot that I have one of these anymore.
03:13.04tzanger4 NFAS groups gets me 668 B channels and 4 D channels on a DS3
03:13.06h3xuser1fn: let me look here...
03:13.17edy2tzanger:uu r so cruel, now what are these DS3 :(
03:13.17tzangeredy2: nope
03:13.33xeet2edy2: a ds3 is 28 t1/ds1 channel
03:13.34tzangerE&M is a signaling.  LS, LSCPD, GS... all signaling
03:13.34xeet2s
03:13.38TESTER2someone can confirm me that CVS-v1-0-06/09/05-19:06:07 works for meetme?
03:13.43user1fnby the way... that server is hardly doing anything at all
03:13.47edy2wow
03:13.48user1fnshowed VERY minimal usage
03:13.53h3xuser1fn: blah.. whats a area code and exchange out there
03:13.56tzangerPRI is a way to take all the signaling bits and put them in ONE spot so your voice/data channelsa re "clear"
03:14.05user1fnI have it working TESTER2
03:14.08tzangerall the other signaling patterns steal bits from the voice path every 6 frames
03:14.10edy2tzanger:u r a one happy man then
03:14.16edy2thanks xeet2, i can always count on you
03:14.17tzangeredy2: whys' that?
03:14.20user1fnarea code is 801
03:14.27edy2tzafrir:so much bandwidth
03:14.32tzangeredy2: not really
03:14.39xeet2hehe
03:14.40edy2heck,
03:14.42tzangerI've got about 3300 voice channels but that's not bandwidth
03:14.57tzangerjerjer's the guy with more bandwidth than god :-)
03:15.05tzangerDS3's "only" about 45mbit/sec
03:15.09edy2u can play with a greater number of calls than otherwise if u had few channels
03:15.24xeet2you can buy an oc3 to level3 in the dc area for about 10k a month
03:15.38xeet2am I the only one that thinks thats crazy?
03:15.52tzangeris that OC3 for data or voice and does it contain bandwidth
03:16.01edy2xeet2:be my teacher man! i love reading over web, how do i know of all these terminonologies wan interconnect related
03:16.06edy2paste a url
03:16.06xeet2data connectivity to the level 3 ip network
03:16.10edy2i ll read to the end
03:16.12tzangerxeet2: nice
03:16.19edy2including all links
03:16.40xeet2its all because of all the government contracts, the spent a ton of money running fiber everywhere
03:16.58xeet2you can get a ptp oc12 for about 25k
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03:17.38xeet2and about 100k will build you your own private sonet ring
03:17.42edy2how much bandwidth is on an OC-x
03:17.55xeet2oc3 is 155mbit minus overhead
03:18.02xeet2oc12 is 622mbit minus overhead
03:18.08edy2wow atm
03:18.08h3xOne of my neighbors has VoIP did's and stuff to/from salt lake city
03:18.14tzangerOC3 is what, 3 DS3?  I forget how that goes
03:18.43user1fnhmmm... can our numbers be ported to a VoIP service?
03:18.45edy2atm hook up offer same 622megs too, right?
03:18.51tzangerDS0 = 64k  DS1 = 24xDS0  DS2 = 4xDS1  DS3 = 7xDS2  OC3 = 3xDS3?
03:18.59h3xI have an OC-12 to Qwest for $400 a month :P
03:19.04xeet2you can fit 3 ds3's into an oc3 but the signalling is a bit different, it doesn't directly translate
03:19.06h3xthe transport piece
03:19.09tzangeryeah
03:19.15h3xlong distance side of course
03:19.20PMantish3x: OC-12?????
03:19.25tzangerwell DS3s don't directly translate to DS1s either because there are slop bits so the DS1s don't have to all be synchronized
03:19.42h3xYeah, well they charge the same price for all OC-x circuits if you are on-net with them so that made the most sense
03:19.57user1fnso your best guess would be that there are line problems, right?
03:20.02h3xI didn't really want to spend $100k on a OC-48 line card for my Cisco ONS 15454 if I don't really need it
03:20.32xeet2user1fn: yes.  try putting your zaptel card in a loop and have them run full intrusive testing
03:20.46user1fnhow do I put it in that state?
03:20.52h3xthey can loop it, zaptel t1 cards emulate a CSU
03:20.54*** join/#asterisk IQ (~IQ@63-230-44-230.omah.qwest.net)
03:20.55xeet2I hate those boxes, they're so picky
03:21.11*** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc)
03:21.12xeet2that and half the cards cisco released didn't really do what they were sold to do
03:21.14dan2PMantis: yo
03:21.17dan2PMantis: whats going on
03:21.27PMantisdan2: Check Jabber. :)
03:21.33user1fncool... thanks for all of the advice
03:21.34dan2PMantis: its running fine here
03:21.35edy2xeet2:can u suggest some reading on wan terms
03:21.42h3xxeet2: well its not really a cisco product
03:21.46h3xit was designed by cerent
03:21.50user1fnthese kinds of problems are frustrating to no end
03:21.56xeet2h3x: right, but the cards they released after cisco bought them
03:22.04edy2where i can find out what the hell is t-x, oc-x,ds-x,....all related stuff
03:22.06h3xwhat like the XC-VT?
03:22.13PMantisdan2: I was just gonna talk about how much Broadvoice sucks - then *YOU* had to show up. LOL
03:22.15newmedian~docs
03:22.16jbotDocumentation 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
03:22.19xeet2for instance the 1 port 802.1q capable ethernet bridge
03:22.22h3xedy2: get a book like telecommunications handbook on amazon or something
03:22.43dan2PMantis: fix your jabber
03:22.46xeet2every time it got a frame for a mac address it didn't know about, it just dropped all frames passing in the box until it figured out where to send it
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03:23.09xeet2when we complained, they said "well thats just what it does"
03:23.10newmedianedy2 e.g. http://en.wikipedia.org/wiki/DS1
03:23.21xeet2"you sold us a switching card, and its obviously not doing real switching"
03:23.26edy2newmedian:opening...
03:24.37xeet2after complaining enough they finally released the dummy 4 port card that didn't do switching, but could still be connected in a circuit with sonet
03:24.46xeet2and swapped them all out
03:24.46opus__xeet2 - can't you do static mac table?
03:25.08xeet2opus: in this case, we're talking about a mac table that was about 20k entries
03:25.18opus__that sucks
03:25.22edy2newmedian: thank you, i am enjoying reading it.
03:25.44newmedianwelcome.
03:25.49xeet2opus: yes, as did the inter-datacenter bridging network at mci hosting for the first 8 months it was up
03:25.55opus__voip-info.org still down lame
03:26.07brookshiregoogle cache?
03:26.08opus__xeet2 whoah
03:26.09brookshire:D
03:26.24opus__brookshire yeah but there's no updates:)
03:26.41brookshiremaybe we kilt it
03:27.35xeet2h3x: never used the xc-vt
03:28.19xeet2we did get stuck with the xc-10g when we used the g1000-4 cards
03:28.30xeet2then of course they eol'd it
03:29.08xeet2I swear, cisco buys companies to screw up the products and then spend years regaining the market
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03:33.56xeet2damn it got quiet
03:34.58xeet2hehe
03:35.28SwKis voip-info down again?
03:35.40Sedorox~wiki-status
03:35.41jbotsomebody said wiki-status was Slow like normal
03:35.56Sedoroxjbot: no... wiki-status is Down Again
03:35.57jbotSedorox: okay
03:36.01Sedorox~wiki-status
03:36.02jbotwiki-status is probably Slow like normal
03:36.06SedoroxHmmm
03:36.16SwKhah
03:36.40SwKjbot ~polycom
03:36.54SwKdamn it I need the manual for the new 1.5.1 firmware on the polys
03:37.17SwK~polycom
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03:40.05blitzragecitats: wow... S.A. is really starting to run away with it now
03:40.47newmedianedy2: this might be useful for you: http://resource.intel.com/telecom/support/documentation/learnabout/documents/t1e1prim.pdf
03:41.04citatsblitzrage: yeah i'm thinking over now.  never would have thought at it would be like this after the 1st
03:42.12TESTER2~wiki-status
03:42.13jbotmethinks wiki-status is Slow like normal
03:42.19edy2newmedian:dloading...
03:42.31edy2oh super
03:42.53edy2newmedian:its all i wanna know about t1/e1,reading...  thanks a ton
03:42.53newmedian:) thought you might like that one.
03:43.01newmedianwelcome.
03:43.16edy2yea i do
03:43.35edy2actually a lot more as info here is intel provided
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03:58.34blitzragecitats: agreed
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04:06.58libpcphi all
04:09.52newmedianjbot seen all
04:09.53jbotall <WeezeyD@206.210.109.233> was last seen on IRC in channel #asterisk, 63d 9h 6m 37s ago, saying: 'hello'.
04:10.55pjzis it possible to set a polycom 500 to autoanser? a line?
04:12.06pjzwiki-status is Down Again
04:12.14pjz~wiki-status
04:12.15jbotit has been said that wiki-status is Slow like normal
04:12.17SwKpjz: yes
04:12.43Sedoroxjbot: no, wiki-status is Down Again
04:12.45jbotSedorox: okay
04:12.49Sedorox~wiki-status
04:12.52jbotwell, wiki-status is Down Again
04:12.53SwKyou need to change a setting in sip.cfg and on ipmid.cfg and set a variable in asterisk before calling dial()
04:12.56Sedoroxyay
04:13.57pjzSwK: ah, okay
04:14.13pjzSwK: can more than one phone register with the same login/password?
04:14.19pjzSwK: at the same time, I mean
04:15.04SwKno
04:15.13SwKpjz: what are you trying to do?
04:15.21SwKall call intercom?
04:15.25pjzyes
04:15.27SwKhah
04:15.56SwKtheres a nasty hack of a way to do that with meetme but most consumers hate it cause they never get used to having to a delay
04:16.22pjzah
04:16.34pjzoh, that reminds me - I need to add meetme.cfg generation to my little app :)
04:18.52SwKheh
04:19.44wazany hope of getting asterisk working with ATT CallVantage?
04:19.45SwKpjz: in sip.conf <alertInfo voIpProt.SIP.alertinfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class="4"...>
04:20.09SwKthats what you want to look for... it wont be set to "Auto Answer" and the class will be blank
04:20.30pjzand that will make line 1 Auto Answer?
04:20.45SwKset it to something and the class is taken from alertinfo section of ipmid.cfg
04:21.17SwKnow to make that fire you need to set the _ALERT_INFO variable from the dialplan prior to calling dial
04:21.25SwKthats in head anyway
04:21.33pjzohh
04:21.41pjzis it in stable?
04:21.45SwKin stable it might be just ALERT_INFO (not sure havent ran stable in forever)
04:21.50pjzah
04:21.56SwKyeah its in stable too
04:22.02pjzcool
04:22.06SwKbut I'm not sure if they backported the update on the variable name
04:22.34*** join/#asterisk techie (gus@antibala.com)
04:24.46SwKpjz: you gotta poly config generator script?
04:25.29pjzyeah
04:25.34SwKcan you share it
04:25.47SwKkeep me from having to write one heh
04:25.49pjzit's totally trivial variable replacement tho
04:25.59SwKthats pretty much what I've done
04:26.16SwKbut it just adds button settings for lines
04:26.38SwKi been meaning to add multi-button/registration but havent done that yet
04:26.50pjzI've got a python script that gens: extensions.conf, voicemail.conf, sip.conf, all the poly files, etc etc etc based on some declarative stuff
04:27.23SwKmine is a php script that takes input from the browser and does just the poly configs
04:27.31pjzah
04:27.44SwKbeen thinking about making it more complete and throwing it on the web so you could download the generated XML
04:28.47pjzwell, my python thing gens even the per-phone files
04:29.02pjzsince it knows everything about everybody
04:29.14pjzit can fill in their login/password, etc.
04:29.17SwKmine gens the per phone files... feed it mac, username/password and what button
04:29.32pjzwhat do you use multiple lines for?
04:29.43SwKwell really you feed it what poly model and then it gives you the right number of blanks for things
04:29.48pjzah
04:30.17SwKwith a little work it could even gen ipmid and sip.cfgs
04:30.27SwKbut they merged a few things in the new firmware
04:30.29pjzI've tried to design a fairly general purpose script
04:30.37SwKjust a phoneX.cfg and ipmid.cfg now
04:30.59pjzoh, yeah, I don't bother to generate those since they're essentially global and don't change per-user
04:31.02SwKi could add other phone support into mine for things like cisco and sipura configs
04:32.25pjzyeah, mine has UIPand IAXy cfg support
04:33.08pjzI need to maske it do meetme stuff and dnsmasq dhcp cfg stuf
04:35.09pjzbut I'm not sure what I've got is suitable for use by anyone but me :)
04:36.29Godseyhow do you enable host cloak?
04:36.51Godseyrandom freenode question;0
04:37.17*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:38.56SwKhah
04:39.01SwKpjz nice
04:39.04SwKwhat do you use that with?
04:39.52SwKGodsey what you mean cloak? like have your host cloaked like ${asterisk_developer}
04:40.32*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
04:40.37Godseyya contributor
04:40.41Godseyfor freenode
04:40.48Pete_Largois voip-info.org down?
04:40.56SwKPete_Largo yes
04:41.02Pete_Largothat sucks
04:41.10newmediangoogle cache time, Pete_Largo
04:41.36Pete_Largonot for good, I hope...
04:41.39xeet2hey pete
04:41.43SwKthe voip-info dude really needs to get some mirrors
04:41.47Pete_Largohey
04:41.54GodseyPete_Largo: high availability is for story books
04:41.58Pete_Largolol
04:42.00kb1_kanobe"VOIP-info.org will be down for a few hours on Thursday June 9 for a system
04:42.00kb1_kanobeupgrade."
04:42.06SwKi know mirroring wikis is a pain in the ass but static mirrors are better then no mirrors
04:42.15newmedianGodsey: http://freenode.net/faq.shtml#pdpc_cloak
04:42.18*** join/#asterisk DaLion (~DaLion@69.156.64.3)
04:42.21*** join/#asterisk af_ (~af@ip-130-170.sn2.eutelia.it)
04:42.32SwKxeet2: I've offered and was turned down
04:43.06Godseynewmedian: the instructions said to find a helper to turn it on
04:43.10Godseybut now how to find one :P
04:43.20SwKGodsey prolly find an oper
04:43.30Godseystaffer not helper
04:43.30Godseyheh
04:43.48GodseyI did /stats o over and over trying to get someone's attn to no avail ;)
04:47.27TESTER2conf => 1,1234,1        and         exten => 8005,1,Meetme(1||1234)         nothing else?
04:47.39*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
04:49.05pjzI grok conf => 123 and conf => 123,456 but what's conf => 123,456,789 do?
04:49.27pjz123 is conference id, 456 is PIN for the conference
04:49.31pjzwhat's 789 ?
04:49.42TESTER2admin pin
04:50.02pjzwhat's the default admin pin?
04:50.07pjzor is everyone admin by default?
04:50.39pjzSwK: oh, hey, how do I set up MWI/voicemail buttons for the sp500 ?
04:52.12SwKwell MWI is just the mailbox setting in voicemail.conf mailbox=vmbox@vmcontext
04:52.43SwKfor the VM button actaully call into voicemail theres a setting for that in phoneX.cfg
04:52.58SwKi for get what it is... grep it from mail heh
04:53.37SwKerrr not voicemail.conf on the mwi but sip.conf
04:53.41pjzright, but where do I have it call? their sip address?
04:54.17SwKno the mwi works via a SIP NOTIFY message
04:54.35SwKand it "just works" when you set the mailbox on the sip peer definition
04:54.40pjzah, okay
04:54.48pjzsweet
04:55.03*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
04:55.23SwKyeah the call into VM thing is pretty nice, just but the extension they dail for VMMain() in there and it "just works" (tm)
04:55.58pjzmm... not sure which one that is
04:56.12pjzeveryone has thier own Vmail ext
04:56.20SwKwith the 1.5.1 firmware (this may work in 1.4.x but I am no so sure) you can restrict calls/button to a number 1-8 and have multiple buttons/registration
04:56.23pjzthat they call to check it right now
04:56.57xeet2is there an app to change the codec for the current call?
04:56.59*** part/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
04:57.07SwKso everyone calls their own extension to check VM? you dont have something like exten => 5555,1,VoicemailMain
04:57.17xeet2or the codec to use on a dial command
04:57.21SwKxeet2 not really but there are ways to do that
04:57.36SwKlike put it on hold change the codec on your UA and the pick it back up heh
04:57.55xeet2unfortunately its for a zap card
04:57.56SwKbut w/out reinviting a call you just cant change the codec midcall
04:57.58kb1_kanobexeet2: not really, though there are patches in mantis to do that. I had to use multiple peers.
04:57.59xeet2iax <> zap
04:58.07xeet2mmm
04:58.11pjzSwK: no, exts are 5xx, their vm is 6xx
04:58.21kb1_kanobexeet2: you want to change the codec on the iax side, yes?
04:58.22xeet2was hoping to avoid multiple peers but I guess I can't
04:58.27xeet2yes
04:58.34SwKon
04:58.39pjzSwK: I can make a vmailmain if needed though
04:58.52kb1_kanobeare you doing fax/data over ip or a low bandwidth backup route?
04:59.13xeet2I normally use g726 so thats the first codec this particular * box combination is using first
04:59.19xeet2but for fax calls I need to use ulaw
04:59.20SwKwit a VMMain you can still have them go straight into VM based on CID
04:59.31SwKw/out login
04:59.32pjzah
04:59.37pjzthat's cool
04:59.40pjzhow do I set that up?
04:59.50pjzthat would be nicer than having to dial 6xx
04:59.54pjzer, *6xx
05:00.24*** join/#asterisk lehel (~lehel@82.79.20.17)
05:00.27pjzor is it automagic without configuration?
05:00.29lehelhello
05:00.34|Vulture|pjz: I do extensions 1xx and VMs to *1xx
05:00.35kb1_kanobexeet2: I have three iax peers on each server, a normal one, a -high suffix (ulaw) and a -low (gsm). Then I can dial() as required... I also use a different source address for each peer so I can use different routes on the network depending on the codec.
05:01.14xeet2in this case I don't have the option of multiple ip addresses
05:01.21pjz|Vulture|: well, xxx numbers are all dialed out, *xxx goes to the vm system
05:01.29xeet2so I'm going to wind up with two peers between the same two boxes
05:01.29pjz|Vulture|: er, to the pbx
05:01.33|Vulture|pjz: ah
05:01.50SwKpjz only dialplan configs
05:02.08pjz|Vulture|: the guys who use the phone a lot didn't want a prefix for outbound stuff
05:02.16SwKunless you want to transfer people straight into someones VMBox
05:02.39pjzSwK: yeah, I'm learning.  I guess I could do xxx with a timeout in the dialplan
05:02.49*** join/#asterisk Cresl1n (~matt@216.207.245.23)
05:02.51*** join/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au)
05:02.58pjzSwK: ah, yeah, that's true, if I go to one vmail ext then that becomes difficult
05:03.05pjzSwK: so I'll stick with what I've got
05:03.12pjzSwK: will it still work witht he vmail button?
05:03.23kb1_kanobexeet2: not a problem. It'll still work. Just don't try and use the 'friend' directive in iax.conf.
05:03.33xeet2mmm
05:03.36xeet2one is already a friend
05:04.51SwKif they want to check their VM?
05:05.44kb1_kanobexeet2: seperate it out into the two halves, otherwise the codec order is unpredictable.
05:06.47kb1_kanobexeet2: have the type=user and type=peer for each on both sides. Thus, two codecs, two hosts; four defines.
05:07.01pjz<PROTECTED>
05:08.01kb1_kanobeduh, that makes no sense. forget I said the last bit... :-/
05:11.29SwKpjz: thats easiest for your configs to set up 1 extensions
05:11.37SwKand call just that 1 extension from the button
05:11.43SwKthe messages button that is
05:13.39pjzSwK: I can do either
05:13.44pjzSwK: just as easily :)
05:14.03pjzSwK: I just edit my generator script one way or another and do 'make'
05:16.13SwKheh
05:17.25SwKsomething like exten => 555,1,VoicemailMain(s${CID}) will throw the caller into their own VMBox... (the variable is the wrong one cant remember what it is off the top of my head
05:17.31SwKbut its the CallerIDNumber
05:18.36pjzSwK: ah, of course
05:18.39kb1_kanobeCallerIDNum
05:18.46SwKyeah thast it
05:18.54pjzSwK: well, you're presuming that you use extensions for SIP names
05:19.04SwKnope
05:19.06pjzer, SIP logins I mean
05:19.08pjzno?
05:19.17SwKit assumes you have caller ID properly set for the extensin
05:19.27pjzoic
05:19.41pjzshouldn't caller ID be the whole phone number though?
05:19.41kb1_kanobeFYI, CallerID = "CallerIDName <CallerIDNum>"
05:19.51SwKthe s tells it not to prompt for password
05:19.54pjzwe all have direct inward dial names
05:19.58pjzer, numbers
05:20.14SwKso voicemailmain(s100) whould throw you into mailbox 100 w/out prompting for password
05:20.25pjzso x555 => 123-456-7555
05:20.33SwKand?
05:20.54pjzso caller ID for my extension is 123-456-7555
05:21.00pjznot just 555
05:21.10SwKI set CID for outbound calls rightbefore I dial the outbound leg... intra company calls are just internal extension
05:21.15pjzso that becomes voicemailman(s123-456-7555)
05:21.27SwKnot nessecarily...
05:21.52SwK${CallerIDNum:8:3)
05:22.01pjzoh, I guess I could do like ${CID:... yeah
05:22.01SwK:ofset:len
05:22.21pjzor ${CID:-3} if it handles negative offsets correctly
05:22.28TESTER2meetme without password works, but doesn't work with pin
05:22.37SwKor is it :len:offset ? hah
05:23.07pjzSwK: it's :offset:len
05:23.14SwKheh
05:23.21SwKthats why i have dialplan cheatsheets
05:23.26SwKi never can remember that shit
05:23.39pjzSwK: b/c to strip leading *s I do goto(newcontext,${foo:1},s)
05:25.11TESTER2meetme without password works, but doesn't work with pin, any idea why?
05:25.32pjznope
05:26.16TESTER2it works once, but not... always bad pin
05:26.23TESTER2not=now
05:26.25pjzSwK: so the problem with this cfg generator is that it's really hard to separate out the customized parts from the generic parts
05:26.40SwKheh
05:26.46pjzSwK: I mean, I don't want to tie anyone into my scheme for doing stuff
05:26.54SwKyeah
05:27.01SwKthats always a problem
05:27.13pjzSwK: b/c asterisk is just *so* customizable
05:27.36SwKhah
05:27.45pjzit really is, if you think about it
05:28.03SwKasterisks greats asset: flexibility... asterisk greatesy liability: its flexibility
05:28.23pjzN-digit extensions, some unknown-but-huge limit on contexts
05:29.43*** part/#asterisk TESTER2 (~Cyber@modemcable219.42-81-70.mc.videotron.ca)
05:33.19*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
05:35.42*** join/#asterisk cjk (~cjk@80.92.64.103)
05:36.08cjkhi, how can i tell my asterisk boxes to not natively bridge between iax channels
05:36.22SwKcjk turn it off in iax.conf
05:36.46cjkSwK: ok that easy thanks
05:37.55*** join/#asterisk shidan (~shidan@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com)
05:38.18cjkSwK: whats the name of that parameter?
05:39.00drumkillanotransfer=yes
05:40.32cjkdrumkilla: thanks
05:42.13pjzanyone mess with digium 'IAXy' S100I boxes?
05:42.26pjzdo you generally just assign them an IP?
05:42.51pjzkind of have to, I guess
05:44.18Cresl1npjz: dhcp baby :-)
05:44.36pjzCresl1n: then how can you know what IP to use iaxyprov on?
05:44.50Cresl1ntcpdump port 9999
05:44.59Cresl1nor, use the new gtkiaxyprov :-)
05:45.05pjzCresl1n: that's a manual process, how do I automate it?
05:45.11Cresl1nsorry, udp port 9999
05:45.19Cresl1npjz: what do you mean?
05:45.26*** join/#asterisk Landrocker (~landrocke@60.234.145.243)
05:45.49pjzCresl1n: well, with the phones, they can dhcp up and I don't care what their IP is b/c they just come and fetch their configs from my tftp server
05:45.57pjzCresl1n: so I can just generate those configs too
05:46.12Cresl1nlol, there isn't anything like that on the iaxy
05:46.18pjzCresl1n: with the IAXy, I can generate its config just fine, but there's no way to feed it to the iaxy
05:46.21pjzCresl1n: right
05:46.42pjzCresl1n: and I can't just feed it to a given IP address b/c the iaxy might not be there yet or whatever
05:47.07cjki did the notransfer=yes on both sides but still i get this -- Attempting native bridge of IAX2/35227273045@35227273045-22 and IAX2/ECO.NSP0.VG-37
05:48.25pjzCresl1n: how often do you have to iaxyprov?
05:48.39pjzCresl1n: just whenver you change the config?
05:48.45Cresl1nyeah
05:48.50pjzhrm
05:48.50Cresl1nthat's pretty much it
05:50.35pjzI'm trying to make my config process foolproof
05:50.44pjzwell, at least automated
05:51.18SwKcjk did you reload chan_iax after you changed the config?
05:51.36Cresl1npjz: you could write a config daemon that sits on a box and automatically does that
05:52.10Cresl1nsniffs udp 9999, if (received_packet) get_macaddress();
05:52.17cjkSwK: yeah restarted even asterisk
05:52.31Cresl1nthen just have a config file that contains all the definitions for each mac address
05:53.11pjzCresl1n: enh, nah, I can just make the config take both a mac and an IP as a param, then I ping the IP, then check the arp table to make sure it's the right device, then run iaxyprov
05:54.07pjzthat's a lot to do though
05:54.35pjzfor a fairly rare use - I only have one iaxy on my whole network :)
05:55.16*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
05:56.30drumkillagtkiaxyprov finds iaxy's on your network for you :)
05:57.48pjzSwK: re: customizablility: it's not just asterisk... even the phone configs... any text config basically either 1) doesn't support being able to config all the features of the phone or 2) is essentially the phone's own config file
05:57.51drumkillaCresl1n: we need to release that ...
05:59.41Romik_somebody knows how to get notification about channel status changes? rining/up/busy/hangup/etc ?
06:00.10SwKRomik_: manager API
06:00.58Romik_SwK: as i found there only polling "channel status xxx" , i need get like even notifications...
06:01.50brookshirerussell, goto bed :)
06:01.56drumkillabrookshire: you first
06:02.04SwKRomik_ you'll probably have to write something then
06:02.11brookshire*pfffft*
06:02.13brookshirei just woke up
06:02.16drumkillaha, bum
06:02.29Romik_SwK: what do you mean - i have to write something? asterisk module?
06:02.36drumkillaRomik_: you shold be able to receive events from the manager interface
06:03.26Romik_drumkilla: but there no such evens as line changing status....notifications....only newexten on originate and hangup, as i found...
06:03.38drumkillaI'd have to look
06:03.45drumkillabut it is event based, not just polling based.
06:03.59brookshirewhere can i get gtkiaxyprov?
06:04.00brookshirehehe
06:04.14drumkillabrookshire: I have it in cvs somewhere
06:04.17Romik_<PROTECTED>
06:04.31brookshiremark said i had to make a homepage for it
06:04.33brookshire:(
06:04.36drumkillabrookshire: hahaha
06:04.41drumkillalike that's hard to do
06:04.53drumkillawe need to get mark to approve it
06:04.54brookshirei'll get collin to do it, yay!
06:04.57drumkillathen i can put it in digium cvs
06:05.02Cresl1nyeah baby
06:05.07Cresl1nlet's get collin to do it
06:05.11drumkillahaha
06:05.20brookshirerussell made collin look something up for him today
06:05.22Cresl1nit's got to have a cool logo
06:05.24Romik_anybody know avaliabily of new quad digium cards with echo cancelling?
06:05.33Cresl1nsoon?
06:05.34Cresl1n:-)
06:05.38drumkillaRomik_: should be available verrrrrrry soon
06:05.42drumkillalike, super soon
06:05.49Cresl1nI think they're trying to release them super-duper soon :-)
06:05.51brookshireas soon as they get delivered from the manufacturing place
06:06.00drumkillalike, in 3 minutes
06:06.04Romik_drumkilla: heheh
06:06.28Cresl1nJackpot!!!!
06:06.33brookshireCresl1n: there is one around your desk
06:06.45drumkillawe have lying around like post it notes
06:06.50drumkillahave them*
06:06.51Cresl1nI think I have acouple of them on my desk
06:06.52Cresl1nhrm...
06:06.54Romik_drumkilla: in serious, this month, next month, or 2006?
06:06.55Cresl1nwhich one...
06:07.01brookshirethose probably don't work though
06:07.05Cresl1nhopefully this (or next) month
06:07.11drumkillaRomik_: this month!
06:07.14brookshirethis month
06:07.23brookshireor 2007
06:07.24brookshire:D
06:07.30Cresl1nor yesterday
06:07.31drumkilla:)
06:07.44drumkillaas did I
06:07.49Romik_drumkilla: and what about prices?
06:07.54Cresl1noooh....
06:07.56drumkillaRomik_: not sure about tat
06:08.01Cresl1nwe aren't in sales
06:08.01drumkillabrookshire: that on the web site yet?
06:08.02brookshirei know this!
06:08.14drumkillaI just write code.
06:08.18drumkillaI don't deal with dollars.
06:08.25Cresl1non drumkilla's back
06:08.31drumkillaheck yeah
06:08.38drumkillait's like a massage at the same time
06:08.43Cresl1n:-)
06:08.48Romik_drumkilla: i need 2 cards, if it much more expensive, i will not wait and will buy now
06:09.15brookshireit's like $825 more.. i think
06:09.16drumkillathey are very reasonably priced
06:09.19brookshiresomething around there
06:09.29drumkillabrookshire is just making things up
06:09.31drumkillaas usual
06:09.34Romik_cool
06:09.37Cresl1nyou guys should come to the office
06:09.42Cresl1nwe can have an all nighter
06:09.45brookshirewell.. that's what the (secret) upgrade price is is
06:09.45pjzanyone got a working PolyReboot script?
06:09.46*** join/#asterisk oej (~oej@213.204.186.40)
06:09.47Cresl1nwell, maybe tomorrow
06:09.54Romik_drumkilla: do digium will offer upgrade for old one'
06:09.55Romik_?
06:09.59drumkillaRomik_: yes
06:10.00*** join/#asterisk Zgarbi (~my@212.58.125.70)
06:10.03brookshireyeah.. but it's complicated
06:10.08brookshireand you have to send your card back
06:10.10drumkillayou have to send the card back for that
06:10.20drumkillain the future, they will be able to be upgraded remotely
06:10.24drumkillabut not for this first round
06:10.27Cresl1nI think we may initially be only doing it for 410s too
06:11.10drumkillaI *ACTUALLY* know that guy
06:11.10brookshire$2195.00
06:11.20drumkillahe's the man.
06:11.32Cresl1nawww
06:11.38Romik_this is digium? 0000:01:04.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01)
06:11.42Cresl1ndrumkilla r0x0rs!!!
06:11.47Cresl1nRomik_: probably
06:11.48Cresl1n:-)
06:11.51drumkillaheh, yeah
06:12.01drumkillaXilinx, sure, sounds good to me
06:12.14drumkillathough I've never seen it like that ...
06:12.18Romik_drumkilla: it 410s or 405s?
06:12.19brookshirei hate cresl1n
06:12.25brookshirebut i hate drumkilla more
06:12.33brookshire:P
06:12.44pjzwho is Tiger Jet Network Inc?
06:12.45drumkillaCresl1n: don't they usually come up as tigerjet?
06:12.53Cresl1ndrumkilla: not the 410/405
06:12.56pjzthat's how my TE110P comes up
06:12.59brookshireno.. te110
06:13.00Cresl1nthey come up as Xilinx something
06:13.00drumkillaok, yeah
06:13.06*** part/#asterisk lehel (~lehel@82.79.20.17)
06:13.09brookshireor tdm400p
06:13.12drumkillathe single spans and tdm cards say tigerjet
06:13.12Cresl1n100/110s use the tigerjet chipset
06:13.24Cresl1nlol, oh yeah, and the TDM400s
06:13.24drumkillaok, word
06:13.32blitzragedrumkilla / Cresl1n: sheeeesh - go to bed
06:13.38Cresl1nblitzrage: !!!!!
06:13.44blitzrageCresl1n: :)  wuz up?
06:13.51drumkillasleep bad
06:14.00pjzso what fun thing could I do if I had a spare TDM400 lying around?
06:14.15Cresl1n:-)
06:14.33Romik_drumkilla: are you sure that i will able to upgrade my card ?  and new 2 that i plan to buy next week from digium?
06:14.34pjzit's all FX.. damn I forget
06:14.43brookshirewrite a driver for macosx
06:14.45drumkillaRomik_: yes, but it will be more expensive
06:14.48Cresl1noooh...
06:14.51Cresl1ngood idea brookshire
06:14.54drumkillaRomik_: if you don't buy it all at once
06:14.56pjzit's all the ones that talk to the CO
06:14.58pjzoh hell no
06:15.04Cresl1nI'm writing the mtp2 stuff now
06:15.09pjzI refuse to write drivers for propritary OSen
06:15.15pjz:)
06:15.34pjzany of you guys ever use a nexpath box?
06:15.39drumkillaRomik_: ask sales about it, they'll work something out with you
06:15.40Romik_drumkilla: now imposible to buy it, but we need 2 cards for 2 our different offices...i can delay purchase for 2 weeks max...but i not sure that will helps
06:15.53pjzthey had a... workable gui
06:16.02drumkillaRomik_: like I said, ask sales, I dunno when they are shipping
06:17.32oejGood morning, drumilla, Cresl1n and blitzrage!
06:17.40Cresl1noej:!!!
06:17.44Cresl1ngood evening
06:17.45Cresl1n:-)
06:18.12SwKolle
06:18.18SwKjust the man i need to bug
06:18.34pjzis there a list of asterisk contractors somewhere?
06:18.49pjzI'm hoping to find someone local who can help me out at work
06:18.51Cresl1nthe wiki I think
06:18.53DaLionyes
06:18.55DaLion;)
06:18.57pjzah
06:18.58Cresl1nmaybe www.digium.com too
06:19.03DaLion#define local
06:19.19drumkillaaround his circular desk
06:19.24drumkillain a ... circular world
06:19.47Cresl1nmaybe the shape of the universe in mark's office is circular
06:19.57drumkillawoah.
06:20.02DaLioni tought the universe was flat
06:20.05DaLionlike earth
06:20.07DaLion;)
06:20.14drumkillaoej: is the patch for 1.0 attached to the sipura bug the one you want merged?
06:20.24oejdrumilla: Yes
06:20.30drumkillahow confident are you in it?
06:20.37oejdrumilla: But not right now in the middle of your night
06:20.45Cresl1noooh....
06:20.47Cresl1ngood idea
06:20.48pjz#define local "Austin, TX, USA"
06:20.58DaLionthanks ppjz
06:20.58oejdrumilla: I'll test it again while you sleep. I haven't looked at it for a while.
06:21.04drumkillahaha, ok
06:21.07Cresl1nugh, it's late
06:21.14DaLion#define "Denver, CO,USA"
06:21.16oejdrumilla: But it was confirmed by the reporter in the bug report
06:21.20drumkillaok
06:21.27drumkillaI might not mind merging it
06:21.29pjzDaLion: bug
06:21.31DaLionis this a a/s/l   irc shit ?
06:21.33DaLionlol
06:21.35drumkillaif we end up making a lot of changes to dtmf behavior ...
06:21.42drumkillaI think we're going to work on that tomorrow
06:21.47pjzDaLion: you forgot the 'local' part :)
06:21.49DaLionok nough playing night
06:21.49brookshirecrestl1n is 12
06:21.50*** part/#asterisk DaLion (~DaLion@69.156.64.3)
06:21.51brookshirehi!
06:21.54oejdrumilla: That's an interesting dilemma
06:22.03drumkillayeah ...
06:22.17drumkilladamin claims that inband dtmf is broken in 1.0, period
06:22.19brookshirethe best damn 12 year old coder ever!
06:22.24drumkillabut I just tried it and it works
06:22.36oejdrumilla: ...and a lot of people use it...
06:22.36brookshireoh wait.. that's file
06:22.38brookshirehehe
06:23.07drumkillaso I'm not sure what to think
06:23.11drumkillabut we'll see tomorrow ...
06:26.53Exstaticaanyone connected to fwd?
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06:28.18Cresl1nooh... gimme some ops :-)
06:28.35SwK<PROTECTED>
06:28.37brookshirethis channel should be antimatt ops
06:28.41Cresl1nheh
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06:29.02SwK<PROTECTED>
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06:39.36Cresl1nhey
06:56.09Romik_tiha v lesu...
06:56.29pjzso I'm using the stdexten macro
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06:56.48pjzhrm, let me rephrase
06:57.01pjzhow do I add a timeout to a Dial() command?
06:58.41Zeeekshow application dial
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07:02.13ta[i]nted!seen ptg123
07:02.17ta[i]nted!seen ptg1234
07:02.24ta[i]nted~seen ptg123
07:02.25jbotptg123 <~PTG123@ip68-106-24-139.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 21d 31m 54s ago, saying: '"replace your phone lines with cheaper voip with better features, oh shit wait, actually scratch that, you need to keep your phone line to have a phone line with us, so forget saving money, pay more :)"'.
07:02.58*** join/#asterisk ta[i]nted (~tainted@adsl-69-227-68-9.dsl.irvnca.pacbell.net)
07:03.03ta[i]nted~seen ptg1234
07:03.04jbotptg1234 <~PTG123@ip68-106-24-139.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 28d 39m 18s ago, saying: 'anyone here use ser with asterisk? :)'.
07:03.22ta[i]nted!seen ptg123
07:03.28ta[i]nted~ptg123
07:03.35ta[i]nted~seen ptg123
07:03.36jbotptg123 <~PTG123@ip68-106-24-139.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 21d 33m 5s ago, saying: '"replace your phone lines with cheaper voip with better features, oh shit wait, actually scratch that, you need to keep your phone line to have a phone line with us, so forget saving money, pay more :)"'.
07:12.23rickard25Extension '200' in context 'pri1' from '0' does not exist, call rejected
07:12.34rickard25What is wrong
07:13.00rickard25first guess is that i have to create a dial-plan for pri1
07:13.24Zeeeksomeone dialed 200
07:13.39Zeeekbut there is no indication of what 200 is in the dialplan
07:13.58rickard25but there is an extension configured as 200
07:14.08Zeeeknot in the right context apparently
07:14.24Zeeekwhere is pri1 coming from?
07:14.35Zeeekzapata.conf ?
07:15.26Romik_does new 411 card will also have AGC?
07:15.28*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
07:15.34rickard25Zeeek, yes
07:18.09rickard25Zeeek, how do I setup the extensions.conf to forward any call from PSTN to SIP extensions?
07:20.47*** join/#asterisk dvshadow (~dvshadow@ip68-96-87-185.oc.oc.cox.net)
07:21.07Zeeekthat's a vast question! I suggest you start by reading the downloadable PDF here: http://asteriskdocs.org
07:21.36ZeeekOr, for a great tutorial intro,
07:21.36Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
07:21.36Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
07:21.45rickard25thanks!
07:22.09Zeeeksee you anon :)
07:29.25Mavvie~fxo
07:29.26jbotforeign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo
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07:29.51Mavvie~fxs
07:29.52jbotextra, extra, read all about it, fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx
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07:39.23Mavviegreat.
07:40.03Mavvieto overcome the problem with the faxes, they have bought an OpenLine4 card.
07:40.15Mavvienot bothering asking me if it works with asterisk or not.
07:43.21ZeeekI hate when that happens
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07:52.34cjkanyone an idea when the BLINDTRANSFER variable will be availabe in iax as well?
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07:59.43festr_hello, i've problem with progress alerting coming from E1. I've asterisk 3 boxes. Box A: (E1 <-> IAX). Box B: (only IAX and SIP) and connected to A. Box C: (IAX and SIP) and connected to B. If i made call from SIP on Box B, progress indication is allright. But when i made call from SIP on BOX C(which is connected IAX to box B and B is connected IAX to A(where is E1)) i get tone progress, instead of progress coming from E1. Any idea?  :)
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08:18.50Thumannhmm.. i'm trying to get asterisk to work with a tdm(2xfxs modules) and one TE110P single span pci cards..
08:19.32Thumannbut when i do the ztcfg command, i get : line 118: Channel 1 already configured as 'Individual Clear channel' at line 114
08:24.31Thumannmy question is to the zaptel.conf file..
08:25.00Thumannfxsks and fxoks lines..  how do i set them up correct?
08:26.06RoyK~pastebin
08:26.07jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
08:26.13RoyKThumann: pastebin your zaptel.conf
08:26.42Thumann:) sec
08:28.47cjkanyone an idea when the BLINDTRANSFER variable will be availabe in iax as well?
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08:33.36ThumannRoyK: http://pastebin.ca/13879
08:33.41Thumannsorry for the delay
08:33.46Thumanndamn phone
08:38.22RoyKThumann: sorry. don't know
08:39.04Thumannnp :-/
08:46.31ThumannRoyK: hmm.. check this out, if you got the time: http://pastebin.ca/13880
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08:50.33djinThumann, goedemorgen (assuming you're Dutch, based on loadzone = nl)
08:50.37darkskiezIf i want to limit the number of calls to an extension, ie, with setgroup, how can I do that for two channels, for both legs of the call to increment the usage counts of both extensions.. I may be going around this the wrong way but I was trying to write a callwaiting disable thingie at the asterisk level.
08:51.09djinwhat is your hardware for fxoks en fxsks?
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08:55.20Thumanndjin: danish actually :)
08:55.27drrayI'm sad, I'm replacing my asterisk box with a 4 port TDM card, and my uptime was 156 days
08:55.28Thumanndjin: well.. i have 2 digium cards..
08:55.46Thumanndjin: 1 TDM card with 2 green fxs modules..
08:56.01Thumanndjin: it's a 4 port card.. but only 2 modules on it...
08:56.22Thumanndjin: and then i have a TE110P single span card also from digium
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08:58.26djinOk, lemme check something
08:58.32Thumannsure
08:59.14djinBut you're configuring 8 channels on a two channel TDM card?
08:59.51djinshouldn't it be: fxoks=32,33
08:59.56djinand that's it?
09:00.37Thumannhmm..
09:00.37Thumannsec
09:01.55Thumanndjin: http://pastebin.ca/13880#comments
09:02.08Thumanndidn't change much
09:02.45darkskiezare you meant to get any documentation, or even a single sheet of paper with your digium hardware?
09:02.56darkskiezI was quite surprised not to get even a packing slip.
09:03.17*** join/#asterisk potsboy (~chrisg@209.212.122.90)
09:03.24darkskiezfrom telappliant in the ulk
09:03.27darkskiezuk
09:03.37djinThumann, 35 channels configured?
09:04.08djinwhat if you start with only the TE110P at first.
09:04.50Thumannpull out the other cards?
09:04.51Thumann*
09:04.55Thumann*the other card
09:05.09djinNo, just remove the references inzaptel.conf
09:06.02djinspan=1,1,0,ccs,hdb3
09:06.03djinbchan=1-15,17-31
09:06.03djindchan=16
09:06.03djinloadzone = nl
09:06.03djindefaultzone = nl
09:06.33djinthen try ztcfg -vv
09:06.43djinif you get 31 channels configured
09:06.46djinadd fxoks=32,33
09:06.51potsboyif i could bend an ear, what the hell is going on here -> Extension '6111' in context 'from-pstn' from '' does not exist.  Rejecting call on channel 0/2, span 4
09:07.09*** join/#asterisk W|NGNUT (~wingnut-n@128.80-203-103.nextgentel.com)
09:07.26djina call to 6111 is received on context 'from-pstn'
09:07.40djinand there is no route for 6111 in the dialplan
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09:08.05potsboyaah, thanks
09:08.13potsboydamn friday brainded!
09:08.15djinand from '' means no caller-id from the caller
09:08.19djin;)
09:08.54djin-> gone for 30 minutes
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09:11.57Thumanndjin: hmm.. got 31 chans configured.. added the line you said. then 33 chans configured.. but still the ' did you forget the bla bla bla. .signaling'
09:12.54darkskiezis there any decent softphone for linux, when I last looked they seemed to suck :[
09:13.55*** join/#asterisk ragnar (~oomphclon@cable72a151.usuarios.retecal.es)
09:14.02ragnarhi
09:14.34_BB_darkskiez: xten lite for linux seems pretty good now
09:14.36ragnargot a little prob.. im doing sip->pstn, and im only getting voice from pstn to sip.. any hints?
09:15.01ragnar(this is with kphone)
09:15.12darkskiez_BB_, I didnt realise that existed even! I was hoping open source though.
09:15.36Ahrimanesdarkskiez: kphone works ok for me
09:17.04Zeeekhey Ahrimanes
09:17.09*** part/#asterisk lehel (~lehel@82.79.20.17)
09:17.23W|NGNUTCheers people. I try to get my head around HA-clustering, but I wonder about the db-file. Does anybody have experience with an active-active *-cluster?
09:17.34_BB_yep it exists...for open source you can grab kphone, gnophone for starters
09:18.30Ahrimaneshey Zeeek
09:18.46W|NGNUTragnar: do you nat?
09:19.09squirrelv5anyone made asterisk work with iaxtel.com?
09:19.21Zeeekiaxtel is not working well at the moment
09:20.09squirrelv5oic..when will it be ok?
09:20.17Zeeekummmm
09:20.24Zeeekit's a free service
09:20.29Zeeekno guarantee
09:20.46AhrimanesW|NGNUT: hm unless you make * store it's internal db in another format i doubt it you'll be able to preserve calls from a crashed server on a failovernode
09:21.31W|NGNUTAhrimanes: I'm not shooting for preserving call, but I hope to be able to mirror db content across the two hosts.
09:21.43W|NGNUTCan the db be stored in another format?
09:21.54W|NGNUTMaybe SQL even?
09:22.01ragnarW|NGNUT: no.. this is over a vpn
09:22.24AhrimanesW|NGNUT: not at the moment i think, but what are you looking to preserve, user info?
09:22.51W|NGNUTAhrimanes: call forwarding, registrations and such..
09:23.22AhrimanesW|NGNUT: ah.. hm well config info can be stored in sql, but not internals i think.. but feel free to patch :D
09:23.58W|NGNUTragnar: you should try to check if the RTP-packets gets through your VPN gateway..
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09:25.20ragnarW|NGNUT: alright
09:26.24squirrelv5is iptel.org allow u to call pstn?
09:28.38*** part/#asterisk Scousey (~chris_cah@82-70-181-205.dsl.in-addr.zen.co.uk)
09:30.52squirrelv5is iptel.org allow u to call pstn?
09:34.12infiI don't know.  perhaps if you ask about 12 more times, someone will answer you
09:34.37W|NGNUTAhrimanes: Well, the db-stuff is not really well suited for mapping to a relational system... is there a distributed b-tree database out there?
09:34.53Zeeekinfi that's what ignore lists are for :)
09:35.30AhrimanesW|NGNUT: yeah i could imagine it being way slow in sql
09:36.49W|NGNUTIf it gave enough payback in clusterablilty, maybe it could be worth it..
09:37.27W|NGNUTBut imagining a table structure that avoided dynamic table generation and still gave the tree-structure of the data . hmmmm
09:37.36W|NGNUTOh well - lunch!
09:37.48AhrimanesW|NGNUT: http://www.cs.yale.edu/homes/shah/html/pubs/skip-graphs.html or http://techreports.library.cornell.edu:8081/Dienst/UI/1.0/Display/cul.cis/TR2004-1926 maybe?
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09:39.00ramthapeace
09:39.19MooingLemur/dev/urandom too
09:39.20ramthawhat reason could cause 99% cpu in asterisk?
09:39.23AhrimanesMooingLemur: i hope you're not listening to it yourself
09:39.30Ahrimanesramtha: what os?
09:39.42MooingLemurit's mostly saying letters and symbol names
09:40.02MooingLemursometimes goes too low pitched
09:40.10Ahrimanesah
09:41.44ramthaAhrimanes: debian
09:41.48Ahrimanesramtha: hm ok
09:42.07ramthaafter starting and stoping asterisk, it works normal
09:42.16ramthadon´t know how long :)
09:42.37Ahrimaneshehe would be good to know things like number of users etc
09:42.42Ahrimanesand asterisk cvs or stable?
09:45.48_omerhi
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09:46.05_omeroopsss is that happened due to my "hi"
09:46.07_omer;)
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09:46.29*** mode/#asterisk [+o twisted] by irc.freenode.net
09:54.36_omerwhat does it mean ?
09:54.37_omerJun 10 02:14:12 NOTICE[3604]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end
09:55.32*** join/#asterisk Pazzo (~Pazzo@host130-250.pool8172.interbusiness.it)
09:57.02Ahrimanesthen he leaves.. clever, hehe
09:59.09Thumann:)
09:59.32Thumannhah
09:59.52*** join/#asterisk tzanger (~tzanger@mixdown.ca)
09:59.58Thumannout salesdude just bought 10 cisco 7960 ip-phones.. and two ZyXel POE-80 hubs..
10:00.05Thumannguess what! :D they dont work togehter..
10:00.08Thumann*together
10:00.16Ahrimanesnice
10:00.17Zeeeksend me the phones
10:00.32ThumannZeeek: agreed
10:00.34Thumannhehe
10:00.48Ahrimaneshm we're getting 10 wooksung 2100w soon
10:01.12Thumanneverything has to be 'the cisco way' so now i've send the dude 120km south to pick me up a cisco poe 3500 :D
10:01.22Ahrimaneslol
10:01.32Ahrimanesplease to learn
10:01.41Thumann;> i'll go to lunch while he brings me the stuff..
10:02.21Ahrimanesooh lunch
10:02.27Ahrimaneswonder what's on the menu today
10:02.31Ahrimanes(hope it's meat)
10:02.55Thumannwe're getting hotdogs.. :>
10:05.13Ahrimanesmm hot dogs
10:06.02Ahrimaneshm maybe we have biuld-your-own-burger again.. that's always nice
10:06.30Ahrimanesalthough others tend too stare a bit when you put in the 4th piece of meat..
10:08.28*** join/#asterisk jerry_hotlinks (~dunno@office.hotlinks.co.uk)
10:08.32ramthaAhrimanes: 8 mutiple calls and asterisk cvs head of 2 weeks
10:08.46ramthaand since several releases i see asterisk defunct
10:08.52ramthabut works stioll fine
10:09.09ramtha250 users in mysql db
10:09.21Ahrimanesramtha: ok, well am not running cvs anymore, so cant help
10:10.19ramthais there any reason for restarting asterisk in some times?
10:10.38ramthawhat errors?
10:10.59jerry_hotlinkspaste here?
10:11.19ramthapastebin..
10:11.20Ahrimanesjerry_hotlinks: noo, pastebin.ca or something like it
10:12.26jerry_hotlinkshttp://pastebin.ca/13883
10:13.01jerry_hotlinksthe files do exist - as they are the sample ones
10:13.09ramthaseems that the file does not exist
10:13.22ramthahm
10:13.27ramtharight format?
10:13.41jerry_hotlinksthey are the original ones that came with the instal
10:14.03*** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net)
10:14.14ramthahmm why should asterisk write that he can not find it..
10:14.18jerry_hotlinksi have this in sip.conf
10:14.19ramthais it played back?
10:14.23jerry_hotlinksnope
10:14.31ramthaah, the its not there
10:14.36ramtharigt permissions?
10:14.39ramtha+h
10:14.45jerry_hotlinksgood question - one sec
10:15.19jerry_hotlinksyup perms are set +r  on all
10:15.33ramthafor user root ore *
10:15.38ramthaor
10:15.42jerry_hotlinksyes for root
10:15.49ramthado it for user *
10:16.04ramthaor do you have * startet as root?
10:16.20jerry_hotlinksyes
10:16.21jerry_hotlinks-rw-r--r--    1 root     root         4323 Jun  9 15:21 tt-weasels.gsm
10:16.58ramthahmm
10:17.02djinThumann, did you get your cards working?
10:17.06ramthawhere are youre sond files located?
10:17.23jerry_hotlinks./var/lib/asterisk/sounds
10:17.27ramthaok
10:17.57jerry_hotlinksi have a live working system and a test system - i have been comparing perms etc
10:18.25djinjerry_hotlinks, dump the .gsm in your extensions.conf
10:18.43djinjust refer to tt-weasels
10:18.57jerry_hotlinkssorry - can you explain what you mean by that?
10:19.10ramthayou have "weasels.gsm" in extensions.con
10:19.21djininstead of "tt-weasels"
10:19.29ramthaput in ""tt-weasels"
10:19.37ramthachange "tt-weasels.gsm" to "tt-weasels"
10:19.43jerry_hotlinksok
10:19.45ramthalike djin says :)
10:20.03djinyou used more words ;)
10:20.07ramtha:)
10:20.30jerry_hotlinksok i know have this
10:20.32jerry_hotlinksexten => 103,1,Background,tt-weasels
10:20.43ramthaok check
10:20.49ramthastill not working?
10:20.55jerry_hotlinksreloading
10:20.58jerry_hotlinksone sec
10:21.02ramtha1
10:21.03ramtha2
10:21.04ramtha3
10:21.05ramtha:)
10:21.10jerry_hotlinkssorted :)
10:21.15jerry_hotlinksthank you so much :)
10:21.18djincool ;)
10:21.36jerry_hotlinksmust remember not to specify extension
10:21.53djinmental note . . .
10:22.06jerry_hotlinkswhen i came here i was half expecting a "rtfm" :)
10:22.14ramthahehehe
10:22.30djinwe're not like that.
10:22.36jerry_hotlinksgood to hear :)
10:23.12jerry_hotlinksnow i got past that hurdle i will try and figure out how to do a callback depending on number entered :)
10:23.25djina mildly UTFS might be used sometimes, because there is no real manual ;)
10:23.50jerry_hotlinksi got  alot of info from the handbook v2
10:24.12jerry_hotlinksand there as well :)
10:24.22djinand mailinglist.
10:24.48jerry_hotlinksah prob with mailing list is its too slow when the boss is on my back to get it done fast
10:25.42tzangerjerry_hotlinks: then learn more :-)
10:26.01jerry_hotlinkswhich is why i came here in the first place :)
10:26.10tzangeror explain to your boss that you are soliciting the help of over 10 thousand people FOR THE COST OF YOUR SALARY.  You can get faster help but it'll cost more.
10:26.19djinjerry_hotlinks, use search on mailinglist. Many questions are already answered in the past.
10:26.27jerry_hotlinksah right
10:26.40jerry_hotlinkssorry quick question - where?
10:26.44tzangerjerry_hotlinks: google
10:26.52djinasterisk.linkx.net
10:26.52tzangersearch terms here site:lists.digium.com
10:26.56jerry_hotlinksthank you
10:27.08djinmy little contribution to the project ;)
10:27.13tzangernice
10:27.27ThumannAhrimanes: so what was it? ;>
10:27.43AhrimanesThumann: smoked meat and some raspberry pie :D
10:28.09ThumannAhrimanes: hehe.. nice
10:28.17AhrimanesThumann: very
10:28.23Ahrimanesbut not much meat left
10:28.30Thumannyuck.. i finished off with this .. white 'champagne peach'
10:28.31Ahrimaneshope food at astricon is good
10:28.39Thumann:-/ most hated fruit now
10:28.54Ahrimaneshehe
10:29.15tzangerwhat is your most hated fruit?
10:29.19Ahrimanesfruit and veggies = what food consumes
10:30.47lterslatest cvs head shows CVS-HEAD without date. Is this the desired way?
10:30.47tzangernonsense
10:31.12tzangerlters: there should be a new command that shows all the source cvs dates
10:31.16tzangernot sure what it is now
10:31.26tzangercheck asterisk -h and also show <tab> in the CLI
10:31.42lters<PROTECTED>
10:32.00ltersbut I miss the quickly visible version on connect to *
10:32.11tzangerlters: it was really not enough
10:32.15tzangerlters: learn to not miss it
10:32.23ltersn/p I will
10:32.26tzanger:-)
10:32.41ltershave you played any with sccp
10:32.48tzangernope
10:33.07Ahrimaneslters: compiled it yourself?
10:33.35ltersyeah, I have. The cisco phones act very different.
10:33.57Ahrimanesok, in the source dir, change the text in .version file.. and rebuild
10:34.30lterslooks like they would want that to still show the last cvs date...
10:34.47tzangerlters: that means you still have to use 'make update" which sucks balls
10:34.51Ahrimaneshm ok
10:35.00ltersmake update ?
10:35.05lterswhy, that is nifty
10:35.22tzangerno it's not
10:35.31tzangeryou use cvs up for every other project out there
10:35.42tzangerand if you forget to do it and you run into problems now your date that you report ot everyone is fucked up
10:35.47tzangerwhich causes pain for not just you
10:35.55ltersdate?
10:36.11lterswhich date
10:36.19tzanger"Hey everyone, I'm running CVS HEAD from about 2 weeks ago and every time I make an SCCP call Cisco calls me up and bitches"
10:36.27Ahrimaneshehe
10:36.33lterslike of the files themselves?
10:36.41lterstzanger, do what ?
10:36.54tzangerlters: well see that's what make update did, it touched the .version file so that the show version date had something to go on
10:37.05ltersoh
10:37.13ltersinteresting.
10:37.19tzangerand $Id$ is great but every file needs it since it's entirely possible (and I do this all the time) to update only bits and pieces of CVS HEAD
10:37.30lterstrue
10:37.31tzangerso the .version file is not in sync with the whole tree, just the last 'make update'
10:37.45ltersare you using cisco phone(s) with sccp or what did you mean?
10:37.51lters:)
10:38.43ltersThat article of Mark's was great. (cnet)
10:39.30Ahrimaneslink?
10:40.08lterson the irc head/title
10:40.32ltersor doesn't your chat show it?
10:40.43Ahrimanesah
10:40.51Ahrimanesloong topic, hehe
10:41.20lterstzanger, are you running latest head?
10:41.29tzangerlters: about 2 weeks old, haen't had time to update
10:41.35tzangerand a lot has changed from the cvs logs
10:42.13ltersyeah, quite a bit, and some serious changes.
10:42.27tzangeryup but that's what drives this whole ship forward
10:42.33ltersI am running 6 02 in prod and have some probs.
10:42.49tzangerif you don't like it either become a dev or run the stable branch :-)
10:42.54tzangerwhat probs
10:43.14ltersabout mid day the sip phones stop working.
10:43.25tzanger??
10:43.30tzangerthis is reproducible?
10:43.32lters:( I saw the bug fix for the channel walk and think that might be my prob.
10:43.51ltersbecause fop/manager seem to hang as well.
10:44.18ltersalso, there were some interesting zap updates.
10:44.29tzangeryup
10:45.44lterssip phones start to do this:
10:45.46ltersWARNING[6658] chan_sip.c: Maximum retries exceeded on call 50decef434f81dd97d1ff83001e215b4@10.1.45.35 for seqno 102 (Non-critical Request)
10:46.00ltersweird
10:46.03tzangerso have you tried head?
10:46.15ltersthat is head from 06/020/05
10:46.26tzangerright but have you tried the latest is what I had meant
10:47.03ltersno, not yet. I was hoping there would be a cvs commit stall showing that all is *stable*
10:47.10lters:)
10:47.17tzangerhahaha
10:47.20tzangergood luck on that :-)
10:47.28tzangerbut I'll give you a hint as to what I do
10:47.59tzangertar -czvf /tmp/asterisk-works-20050610.tgz /etc/zapata.conf /etc/asterisk /usr/sbin/ast* /usr/lib/asterisk
10:47.59lterskram, I am all ears
10:48.08tzangerthat gives me a snapshot of a working system
10:48.31tzangerthen I move my source directories out of hte way in case I need to go back to the exact source that I had working (I don't trust cvs -D)
10:48.42tzangercvs co or up, make, make install, give 'er a whirl
10:48.53lterskram, just makes a tar ball. fancy
10:49.05tzangerif it works, great.  If not I back up to the last known state easily with tar -xzvf /tmp/asterisk-works-20050610.tgz -C /
10:49.12tzangerI'm tzanger, not kram :-)
10:49.25trasherrmy asterisk box is connected to S0 via CAPI - is it possible to do more than one calls via the S0 at the same time?
10:49.27lterssorry, I hit k for ok, an must have hit tab
10:49.47tzangeris ISDN BRI not two B chan, trasherr?
10:50.05trasherrif i try a second call, when another one is active, my software-SIP-phone shows "503, service unavailable"
10:50.08lterstzanger, thanks for your tips. gotta go
10:51.18trasherrtzanger: i dont know exactly what you're talking about.. devices is set to 2 in capi.conf.. so it have to work, haven't it?!
10:51.25*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
10:51.35tzangertrasherr: I've never used BRI, I was just stabbing in the dark
10:51.36rickard25How do I execute commands in asterisk, Example I have one person on the line, and want to play a mp3 or run festival
10:52.16trasherrhmm :/
10:52.52trasherrso i guess you can't help me with configuring asterisk to show the ISDN-MSN on outgoing calls..
10:53.01tzangerno sorry
10:53.38trasherroks
10:55.32rickard25how do i exec a asterisk app while having a person on the line?
10:56.02trasherrhm i want to know that too ;)
10:56.24ragnarstart the console? :)
10:56.35rickard25yes and in the console?
10:57.04ragnarwell.. my very wild guess is that if help doesnt show something useful, you cant
10:57.42rickard25k, so its creating extensions and macros and then conference with the macro?
10:57.47ragnarmaybe you could change the extensions in realtime, and reload them
11:01.07*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
11:09.53*** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net)
11:10.40robl^morning
11:11.11*** join/#asterisk Marlow (~martin@cerberus.bluetree.ie)
11:16.26trasherrhm with ISDN cards with cologne chipset it's possible to use it with asterisk and plug a isdn telephone into it, right?
11:24.51Marlowtrasherr: jup
11:25.52trasherrthx
11:26.54trasherrMarlow: another question.. assume that my ISDN phone number is "123456" - now is it possible to create own extensions like "123456-10"?
11:28.21Marlowtrasherr: that depends if your telco allows you to add numbers behind
11:28.27Marlowtrasherr: but if, then yes
11:28.55trasherrhm, what do you mean with telco?
11:30.13cjkanyone an idea when the BLINDTRANSFER variable will be availabe in iax as well or is there something similar
11:33.13*** join/#asterisk zoa (zoa@82.103.76.147)
11:37.45djinAnyone going to Astricon Madrid as well.?
11:37.52newltrasherr: Sure, buy an indial range from your carrier.  Some sell blocks as small as 10.
11:38.47cochihehe bad chances with arcor i guess, trasherr
11:38.59cochiat least it won't be cheap for a private person
11:39.04cochiif it's for a company, then no prob
11:39.35cochiwas a common option back then with national 1TR6 anyway *sigh* had some advantages over EDSS1/ETSI :|
11:40.51Ahrimanesdjin: i am
11:42.02djinAhrimanes, cool.
11:42.13Ahrimanesdjin: when are you arriving in madrid?
11:42.19djinTuesday
11:42.37Ahrimanesdjin: ok same here.. am trying to gather some people for socail activities tuesday evening
11:42.50djinOh
11:42.58Ahrimanessocial
11:43.04Ahrimanesgod my spelling is crap today
11:43.47Ahrimaneshehe
11:44.12djinare you from Europe?
11:44.20Ahrimanesdjin: yes, denmark
11:44.30djinah, ok.
11:44.47Ahrimanesah nice
11:44.58djinare you up for dCap as well?
11:45.20Ahrimanesdjin: no, boss didnt want to spend the money now, so maybe in nov in the us
11:46.00djinoh, ok. ($275 wasn't that much, so I'll give it a try)
11:46.27Ahrimanestried to tell the boss that $275 wasnt much too.. but even going to astricon was much in his terms...
11:47.00*** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl)
11:47.05onkeltimmhelay
11:47.15Ahrimanesdjin: hehe
11:47.20newluntil tax time 8)
11:47.26djinHi, fellow Dutch man
11:47.33djinnewl, true
11:47.41Ahrimanesdjin: which part of holland are you from ?
11:47.42djinPaying taxes is nice.
11:47.48djinMeans you made money ;)
11:47.57djinI'm from Haarlem, near Amsterdam
11:48.12djinWell NL is small, so everything is near Amsterdam ;)
11:48.24Ahrimanesdjin: ok.. strange custom that when you order a beer you get the small one pr default.. dont like that, heh
11:48.51Zeeekbeer?
11:48.59djinwakey wakey .
11:49.05ZeeekBeer!
11:49.07AhrimanesZeeek: you have hightlight on beer? hehe
11:49.16Zeeekno just an eye for detail
11:49.29*** join/#asterisk stkn_ (nobody@stkn.developer.gentoo)
11:49.33Ahrimanesah
11:49.40djinAhrimanes was referring to small ones, so go back to sleep, Zeeek.
11:50.06Ahrimanesdjin: Zeeek is currently the other person up for beer on tuesday night, hehe
11:50.12Zeeekbleh small beers
11:50.28djinbetter
11:50.38Zeeekmy father used to say "colder in the mass"
11:50.42Zeeekso bigger
11:50.42Ahrimanesbut there's hardly a mouthful in the small ones down there
11:50.50djinAhrimanes, you said 'social', not 'beernight'.
11:50.55*** part/#asterisk jeffik (~Jeff@69.158.17.52)
11:50.59Ahrimanesdjin: social = beer
11:51.08Ahrimanesmaybe even food
11:51.10Ahrimanestoo
11:51.13Ahrimanesdjin: lol
11:51.17djinwow, tapas .
11:51.54Ahrimaneswould be sweet
11:51.56Ahrimanestapas with beer
11:51.58djinZeeek will be in Madrid as well?
11:52.05Zeeekthat is correct
11:52.07Ahrimanes<homermode>mmmm beer</homermode>
11:52.20Zeeekafter much tribulation, I decided to brave the company
11:52.42AhrimanesZeeek: you found out that at least one other person would be there an jumped at the chance
11:52.53djinnot sure what you means, but sounds good.
11:53.04djinyou mean/it means
11:53.17Zeeekdjin I mean I have already been to asterisk lunches for example
11:53.35ZeeekI know the kind of geek that goes to these gatherings :)
11:53.55djin:)
11:54.12djinAre there Playstations van bootbabes?
11:54.17djinvan = and
11:54.20djindamn!
11:54.49trasherrso.. i can call ISDN numbers via SIP from my workstation through my asterisk PBX.. but on an active call, i hear an echo (i can hear myself on the speaker).. how do i avoid that?
11:54.58ZeeekI'm always interested in counting the number of women at these meeting, usually < 0.1%
11:55.29djindon't talk?
11:55.36djini'm sorry
11:55.41djinnot funny
11:55.59Zeeektrasherr the speaker of what, your phone? What fone?
11:56.09djintrasherr, could be because on multiple things.
11:56.20djincodec, network, etc.
11:56.36Zeeeksound card, person on other end with speaker
11:56.45Zeeekshitty hardphone
11:57.04Zeeekpoorly adjusted drivers/hardware...
11:57.11djinto name a few.
11:57.18Zeeekechocancel configuration in .conf files
11:57.38djinto name a few more
11:57.54Zeeekbut the "in my speaker" intrigues me
11:58.02Zeeekturn the speaker off!
11:58.04*** join/#asterisk telenieko (~telenieko@62-15-139-146.inversas.jazztel.es)
11:58.15Zeeekah, but then we'd have one-way audio
11:58.33djinI talk, you listen.
11:58.36teleniekohi. I just read a post on the mailling list about an IM Application, is there anything coded about that? What I want is to get an user status from jabber to act accordingly
11:58.37trasherrwith SJphone @ my pda .. but i guess it was the speaker.. now i plugged in a headset, seems to work better now.. but the person i called hears an echo of me too
11:58.54Zeeekwhat are they using?
11:59.12Zeeeksoftphones are hardly ideal btw
11:59.21trasherrthe person i called is using an normal ISDN phone
11:59.22djintelenieko, isn't SER doing that?
11:59.25Zeeek2x for linux softphones
11:59.46teleniekodjin I use asterisk only, not sER ;(
11:59.53djintrasherr, perhaps he hears your echo ?
12:00.03Zeeektrasherr check the mailing list and wiki for all the echo cancel stuff. Sometimes it's as easy as reducing the volume of audio going out
12:00.05JerJerwhy not SER?
12:00.15djintelenieko, ok.
12:00.29trasherroks thx
12:00.47teleniekoany way to get jabber user status from asterisk or a unix shell ? :)
12:02.38djinDoes anyone have dCap-exam experience?
12:02.46*** join/#asterisk nitram (foo@superblob.com)
12:03.19AhrimanesZeeek: what type of beer do you favor?
12:03.38*** join/#asterisk Sanguis (Sanguis@cpc4-kemp1-5-0-cust222.lutn.cable.ntl.com)
12:03.53ZeeekMany, many, many kinds depending on country I'm in
12:03.59ZeeekI also like wine
12:04.14ZeeekI kind of work in the wine industry in a way
12:04.20Zeeekbut that's far OT
12:04.24Zeeekwhereas beer is not
12:05.47W|NGNUTAhrimanes: Thanks for the links. I think the skip-graphs looks most promising from a 2-minute browsing. Are there more people that wish to have the possibility of spreading the databases?
12:06.05*** join/#asterisk BerndR (~konversat@mich2-145-8.utaonline.at)
12:06.07djinAstricon -> Madrid -> Spain -> wine
12:06.09AhrimanesW|NGNUT: i'd love it
12:06.19djinThat's not that OT
12:06.40AhrimanesZeeek: ok, hoegaarden and newcastle are pretty much the best i like
12:06.48BerndRhello all
12:06.51W|NGNUTToo bad I'm not a hacker... I'm going to ask in linux-ha if there is a way to cluster db-files.. they should know such things.
12:07.00AhrimanesW|NGNUT: hehe
12:07.21AhrimanesW|NGNUT: hm my boss likes giving money to opensource development.. i might ask him to fund it
12:07.35BerndRdoes anyone know how to pass multiple options by agi to asterisk?
12:07.52BerndRi tryed ast.appexec('system','asterisk|-rx|extensions|reload')
12:08.07W|NGNUTAhrimanes: He would have to fund some C training too in my case, but hey....
12:08.11BerndRor ast.appexec('system','asterisk -rx extensions reload')
12:08.43AhrimanesW|NGNUT: will i'd probably find someone more experienced for the task :)
12:09.03BerndRby using spaces the cli says "(system) Options: (asterisk)"
12:10.47AhrimanesW|NGNUT: and i was trying so hard.. sigh
12:11.42W|NGNUTAhrimanes: I could make it easier for you and show you the C agi-program I'm messing around in... :P
12:12.06AhrimanesW|NGNUT: hehe, why bother to make agi in c?
12:13.04W|NGNUTI like pain..
12:13.26Ahrimanesah
12:13.30Ahrimanesmasochist you are
12:13.55Ahrimanesoh, and like yoda i speak
12:13.56Ahrimaneshm
12:15.06*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
12:16.29BerndRi also passed ast.appexec('system','"asterisk -rx extensions reload"') with double quotes because asterisk internal uses argv[1]
12:16.50BerndRbut nothing helps :(
12:16.56*** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com)
12:17.27robl^somoene has watched one too many sci fi flicks :)
12:17.31Ahrimaneshehe
12:19.02djinnot sure posting the link doesn't at least break 2-3 IRC rules ;)
12:19.25ThumannxD
12:19.33robl^djin, rules?!!? in here?!?!? LOL!!!!
12:19.33*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
12:19.39Ahrimanesdjin: goatse?
12:19.48Thumannwho said my name?!
12:19.49djinno, not that bad.
12:19.51Ahrimaneshehe
12:19.58AhrimanesThumann: ew.w.
12:20.01Thumann;X
12:20.02djinOk, look at your own risk
12:20.05Thumannhehe
12:20.06*** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net)
12:20.10djinStar Trek will never be the same
12:20.12djinhttp://www.btinternet.com/~skinhead1/klingons.htm
12:20.22cluecon~cluecon
12:20.26jbot[cluecon] 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.
12:20.29ThumannAAAH!
12:20.37djinwarned you!
12:20.44Thumanndjin: erasing my old vhs tapes atm!!!
12:20.52djin:)
12:21.05Thumanndamn.. almost goatse
12:21.09djincame across this link on a weblog a hour ago.
12:21.18JerJercluecon:  isn't cluecon in the fucking topic enough?
12:21.19Ahrimanesmust .. get .. ops .. and .. ban .. djin
12:21.20Ahrimaneshehe
12:21.21djinthe moment to post it was there ;)
12:21.29*** join/#asterisk jeffik (~Jeff@69.158.17.52)
12:21.44tzangerthat link is *so* not right
12:22.15Ahrimanes..who ruined star trek
12:22.25tzangerno that wasn't you
12:22.38tzangerthat was shaner
12:22.40tzangerer shatner
12:22.42Ahrimanesthat was an addition to what djin said
12:22.42Ahrimaneshehe
12:22.43Ahrimanesok
12:23.25cluecondjin: we know who you are and you will soon be met by members of the federation council to discuss your distribution of classified material.
12:23.54*** join/#asterisk trikk (~trikk@webnovative.demon.nl)
12:24.12*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
12:24.31djinok, I'm sorry
12:24.43Ahrimaneshehe
12:24.53Ahrimanesyou buy the first round of beer tuesday then
12:25.12*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
12:25.13djinOk, it's on me ;)
12:25.27*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
12:25.33djinHe, where's Zeeek?
12:25.42djinbeer, beer, beer . . . . beeeeeeer.
12:25.48Ahrimaneshehe
12:25.54Ahrimanes3
12:25.55Ahrimanes2
12:25.56Ahrimanes1
12:25.58Ahrimanesdamn..
12:25.59clueconI guess Zeeek doesn't get any beer.
12:26.07ZeeekI was away
12:26.19clueconsorry, too late, no beer for you tardy boy.
12:26.30ZeeekI got an URGENT email from the agency to pay my hotel in advance
12:26.33Ahrimanesclose but no beer
12:26.42AhrimanesZeeek: lol
12:26.47ZeeekIt wxasn't clear to me that we were supposed to pay it
12:27.32Marlowtrasherr: telco is you telephone company ..
12:27.34newlDid you ask them to cook your free continental breakfast in advance if you do?
12:27.48Zeeekprolly won't be cooked
12:27.54Zeeekls -l
12:27.59Zeeekno, no, no
12:28.10Marlowtrasherr: in Denmark or Ireland for example you have to buy each and every DID from your Telco ..
12:28.27AhrimanesZeeek: better that than a login and password :D
12:28.38AhrimanesMarlow: huh? i'm in denmark
12:28.59Marlowtrasherr: in Germany you can buy a P-P configured line and a prefix  ... hang all the numbers you want after that prefix ..
12:29.06*** join/#asterisk RoyK (~roy@42.80-203-178.nextgentel.com)
12:29.22Zeeekroot
12:29.24Zeeeksystem
12:29.28ZeeekOH SHIT!!!!
12:29.32clueconroot
12:29.33cluecongod
12:29.35Marlowtrasherr: not sure, if that also works on P-MP ..
12:29.36Zeeek^H^H^H
12:29.42Ahrimaneslol
12:29.46clueconyeah, we all know that somebody is stupid enough to use it.
12:30.15Zeeekapparently a lot of people use root and root
12:30.28Zeeekeasy to remember
12:30.38MarlowAhrimanes: han havde nogle spørgsmål vedr. numre, om man bare kan lave lokalnumre og DIDs når man nu har en asterisk boks .
12:30.49MarlowAhrimanes: og det er jo nu lidt forskelligt fra land til land
12:30.51ThumannDanish powah
12:30.57AhrimanesMarlow: ah ok
12:30.58MarlowThumann: eh :)
12:31.08ThumannMarlow: hey igen! :)
12:31.17MarlowThumann: davser ..
12:31.18AhrimanesMarlow: argh, hvad sker der for din revdns..ikke i dk?
12:31.23cochi:|
12:31.24Ahrimaneshey Thumann
12:31.27MarlowAhrimanes: jeg ER ikke i DK ..
12:31.30ThumannAhrimanes: :D min er i dk
12:31.31Thumannhehe
12:31.35cochi<- wants to learn danish instantly
12:31.35AhrimanesMarlow: hvor så, hehe
12:31.41AhrimanesThumann: hvor arbejder du?
12:31.41MarlowAhrimanes: IE = Ireland
12:31.45AhrimanesMarlow: ah
12:31.49Thumanncochi: hehe
12:31.57Ahrimanescochi: but why
12:32.00Ahrimanesit blows
12:32.04cochii like languages
12:32.08Ahrimanescochi: ah ok
12:32.09cochigenerally
12:32.12JerJer<-- wants to learn English
12:32.18Ahrimanescochi: try russian
12:32.19Zeeekhow is Spanish spelled in Spanish? Español ?
12:32.21cochii'm just not up on learning vocabulary
12:32.28AhrimanesThumann: uhm.. no thanks?
12:32.30cochirussian!? sry, that'd exceed my lifespan ;)
12:32.35Zeeekyes, that looks  right
12:32.40djinEspanol
12:32.42cochijust as chinese did. so i cancelled after a year ;)
12:32.42Ahrimanescochi: i learned quite a bit in 2 months
12:32.48MarlowJerJer: that's not easy .. there is really no place in the world, where they speak proper english
12:32.49ThumannAhrimanes: dansk er vejen frem.. vi skal bare ha' konverteret resten af verden...
12:32.53AhrimanesThumann: haha
12:32.59AhrimanesThumann: arbejder du med voip i dk?
12:33.12cochiat least i can guess the danish stuff ;)
12:33.23djinwido you work with voip in Denmark?
12:33.30djinwido = do
12:33.35djinam I right?
12:33.35cochiyep :)
12:33.36Ahrimaneshehe
12:33.52Marlowdjin: maybe you should specify, who you are asking .
12:34.00Marlowdjin: there are at least 3 danes in here ..
12:34.05ThumannAhrimanes: både og.. jeg har fået til opgave at lave et par kasser der skal sendes til udlandet.. finland nærmere bestemt.. og jeg har det tekniske ansvar for en box her i landet.. ellers er jeg systemkonsulent i aalborg i windåse verdenen
12:34.16djinok then
12:34.20AhrimanesThumann: windåse.. argh
12:34.26cochi(k, can't guess that much +g+)
12:34.27ThumannAhrimanes: Mcp all the way ;>
12:34.42ThumannAhrimanes: naah.. jeg er den eneste der er linux horny i det her firma.. :D
12:34.56AhrimanesThumann: hehe ad linux.. FreeBSD er vejen frem... :P
12:35.05*** join/#asterisk savag3 (~rfairhall@link4.amg.net.nz)
12:35.06MarlowAhrimanes: eh .. ikke når vi taler Asterisk :)
12:35.16AhrimanesMarlow: hvorfor ikke?
12:35.20ThumannAhrimanes: 3 dead throlls in a baggie - Every OS sucks
12:35.20cochipfff. irgendwelche deutschen hier? lasst man nen gegenpol zu der daenerei machen ;)
12:35.21djinlinux horny?
12:35.27MarlowAhrimanes: fordi FreeBSD porten er lang bagud .
12:35.31AhrimanesThumann: hehe not mine
12:35.33AhrimanesMarlow: 1.0.7 ?
12:35.46Thumanndjin: ~ the only one in the company who likes linux.. hehe
12:35.46MarlowAhrimanes: zaptel support er i BETA og Alpha stadie .
12:35.56AhrimanesMarlow: tjaeh, men virker nu ok
12:35.56Thumanndjin: or.. translated.. who isn't afraid of it..
12:35.58MarlowAhrimanes: hvis du kun vil lave VoIP til VoIP, så virker det .
12:36.00clueconJerJer: which dialect?  Ya got Southern, Northeastern, Central, Western, SoCal, NoCal, LA, San Fran, Jersey, East River, Manhattan...and that's just the beginning.
12:36.04djinThumann, ah ok ;)
12:36.16Thumannmost people here run when they see a CLI
12:36.17savag3hello
12:36.17MarlowAhrimanes: sure .. men ikke så stabil som i Linux .. og jeg ville ikke basere min switche på det ..
12:36.27ThumannWHAT?! where du i set the check mark?! ;)
12:36.44savag3ive got a question regarding h323
12:36.47djinThumann, just reboot ;)
12:36.57MarlowAhrimanes: der er diverse ting, som ikke er blevet porteret og ikke alle kort er supporteret ..
12:37.02AhrimanesMarlow: hm ok, anyways... pstn lader vi andre om :D
12:37.06savag3when asterisk sends a progress pdu the display ie is set to root
12:37.11savag3how do i stop that happening?
12:37.26MarlowAhrimanes: det kan være, at det gælder for dig .. jeg som ITSP skal have linier ind i boksen ...
12:37.27AhrimanesMarlow: og ports gør livet meget nemmere mht h323 og lign
12:37.33MarlowAhrimanes: og det skal virke hver gang .
12:37.37Thumann:D lol.. 11 calls allready regarding the new win-hotfix who doesn't like intel onboard gfx cards.. (mostly on dell comps.. )
12:37.46MarlowAhrimanes: og H323 er bare evil, så det holder man sig fra ..
12:38.02AhrimanesMarlow: i know.. men vi leget med video, så det har vi et par hardphones der kræver
12:38.06MarlowAhrimanes: desuden .. hvis du vil have ports lign på Linux, så er der jo Gentoo og Debian
12:38.22ThumannThumbs up for debian
12:38.35cochigentoo sucks. just problems. sry :|
12:38.39AhrimanesMarlow: ja, men jeg kan ikke lide linux og deres måde at release på.. og ideen med base seperat fra kernel, så holder mig til BSD så vidt som muligt
12:38.41MarlowThumann: jepper ..men det kender du jo også min holdning til :)
12:38.44*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
12:39.20MarlowAhrimanes: base seperat fra kernel er fordi du kan vælge base eller bygge den selv ..
12:39.31ThumannMarlow: indeed! :)
12:39.35MarlowAhrimanes: det ligger i Linux historie, ikke fordi man har besluttet det
12:39.37AhrimanesMarlow: linux = kernel, distro = base
12:39.44ThumannMarlow: bare tit noget gammelt shit man får med apt hvis man kører woody
12:40.00ThumannMarlow: kører nu også selv sarge...
12:40.09MarlowThumann: det er fordi folk ikke forstår meningen i Debian opsplitning .
12:40.13AhrimanesMarlow: har kørt debian i 97-99 selv, men er blevet for glad for freebsd, så holder mig der
12:40.39MarlowThumann: stable vil altid være gammel, fordi den er frozen .. der kommer kun sikkerhedsopdateringer og de bliver altid porteret tilbage til de versioner, som er der i forvejen ..
12:41.01MarlowThumann: så du som sysadm roligt kan rulle en upgrade til 2000 workstations ud, uden at det knækker
12:41.12*** join/#asterisk Meaty (~cp_simbul@office.abi.ca)
12:41.25MarlowAhrimanes: eh .. dengang var jeg Slackware ..
12:41.25ThumannMarlow: true.. syntes bare det er forkert at alle bliver ved med at råbe op at.. DU SKAL HOLDE DIG TIL STABLE!!! for den alm. bruger har det nada at sige..
12:41.32MarlowAhrimanes: jeg kom først til Debian meget senere ..
12:41.35AhrimanesMarlow: var jeg så fra 96
12:41.37Ahrimaneshehe
12:41.39Ahrimanesaltså slack
12:41.43drrayI think I buggered up my voicemail folders by trying to copy the folders over
12:41.59MarlowAhrimanes: startede i '94 på Slack og i '01 over på Deb
12:42.01Ahrimanesdrray: in what way buggered?
12:42.05AhrimanesMarlow: ok
12:42.30MarlowThumann: det er så dem, som ikke har fattet det, der råber ..
12:42.32AhrimanesFreeBSD fra 99 og fremefter :D
12:43.13drrayI think the file structure has changed since I installed asterisk on the other box 200 days ago
12:43.21drrayit can not find greetings or play messages
12:43.24Ahrimanesdrray: hm ok
12:43.29drrayit knows how many old messages
12:43.38drrayI'm just going to flush the folders
12:43.42drrayand maybe make clean
12:44.00MarlowAhrimanes: jeg har ikke noget imod freebsd .. bruger det selv hist og her, men med asterisk er der altså mest udvikling på Linux .. fordi det er Asterisk native platform .
12:44.08MarlowAhrimanes: og alt andet halter bagefter
12:44.17ThumannBeer i say!!! beer
12:44.21MarlowThumann: herovre er det pints ..
12:44.26AhrimanesMarlow: jeps, men ud over zaptel har jeg ikke mødt nogle forskelle..
12:44.29Marlowand pints are GOOD
12:44.45MarlowAhrimanes: i lang tid var FreeBSD porten altid mange versioner tilbage .
12:44.48Ahrimanesbeer in quantities larger than .4l is good
12:44.57AhrimanesMarlow: jeps.. men den kan man jo bare selv opdatere
12:45.30MarlowAhrimanes: sure ..men når man har en infrastruktur på Linux fordi man startede tidligt, så er der ikke mening i at skifte
12:45.42cochi...
12:45.49AhrimanesMarlow: slet ikke
12:45.50MarlowAhrimanes: jeg har været på * vognen i næsten 3 år ..
12:45.57AhrimanesMarlow: man skal holde sig til det man har det godt med
12:46.18MarlowAhrimanes: jepper .. og det er sansløs at skifte, bare for at skifte
12:48.07Zeeek2 minutes - [14:44] <Thumann> Beer i say!!! beer
12:48.20AhrimanesMarlow: meget.. mere vigtigt at man kender sit system..
12:48.21Zeeekoh, 4 minutes
12:48.25Ahrimanesøøøøøøøøøøøl
12:48.36Zeeeksome of us have work to do
12:49.31AhrimanesZeeek: so?
12:49.42drrayif I make clean will it wipe out my .conf files?
12:49.46ZeeekNO
12:49.54Zeeekbut you're right to be afraid!
12:50.06Mocdont do make samples thought ;)
12:50.10drrayko
12:50.19drrayIt's been 6 months since I've compiled it
12:50.21drraythanks
12:50.22JerJera make samples would blow them away
12:50.33JerJerbut you should always have backups of config files
12:50.42drrayI do, I'm migrating asterisk
12:50.51drraywhich is a pain, or I'm an idiot
12:50.56drray:)
12:51.25Mocmy * migration are easy, I do my install in /home/asterisk/root, so I have 1 folder to copy that voila ;)
12:51.50drraycopied zaptel.conf, zapata, sip, extensions, voicemail, and manager
12:52.11Mociax ? copy everything
12:52.12drrayMoc that's a good idea, then just make the new version
12:52.18drrayI am not running Iax
12:53.02drrayI have a working 1t1 (asterisk box) then I am trying to replace with another 4t1 card box
12:53.04Mocyou change the PREFIX in the makefile, also I removed the asterisk name for the sub directory, like /etc/asterisk, it now only /etc
12:53.13drrayI don't want to break the working asterisk box just yet
12:53.29*** join/#asterisk oej (~oej@apollo.webway.se)
12:54.09drrayit's been rock solid with 5 fxo and 16 fxs lines
12:54.19drraynow we go to 72 fxs lines
12:55.03[TK]D-FenderFXS *shudder*
12:55.45drray:)
12:55.53drrayreplacing a mitel sx-50
12:55.57drraybaby steps
12:55.59clueconcan * use a full T1 (24 voice lines) or does it need the D Channel?
12:56.04[TK]D-FenderI'd sooner run digital sets and a SIP converter though the cost would be "unfortunate"
12:56.22[TK]D-FenderBut would forgo the woes of analog.
12:56.28drraywe don't have broadband to the units
12:56.59drrayour lobby payphone is on an Iaxy
12:59.28BerndRwho knows  app_conference? is it better than meetme?
13:00.21fenlanderBerndR: it doesn't have all the fancy options
13:00.38fenlanderBerndR: but it works well for me
13:00.47rickard25exten => 1234,1,MP3Player(/var/lib/asterisk/mohmp3/QuajiroPromo.mp3)
13:01.02rickard25what is wrong with that line should it be " involved?
13:01.27BerndRfenlander: meetme does not realy have fanzy options :(
13:01.39rickard25yeah it workes!
13:01.50BerndRfenlander: poor control by dtmf
13:01.53fenlanderBerndR: then you are probably not going to like app_conference!
13:02.03*** join/#asterisk naif (~User@213.155.196.233)
13:03.15BerndRfenlander: It's a pity but thanks
13:04.06Ahrimaneshm daewoo dect voip phone
13:04.15naifwhere?
13:04.20Ahrimanesin my hand
13:04.43Ahrimanesgoogl efor daewoo crofo 9090 if you want info it seems
13:04.48Ahrimanesbut no manual, damn
13:05.23drraythanks, make clean fixed my voicemail
13:05.41drrayI guess I'll just make them record greetings
13:07.03*** join/#asterisk cpatry (~grepmoo@65.39.228.5)
13:08.53*** part/#asterisk Dus10 (~Dus10@68-248-179-130.ded.ameritech.net)
13:10.43W|NGNUTHi all you danes. Does anyone of you know sh@warma.dk?
13:11.23*** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net)
13:13.37[TK]D-Fenderrickard25 : What error does it give you?
13:15.37[TK]D-Fenderrickard25 : Your line looks properly formatted.  Make sure it doesn't have an ID3 tag (thats a no-no) and while I'm not certain of this, VBR is probably a bad idea
13:16.02[TK]D-Fenderrickard25 : And of course make sure the files is exactly as named :)
13:16.18*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
13:17.50*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
13:20.56RoyK~seen W|NGNUT
13:20.57jbotw|ngnut <~wingnut-n@128.80-203-103.nextgentel.com> was last seen on IRC in channel #asterisk, 45d 23h 48m 14s ago, saying: 'Actually, I should have said Fujitsu Siemens...'.
13:22.05*** join/#asterisk t3chie (~jbest@66.77.171.210)
13:22.26t3chiehello
13:22.30AhrimanesW|NGNUT: am trying to see if i know the guy making it, hehe
13:22.51*** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au)
13:23.10*** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net)
13:23.26t3chiehas anyone here set up a vonage type solution?\
13:23.30W|NGNUTDamn rotating nicks..
13:24.03*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:24.36jerliquet3chie: no but I'm here with you trying to :)
13:25.13t3chiecool...we are setting up a service for businesses, using SER and asterisk for VM
13:25.38jerliquehaving any luck with it?
13:26.05t3chieso far it is going pretty good.  we have a few ata's out right now beta-testing
13:26.22jerliquecool!!!
13:26.48*** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com)
13:27.02jerliqueso how do your users get dial tone?  Do you have a gateway to the pstn or you voip'ing to another carrier?
13:27.24t3chiewe are using a carrier RNKVoIP
13:27.30t3chiethey have been great so far
13:27.45t3chietried to get on to level 3 but they were taking too long
13:27.53t3chieRNK had us set up that same day
13:28.12jerliquecan your end-users see theRNK gateway, or is that totally behind the scenes?
13:28.21newlThat's the difference between small fries and a real carrier. B)
13:28.30*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
13:28.30*** mode/#asterisk [+o twisted] by ChanServ
13:28.34HmmhesaysI have set up a vonage solution
13:28.54t3chiebehind the scenes as far as sip goes, but the RTP is peer to peer
13:28.56Hmmhesaysbut I'm pretty sure, I have access that's not publicly available yet
13:29.13Lee__RNK looks good on paper. Good to hear they also deliver for real.
13:29.14t3chieunless they are behind nat
13:29.37t3chiethey have been extremely helpfull
13:29.48jerliquewhere are they, US?
13:29.49t3chiechange things on their server on the fly for us
13:29.59t3chiethey are in massachussets i believe
13:30.32jerliqueok cool
13:31.17t3chiehmmhesays do you use asterisk only or are you using SER?
13:31.41jerliquet3chie: do you care to share your config  files for ser and * ??
13:32.19t3chieI could email them to you
13:33.15t3chienp
13:33.27Hmmhesaysis it more or is dialplan.xml completely not needed in a 7960
13:35.45*** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146)
13:35.57*** join/#asterisk pif (ldm@zenon.apartia.fr)
13:37.36jsharpWhy?  Makes sense to me.
13:37.41HmmhesaysI can't seem to find the wiki link I found yesterday, in reference to configuring 7960's
13:37.50Hmmhesaysdialplan should be handled at a centralized location
13:37.59Hmmhesaysin a pbx setup
13:39.48drumkillajsharp: I understand why it is there for SIP - I just think SIP is a terrible protocol
13:39.56drumkillathat's really the point here ;)
13:40.01jsharpah.
13:40.02jsharpGotcha.
13:40.18drumkillausing IAX, that sillyness is not needed
13:40.31jsharpSIP - designed by network weenies with no concept of telephony.  H323 - Developed by telco weenies with no concept of networking
13:41.02drumkillaheh ... I'm not too into the history of how all this happened ...
13:41.02[TK]D-FenderHey drumkilla :)  I just refreshed my CVS download and am getting ready to wipe my * modules out for recompile witha 1 CHARACTER mod to see if I can do anything at all with * source code....
13:41.09drumkillabut the specs for SIP are getting out of hand.
13:41.23drumkilla[TK]D-Fender: hahahaha ...
13:41.39drumkillaI still don't know what you did to make that not work.  :)
13:42.30[TK]D-FenderHey, I downed * but mpeg123 is floating free in a funny way, can anyone tell me what to do to fix it short of rebooting?
13:42.30[TK]D-Fender<PROTECTED>
13:42.30[TK]D-Fender30997 ?        1-17:13:10 mpg123
13:42.30[TK]D-Fender10761 ?        00:00:00 smbd
13:42.45[TK]D-Fenderthats from a ps -A
13:42.51clueconkill 30997?
13:42.53Hmmhesaysdid your machine explode?
13:42.55[TK]D-FenderTried, failed
13:43.01jsharpkill -DIERIGHTNOWDAMMIT
13:43.02[TK]D-Fenderno Hmmhesays, no smoke :)
13:43.09jerliquetry kill -9 30997
13:43.10clueconkillall mpg123?
13:43.34[TK]D-Fenderjerlique : worked, thanks.
13:43.47[TK]D-Fender<- n00b
13:43.49*** join/#asterisk newl (~newlook@203-59-15-197.dyn.iinet.net.au)
13:43.52clueconapply direct 120V to the cmos reset?
13:44.10[TK]D-FenderHA.... don't be an ass..... or anymore of one that absolutely necessary ;P
13:44.20jerliquempg always locks up. Just mkae sure that you stop asterisk 'nicely' and you will generally avoid this
13:44.38clueconTKD: It's Friday and the office is full of asses.  I'm just trying to fit in.
13:44.49jerliqueha!
13:44.53[TK]D-FenderI used "stop now" guess maybes should have used "stop gracefully" but that takes too many charachters ;)
13:45.25[TK]D-Fender<cluecon> TKD: It's Friday and the office is full of asses.  I'm just trying to fit in <- Sounds like something BKW would say ;)
13:45.31[TK]D-Fender:O
13:45.53clueconTKD: I would thank you for the compliment but I'm not sure it was meant as such.
13:45.57jerliquetype 'stop gra' and press TAB
13:47.06dalaberaHey Guys Quick Question: I will Install Fedora Core3 and would like to use Kernel 2.6 for first time, I have read that asterisk work with it, but some people have found problems with zaptel compliling. It is safe to use kernel 2.6???
13:47.31cluecondalabera: i use FC3 with the 2.6 kernel and it's fine.
13:49.20*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
13:49.25yaaarword
13:49.32clueconsentence
13:49.46yaaarcluecon: at least my comment was what it purported to be....
13:49.53yaaar(a word)
13:49.58yaaar;-)
13:50.17cluecondepending on where ya might be, mine could have been a sentence.
13:50.28yaaarit lacks a verb
13:50.31[TK]D-FenderSentence could be a sentence.  If demanded by a judge to a jury :)
13:50.39yaaarhrm
13:50.42[TK]D-FenderIt is a verb in that sense
13:50.42*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
13:50.42*** mode/#asterisk [+o bkw_] by ChanServ
13:50.45yaaari suppose
13:50.48clueconand depending on how gullible the crowd is, novel could pass for a novel.
13:51.19dalaberacluecon, Cool!! so definitely it work without a problem on 2.6!!
13:51.20[TK]D-Fendercluecon : sis you know the word "gullable" isn't in the dictionary?
13:51.29yaaarible?
13:51.35yaaarhehe
13:51.37clueconwho says you actually have to fill 300+ pages with words, just put one word in page 153 and leave it be.
13:52.16yaaarso, can i have the voicemail system email .mp3 attachments, instead of .wav?
13:53.23[TK]D-Fenderdrumkilla : compiling now....
13:53.59[TK]D-Fender<- former Grammar Ranger :D  Old-shool.....
13:54.15*** join/#asterisk mjman (~mike@205.158.42.66.ptr.us.xo.net)
13:54.58t3chieanyone in the states have a solution for 911?
13:55.04mjmanHi, I have two phones, one at ext. 7037, and one at ext. 8037. I want to set it up so that when someone dials 7037, both extensions ring. How can I do this??
13:55.16mjmant3chie: whats the problems?
13:55.21mjmanproblem
13:55.29clueconTKD: I used to really care about spelling and grammar, but I've learned that most of what I write these days wouldn't pass for much of anything (except maybe some arcane form writing) and therefore doesn't really matter.
13:56.00t3chieall voip providers need to support 911
13:56.03cluecont3chie: don't call it, they seem to get upset if you tell them you are just testing system compliance due to new FCC regulations.
13:56.49*** join/#asterisk yxa (empty@cm121.gamma228.maxonline.com.sg)
13:56.56t3chienot sure how to go about providing it
13:56.57clueconmjman: thats just a matter of setting up your extensions.  define 7037 to dial both phones.
13:57.17Hmmhesaysanyone know if 7960's have support similar to a sip subscribe... using sccp?
13:57.40mjmancluecon: well the thing is, I am using AMP, so I cannot edit the configs or else it gets overwritten
13:58.01cluecont3chie, haven't had a chance to think it thru, but it should be pretty easy to do.  just a matter of matching the phone number with a registered physical address.  that's how the bells do it and it should be same for voip.
13:58.12MeatyAnyone can help me with that ? -> http://pastebin.com/297328
13:58.15clueconi said should, not that it is of course.
13:58.42clueconmjman: not familiar with AMP or I might be able to provide more insight.  How do you configure extensions in amp?
13:58.56*** join/#asterisk trikk (~trikk@webnovative.demon.nl)
13:59.06clueconmjman: are they configured seperately from device setups?
13:59.11*** join/#asterisk mjmac (~mjmac@cpe-24-198-203-132.maine.res.rr.com)
13:59.45mjmancluecon: its a web configuration frontend for asterisk. The way it works is that when you create an extension, AMP creates an entry in sip_additional.conf and one in extensions_additional.conf
13:59.53mjmacanyone have a sip softphone and a few minutes to help me test a conference?
14:00.01Hmmhesaysor does the sccp firmware itself have any line status type support
14:00.09yaaarcluecon: what about the fact that voip connections can move around? if i take my laptop across the country, it still dials 911 from the same number.....
14:00.52yaaarmakes it a bit trickier than the bells' situation...
14:01.15*** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net)
14:01.32clueconyaaar: that's where the disclaimer has to come in.  we realize you can use it anywhere, but if your dialing 911 on your laptop and your 500 miles from home, then you really need to do one of 2 things: use a web interface to update your location everytime you travel (lat and long would work) or realize that you will never get 911 unless you are in your local area.
14:02.11yaaarcluecon: that's all well and good, except that the FCC has ruled that that's not good enough and that the voip providers have to provide accurate info.
14:02.34yaaarunless i've been misreading something someplace
14:03.15clueconthe FCC is a bunch of crusty old geezers who haven't a clue.  I haven't actually looked at the wording but that sounds about right.  Politicians just don't have a clue.
14:03.38yaaarwell, either way, we're the ones stuck fixing things to their specifications......
14:03.59yaaarin my case, i don't care as much, because i'm really looking to do stationary installations
14:04.05newlIf you don't like them, vote them outta office. B)
14:04.08clueconLogic will prevail.  There is a way to do it, but I don't have the time to think about it at the moment.
14:04.11Nuggethttp://www.livejournal.com/users/jwz/494040.html  <-- spiffy
14:04.15clueconnewl: i do.
14:04.18yaaarnewl: vote the FCC out of office?
14:04.27yaaarnewl: that would be a neat trick
14:04.41newlNo, the bastards that put them there.  Crap rolls down hill ya know.
14:04.45clueconyaaar: the politicians.
14:05.21jsharpWhen you find non-sleazy lesser-of-two-evils politicians, let me know.
14:05.30yaaarthe FCC has been operating more-or-less autonomously for some time, and these days doesn't even bother to ask congress when it wants to completely overhaul broadcast fair use
14:05.37*** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net)
14:05.46yaaarjsharp: they're called "Democrats"
14:06.13*** join/#asterisk asdfblah (~UNIX@pcp04541662pcs.brmngh01.mi.comcast.net)
14:06.16robl^anyone know if there is a MONO / .NET wrapper for iaxclient?
14:06.19clueconjsharp: i would but I have too many skeletons that would creep out normal america.  If it weren't for the fact that all the freaks would love me while all the regulars ran for the hills, i'd be running for office somewhere.
14:06.36clueconrobl: don't think there is.
14:06.36asdfblahwill the Avaya 4624 work with asterisk/
14:06.37yaaarcluecon: yeah i heard that
14:06.37asdfblah?
14:07.21newlrobl^: not afaik.  I do know someone who was writing a wrapper for libiax though but he ran into a snag with char arrays in unmanaged code and callbacks from the library.
14:07.50yaaarso, can asterisk send voicemail as .mp3 instead of .wav?
14:08.14robl^newl, yeah,  I started to do a quick hack but ran into those snags too..  that's why I was curious if someone else had worked around it
14:09.14newlrobl^: maybe you and this other bloke can put your heads together and maybe come up with a solution.
14:09.39newlIf there is one that is..
14:10.27robl^newl, if nothing else, I'd be willing to try porting iaxclient to c# instead of being a wrapper
14:10.54newlThat thought had crossed my mind as well.  It'd be more portable that way.
14:11.38robl^newl, yeah but the downside is loosing the ability to keep in sync with the original library for bug fixes and enhancements
14:11.46newltrue
14:12.49newlOTOH, if the required changes were made to the library could be done with a patch, maintaining that patch alone wouldn't be quite as bad as backporting patches library->c#
14:13.50robl^there is a java binding..  hrmm
14:14.13mjmanOK, I figured out how to make it ring the two extensions in succession, but not at the same time. How can I run two exten => lines at the same time
14:14.21*** join/#asterisk Katty (~angela@68.112.15.110)
14:14.46Hmmhesaysheh these 7960's are pretty fun
14:14.48robl^mjman, easy.  Dial(SIP/phoe1&SIP/phone2)
14:16.05mjmanrobl^ thanks!
14:16.16*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
14:16.50*** join/#asterisk gtigene (~chatzilla@70.89.216.41)
14:17.33*** join/#asterisk trasherr (trasher@dsl-084-058-010-193.arcor-ip.net)
14:17.33trasherrre
14:18.29trasherrdoes anybody know, if it's possible to answer a call with a hotkey @ x-lite?
14:18.38trasherr(sorry for ot ;p)
14:18.41[TK]D-FenderGood friggen grief!!!!! My 1 line mod to app_queue.c just doesn't work!
14:18.53[TK]D-Fenderlike it never sees it.
14:19.05jerliqueso what happens if you want to dial 100 extensions, (not a queue)
14:19.20robl^I like this filename in iaxclient "winpoop.h"
14:19.33KattyHmmhesays: i'm in conference if you want to call (=
14:19.58Hmmhesayshmm where?
14:20.06cluecon#996
14:20.12Hmmhesaysahh, that's right
14:20.15gtigeneThe wiki page for Asterisk variables mentions "one touch recording" but does not explain. Where can I read more about this, or can someone describe this feature to me, please?
14:22.37*** join/#asterisk blop (blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb)
14:27.02*** join/#asterisk brookshire (~matt@207.111.174.1)
14:29.09*** join/#asterisk denon (denon@synapse.subneural.net)
14:29.10*** mode/#asterisk [+o denon] by ChanServ
14:29.25[TK]D-FenderAny * coders here who could help me with my little problem?
14:29.26*** join/#asterisk ArkyLady (ArkyLady@adsl-208-191-242-96.dsl.ltrkar.swbell.net)
14:29.56Ahrimanes[TK]D-Fender: which problem?
14:30.12*** join/#asterisk PakiPenguin (~pingu@202.147.163.79)
14:30.16*** join/#asterisk coppice (~chatzilla@14.198.17.210.dyn.pacific.net.hk)
14:30.26*** join/#asterisk minded (~minded@65.211.26.66)
14:30.27clueconTKD has lots of problems.
14:30.27mindedhey
14:30.47[TK]D-FenderI added 1 line to app_queue.c for a 2nd verbose and it just doesn't take.  I "make clean", and "make install" after wiping out the lib/modules for it and its like its not even looking at the .c file
14:30.48mindedhow do i set up asterisk so that i can make outgoig calls?
14:30.59mindedright nwo i just get a busy signal
14:31.10sivanais there any way to fax over IP?
14:31.16PakiPenguinyes sivana
14:31.19PakiPenguinmany ways!
14:31.29PakiPenguinthe one i tried was using a sipura :)
14:31.33Ahrimanes[TK]D-Fender: and of course you reloaded app_queue.so or asterisk?
14:31.38minded:)!
14:31.49sivanaI've had no luck with Sipura over a dedicated ethernet bridge
14:32.08[TK]D-FenderAhrimanes : I shut * down prior, did the recompile and install over and then completely restarted * with "safe_asterisk &"
14:32.56*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
14:32.57PakiPenguinnope , use a modem + sipura ( sipura registered to * ) while otherside connect to fax modem  , send the fax to the modem , it will dial through sipura -> ip :)
14:33.04PakiPenguinits twisted , but works okay for me here
14:33.10[TK]D-FenderI know little about C programming, esp in linux w/ MAKE etc, but is SO an intermediay mode that needs to be rebuild for my changes to take?
14:33.50sivanaPakiPenguin: Fax Machine -> 56K Modem -> Sipura -> Internet?
14:33.51yaaarfrom a regulatory/legal standpoint, is it ok for me to simply setup voip accounts with a media gateway, setup asterisk for my clients, and then bill my clients for the phone service? or do i need to be like a clec or similar to do such things?
14:34.06PakiPenguinyes sivana
14:34.08mindedanyone have any idea how to enable outbound calls with asterisk
14:34.13PakiPenguinno no
14:34.13PakiPenguinsorry
14:34.21mindedas of right  now i get a busy signal when i try to place a call
14:34.25PakiPenguinfax machine -> sipura -> * -> Internet
14:34.39PakiPenguinminded, what are you using for outgoing connectivity? x100p?
14:35.02Ahrimanes[TK]D-Fender: the .so file is the file installed by make install
14:35.13sivanaPakiPenguin: I've tried this:  Fax Machine -> Sipura -> Ethernet Bridge (2MB) -> *
14:35.14mindedwhat do you mean paki
14:35.46yaaarminded: he's asking how you're planning on getting your asterisk box attached to the telephone network
14:35.57mindedit is attached through broadvoice
14:36.03[TK]D-FenderSo I guess I have to rebuild the SO somehow?
14:36.10PakiPenguinsivana, mind telling me what was the problem you were facing with that setup
14:36.28PakiPenguinminded, then check if your * is registered with broadvoice
14:36.41mindedi believe it is
14:36.49mindedwe can accept incoming calls
14:36.51Ahrimanes[TK]D-Fender: .so is built by make/make install
14:36.52yaaarminded: from the asterisk console, do "sip show channels"
14:36.55sivanaPakiPenguin: I get line errors... the ethernet bridge is between two locations in the same city on dedicated copper pair
14:36.55mindedfrom my phone through broad voice
14:37.06yaaarminded: oh never mind.....incoming works you must be registered
14:37.11bkw_or show channels
14:37.11PakiPenguinyes
14:37.20mindedyaaar: 0 active SIP channels
14:37.25sivanahow is Vonage offering fax service on their ATAs?
14:37.30rickard25what is the best sip-phone to use? X-ten,?
14:37.36yaaarminded: i'm sorry....brain fart.....that's sip show reg<tab>
14:37.38bkw_go read
14:37.41PakiPenguinhmms x-lite is good
14:37.47yaaarregistered or registry or somesuch
14:37.52bkw_registry
14:37.53PakiPenguinone sec sivana
14:38.06rickard25but x-lite cant seem to user conference.
14:38.09mindedyaaar: i see *CLI>
14:38.10*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
14:38.14mindedand its givin me problesm
14:38.20mindedow do i get off that
14:38.21minded;)
14:38.21*** join/#asterisk Cresl1n (~matt@216.207.245.23)
14:38.30jontow*CLI> is what you SHOULD see
14:38.30yaaarminded: that's just the prompt. ignore it
14:38.36*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:38.36*** mode/#asterisk [+o anthm] by ChanServ
14:38.45mindedthan sip show reg hmm
14:38.50mindedok
14:38.50[TK]D-FenderAhrimanes I used "make clean" and then "make install" from the * source directory and not APPS.    Is changing my MAKE line to exclude "clean" all I should need?
14:38.55*** mode/#asterisk [+o Cresl1n] by drumkilla
14:39.10mindedok yaaar: i see a url:port user name refresh and state
14:39.36Ahrimanes[TK]D-Fender: uhm clean doesnt delete .c files.. they're kept for building .o and .so
14:39.39*** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5
14:39.41yaaarminded: does it say you're registered?
14:39.46Ahrimanesoh sorry, have to leave.. back in an hour or so
14:39.51mindedum
14:39.54mindedi dont believe so
14:40.07mindedwhere would it say that?
14:40.20yaaarminded: under "state"
14:40.24mindedFailed
14:40.27minded:(
14:40.35yaaarbunk
14:40.56mindedso i take it step 1 is to register
14:40.56minded:)
14:40.56yaaarwhat's your sip.conf looking like? (pastebin)
14:41.09mindedwhats the url for pastebin again
14:41.15[TK]D-Fenderdangit....
14:41.23yaaarminded: first things first, do you have a 'register =>' line for broadvoice in sip.conf?
14:41.28yaaarminded: http://pastebin.com
14:41.33yaaar(it's a tricky one)
14:41.47mindedi do have a register line
14:41.52mindedheh yaaar:)
14:42.04yaaark
14:42.18ThumannAhrimanes: Tuc kiks owns
14:43.05*** part/#asterisk dan2 (dan@dan2.active.supporter.pdpc)
14:43.17*** join/#asterisk inspired (mikael@213.197.167.61)
14:43.51inspiredI recently discovered that ssh was killed due to lack of memory. is there some way to prevent ssh from being killed if there are memory problems?
14:44.10mindedyaaar: am i supposed to pastebin this for ya still?
14:44.18*** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
14:44.26*** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
14:47.39yaaarminded: yeah, but also post that pastebin url in here so somebody who really knows what their doing (as opposed to me) can take a look
14:47.49mindedk :)
14:48.05mindedhttp://pastebin.com/298400
14:49.22W|NGNUTOkay; agi-newbie question; why doesn't get_variable work? I have tried in both C and Perl...
14:49.52Hmmhesayswhat is the problem minded?
14:50.09mindedi am trying to setup outbound calling
14:50.12mindedbut i dont know how
14:50.18clueconwingnut: pastebin the agi.
14:50.20mindedsip show registry, state is Failed
14:50.38Hmmhesayswell i've never actually used boradvoice, but that register line looks a little off
14:51.27yaaarminded: it looks to me like you need a /<extension> on the end of your 'register' line in sip.conf. can someone more knowledgable verify that? i'm just basing it on: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
14:51.31Hmmhesays*broadvoice even
14:52.38mindedi just reloaded
14:52.43mindedand the state is now Registered
14:52.55*** join/#asterisk fantomax1 (~fanto@81.208.114.250)
14:52.58fantomax1hi all
14:53.25fantomax1can anyone tell me how to eliminate the audio message when an extension is not reachable ?
14:53.32yaaarminded: cool....can you call out?
14:54.45mindednope
14:54.48mindedstill getting a busy signal
14:55.07mindedthe console says
14:55.23mindedpbx.c:1689 pbx_extension_helper: Cannot find extension context 'from-sip'
14:55.26mindedwhen i dial out
14:55.35Hmmhesaysthere's your problem
14:55.42Godseyso in extensions.conf you don't have [from-sip]
14:56.06Hmmhesayssounds like you are using *@home
14:56.23BerndRare there some improvements in meetme from version 1.0.5 to 1.0.7 ?
14:56.30mindedso what are you aying Hmmhesays
14:56.53Hmmhesaysi'm a staunch believer newb's should not use *@home
14:57.03BerndR1.0.5 allows a caller just to mute and unmute himself
14:57.08mindedwell, i am a newb
14:57.32Hmmhesaysit will create complex config files that you don't understand
14:57.35Hmmhesaysand can't troubleshoot
14:57.57mindedwell, my boss is pretty good with this stuff, i just try to get what i can done, done, and he takes care of what i cant do
14:57.58rickard25Hmmhesays, true
14:58.00minded'_
14:58.01minded:)
14:58.06BerndRwhat about "mute all", "kick all" and so on for admins in a conference just by pressing a number?
14:58.15Hmmhesayshonestly I would suggest to you, dictch *@home, grab cvs head....compile .. enjoy
14:58.23Hmmhesays*ditch even
14:58.29mindedi dont have the authority to make a decision like tha
14:58.41Hmmhesaysgotcha
14:58.46Hmmhesaysso fix your extensions.conf
14:58.55Thumannkill who ever is in charge
14:58.57Thumanno.O
14:59.26Hmmhesaysit is the klingon way
14:59.28mindeda wiki anywhere on doig that Hmmhesays ?
14:59.33ThumannHmmhesays: indeed! :D
14:59.35squirrelv5whats my asterisk version? Connected to Asterisk CVS-HEAD currently running...
15:00.07Hmmhesayscontexts are explained in the wiki
15:00.14squirrelv5whats my asterisk version? Connected to Asterisk CVS-HEAD currently running...
15:00.15*** join/#asterisk lters (~lters@eg1.ekn.com)
15:00.37mindedHmmhesays: im lookin through the wiki for setting up extensions.conf
15:00.41mindeddont see anything about outbound calls :(
15:00.42Hmmhesaysthe *@home extensions.conf will confuse you though
15:00.49Hmmhesayshas nothing to do with outboud calling
15:00.52Hmmhesays*outbound
15:00.58Hmmhesaysyour error is context related
15:01.39mindedalright... where is the wiki at
15:01.59Hmmhesayswww.voip-info.org
15:02.34mindedya thats what im reading
15:02.36mindedargh.
15:02.59Hmmhesaysset verbose 5
15:03.07Hmmhesaysdial out, and post the console output on pastebin
15:03.24mindedset verbose 5
15:03.33mindedlike. in the asterisk console? or put that in extensions.conf
15:04.09Hmmhesaysyeah in the asterisk console
15:04.45ThumannBye bye ZyXel POE-80, Hello Cisco Catalyst 3550
15:05.03PakiPenguinhehe
15:05.20Nuggets/POE/POS/
15:05.30*** part/#asterisk Marlow (~martin@cerberus.bluetree.ie)
15:05.53Thumann10x Cisco 7960 ip-phones + 2 x Zyxel poe-80 = disaster
15:07.31*** join/#asterisk iq (~WebGuest@ext-B14-117.omhq.uprr.com)
15:07.33iqHi all
15:07.49Thumannactually cisco is the bad guy here.. their products are compatible to nothing! cisco all the way.. or no way at all.. :S
15:07.56yaaaranybody know if asterisk can send voicemail as .mp3 instead of .wav? and how to go about it?
15:07.58Thumann*with nothing
15:08.02jeffikHmmhesays: Got a question
15:08.18mindedHmmhesays: http://pastebin.com/298414
15:08.19Thumannyaaar: out of the box?
15:08.29yaaarThumann: it didn't come in a box!
15:08.45Thumannyaaar: hehe.. i know.. it came wrapped in barbwire..
15:08.50yaaaryeah
15:08.50Hmmhesaysok post that massively complex extensions.conf
15:09.15mindedheh
15:09.16mindedmines not that big
15:09.21Hmmhesaysmkay, post it
15:09.35mindedonce i set it up so i can recieve calls last night, i was like ALRLIGHT IM DONE!! so its probably shorter than it will be when im done:)
15:09.43Thumannyaaar: well.. change the format to mp3 instead of waw.. and it should work without a fuss..
15:09.48Thumannyaaar: as far as i remember..
15:09.53Hmmhesaysif you deleted some of the *@home stuff... you are in trouble
15:10.03mindedHmmhesays: http://pastebin.com/298416
15:10.25Hmmhesayswhoa
15:10.30Hmmhesaysthat's it?
15:10.34mindedya
15:10.35Thumannyaaar: our was set to it once.. but our licensewatch (spam, office control,bigbrother) thingie went bonkers.. :D
15:10.39Hmmhesaystype show dialplan on the cli and post that
15:11.19mindedhttp://pastebin.com298417
15:11.20Thumannyaaar: o.O mp3 is baaaad the boss says.. :>
15:11.22mindedhttp://pastebin.com/298417
15:11.49Hmmhesaysyou are typing from a sip phone?
15:11.56Hmmhesayser.. calling
15:11.59mindedim typing from a keyboard
15:12.02minded:)
15:12.08mindedya
15:12.11mindedits a sip phone
15:12.16Hmmhesaysshow me your sip.conf
15:12.22yaaarThumann: ok, if i change format to mp3, i get errors in the * console saying "format_mp3.c:299 mp3_rewrite: I Can't write MP3 only read them."
15:12.22Hmmhesaysyou can pm it if you want
15:13.09Thumannyaaar: asterisk-addons/format_mp3
15:13.13Thumann<PROTECTED>
15:13.13yaaarneat
15:15.10yaaarhrm....actually, how do i get that? i'm on gentoo, built from ebuild, but there's no ebuild for that addon
15:17.02Thumannyaaar: let me think
15:17.13Thumannyaaar: as i recall.. you need to have lame installed
15:17.17Thumannthe mp3 encoder
15:17.20yaaaryeah
15:17.29Thumann:D found the script
15:17.30Thumannsec
15:17.32drrayI just send the .gsm file to them instead
15:18.13yaaardrray: will .gsm files play in say, winamp?
15:18.27drraythere is a winamp plugin
15:18.28drray:)
15:18.52yaaaryeah, see, i'm not too keen on telling whole offices full of people "go install this plugin"
15:19.08yaaari'd rather have it in a format that i can be reasonably certain everybody can already play
15:19.12Thumannyaaar: f' that.. here's the recipe -> http://snippsnapp.polite.se/wiki?action=browse&diff=1&id=RecordingWithAsterisk
15:19.18yaaarsweet thanksl
15:19.18Thumanninsane link.. but it's there
15:19.36drrayI hear you, but I was not keene on spending cycles on my asterisk box encoding mp3's
15:19.50Thumannyes.. it does require some cpu usage..
15:20.18Thumannbut.. the format is usually quite.. sh!tty.. so.. if you have a reasonable box with little load on it.. shouldn't be a problem
15:20.22drrayI was just trhowing it out as an option
15:20.30drraynot saying don't use mp3
15:21.36rristroph2When recording with Record() or Monitor(), can you select different bit rates or are you always stuck with 8khz ?
15:22.07*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
15:22.53*** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
15:23.02PBXtechi have tftp with no nat/firewall.. and a cisco phone behind a firewall. gettinger Connection refused errors in the tftp log.. is that because of the NAT?
15:23.11*** join/#asterisk zoa (~zoa@pirus.securax.be)
15:24.09drrayare the tftp ports open?
15:24.16PBXtechyup
15:24.26PBXtechservice is running
15:24.29PBXtechno firewall
15:26.06*** join/#asterisk bannerman (~bannerman@209.216.176.42)
15:26.33bannermanAnyone know what the return policy is on Digium hardware? I purchased a TDM04B about three weeks ago, and I'm not satisfied with it.
15:27.45trasherrhm i am using X-Lite softphone.. does anybody know a better one?
15:28.22bkw_trasherr, google for it
15:28.55bkw_bannerman, was support not able to help you?
15:29.11zoaheloa brianella
15:29.25bkw_zoa update that jitter buffer patch for sip please please please please please
15:29.27zoawhats the problem bannerma ?
15:29.29zoayes
15:29.41zoai will try to do so for astricon
15:30.38bkw_If I had 10 bucks for every time someone came to me that had a problem getting support out of digium... I could retire.
15:31.00bkw_must be a busy company
15:31.37zoahehe
15:32.07Nuggetbannerman: http://justfuckinggoogleit.com/?q=site%3Adigium.com+return+policy
15:32.17Nuggetit's the top hit for christ's sake
15:33.03Hmmhesayssometimes tech support in general is not good..... most of the time people just want a fix without getting there head around the problem to understand the cause
15:33.07bannermanNugget: danke
15:33.36clueconNugget: that is priceless.
15:33.50drrayI got a "bummer dude" from the digium email support guy about MWI not working on an analog phone
15:34.12coppicebkw_ of venture capital your own support company :-)
15:34.52bannermanzoa: A few problems, the biggest being incorrect fax tone detection causing echo cancellation to be cancelled mid call
15:35.12bannermanzoa: the second biggest problem for me is just echo cancellatino in general, it sucks
15:35.42drraybannerman - did you try moving the card in the case? to a different PCI slot?
15:36.02bannermandrray: I moved the card to three different systems over the course of three weeks. I've been very methodical.
15:36.38drraywhat number does zttest give you?
15:37.07bannerman99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
15:37.08zoaic
15:37.12bkw_zoa do the words cross platform mean anything to you?
15:37.15bkw_bannerman, thats normal too
15:37.18zoayes
15:37.25zoabut not for this :)
15:37.28bannermanI've never seen zttest, I guess I haven't been as methodical as I thought.
15:37.31bannermanWhat is it?
15:37.39drraythat's a good number
15:37.44bannermanok
15:38.08bannermanI get the feeling I have a pretty nice setup, but I still just get crappy calls and issues that I can't resolve.
15:38.10drrayI solved my echo problem by changing the PCI slot
15:38.29bannermanI can change my echo problem by tweaking settings, and it's not an all the time problem, I do have calls that are flawless
15:38.43bkw_oh
15:38.44bannermanbut the fact is, I can downgrade to a 4-line no-pbx system for my office and be done with this BS.
15:38.45bkw_you hear echo?
15:38.48bkw_not the caller right?
15:38.52bannermanbkw_: right
15:38.56bkw_its not your end doing it
15:39.33bkw_did you recompile with agressive echo can?
15:39.36bannermanbkw_: It really doesn't matter to me which end is doing it. What matters is the fact that my boss picks up to call our client's CEO, halfway through the call he hears massive echo.
15:39.46bannermanNo, I heard more bad than good about aggressive echo cancellation.
15:39.57zoaaggressive was fine for us
15:40.04*** join/#asterisk PBXtech (~nik@67.107.241.9.ptr.us.xo.net)
15:40.05zoaand disable that fax detection
15:40.11drray:)
15:40.14bannermanI wanted fax detection.
15:40.28bkw_faxing wont' work thru a tdm board worth a flip anyway
15:40.29bannermanEither way, I'm done. From higher up, time to go back to regular phones.
15:40.48bannermanThis project was a disaster from inception
15:41.09drraymaybe you jumped the gun a bit?  it took us 60 days to roll out our asterisk box
15:41.13bkw_thats why it gets called disasterisk at times.
15:41.15bannermanWhat with TelIAX giving our number to another client and not answering my email/voicemail for three days, LiveVoip's service quitting ringtone...
15:41.31pjzohhh
15:41.44bjohnsonsixtel didn't respond to me for close to 2 weeks
15:41.45bannermandrray: I did jump the gun, but we've had 60 days to iron things out in production, and I've spent many a night up til 4 am trying something new.
15:41.57pjzbannerman: were you using voip for pstn service or just internally?
15:42.20drraybannerman, that really sucks.  I had some headaches that got solved, I've been pleased as punch since I rolled it out
15:42.40bannermanpjz: I don't undestand the question. We started out buying termination from Nufone, Livevoip, etc and wound up purchasing a TDM04B because of bad serivce.
15:42.52bannermandrray: I'm glad for you :)
15:43.34pjzbannerman: I see.  What kind of phones are you using internally?
15:43.36*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
15:43.39Nuggetnearly all of my asterisk hemorrhoids would be solved if I could find a voip did provider that didn't suck
15:43.47bannermanNugget: yes.
15:44.19*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
15:44.19*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5
15:44.38*** join/#asterisk Nix (~Nix@81.213.125.220)
15:44.43pjzbannerman: I've got all IP500s, very few problems.
15:44.55bannermanpjz: yeah, it's a great phone
15:45.08bannermanpjz: regardless, if I can't get good termination, it just doesn't matter
15:45.24bkw_we do termination
15:45.32bkw_and it doesn't suck
15:45.32pjzbannerman: yeah, I've got two 4port FXO cards too
15:45.44bannermanbkw_: that instantly puts you in the "suspect" category. Are you a homicidal maniac?
15:46.03bkw_suspect?
15:46.05bkw_how so?
15:46.14bannermanbkw_: Yes, you must be evil. All people who provide termination are evil.
15:46.17bkw_I use our service DAILY to make and take calls it works great.
15:46.30bkw_bannerman, I actually answer the phone and reply to emails.. is that evil?
15:46.39bannermanbkw_: No, that's very good.
15:47.03DeeJayTwoI see some cable companies offering for 30$/month VoIP... and for 5$ more (no fees for a whole country)..
15:47.10PakiPenguinumm
15:47.19DeeJayTwohow do they make to have so lower prices?
15:47.21bannermanbkw_: If I was still working with VoIP, I'd give you a shot
15:47.27DeeJayTwo(for long distance calls)
15:47.43PakiPenguinbkw_, which company? website?
15:47.49bkw_www.asterlink.com
15:47.53DeeJayTwoCable companies aren't usually that big..
15:49.06Hmmhesaysinteresting color scheme there
15:49.48PakiPenguinyeah
15:50.47PBXtechcan you not pull tftp config with a phone behind a NAT?
15:50.54bkw_yes
15:51.13Hmmhesaysif your isp isn't dropping tftp packets
15:51.16PBXtechdamn it. i keep getting connection refused
15:51.44*** join/#asterisk asteriskDOTbz (~logger@telux.net)
15:51.44asteriskDOTbz<PROTECTED>
15:52.11yaaarNugget: i second bkw_
15:52.13HmmhesaysHello
15:52.17NuxiActually it depends on how the tftp server operates as to whether it passes through NAT well.  most don't work well through NAT.
15:52.30*** join/#asterisk cmk (~cmk_@p54A3F397.dip.t-dialin.net)
15:52.41PBXtechtftp-hpa 0.39
15:52.58bkw_that one should work
15:53.01bkw_its one of the better ones
15:53.02PBXtechno nat on * box. just remote phone
15:53.23PBXtechhmm
15:54.02PakiPenguinwhat configuration from mysql method is better ( realtime? res_config ? or any thing else? )
15:55.10*** join/#asterisk Error_X (~alexander@217-131-211.5001.adsl.tele2.no)
15:55.46PBXtechman someone is bruteforce'ing my box via ssh
15:56.03sylesorry that was me
15:56.37sylej/k
15:56.48jontowso change the port number for sshd
15:57.04jontowway more difficult to scan a system where you know naught the port numbers :)
15:57.05Nuxiuse port 80.
15:57.08HmmhesaysI left a box up on a public IP, no firewall just to see how long it would take for someone to bust in
15:57.36Hmmhesaysit was like 9 months before anything really happened
15:57.40jontownuxi; i think that'd log all the requests still, but they'd be by webbrowsers :)
15:57.53PBXtechssh uses hosts.deny right?
15:57.59jontowit can..
15:58.07ChkDigitHmmhesays: Sample of size 1 has a large stddev...
15:58.17jontowwhat the hell.. my PRI is being ultra-flaky
15:58.25ChkDigit...sorry, other term ued for small sample sizes... I can't remember it...
15:58.36Hmmhesayswhat in the hell are you talking about?
15:58.47Hmmhesaysor do you have auto nick complete on
15:58.53*** part/#asterisk squirrelv5 (~squirrelv@202.57.81.146)
15:59.15PBXtechphew that put an end to that
15:59.27bannermanPBXtech: heh, I had the same thing when I opened up ssh
15:59.39PBXtechok back to tftp
16:01.18ltersanyone try using ast proxy manager?
16:01.30lterscan fop work via the manager on the wiki?
16:01.40ltersproxy manager that is.
16:02.52*** join/#asterisk ZOP (NachoGod@ZOP.sustaining.supporter.pdpc)
16:03.09*** join/#asterisk AgiNamu (~Michael@200.6.219.188)
16:04.51*** join/#asterisk mjman (~mike@205.158.42.66.ptr.us.xo.net)
16:05.22PakiPenguinhey AgiNamu
16:05.33*** join/#asterisk Alphahelix (~alphaheli@63.135.32.200)
16:05.40mjmanHi, I am having problems with setting up a softphone. When the softphone calls in, the destination extension rings, but when I answer, I cannot hear anything. However, the guy on the softphone can hear me
16:05.48mjmanWhat could cause this?
16:05.48AgiNamuhey
16:05.51AgiNamuwhat's new?
16:05.54jsharpNAT?  Firewalls?
16:05.58Alphahelixnat
16:05.59AgiNamumjman, SIP
16:06.07AgiNamuuse a protocol that doesn't suc
16:06.08AgiNamuk
16:06.17mjmanPlease, can I have more than one-word answers
16:06.27PakiPenguinnot much , i am having a hard time deciding what to use ( realtime , res_config or something else )
16:06.27AgiNamusure, "google NAT and SIP"
16:06.35ChkDigitmjman: Restrict the codecs in your softphone to just ulaw or alaw.
16:07.02PakiPenguinmjman, thats caused by SIP and NAT  , use a stun , or use iax!
16:07.22PakiPenguinAgiNamu, what do you suggest?
16:07.22Qwellor setup your config for NAT and SIP properly.  Do what AgiNamu suggested
16:07.22joe[TK]D-Fender: www.voipsupply.com was on of the vendors you recommended in the US, what was the other again? it went out of scope in my logs...sorry
16:07.48[TK]D-Fendervoipstore.atacomm.com
16:08.04mjmanOK
16:08.09mjmanthanks guys!
16:08.25*** join/#asterisk Weezey (Weezey@206.210.111.117)
16:08.27*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
16:08.28*** mode/#asterisk [+o bkw_] by ChanServ
16:08.32mjmanHow do I set up a STUN server?
16:08.37AgiNamuPakiPenguin, i write my own config stuff
16:08.43PakiPenguinstun.xten.net
16:08.43*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
16:09.13PakiPenguinAgiNamu, i need automatic signup , stuff , and like dum user - administration stuff , so i have to use something with mysql + php / perl
16:09.24AgiNamuPakiPenguin, I wrote something like that
16:09.24joe[TK]D-Fender: thanks again
16:09.32AgiNamucomplete 'softswitch', so to speak, on top of asterisk
16:09.33AgiNamuin C#
16:09.38AgiNamuless than a month
16:09.41PakiPenguinAgiNamu, is it opensource?
16:09.44AgiNamuuse Mono on Linux
16:09.49PakiPenguini mean free to look?
16:09.50AlphahelixI just curious if my asterisk is on a 192.168.0.0 network and my phones are on a 10.0.50.0 behind two different routers and external ip set up properly and the phones registered in asterisk why I cannot get communication between the phones\
16:09.51AgiNamuto make webservices, then connect
16:09.55*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
16:09.57AgiNamuno, it's a private project shelved
16:10.00yaaarword
16:10.01jontowjsharp; they won't take my CC because I had to move when my house burned down :-/ they refused $250 of my money yesterday
16:10.04AgiNamucause 911 had more potential :)
16:10.13mjmanPakiPenguin: that link is broken
16:10.25PakiPenguinits not a link , its a stun server address
16:10.33mjmanoh ok
16:10.35PakiPenguinAgiNamu, can we pvt?
16:10.36mjmanhaha
16:10.47jsharpI ordered a bunch of Sipuras with the 4 line upgrades.  THey shipped two line phones and wanted to charge me again to get the 4 line upgrades.
16:10.50jontowi haven't yet had a chance to change the billing address on the card; so they'll only ship it to the burnt house... which does me little good; as the carriers refuse to drop there :/
16:11.12jsharpEeeek.
16:11.13jontowso i said fuck it; i'll either go somewhere else or they'll wait
16:11.17bannermanjontow: It takes about 5 minutes to call my bank and change my billing address.
16:11.28AgiNamupvt?
16:11.32PakiPenguinprivate :)
16:11.35jontowyeah, and im damned lazy and i don't like banks :)
16:11.52jontowgetting me to deal with banks is roughly akin to pulling out my fingernails
16:12.24jsharpTreat em like a phone company.  Call the "supervisor" every 5 minutes until you get the answer you want.
16:13.04jontowheheh
16:13.40jontowwee ;P
16:13.51jontowdaylong project it is :)
16:14.26jontowsomeone is uh.. 'scanning' my PRI
16:14.30PakiPenguinjontow, banks here rely on the phone banking too much
16:14.37*** join/#asterisk waz (~tjs@208.218.27.24.cfl.res.rr.com)
16:14.58jontowi don't trust telephones.. i play with them ;)
16:15.04*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
16:15.08PakiPenguinhahah
16:15.36jontowknowing a bit about the internals makes you not want to send your private data across any distance using them..
16:15.36PakiPenguin<PROTECTED>
16:15.59jontowor at least; that's what i've picked up ;)
16:17.10PakiPenguin:)
16:17.25jsharpWe all need encrypted phones.
16:18.06greg_workjust playing with festival .. i have  Answer, Wait(1), Festival(this is a test of festival text to speech), but its only saying "-est of festival text to speech".. any ideas why?
16:18.07jontowi'd settle for one of those 80s voice scramblers
16:18.24jontowgreg; put a few .'s at the beginning of your text.. that worked for me
16:18.47greg_workjonas: thats a kludgy solution...
16:18.55jontowi didn't say it wasn't
16:18.59*** join/#asterisk morex (morex@host81-157-228-171.range81-157.btcentralplus.com)
16:19.00jontowfestival is a kludgy solution ;)
16:19.03greg_workheh
16:19.04morexHello all
16:19.12morexI have some news for you
16:19.25morexYou might be aware of the open-source JAGIServer project
16:19.39morexWhich is a FastAGI application for Asterisk
16:19.56morexWe're preparing the successor to JAGIServer, which is called OrderlyCalls.
16:20.11morexOrderlyCalls offers support for both FastAGI *and* the Manager protocols
16:20.16greg_workhm. it only happens with cmu_us_bdl_arctic_hts .. kal_diphone and ked_diphone are ok (though sound crappy)
16:20.35morexAnd offers a simple and extensible Named Service approach to providing telephony services to your Java applications.
16:21.06jontowmorex; i don't really like java; do you have one written in Tea instead?
16:21.14morexOrderlyCalls also features a Web Deployer, allowing you to proved telephony services to your web applications in any J2EE Servlet Container (such as Tomcat).
16:21.28jontow;)
16:21.34morexYou can get a sneak-peek look at OrderlyCalls at http://www.orderlyq.com/OrderlyCalls.zip
16:21.38morexWhew!
16:22.31morexJontow: No. :-P
16:24.43Weezeyif I want to see if a user is on the phone, what should I be using to interface with asterisk?
16:24.49*** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net)
16:25.01morexWeezey: You can do it with OrderlyCalls
16:25.29Lee__Weezey: show channels at the command line
16:25.43outtoluncdoes it wash dishes too? <G>
16:26.09Lee__Can OrderlyCalls make phone calls and meet people so I don't have to?
16:26.09WeezeyLee__: cool, thanks.  Can I do that with AGI somehow?
16:26.18morexOuttolunc: You can easily extend the OrderlyCalls Service class to provide any service you link, including robotic automation and dishwashing ;-)
16:26.32clueconouttolunc: only if you compile it in.  use --wash-dishes=yes during the make.
16:26.42Lee__Weezey: yes, you'll have to read up on the Manager API. I don't really know where good docs are for that. I'd assume the wiki.
16:26.49outtoluncsaweeeet
16:26.52morexLee: OrderlyCalls can make phone calls.  You can have it connect to you, or play them a message.
16:27.03WeezeyLee__: wiki was down when I tried to look up yesterday.
16:27.10morexLee/Weezey - um, OrderlyCalls has full javadocs on Manager.
16:27.19morexNot that I'm pushing it or anything :-)
16:27.23Lee__I want a computer program that can /think/ for me, so I don't have to speak.
16:27.31Weezeymorex: sure you are you OrderlyCalls whore.
16:27.46morexOrderlyCalls is by bitch...
16:27.52morexmy bitch even...
16:27.53Juggieshut up
16:27.58Juggiestop pushing your damn product
16:28.04clueconLee__: thats called wife 2.0
16:28.25morexJuggie: Sorry.  Am a bit excited as have just finished/released it.  It's open-source/free...
16:28.45Juggieput it on the wiki
16:28.46Juggiesay it once
16:28.50AgiNamudoes orderlyQ compensate for small equipment?
16:29.01morexAgi: No.
16:29.07morexNot that I'd know.
16:29.11*** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc)
16:29.30morexJuggie: I'm doing that next week - I thought I'd do a prerelease for my registered users and everybody here first
16:29.36morexJust in case anything major comes up.
16:29.53Lee__morex: it sounds cool. I'll have to check it out once I get this dev system finished.
16:30.00morexThanks Lee
16:30.16morexIt'd designed to be cool.
16:30.28morexWe use it...
16:30.32clueconmorex: what is it and where is it?
16:30.48*** join/#asterisk jimmy_deanPB (~jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
16:30.54outtolunchttp://www.orderlyq.com/OrderlyCalls.zip
16:31.07morexBeat me to it... :-)
16:31.19clueconok, so what kind of virii is in that zip?
16:31.29Lee__Click here now!!!
16:31.42AgiNamuAST.QVirus
16:31.54outtolunc.almode
16:32.00morexClue: Nothing at all, I hope.
16:32.15*** join/#asterisk cpatry (~grepmoo@65.39.228.5)
16:32.59*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
16:33.11jimmy_deanPBI am trying to get a tdm400p card going with 2 FXO and 2 FXS modules going but the card seems to lock up my computer after a day of running and ztcfg seems to be complaining as follows: "ZT_CHANCONFIG failed on channel 1: No such device or address (6)"...any ideas?
16:34.40QwellAgiNamu: If anyone actually believed that, they don't deserve to run it anyways :p
16:34.48AgiNamuLOL
16:34.54AgiNamuthat's so funny , because of a PM I just got
16:34.58Qwell...
16:35.00AgiNamuasking if it was real
16:35.23Qwellc'mon, who was it?
16:35.33Qwellpublic flogging :p
16:35.41morexI was terrified I'd distributed a virus...
16:35.58Qwellok, see, thats slightly different
16:35.58AgiNamuwell, I guess he can't run it... just debug it eh
16:36.01morexThought I'd better check.
16:36.03robl^public flogging?!?!
16:36.20AgiNamuHey, can I push my product here? It's a service.
16:36.23morexJeez, you try to do an open-source community a favour...
16:36.30Qwellcluecon: that was good :p
16:36.33robl^I wanna get flogged too!!
16:36.35AgiNamuI changed our E911 connection method. No more Internet connection required.
16:36.48robl^AgiNamu, a POTS line :)
16:36.49clueconPushers usually get flogged pretty quick round here.  Not recommended.
16:37.03morexI'm not trying to push my service.
16:37.05AgiNamuWe just assign you a DID, you dial in, with the caller ID correct (i.e., the caller ID of your subscriber)
16:37.13AgiNamuand we send it to the right selective router
16:37.15morexJust announce a new open-source project for Asterisk, is all...
16:37.21AgiNamumorex, im just razzin ya
16:37.26robl^some of us enjoy flogging :)
16:37.29AgiNamualhtouhg, I am pushing a real service :D
16:37.33morexLOL
16:37.47robl^AgiNamu, that MAY not work.  E911 doesn't use caller ID.  it uses ANI
16:37.54Alphahelixgot a quick question
16:37.58AgiNamurob^, no, we provide a gateway
16:37.58clueconmorex: looks interesting.  Will have to make some time for it so that I can see what it really does.
16:38.14AgiNamuwe read the caller ID and use that. then we use ip interconnects and so on
16:38.23AgiNamuand ALI steering agreements, etc. etc.
16:38.36morexClucon: Thanks. Let me know if you have any questions - email is in the dist.
16:38.40AgiNamubut it's easier for our customers (voip providers) to implement, if all that have to do is dial a number
16:38.41jontowAgiNamu; McPherson Struts, too?
16:39.02AgiNamu?
16:39.11jontowfor your steering agreements; of course :)
16:39.19robl^I got new struts for my Ford Escort.  does that count?
16:39.23AgiNamuoh hehe... i dont know anything about cars.
16:39.25jontowrobl; sorta ;)
16:39.28AgiNamuso i didnt get the joke
16:39.50jontow'tis ok, it was a pretty poor joke
16:39.52jontow;)
16:40.01jontowmy jeep has a seat in it from a junkyard pinto
16:40.15jontowwas the best part on the car, i heard.. :)  (was put in the jeep before it ended up in my hands)
16:40.23jimmy_deanPBanybody here that I could talk to about installing a TDM400P card for asterisk?
16:40.25robl^from a Pinto?!?!  and the Jeep hasn't exploded yet?
16:40.30jontowhahahah
16:40.35jontownah.. it runs kinda funny though
16:40.45jontowwell. actually.. yeah it did (the muffler, anyway)
16:40.51jontowbut i fixed that
16:41.01robl^its the Pinto curse!
16:41.14jontowfigures.. if i had $200-500 i'd buy new seats
16:41.28*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
16:41.30jontowthe racing-style rubicon ones.. just wish they reclined (as SO many jeep wrangler seats do not.)
16:41.32clueconHere's a thought...Why not have a queue that will call me when an agent is ready?  I mean, it would be nice to call into say the phone company and have it call me on my cell when an agent is actually ready to talk to me rather than me having to sit there for 30 minutes listening to some god awful muzak.
16:41.39jontowbut .. aside all this ;)  what about the ASTERISK!?
16:42.06jontowcluecon; 'cause they'll hangup and you'll have to wait another 30 :)
16:42.18AgiNamuWhat about if while on hold, the queue directed you to 900-hottimes?
16:42.34robl^jontow, actually.. ya wanna know the funny thing.. my parents had a pinto (1978) the engine has been in 4 bodies, including a Mercury Bobcat..  the egine is stiull in perfect condition.. but the rest of the cars keep falling apart
16:42.36jontow:)  my friend used to have a mid-70s comet.. with 50gal. drums mounted in the back to run shine with :P
16:43.00clueconno, no, no...if the agent hangs up (bad agent) then the agent gets dinged but you still get connected.
16:43.01jontowactually.. i hear he still has it stashed somewhere remote
16:43.05morexCluecon: There's a product called QueueBuster that'll do that, but it's hell-of-expensive (IMHO)
16:43.21morexWith OrderlyQ, you don't have to wait on hold
16:43.25PBXtechdang this guy that was bruteforcing my subnet got into 2 of my box's and changed the root password
16:43.26morexBut you do have to call back.
16:43.29AgiNamutelephony stuff seems to be expensive cause everyone believes its some kind of magic
16:43.34morexWorks out a lot cheaper to provide.
16:43.43AgiNamuthat's why so many companies would like to downplay asterisk
16:43.47AgiNamuit lets an idiot setup a PBX
16:43.50AgiNamuor at least try to
16:43.54jontow<PROTECTED>
16:43.58clueconAgiNamu: Telephony is magic.  * is Black Magic.
16:44.01jontowi've met a lot of idiots trying to setup asterisk.. they didn't get it :)
16:44.13clueconBlack Magic is so much cooler.
16:44.22AgiNamuWell, from the postings I read, it seems like a lot of idiots have setup astersik
16:44.29AgiNamuapparently, some  even have looked at the code ;)
16:44.31jontowhehe
16:44.51robl^that remeinds me I need to sacrifice a chicken to my Asterisk box tonite.  Does Asterisk prefer original recipie or extra crispy?
16:44.57jontowso, for some company stress-testing.. i guess i need to be uh.. generatin' some call files
16:45.00jontow.. after a cigarette
16:45.18AgiNamurobl^, any kind, so long there are biscuits
16:45.25clueconjontow: we can help you stress test, just post a phone number and see how many of us give it a try.
16:45.33AgiNamuwell, later
16:46.35jontow:) i did post a number once.. nobody called and i got lonely :(
16:46.54jontowbut now i have.. the guess who, on cd
16:46.55robl^cluecon, the Asterisk equivelent of slashdotting?
16:47.12clueconjontow: we aren't going to call your phone sex line.
16:47.18clueconrobl^: exactly.
16:48.19robl^"Press 2 for Teletubbies in heat! Tubbie-custart just the way you like it - WARM!  Remeber calls are $3.99/per minute."
16:48.51clueconthere's an idea for a website.  it will be called astdot and the purpose will be to post your did for an asterisk server you want load tested.  get a bunch of nerds to hit your box all at once while you sit back and enjoy the pretty colors.  just make sure you got the fire dept on speed dial on the pots line.
16:48.53jontow(no.)
16:49.13wazheh
16:49.24jontow(to the teletubbies deal.. just, no.)
16:49.26jontow;)
16:49.49robl^jontow, c'mon.  how can you not love the teletubbies??
16:52.00jontow.. no heat.
16:52.39robl^ok.. I've had too much caffeine :)
16:53.16jontowbash-2.05b# uptime
16:53.17jontowuptime: couldn't get boot time: No such file or directory
16:53.18jontowthat sucks..
16:54.21greg_workjontow: cat /proc/uptime
16:55.44greg_workdoes anyone know an automated way to test the quality of a voip link? i basically am looking for a way to check a couple of voip providers every x minutes and decide which is the best to use
16:56.26cianhugheshey i'm having a problem with iaxtel
16:56.37cianhughesJun 10 17:55:26 WARNING[730]: chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/69.73.19.178:4569/5 (type = 6, subclass = 1, ts=7, seqno=0)
16:56.47cianhughesalso from the same box:
16:56.53cianhughestelnet iaxtel.com 4569
16:56.53cianhughesTrying 69.73.19.178...
16:56.54cianhughestelnet: connect to address 69.73.19.178: Connection refused
16:56.54cianhughestelnet: Unable to connect to remote host
16:57.01cianhughescould their server be down?
16:57.17citatscianhughes: iax2 is not tcp so telnet wont show you anything
16:57.59cianhughesyes of course, it won't
16:58.07jontoww | head -1 <-- works better than cat /proc/uptime in this case :(
16:58.10*** join/#asterisk zeut (~zeutzeut@ool-4350c7ad.dyn.optonline.net)
16:58.24*** join/#asterisk Mw3 (mw3@daisy.chains.ch)
16:58.30greg_workjontow: w works but uptime doesnt? weird
16:58.58zeuthello... would anyone care to comment / compare asterisk to the Nortem BCM or the Avaya IP Office?
16:59.20Nugget---
16:59.20Nugget<PROTECTED>
16:59.20Nugget<PROTECTED>
16:59.20Nugget<PROTECTED>
16:59.21Nugget<PROTECTED>
16:59.24zeutMy company is setting up a anew office and would like to use asterisk but I'm nto sure if it's right for us.
16:59.24Nugget---
16:59.27Nugget^ OMG TEH ROCK
16:59.43Nuggetwho did that so I can buy them a beer?
17:01.44*** join/#asterisk wooden (usydth@p54AA1B02.dip0.t-ipconnect.de)
17:01.52woodenhi
17:01.58woodensomeone from germany here?
17:02.26*** join/#asterisk Tili (~Tili@202-133-65-13-dialup.sat.net.pk)
17:04.25zeutAlso, what are the relative merits of IP phones like the polycom IP300 versus a standard digital phone
17:04.40*** join/#asterisk sivana (~sivana@mixdown.ca)
17:05.10woodenive a asterix after a router and forwarded 5060 and 10000-11000 (like the rtp.conf) i can receive calls from a phone but when i want to call out via sipgate.de then i got a error.... but the call is listet by sipgate.de
17:05.24WeezeyNugget: haven't installed lately have you?
17:05.31*** join/#asterisk bofh42 (~bofh42@p54820135.dip0.t-ipconnect.de)
17:05.32Nuggetnope
17:06.00WeezeyNugget: it's the goods
17:06.11WeezeyI can't get chan_h323 to run though.
17:06.28woodenbut the service number (testnumber) means welcome by sipgate....
17:06.45Tiliwhat is pseuodo channel in Zaptel
17:07.29woodenis on freenode a german asterisk channel?
17:08.17denondoubt it
17:08.21denondont know though
17:08.31denontry #asterisk-de
17:08.36denonthere's someone in there
17:08.38*** join/#asterisk mdeneen (~mdeneen@151.198.54.74)
17:09.10cianhughesanyone here familiar with iaxtel
17:09.57cianhughesi have put a register line in iax.conf
17:10.06cianhughesregister => cianhughes:xxxx@iaxtel.com
17:10.08anthmTili, psuedo is a virtual zap channel used to allow software to interact with a real zap channel
17:10.40cianhughesand a context for incomming calls:
17:10.46cianhughes[iaxtel]
17:10.46cianhughestype=user
17:10.47cianhughessecret=xxxx
17:10.47cianhughescontext=iaxtel-in
17:10.48cianhughesauth=rsa
17:10.48mdeneeni saw that 1.0.8 may be released soon.  Where can I find a changelog for it?
17:10.50cianhughesinkeys=iaxtel
17:11.10cianhughesand have specified username & password in extensions.conf
17:11.19Tilianthm: so opening a pseudo channel will open any free channel. as in zt_open
17:11.20cianhughesas well as doing iaxtel-in context
17:11.28anthmon its an imaginary on
17:11.40cianhughesbut when I dial out 917009999613 I get this on the console:
17:12.18cianhughes<PROTECTED>
17:12.21cianhughes<PROTECTED>
17:12.23cianhughesJun 10 18:08:55 WARNING[730]: chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/69.73.19.178:4569/3 (type = 6, subclass = 1, ts=6, seqno=0)
17:12.27cianhughes<PROTECTED>
17:12.29cianhughes<PROTECTED>
17:12.30Tilianthm: where can i get more information on how this zaptel stuff works. i really like it all and would want to learn it
17:13.06*** part/#asterisk cpatry (~grepmoo@65.39.228.5)
17:15.28woodenwhen im after a router and nat is 5060 and 10000-10000 port-forwardings enough?
17:15.35wooden10000-11000
17:15.41woodenor more ports?
17:16.12anthmTili, nowhere that i know of besides using it until you can't help but get it
17:16.35Tilianthm: thanks
17:17.27jontowtili; http://www.zapatatelephony.org/
17:17.41Tilijontow: thanks a lot. asterisk rocks!!!!
17:17.43jontow:)
17:17.51jontowvery welcome; i like that site and found much good reading there :)
17:17.55jontowits all the older stuff
17:18.00jontowbut it'll give you an idea where this started..
17:19.49*** join/#asterisk kimosabe (~natt@dsl-200-67-12-220.prod-empresarial.com.mx)
17:20.38kimosabecould some one lead me in the correct direction for a billboard so that some one can call in and recieve a group of instructions to read option 1 for sales 2 for acounting etc
17:21.22clueconkimosabe: you want ivr help?
17:21.55kimosabenot shure what ivr is im setting up a box at home that i want to hve several options in order to see if i have mastered asterisk
17:22.07clueconhttp://www.voip-info.org/wiki-Asterisk+tips+IVR+menu
17:22.14clueconyou want ivr.
17:22.18kimosabethanks man
17:23.29jontowheh !
17:23.54jontowmy p3/450 just allowed me to shoot up to 47 channels in use (1zap, 1iax2 per call)
17:24.09jontowload average hit 6.8 and its still responsive, but definitely out of zap channels.. ;) single PRI
17:24.16docelm0I have 800 channels of SIP on my * box
17:24.25docelm0And thats just 1 box.. :)
17:24.38jontowyeah.. i have uh, 17.0 load averages now
17:24.38jontowhehe
17:27.14cyburdinels
17:27.15cyburdineuptime
17:27.15cyburdinels
17:27.22docelm0When I am aound 800 channels Im only pushing like 40% CPU usage
17:27.24cyburdinedamn wrong window sorry
17:27.30docelm0hehe
17:27.54cyburdinethank goodness I wasn't out to surf my porn collection
17:27.55yaaardocelm0: jesus. what hardware is that on?
17:28.04ChkDigitssh root@localhost
17:28.07ChkDigitpasswordsaregay
17:28.11ChkDigitWoops!
17:28.15docelm0Dual Opteron 248 w/ 8GB ram
17:28.25yaaaryeah...that oughta do it
17:29.27docelm0thats just my customer service server for where I Work. for my own network I am building 10 Dual 3.6 8GB ram clustered.. should handle about 5000 calls / second give or take
17:30.09Weezeydocelm0: g729?
17:30.10Lee__ChkDigit: why are you sshing, as root, to localhost?
17:30.41Lee__I do agree that passwords are gay, though.
17:30.58ChkDigitI was just playing fun games after cyburdine goofed up.
17:31.02WeezeyLee__: he just wants people to try sshing to his host so the have to type regina
17:31.17Weezeys/the/they/
17:31.33ChkDigitWeezey: Nice.
17:32.02ChkDigitI've never heard anyone make fun of Regina before.
17:32.18Weezeypssha
17:32.35Weezeyyou live in Canada, you've heard people make fun of the word Regina
17:32.56ChkDigit=)
17:33.17Weezey<sarcasm>has no place here</sarcasm>
17:34.34Weezeyheh, only when Grandpa docks there.
17:35.01*** join/#asterisk file[class] (~jcolp@66.199.241.90)
17:35.04file[class]helllllo
17:36.27*** join/#asterisk wooden (sxrooj@p54AA2334.dip0.t-ipconnect.de)
17:36.36woodenis this looking right?
17:36.37wooden[calling]
17:36.37woodenexten => _1.,1,Dial(SIP/${EXTEN:1}@sgmrnat80,60,tr)
17:36.37woodenexten => _1.,2,Congestion
17:36.39woodenexten => _1.,102,Busy
17:36.47*** join/#asterisk sphing (~sphing@64-238-183-42.hdlk.apt.gru.net)
17:37.39Weezeywooden: exten => _1.,1,Dial(SIP/sgmrnat80/${EXTEN:1}|60|r)
17:37.41Weezeywhat's t do?
17:37.47cyburdineI'd add exten => _1.,2,TakesSudafed right after exten => _1.,2,Congestion
17:37.48file[class]transfer magic
17:37.49jontowwow i just beat on that machin ehard ... hit 25.0 (load avg) at one point with debugging on :)
17:38.57woodenWeezey: sgmrnat80 is a entry from my provider in sip.conf
17:39.10woodenthe call is listed but every time i become a error
17:39.19woodenafter the first ring
17:39.26woodenbut i can receive calls
17:39.56woodeni will try your example now
17:40.14Weezeywooden: if it's already ringing that probably won't help you
17:40.24Weezeysounds like you need to get around nat
17:40.26woodenonly somtimes
17:41.04asdfblahanyone know how to make asterisk record voicemail in mp3 format?
17:41.11woodenno same error
17:42.16woodenthere must be something wrong with port forwards
17:42.48woodenasterisks tells sipgate.de call the number then i got a error.... the testnumber works ... welcome blabla
17:43.13woodenive nat in general on and by sgmgrnat80
17:43.19woodenat the phone, no
17:45.26kimosabeim trying to set up this ivr menu i have some doubts though if i take the one from wiki and i copy paste it how should i access the menu ?
17:46.30file[class]Nugget: picky picky.
17:46.43file[class]kimosabe: uh, you have a number that goes to the menu?
17:46.56file[class]because if you do, well then I would hypothesize - it would go to the menu.
17:47.28MikeJ[Laptop]hehe
17:47.31MikeJ[Laptop]nice
17:47.32kimosabehttp://www.voip-info.org/wiki-Asterisk+tips+IVR+menu this is what im trying to follow please correct me
17:47.55Cresl1nfile[class]: how do you get in irc in class?
17:48.09file[class]Cresl1n: SSH to a machine in San Jose, then to one in New York :)
17:48.12file[class]then use BitcHX
17:48.17Cresl1nhuh?
17:48.25file[class]SSH.
17:48.26Cresl1nyou guys have net access in class?
17:48.28Nuggetdigium needs to put up a public cvsweb
17:48.30file[class]yeah
17:48.33file[class]I'm in a computer lab
17:48.37Cresl1noh, ok
17:48.47Cresl1nwhat class is it?
17:48.59file[class]none really
17:49.04file[class]it's the last day of classes so I'm everywhere
17:50.34*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
17:51.46shidoerf?
17:52.02*** join/#asterisk YoWoudi (~myassisbi@208.205.181.170)
17:52.03citatsNugget: i've got one if you need
17:52.26YoWoudihello everyone
17:55.12shidohello
17:56.33YoWoudiive been out of the asterisk game for i think more than a year or so and want to get back into it
17:57.07*** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net)
17:57.20mike^^theoretically i could modify g729 codec and add my own trans code ?
17:57.27mike^^<- mac os x
17:57.28mike^^:-\
17:57.30mike^^thats why..
17:57.41file[class]mike^^: what?
17:57.50mike^^im asking if i could do trans encoding myself
17:57.58file[class]you can write your own codecs...
17:58.00mike^^in theory
17:58.01mike^^but
17:58.07mike^^im in no place to write a brand new codec
17:58.16mike^^im saying use the g729 and make it able to transcode from alaw -> g729
17:58.16mike^^etc
17:58.19mike^^vice versa
17:58.22file[class]it already does that...
17:58.25mike^^not the mac os x
17:58.32MikeJ[Laptop]hey.. give me my name back,
17:58.34file[class]correct, Digium hasn't compiled it for OSX
17:58.39*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
17:58.39mike^^yeah
17:58.40mike^^thats my problem
17:58.42*** join/#asterisk NormAst (~NormAst@CPE000625ee7e4e-CM0012c90d3496.cpe.net.cable.rogers.com)
17:58.52mike^^hmm
17:58.54mike^^shouldnt be too hard heh
17:58.58mike^^dont understand why they havent
17:58.59MikeJ[Laptop]mike^^, I hear it is in the works
17:58.59Ariel_hello everyone
17:58.59file[class]uh, yes it will
17:59.03file[class]you don't have the source code
17:59.07mike^^no source
17:59.08mike^^?
17:59.15mike^^i have the src in the asterisk source
17:59.16mike^^for g729
17:59.17file[class]no source code to compile it for OSX...
17:59.24shidohrmm
17:59.26mike^^so i can add transocidng myself
17:59.26file[class]you shouldn't have it...
17:59.30mike^^its nott here?
17:59.32file[class]as it's licensed and such
17:59.35mike^^im only guessing because its a module
17:59.36shidoyou have g729 source, mike^^ ?
17:59.40mike^^heh
17:59.42file[class]Digium doesn't release it
17:59.52mike^^no
17:59.56mike^^i dont mean the one that transcodes
17:59.58mike^^im saying basic one
18:00.00file[class]format_g729?
18:00.04mike^^yes
18:00.09file[class]that doesn't do transcoding or anything with G729
18:00.14mike^^i know
18:00.15mike^^but
18:00.16file[class]you can't base a codec off of it
18:00.20mike^^how come?
18:00.26file[class]it has none of the G729 codec
18:00.29mike^^ah
18:00.32mike^^i see
18:00.33MikeJ[Laptop]cuz there is no code for transcoding in it
18:00.36file[class]exactly!
18:00.38YoWoudiariel arent you oldschool here?
18:00.41Juggiethere is an opensource codec you could find for g729
18:00.45mike^^well im sure there is some g729
18:00.47MikeJ[Laptop]it is just the code to save g729 stuff to file
18:00.47mike^^yeah
18:00.48mike^^exactly
18:00.51Ariel_YoWoudi, old school?
18:01.00mike^^exactly
18:01.02Juggienone the less, you could maybe ask digium for a build.
18:01.04mike^^thats all i need
18:01.06MikeJ[Laptop]file is in school?
18:01.13mike^^i can take another codec and learn how transcoding works in those
18:01.16mike^^and base my idea off of that
18:01.20MikeJ[Laptop]I weant to the schools
18:01.24mike^^using that instead of a file
18:01.28mike^^heh
18:01.30Ariel_YoWoudi, as been on the channel for years yes and been working with computers and asterisk for years yes.
18:01.34mike^^digium for a buid? theres a petition going on
18:01.43mike^^i doubt im gonna be a "special" person ;)
18:01.56MikeJ[Laptop]if you pay for it, it will come
18:01.59mike^^heh
18:02.01mike^^uhm
18:02.05mike^^im not paying for something i cant use
18:02.08MikeJ[Laptop]or somthing like that
18:02.09mike^^and this is personal use anyway
18:02.12mike^^i'd rahter code it
18:02.14mike^^rather
18:02.21MikeJ[Laptop]you need to pay for g729
18:02.22mike^^and there is examples for outputting to files and vice-versa
18:02.32mike^^therefore I could code my own, and yeah so what pay $10 royalty so what?
18:02.38file[class]mike^^: as for making your own still, you don't have the G729 codec and it costs money
18:02.38mike^^of course i would be using their librarys
18:02.45mike^^its out there ;)
18:02.50file[class]legally it costs money
18:02.50mike^^almost all propietary sourcei s
18:02.53mike^^is
18:02.55mike^^yeah iknow
18:03.02mike^^i dont plan on trying to bypass that law
18:03.10blitzragetoo many mikes
18:03.11mike^^im saying since nobody else has built it for OS X ppc
18:03.14mike^^rather than linux PPC
18:03.21blitzragefile[class]: zup
18:03.25MikeJ[Laptop]I told him to give me my name back, he won't
18:03.28file[class]blitzrage: waiting for school to be over
18:03.36file[class]to see if my Powerbook came in @ home
18:03.36blitzragefile[class]: haha... schools been over for months! :)
18:03.43mike^^hehe
18:03.44blitzragefile[class]:  too bad powerbooks suck
18:03.46file[class]I wish.
18:03.48file[class]POWERBOOKS ROCK.
18:03.50mike^^im rebuilding my powerbooks volume structure =/
18:03.53mike^^screwed up on me
18:03.57mike^^drive 10 is my last resort
18:04.01mike^^disk warrior had no luck
18:04.06blitzragesaw*
18:04.25file[class]extreme sillyness
18:05.12mike^^i do plan on using the linux g729 for a company pbx although thats in the works
18:05.16mike^^this is my personal home phone sys
18:05.17*** join/#asterisk Dus10 (~Dus10@68-248-179-130.ded.ameritech.net)
18:06.01mike^^i wonder how hard the g729 compression is..
18:06.03mike^^as far as in asm
18:06.18mike^^it could be replicated if someone understood it well enough
18:06.25*** join/#asterisk qwertyiop (~Cain@81-178-229-82.dsl.pipex.com)
18:06.35mike^^and no im not trying to "Steal" or get it for free
18:06.47mike^^it just sucks i cant use it as far as transcoding at home
18:06.52*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
18:07.24Juggieis voip-info.org down or slow?
18:07.50|Vulture|yea I kept getting errors yesterday
18:08.04|Vulture|and now its just slow
18:09.20|Vulture|oh fun... if you send HEAD a 2 Hangups for a single channel * crashes :(
18:09.21blitzragelates :)
18:09.24|Vulture|later
18:09.27qwertyiopI've been trying to set up IAXTel in my dialplan for a couple of days, it was working for a while then stopped and started, is it me? or is it down now?
18:09.49blitzrageqwertyiop: you're missing a letter in your name
18:09.54*** join/#asterisk zoa (zoa@82.103.76.147)
18:10.10qwertyiopheh, got that the other day
18:10.22file[class]blitzrage: don't drown
18:10.29robl^qwertyiop, iaxtel has.. umm. or at least had.. MANY issues.  I know they have been working hard to fix it recently.  not sure of its current status
18:10.42mike^^yest it was down
18:10.43coppicefile: why not? its a free country
18:10.53file[class]coppice: it's very unproductive
18:11.02blitzragefile[class]: I won't, I'm a former lifeguard :)
18:11.08coppicehow authoritarian
18:11.14*** join/#asterisk anti (russ@anti.developer.gentoo)
18:11.18Cresl1nblitzrage: !@!!
18:11.19qwertyiopcool thanks, I think my setup is correct, I'll stop messing with it then.
18:11.22file[class]coppice: yup, now back to work DSP slave!
18:11.24blitzrageCresl1n: yo yo yo :)
18:11.35zoajow creslin
18:11.42Cresl1nzoa, what's up?
18:11.43coppiceblitzrage: go on. drown. exercise your freedom
18:11.46file[class]zoa!
18:11.49blitzrageWHO'S EXCITED FOR ASTRICON?! :D
18:11.49zoaand file
18:11.52zoaand coppice
18:11.53blitzragecoppice: LOL
18:11.54zoaand blitzrage
18:11.55Cresl1nwho's going to be there?
18:11.58blitzragezoa: hehe... zup :)
18:12.00zoablitzrage: are you going ?
18:12.01mike^^something diff in cvs head and 1.0.7 dealing with codecs? i tried using g729 and it said cant trans from gsm -> g729
18:12.01mike^^but
18:12.02blitzrageCresl1n: well... me :)
18:12.04blitzragezoa: yes sir
18:12.06mike^^1.0.7 lets me make calls just fine
18:12.19zoacool, me too
18:12.20robl^actually.. what would be nice if the nice Digium / IAXtel folks would update the webiste.  :)  maybe some sorta real time status of the service?  :)
18:12.21file[class]not Europe thati s.
18:12.22zoabring a book for me
18:12.23zoa"_
18:12.35file[class]the book won't appear for 2 months!!!
18:12.37file[class]:(
18:12.39qwertyiopwould be nice
18:12.46blitzragefile[class]: truw
18:12.50blitzragetrue*
18:12.55file[class]15 minutes
18:12.58file[class]T-minus 15 minutes!
18:13.07jontowwow it is hot and nice out
18:13.07*** join/#asterisk santiago (~santiago@63.245.86.198)
18:13.12mike^^i didnt look into it too much couldnt miss the calls i was wiating on although im curious if for some reason cvs head does somethign in gsm automagically ;)
18:13.18blitzragejontow: agreed - just got back from a couple of hours of tennis
18:13.28jontow:)
18:13.51jontow[11:14] <karr>  Potsdam, New York (13676) -- 2:13 PM EDT
18:13.51jontow[11:14] <karr>  Temp: 83.8 F        Humidity: 79%
18:13.51jontow[11:14] <karr>  Conditions :  with wind  at 2 mph
18:13.52jontow(woi.)
18:14.01blitzrageok... and now I have swim shorts on... so I'm off to the ool!   (notice there is no p in it - please keep it that way)
18:14.09file[class]pfft
18:14.10coppicelooks cool and dry
18:14.11Ariel_it's raining a really allot.
18:14.22*** join/#asterisk santiago (~santiago@63.245.86.198)
18:14.23blitzrage28 Centigrade here
18:14.24file[class]PeAcHy
18:14.24Ariel_Watch out Gulf coast it's coming your way.
18:14.46Cresl1ncoppice: hey, do you know where some tone generation code samples are?
18:14.49|Vulture|I think were gunna miss it here in Jacksonville Ariel_
18:15.15Ariel_It's raining so much right now here that they have flash floods warnings out.
18:15.21coppiceCresl1n: tone generation in what context?
18:15.51Cresl1nbasically pass it a frequency at a fixed sample rate (8000 samples a second) and it returns the samples
18:15.54MikeJ[Laptop]he wan't to make a beat box :)
18:16.20|Vulture|Ariel_: not good... and I was planning on driving back to Orlando on Sat.
18:16.34file[class]uh oh
18:16.36file[class]announcement
18:16.46*** join/#asterisk daXas (~on@85.96.199.40)
18:16.56MikeJ[Laptop]??
18:17.25file[class]need to clean out locker
18:17.36cluecon~cluecon
18:17.37jbotextra, extra, read all about 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.
18:17.45clueconbecause we all know file was gonna say it.
18:17.53Nuggetheh
18:18.21zoacreslin, slav is coming!
18:18.40*** join/#asterisk IAmSam (~Sam@c-24-20-237-108.hsd1.or.comcast.net)
18:19.04clueconI do not like green eggs and ham!
18:19.13*** part/#asterisk Dus10 (~Dus10@68-248-179-130.ded.ameritech.net)
18:19.31*** join/#asterisk freat (~freat@h-69-3-229-184.chcgilgm.covad.net)
18:19.34ltersjbot, with worthwhile Meetme for those of us not able to be there?
18:19.46freathello all
18:20.02|Vulture|sup freat
18:20.15freathey long time no see
18:20.22IAmSamI do like green eggs and ham
18:20.29freatdid make it to supercomm the other day
18:20.33Ariel_argh my kitchen has a leak.
18:20.37freatlots of pretty phones
18:20.45|Vulture|Ariel_: in the cieling?
18:20.46Cresl1nzoa: cool!  you'll have to introduce me
18:20.47kimosabedoes any one know whrer i can find a simple example for an ivr menu with two options
18:20.56Ariel_|Vulture|, yepper
18:20.59freatjambo kimosabe
18:21.05Nuggetwhat is the application behind #996 and the corresponding conference line?
18:21.06freatvoip-info.org
18:21.08|Vulture|Ariel_: thats not good at all
18:21.13kimosabefreat whats jambo
18:21.20freathello
18:21.26freatin swahili
18:21.27|Vulture|Ariel_: there are still ppl in Orlando that have tarps over their roof
18:21.46Ariel_|Vulture|, no it's dripping down the light.
18:21.51kimosabefreat can u lead me in the correct maner to a link  thanks man
18:22.00clueconNugget: it is customized conferencing and jitterbuffers i think.  run by asterlink.
18:22.09|Vulture|Ariel_: may want to kill power to the kitchen
18:22.38clueconkimosabe: did you check the link I posted previously?
18:22.48freathttp://www.voip-info.org/tiki-index.php?page=Asterisk+tips+IVR+menu
18:22.49Ariel_|Vulture|, shit I guess I am going to have to get on the roof as soon as the rain stops.
18:22.55kimosabecluecon yes but can get it to run i get busy signal
18:23.06freatany idea why outside calls can't be transferred by SIP phones, but internal calls can?
18:23.08kimosabei dial 205 and it sends bussy
18:23.14clueconkimosabe: what do you get on the CLI?
18:23.37Ariel_big T little t....
18:23.38kimosabecluecon i pick up phone on sip and dial 205 and cli doesnt send a thing
18:23.49freatsip debug
18:23.55kimosabeoki
18:24.09clueconincrease your verbosity.
18:24.21*** join/#asterisk Wonka (produziert@wonka.support.madwifi)
18:24.40Wonkare
18:26.13mike^^intel has open source g729 code heh
18:26.33mike^^might as well just do that until digium creates a better/cleaner * build
18:26.34mike^^asterisk
18:26.57mike^^for 1 personal line i dont see a problem myself?  and if so even then i could buy a license either way
18:27.01mike^^and wait for a build
18:27.13mike^^but i dont understand if they used PPC linux it obviously isnt an asm issue
18:27.18mike^^its more or less just compiling
18:27.34mike^^am i wrong here?
18:27.41mike^^they built on freebsd... os x is based on BSD
18:28.19*** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net)
18:28.31focksanyone using TAPI?
18:28.46coppicemike^^ you can't use the Intel code on PPC
18:29.03focksI'm trying to find a screen-popper but it seems everything is commercial software
18:29.26mike^^then
18:29.33mike^^why is there a digium formatted Linux PPC?
18:29.38mike^^im speaking of digium itself in that term, sorry
18:29.59mike^^if they can compile for Linux intel and also PPC why not BSD PPC?
18:30.04mike^^when there is a BSD x86
18:30.06coppicemike^^ the Intel G.729 is full of MMX code
18:30.15mike^^what about the PPC ver?
18:30.24*** part/#asterisk IAmSam (~Sam@c-24-20-237-108.hsd1.or.comcast.net)
18:30.55mike^^lol next year apple might have OS X on x86.. that'll be a little crazy
18:31.18mike^^even though there is already a darwin compiled for x86...but not their propietary Aqua/etc
18:31.37Hmmhesaysheh suddenly my 7960 freaks out
18:31.43coppiceI don't think anyone has done G.729 for PPC Asterisk. The G.729 reference code should build on PPC, but you'd need to turn into an asterisk module
18:31.48Hmmhesaysactually I thnk it's asterisk
18:32.00mike^^which shouldnt be hard
18:32.11shidoHmmhesays,
18:32.14mike^^im saying digium has PPC Linux
18:32.15shidowhat do you mean freak out
18:32.19shidoreorders voice?
18:32.28coppiceyou can't do it legally, though
18:32.30mike^^they built a PPC Linux binary
18:32.32mike^^how come?
18:32.34Hmmhesaysnaw, it won't register after I rebooted the pc
18:32.39shido"pc"
18:32.40shido?
18:32.43coppicewho did?
18:32.44shidowhat are you talking about?
18:32.48mike^^power pc
18:32.52mike^^diff arch
18:32.54Hmmhesaysok, the hardware running asterisk
18:32.54shidoits probably a simple sip.conf problem, Hmmhesays
18:33.02shidopastebin.ca your sip.conf
18:33.05Hmmhesaysnm, I was just making a statement anyway, I'll figure it out
18:33.07mike^^im speaking of the diff between intel and power pc (mac's arch)
18:33.13shidosure...
18:33.24mike^^if digium has a linux power pc module for asterisk built
18:33.30coppiceasterisk itself has been built for PPC,  but I don't think the G.729 codec has
18:33.33mike^^then how far can we be and why not right now from a os x ?
18:33.50mike^^yeah i know
18:33.52mike^^i use it ;)
18:34.00mike^^thats the whole reason i brought up the topic
18:34.01HmmhesaysI don't like how you assume I'm helpless shido
18:34.13shidoIm not assuming anything
18:34.16shidoyou said you had a problem
18:34.21shidoI've encountered the problem before
18:34.30shidoI CAN help.
18:34.30mike^^i'd like to use my cablemodem for a phone line without hearing voices sound like static
18:34.40*** join/#asterisk jskcr (~jskcr@jskcr.user)
18:34.43mike^^Hmmhesays: why not just pastebin?
18:34.52Qwellmike^^: So use a supported arch/distro...
18:34.54Qwellerm, os
18:34.57Hmmhesayshaha, cause i'm not a newb and i'll figure it out
18:35.01mike^^uhm,..
18:35.08mike^^for a business, sure i have no prob with that
18:35.14mike^^thats why my business setup will be x86 linux
18:35.16mike^^althoughf ro my home
18:35.29Qwellahh, so you want them to compile it for just one person, so they can sell one license
18:35.30mike^^i dont see what the big deal in compiling for os x power pc rather than linux power pc is?
18:35.31Qwellthat makes sense
18:35.32shidothen why come here if you're going to figure it out all by your self?
18:35.35mike^^uh no
18:35.39mike^^several ppl signed the online petition
18:35.44Hmmhesayscause I help other people
18:35.51mike^^if you would search for os x asterisk g729 on google
18:35.51Qwellseveral, like 5?
18:35.51shidoyou cant help yourself how can you help other people?
18:35.52Hmmhesaysand... if it's something I can't figure out, I'll ask
18:36.01Hmmhesaysheh
18:36.07mike^^so in other words... since its not as many ppl, its not something they should do?
18:36.08shidoblowing off steam, sorry Hmmhesays
18:36.09*** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)
18:36.15mike^^why would they do power pc linux and not OS X?
18:36.16Qwellmike^^: pretty much, yes
18:36.18coppicemike^^: how well does MMX and SSE code run on PPC?
18:36.19mike^^it cant be much different
18:36.22shidodont take it personally, I'll go hide back in my corner now :)
18:36.27Hmmhesaysoh you think I can't help myself?
18:36.28mike^^no idea, they did the linux version though
18:36.32*** join/#asterisk lordcian (~john@209.194.32.59)
18:36.32Hmmhesaysheh
18:36.33mike^^which means it can tbe much harder to do the os x
18:36.37mike^^since they DID do FBSD x86
18:36.39jdv79its perfectly possible to make a automated telemarketting thing with asterisk, right?
18:36.43mike^^therefore, that proves its not too much harder
18:36.44mike^^heh
18:36.44shidoheheh - no I just wanted to see if you would get heated enough to bitch loud enough
18:36.51mike^^just some code changes if that even?
18:36.55coppicethey didn't do G.729 for any PPC machine
18:36.57shidojust messin with you Hmmhesays
18:37.04mike^^maybe changing their registration program to use diff checksum functions?
18:37.07jdv79like dial all these numbers and play this wav file and let them dial out of it too...
18:37.08mike^^eh
18:37.08mike^^sec
18:37.21lordciananyone like/use a 4port fxo card NOT by digium?
18:37.24Hmmhesaysyou won't irritate me that much shido
18:37.28shidoLOL!
18:37.29HmmhesaysI got co-workers for that
18:37.32shidodamnit...
18:37.38mike^^http://www.digium.com/index.php?menu=product_detail&category=extras&product=G729
18:37.41Hmmhesaysone has been trying to get me fired for 3 days now
18:37.41mike^^linuxppc at the bottom
18:37.43Qwellshido: gotta try harder man
18:37.43shidojdv79, yes you can.
18:37.47mike^^there IS a linux power pc version, thanks
18:37.49Qwellmike^^: linux ppc != osx
18:37.53mike^^uhm
18:38.00shidohold their pbx hostage
18:38.00mike^^PPC = same arch as OS X
18:38.01jdv79also, is there a way to detect whether a real person or an answering machine has answered?
18:38.01mike^^which means
18:38.07shidono calls forrrr yewww!
18:38.07Hmmhesaysno... I won't get fired
18:38.09mike^^just like they compiled a fbsd x86
18:38.11Qwellmike^^: Does it run on beos?
18:38.12mike^^they can do a BSD ppc
18:38.14Hmmhesaysno chance at all of that
18:38.14Qwelldoes it run on solaris?
18:38.25Qwellits all x86
18:38.25mike^^heh
18:38.31coppicemike^^: they've changes the codec they use, then. In that case it should be easy for them
18:38.33Qwellshould be easy, right?
18:38.54mike^^if they cmopild for fbsd x86 and linux x86, then made a power pc linux ver
18:38.59mike^^that means if there was assembly code that needed to be changed
18:39.00Qwellmike^^: Your logic is highly flawed
18:39.03mike^^it has already been changed
18:39.07mike^^OS X is built on BSD
18:39.16mike^^therefore it shouldnt be hard to compile a BSD based version
18:39.22*** part/#asterisk jskcr (~jskcr@jskcr.user)
18:39.24mike^^since it was already created on x86 arch
18:39.29mike^^how is it flawed?
18:39.31Qwellbuilt on...huge difference
18:39.33coppicemike^^ you really are entirely clueless
18:39.37mike^^im clueless?
18:39.38Qwellbuilt on != IS
18:39.38shidonot quite jdv79 http://lists.digium.com/pipermail/asterisk-biz/2005-March/003167.html
18:39.42mike^^ok
18:39.48mike^^i can install linux on my mac right now
18:39.52mike^^and get it working on there
18:39.52mike^^but
18:39.54shidohttp://lists.digium.com/pipermail/asterisk-dev/2005-March/010060.html jdv79
18:39.55*** join/#asterisk DaLion (~DaLion@69.156.64.3)
18:39.56mike^^i cant use it on my OS X which is BSD based?
18:40.00Qwellno
18:40.05QwellNEXT!!!
18:40.12mike^^yes i know i cant
18:40.13mike^^becaquse
18:40.15mike^^they havent compiled it
18:40.15DaLionanyone have an idea why im not getting VM emails no moe.. its not even hitting my MX..
18:40.16mike^^;)
18:40.23Qwellmike^^: why would they spend the time?
18:40.32DaLionanyway to chck that ?
18:40.37QwellThey'll sell 5, maybe 10 licenses
18:40.40mike^^why not? they have no ideaon the licensing until someone buys it
18:40.46mike^^X-Servers are starting to get popular
18:40.48mike^^i hated apple
18:40.49shidoDaLion, did you cvs update lately?
18:40.51mike^^until last year
18:40.57mike^^5 maybe 10?
18:40.58mike^^yeah right
18:41.01mike^^i'll buy 10 myself
18:41.07DaLionnot sure let me check
18:41.13mike^^and i KNOW im not one out of a small bucket complaining
18:41.17mike^^and WHATS the big deal?
18:41.23DaLionCVS-HEAD-06/10/05-01:27:10
18:41.26mike^^if they have the registration code wrote, and codec wrote
18:41.30mike^^why not just compile it?
18:41.36mike^^WHY NOT is actually a better question
18:41.39mike^^they wont LOSE money
18:41.44Qwellbecause it doesn't work like that
18:41.48mike^^heh
18:41.53DaLionshildo any issue ?
18:42.01Qwellbsd is NOT osx
18:42.07mike^^you are obviouslly clueless... i have experience to back me up on coding in general on my resume
18:42.08mike^^yes i know
18:42.17mike^^they'd have to modify some of it, including headers, etc
18:42.17Qwellso it won't "just compile"
18:42.18mike^^but
18:42.20mike^^it can be done easily
18:42.24mike^^it won't be that hard whatsoever
18:42.35*** join/#asterisk fugitivo (~ajf@201.255.104.46)
18:42.38fugitivohello
18:42.41DaLionshido.. still i ddidnt touch a thing sometimes it works some others it doenst
18:43.00lordciandoes anyone use or like any 4-port fox cards that not made by digium?  more specifically, that work while allowing xwindows, frambuffer and acpi?
18:43.11mike^^chcek out darwinports
18:43.12mike^^heh
18:43.15mike^^fink...
18:43.26mike^^think those guys did all that if it took so long?
18:43.34mike^^simple patches make each one of those apps compile
18:43.40shidolordcian, quite frankly I've always ran Xwindows with asterisk but i tell everyone else not to because they use stupid hardware
18:43.53DaLionso.. why could vm stop being mailed ? basically not hitting my exim..
18:43.54mike^^it might just be a few simple sed usages and code changes
18:43.57shidoerr, lesser quality hardware
18:43.57mike^^sed for header files
18:43.58Qwella server with X is silly anyhow
18:44.03MikeJ[Laptop]mike^^ to my understanding the g729 codec in use makes use of a bunck of x86 specific code, the porting issue is not osx, its ppc
18:44.09mike^^X Server does NOT need to run X ;)
18:44.12shidoDaLion, is that a M$ product not working again for you?
18:44.28mike^^MikeJ: but digium already released a PPC linux version
18:44.28mike^^http://www.digium.com/index.php?menu=product_detail&category=extras&product=G729
18:44.32mike^^thats my whole point
18:44.38mike^^bottom of that link
18:44.43lordcianshido, maybe thats my issue, im on an hp ml110 2.4gig server with default video, but my t100p drops calls regularly if X,framebuffer on
18:44.53MikeJ[Laptop]well hell, I didn't know that
18:44.55lordcianscsi not ide, so its not dma.
18:44.57mike^^I have ran OS X without running the X server
18:45.05Nuggetafaik, the only blocker is determining the mac address.
18:45.05mike^^MikeJ: exactly
18:45.13Nugget(no pun intended)  :)
18:45.16zoahaha
18:45.17Qwellmike^^: I wasn't talking to you when I said that. :p
18:45.18MikeJ[Laptop]sigh... stop your bitching on irc, call digium and ask for it
18:45.23zoaim reading in this bulgarian take away menu
18:45.27mike^^and reguardless, who are you to say I can't use my desktop OS X machine as my * box?
18:45.28DaLionshido ????
18:45.31DaLionMS?
18:45.33DaLionlol
18:45.34zoa- pork fillet from 1930
18:45.36zoaWTF
18:45.37mike^^oh
18:45.37mike^^;)
18:45.38mike^^sorry
18:45.40shidolordcian, dood, you dont really need framebuffer
18:45.40Qwellzoa: eww
18:45.43DaLionno its asterisk not sendiing to my mail server
18:45.43shidoand STILL have a gui
18:45.45mike^^im VERY against X on production machines
18:45.48mike^^or live machines
18:45.52zoa- live bonbons from germs
18:45.52Qwellzoa: fine aged pork?
18:45.53kimosabefreat u around
18:45.57mike^^not OS X.. by saying X i mean
18:45.59DaLionthe hell is M$ in this for .. you know i hate that bit..h
18:46.02mike^^Xfree86... X-Org... etc
18:46.04jdv79thanks shido
18:46.07zoaarent germs something else ?
18:47.06lordcianyeah, i was just working if this is symptomatic of all * compatible cards, or just digiums
18:47.12lordcianworking=wondering
18:47.26mike^^i wouldnt have typed one thing about it if i didnt see linux ppc binary
18:48.28mike^^as you see it says FBSD is unsupported, all unsupported except linux x86 is optimized
18:48.36mike^^which means that might be the only one using the optimization (mmx, etc) code?
18:48.40mike^^possibly, i dont know for a fact
18:48.43mike^^but of course that PPC bin isnt using it
18:49.28freatkimosabe: hey
18:51.31lordcianmike^^ were you asking about asterisk for mac os x?  its available..  http://www.asteriskdocs.org/modules/news/article.php?storyid=31
18:53.06lordcianso, noone has used non-Digium multi-port fxo cards?
18:53.16lordcianor knows of any?
18:53.23*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
18:53.27Hmmhesaysi do believe this 7960 is my new favorite phone though
18:54.32*** part/#asterisk gtigene (~chatzilla@70.89.216.41)
18:54.57Hmmhesaysuntil I get something more shiny and new
18:55.12|Vulture|IP500
18:55.16|Vulture|:P
18:56.02*** join/#asterisk bugaloo ([U2FsdGVkX@200.214.61.137)
18:56.05kimosabefreat i have it recording the message know how can i make that the main menu sound
18:56.08Hmmhesaysyeah, I hear those are nice
18:56.30|Vulture|use to use 7960s now I use IP500s
18:56.32bugaloohi there! can some help me to detect a modem in asterisk?
18:58.03Hmmhesayswell don't be stingy send me one
18:58.07HmmhesaysxD
18:58.34*** join/#asterisk bugaloo ([U2FsdGVkX@200.214.61.137)
19:00.47bugalooanybody knows how can I configure a voice modem in asterisk?
19:01.39|Vulture|Anyone here use spandsp internally like fax machine--->FXS-->*-->spandsp via and internal extension
19:05.11*** join/#asterisk jeremywhiting (~jeremy@70-56-99-134.slkc.qwest.net)
19:05.20mindedhey
19:05.37mindeddo i have to move the demo-instruct and demo-congrats in order to use them?
19:05.41mindedbecause as of right now they arent being used at all
19:05.47mindedeven if i try to reference them
19:07.25*** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net)
19:08.22*** join/#asterisk bugaloo ([U2FsdGVkX@200.214.61.137)
19:09.05Hmmhesaysminded
19:09.08Hmmhesaysleave them in there
19:09.19Hmmhesaysleave the whole demo context in
19:09.21Hmmhesaysit comes in handy
19:09.50Qwelluninclude it...
19:10.20Hmmhesaysyeah you can do that too
19:16.51clueconyeah, use the demo to call digium and make your boss wig out cause he didn't expect to reach a live person.
19:19.15*** join/#asterisk meppl (~mephisto@p54AAC747.dip.t-dialin.net)
19:19.21mindedwhat does this error tell you
19:19.36*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
19:19.55mindedWarning[8006] pbx.c:1648 pbx_extension_helper: No application ' ' for extension (context, s, 5)
19:20.03QwellYou screwed up.
19:20.20mindedHmmhesays: ^
19:20.21QwellWhat does your exten => s,5, line look like?
19:20.28mindedsec
19:21.34mindeddont have a s,6 but
19:21.36minded5 i mean
19:21.45mindedthe 4th n, which acts as 5 says
19:22.03mindedexten => s,n,(restart),BackGround(demo-congrats)
19:22.31Qwellwhats (restart) do?
19:22.55Qwelland was that a direct copy/paste, or a retype?
19:23.44*** join/#asterisk PakiPenguin (~pingu@202.147.163.122)
19:23.48PakiPenguinhello everyone
19:24.21PakiPenguinquick help , i need to call us , right now , havent signed up with any provider yet ( except broadvoice ) I need only 5 - 10 minutes , tell me some provider that accepts creditcards please
19:24.22*** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net)
19:24.36Derkommissarim using an perl script for routing
19:24.43Derkommissarnow with a lot of traffic i get this error.
19:24.45DerkommissarJun 10 09:00:55 WARNING[6316]: res_agi.c:224 launch_script: unable to create fromast pipe: Too many open files
19:24.52jeremywhitingPakiPenguin: junction networks does: jnctn.net
19:25.03mindedQwell: retype
19:25.06jeremywhitingand they charge by the minute pre-paid too
19:25.07Derkommissarhow can i set the agi to accept more sumultanius connections ?
19:25.09mindedand im not sure what the (restart) does
19:25.31*** part/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)
19:25.44PakiPenguinumm jeremywhiting it didnt open any page
19:25.56PakiPenguinoh now it did
19:26.16Qwellminded: paste it
19:26.33mindedits on a diff box
19:26.57jeremywhitingI haven't been impressed with junction networks quality, but that is probably my server's problem, and not theirs
19:27.18shidocat /proc/cpuinfo cat /proc/pci cat /proc/meminfo on pastebin.ca jeremywhiting
19:27.57Qwellminded: is there a random char in there, screwing things up?  a space perhaps?
19:30.09PakiPenguinbut they only take US creditcards :'(
19:31.19mindedi got it
19:31.21mindedqwell
19:31.26mindedthat (restart) was notes
19:31.28minded:)
19:31.41mindedare any of the files in there the ones that say like
19:31.48mindedplease dial the extension you are trying to reach
19:33.05*** join/#asterisk file[desk] (~jcolp@mctn1-3528.nb.aliant.net)
19:33.56*** part/#asterisk lordcian (~john@209.194.32.59)
19:35.00*** join/#asterisk klictel (~klictel@207.107.208.137)
19:35.07*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
19:35.11klictelHi all
19:36.52klictelall quiet on the asterisk front
19:38.24*** join/#asterisk stkn (nobody@stkn.developer.gentoo)
19:42.05Derkommissarwho is going to astricom in spain on tuesday ?
19:42.07Derkommissar:-)
19:42.43Qwellsend me a ticket, and I'll be there. ;]
19:46.13Ariel_Derkommissar, not me. I don't have the cash for it. Are you going?
19:46.34cyburdinehey guys and gals, how does one setup an asterisk box to handle 50-100 analog phones
19:46.54cyburdineare there compatable cards that handle that many phones
19:47.19cyburdineI'm looking for hardware solutions not software configurations
19:47.34cyburdineis the answer sip or mgcp?
19:47.36Ariel_cyburdine, that is easy.  asterisk plus digium TE410 board then a few Adtrans Channel Banks 750 or 850 are best
19:47.46*** join/#asterisk FuriousGeorge (~ads@pool-138-89-118-49.nwrk.east.verizon.net)
19:48.35cyburdinegotcha so we'd need a channel bank as an intermediate between the handset and E1 card
19:49.02cyburdinewhat is the average cost of an Adtrans CB 750 or 850?
19:49.16JerJerdepends how one acquires it
19:49.23Qwelltheft
19:50.08cyburdineyeah... theft is good...
19:50.22klictelDerk: I am
19:50.29Ariel_ebay between 300 and 500.
19:51.02*** join/#asterisk bjohnson (~bjohnson@66.11.165.163)
19:51.13Derkommissaryes
19:51.17Derkommissarsup Ariel_
19:51.25Derkommissari hope its worth my time
19:51.55Ariel_Derkommissar, hope you have fun.  (Got room in your bags for a large person?) hehehe
19:52.05*** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com)
19:52.35*** join/#asterisk dsfr (~dsfr@207.111.174.1)
19:56.55woodenwow now i can call out and receive calls :>
19:57.04woodenbut when somebody call me i cant hear em
19:57.09woodenwhats the problem?
19:57.10Qwellsip?
19:57.13Qwellnat+firewall
19:57.14woodenim after a router with nat
19:57.18Qwellmmhmm
19:57.34Qwell!google sip nat site:voip-info.org
19:57.39Qwell~google sip nat site:voip-info.org
19:58.32wooden?
19:58.40Qwelldunno, bot is being slow
19:58.43woodencan u tell me whats the problem?
19:58.46Qwellyes, nat
19:58.49woodenor whats to do
19:58.56Qwellgoogle the above
19:59.58*** join/#asterisk ThiagoDamas (~ThiagoDam@vetnet4.vetorialnet.com.br)
20:00.29ThiagoDamashas someone used asterisk with R2 signaling?
20:00.31*** join/#asterisk Madkiss (madkiss@madkiss.staff.freenode)
20:00.34Madkisshi all.
20:02.04*** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)
20:02.17jdv79can asterisk talk to something like a GSX9000?
20:02.34*** part/#asterisk Cresl1n (~matt@216.207.245.23)
20:02.35jdv79i guess that'd be sip trunking or something
20:02.43JerJersure - gsx9k talks sip
20:02.56Madkissif I have two accounts for dialing others, e.g. FWD and german Freenet, is there something I can do to make sure that outgoing calls to other fWD-users are routed via FWD, and calls to everybody else take the frreenet-waY?
20:03.07jdv79want to use the GSX for its LCR and stuff but need an appserver
20:03.13jdv79asterisk can fill that role, right?
20:03.25JerJerall day long
20:03.33jdv79break a sweat?
20:03.35jdv79;)
20:03.46*** join/#asterisk kimosabe (~nat@dsl-200-67-12-220.prod-empresarial.com.mx)
20:04.05jdv79this is cool, all that's missing is comprehensive answering machine detection
20:04.26JerJeropen your favorite text editor and go at it
20:04.40JerJerthere is nothing stopping you
20:04.58kimosabefreat im recording my ivr now and i can hear it i have also moved it to /var/lib/asterisk and i named it sai-welcome   how exactlly do i do it so that i can know play it back as a message menu ?
20:05.04jdv79the fact that i am not skilled at whipping that up kinda does:)
20:05.30jdv79i've read that its not as simple as it may seem
20:05.48JerJerand don't tellme you cannot code - bkw didn't know any C before finding asterisk, now he's writing asterisk modules and bug fixes like they are going out of style
20:06.03JerJerits just math
20:06.09jdv79i know C
20:06.11*** join/#asterisk milkyflava (~milkyflav@20-156-237-24-mvl.ewc.gci.net)
20:06.14JerJerthere ya go
20:06.15jsharpThen you can code Asterisk.
20:06.18jdv79i just don't know the theory behind machien detect
20:06.35jdv79and frankly i don't have time at the moment either
20:06.51JerJerthen find a better market to target
20:06.53kimosabecan some one help me this is a menu im trying to make i am to the part where i can record the message but dont know how to call the message can some one tell me how and where is the defualt backgroun ??
20:06.59JerJeri hate telephone spammers
20:07.16JerJerdefault ?
20:07.25milkyflavaWhere can I find hardware requirements i.e RAM, proc speed, etc. etc? I want to just set up one line using my POTS line to begin playing around with asterisk.
20:07.42jdv79JerJer, so do I but i'm being paid well:)
20:07.54JerJeruntil someone sues them
20:08.02kimosabejerjer yes the route to default background   exten => 5,1,MusicOnHold(default)
20:08.07jdv79supposedly the DNC used this guy last year
20:08.16jdv79he's still in bix
20:08.18jdv79biz
20:08.22JerJermilkyflava: there are no real minimum requirements
20:08.35milkyflavamaybe a suggestion?
20:08.35jdv79http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51268&item=5779159340&rd=1&ssPageName=WDVW
20:08.46JerJermilkyflava:  you just won't be able to process that many simultaneous calls if you don't have a bunch of cpu power and a decent amount of ram
20:09.11JerJerjdv79:  its junk
20:09.12jdv79ah, the dot-com days - we had 5 or 6 of those just kickin around!  pretty well stocked too:)
20:09.13milkyflavawould 500Mhz and 256MB RAM for a single line test box would be fine?
20:09.14jsharpSingle line?  Bottom end p2, 256MB ram.
20:09.23jsharpOhyeah.
20:09.28jdv79huh?  why does qwest use them then?
20:09.38milkyflavaexcellent, thats all i have left in the house to use. :)
20:09.40JerJerthe WRT's are less than a p2's worth of power
20:09.46JerJerand only have 32 meg ram
20:09.49jdv79we went through something like a dozen vendors back in '01 and sonus trumped them all
20:09.54JerJerbleh
20:10.01jdv79that's because its a hw box...
20:10.08jdv79its not a computing platform:)
20:10.09JerJernot even global crossing -needs- something like that
20:10.34DaLionstill nothgin
20:10.35jsharpThe only real "requirement' is if you're using a TDM400 card for your pots line.  Then you need a motherboard that supports PCI 2.2
20:10.37JerJerbut if you got money to blow i'm sure that ebay seller would love to get off of that pile of junk
20:11.00klicteljdv79 answering machine is simple to code... just base it on the durations of voice vs silence
20:11.02DaLion* box doesnt send VM but its not hitting MX at all while a #>exim -v myemail@provider.com works
20:11.02jdv79i dont want it - i already have some
20:11.23jdv79klictel, are you absolutely sure about that?
20:11.47milkyflavaBut it can be done without PCI2.2, correct?
20:11.48klictelsure am... base it onthe duration of the greeting
20:11.57jdv79uh, ok
20:12.04JerJerthen i wold think one could also listen for a tone on the line
20:12.23jsharpRight.  The PCI 2.2 requirement is only valid if you want to run a TDM400P card.  If you get one of the older X100P cards, then you're fine without it.
20:12.33JerJerbut yes reading the voice energy over say 4 seconds should give you a very reasonble guess
20:12.38klictelJerJer sure can, but you want to detect it asap to relase the line asap, not necessarely wait for the end and the beep
20:13.31milkyflavaOK, so would one of the X100P be all that I would need to be able to make calls using my computer over the POTS line then?
20:13.42JerJerno
20:13.46JerJerTDM01B
20:14.00JerJeryou do not want an x100P, trust me - espcially any of the ones that are out there today
20:14.48jdv79i think, if memory serves, the GSX can handle 12 DS3s
20:14.55jdv79does that sounds right?
20:14.58milkyflavaOk, I have one telephone line coming into my house and a box that doesn't support PCI2.2, what card would be suggested so I could play around with asterisk?
20:15.10JerJerbuy a new box
20:15.15milkyflavalol
20:15.23jdv79boxen are dirt cheap
20:15.26milkyflavatrue
20:15.35JerJerget a mini-itx box
20:15.47milkyflavalike the via epia boxes?
20:15.57JerJeror hell just get a sipura 3000
20:16.06jsharpGrandstream.
20:16.17*** part/#asterisk ThiagoDamas (~ThiagoDam@vetnet4.vetorialnet.com.br)
20:16.23JerJerdoes shitstream have FXO?
20:16.48jsharpthe 486 does. I think.
20:17.06JerJerahh - didn't know that
20:17.12*** join/#asterisk jets (~jets@guardian.pmt.org)
20:17.17JerJerbut many know my feelings on their hardware
20:17.22jetsis the v option in meetme() not coded in yet
20:17.23jdv79?
20:17.27jetsvideo on the sip channel does work though?
20:17.29milkyflavaI was going to order a via M10000 for my mythtv box but I could use it to test * and the TDM400P and that would be powerful enpugh?
20:17.30JerJeri still call their phones BarbieTones
20:17.32jsharpYou've got a lot of feelings on a lot of things.
20:17.36jdv79haha
20:17.37jdv79nice
20:17.44woodencant find on google.de the problem with incoming calls and no voices
20:17.46jdv79i think i have one around here somewhere
20:17.58jsharpmilkyflava: Yes, it would work.
20:18.35jdv79is AGI or whatever its called slow?
20:18.40milkyflavathanks for the help, I really appreciate it.
20:19.37bkw_ya know what
20:19.46bkw_realtime isn't so bad if you use res_perl :P
20:19.57bkw_it actually works correctly :P
20:20.05Qwelllies!
20:20.18jdv79res_perl?
20:20.18MikeJ[Laptop]ummm,
20:20.23jdv79is that like mod_perl?
20:20.31jdv79embedded interper?
20:20.37*** join/#asterisk Kernel_core (Raph@93.230.dial-up.xter.net)
20:21.12Kernel_coreI am useing SIP , some packets get duplicated ! what should I do ?!
20:21.12bkw_hehe ya
20:21.18DaLionwooden join the lub
20:21.19bkw_Kernel_core, YOU RUDE ASS
20:21.19DaLionclub
20:21.24bkw_RUDE RUDE RUDE
20:21.31DaLionbkw
20:21.38DaLioncan i ask you sometin brian
20:21.40bkw_bust in and just start asking questions
20:21.45bkw_DaLion, sure
20:21.48Qwellwooden: nat sip site:voip-info.org
20:21.50Qwellwooden: google it
20:21.59MikeJ[Laptop]hey Kernel_core, why don't you walk in to my house and demand attention, don't worry, don't say hi or anything
20:21.59DaLioni got nat=yes canreinvite=no and qualify=1000
20:22.17DaLionbut SDP is not reewritten
20:22.22DaLionalways show as 172.16.0.200
20:22.27DaLionwich is my carp ;)
20:22.31bkw_you didn't set localnet and extern ip?
20:22.33Qwellfish ip?
20:22.34milkyflavabummer
20:22.40milkyflavaim guilty of that also
20:22.46Kernel_core!
20:22.54jdv79where is res_perl doc'd?
20:22.54*** join/#asterisk epoch (epoch@octane.breakbeats.org)
20:22.58bkw_Kernel_core, its ok..
20:23.01DaLionfuck brian your DA MAN!
20:23.02citatsomg this is the funiest thing i've seen in a long time.  http://www.killsometime.com/Video/video.asp?video=Nunchucks-Pro
20:23.06DaLionlocalnet wasnt set
20:23.10DaLionMAN
20:23.15DaLioni would kiss you but im a guy
20:23.16DaLionlol
20:23.18bkw_I just think its rude to bust in a channel and ask a questions the second you join
20:23.25bkw_DaLion, thats ok.. i'm into that!
20:23.29MikeJ[Laptop]hehe
20:23.43bkw_:P
20:23.45Kernel_corebkw_: OK I don't ask !
20:23.46jdv79i've seen that - guy's afro is awesome
20:23.55bkw_Kernel_core, just say hi ;)
20:24.00bkw_thats all i'm sayin
20:24.35milkyflavaHello and thank you..makin up for my rudeness also. :)
20:24.40Kernel_corebkw_: HI DEAR ! how are you ?
20:24.44Kernel_coreis it ok now ?! :P
20:24.49bkw_hehe ya
20:24.51jetsbk dubyaaaaaaaa
20:25.06Kernel_coreok
20:25.13bkw_I really want someone to rush up to a group of people in public they don't know.. bust in and ask a question... see how that group would react.
20:25.24bkw_they might beat your ass into the dirt
20:25.25bkw_haha
20:25.28Kernel_corenow can I ask my question !?
20:25.31bkw_sure
20:25.39jetswhich wouldn't necessarily be a negative thing.
20:25.45jdv79yeah but they're not holding up a sigh that says "asterisk people here" either
20:25.48kimosabecan some one help me with this please http://pastebin.com/298547
20:25.49jdv79sign
20:25.59kimosabeits an error on an ivr
20:26.25JerJerNo such file or directory
20:26.41kimosabeim putting them in /var/lib/asterisk
20:26.45JerJerno
20:26.53JerJertry again
20:27.05JerJer<PROTECTED>
20:27.06klictelare these real life videos or done on purpose?
20:27.13kimosabeoki thanks
20:27.31QwellThat wasn't even slightly funny
20:27.35citatsklictel: real life.  look at the Grape-Stomp one
20:27.49*** join/#asterisk sangee (~rkuru@CPE0040055ca746-CM000e5c70979a.cpe.net.cable.rogers.com)
20:28.01milkyflavafor funny videos check out www.angryalien.com
20:28.17kimosabejerler no file called sound do i just create it and it will work ?
20:28.34klictelsound is a dir
20:28.47klictelsounds
20:28.52kimosabei have /var/lib/asterisk
20:29.03kimosabebut no sounds directory in there can i just create it
20:29.12*** join/#asterisk adker (~adker@67-136-212-8.dsl1.glv.ny.frontiernet.net)
20:29.12DaLionoh hmm ok im not
20:29.20Kernel_coreI have a Problem with DTMF and asterisk , sometimes it is duplicated ! I debug my SIP here ( I pressed 1 time 6 and I got 2 times 6 in my  SIP )
20:29.29Kernel_core10 headers, 0 lines
20:29.30Kernel_coreUrgent handler
20:29.30Kernel_coreRFC3389: 1 bytes, level 4...
20:29.30Kernel_coreset_destination: Parsing <sip:1234@mycisco:5060> for address/port to send to
20:29.30Kernel_coreset_destination: set destination to mycisco, port 5060
20:29.30Kernel_coreReliably Transmitting:
20:29.31Kernel_coreINFO sip:1234@mycisco:5060 SIP/2.0
20:29.33Kernel_coreVia: SIP/2.0/UDP myasterisk:5060;branch=z9hG4bK676c6171
20:29.36Kernel_coreFrom: "2047001" <sip:2047001@myasterisk>;tag=as79511eeb
20:29.37Kernel_coreTo: <sip:1234@mycisco>;tag=CB548BC0-14D
20:29.39Kernel_coreContact: <sip:2047001@myasterisk>
20:29.41Kernel_coreCall-ID: 34fc7e925f996a0b5b551e5b6949d82b@myasterisk
20:29.43Kernel_coreCSeq: 108 INFO
20:29.45Kernel_coreUser-Agent: Asterisk PBX
20:29.47Kernel_coreContent-Type: application/dtmf-relay
20:29.49Kernel_coreContent-Length: 24
20:29.50jsharpAieee.
20:29.51Kernel_coreSignal=6
20:29.53Kernel_coreDuration=250
20:29.55Kernel_core<PROTECTED>
20:29.57Kernel_coreUrgent handler
20:29.58Qwell~pastebin
20:30.35*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
20:30.35*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5
20:30.43Qwellgonna need to kiss some serious ass to get help now.  heh
20:30.50DaLion~pastebin
20:30.51jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
20:30.57jdv79$$$ always smoothes things out
20:31.20milkyflavalol
20:31.30jdv79thanks guys
20:31.35*** join/#asterisk Tester2 (~Tester@modemcable026.175-200-24.mc.videotron.ca)
20:32.33JerJerthank you, drive thru
20:32.52QwellJerJer: I'm being lazy today.  Do you guys pass cidname?
20:32.56Tester2how to specify CVS-HEAD-06/04/05-20:08:02 when retriving cvs? (or is the last HEAD compiling?)
20:33.16*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
20:33.32NuggetI compiled HEAD about four hours ago, no problems.
20:33.53sangeecan someone help me? i got ring back tone when second leg is busy (until the dial timeout)
20:34.07Tester2and how to specify a date?
20:34.14milkyflavaIS there a knoppix type cd for the install or should I use cvs or compile by source? Is one way more common than others when asking for help?
20:34.15*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
20:34.41milkyflava-S +s
20:35.07Kernel_corehttp://217.218.80.13/kernel/sip.txt here is the link , I pressed one time 6 and I got 2 6 in my SIP debug , what is wrong ?!?! ( my client is Xten )
20:35.11harryvvAfter having been a user of sixtel services for many months now got my first email from them that thay are going to clean up there problems and promise better customer service within a cirtain time period or will get credited more time to my account.
20:35.36xeet2so they're offering an sla
20:35.36*** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com)
20:35.39xeet2hmm
20:36.07sangeecan anyone tell me how to solve the ring back tone?
20:36.09harryvvI guess
20:36.32DaLionBKW not working
20:39.38Tester2when downloading a cvs version how to check which date is it?
20:39.45kimosabejerjer i created the sounds directory in /var/lib/asterisk and i placed the sound file in there but it still will not call it
20:40.01milkyflavaThanks for all the help. Good Night.
20:40.49kimosabejerjer http://pastebin.com/298557
20:41.16*** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca)
20:41.27*** join/#asterisk stocke2 (~eric@fl-64-45-235-241.sta.sprint-hsd.net)
20:41.28DaLioncan i show u the ngrep of 5060 ?
20:41.31stocke2hello
20:41.38stocke2I have a question
20:42.04Tester2cvs checkout -D "June 4, 2005 20:08:02" asterisk ?
20:42.49shidostocke2, shoot
20:43.06stocke2we have a pbx set up in one building and we are moving some offices into a new building next door, we are going to move one t1 over there, can I set up an asterisk server over there that will allow us to transfer phonecalls over the other pbx?
20:43.10*** part/#asterisk Tester2 (~Tester@modemcable026.175-200-24.mc.videotron.ca)
20:43.24Qwellstocke2: sure
20:44.08*** join/#asterisk zotz (~zotz@208.196.247.140)
20:47.06DaLionwat does nat=route do ?
20:47.19kimosabewhat permision should the gsm files have should asterisk be the owner
20:47.54DaLionicmp 36: host 192.168.1.30 unreachable
20:47.59DaLionand this . icmp 36 ?
20:49.54Nuggetso....  some random cisco 7960 out on the internet is pulling its logo.bmp from my webserver.
20:50.01QwellNugget: nice
20:50.06Nuggetshould I replace it with goatse or tubgirl?
20:50.09QwellNugget: I would
20:50.22Qwellor...was that a multiple choice question?
20:50.33Nuggetwell, I'm open to other suggestions
20:50.44QwellYou could randomize it
20:50.48Qwellthat'd be fun for a few days
20:51.05Qwellmod_rewrite?  heh
20:51.57DaLionbrian ?
20:52.01DaLioni added localnet
20:52.03DaLionno luck
20:52.18DaLionlol
20:52.25DaLionnugget replace by your company logo
20:52.32DaLionand url
20:52.35QwellDaLion: what about externip?
20:52.57DaLionexdtrn ip is the ip of my carp setup outside.. or the FW ip if u want
20:54.03kimosabesome one here the file we call for voice have a .gsm ext should i eliminate that so that asterisk can call them becuase i have one called sai-welcome.gsm im calling it from extensions.conf with sai-welcome but it dont find it it says no file with that name
20:54.37*** join/#asterisk iswm (iswm@iswm.user)
20:54.41Qwellwow, not a single comma
20:55.00kimosabecan some one help me out with this
20:55.00DaLionhttp://pastebin.ca/13913
20:55.02DaLionqwell
20:55.10Kernel_coreI have over 700ms delay from my SIP phone to my asterisk , is there any configuration to optimize asterisk for this big latency?!
20:55.10jsharpis sai-welcome.gsm in /var/lib/asterisk/sounds
20:55.18kimosabejsharp yes
20:55.27jetsdidn't i read someone had a working skype channel or was I just high?
20:55.41QwellDaLion: I can't hit that domain, until slePP adds an ipv4. host...
20:55.49kimosabeand i also chown this file to asterisk:asterisk
20:55.52DaLionpastebin ?
20:55.56Qwellyeah
20:56.01DaLionnywhere else i can paste ?
20:56.11Qwelluuoc.com perhaps.  I probably won't be able to help though
20:56.24kimosabedalion www.pastebin.com
20:56.32Qwellpastebin.com kinda sucks
20:56.34Nuggetgoatse is unrecognizeable at that resolution and in greyscale.
20:56.37NuggetI just did http://slacker.com/~nugget/stuff/asterisk-cow.bmp instead.
20:56.46DaLionhttp://pastebin.com/298565
20:56.59kimosabejsharp u familiar with this problem ?
20:57.00QwellNugget: weak
20:57.01Qwell:p
20:57.03Nuggetyeah
20:57.09Nuggetbut there's just not enough room to work with
20:57.14Qwellyeah...
20:57.46NuggetI moved the real one to http://slacker.com/~nugget/stuff/asterisk-cow-real.bmp
20:57.59Qwellcool, time to update the config on my 7960
20:58.04Qwellerm, I mean...
20:58.05Nuggethah
20:58.05DaLionwahts specs on logo ?
20:58.10DaLionsize and quality needed ?
20:58.29DaLionand wat happens if no tftp on eboot ?
20:58.31DaLionu die ?
20:58.33QwellNugget is /whois'ing me right now, trying to match my IP up to the one in his log. :p
20:59.48Nuggetwell, I could screw him up by symlinking the logo to /dev/urandom or something
20:59.57NuggetI'll bet the phone would just keep loading and keep loading
21:00.45citatsNugget: you could try to call the phone with sip by ip :)
21:00.51Nuggethaha
21:01.19Qwellthat would be fun
21:01.27Qwellboth of those ideas
21:01.47DaLioncona nyone help
21:03.20jetsskype?
21:03.33jets~skype
21:03.34jbotfrom memory, skype is um programa de bate-papo via voz, proprietário e fechado, que usa padrões proprietários e fechados, dos mesmos autores do (spyware) kazaa; procure usar alternativas (pelo menos) com padrões abertos/livres, como os do projeto openh323 <http://www.openh323.org/>, speakfreely <http://www.speakfreely.org/> ...
21:03.41Qwellheh
21:04.04DaLionhehe its a spyware that get more spyware
21:04.05DaLionlol
21:04.32jeremywhitinghi all, anyone here have 911 set up in asterisk without pstn lines?
21:04.58jetsskype is at http://www.skype.com/ it is rumoured to be a great P2P voip service, but in all reality it's just hype no skype channels exist for * as of yet.
21:04.59*** join/#asterisk dsvanlund (~david@ua-83-227-224-57.cust.bredbandsbolaget.se)
21:05.03jets~skype
21:05.05jbotextra, extra, read all about it, skype is um programa de bate-papo via voz, proprietário e fechado, que usa padrões proprietários e fechados, dos mesmos autores do (spyware) kazaa; procure usar alternativas (pelo menos) com padrões abertos/livres, como os do projeto openh323 <http://www.openh323.org/>, speakfreely <http://www.speakfreely.org/> ...
21:05.15jetsi hate you jbot.
21:05.35Qwelljets: jbot: no, skype is blah
21:05.52Qwellmight need to msg him
21:07.05jeremywhitinganyone?
21:07.16Nuggetit just pulled the new logo.
21:07.24Qwelljeremywhiting: just call your local dispatch
21:07.29jeremywhitingor should we just wait until the fcc regulation forces juction networks to provide it for us
21:07.36shidoheh
21:07.39Qwellor, call the local PD, and ask them what you should call
21:07.43shidojeremywhiting, 911 is a touchy subject
21:07.46jeremywhitingso look up the local dispatch number in the phone book then? and set the 911 extension to go there
21:07.54Qwelljeremywhiting: I would call and ask, honestly
21:08.00jeremywhitingok, thanks
21:08.04Qwelldon't call the dispatch though :p
21:08.04shidountl the fcc basically ordered it
21:08.07Qwelljust call the PD
21:08.11shidowe werent going to touch it with a 50k ft pole
21:08.26Qwellshido: its gonna end up costing everybody more, isn't it?
21:08.38shido$10/mo
21:08.40shidofor 911
21:08.42shidoor similar
21:08.43jeremywhitingno, I wont dial 911 to find out the local number of 911, I'll call local pd
21:08.44Qwellper user?
21:08.59jeremywhitingshido: is that from v911.us or something similar? that price?
21:09.00shidothats what the user will be charged
21:09.08Qwellno opt out, eh?
21:09.32*** join/#asterisk jsharp (~jsharp@65.88.254.38)
21:09.34QwellI'd just as well cancel my service, honestly
21:09.58jsharpDidn't the FCC mandate E911 service, though...and doesn't that include address delivery to the PSAP?
21:09.59QwellI don't need a home phone that bad
21:10.09QwellI'll get a cell socket or something
21:10.23yaaarhey guys, if i install AMP or the voicemail web-access, is that going to screw up my existing extensions/voicemail/dialplan?
21:10.33shidono you can opt out
21:10.36Qwelloh?
21:10.39Qwellexcellent
21:10.39shidoyes
21:10.41shidoabsolutely
21:10.49yaaarand would you recommend running the webserver on the same box, or do i need to use a different one?
21:10.53Qwellsee, I was wondering about that the other day, and nobody had an answer
21:11.05QwellI absolutely don't want or need 911 at home...especially for that price
21:11.11shidowe might ask for a signature or something that says THIS person doesnt want our e911
21:11.19QwellI'd gladly oblige
21:11.27*** part/#asterisk DaLion (~DaLion@69.156.64.3)
21:11.31Qwellin fact, get me a form, and I'll send it in advance :p
21:12.12yaaarheck, it should be easy enough to have asterisk dial the local cop shop when a user dials 911 anyway, right?
21:12.18Qwellyaaar: exactly
21:12.39jeremywhitingthat's what we'll probably end up doing
21:12.40Qwellor, I was thinking, if somebody DOES opt out, and they call 911, and it goes to the provider
21:12.49Qwellhave a "This call will be very expensive." message
21:12.54jeremywhitingI sent an email to sales@v911.us just for kicks to see what rates they have though
21:13.07jeremywhitingmostly to show my boss that I looked into other options
21:13.16Qwellaginamu has been promoting a service...looks like < $1.50/DID
21:13.31yaaarbtw, from a regulatory/legal standpoint is it ok for me to buy voip connections from a media gateway, install them for my clients, pay the media gateway, and then bill my customers (more) for the minute-usage?
21:13.44yaaaror do i need to register as some kind of clec or similar to do that?
21:13.51Qwellyaaar: You technically become a provider at that point, I'd think
21:13.55QwellI have no clue though...don't listen to me
21:14.39yaaarwell, yeah i'd call myself a provider....i'm just asking if it would be *legal* for me to become a provider
21:14.45*** join/#asterisk smash- (~smash@198.107.16.189)
21:14.46Qwellwhy wouldn't it?
21:14.46smash-hey
21:14.55smash-<PROTECTED>
21:15.02Qwellsmash-: I think it was 300meters
21:15.05jsharp100 meters...for ethernet.
21:15.08Qwellerm, heh
21:15.11shidoyaaar, I know the feeling, the idea is so simple - why arent there more ppl becoming providers?!?!?1
21:15.12Qwell300 feet
21:15.15shidowell - its fun at first
21:15.17smash-<PROTECTED>
21:15.18smash-<PROTECTED>
21:15.19Qwellyard ~= meter, right?
21:15.22smash-<PROTECTED>
21:15.23shidothen when you go bast 500,000 customers
21:15.24jsharpAbout.
21:15.29shidoit becomes ... interesting
21:15.29yaaarshido: totally
21:15.42smash-<PROTECTED>
21:15.43Qwellshido: You guys have that many?
21:15.47shidovoicemail collects like flies on shit
21:15.48smash-is that a patch cable or cross over?
21:15.57Qwellsmash-: look at the colors
21:16.00smash-lol
21:16.02Qwellare they the same order on both ends?
21:16.05jsharpsmash-:  Ethernet or T1 or what?
21:16.05shidoemail and tech support become... fun
21:16.06smash-yeah qwell
21:16.10Qwellstraight through
21:16.11smash-<PROTECTED>
21:16.16jsharpstraight through.
21:16.19shidowhich is why we're spanking a few people around here
21:16.20yaaarshido: but see, i don't want to do call origination or media-gateway stuff or any of that....i just want to install the * server for the customer and have it drop call-detail to my billing postgres server. and bill them.
21:16.22smash-<PROTECTED>
21:16.28smash-<PROTECTED>
21:16.33shidoto turn out more quality tech support and customer service
21:16.52smash-when i use a tester i get signal, but when i plug cable into switch on 1 end and into laptop on otherend i cant get connection.
21:16.58smash-its only like 170 feet
21:17.04smash-<PROTECTED>
21:17.06yaaarshido: well, we're already an ISP, so email and tech support are already our big business
21:17.20jsharpI'm wondering if the company I work for qualifies as a "voip provider" since we offer voip services to the customers at the end of our VSAT connections.
21:17.21QwellI think I need to go work for an ISP, or a telco or something
21:17.31Qwellanybody in the southern california area hiring? ;/
21:17.37smash-<PROTECTED>
21:17.38yaaarjsharp: what's VSAT
21:17.39smash-qwell
21:17.44Qwellsmash-: ?
21:17.51smash-u asking for job i socali
21:17.52*** join/#asterisk core-ix (~saber@2001:618:495:3:2d0:b7ff:fe0f:ba6)
21:17.55yaaarQwell: rethink that.....
21:17.57Qwellsure, why not?
21:17.57smash-yeah
21:18.03smash-u dun want a job for telco company
21:18.09jsharpVSAT - Very small aperature terminals.  Small, relatively low bandwidth two-way satellite terminals.
21:18.10smash-they will have u wiring copper all day
21:18.15Qwellpsh
21:18.25QwellI'm no cable monkey
21:19.02Hmmhesaysheh proxy authentication required hell
21:19.11jsharpWe've got customers with mobile satellite packs with voip phones on the end of them and I'm wondering if we have to constantly update E911 information everytime they move a unit.
21:19.11Himekoare the sats geosync?
21:19.12*** join/#asterisk DaLion (~DaLion@69.156.64.3)
21:19.16jsharpYes.
21:19.33Hmmhesaysstupid vonage puts the src IP in the from field, so I can't match it to a user
21:19.35Qwelljsharp: currently, I don't believe you do
21:19.39DaLionexten => _9,1,Dial(Zap/1/${EXTEN:1})
21:19.39Himekothe delay is not a problem?
21:19.42DaLionwoul this work ?
21:19.47DaLiondialing 9 goes trough zap /
21:19.59Qwell_9. perhaps
21:20.05jsharp_9.
21:20.14DaLionoh
21:20.14jsharpHimeko:  No, delay isn't much of a problem.
21:20.34jsharpIts a constant delay, not a jittery internet connection.
21:21.08shidoerr
21:21.08shidono
21:21.22shidohow many more digits DaLion ?
21:21.27shido7?
21:21.28shido10?
21:21.58DaLion?
21:21.59shidoexten => _9XXXXXXX,1,Dial(Zap/1/${EXTEN:1})
21:22.03shidofor 7 digits
21:22.04QwellI guess . out zap is bad
21:22.08shido9 555-1212
21:22.19DaLionnah i need internaltionsla with 1
21:22.24shidou can do . if you dont have an extension starting with 9
21:22.27DaLionso if i dial 915555551212
21:22.33DaLioni need zap/1/15555551212
21:22.48shidothen do a exten => _91XXXXXXX.,1,Dial....
21:22.49shidoor
21:22.53smash-hey what about on a PRI > DSU/CSU|sangoma card. strait through cable qwell?
21:22.57DaLiongot it
21:22.58Qwellsmash-: dunno
21:23.06jsharpStraight through, most likely.
21:23.09smash-yeah
21:23.12smash-i figure as much
21:23.14shidoexten => _9.,1    if you dont have any other extensions with a 9
21:23.16shidoetc
21:23.19DaLion;)
21:23.20DaLionuep
21:23.21DaLionyep
21:23.23DaLionworked
21:23.29DaLiontrying to debug RTP stream bug
21:23.37DaLionwhere nat=-ey not rewriting the SDP info\
21:23.48smash-<PROTECTED>
21:24.25smash-<PROTECTED>
21:24.28kimosabecan i play mp3 as defualt on the music on hold if so how would i call it set music on hold (mozardt.mp3)   ??
21:24.41shidoif its having issues
21:24.52shidoI hope you dont have staples going through your cat5
21:25.00smash-<PROTECTED>
21:25.05smash-<PROTECTED>
21:25.07smash-<PROTECTED>
21:25.16smash-<PROTECTED>
21:25.24infizip ties are also not spectacular, velcro is better
21:25.32shidothere's different kinds of cat5 but they should all do 300 m
21:27.31infiremind me never to host my servers with you, shido
21:27.44xeet2bah
21:27.56Qwellahh, I knew there was a reason I've been helping this guy admin his server for 2 years...
21:28.10QwellI tell him I'm saving up for a 7960, he says he and his partner will buy it
21:28.22joeumm 100m is maximum iirc
21:28.28smash-<PROTECTED>
21:28.36Qwell898 feet?  damn
21:28.46*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
21:28.51smash-<PROTECTED>
21:29.43Qwellsmash-: the tester is saying 898 feet?
21:29.56smash-the writing on the cable says 898 feet
21:30.00smash-<PROTECTED>
21:30.10Qwellhand writing, or a label that says the maximum or something?
21:31.04shidoLOL
21:31.18smash-<PROTECTED>
21:31.25smash-<PROTECTED>
21:31.27smash-<PROTECTED>
21:31.32smash-<PROTECTED>
21:31.39smash-<PROTECTED>
21:31.44smash-<PROTECTED>
21:31.45Qwellumm
21:31.49smash-<PROTECTED>
21:31.52Qwelldoes it say "feet", or "'"
21:31.52smash-<PROTECTED>
21:31.59smash-says FEET
21:32.03Qwelloh, dunno
21:32.12DaLioncan we reload queues ?
21:32.14DaLionwithout
21:32.17smash-<PROTECTED>
21:32.20smash-<PROTECTED>
21:33.58DaLionreloadong all
21:35.52rristroph2is there such a thing as a java applet soft phone ?
21:36.06*** join/#asterisk gtigene (~chatzilla@70.89.216.41)
21:37.02rristroph2i.e., a java applet in my browser that could access the speaker/mic do the right magic with SIP ports and stuff -- do java applets even have permission to do that sort of stuff ?
21:37.10woodenhow to resolv the 24h disconnect problem? is asterisk refreshing the given dynhost or not?
21:37.23woodenwhen not i will mak a restart every night
21:37.27gtigeneWhen I boot to official 2.6.7 kernel image it says "/etc/modprobe.d/zaptel: ignoring bad line starting with 'post-install'" and Asterisk won't start. Does anyone know of changes to these files for kernel 2.6?
21:39.35*** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net)
21:40.23*** join/#asterisk zoa (zoa@82.103.76.147)
21:42.37Pete_Largogtigene, were you using 2.4 or 2.6 prior to 2.7?
21:42.55Pete_Largoer, should have read 2.6.7 sorry
21:43.08kimosabeif im creating a menu and from the menu i call a sip the sip ext is 2203 its in a hunt for 4 trunks 2203,2204,2209,2210  they will automatically roll how can i make it call 2203 instead of sip/phone/1/20
21:43.09gtigenePete_Largo, I was using 2.4 prior to 2.6.7
21:44.07Pete_Largothe setup for zaptel is different for 2.6.  there is some good reading in the zaptel directory.  there are some things that need to be done for udev, and also there is a compile argument (make linux26)
21:44.54*** join/#asterisk tzanger (~tzanger@mixdown.ca)
21:44.59gtigenePete_Largo, thank you.
21:45.31Pete_Largoany time
21:46.05*** part/#asterisk jeffik (~Jeff@69.158.17.52)
21:48.25kimosabeany one here in an ivr menu if i want it to dial an ext how would be the syntax
21:49.03kimosabeExten => 1,2,Dial(2203)  is this the correct manner ?
21:49.23kimosabeif i press 1 would this dial 2203 ?
21:51.07Hmmhesayscrap I have to use insecure=very
21:52.45*** join/#asterisk Cresl1n (~matt@216.207.245.23)
21:54.52kimosabeis any one any good with ivr menus
21:58.13*** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net)
21:59.56Pete_Largokimosabe, I think you need to tell it HOW to dial 2203.  I _THINK_ this would work for example: exten=>1,2,dial(SIP/2203)
22:00.39Pete_Largoor you could replace SIP with IAX2 or ZAP or.....
22:01.02kimosabethanks man
22:01.08Pete_Largoany time
22:02.51slePPQwell: which domain?
22:03.03QwellslePP: pastebin.ca
22:03.09slePPyou can't hit the ipv6 at all?
22:03.09Qwellipv4.pastebin.ca would rock my socks
22:03.12slePPwhat's the traceroute to it look like?
22:03.15slePPi thought there was.... heh
22:03.27Qwellone without an AAAA record? :p
22:04.05kimosabepete largo do you know a good how to for ivr messages menus somthing that will allow multiple options becuase know i hear it but when i press the key per say 1 it doesnt tranfer me to the following option it just rolls into the following option
22:04.23slePPQwell: just added it. ipv4.pastebin.ca. be about 2 minutes for the sync to happen
22:04.32*** join/#asterisk Darwin35 (darwin35@cas-11.ftsm-noc.mtlnk-152.valuelinx.net)
22:04.32slePPbut what's the traceroute to pastebin.ca look like? (ipv6)
22:05.13Qwellno, its almost certainly mine.   freenet acts up sometimes
22:05.20slePPfreenet always acts up :>
22:05.26rristroph2has anyone here successfully used the sipXphone from sipfoundry.org with asterisk ?
22:05.28Qwellthat too
22:05.41*** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com)
22:05.43slePPQwell: interested in a real uplink, then?
22:05.45slePPgre/sit
22:05.57QwellslePP: nah, this is free...it suits me for my purposes
22:06.09slePPso are the tunnels i can get you :P
22:06.14Qwelloh
22:06.25Qwellgonna head out in a few minutes here...discuss Monday or so?
22:06.33slePPsure. you're in NY
22:06.33slePP?
22:06.36*** join/#asterisk folsson (~filip@h82n1fls32o985.telia.com)
22:06.36Qwellno, CA
22:06.46slePPoh, so you're close to the SFO POP
22:06.54slePPyou'll like that way more than freenet6
22:06.56Qwella few hundred miles
22:07.05Qwellcool, yeah, I'll hit you up when I get back
22:07.12slePPk
22:08.08slePPyou could also peer up here to me in edmonton, since my west coast routes are good, but you'd be better off with the san fran point. (OCCAID, btw)
22:11.43PBXtechwest coast rules
22:15.03h3xwest coast west coast...
22:15.30Pete_Largokimosabe, try reading this page http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+IVR+menu
22:18.22Pete_Largokimosabe, look especially at this line... exten => 0,1,Goto,from-sip|1000
22:18.22Pete_Largo<PROTECTED>
22:18.41Pete_Largogeez, I hope that makes sense...
22:20.35*** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net)
22:21.21harryvvanyone try out that asterisk+ser live cd?
22:21.50harryvvafternoon slepp
22:22.49*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
22:28.35*** join/#asterisk flotox (jovan@host112-44.pool80181.interbusiness.it)
22:28.54*** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net)
22:29.50*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
22:32.09WilliamKhas anyone been able to get the 2B transfer to work properly on PRI?
22:33.01MikeJ[Laptop]WilliamK, to my knowledge, they are on 5ess cpe side only
22:33.18MikeJ[Laptop]so that's the first step
22:35.11jackxhe. What is Dundi exactly?
22:35.13WilliamKweird, CLEC enabled it on my NI-2
22:35.22WilliamKjust can't make it work
22:35.37WilliamKthey also enabled call controll on the PRI for both sides, not just them
22:35.58harryvvwilliam what did that t-1 pri cost you
22:36.07tzangerCLEC enabled DUNDi on your PRI?  wow I never thought I'd see the day :-)
22:36.52harryvvso much for a cheap source of ip500s on tiger direct put on hold for 10 min now :)
22:37.03WilliamKtzanger, not DUNDi, dunno where you got that =)
22:37.11WilliamKI'm talking about 2B chan transfer
22:37.18tzangerahhh
22:39.30*** join/#asterisk Slup (~Slup@62.99.100.238)
22:39.41Pete_Largo2B Chan transfer is available on DMS100/500 as well
22:40.18WilliamKthey enabled it on the Siemens switch, but I can't make it work properly under * yet
22:41.07harryvvip500 299.00 voipsupply USD and 199.00 CDN tigerdirect.ca
22:41.27SlupHi all
22:41.31harryvvThats a huge spread in price
22:41.41Pete_Largobig difference
22:41.52Slupi'm getting a bit lost with * and hoping you can help
22:42.33*** join/#asterisk tessier (~treed@146.82.146.22)
22:42.35tessierHello all
22:42.42jackxmorning
22:42.44tessierAnyone know how to make a Cisco 7960 dial direct ip to ip?
22:43.03tessierIf I go into url mode and try to enter a SIP URI it does not seem to work. But I'm really not sure what the URI should be. something@1.2.3.4?
22:43.06jackxhm. I assume Dundi only working for sip urls. not for normal extension
22:43.20jackxsip://something@something
22:43.23tessierOr do I need to say sip:something@1.2.3.4?
22:43.31tessiersip://something@something? Ok...
22:43.31MikeJ[Laptop]WilliamK, Pete_Largo, my understanding is that 2bct is only on 5ess IN ASTERISK..
22:43.51tessierThat is highly inconvenient to have to type all of that. Of course you will be calling sip://
22:43.53Slupi don't need to know exactly how to just if it's posible although basicaly how to would be nice if i have a domain name and point a sub domain to my * box can i them be name@sip.mydomain.com
22:43.54*** join/#asterisk fugitivo (~ajf@201.255.104.46)
22:43.58MikeJ[Laptop]it is available on some other protocols, but I do not beleive in asterisk
22:44.04fugitivohi
22:44.31MikeJ[Laptop]high
22:44.42*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
22:45.16fugitivoi'm having problems with sip remote clients, registering with the domain name, they can register, but can't listen any sound, registering with ip address, works ok
22:45.24jackxWhy is it inconvenient?
22:45.30xbmodderdoes anyone here have a simple extensions.conf?
22:45.39xbmodderthat they could send?
22:45.40WilliamK=)
22:45.57fugitivoany idea?
22:46.02jackxi have simple one, but i use the wrong *correct* way of building it ;)
22:46.22*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
22:46.38Slupif i have a domain name and point a sub domain to my * box can i them be name@sip.mydomain.com
22:47.05jackxwhy wouldn't it be?
22:47.25jackxworks great for me
22:47.43Slupjackx you replying to me?
22:47.52jackxyes
22:47.55Slup:)
22:48.02MikeJ[Laptop]WilliamK, it can be done... you willing to fund it?
22:48.04kimosabepbstech im trying to create an interactive ivr menu it works it ansewers it pases me to 2 diffrent menu languages  but i have 4 ext per say in this order 2203,2204,2209,2210 they are in a hunt group the only ext any one knows is 2203 per say i want the menu to allow me to dial 2203 well it does but since 2203 is busy it doesnt send me to the following option
22:48.40WilliamKMikeJ, are you offering to do it for a decent price?
22:48.40WilliamK=)
22:49.04kimosabecan anyone help me with this decision making thanks
22:49.26Slupmy brain is a bit melted so forgive if the obvoius isnt to me atm i think i just cant see the woods for the penguins jackx
22:50.10jackxWell. i use the method as well. But when i try to call myself trought sip url. no action is being taken
22:50.15jackxthat's weird
22:50.40Pete_Largokimosabe, how did you build the hunt group?
22:50.51jackxyou can try: jack@pbx.voipbroker.nl
22:51.02jackxif it starts ringing at my place, then you'll now it works :)
22:51.15jackxs/now/know
22:51.40Slupalso i think i may be confising the role of * if i want to call somone on another server/provider say bob@somevoip.com is it the * box that connects the call?
22:52.12jackxyou mean when you call with x-lite?
22:53.01jackx(or other softphone)
22:54.05Slupjackxx i have snoms
22:54.05fugitivoi'm having problems with sip remote clients, registering with the domain name, they can register, but can't listen any sound, registering with ip address, works ok
22:54.18Slupwhen i call your number it comes up NOt found
22:54.33jackxThat's pretty weird
22:54.58kimosabepete largo http://pastebin.com/298639
22:55.24Slupit's weard too how we still call it a number even though there is no numbers in it
22:55.40jackxit's an uri
22:56.09*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
22:56.48Slupjackx ya but that sounds odd
22:57.15jackxtrue
22:57.29Slupi suppose in a few years it wont
22:58.09Slupany ideas why your sip address can't be found
22:58.27kimosabepete_largo u check that out ??
23:02.06Slupany ideas why your sip address can't be found jackx
23:02.26jackxactually no.
23:02.35kimosabeany one good with decision making im ivr menus http://pastebin.com/298639
23:02.43jackxtrying some dirrent settings, they do not work
23:03.05Slupmmm what you been playing with recently
23:03.46kimosabeivr menu and a hunt need to access a sip via ivr but for it to hunt
23:03.51*** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net)
23:04.13jackxi quess: you -> * (pbx.voipbroker.nl) -> me (x-lite) should work just fine
23:04.19Pete_Largokimosabe, here is my quick and dirty and untested best guess to make an IVR transfer to a hunt group... http://pastebin.com/298642
23:05.27*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
23:05.28*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5
23:05.47kimosabei think this one might ring all ext but let me try
23:06.21jackxtry
23:06.23jackxjack@137.224.227.68
23:06.24Pete_Largocorrected the s to 1... http://pastebin.com/298646
23:06.35jackxmaby that'll work. if not.. we are even further from home :
23:07.09Pete_Largoif you want to ring 1 phonen at a time, just change the context like so...http://pastebin.com/298649
23:08.25tessierjackx: It won't try to go through my asterisk box which it is registered to if I dial by uri correct?
23:09.00jackxwell. currently i am not so sure of that
23:09.24jackxi think the stream will go between the to clients
23:09.38jackxbut auth and call control and stuff we be done by asterisk i think
23:10.09tessierMy * does not have a jack@
23:10.25tessierI think it should not go through * if I dial by uri
23:10.34tessierWish I could run ethereal on this thing...
23:12.17jackxi think you should try sip:// infront of it
23:14.41*** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca)
23:14.54tessierI did try sip:// in front
23:14.56tessierhttp://slacker.com/~nugget/asterisk7.php
23:15.15tessierI found a whole thread on the asterisk mailing list of people trying to work this out. Apparently it is not possible.
23:15.27tessierThey end up hacking the asterisk dialplan to parse and recognize the sip uri
23:16.13SarahEmmhihi
23:16.40jackxhmm
23:16.48jackxcool
23:17.15tessierhttp://www.pch.net/resources/discussion/inoc-dba/archive/2005-February/001251.html
23:17.21tessierCheck that thread out. It is what lead to that url.
23:17.31tessierBut I am trying to take asterisk out of the loop. I don't want it processing the media stream.
23:18.45Slupjackx tried that in a number of configurations and nada
23:19.01Slupbrb got to call some one
23:19.14jackxhe states: "to a remote sip client"
23:19.34jackxwhen i call local, i also get not found
23:21.42jackxwell... asterisk isn't a fully featured proxy server
23:22.48tessierjackx: Which is why we are moving to SER
23:23.44drumkillaasterisk doesn't have to process the media stream
23:23.47*** join/#asterisk jdg (~jdg@CA03F83B.adsl.mana.pf)
23:24.32jackxwhy not?
23:25.04drumkillait will issue a reinvite to tell endpoints to talk to each other
23:25.08drumkillain the case that it is possible
23:25.22drumkillait only processes media when it has to ... like codec transcoding
23:27.59jackxunderstandable
23:28.41drumkillaanyway, just wanted to make sure that was clear ...
23:28.48drumkillathough SER is still a nice complement to asterisk
23:31.25*** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com)
23:31.28*** join/#asterisk Sephen (~Sephen@proxy5.med-web.com)
23:32.14*** join/#asterisk Hmmhesays (~Neg@24-117-130-121.cpe.cableone.net)
23:33.07*** join/#asterisk Weezey (WeezeyD@206.210.109.233)
23:33.10*** join/#asterisk CoolAcid (~jk@216.99.98.39)
23:33.21*** join/#asterisk morex (morex@host81-157-123-72.range81-157.btcentralplus.com)
23:33.51Weezeyhas anyone compiled chan_h323 and got it working?  mine just crashes (no message) when I load it on two different boxes.  CVS HEAD
23:35.36harryvvdrumkilla you have alot of experaince with ser?
23:36.12jackxHow is asterisk compared to commercial fabrications? (avaya/cisco)
23:36.48*** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net)
23:36.49Sephencheaper. =)
23:37.01jackxthat's one thing ;)
23:37.04Miccdoes asterisk work on linux smp kernel?
23:37.10harryvvjack, i just talked to a company that said thay had installed several asterisk boxes at the university of california to run its 19 thousand phones. I dont know if there is alot of truth to that though.
23:37.14jackxcompared to stability and scalability
23:37.27*** join/#asterisk niZon (~ilt@S0106deadbeefbeef.wp.shawcable.net)
23:37.34jackxhmmm
23:37.43jackxwhats several? :)
23:37.45Sephenjackx: I've heard similar things with hotel chains.
23:38.10drumkillaharryvv: what campus?
23:38.22Sephenjackx: We're in the process of converting our company to VoIP using *, but we're not there yet.
23:38.55xeet2micc: it works *best* on an smp kernel, with multiple cpu's of course
23:39.13harryvvjacm, I did some unrelated work at a city hall and thay were one of the first city halls in the region to install 1,900 ip phone in 9 sites and tied them together. It is a nortal network ip setup and as the phone administrator told me thay have nothing but problems. alwyas going down ect. She would have to stay after work and and wait for the tech to come in and configure everything or troubleshoot it via a serial port.
23:39.18Sephenxeet2: Then the zaptel SMP bug has been fixed?
23:39.25harryvvuniversity of califonia.
23:39.46xbmodder?
23:39.59xeet2sephen: which bug are you referring to?  I've got a few multicpu boxes with zaptel cards that work fine
23:40.45Sephenxeet2: Its more than a year old since I've tried playing with Asterisk on an SMP box, and that was with bad luck. Since then, I only have 2 boxes that are running Asterisk, and they are both singles, so I haven't had a change to revisit it yet and see.
23:40.46harryvvxee2 nice
23:41.00jackxwell.. sounds pretty good
23:41.16xeet2sephen: for smp stuff you should run head though =)
23:41.42Sephenxeet2: I typically do.
23:41.53Sephenxeet2: I'll give that a try next time I reconfig this box. =)
23:41.55harryvvI have been thinking about buying clone boxes for multiple asterisk or ser but want to know what motherboards have been the most dependable. I know there are a number of dell boxes that are not.
23:42.34xeet2harryvv: most of the issues with dell incompatabilites have come from the pci bus, or the fact that many of them only have 64 bit or higher pci slots
23:42.52xeet2the poweredge 750's work great
23:43.08harryvvhow many have you worked with?
23:43.37MiccWhat is the iax timing interface? I'm getting a warning about it when I start asterisk.
23:43.37xeet2i've only done the 750's and the sc1425's
23:43.41Sephenharryvv: We use SuperMicro stuff almost exclusively, and we have 30+ linux servers spinning away now on that hardware.
23:43.45Weezeywhat kind of motherboard should I be using for multi-processor?  I notice Asus' site lacks in the amount of dual and lack of quad boards
23:43.52*** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca)
23:44.25JerJerdell
23:44.30harryvvSephen, okay thats cool. I want to buy some more boxes but need to keep the price to a minimum.
23:44.34xeet2building your own box is a great way to go, but dell certainly has some great servers if you don't want to bother with any of that
23:44.56JerJersupport
23:45.16WeezeyDell is pretty much the same price as building it myself.
23:45.24Weezeyfor desktop systems anyway
23:45.43SephenI've never known a sysadmin who was worth his paycheck have to rely on a *hardware* vendor for support for pcs/servers.
23:45.44*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
23:46.11Weezeyheh, good point.
23:47.53xeet2sephen: you're wasting your money paying a good sysadmin to build a pc from scratch
23:48.43xeet2yes you want someone who *can*
23:48.50xeet2but that doesn't mean you should pay them to do it
23:48.57xeet2costs you more in the long run
23:50.46*** join/#asterisk kFuQ (~somedude@c-67-185-114-199.hsd1.wa.comcast.net)
23:51.49xeet2for the same price, as you say weezey, why not pay a company who's whole business is building pcs and servers and obviously do it well, rather than spending all your time researching compatability and then having to build it yourself
23:52.11*** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net)
23:52.26Sephenxeet2: Building the server is grunt work, which is why it gets handed down to grunts. I wouldn't use any linux load that someone like Dell did, for instance. I'd wipe and reload that box. Besides, the SuperMicro boxes don't take much to assemble - the boards are mounted in the chassis. Add CPUs, add memory, add HDs (already has CDROM/Floppy for the 1Us). Takes a whole 15 minutes to add those componets and start the load.
23:52.34Juxtgood evening
23:52.57Juxtis there some sort of an IAX load balancer?
23:53.40xeet2sephen:  oh yeah, I don't mean let dell install the OS =)  I mean just the hardware side of it
23:53.46xeet2sorry I didn't explain myself on that
23:54.24*** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net)
23:54.33milkyflavahello everyone!
23:55.24xeet2juxt: mmm, nope, but you could in theory use a bigip load balancer to do it
23:55.53Juxtxeet2: never heard of bigip
23:55.54xeet2iax works fine through nat, and is only one port, and the bigip's can balance udp
23:55.55Sephenxeet2: =) We have grunts for that kind of work.. I've not found another manufacturer who sold a 1U server with 4 SATA or 4 SCSI hot swap bays, 2 PCI slots, and dual processor except SuperMicro. Dell, HP, IBM, etc doesn't do it. Neither do they offer a 3U or 4U that handles 8 drives. To me, thats just freaking cool.
23:56.05Slupxeet2 harryvv i read the other day on the an * site that the don't recomend dells
23:56.19xeet2juxt: they're pretty big =)  go look them up.  but an average box will cost you about 5k
23:56.32Slupcan't remember the specific issue but under load they bailed
23:56.48Juxthmm i guess i'll have to play with SER instead
23:56.49*** join/#asterisk adker_ (~adker@67-136-212-8.dsl1.glv.ny.frontiernet.net)
23:57.02xeet2slup: alot of the hardware used in the dells require 2.6 to really work right
23:57.07xeet2might have something to do with it
23:57.40Juxti run asterisk on 2.6 with no issues
23:57.58xeet2sephen: mmm, I didn't know supermicro had something like that out
23:58.01jackx2.6.9. no problemens
23:58.05harryvvtalking to a dell sales rep right now.
23:58.22jackxknow where i can cheap 1u mounts?
23:58.23Juxtthe only issue i found with 2.6 is that my raid slowed down by about 20%
23:58.38xeet2juxt: what card?
23:58.59Slupdrumkilla what you was saying about * issuing a reinvite for external addresses where would i set that up or what would be the best thing to google for?
23:59.01Juxtit's some ATA raid card, not sure really
23:59.09Juxtraid1 config
23:59.13xeet2juxt: sata?
23:59.22Juxtno just ATA
23:59.23drumkillaSlup: it's automatic, unless you explicitly disable it
23:59.28xeet2hmmm
23:59.29Juxtactually IDE
23:59.37xeet2rocketraid?
23:59.37*** part/#asterisk morex (morex@host81-157-123-72.range81-157.btcentralplus.com)

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