irclog2html for #asterisk on 20050309

00:00.23*** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com)
00:00.23Hmmhesaysthen there's me who can't construct a sentence
00:00.24Hmmhesayslol
00:00.25MicH323Hi
00:00.32Nuggetyour a literate.  :)
00:00.57MicH323Me having connection problems with BroadVoice and Asterisk... It wont authenticate since yesterday...
00:01.12gr8nashMicH323 hey me to!
00:01.35gr8nashNo way!! i have been struggling for 4 hours now on this problem
00:01.48harryvvim begining to hate 64bit os. Some binaries are just not built for it.
00:01.49Smytheits at http://pastebin.ca/7045
00:01.54Hmmhesaysi've spent most of the day writing a gui
00:02.16slePPSmythe: enter the draw while you're on the pastebin :>
00:02.31MicH323Hmmm... I this BroadVoice changed their authentication again!!!
00:02.41gr8nashMicH323 could you get ahold of them.. they are all busy.. i sat on hold for 2 hours
00:02.45gr8nashuntil i hung up
00:02.48modulus_broadvoice sucks
00:02.48Smythepastebin is sweet - I've never used it before
00:02.54slePPpastebins are fun
00:02.59gr8nashmodulus_ who is better
00:03.03mandrekoi was looking at broadvoice... if they suck, who doesn't? ;)
00:03.17newpersyeah, i haven't heard you all mentino a good voip company
00:03.21newpersmention
00:03.29Groobywhere are you guys at?
00:03.29modulus_gr8nash, voip sucks in general
00:03.37Groobymy BV still work great
00:03.48newpersmodulus_, why is that?
00:04.01*** join/#asterisk cbachman (~chatzilla@129.105.7.250)
00:04.07MicH323BroadVoice was woring brilliantly up until yesterday! They have great unlimited international plans... My wife stays on to her sister in New Zealand for hours...
00:04.14MicH323Now she is nagging me... :(
00:04.26opus___why\
00:04.27GroobyMich, where are you guys located?
00:04.33GroobyI am here in DC and I have no problems at all
00:04.44gr8nashim in Washington west coast
00:04.44modulus_newpers, b/c the technology sucks
00:04.47MicH323I am in London
00:05.06mandrekowell, any new technology has it's quirkes
00:05.18harryvvgr8nash, in seattle?
00:05.21Groobywierd
00:05.25gr8nashjust south of seattle
00:05.28gr8nash2 hours
00:05.28modulus_voip is hardly new
00:05.30Groobybbiab
00:05.30MicH323Grooby: Are you connected to BroadVoice?
00:05.39mandrekowell, not new as in last month, but newer than analog lines ;)
00:05.41harryvvgr8 you mean olympia /
00:05.43harryvv?
00:06.03*** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net)
00:06.14gr8nashharryvv im in Vancouver WA.. 2 hours south of seattle
00:06.23opus___dude
00:06.26opus___thats like 3 1/2 hours
00:06.33gr8nashnope. =)
00:06.37gr8nashdrive it all the time
00:06.40opus___you drive like 80mph
00:06.41newpersmodulus_, so voip sucks?  or our broadband speeds?
00:06.50mandrekoi'm just wondering if i got asterisk setup, if i could have a remote machine (my work) vpn into my home and be connected with a softphone somehow...
00:06.52opus___<--- pdx .. hi gr8nash!
00:06.53harryvvthats not 2 hours south of seatte!
00:06.58MicH323No My BroadBand ics ass!!! 4Mbs
00:07.18gr8nashopus.. portland huh?
00:07.25harryvvwith traffic the way it is it takes 2 hours to go from seattle to tacoma.
00:07.25gr8nashharry where do you live?
00:07.31opus___yup
00:07.32harryvvim in the other vancouver
00:07.56gr8nashok in rush hour yes. your screwed.. but i dont drive during rush hour
00:08.20gr8nashDAMN
00:08.23gr8nashyour right 3 hours
00:08.24*** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
00:08.25gr8nashmy bad
00:08.27harryvvit still takes alot longer like 3 hours.
00:08.28harryvv;)
00:08.50gr8nashyes its 180 miles north of my house.. woodland wa
00:08.53opus___unless you drive like gr8nash, 120mph+
00:08.55CoaxDYou know, I ordered "3 Megabit Service" from Charter..  They showed up and said "Just so you know, they don't have 3 megabit service here."  --  apparently, cablemodem in Rural Wisconsin translates to "44kbyte/sec. Max."
00:09.05harryvvwe almost moved to battle ground from beverton but moved to tacoma instead from pooortland.
00:09.19gr8nashwow. cool lots of locals
00:09.19harryvvanyway
00:09.35gr8nashso noone has any suggestion of any VOIP provider
00:09.43opus___i use broadvoice
00:09.51opus___The $5.95 plan
00:09.54CoaxDApparently, they've translated "cablemodem service" to "blazingly fast local service, but we route your connection via a T3 that is shared between 250 people, all downloading files at top speed from kazaa"
00:09.54harryvvgr8 iax.cc voipjet
00:09.57gr8nashi cant connect to them.. even though i have an account
00:10.00tzangerI heartily reccomend nufone
00:10.03MicH323is your broadvoice still woring with Asterisk?
00:10.03mandrekowhat's the $5.95 plan? i didn't see that on their site
00:10.22opus___Its under BYOD
00:10.26tzangerjust look at the list archives to see the issues with iax.cc, livevoip and broadvoice
00:10.29Groobymich, i am connect to bV
00:10.29opus___bring your own drugs
00:10.30opus___i mean
00:10.33opus___device
00:10.33harryvvIf i had the cash i would put my own termination point in downtown vancouver
00:10.49mandrekoahh, very interesting
00:10.51Groobyman..i almost got spandsp to work over ulaw
00:10.52Groobyhehehehe
00:10.53tzangerharryvv: yeah?
00:10.58tzangeryou terminate a lot to vancouver?
00:11.00CoaxDhaha. 1.43c/min if you prepay $200 or more
00:11.00mandrekowith the $5.95 plan i could do this ;)
00:11.07CoaxD(www.iax.cc)
00:11.47mandrekoBYOD means you have asterisk, and don't need a SIP/softphone/adapter ?
00:11.47*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
00:11.57Groobymeans bring your own device
00:12.04Groobybe it ATA, asterisk, soft phone
00:12.08*** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
00:12.41mandrekoi apologize, i'm new to this stuff, still in a research stage
00:13.17gr8nashwell i will probley look at it with semi-fresh eyes tomorrow
00:13.33Grooby?!?!
00:13.39gr8nash??
00:13.46Nugget]:8)  Cows are cool
00:14.00Groobyso are monkeys
00:14.06Groobythey roar
00:14.08Grooby:-D
00:14.12gr8nashcommon Grooby do a ascii monkey!
00:14.12modulus_jbot dogcow?
00:14.13jbotMOOOFF!!
00:14.20NuggetGrooby is uncommon.
00:14.37Groobygrooby's are very common
00:14.40mandrekois there any way to hook up a remote phone up, to have my number follow me when i'm at work?
00:14.41Groobyjust checkout grooby.com
00:14.43Groobyif it still exist
00:14.50harryvvtzanger thats a wish list right now. But know people that could help me make it happen. Still worry about the up and comming cable carriers and other larger companies that are now rolling out voip.
00:14.58Nuggetmandreko: yes, of course.
00:15.22tzangeryeah
00:15.27mandrekoNugget: how would i do something like that?  I don't need the specifics, but what's the general idea?  I can RTFM if i know what i'm looking at
00:16.00Nuggetmandreko: I don't even know where to begin, your question is so broad.  first step would be getting asterisk handling your number when you're not at work -- then go from there.  :)
00:16.13Nuggetbut the bottom line is, yes, that's possible.
00:16.21Groobymandreko, you understand the concept of having a central PBX box right?
00:17.05mandrekohaving the menu system, possibility of multiple lines, voice mail handling, right?
00:17.13Nuggetasterisk does all those things, yes.
00:17.33elricI am getting very low volume on out going calls to cell phones, is there a way to increase call volume?
00:17.41Groobyelric, vol+?
00:17.44Grooby:P
00:17.51Nuggetasterisk can enable you to make your exposure to the phone system as ridiculously complicated as you wish.  the limit is only your imagination.  :)
00:18.14Groobyand voip replaces hardwire lines connecting your phone to your central PBX box
00:18.22mandrekohehe, yea, i know my work's pbx is quite complex, and we've been talking about replacing it with a voip pbx like asterisk, however i wanted to make it work at my house first
00:18.24Groobyso you can have a softphone/hardphone that you carry w/ you
00:18.47Groobyand as long as you have internet, you can be reached (voice quality may vary)
00:18.52Nuggetmandreko: that's a good approach.  asterisk has some downsides and it's an excellent idea to gain familiarity at home where jobs are on the line.
00:19.00bandrewWhy would you want a two port VOIP phone?  So you can pass signal through it to a computer or something?
00:19.04Groobyback to help my gf
00:19.14Nuggetbandrew: right.  for desks where there's only one ethernet drop.
00:19.29CoaxDbandrew: Two ethernet ports == only 1 ethernet cable run needed to a cubicle or office
00:19.40CoaxDbandrew: (and no switch or hub needed in the office)
00:19.50bandrewthanks
00:19.51mandrekoyea, i'm a geek, and use linux quite a bit (I work for a small ISP), so i figure I can learn.  I just need to try it I gues
00:20.07Nuggetmandreko: I documented my experimenting at http://slacker.com/~nugget/asterisk.php
00:20.15NuggetI did pretty much exactly what you describe
00:20.17CoaxDmandreko: You too, 'eh?
00:20.21CoaxDMandreko: (I own one.)
00:20.29mandrekoNugget: very cool ;)  I"ll check it out!
00:20.34elricGrooby, :P haha well its fine for any thing else just cellular phones, like Zap 3 gets an incoming call and calls out if the person's cellular phone if he isnt in. the volume on that is very low but if its a land line thats called out it works fine
00:20.43NuggetI don't work at an isp any more though, but I assure you my experience is still valid!  :)
00:20.44mandrekoCoaxD: I'm the only non-owner at mine ;) it's a small family owned one that hasn't been sucked up yet by a big guy
00:20.48CoaxDmandreko: A lot of ISPs are moving into voip services..  (Which is funny, because most of us ISP folks moved into ISP stuff from BBS stuff)
00:21.06*** join/#asterisk jmhunter (~jmhunter@64.77.199.223)
00:21.06*** mode/#asterisk [+o jmhunter] by ChanServ
00:21.12CoaxDmandreko: Heh :) Yeah, we're small, and family owned
00:21.20jmhuntersup coax
00:21.22mandrekoyep, at work, we're talking about doing VOIP services as well, so I'm very interested in learning more ;)
00:21.25CoaxDmandreko: (By small, I mean 'less than 1000 users')
00:21.34bandrewAnyone know of a cheap switch that supports QoS?  Also, does the phone you buy need to support QoS too?
00:21.34CoaxDjmhunter: Hey dude!  Not much. You?
00:21.42mandrekoCoaxD: hehe, we're a _little_ bigger than 1000, but not TOO much
00:21.49jmhunterim trying to get in touch with the crew at von
00:21.52CoaxDmandreko: Yeah, just enough to require a fulltime sysadmin :)
00:22.13mandrekoCoaxD: yeppers.  All our servers will fit in 2 racks at our datacenter, so we're not huge
00:22.30CoaxDmandreko: Yeah, same on this end. *lol*
00:22.33SmytheKaID?
00:22.52CoaxDmandreko: I do have a bunch of tower boxes we have on a table, tho. So i guess its a wee bit bigger, hardware-wise! *lol*
00:22.55Smythethe mitel side shows locked out when I try and connect form the ast box
00:23.15Nuggethttp://slacker.com/photos/misc/pophell  <-- tiny ISP  :)
00:23.17mandrekoCoaxD: yea, the low priority servers in our local office don't count either ;)
00:23.22CoaxDmandreko: I've run the damn thing since 1997.  Getting sick of ISP business. Doesn't pay as much as it used to. *g*
00:23.30CoaxDmandreko: Hehe
00:23.35CoaxDmandreko: Yeah, i wasn't counting those. *g*
00:23.35mandrekoNugget: that's horrible..
00:23.38Nuggetindeed
00:23.52mandrekoCoaxD: yea, we've been going since 1996, still having fun though
00:24.04CoaxDmandreko: Heh :)
00:24.04mandrekoNugget: I'd get shot if my racks looked like that
00:24.13CoaxDoh jesus
00:24.14CoaxD(the small ISP picture)
00:24.20*** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net)
00:24.29Nuggetmakes you realize why it's important to do things properly, eh?
00:24.34CoaxDand I thought *mine* was bad
00:24.49CoaxDsportsters hanging everywhere! *lol*
00:24.52NuggetI love the modems just hanging in the air, and how it appears to all be plugged into that single outlet.
00:25.27mandrekomine's not _much_ better, we just moved datacenters, and our wiring's not _great_ http://www.ori.net/mandreko/servermove/
00:25.36*** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com)
00:25.43CoaxDNugget: *lol* Indeed!
00:25.47CoaxDmandreko: Its the same on this end
00:25.58CoaxDmandreko: (We just moved too)
00:26.00mandrekowe at least have zip ties ;)
00:26.06mandrekozip ties make a rack... hehe
00:26.26CoaxDOOH! I see Dell PowerEdge 350's!
00:26.30*** join/#asterisk FaulHel (~a@24.153.115.117)
00:26.36CoaxD(or somesuch)
00:26.38mandrekoshould be 2650's
00:26.48harryvvhahah that rack looks like what the hp blade commerical is going after :)
00:26.48CoaxDhehe yea
00:27.04CoaxDyou've actually got cabinets
00:27.11CoaxDi've got 2 post 19" racks with a gazillion things hanging on them
00:27.17modulus_woohoo! i have 14 dids
00:27.18mandrekoyea, we are in a huge datacenter... same one that does monster.com's alternate site
00:27.28CoaxDmodulus: Um, good for you? :)
00:27.33modulus_no treally
00:27.37FaulHelcan anyone help me install asterisk on a machine that cannot boot from a cd...or give me some tips
00:27.42modulus_that's actually kind of lame
00:27.45modulus_voip sucks
00:27.49CoaxDmandreko: ours isn't in a data center. Its in the basement of a small mall
00:27.56CoaxDmodulus: *rotfl*
00:28.04CoaxDmodulus: At least you have a sense of humor.. :)
00:28.06shepherdfaul: debian minimal install
00:28.06harryvvMost impressive rack setup was when I interviewed for amazon.com data center. Very clean and setup well.
00:28.07mandrekoCoaxD: that has it's advantages and disadvantages ;)
00:28.07shepherd:)
00:28.09modulus_btw wtf are all those modules loaded for by default?
00:28.12CoaxDmandreko: INDEED!
00:28.13modulus_jesus christ
00:28.29CoaxDmandreko: (keeping in mind of course that the nearest datacenter is about 80 miles away from here.  And thats small class.)
00:28.33modulus_app_festival is default loaded
00:28.33modulus_wtf
00:28.36CoaxDmandreko: *g*
00:28.45CoaxDmandreko: (My first T1 cost $2200 a month.  And that was just loop.)
00:28.49modulus_chan_modem_i4l
00:28.50modulus_wtf
00:28.53mandrekoCoaxD: that sucks, we have 2 near me in Indianapolis...
00:29.03mandrekoCoaxD: 2200 for a LOOP!?!?
00:29.10tzangerwtf
00:29.17tzangeralison's not doing TV news now is she?
00:29.33CoaxDmandreko: (And I had to get down on my knees, bend over, and give the manager of USWest (now Qwest) really nice sweet head in order to get it, too)
00:29.34modulus_hi tz
00:29.46mandrekoCoaxD: sounds like us with SBC ;)
00:29.50CoaxDmandreko: Yeah, $2200 for loop. and that was '98
00:30.10mandrekoCoaxD: i don't remember our prices from then, i just know our OC3 is pricey now.  we pay a premium for good bandwidth
00:30.21CoaxDmandreko: Why the hell do you need an OC3?
00:30.25CoaxDmandreko: Are you a DSL ISP?
00:30.32modulus_dick sucking lips isp?
00:30.39Nuggetmandreko: you're in indy?  cool.  I do some work now and again for iquest.
00:30.42mandrekoCoaxD: yes, 3 dsl atms, several dialup pools, 500+ websites
00:30.46modulus_ISDN: it still does nothing
00:30.48CoaxDmandreko: ahh
00:30.55CoaxDmandreko: You're *way* bigger than us.
00:30.55mandrekoNugget: woot ;) I used to use iquest
00:31.05Nugget<-- grew up in broad ripple
00:31.05denonmandreko: I hope you didnt have anything to do with that flash on http://www.ori.net/mandreko
00:31.13CoaxDdenon: Get me a T1 for $500/mo.
00:31.22mandrekoNugget: I do the Westfield/Noblesville thing up north
00:31.25Nuggetcool
00:31.34NuggetI haven't lived there since the early '90s
00:31.34r0d3nt|mAnyone have the SiP firmware for the 7912's ??
00:31.36denonmandreko: 'cause uh .. I want those 2 minutes of my life back please :)
00:31.41mandrekodenon: no, that's actually something i found and mirrored for myself ;)
00:31.55denonno, really .. I want those 2 minutes back
00:31.57CoaxDdenon: Data. To a nice backbone. Yes. you heard me.  $500/mo.  With bandwidth charges.
00:31.58mandrekohehe
00:32.16CoaxDdenon: Make it happen, sir, and I'll send you a pizza.
00:32.19denonCoaxD: could do it for much less than that .. if you lived in a metro
00:32.27mandrekoCoaxD: we're doing T1's for ~600 anymore as long as the loop's not atrocious
00:32.28CoaxDdenon: Hehe
00:32.40harryvvfor those wanting have more choices in the iphones on the market came across this site. http://www.sipcenter.com/sip.nsf/html/SIP+Phones+and+Adaptors
00:32.42CoaxDhell, I'd even consider $600
00:32.50denonmandreko: dont even bother .. his last mile loop includes twigs and leaves
00:32.51ScythelXhow much does a t1 long distance pri cost
00:32.58r0d3nt|m$485 with loop is the best I can do a T1,data. anywhere in the USA...
00:33.00CoaxDdenon: Get me a T1 for $500 or $600.  You know you wanna do it
00:33.11modulus_how far from the co are you?
00:33.19denonr0d3nt|m: highly doubtful .. anywhere
00:33.36r0d3nt|mdenon, I haven't qualified an address I couldn't install yet...
00:33.49denonreal t1? or atm over dsl?
00:33.54r0d3nt|mTeee ONE
00:34.00r0d3nt|m1.54mb 24 channels
00:34.03r0d3nt|mthe real deal.
00:34.17mandrekois that only for the channelized ones then, or you doing DS1's for the same?
00:34.22r0d3nt|m23+D or 24 how ever you want it
00:34.26harryvvr0d3nt|m,  any idea who I should call up here in canada? not to many choices other then telus and i think level3 but I think level3 left the area after bankrupty.
00:34.33CoaxDI guess denon dont wanna sell me a T1 :)
00:34.41harryvvLooking for pri
00:34.47denonCoaxD: I could quote you again
00:35.01CoaxDdenon: You gotta get the $$ down to where I can justify the cost, man! :)
00:35.02r0d3nt|mI need Cisco 7912 SIP Firmware... Can anyone help ???
00:35.07CoaxDdenon: err, the changeover!
00:35.25denonmsg me the info again
00:35.27CoaxD(and the damn network renumber!)
00:36.01mandrekoI wonder what it'd cost for me to get my own system to become like a broadvoice..
00:36.09CoaxDmandreko: Done to death, man
00:36.15CoaxDmandreko: You will not compete
00:36.25CoaxDmandreko: You can, however, compete in local markets
00:36.31mandrekoCoaxD: who needs to compete when it is just another geek toy? ;)
00:36.37CoaxDmandreko: Hmm. alright :)
00:36.39harryvvman you need pri  pstn and data
00:36.39denonthat what your gf said?
00:36.57FaulHelT1's don't use 24 channels do they...thought they were 4 channels and they used 64k on each one to produce the broadcast. ISDN PRI is 24 channels (23+D)
00:36.59harryvvCoaxD,  you mean in a local area.
00:37.06mandrekoCoaxD: yea, i know it's been done to death, however you're right, we were considering it for local use to compete with SBC phone service
00:37.13r0d3nt|myes.. a PRI is 23+D
00:37.16CoaxDfaulhel: Um, T1s do use 24 channels, yes
00:37.20r0d3nt|mchannelized T1 is 24, 56k channels
00:37.25CoaxDharryvv: Yes
00:37.30harryvvFaulHel,  64 is data+overhead.
00:37.31ScythelX24 ds0's
00:37.33FaulHelah
00:37.34JunK-Cnot 56, 64!
00:37.37CoaxDmandreko: You bet.  Local phone service, you can certainly compete
00:37.45CoaxDmandreko: Especially if you're in a high-cablemodem/low-dsl area
00:38.00DaminT1 is 24 64k channels. PRI is 24 64k channels, but one is used for a D channel for signalling (call setup and whatnot)
00:38.06harryvvCoaxD,  but as a long distance voip carrier its way to compedative to make a profit right?
00:38.08r0d3nt|mI'll argue all you want about the symantecs of telco circuits.. but I need the SIP Firmware for the Cisco 7912... can anyone help ???
00:38.17mandrekoCoaxD: yea, definately here.  we're near a lot of places that just can't get DSL, because SBC's not putting out DSLAMS
00:38.18CoaxDChannelized T1 is 24 56k channels, with the last 8k of each channel bit-robbed for signalling
00:38.28CoaxDharryvv: Correct.  There's no way to make money there right now
00:38.38Daminr0d3nt|m: A T1 is only 56k if you are doing robbed bot signaling.
00:38.42DaminBit raterh.
00:38.45CoaxDmandreko: OH, sure. Yeah.  so everybody gets cablemodems
00:38.53CoaxDmandreko: Put the kabosh on their phone lines and sell 'em good voip service
00:38.59harryvvCoaxD,  do you know of any pri carriers for in the vancouver area?
00:39.07tzangerDamin: yeah CAS T1 is 24 56k channels
00:39.10tzangerthey're still 64k though
00:39.10mandrekoCoaxD: those that can, sadly lots of the people out in the boonies don't have cable either, but we're working on it, and fixed p2p wireless
00:39.15tzangerthey're just not 8-bit "clean"
00:39.16CoaxDharryvv: Given that I don't actually live in the vancouver area, that'd be pretty tough
00:39.27harryvvI wonder if shaw will be comming out with cable + pstn modems in the future.
00:39.28harryvvyea
00:39.37CoaxDmandreko: hmm
00:39.39DaminESF B8ZS Babay! :)
00:39.47CoaxDmandreko: If you can do p2p wireless, you can do voip to them too
00:39.56CoaxDtzanger: Correct
00:39.57mandrekoCoaxD: yep.  that'd be fun ;)
00:40.04TrepaliumWouldn't be surprised with all the phone companies intruding into their market.
00:40.14CoaxDmandreko: We've researched doing wireless here, but the prob is, to blanket everything, you'd be talking thousands for an income of zilch
00:40.17harryvvOhh this is slick. a Biometric fingerprint IP box mounted on a wall by ip-ware. http://www.ip-ware.net/www02/index2.html
00:40.17mandrekoCoaxD: i swear i just like it for the geek factor.
00:40.26CoaxDmandreko: Plus, bandwidth is still way too expensive
00:40.32mandrekoCoaxD: yea, I'm afraid of that here too, but we're working on it
00:40.41CoaxDmandreko: (Unless of course denon here can get me a T3 w/ 45mbit bandwidth for $1000/mo)
00:41.00mandrekoCoaxD: if he can get you one for 1000/month, i'll have to talk to him ;)
00:41.07CoaxDmandreko: Hehe. for sure
00:41.18CoaxDmandreko: Is your OC3 delivered full-bore?
00:41.28mandrekoCoaxD: yep ;)
00:41.33mandrekoCoaxD: and more's available
00:41.34CoaxDmandreko: i.e. full 192mbit/sec? or whatever it is?  (45*3 i guess)
00:41.49mandrekoCoaxD: we don't come close to using it, but yea, we can
00:42.04CoaxDmandreko: Nifty
00:42.14mandrekoCoaxD: we average about 8Mbps constant
00:42.21CoaxDmandreko: I wonder.. Do you know how much you pay for that OC/3?
00:42.34CoaxDmandreko: Hmm.  Well, why do you need an OC/3 if you only use 8mbit/sec?
00:42.41mandrekoCoaxD: i dont' do much with the finances, but i know we pay SBC about $20,000/month for everything.
00:42.47CoaxDmandreko: (or is that just what the datacenter dropped for you)
00:42.53CoaxDmandreko: Ah
00:42.55CoaxDmandreko: Not bad
00:42.57mandrekoCoaxD: that's what the datacenter has
00:42.58puppetHmm anyone free for a simple question?
00:43.05CoaxDmandreko: Ahhh. okay. see, thats what I was wondering
00:43.23mandrekoCoaxD: our equpiment wouldn't handle OC3 traffic at this point, we'd have to upgrade our core
00:43.55puppetI wanne do something like IF ${CALLERIDNUM} = "12345" then Goto(voicebox,1)
00:44.12JunK-Cpuppet: go in the wikis, its already there
00:44.17JunK-Clot of infos related to that.
00:44.34JonR800puppet, no need to use an if.
00:44.39puppetthanks :)
00:44.44puppetjonr800: was psuedo code :)
00:44.55puppetIm having trouble with my cellphone
00:45.02mandrekomethinks I"ll have to setup an asterisk box here, and play until I'm a little more comfortable with it
00:45.22puppetIts some strange stuff, when i decline a call I get a new call from my own VoIP number not from the cellhpone ;p SO i get a new session kinda
00:46.08harryvvmandreko,  are you a hosting company
00:46.33*** join/#asterisk mesi (~player@dsl-082-083-155-079.arcor-ip.net)
00:46.45mandrekoharryvv: yea, we're full service
00:46.53*** part/#asterisk Smythe (~Smythe@spock.cbcag.edu)
00:46.53harryvvI see
00:47.20mandrekohosting, dialup, dsl, computer repair, onsite support.. you pay us, we do it ;)
00:48.00mandrekosadly, we're sorta like technology whores
00:49.23mandrekoheya Nugget, where did you learn most of what you're doing?  I'm just reading through your site
00:52.01mandrekohmmm, I wonder how hard it'll be to setup extensions for me and my roomie, so I don't have to always answer the phone...
00:52.06*** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
00:57.28shmaltzanybody here using billing with asterisk?
01:00.07eKo1What billing?
01:01.14*** join/#asterisk pulu (~chatzilla@65.77.78.3)
01:01.48*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
01:06.27NuggetI bill for my time spent in here, but nobody pays.
01:06.46Nuggetbunch of cheapskates, the lot of them
01:07.37moonwickyou're just a slacker about collecting.
01:07.57Nuggetthat must be it.
01:08.28harryvvnugget just go sell your services to the biz side of things.
01:09.32*** join/#asterisk coldfeet (~c@213.78.240.109)
01:10.42puppetexten => s,6,GotoIf($[${CALLERIDNUM} = "31337"]?1|3:7) the expression turns out to 0 even when calledidnum = 31337
01:10.53CoaxDYou know, i REALLY hate it when i have to pause a TV show so I can go and lookup a word they use
01:11.08CoaxDThey really shouldn't use words like 'dearth' (you know, words NOBODY EVER USES) in a tv drama
01:11.48coldfeethi all, got a question which i cant find in the mailing lists....
01:12.03coldfeetI have two accounts both for xlite set in sip.conf
01:12.25coldfeetone registers fine, the other one a different network fails
01:13.17coldfeetxlite debug shows 401 then trying then 402 forbidden, and asterisk shows ...
01:13.29*** part/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
01:13.39coldfeetRegistration from 'home <sip:home@sip.dom.com>' failed for 'a.b.c.d'
01:13.40*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
01:14.13coldfeeti checked the user/secret combo and there is no allow/disallow in case it weas a codec mismatch
01:14.33coldfeetI even tried the account that was working in the xlite that wasnt but still same error
01:15.28coldfeetis there any place i can check exactly what is being sent and what is being compared to
01:18.13DyOShas anyone experienced a problem where you dial the directory on asterisk then it asks you to spell the last name and after you put in teh 3 letters it just beeps like an invalid beep tone?
01:19.27Groobyyawn
01:19.30Groobyback
01:21.47coldfeetif this is a answer on google, let me know cause i aint found it all day
01:22.18*** join/#asterisk kks (~kks@203.115.210.253)
01:22.20Groobyin CLI
01:22.31Groobydo sip debug peer <exten number>
01:22.36Groobyi.e. sip debug peer 200
01:23.16*** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
01:23.17rvhiis there a way to change the soft key on polycom phones?
01:23.30rvhie.g. some keys need to press 'more' to get
01:23.37MikeJ[Jayden]coldfeet: pastebin a sip.cong
01:23.39MikeJ[Jayden]conf
01:23.43MikeJ[Jayden]~pastebin
01:23.44jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca
01:23.49rvhii'd like to move some in the first screen
01:23.55r0d3nt|mAnyone have the Cisco SiP firmware for the 7912's ??
01:24.26coldfeetit cant get the IP address ,hence the debug dont show, it doesnt register so in sip show peers host is unspecified
01:24.39Groobysip show debug peer IP?
01:24.40MikeJ[Jayden]rvhi, there is a VERY detailed admin manual at polycoms site... if moc is around, he knows the polycoms well
01:24.44Groobyi think you can debug ip too
01:24.52justinnnnhey ppls
01:24.58justinnnni installed the new mp3 thing..
01:25.04justinnnnhow do i spot the old mpg123 from running ???
01:25.17coldfeetbut the IP is unspecified, cause it aint registered , and its dynamic
01:25.18rvhii looked at admin guide of polycom, can figure out
01:25.21Goshenpeople on Broad Voice....how are you routing your 911? keeping a landline?
01:25.41*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
01:25.42MikeJ[Jayden]coldfeet, if you want me to help, you need to respond to what I said above... it sounds like a routing\nat\or firewall issue at first glance,
01:25.43*** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
01:25.48MikeJ[Jayden]Manx!
01:25.56firestrmhelp
01:26.12MikeJ[Jayden]firesrm, NO! :)
01:26.31Groobynothing a "smooth jazz" can't fix
01:26.43Groobyespecially as background music on some good old pron
01:26.58Grooby:)
01:27.05*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
01:27.21ManxPower~docs
01:27.22jbotDocumentation 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
01:27.37coldfeetguys my two contexts are in pastebin
01:27.48ManxPowerfirestrm, Do the drivers load?
01:27.55Groobywhat's the url?
01:27.57coldfeetcoldfeet sip is the title
01:28.09coldfeetpastebin.ca/7050
01:28.12firestrmanyone have a working example of zapata.conf with tdm400 2 fxs 2fxo? i cant get the dang fzs ports to go to ANY context when i dial an extension..
01:28.14ManxPowerfirestrm, remember that if you plug a phone line into the FXS modules you will blow up the module the first time the line rings
01:29.49MikeJ[Jayden]coldfeet, are you logging in these phones from diff subnets?
01:29.52firestrmManxPower, no i havent done that.. but good advice to remember.. :\
01:29.59coldfeetyup
01:30.06coldfeetdiff networks all together
01:30.22MikeJ[Jayden]ok, so if you use softphone or whatever to log into the account that "isn't
01:30.33MikeJ[Jayden]" working, with the phone that is, that works rihgt?
01:30.34Brixiushello
01:30.48*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
01:31.14firestrmManxPower, Ive just installed the samples, and it answers the fxo no problem, does the demo.. im trying to get it to allow me to dial an extension from the fxs port, that simply executes goto(demo,s,1) but no joy.. fast busy on first digit
01:31.28TrepaliumI can't seem to send audio using X-Lite and the Speex codec, and sending via iLBC sounds very, very bad.  Is this normal?  The other three codecs seem fine.
01:31.28coldfeetthe softphone works cause i tried it earlier in the day with SER account
01:32.09*** join/#asterisk liquide (~havard@liquide.user)
01:32.34MikeJ[Jayden]coldfeet, so you are saying both accounts authenticate fine, on the local network, correct?
01:33.09firestrmmanx ive added a context = internal and channel -> 1,2 to the zapata.conf..
01:33.22rvhianyone has a working example of multiple level auto attendent?
01:33.34rvhipress 1 goto this
01:33.36firestrmmanx, and the corrosponing handler in the extensions.. but no joy
01:33.46rvhithen has another set of menu
01:33.52coldfeetMikeJ, confused as to what u mean by local, my asterisk is on public IP, xlite1 is one one laptop , and xlite2 on another both of different wan IP (behind a router)
01:34.14coldfeetcan I test the registeration from the cli
01:34.14rvhii wonder how you handle same number in different levels
01:34.38coldfeetalso in SIP in general is there a way of telnetting to the port and testing basic stuff like register
01:34.50cypromisrvhi: you sent the call to a different contet
01:34.54cypromiscontext that is
01:35.49coldfeetalso i can use the xlite to register to SER no probs,
01:36.09MikeJ[Jayden]coldfeet, are both clients on diff subnets?
01:36.10rvhioh, i c, that makes sense now. thx
01:36.25MikeJ[Jayden]and is there any NAT or firewalls involved?
01:36.27firestrmany have a zapata sample with a 2x2 tdm400?
01:36.33*** join/#asterisk mesi (~player@dsl-082-083-129-089.arcor-ip.net)
01:36.40coldfeetMike, yup, they are on private IP's behind different routers which connect to different ISP's
01:36.51coldfeetno firewalls, NAT yes
01:37.07rvhiany php package to create extensions.conf online?
01:37.36coldfeetthe register command is being sent to asterisk i checked and reply also comiing back, and REGISTER is on 5060/5090 and that is all open
01:38.04MikeJ[Jayden]with sip debug, can you see the register come in from both?
01:39.23coldfeetyup
01:40.03ManxPowerfirestrm, Paste your /etc/asterisk/zapata.conf to pastebin.ca
01:40.37coldfeetcaroline registers instantly as soon as I startup, is there any cache or nething in asterisk
01:40.43ManxPowerOh kram where are you!
01:40.55MikeJ[Jayden]von ;)
01:41.15ManxPoweruncontrollably, too.
01:41.34MikeJ[Jayden]ok, pastebin the sip debug of the broken one registering
01:41.54MikeJ[Jayden]it will be ok
01:42.08MikeJ[Jayden]I found the best use for windows xp ever....
01:42.19TrepaliumFrisbee?
01:42.38MikeJ[Jayden]Spider solitaire while helping users on #asterisk.... it's multitasking baby
01:42.57tuxinator_linuxhmm,
01:43.03MikeJ[Jayden]I love me some spider solitaire, and it's free......with the purchase of xp that is
01:43.09tuxinator_linuxneed to  buy a new gay MikeJ[Jayden]
01:43.15tuxinator_linuxoops
01:43.17tuxinator_linuxgame
01:43.26MikeJ[Jayden]but xp is free too... with the purchase of a pc that is...
01:43.29MikeJ[Jayden]but what is free
01:43.54MikeJ[Jayden]txinatore_linux aparantly needs to buy a new gay MikeJ[Jayden]
01:44.08MikeJ[Jayden]does that mean that I am to strait for you?
01:44.10MikeJ[Jayden]hehe
01:44.24tuxinator_linuxI prefer guys to be strait
01:44.39justinnnnanyone wana help me with txfax :) ?
01:44.47justinnnnive tried soooooo many versions of libtif/spand/asterisk
01:44.48tuxinator_linuxThat is a common question on here
01:44.53justinnnnit always just segfaults when it sends a fax
01:45.00tuxinator_linux~txfax
01:45.03coldfeetI am also confused as to why I would get a 401..then trying, and then a 403 forbidden
01:45.14Silik0nif its so free why are peecees w/out XP cheaper?
01:45.47tuxinator_linux~what is txfax
01:45.49jbottuxinator_linux: what are you talking about?
01:46.18MikeJ[Jayden]justinnn... toss asterisk@home on a test box.. it has it all in there for you, and you can get a feel for how it works
01:46.30tuxinator_linuxI don't have any experence with txfax, sorry justinnnn
01:46.41MikeJ[Jayden]~spandsp
01:46.45tuxinator_linux~linux@home
01:46.51firestrmManxPower, http://pastebin.ca/7051
01:46.53tuxinator_linux~asterisk@home
01:47.10MikeJ[Jayden]tux, check out the wiki
01:47.25tuxinator_linuxI already dowloaded it
01:47.44MikeJ[Jayden]coldfeet, you get a 401, because you refuse the willing help of those that can fix it...
01:47.45firestrmManxPower, its about as minimalistic and simple as one can get.. maybe too simplified.. i probbly forgot somthing..
01:47.46*** join/#asterisk syamajala (~syamajala@h0030bd1e9b96.ne.client2.attbi.com)
01:47.54syamajalaKalD|Work, ok
01:47.59ManxPowerfirestrm, What modules do you have on the card?
01:48.11firestrm1,2 fxs 3,4 fxo
01:48.17ManxPowerfirestrm, post your /etc/zapata.conf.
01:48.24ManxPowerfirestrm, Are any of the module working?
01:48.26MikeJ[Jayden]~pastebin
01:48.27jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
01:48.29tuxinator_linuxfirestrm: Not on here
01:48.33firestrmManxPower, http://pastebin.ca/7051
01:48.40tuxinator_linuxbueno
01:48.51coldfeetpaste bin the debug http://pastebin.ca/7052
01:48.57syamajalai just found the asterisk site a few hours ago and i am really interested
01:49.10firestrmManxPower, the fxo works fine.. fxs gives fast busy when dialing extension
01:49.16tuxinator_linuxsyamajala: Its nice
01:49.26syamajalai don't know anything about phone systems
01:49.29syamajalabut my mom does
01:49.39tuxinator_linuxsyamajala: go grap asterisk@home to try it out
01:49.44syamajalamy parents own a doctor's office and they wanna replace their phone system
01:49.46coldfeetMike posted it...hoping not to get anymore 401
01:49.46ManxPowerfirestrm, what is the output of ztcfg -vvv?  did you remember to STOP NOW and then start Asterisk after making zap changes?
01:49.50*** join/#asterisk vipor (vipor@ip68-111-216-186.sd.sd.cox.net)
01:49.53viporHello..
01:49.59vipor-/?
01:50.02ManxPowersyamajala, I am currently replacing the phone system at a doctor's office.
01:50.03tuxinator_linuxasterisk@home is found at http://asteriskathome.sourceforge.net/
01:50.07firestrmManxPower, no i only did reload... didnt know i had to stop
01:50.15syamajalao
01:50.16ManxPowerfirestrm, that's your problem.
01:50.19firestrm:)
01:50.22firestrmthanks..
01:50.24ManxPowerfirestrm, for zap changes you have to stop and start asterisk
01:50.39firestrmi'll try that..
01:50.40syamajalai don't know much about phone systems
01:50.52tuxinator_linuxsyamajala: Pretty quick to pick up
01:51.00tuxinator_linuxsyamajala: I didn't know much either
01:51.01modulus_voip sucks
01:51.07syamajalao
01:51.11tuxinator_linuxmodulus_: oooookay
01:51.18syamajalatuxinator_linux, so what is the basic stuff i need
01:51.27syamajalai mean in terms of hardware besides a system
01:51.38viporI was wondering if asterisk can be setup to replace my stander answering machine, via generic pci modem??
01:51.39tuxinator_linuxsyamajala: ~rtfx
01:51.39modulus_mariah carey is hot
01:51.44tuxinator_linux~rtfw
01:51.45jbot[rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
01:51.56syamajalathanks
01:52.06tuxinator_linuxvipop: expensive answering machine
01:52.56CoaxD'
01:52.59CoaxDerp
01:53.00ManxPowervipor, no.
01:53.22ManxPowervipor, some IA-92 PCI Winmodems are compatable with the Zaptel drivers.
01:53.23*** join/#asterisk Skysky (~Miranda@Toronto-HSE-ppp3731995.sympatico.ca)
01:53.35ManxPowerIf you want to use any other modem then you'll have to write your own driver for it.
01:54.49viporo
01:54.51viporic
01:55.35viporis there a list of modem's that are compatable w/Zaptel drivers?
01:55.46nestArMD3200 chipset
01:56.10TrepaliumSearch ebay for "x100p clone" cards.
01:56.40viporok
01:56.43viporthanks
01:56.55nestArlol
01:56.55eipii need iaxy install docs... who have it?
01:57.07nestArwww.digium.com i bet..
01:57.19coldfeetMike I have also just pasted pastebin.ca/7053 my sip startup, just the part from the sip channel
01:57.21coldfeetwhat does
01:57.22coldfeetUnable to find key 'home' in family 'SIP/Registry'
01:57.25coldfeetmean
02:01.25*** join/#asterisk Eight (~blake@12-205-155-39.client.mchsi.com)
02:03.01ManxPowerhttp://bugs.digium.com/bug_view_page.php?bug_id=0003743
02:03.08DyOShas anyone experienced a problem where you dial the directory on asterisk then it asks you to spell the last name and after you put in teh 3 letters it just beeps like an invalid beep tone?
02:04.14goatmilkVolcano Mount St. Helens in the United States apparently erupts. A plume of smoke is seen as far away as Portland, Oregon
02:05.48JuggieManxPower, have you ever used 5ess switchtype?
02:06.02*** part/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
02:11.43coldfeetwebcam for volcano http://www.fs.fed.us/gpnf/volcanocams/msh/
02:14.41*** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com)
02:14.50MikeJ[Jayden]sorry... my inlaws called
02:15.39AgiNamuMar  8 21:15:27 WARNING[1800]: app.c:599 ast_play_and_record: No audio available on IAX2/TLF11004@TLF11004/16??
02:15.45AgiNamuWTF can do I do about this?
02:15.49AgiNamuNo audio available crap?
02:15.54AgiNamuI tried using G729 and GSM
02:15.56AgiNamusame error
02:16.00AgiNamurecording to wav
02:16.08coldfeetMike...and I thought i had problems :-)
02:16.36coldfeetnehow if u missed the paste bin it all at pastebin.ca/7053
02:18.40AgiNamuHEAD or STABLE... HEAD or STABLE... sigh... what to do...
02:19.05Juggiecoldfeet, why am i looking at the valcano
02:19.11Juggiei dont see anything exciting
02:19.57EightRandom preference pollish question: What distro do you prefer for Asterisk servers?
02:20.01ManxPowerJuggie, No need.  Everything here is NI2
02:20.23*** join/#asterisk zignig (~simon@203.217.15.10)
02:20.33ManxPowerAgiNamu, You should not need to, but try answering the line first
02:20.34coldfeetJuggie, because it one the us newsstations that a plume just shot up from it...about 10 mins ago
02:21.20Qwellmaybe it erupted and killed the webcam?  heh
02:21.31*** join/#asterisk Funbags (~Funbags@ool-18be223d.dyn.optonline.net)
02:21.37Qwellwebcam shows a happy picture.  reality, millions dead
02:22.22Funbagsif i see the call coming in from broadvoice in the debug in asterisk
02:22.39MikeJ[Jayden]coldfeet... looks like somthing weird is going on with your sip config... can you do a reload and see if you still get that on sip load
02:22.42Mavviehappy view!
02:22.56zignigQwell: mmmm , pyroclastic flows , mmmmmmm yummy.
02:23.02GoshenBroadvoice doesn't handle 911 calls...what are other broadvoice users doing about this?
02:23.18GoshenI thought about mapping 911 to call the local dispatch...I could do that
02:23.25Funbagsif i see the call coming in from broadvoice in the debug in asterisk ( it says invite and i can see the caller id and stuff
02:23.49Funbags) but it doesnt pick up the call..
02:24.01Goshentype sip debug
02:24.13Goshenat the asterisk console, then copy and paste it to pastebin.ca
02:24.13modulus_mariah carey is hot
02:24.13coldfeetMike, I have basically stop and restarted the server about a million times, just incase reload didnt do it
02:24.27coldfeetGoshen is at pastebin.ca/7053
02:24.42GoshenFunbags error is?
02:24.46*** join/#asterisk cybertheq (~infra@216-251-177-106.ips.cpinternet.com)
02:24.53FunbagsGoshen, no error
02:24.56coldfeetum...nope just me being confused :)
02:25.10FunbagsGoshen, it just doesnt take the call
02:25.26coldfeetanyone know how test a register request from the cli
02:25.37Goshenwell, we can't help you unless we see the error ;)
02:26.00justinnnnppl
02:26.00FunbagsGoshen, well i mean i dont see the erro
02:26.02AgiNamuManxPower: nope. tried answer, same thing :\
02:26.06Funbagsor :)
02:26.07justinnnnhow do i  get a g++ compiled in redhat.. 7.3 ?
02:26.14ManxPowerfirestrm, did it work?
02:26.22MikeJ[Jayden]Manx, ... you know what this would be?  Mar  9 01:58:15 DEBUG[13063]: db.c:163 ast_db_get: Unable to find key 'home' in family 'SIP/Registry'
02:26.37AgiNamugonna go get CVS stable
02:26.48ManxPowerMikeJ[Jayden], No idea.
02:26.59MikeJ[Jayden]on sip startup.... that's what coldfeet is getting...
02:27.04MikeJ[Jayden]no, me either
02:27.13MikeJ[Jayden]sounds kinda like screwed up astdb
02:27.26MikeJ[Jayden]coldfeet, what version of * are you on?
02:27.42coldfeet1.0.6 ftp one
02:28.14MikeJ[Jayden]coldfeet, pull stable from cvs and recompile... make sure to do a make clean
02:28.28coldfeetbut b4 that line i get a successful seed of the other context namely caroline...i will post it here
02:28.29coldfeetParsing '/etc/asterisk/sip.conf': Found
02:28.30coldfeet<PROTECTED>
02:28.41MikeJ[Jayden]I know..
02:28.48MikeJ[Jayden]I am guessing at this point
02:28.50coldfeetMike what the cvs command
02:28.55coldfeetlet me try it now
02:28.56FunbagsGoshen, SIP/2.0 407 Proxy Authentication Required -- maybe thats the error
02:29.07MikeJ[Jayden]coldfeet:  it's on the wiki
02:29.11MikeJ[Jayden]~docs
02:29.12jbotDocumentation 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
02:29.45*** join/#asterisk lildivil (~lildivil@ool-18bc24d7.dyn.optonline.net)
02:30.04AgiNamuVoicemail is recording a 102byte wav file and eamiling it
02:30.34*** join/#asterisk ionix (ionix@MTL-HSE-ppp184758.qc.sympatico.ca)
02:30.36coldfeetwill make clean remove my config files or can i leave em there
02:31.18Qwellmake clean should just clean the local source
02:31.36Qwellnot sure why you would need to do so on a release version, if you've just downloaded it
02:32.20ionixHey, anyone has an idea on how to handle multiple callers in a prepaid situation ?
02:32.23coldfeetjust did it tonight when the cvs was failing
02:32.32coldfeetnehow let me try
02:32.47QwellIf you're upgrading, you'll need to remove the existing modules first
02:33.14AgiNamuHow do I read a patch .ref file?
02:33.25AgiNamuas in interpret
02:33.34AgiNamu.rej
02:34.20coldfeetdone a make clean in the old install dir
02:34.21*** join/#asterisk Smythe (~Smythe@spock.cbcag.edu)
02:35.57zignighas anyone tried destar?
02:36.47coldfeetokay done Asterisk CVS-v1-0-03/09/05-02:37:40
02:36.54coldfeetand same error :)
02:39.46coldfeetguys does asterisk keep a db somewhere, cause even when i restart it it seems to register caroline immediately surely that should not be the case until the xlite user sends the register request
02:40.11MikeJ[Jayden]yes, ast db...
02:40.23coldfeetwhen does that get cleared
02:40.27MikeJ[Jayden]there are cli commands for it..
02:40.33coldfeetokay let me look
02:41.07ManxPowercoldfeet, YES!
02:41.35MikeJ[Jayden]http://www.voip-info.org/wiki-Asterisk+cmd+DBdeltree
02:41.46ManxPowercoldfeet, "seeding" means "loaded the info from the on disk cache/database".  The next time the phone registers that new info will be saved in the cache/databae.
02:42.30*** join/#asterisk soulz- (~soulz@host-137-132-45-219.imcb.nus.edu.sg)
02:42.34soulz-hello all
02:42.41AgiNamuanyone have any ETA on asterisk 1.1 or something?
02:42.56ManxPowerAgiNamu, 3 - 6 months last I heard (and that was only a rumor)
02:42.57AgiNamuSomething relatively stable, but with all the changes that have happened to the damn api?
02:42.59coldfeetok in the db I see that caroline is in there, so thats where the seed comes from I guess..in fact just read post
02:43.01MikeJ[Jayden]next version of stable will turn into asterisk 1.2
02:43.02AgiNamuahh fuck
02:43.10AgiNamuso 1.1 is nothing?
02:43.26ManxPowermikegrb, I still have not seen KRAM say what the next version of Astersik will be.
02:43.28soulz-manxpower: dude, the bv error went away, but another error came, http://pastebin.ca/7054
02:43.30MikeJ[Jayden]1.1 is the curent head, it will turn into 1.2 after it freazes
02:43.32AgiNamuor 1.0.7 is 1.2?
02:43.39MikeJ[Jayden]and gets "stable"
02:43.40AgiNamuoh ok
02:44.00ManxPowerAgiNamu, As I said I've not seen Digium say that, only non-digium people.
02:44.01MikeJ[Jayden]no
02:44.05soulz-manxpower: when i dial a number, it dials and disconnects, while attempting to bridge the call, and i get a 404 error message
02:44.10*** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net)
02:44.14ManxPowersoulz-, I cannot help you
02:44.22soulz-manxpower: ok :P
02:44.32soulz-can anyone see where i am going wrong?
02:44.34MikeJ[Jayden]mark has said that in dev calls, but that is just a progression, it tells you nothing other than numbers
02:44.35coldfeetnow I have deleted them both...caroline also now gives an error
02:44.36brycecI have a question regarding the use of variables in extensions.conf (or similar)
02:44.46coldfeet<PROTECTED>
02:44.46coldfeetMar  9 02:47:02 DEBUG[19777]: db.c:163 ast_db_get: Unable to find key 'caroline' in family 'SIP/Registry'
02:44.46coldfeetMar  9 02:47:02 DEBUG[19777]: db.c:163 ast_db_get: Unable to find key 'home' in family 'SIP/Registry'
02:44.50coldfeet:)
02:44.51*** join/#asterisk SkySky (~Miranda@HSE-Toronto-ppp286966.sympatico.ca)
02:44.57MikeJ[Jayden]now register..
02:45.08coldfeetI did and I get error again
02:45.28AgiNamuMar  8 21:45:12 WARNING[26673]: app.c:619 ast_play_and_record: No audio available on IAX2/TLF11004@TLF11004/1?? <damm CVS HEAD and 1.0.6 have this issue :@
02:45.29MikeJ[Jayden]as for versions, there is no defined "these features will be in x"
02:45.44brycecCan I set a variable to another variable's value? I've tried the obvious VAR!=${VAR2} but that seems to leave a null value. Can I do this, and how?
02:45.44AgiNamuMikeJ, I'm aware of that tragicality
02:45.50SkySkyhi, i wonder if i need to install zaptel for a virtual machine thats running asterisk if i wanna use meetme and iax
02:46.09AgiNamubrycec... SetVar?
02:46.18xkevbrycec SetVar(newvar=${oldvar});
02:46.25brycecSkySky, No, not unless you're using a T1 card.
02:46.25ionixHey, anyone has an idea on how to handle multiple callers in a prepaid situation ?
02:46.27xkevexpressions are different
02:46.33brycecthanks AgiNamu, xkev
02:47.01xkevGoto($["${newvar}" != "${oldvar}"]?here:there);
02:47.01ManxPowerZaptel is REQUIRED for two asterisk functions.  Meetme, and IAX2 TRUNKING.  Regular IAX2 does not need Zaptel, and neither do some of the trd party conferencing apps for Asterisk.  No, Music on Hold does not need Zaptel.
02:47.09xkevI recommend the quotes, for null vars
02:47.22SkySkybrycec: i've read from wiki saying that I need a timer in order to use meetme and iax trunking
02:47.31AgiNamuSo, anyone have any clues to this VoiceMail issue?
02:47.42cybertheq?
02:47.50brycecsorry SkySky, see ManxPower's reply
02:48.58SkySkybrycec: ops sorry .. i missed his reply~ thank you by the way~^^
02:49.07shmaltztzanger, you around?
02:49.14shmaltz~seen tzanger
02:49.17jbottzanger is currently on #asterisk.  Has said a total of 10 messages.  Is idling for 2h 10m 2s
02:49.17coldfeetwell the all fail now :-)
02:51.25AgiNamuAHA!~!! I found the voicemail issue
02:51.28AgiNamuIt's actually an asterisk issue
02:51.31AgiNamuIf you call yourself
02:51.35Funbagswhy would the console keep displaying destroyed call yad yada...?
02:51.36shmaltzis anybody using a post billing solution with asterisk?
02:51.43AgiNamulike, IAX2/UserA dials IAX2/UserA
02:51.49AgiNamuthen it fucks up and gets it's channels mixed up
02:51.56QwellAgiNamu: nice
02:51.57AgiNamushmaltz, I wrote one
02:52.08shmaltzwhat is it called?
02:52.13shmaltzAgiNamu?
02:52.20AgiNamuTelefinity
02:52.23cybertheqhi all, can someone recommend which linux kernel should be used for CVS HEAD and/or CURRENT?
02:52.24AgiNamuIt's for my own company
02:52.26AgiNamuAll C#
02:52.27AgiNamuand SQL Server
02:52.37AgiNamuand a patch to cdr-csv to make it not suck
02:52.53shmaltzyou saying M$ SQL?
02:53.01AgiNamuNo, I'm saying Microsoft SQL Server 2000
02:53.08AgiNamuM$ SQL is for slashdotters
02:53.15cypromisg
02:53.15SkySkyim trying to use ms virtual server to install asterisk so that my home windows station can use zap lines from office.  but virtual machines doesn't has usb-uhci.  so i wanted to use zaprtc~ i wonder if anyone here tried zaprtc? because im worrying if this would cost a lot of system resources
02:53.16Funbagsheh
02:53.37AgiNamuyea, you'll probably call mono M$ono right?
02:53.39AgiNamuand C# C$?
02:53.44AgiNamucause MS makes money
02:53.44shmaltz:)
02:53.53AgiNamuinstead of being altuistic like digium
02:53.54AgiNamu:D
02:53.56ManxPowerSkySky, Just don't use IAX2 trunking or meetme.
02:54.06JonR800haha
02:54.10cypromisskysky: zaprtc works fine
02:54.13ManxPoweryou'll never get ztdummy to work WELL inside a virtual machine.
02:54.14TrepaliumWho said digium is altruistic?  Buy their hardware!
02:54.27JonR800it's not about making money, it's about product quality and concern for the customer.. :)
02:54.28ManxPowerDo you really think your virtual machine can generate 1,000 interripts per second
02:54.33cypromisshould work inside a virutal machine as well
02:54.43cypromisManxPower:depends
02:54.48cypromis;)
02:55.07ManxPowerSkySky, Honestly, don't spend the time trying to get a timer.  You don't need it for most things.
02:55.23SkySkyic^^ i will give it up then~
02:55.32SkySkythx all
02:55.39SkySky^_^
02:55.47ManxPowerSkySky, once you get everythng else working then try to get a timeer.
02:55.59SkySkyyep. i think i will do that
02:56.51ManxPowerStart with something simple like solving world hunger, then go on to the harder stuff like making a zaptel timer run in a virtual machine.
02:57.18AgiNamuWell... compare MS's test labs to Digiums... :)
02:57.41TrepaliumI like C# as a language, but a lot of MS's .NET framework just ends up frustrating me.  Sealed classes that would be more useful unsealed, piss poor examples that don't show the RIGHT way to do something, some stuff that copies VB's design errors from the past, etc.
02:57.46ManxPowerAgiNamu, If reliability is any indication, Digium has the slightly better lab.
02:58.04*** join/#asterisk PBXtech (~nik@71-32-207-98.slkc.qwest.net)
02:58.23AgiNamuManx, oh they run stress tests on a variety of hardware for months straight?
02:58.29PBXtechwhats this asterisk biz edition about?
02:58.38MikeJ[Jayden]huh?
02:58.39AgiNamuTrepalium, yea, there are some shits in there.
02:58.43AgiNamuWhere they aim for "Mort" developers
02:58.51AgiNamuI have a friend who works on Visual Studio
02:59.04AgiNamuAnd we argue about that a lot. I think the VB idiots should stfu and go away
02:59.11AgiNamuand he realises they have a lot of customers like that.
02:59.28Smytheok - I dial from an ip phone to an extension on my Mitel, via the astrisk server, Ast is connected to the Mitel via a T1.  The Mitel show the incoming channel as locked out, and you hear the trying to ring through from the ip phone - any ideas????
02:59.54AgiNamuTrepalium, any specifics where you feel pain?
03:00.09MikeJ[Jayden]pbxtech. huh?
03:00.18Smythehang up the ip phone and the channel shows idle again on the mitel
03:00.42TrepaliumThe ImageList control keeps pissing me off when doing SWF apps.  Non-extensible, and mangles images way too often.
03:01.24*** join/#asterisk Rival (~rival@66.177.249.219)
03:01.28TrepaliumAnd when doing ASP.NET, the drag-and-drop editor gets in the way more often than not.  Usually just have to write HTML directly instead.
03:01.34PBXtechhttp://blog.tmcnet.com/blog/tom-keating/voip/voip-blog/asterisk-business-edition-launched.asp
03:02.12Qwellugh!
03:02.26QwellDon't EVEN get me started on the god awful designer for asp.net
03:02.59QwellTrepalium: try to put a datagrid inside of a datagrid
03:03.11TrepaliumWhy would I.  I know it does not work.
03:03.18Qwellit "works"
03:03.31Qwellbut in order to design the internal one, you have to pull it out, make your changes, then put it back in
03:03.37MikeJ[Jayden]PBXTech, it's von... they had to anounce somthing :)
03:03.47PBXtechHaHaHa
03:04.11TrepaliumI love the fact that HTML tables are fully editable, but "ASP" Tables are not except through the idiotic collections editor.
03:04.19AgiNamuImageList sucks, yea.
03:04.23AgiNamuSWF sucks actually
03:04.26AgiNamuIn v1
03:04.26Qwellheh
03:04.36AgiNamuVS 2005 is WAAAY better
03:04.41AgiNamuI've been using it for over a year
03:04.50Qwellless designer BS?
03:04.54AgiNamuthe editors are lightyears enhanced
03:04.54AgiNamuoh yea
03:04.57AgiNamuthe editor is a joy now
03:04.59QwellDoes it actually format HTML *PROPERLY*?
03:05.10AgiNamuyes
03:05.13TrepaliumI'm sure it is, but it's not released yet, and the betas are not supposed to have programs developed in them distributed or something.
03:05.14AgiNamuand doesnt fuck with your formatting :)
03:05.18*** part/#asterisk Smythe (~Smythe@spock.cbcag.edu)
03:05.21Qwellabout time
03:05.23AgiNamuTrepalium, nope. Beta2 has a go live license.
03:05.30AgiNamuAnd anyways, I Dont care. MS can sue me.
03:05.33AgiNamuI doubt they will.
03:05.34TrepaliumOh, really...  Guess I
03:05.37Qwellwe won't get to use it until at least 2007 though, heh
03:05.38Trepalium'll need to download.
03:05.48AgiNamuBeta2 is coming really really soon
03:05.55Qwellwe're lucky we have the 1.1 framework
03:06.08Qwellbut...I guess you could just make the code backwards compatible, right?
03:06.13AgiNamuYea
03:06.17AgiNamuI edit Asterisk with VS2005
03:06.17Qwellhmm
03:06.28AgiNamuMS VC++ kicks ass
03:06.29TrepaliumI know VS 2005 finally has theme (or master pages) support so these awful usercontrol hacks to maintain consistency can finally go away
03:06.38AgiNamuTrep: Master pages rocks
03:06.42AgiNamutheming is another cool feature too
03:06.44Qwellwhat are master pages?
03:06.53AgiNamuAllow you to define a main page, and content regions
03:07.03AgiNamuso you can have same layout, look-n-feel, etc. on all pages
03:07.06AgiNamujust replacing the content sections
03:07.08Qwelloh
03:07.31AgiNamuSQL 2005 looks awesome too.
03:07.32TrepaliumI actually fought with CSS for weeks getting the layout right because VS kept mangling my usercontrols by completing the tables on them.
03:07.36AgiNamuC# Stored Procedures
03:07.40AgiNamuyea, VS 2003's editors sucked
03:07.47AgiNamureally uinstable
03:08.06TrepaliumI should've abolished the tables from the start, but CSS is so finicky.
03:08.43TrepaliumPlus the fact that IE is so uncompliant with the CSS spec it takes forever to get it working.
03:08.44MikeJ[Jayden]../quit #html
03:08.56Trepaliumsorry.
03:08.58MikeJ[Jayden]damn.. that didn't work :)
03:09.01ManxPower.part #i-hate-microsoft
03:09.01MikeJ[Jayden]hehe
03:09.27Qwell/part #so-do-I-,-but-it-pays-the-bills
03:09.38MikeJ[Jayden]OKAY... listen up everyone...
03:09.56MikeJ[Jayden]okay Mank, I have their attention, your turn ;)
03:10.09AgiNamu./something #ms-kicks-ass-and-you-know-it
03:12.28AgiNamu....
03:12.29TrepaliumMS kicks ass, and then ends up in court on antitrust issues.
03:12.52AgiNamuMS ends up in court on antitrust, and guess what? today the US government goes and buys Corel for their office suite
03:12.56AgiNamudisproving their own argument
03:13.07AgiNamuMS ends up in court cause they have lame comptetion who cant keep up
03:13.29GoshenI don't understand why they didn't just go with Oo and save themselves millions of dollars
03:13.35Mavvietrue. just like they did with the rail barons and the oil barons.
03:13.37TrepaliumThat is because lawyers are the one corner keeping Wordperfect alive.  Most law firms use wordperfect and have for years.
03:13.38*** join/#asterisk xFuck3r (~OwneD@200-232-182-63.dsl.telesp.net.br)
03:13.43*** part/#asterisk xFuck3r (~OwneD@200-232-182-63.dsl.telesp.net.br)
03:13.58AgiNamuOo?
03:14.06AgiNamuOpenOffice?
03:14.17AgiNamuanyone who thinks there's actually competition to Word is smoking crack
03:14.18TrepaliumThe IE/Netscape thing was pretty scummy.
03:14.22AgiNamuor doing LSD
03:14.31*** join/#asterisk The_Ball (~alex@static-100.35.240.220.dsl.comindico.com.au)
03:14.35AgiNamuScummy? Yea, Netscape had an incompetent, arrogant, product team.
03:14.39AgiNamuNetscape sank themselves.
03:14.50AgiNamuMS knows how to do products.
03:14.59AgiNamuJust cause you're a hotshot coder with some VC money doesnt mean shit.
03:15.11TrepaliumYes, but bundling it with the OS, and forcing OEMs to install it is pretty anti-competitive.
03:15.12MikeJ[Jayden]damn, I'm sunk now.
03:15.13The_Ballis it possible to configure asterisk so people don't have to use iax/user:pass@domain.com/extension but can use just iax/user:pass@domain.com ?
03:15.24ElsharHmm, when was 1.0.6 released?
03:15.34MikeJ[Jayden]cuz I'm not a hotshot coder, and I have no money
03:15.46*** join/#asterisk Shaneful (~user@24.85.246.178)
03:15.50MikeJ[Jayden]the_ball... yes
03:15.54AgiNamuTrepalium, having an OS with a browser built-in is just common sense.
03:15.58TrepaliumNetscape would've likely lost one way or another, but MS was too impatient and sunk to dirty tricks.
03:16.07AgiNamuLinux desktop OSes bundle a browser.
03:16.10Funbagswhen asterisk registers with broadvoice what should the debug responce be?
03:16.22AgiNamuMS's mistake was to assume HTML was so damn cool and integrate IE and shit soo far into the browser.
03:16.25zignigThe_Ball: yes you can
03:16.30AgiNamui mean, into their other products :P
03:16.52ShanefulAnyone use astgui?
03:16.55zignigThe_Ball: http://www.iaxtel.com/setup.html , you just set it up to route all calls to iaxtel numbers to the trunk
03:16.57AgiNamuMS did us a favour by ending Netscape earlier for us.
03:17.10AgiNamuIf they hadn't we'd be using Netscape Vx now
03:17.14AgiNamuinstead of FireFox
03:17.28MikeJ[Jayden]zing, what the hell are you talking about
03:17.39AgiNamuNetscape also did not understand developers. they didn't package their browser as a reusuable component
03:17.40ElsharAnyone good at figuring out wacky * problems? Can't seem to get to the xvoip forums atm :/
03:17.41AgiNamumaking even MORE people use IE
03:17.44AgiNamulike AOL :)
03:17.51MikeJ[Jayden]the_ball, if you don't specify an extension it will go to the s extension
03:17.56The_Ballzignig, something like exten => _,1,Goto(incoming,s,1) ?
03:18.05AgiNamuanyways, that's irrelevant to how cool .net is
03:18.10TrepaliumAt least we have firefox now...  Now if I could only get rid of IE for good (especially since I accidently installed spyware on a server by mistyping a URL when trying to get drivers).
03:18.12AgiNamudespite the stupid name.
03:18.22The_BallMikeJ[Jayden], oh
03:18.25AgiNamuTrepalium, if you got rid of IE, a ton of 3rd party apps would stop working
03:18.25MikeJ[Jayden]the_ball... exten => s,1,whatever
03:18.29AgiNamugo delete iexplore.exe
03:18.37AgiNamumshtml is required by a lot of stuff
03:18.46AgiNamuIn fact, the new Netscape will actually use MSHTML to render!
03:19.00zignigThe_Ball: same idea but routes iaxtel numbers specifily
03:19.01MikeJ[Jayden]ok.. I'm sick of this.. go take the browser converation somwhere else...
03:19.13MikeJ[Jayden]or I will make you run * on windows
03:19.14The_BallMikeJ[Jayden], can you try a iax call to guest@au.wigen.net just to see if you get an answer?
03:19.27AgiNamuMikeJ -- I'll give you $20,000 right now for a port of * to Win32.
03:19.28Trepaliummikegrb: Done that.  Wasn't much fun.
03:19.36AgiNamuThen I could get rid of these damn linux machines
03:20.02Trepaliumack.  nick complete failure.
03:20.02MikeJ[Jayden]I have * on win32
03:20.02ElsharAgiNamu, Just use freebsd :P
03:20.02AgiNamuand run one nice envrionement
03:20.02MikeJ[Jayden]its not a port, it's called compiling
03:20.03ElsharIf you don't need the zaptel hardware, shouldn't be a problem.
03:20.03AgiNamuWith the MS CLR on all machines, instead of Mono
03:20.08MikeJ[Jayden]it's cygwin, but it runs
03:20.11AgiNamuMikeJ, you've done it under cygwin?
03:20.15AgiNamuand it works o.k.?
03:20.15coldfeetgoodnite people, tks for the help, will try again tommorrow
03:20.19coldfeettks mike
03:20.31MikeJ[Jayden]it works, I wouldn't say "well"
03:20.50AgiNamuyea, that's why it's called a port.
03:20.50MikeJ[Jayden]but enough that I can say, I can run * on windows...
03:21.13MikeJ[Jayden]really, there are a few minor changes to the code, and a hacked to sh*t makefile
03:21.58ShanefulI want to trying to set setvar in the asterisk manager api, but it doesnt seem to be working, does anyone have any exp with it?
03:22.06ElsharAnyone have a vague idea as to why asterisk might be consistantly dropping calls after 20 seconds?
03:22.19MikeJ[Jayden]I am waiting on a few things from cygwin cvs to hit release before I do any patches, specifically there is some stuff for process priority that will be in cygwin soon that don't work right now... it also fails compile if you don't have unixodbc installed.
03:22.23TrepaliumI wonder how mono's implementation of .NET remoting is.  I need a manager proxy/tracker of some sort for this call control app.
03:22.30*** part/#asterisk Moc____ (~mochouina@64.235.210.66)
03:22.49MikeJ[Jayden]aginamu, why do you want * on win32 so bad
03:22.50MikeJ[Jayden]??
03:23.40MikeJ[Jayden]Moc.... Nooooooo
03:23.52ShanefulI have a NoOp in the extensions.conf but the variable prints out blank
03:23.52MikeJ[Jayden]wow, * on windows shuts everyone up :)
03:24.02The_Ballanyone care to test if iax/guest@au.wigen.net works?
03:24.13Goshensure
03:24.14MikeJ[Jayden]shaneful, that's becuase the variable is not set :)
03:24.23ShanefulI figured that.
03:24.28MikeJ[Jayden]hehe
03:24.30ShanefulI am prabably doing something stupid
03:24.32MikeJ[Jayden]glad to help
03:24.34Shanefulbut it elludes me
03:24.54Shanefulcan I paste something to you?
03:25.03*** join/#asterisk myconid (~myconid@12.177.133.243)
03:25.07MikeJ[Jayden]well.. the only guy I really tried to help today I borked even worse, do you want to risk it
03:25.14Shanefulsweeeet
03:25.16Shanefultwice as fun!!
03:25.16myconidis anyone running cvshead w/ broadvoice?
03:25.22MikeJ[Jayden]~pastebin
03:25.23jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
03:25.51MikeJ[Jayden]I cancelled my broadvoice yesteday, decided I wanted IAX cuz I am sick of screwing w/ sip and nay
03:25.53MikeJ[Jayden]nat
03:26.02myconidwhod you switch to?
03:26.17zapahi all, is posible to change default context of a Sip extension in dial plan.
03:26.18ManxPowerSIP?  Nay!  Nay!
03:26.19MikeJ[Jayden]nufone for now.. we will see how I like them
03:26.32ManxPowerzapa, default context?  No.
03:26.35MikeJ[Jayden]zapa. goto?
03:26.35GoshenThe_Ball: "Welcome to Alex an Paul's house......" getting your phone tree...
03:26.42Shanefulhttp://pastebin.ca/7055
03:26.52cybertheqhi all, anyone familiar with quicknet cards?
03:27.05*** join/#asterisk Damin_Mobile (~pocketirc@152.sub-70-214-30.myvzw.com)
03:27.09The_BallGoshen, thats great, the phone rang as well
03:27.11MikeJ[Jayden]I'm sure somone from quicknet is ;)
03:27.11jetsI love broadvoice
03:27.23GoshenThe_Ball: left message for Alex
03:27.34cybertheqI need advice on which linux kernel to use with the 'ixj' driver.
03:27.38GoshenThe_Ball: its ... iax2/guest@au.wigen.net
03:27.39The_Balli'll get it in my email inbox :)
03:27.42MikeJ[Jayden]I hate sip and nat... if it's any consolation, they were sorry to lose my business too.. an auto e-mail told me
03:27.47Goshennot iax/
03:27.48MikeJ[Jayden]ixj?
03:27.57*** join/#asterisk syslod (~yurplsl@65.114.0.198)
03:27.59cybertheqquicknet internet phone jack
03:28.05The_BallGoshen, oh, i c
03:28.08ManxPower"Thank you for calling The Center for Total World Domination, a department of the NSA.  If you know your party's extension, you may dial it at any time...."
03:28.10syslodHello
03:28.17ManxPowerI should put that on my boyfriend's Asterisk IVR.
03:28.34ManxPowerWe should get Alison to record that.
03:28.41MikeJ[Jayden]dirty, dirty tricks...
03:28.58MikeJ[Jayden]I need to scam that one great recording from bkw
03:29.24MikeJ[Jayden]the one about we will hangup on you as soon as * segfaults
03:29.28MikeJ[Jayden]I like it.
03:29.39myconidI dont ever get voice from broadvioce, but I transmit just fine
03:29.40myconidany ideas?
03:29.46zapaI need to user can dial long distance call, only when the user is sit at his desk, and when he leaves put again password, like a block of the telephone extension. But i don´t want the user always put a password
03:29.58The_BallGoshen, haha, that's really cool
03:30.19Goshenfun stuff isn't it? :)
03:30.35MikeJ[Jayden]shaneful, wtf did you pastebin....
03:30.35Goshennow whenever I want to call Alex and Paul's house I just dial ext. 40
03:30.41MikeJ[Jayden]hehe
03:30.42Shanefulheh
03:30.45*** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
03:30.54Shanefultelnet to the asterisk server
03:30.59Shanefuluse manager api
03:31.12Goshentelnet? people still use telnet?
03:31.21MikeJ[Jayden]ummmm.... how about the bit of dialplan that's borked
03:31.38MikeJ[Jayden]Goshen is an SSH snob
03:31.46The_BallGoshen, now i can add the iax2 url to my webpage to
03:31.58ShanefulI didnt think you could compile asterisk with ssh...
03:32.00Shanefulheh
03:32.35*** part/#asterisk zignig (~simon@203.217.15.10)
03:32.48MikeJ[Jayden]Manx, where did you go, I'm dying here
03:32.57shmaltztzanger, you around?
03:33.07MikeJ[Jayden]and where is tzanger... he should be answering these questions :)
03:33.14cybertheqanyone? what kernel to use with 1.0.6/CVS/HEAD/CURRENT/etc?
03:33.15shmaltz~seen tzanger
03:33.17jbottzanger is currently on #asterisk.  Has said a total of 10 messages.  Is idling for 2h 54m 2s
03:33.30shmaltzalmost 3 hours that he is sleeping already
03:33.32MikeJ[Jayden]cybertheq, 2.4, 2.6 are fine
03:33.35ManxPowerMikeJ[Jayden], Now you can experience another fun thing: All the old timers stop answering questions, leaving you with a flood of newbies wanting help.
03:33.48MikeJ[Jayden]hehe
03:33.54shmaltzManxPower, you sure enjoying this
03:34.04MikeJ[Jayden]I will just go to my spider solitaire
03:34.09MikeJ[Jayden]yes, of course he is.
03:34.13shmaltzManxPower, if my memory server me right you a BSD fan, no?
03:34.33ManxPower"I'm having a miserable day.  There's not reason your's should be any different." --Unknown
03:34.56MikeJ[Jayden]:( Shaneful does not want any real help cuz he won't pastebin me anything useful... isn't that sad
03:35.03shmaltzManxPower, of course there is, its the fact that your is
03:35.16Shanefulok mate.
03:35.17Goshencybertheq: I am using 2.6.10 with 1.0.6 stable
03:35.25cybertheqthanks goshen
03:35.34tzangereh?
03:35.37MikeJ[Jayden]hehe... he said mate... are you one of those aussies...
03:35.38shmaltzhelo
03:35.40Goshenjust make sure you compile in the ccitt stuff
03:35.41MikeJ[Jayden]tzanger!
03:35.43cybertheqdo you know of any probs with the 'ixj' driver on 2.6x?
03:35.45tzangerwhat's up
03:35.48MikeJ[Jayden]I'm taunting people...
03:35.58tzangerixj?  quicknet stuff?  I have one but haven't used it in forever
03:36.00shmaltztzanger, slackware is running and rockin
03:36.02tzangernever got it to work nicely with asterisk
03:36.05tzangershmaltz: excellent
03:36.15Shanefulhttp://pastebin.ca/7056
03:36.21MikeJ[Jayden]anyone bumb me a smoke?
03:36.27cybertheqread my posts on [asterisk-bsd] regarding quicknet for a cry...
03:36.29ShanefulI will buy ya a carton.
03:36.35shmaltzbut I'm trying to recompile without usb, and for some reason it gest messed up
03:36.41tzangerwhy
03:36.48Shanefulgimme address I UPS overnight them if ya find me solution
03:36.50tzangerMikeJ[Jayden]: heh
03:37.24Qwellheh, I had smokes UPS'd to me once
03:37.28viporwitch one is better the X100P or X101P or does it not madder/?
03:37.29MikeJ[Jayden]I won't solve you problem if you won't help me...
03:37.31shmaltztzanger, I wish I knew
03:37.34Qwellhe didn't spring for overnight though. :(
03:37.38ShanefulI am using VICI dial
03:37.39tzangershmaltz: no why are you compiling without usb
03:37.47Shanefulit passes all its commands thru the asterisk manager api
03:37.49shmaltzI can't modprobe anything, and lsmod shows nothing is loaded
03:37.52MikeJ[Jayden]do you have some dialplan with the noop and stuff... or just what you are dumping to manager
03:38.02shmaltztzanger, b/c I need ztdummy
03:38.03MikeJ[Jayden]so where is this noop you are talking about
03:38.08GoshenMikeJ: Here you go....http://www.smokehelp.org/assets/images/Smokers_Lungs.jpg
03:38.14tzangershmaltz: ah
03:38.19tzangershmaltz: want a trick
03:38.24shmaltzpls
03:38.35tzangershmaltz: copy the /boot/config-whatever-whatever to .config and make oldconfig
03:38.35syslodSmokes kill, drink a beer.
03:38.46Qwellsweet
03:38.47tzangerthis will get you the same configuration as the original slackware kernel
03:38.47Trepaliumvipor: I don't believe it matters.
03:38.51QwellGoshen: I need a smoke now.
03:38.58shmaltzit didnt work for  me
03:39.00tzangershmaltz: then you can make menuconfig and select/deselect the speciific bits you want
03:39.02Qwelland I'm about to eat red meat too...its a good night for me
03:39.09cybertheqwell anyone, since linux 2.4 is known to work with 'ixj', where can one download fc1 or some such (not at fedora-legacy...
03:39.11tzangerQwell: I had a roast tonight
03:39.15vipor<Trepalium>  ok thanks
03:39.22shmaltztzanger, I tried it and my boot didnt go past Loading bzImage ........................
03:39.24Qwelltzanger: we're having sloppy joes :p
03:39.35tzangershmaltz: you reran lilo?
03:39.41*** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net)
03:39.42MikeJ[Jayden]I'm having gas, what's your point
03:40.02shmaltztzanger, I didnt' need to, b/c the files name didn't change (just overwrote them, after saving the old ones)
03:40.13tzangershmaltz: you MUST rerun lilo
03:40.16tzangerMUST MUST MUST MUST
03:40.19tzangerafter EVERY kernel update
03:40.20MikeJ[Jayden]shaneful, I am still not getting what's not working
03:40.25tzangerlilo != grub
03:40.29shmaltztzanger, I know how to compile the kernel
03:40.34shmaltzwhy?
03:40.35tzangerlilo is much much dumber than grub (one of the things I prefer anyway)
03:40.44Qwellyeah, I like grub alot more
03:40.51TrepaliumMuch dumber, and much more difficult to kill.
03:40.52ShanefulThe var isnt being set.  I shell into server.
03:40.52MikeJ[Jayden]tzanger likes em dumb...
03:40.53shmaltzactualy I think grub is dumber
03:41.02viporfor freebsd does Asterisk like KDE, or does it also not matter what X-windows I use
03:41.09tzangerwhen you overwrite the kenrel you aren't using the exact same inode map, and rerunning lilo will have it grab the current inode map for the kernel and update lilo with it
03:41.13shmaltzbut why do I have to rerun lilo?
03:41.15Qwellvipor: You could run it without X
03:41.18tzangershmaltz: no grub is FAR more intelligent with this
03:41.20MikeJ[Jayden]how do you know the var is not set, where do you see that?
03:41.30tzangershmaltz: you need to tell lilo the kernel's new inode map
03:41.34shmaltztzanger, maybe with this
03:41.36tzangerlilo has *no* concept of filesystems
03:41.37viporic
03:41.44GoshenDoes anyone have any idea how to set the "forwarded call" flag on a call? When forwarding a call on my PSTN Qwest line, my cellphone(AT&T) shows an arrow next to the caller ID, letting me know it is a forwarded call
03:41.47puppetAnyone here using festival anything or is it aint that good?
03:41.49tzangergrub has filesystem drivers so it can actually "seek out" the kernel... lilo cannot
03:41.52Trepaliumvipor: I'm not sure X100Ps work in FreeBSD, FYI.
03:41.53GoshenI want to duplicate that behavior on my asterisk box
03:41.57shmaltzbut, grub still thinks its hd0, when it's actualy hda
03:42.02viporo
03:42.11tzangershmaltz: huh?
03:42.13MikeJ[Jayden]puppet, not that good... but funny at times
03:42.32ManxPowergrub always confused me.  lilo doesn't.
03:42.32tzangerare you running lilo or grub?
03:42.38tzangerManxPower: exactly
03:42.44shmaltzin grub I have to tell it to look in hd0, while in lilo I can tell it to look in /dev/hda
03:42.44MikeJ[Jayden]I'm running... to the store
03:42.48shmaltzlilo
03:42.50tzangerI see no reason to have the boot loader understand fileysstems
03:43.05shmaltztzanger, you are right about that
03:43.10tzangershmaltz: then please, after you update the kernel, run lilo so that it loads the new inodemap into the bootloader
03:43.15The_Ballshmaltz, grub uses hd0 hd1 hd2 for you harddrives
03:43.17Goshenwow, do a search on google image for asterisk...now thats a fan :)
03:43.18puppetmikej[jayden]: ok, having some trouble with it if its worth putting up
03:43.24MikeJ[Jayden]shaneful... PM me
03:43.27TrepaliumWith grub you can sometimes avoid using a rescue disk versus lilo.  But who doesn't like li 01 01 01 01 01 ...
03:43.34shmaltzthe_ball, thanks, thats exactly what I said
03:43.37ManxPowerthe kernel RPM SHOULD run lilo for you, but NEVER trust it.
03:43.40MikeJ[Jayden]ummmm. I don't use it any more
03:43.53tzangerGoshen: hahahahha
03:44.03shmaltztzanger, so you saying the loading hang was b/c of the inode maping in lilo?
03:44.13ManxPowerIt would be nice if lilo installed a checksum of the inode numbers and complain if it changes.
03:44.17tzangerTrepalium: the *only* time I have that hapen is when I confuse the shit out of the boot process
03:44.21tzangershmaltz: exactly
03:44.22The_Ballshmaltz, yes, but it sounded by you as that was wrong... it's supposed to be that way
03:44.25tzangerManxPower: it can't
03:44.29myconidanyone have the one way audio issue?
03:44.35tzangerManxPower: you still need to run lilo to have it check the inode map
03:44.42tzangerManxPower: so why bother checksumming it
03:44.43*** part/#asterisk cybertheq (~infra@216-251-177-106.ips.cpinternet.com)
03:44.49TrepaliumI've seen it way too often..  Usually when I'm playing around with partitions, and resizing them.
03:45.05ManxPowertzanger, A more informative and user friendly message than: LI
03:45.13MikeJ[Jayden]myconid, one way audio is often firewall issue
03:45.14tzangerManxPower: :-)  well yes
03:45.19shmaltztzanger, but why didn't the compiled kernel allow me to load anything?
03:45.19tzangerManxPower: I never said it wasn't too terse
03:45.21Goshenmyconid: yup I did before I figured out how to configure rtp.conf, port forward, nat=yes(in sip.conf), and generally BEND OVER BACKWARDS FOR SIP
03:45.28tzangershmaltz: well two things
03:45.34tzanger1) you overwrote the working kernel
03:45.43tzanger2) it didn't complete loading the kenrel from the looks of it
03:45.54Goshenmyconid: what provider?
03:46.01shmaltztzanger, 2 make sense
03:46.01myconidGospen: broadvoice.
03:46.06myconidMikeJ: I killed my firewall.
03:46.20tzangerI always install the new kernel as /boot/bzImage.2610 (or whatever) and then add an entry to lilo.conf called linux and rename the WORKING one to oLinux or something (old linux) -- that way if the new kenrel fucks up you can alway sboot the old one
03:46.29Goshenmyconid: how do you handle 911 calls?
03:46.48myconidGoshen: I walk down the street to the State Police HQ
03:46.49*** join/#asterisk techie (gus@asterisk.horizonte.us)
03:46.57myconidor pick up my ham radio
03:47.03QwellWhy not just get an FXO, and use the PSTN?
03:47.04myconidand talk to dispatch
03:47.11tzangershmaltz: also a trick I learned for remote is to install the NEW kernel as the second kernel in the lilo.conf list and use lilo -R Linux to make it "point" to the new one for the next boot so if it fails I can have someone press the big red button and it'll boot the owrking kernel automatically and I can fix what I screwed up
03:47.17mikegrbme too!
03:47.18Goshenmyconid: I am going to map 911 to the local dispatch
03:47.19QwellYou don't need a provider to have 911, do you?
03:47.22shmaltztzanger, if i cp /boot/config* /usr/src/linux/.config and run make menuconfig in /usr/src/linux will it show the old config?
03:47.29tzangershmaltz: no
03:47.31tzangermake oldconfig first
03:47.32mikegrbexcept I pick up my cell phone or my ham radio
03:47.34Goshenqwell: babysitter might need to dial 911
03:47.37tzangerthen make menuconfig
03:47.41myconidI think you can get a POTS line for 911 only.
03:47.48QwellI mean...
03:47.49tzangeryou want it to "sanitize" the old config against the new kernel
03:47.49myconidlike a dead cell phone cal call 911
03:47.54QwellYou don't need a POTS provider to have 911, right?
03:47.59Qwellit "Just Works"?
03:48.01shmaltzso what does make mrproper do?
03:48.07myconidQwell: On a POTS line?
03:48.11mikegrbQwell: /if/ the pair is connected
03:48.11myconidor cell
03:48.12*** join/#asterisk JerJer[mobile] (~jj@65.173.197.109)
03:48.13tzangermrproper cleans out everything and zaps the .config file while it's at it
03:48.15Qwelldunno, thats what I've always been told
03:48.16myconidsweet.
03:48.16Goshena girl in Florida tried to call 911 but their voip provider didn't provide it, and her parents were both shot and killed
03:48.25shmaltzI see
03:48.27tzangerGoshen: got a link to that
03:48.28Qwellnice
03:48.28JerJer[mobile]hell yeah
03:48.30JerJer[mobile]good stuff
03:48.31tzangerI don't buy it
03:48.36Goshenlet me get it
03:48.49shmaltzGoshen, I'm with tzanger on this one
03:49.00shmaltzthe media loves to overdramatize things
03:49.01GoshenI keep an old cellphone for 911 too, keep it at the head of the bed, and keep the battery disconnected
03:49.21myconidi did it tonight infact.
03:49.23shmaltztzanger, thanks alot
03:49.26ManxPower*shrug* I need POTS for DSL, I figure it's there for 911 use.
03:49.32tzangernews.googling for 911 voip murder and variants turns up 0 hits
03:49.40ManxPowerI even have a big red phone connected directly to it with the ringer turned off.
03:49.42mikegrbQwell: sometimes you might have tone on a disconnected pots line then 6 months later, they need that pair for somebody else
03:49.42myconiddriving home.. a suv rolled twice off the highway.
03:49.46Goshenjsmith told me about it...
03:49.46TrepaliumThat was the propaganda that PSTN providers have been using for ages to complain about VoIP.
03:49.50mikegrbmyconid: heh I did that a lot a few months ago
03:49.56tzangerManxPower: exactly my thinking too -- I also sell service and say it is SECOND LINE SERVICE ONLY and htat 911 is NOT SUPPORTED
03:49.59mikegrbmyconid: and worked on the other end as well
03:50.03shmaltz~google murder voip 911
03:50.43myconidcan someone share their incoming extensions file?
03:50.48myconidim curious what normal ppl do
03:51.20tzangermyconid: my incoming extensions.conf is very sparse
03:51.27tzangerjust dials my zap group 1
03:51.30tzangerI don't do any IVR
03:51.36myconidtzanger: do you have transfer setup? and voicemail?
03:51.44tzangermyconid: yes and yes
03:51.51shmaltztzanger, you went to that second link?
03:51.51myconidhow do i do transfer and voicemail? :)
03:51.57myconidwana send me that ? :)
03:51.58shmaltzbottom story
03:52.11tzangerjust dial with 't' option
03:52.19Mocis sixtel down now ?
03:52.28tzangerMoc: It's bouncing on my system (just noticed it)
03:52.50GoshenMoc: website appears to be
03:52.51Mocdamn.. I did the mistake to use the DNS name ..
03:53.03Mocdns doesnt resolve, and it causing major problem on my box now ..
03:53.09Mocdont try to reload your config ...
03:53.29tzangershmaltz: interesting
03:53.36shmaltzyep
03:53.47tzangersounds like someone didn't read their T&C correctly
03:54.17shmaltzexactly
03:54.21Mocok now I can't register to my asterlink account .... damnit .. it aint my provider day today..
03:54.26TrepaliumI can't seem to send audio using X-Lite and the Speex codec, and sending via iLBC sounds very, very bad.  Is this normal?  The other three codecs seem fine.
03:54.27mog_homemake samples
03:54.28shmaltzmost providers tell you no 911 services
03:54.29mog_home^_^
03:54.45myconidhow do i transfer a call?
03:54.54tzangerwith #
03:54.58tzangeror hookflash for zap
03:54.59myconid#102 ?
03:55.04shmaltztzanger, you seen this:
03:55.05tzangermyconid: pretty much yes
03:55.06shmaltzhttp://home.exetel.com.au/azyc/asterconf/
03:55.18Goshenjust map extension 911 to dial your local dispatch
03:55.18tzanger# <alison sez "Transfer"> 102
03:55.26myconid<3 alison
03:55.29tzangerI actually dialled that exact sequence today
03:55.33tzangertalking to my romanian hottie :-0
03:55.57*** join/#asterisk alakdan (~alakdan@210.213.173.63)
03:56.09Mocall the provider are at Von, so arnt there to keep their service running ;)
03:56.39myconidIt says invalid extension
03:56.49Goshensixtel's dns servers down too..shesh
03:57.08marlowesixtel is up
03:57.37marloweuhh
03:57.37marlowenevermind :)
03:57.40Goshen:)
03:57.48marlowegood thing i stopped using them
03:57.56shmaltzA man is getting into the shower just as his wife is finishing up her
03:57.57shmaltzshower, when the doorbell rings.
03:57.59shmaltzThe wife quickly wraps herself in a towel and runs downstairs.
03:58.00shmaltzWhen she opens the door, there stands Bob, the next door neighbor.
03:58.02shmaltzBefore she says a word, Bob says, "I'll give you $800 to drop that
03:58.03shmaltztowel,"
03:58.05shmaltzAfter thinking for a moment, the woman drops her towel and stands naked
03:58.06shmaltzin
03:58.08shmaltzfront of Bob.
03:58.09shmaltzAfter a few seconds, Bob hands her $800 dollars and leaves.
03:58.11shmaltzThe woman wraps back up in the towel and goes back upstairs.
03:58.12shmaltzWhen she gets to the bathroom, her husband asks, "Who was that?"
03:58.14tzangershmaltz: yeah I heard this one.  It's good though :-)
03:58.14shmaltz"It was Bob the next door neighbor," she replies.
03:58.15shmaltz"Great!" the husband says, "did he say anything about the $800 he owes
03:58.17shmaltzme?"
03:58.19shmaltzMoral of the story: If you share critical information pertaining to
03:58.19myconidwhere do i enter extensions that are vlaid for transfer
03:58.20alakdanjust wondering what would be the possible error here. the 866 number works but the 877 does not
03:58.20shmaltzcredit
03:58.21shmaltzand risk with your shareholders in time, you may be in a position to
03:58.22alakdan[inbound]
03:58.22alakdanexten => 8774673648,1,Goto(DonateX-alohamm,s,1)
03:58.22alakdanexten => 8668247992,1,Goto(DonateX-alohamm,s,1)
03:58.23shmaltzprevent avoidable exposure
03:58.25shmaltz:)
03:58.27marloweMar  8 22:58:19 NOTICE[17862]: chan_iax2.c:5667 socket_read: Peer 'sixTel' is now REACHABLE! Time: 35
03:58.29marlowethere ya go
03:59.08mikegrbIf you're not sure if you have 911 service, do not call the operators at the 911 emergency center. It is against the law to call 911 just to test it.
03:59.11mikegrbIt is best to call your Internet service provider. Time Warner Cable is reportedly the only provider that offers a full emergency 911 service.
03:59.15mikegrbbullshit on both counts
03:59.15marloweIm still waiting on a DID I ordered weeks ago or something
03:59.17SexyKenHey guys...
03:59.22mikegrbtherefore I don't believe and damn thing in that article
03:59.23SexyKen....does realtime support nat=yes
03:59.24*** join/#asterisk qwerp (~abc@219.93.57.58)
03:59.28shmaltzA priest offered a lift to a Nun.
03:59.30shmaltzShe got in and crossed her legs, forcing her gown to reveal a leg.
03:59.30qwerpharlo..
03:59.30tzangermikegrb: yeah but I can fully unerstand why they'd say that
03:59.31shmaltzThe priest nearly had an accident. After controlling the car, he
03:59.32shmaltzstealthily
03:59.34shmaltzslid his hand up her leg.
03:59.36shmaltzThe nun said, "Father, remember Psalm 129?"
03:59.37shmaltzThe priest removed his hand. But, changing gears, he let his hand slide
03:59.39shmaltzup
03:59.40shmaltzher leg again.
03:59.42shmaltzThe nun once again said, "Father, remember Psalm 129?"
03:59.43shmaltzThe priest apologized "Sorry sister but the flesh is weak."
03:59.45shmaltzArriving at the convent, the nun went on her way.
03:59.46shmaltzOn his arrival at the church, the priest rushed to look up Psalm 129.
03:59.47Goshenpacket8 supports 911
03:59.48shmaltzIt said, "Go forth and seek, further up, you will find glory."
03:59.49shmaltzMoral of the story: If you are not well informed in your job, you might
03:59.50SexyKenWhat the fuck dude
03:59.51shmaltzmiss a great opportunity.
03:59.51SexyKenShut up
03:59.57shmaltz:)
04:00.00shmaltzI'm bored
04:00.06SexyKenAnd you're fucking retarded.
04:00.13tzangerhmm
04:00.17tzanger27MB/sec on firewire
04:00.20qwerpharlo..
04:00.21tzangerthat's not great
04:00.23GoshenI would rather listen to a joke then your swearing
04:00.24tzangerbut not bad
04:00.33qwerpany good recommendation other than IP500 phone sets?
04:00.39mikegrbI'd rather listen to anything but SexyKen any day
04:00.41marloweIP300? :)
04:00.47qwerperr..
04:00.47shmaltzSexyKen, i'll stop for now
04:00.49marloweIP600? :)
04:00.50qwerpcheaper ones?
04:00.55mikegrbIP501?
04:00.57marloweAhh, more precise question
04:00.58marlowelol
04:01.03mikegrbIP499?
04:01.07marlowelol
04:01.09tzanger224Mbps on a 400Mbps link
04:01.10marloweIP300
04:01.19tzangerI wonder what hitting 5 drives on it would do
04:01.28marloweYou can buy my cisco .. uhh what is it
04:01.29tzangerif it's 224Mbps total or just that controller is slow
04:01.29marlowe7905g
04:01.38marloweits been sitting here, unused forever
04:02.25Shanefulouch
04:02.25*** join/#asterisk JerJer[mobile] (~jj@65.173.197.109) [NETSPLIT VICTIM]
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04:02.27marlowewoahh
04:02.27qwerpip300 supports ADSI, right?
04:02.27SexyKenNow that was funny.
04:02.27Shanefulits like a zipper
04:02.27TrepaliumWhoa.  Visions of efnet.
04:02.28tzangerhaha
04:02.28tzangershhhh I'm on efnet
04:02.37`SauronThat's dirty. :)
04:02.37Goshenhey, sitel came back online once that netsplit came back
04:02.37Goshen[21:02] * Dns resolved iax2.sixtel.net to 205.234.133.203
04:02.42*** join/#asterisk footnote (~jhicks@67.141.135.121)
04:02.44marlowedood
04:02.45GoshenSixtel
04:02.46marlowesixtel came back a while ago
04:02.50marlowelol
04:02.57SexyKenFor some reason I can't get my IP 600 to dial international numbers, dont even see it try to dial in asterisk cli.
04:02.58marlowei pasted it like 10 minutes ago they were up
04:03.06footnotesixtel > pachell
04:03.09marloweSexyKen: Check your dial plan in the phone
04:04.17footnotehrm, sixsixsixtel
04:04.30shmaltzSexyKen, sorry I think this one is realy good:
04:04.32shmaltzA turkey was chatting with a bull. "I would love to be able to Get to the top of that tree," sighed the turkey, but I haven't got the energy." "Well, why don't you nibble on my droppings?" replied the bull. "They're packed with nutrients." The turkey pecked at a lump of dung and found that it gave him enough strength to reach the lowest branch of the tree. The next day, after eating some more dung, he reached the second branch. Finally after a fourth ni
04:04.45SexyKenmarlowe -> The dial plan looks as if it allows international dialing.
04:05.02marlowei dunno i gotta go
04:05.09footnoteshmaltz: truncated
04:05.11marlowedouble check it :)
04:05.17marlowedouble check your asterisk plan
04:05.25marlowemake sure it works from another phone first
04:05.30marlowethen you can elimate it being asterisk
04:05.34marlowethen its obviously the phone
04:05.41marlowei do internetional dialing from my polycoms all day
04:06.05qwerpother than polycom,
04:06.21qwerpwhat other brands ip phone is good?
04:06.39footnotet polycom plz b 4 oss phonez k plz thx
04:06.57SexyKencisco
04:07.13qwerperr.. and cheap :D
04:07.16footnoteSexyKen: but it says cisco all over it
04:07.32TrepaliumAt least it doesn't say Linksys.
04:07.46footnoteyou'd have to be an ex-employee to understand :)
04:07.59shmaltzqwerp, cisco is very good
04:08.05footnotenonono
04:08.08footnotecisco evil
04:08.24*** join/#asterisk Anjelikus (Anjelikus@ool-18bc68c9.dyn.optonline.net)
04:08.27shmaltzpolycom isn't that much cheaper than cisco
04:08.32Anjelikushi all...
04:08.39Anjelikusam I allowed to ask a stupid question?
04:08.42myconidno
04:09.09shmaltzAnjelikus, not if you ask if you are allowed, but go ahead, nobody will throw any stones at you
04:09.13QwellAnjelikus: If you do ask one, make sure you go all out
04:09.14ShanefulSexyKen: I think there is a digit timeout on the phones config
04:09.20Anjelikuslol
04:09.34SexyKenShaneful - there is but it's not the problem.
04:09.37shido6polycom is 195.00
04:09.39shido6ip500
04:09.44footnoteAnjelikus: correct answers cost extra
04:09.48Anjelikusok...so, I'm VERY new to the whole PBX thing...and I actually came across Asterisk@Home and used that to install onto an old PII that I have...
04:09.48TrepaliumThere are no stupid questions, but there are a lot of inquisitive idiots.
04:09.52shmaltzcisco 7960 is 235
04:10.06SexyKenshido6 -- where is my vanity toll-free #?
04:10.32Anjelikusnow, I'm not really sure what other hw I need to install to get this working on a phone....where can I get pointed in the right direction here in terms of finding hardware for this kind of project
04:10.33Anjelikus?
04:10.43SexyKen12/21/04 you said it'd take 7-10 business days....I think we're a couple days passed.
04:10.45shmaltzuse xlite
04:10.52shmaltzand some sip account
04:11.00shido6am I to guess your vanity number?
04:11.01shmaltzlookup the rest on voip-info.org
04:11.05*** join/#asterisk lhand (lhand@216.86.207.139)
04:11.09shmaltz~docs
04:11.10jbotDocumentation 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
04:11.11TrepaliumOkay, where is the Cisco 7960 being sold for 235.
04:11.25shmaltzAnjelikus, follow jbot's advise
04:11.29niZonare there any phones other than the cisco 7900 series that support XML services?
04:11.45mog_homehey anyone here run asterisk on bsd, or mac os x?
04:11.49shmaltzniZon, I think the polycoms do
04:12.02mog_homeis there any fix for the new threading stuff mark put in so that we can build it?
04:12.03SexyKenshido6 ->  You should have it memorized by now as far as I'm concerned.  18778964578
04:12.08mog_homeor should i just roll back a week?
04:12.09*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
04:12.15mog_homethis is in head
04:12.23shido6what happens when you ring it from the PSTN, SexyKen
04:12.36footnoteDoes Jim Dixon irc?
04:12.41DyOSi have my exten for my 800 through nufone set up like this exten => 86623310XX,1,Dial(SIP/666,20,tr) when i call that # it doesn't dial my ext 666 it just says person is not available please call your try again...any ideas on this someone please help
04:13.01SexyKenshido6 -> the same thing that happened 3 months ago. It says the number you are dialing is not real.
04:13.19shido612/21/2004 2:02 PM your number was active
04:13.20ManxPowerDyOS, Asterisk is not registering.
04:13.27*** part/#asterisk lhand (lhand@216.86.207.139)
04:13.28DyOSso what does that mean exactly?
04:13.31DyOShow do i go about fixign?
04:13.40SexyKenUh
04:13.41*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
04:13.44ManxPowerDyOS, Your register => in iax.conf is not correct.
04:13.50DyOSthanks
04:13.54SexyKenShido -> Now it says 'you have reached mci world wide conferencing services'
04:14.17SexyKenOh wron gnumber
04:14.17SexyKen18778964578
04:14.22SexyKen18778964678
04:14.38DyOShey i looked in my iax.conf file but other 800's through that same iax provider work perfectly fine....
04:14.41DyOSso i don't think it's that
04:14.44SexyKenwow -- it might work now.
04:14.45DyOSor they wouldn't work
04:14.51Anjelikusthanks...
04:15.09`SauronI wonder what happens if you call a 800 number through FWD or something
04:15.15`Saurondoes it pass on your FWD number?
04:15.47channanhi... anyone knows why we can't get FREE long distance to Mexico just like we did with Canada (ie. thru Vonage)? Is it monopoly thing?
04:15.57moonwickmexico has phones?
04:16.04channanthat's funny LOL
04:16.05footnoteTelmex state run, right?
04:16.11`Sauronmoonwick: they actually have a very well built out GSM network
04:16.32footnotehold the guacamole
04:16.38ManxPower`Sauron, countries with poor PSTN service frequently have good cell networks.
04:17.10niZoneverything hardwired in that country sucks
04:17.23`Sauronshrug, I didn't have a problem with their PSTN stuff either
04:17.32niZonthey don't even lock their distrobution boxes
04:17.55shido6SexyKen, hopefully you didnt send us a wrong number when you requested a status on the number.
04:18.03DyOSwhat i the proper syntax to forward an incoming call to a certain extension...
04:18.06techiemust be those burritos
04:18.31NukemizerAny AMP users here tonight ?
04:19.21SexyKen•shido6• I didn't send you a wrong number when I requested status you sent me back a wrong number.
04:21.00*** join/#asterisk lucca (~lucca@export.accela.net)
04:21.18shmaltzc ya all guys
04:21.20shmaltzgtg
04:21.42shmaltz~bye
04:21.43jbotcya
04:22.57footnote~cya
04:22.58jboti guess cya is Cover Your Ass
04:23.02footnotehaha
04:23.31footnoteshmaltz: monsters, under the bed.
04:23.33luccasage advice
04:24.09shmaltzlol
04:24.11channanNukemizer-I used it a little bit
04:24.52SexyKenAnyone know why Asterisk would get this when I attempt to login with my Polycom IP 600: Mar 8 21:18:52 NOTICE[5990]: rtp.c:317 process_rfc3389: RFC3389 support incomp
04:25.14NukemizerChannan, looking for a way to make Zap/3-1 ring to Zap/1-1    Telco in to House Wiring using my TDM22B card
04:25.21*** join/#asterisk Newbie___ (some@60.48.50.70)
04:26.02The_BallNukemizer, you mean when the phone connected to Zap/3-1 dials something?
04:26.09NukemizerChannan, only provisions for sip and IAX2 type devices.. and AMP likes to re write files
04:26.35NukemizerIncoming telco rings to Zap/3-1 for incoming calls
04:26.37channannukemizer-hmm.. not sure.. let me think about it.
04:26.44*** join/#asterisk setjmp (setjmp@ip-wv-66-190-126-051.charterwv.net)
04:27.50Nukemizersince I have only Analog house wiring I want to ring Zap/1-1 ( the station side) so my house phones will ring.. Over powered answering machine :)
04:28.17Nukemizerthen i can start to migrate to sip and IAXY
04:30.41NukemizerOr how about , could I make a call group with Zap/1-1 in it ? humm.. perhaps front end with auto attendant ? and time out to Zap/1-1 to stop or slow solicitation calls
04:31.04MocSexyKen, you have use the original polycom config ?
04:31.29*** part/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
04:31.44SexyKenMoc -> What do you mean?
04:32.13Moc3389 is the silence supression thing... it aint configured by default in the default polycom config
04:32.19setjmpAnyone have a suggestion for "best concept documentation" for Asterisk?
04:32.59Mocsome called kris made bad .config for asterisk available somewhere, you probably took them
04:33.08footnotesetjmp: source code!
04:33.17MocI mean config for polycom phone with *
04:33.18JerJer[mobile]moc: bad .config ?
04:33.19setjmpthere :)
04:33.32*** part/#asterisk Anjelikus (Anjelikus@ool-18bc68c9.dyn.optonline.net)
04:33.33SexyKenMoc -> I did download someone elses config files yea...
04:33.51Mocthere we go  ;)
04:34.06Moctake the ORIGINAL config, and only modify your Voicemail stuff and your register... and try that
04:34.17Moconce you get it working do a backup, THEN change stuff one by one ;)
04:34.17SexyKenMoc -> How do I do that?
04:35.16*** join/#asterisk goldenoldies (~goldenold@c-67-160-85-227.client.comcast.net)
04:35.21goldenoldieshi all
04:35.36*** join/#asterisk techie (gus@asterisk.horizonte.us)
04:36.05goldenoldiesanyone have any suggestions for presence servers to be used in conjunction with eyebeam/x-ten pro?
04:37.00JerJer[mobile]eh?
04:37.28Mocgoldenoldies, you mean make * a IM server É
04:37.34mikegrbJerJer[mobile]: I think he wants to give you a present!
04:37.37goldenoldieswell, no
04:37.45goldenoldiesthey have a presence agent option
04:38.01SexyKenMoc -> Did I mention that I have 2 other lines that login to different Asterisk servers properly?
04:38.02goldenoldiesSubscribe to contact list: $username$-list@$domain$
04:38.03Mocwell presence use SIMPLE message
04:38.11MocSexyKen: http://www.freedomphones.net/polycom/files/SoundPointIP_SIP_1_4_1.zip
04:38.35Moc* have BASIC LAMP option.. I'll see if more can be added... but SIP seem very bad as it is..
04:38.43GoshenJerJer[mobile]: I want to set the "forwarded call" flag when I forward a number from my * box, is there some way to do that with NuFone?
04:39.14GoshenJerJer[mobile]: so when the forwarded call comes in on my cell, it shows an arrow next to it showing it was forwarded
04:39.15SexyKenMoc -> Wouldn't the fact that my other Asterisk SIP accounts work fine from the same phone eliminate the config being the problem?
04:39.50Damin_MobileJerJer: You should be here.
04:40.16Mocit SHOULD.. it your voip provider ? your calling a X-ten phone ???
04:40.20Mocor you get that on the register ?
04:41.42Newbie___does anyone know a good and cheap voip provider for making IDD calls?
04:43.42MocIDD É
04:43.44Moc?
04:43.48SexyKenMoc -> I get that when I try to reg.
04:43.57Newbie___as in overseas calls, Moc
04:43.58footnoteNewbie___: good, fast, cheap. pick any two.
04:44.20Newbie___damn, life is full of decision
04:44.29MocNewbie___, cheapest I found is lookieloo, but they force a g729
04:44.47Newbie___ok i will go for good and fast
04:44.50Mocthen there is voiceconduits, but they doesn't seem to work rightnow.  Then you got nufone which work today ;)
04:45.11SexyKenWhat port does asterisk listen on?
04:45.18Mocdepend for what
04:45.21`SauronSo, did they ever figure out what the deal is with the new BV patch?
04:45.25SexyKenFor registrations
04:45.58Newbie___my calls will origin from asia though
04:47.09MocNewbie___, well search voip-info.org
04:47.17roamer323newbie - you'll have delay galore - originating from asia
04:47.33Newbie___www.nufone.com is not available
04:47.41Mocnufone.net
04:47.51Newbie___tks
04:47.52`SauronHurm.
04:47.56`SauronAnyone know where to find the patch?
04:48.23`SauronI guess I could see if it's in -head?
04:48.26`Saurons/?//
04:48.34roamer323Newbie - I'd do a ping or traceroute before to the host BEFORE laying down any $   - you're easily talking about 400 ms latency ... almost unusable
04:49.22footnoteno guarantees things won't change
04:49.55Newbie___i did ping to broadvoice, i got 200-300 ms
04:50.29goldenoldiesmost of our international customers have to use VPNs as a lot of the international telcos are blocking voip traffic
04:52.07roamer323newbie - that's pretty bad - also when you do a traceroute - count the number of routers in between - each and every one may (a) block traffic (b) drop your packets
04:52.46roamer323newbie - your best bet is to find a provider in your country ... hopefully voip is legal in your country :-)
04:53.36goldenoldiesI am doing 150 clear voice calls between Manila and Los Angeles over an IPLC E-1
04:54.05EightIs there a consensus on the best place to buy a single TDM400P card?
04:54.18`SauronEight: Check google
04:54.20`Sauronor froogle
04:54.31EightYa, got a bunch of different links out of that already.
04:54.37EightJust wondering if there was some particular store everybody liked.
04:54.39Newbie___tks roamer323
04:55.07QwellHow about the digium store?
04:55.52EightQwell - Sometimes manufacturer's stores are the *worst* place to get a product ;)
04:56.13Qwellsometimes, sure
04:56.23j0sorry for the basic question, but whats the difference between PSTN and POTS? even the dictionary i googled said they were both the same thing.. yet the asterisk handbook refers to them as needing different hardware to connect to
04:56.27Qwellbut when they're also going to support it...why not support them directly?
04:56.27EightBut I don't know in this case. Thus the Q =)
04:57.37EightQwell - well, some times a manufacturer doesn't even *like* to sell direct. It costs them more per unit sale than they recoup in the difference between wholesale and retail pricing.
05:03.15*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
05:03.40*** join/#asterisk tholo (~tholo@g4.sigmasoft.com)
05:04.23MocPOTS = old analog line that let you access the PSTN Network
05:04.49footnotePOT = programmer's supplies
05:04.58EightPO = Police Officer
05:05.10MocPOT = analog control
05:05.17footnoteFive-O!
05:05.21tholoPOTS == Plain Old Telephone Service
05:05.26EightMoc - bah, you broke the pattern.
05:05.28`SauronDum di dum
05:05.30Moc;)
05:05.37SexyKenAnyone know why Polycom would report 'Couldnt' register user: temporary unavailable'
05:05.45`Sauronthey need to fix the patch submission stuff for *
05:05.52Moc?
05:06.01`Sauronsucks having to apply patch after patch to * everytime you do a major upgrade
05:06.16footnote`Sauron: life's a bitch, then you die :)
05:06.22Moci why I stick to cvs head
05:06.29`SauronMoc: I am running -head
05:06.42techieor you die with a bitch
05:06.50footnoteHey, can you set up cvsup for asterisk a la freebsd-mirror?
05:06.58footnoteso you can keep a local repo?
05:07.12JunK-Ci love head.
05:07.13footnotethat helps a bit with merging, etc
05:07.32MocI do tar.bz2 ok asterisk head everynight now
05:07.32footnoteJunK-C: tis better to receive than give.
05:07.39footnoteMoc: bah
05:08.03*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
05:08.05`SauronI had to re-download -head, and re-patch
05:08.14MocI know I can do the same with cvs but I ratter have a full copy
05:08.24JunK-Cfootnote: huh?
05:08.26footnoteIf you had the repo locally....
05:08.32`Saurondoes anyone know if the latest BV patch made it into head?
05:08.35footnoteJunK-C: head :)
05:09.06Mocfootnote he is french ;)
05:09.09footnoteaha
05:09.18footnotehe should know what i mean then!
05:09.21*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
05:09.33JunK-Cnot really :)
05:09.35footnotehehe
05:09.38Mocyes he should, I did and Im french too ;)
05:09.57*** join/#asterisk santiago (~santiago@63.245.86.95)
05:10.36Mocgive a head = donner une pipe ;) ou receive a head = se faire faire une pipe ;)
05:10.38`Sauronhas anyone tried to send sms messages over sip?
05:11.16`SauronI'd imagine you'd need the (nonexistant) ss7 code to do any serious sms stuff
05:11.33JunK-Ci prefer say: suce ma chienne :)
05:11.38footnotehehe
05:11.54JunK-Cor: kessé tu criss, ta mere fait ca mieux que toé!
05:11.55JunK-C:P
05:11.56Mocbut Im polite ;)
05:12.06JunK-Cwhat, i am not ? :P
05:12.11Mocwell hehe
05:12.51footnoteok, i gotta go figure out this bias range resistor.
05:13.09footnoteOhm's Law is hard.
05:13.53`SauronHum.
05:13.58`SauronWell, that's broken.
05:14.00`SauronSigh.
05:14.04setjmpnot really.
05:14.09setjmpohms law that is.
05:14.18setjmpat least the math :)
05:14.23footnotesetjmp: I'm a burnout :)
05:14.29setjmpbeter then the deeper reactance calculations :-D
05:14.36footnotevoodoo!
05:15.09setjmpKept missing the Amateur Extra Class License exam here by 1-2 questions back in the late 80s.
05:15.13footnotesetjmp: can i borrow some crow's eyes?
05:15.21setjmpgot the Advanced within 2 months.
05:15.26footnoteshopping tomorrow
05:15.27setjmpNever the extra though, hehe.
05:15.30JerJer[mobile]73 de n8twj
05:15.40setjmp:)
05:15.49setjmpde ki9kh
05:15.54footnotesetjmp: amateur radio is what led me into computers.
05:15.55setjmppreviously ke2nm
05:15.56footnote#$#@$
05:16.13setjmpCB and computers got me into amateur radio, hehe
05:16.22footnotehaha
05:16.22setjmpWish I was a lot more active though.
05:16.28footnoteI never got my ticket
05:16.35setjmpWhy not?
05:16.49JerJer[mobile]i've got a 2m and 70cm repeaters fired up back home
05:16.49*** join/#asterisk walnuck (~walnucky@modemcable106.82-200-24.mc.videotron.ca)
05:16.49footnotelost interest after learning 8080 assembly
05:16.51walnuckhi
05:17.01SexyKenAnyone know what the FUCK this means
05:17.01SexyKen|4|00|Registration failed User: proggod, Error Code:480 Temporarily not available
05:17.08SexyKenThat's an error from the logfile
05:17.08MocJerJer[mobile] you got my msg about the 1800# ?
05:17.13JerJer[mobile]no
05:17.19setjmpahh. 8080 wasn't that bad. My all time favs were 6800 and 68000
05:17.21setjmpand vax
05:17.25footnoteyeah
05:17.32footnote68K >*
05:17.35walnuckanyone using fedora?
05:17.37setjmpThe times have changed much since back then
05:17.47footnoteall i ever wanted out of life was a 3GHz 68010
05:18.11Eightwalnuck: I am, but I doubt I'd be much help. I've just now installed Asterisk.
05:18.17setjmpI was satisfied with a 8Mhz 68000 on my old Amigas. Did well.
05:18.25`SauronSigh.
05:18.27footnotewalnuck: White Box Linux!
05:18.54`Sauronhum di dum
05:18.59setjmpToo bad the 6800 didn't get popular till so late in the 8 bit days. It really had a lot of good stuff beyond others (65xx), and made a lot more sense then others (8080/z80/etc)
05:19.14footnotethe 6809 came around at just the wrong time
05:19.16setjmpabout the only thing the coco's had going for them, hehe
05:19.19*** join/#asterisk paulc (~paulc@251.134.218.209.transedge.com)
05:19.20footnoteit was a neat processor
05:19.40footnotegreat for forth
05:20.11setjmpStill have an amiga 500, though I haven't powered it up in a long time now.
05:20.20footnoteguru meditations
05:20.51setjmpI really need to get my hamshack more then "EMPTY" now :-D
05:21.27setjmpBack in the Packet Radio earlier days, and BBS times :)
05:21.50*** join/#asterisk mh- (~mh@202.5.145.13)
05:22.05footnoteI had a Henry Tempo One
05:22.13footnotelast rig i had
05:22.14*** join/#asterisk alakdan (~alakdan@210.213.173.63)
05:22.14setjmpI ported STadel (customized version of Citadel for the Atari ST) to the amiga. Then began rewriting it and rbbs in m68k, hehe
05:22.25setjmphow old?
05:22.27footnote43
05:22.34setjmpjust turned 37 here last week
05:22.44setjmpmy first born turns 18 next week :-D
05:22.48footnotenever trust anyone under 30
05:23.12par<--34
05:23.16footnoteI got my first grandson this weekend :)
05:23.19footnotewoop
05:23.22alakdanhello, is it possible in an agi script to wait for say 3 secs before streaming the file?
05:23.22setjmpUsed to think never trust anyone over 30, till I left NY, hehe. Been in West Virginia a couple years now, and seem to have had a change of heart though, hehe
05:23.39footnoteWhat part of WV?
05:23.46setjmpBeckley now
05:23.57setjmpThough was out in Wyoming County till last July
05:24.01footnoteI have friends in Charlottesville
05:24.14setjmpHmm, trying to remember where that is.. HEard of it for sure.
05:24.20footnoteSupposed to go visit in a couple weeks
05:24.34setjmpThat closer to charleston?
05:24.36*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
05:24.36*** mode/#asterisk [+o bkw_] by ChanServ
05:24.47footnotedunno, haven't been yet or looked it up
05:25.05setjmpI'm like an hour~ to virginia or kentucky here.
05:25.05Mochi bkw_
05:25.24footnoteI live in the southern stretches of appalachia
05:25.34setjmpWhereabouts?
05:25.35footnotewell, blueridge actually
05:25.40setjmpKY eh?
05:25.44footnoteDawsonville Georgia
05:26.01footnote(home of Bill Elliot!)
05:26.07setjmpActually, technically they run like from southern NY all the way to Missippii if memory serves me..
05:26.10setjmpCool.
05:26.29footnoteIt's a NASCAR town, museum is here
05:26.35footnote"Thunder Road"
05:27.38footnoteok, i gotta go play guitar.  nite
05:28.05bkw_thats hot!!
05:28.07setjmpahh
05:28.11setjmpwish I had time for mine, hehe
05:28.21footnoteI've been building tube amps
05:28.24setjmpOle Ibanez RX-20 (nothin fancy)
05:28.30Mocbkw_, you saw some french canadian ? ;)
05:28.32setjmpI need to be doing that too :)
05:28.37bkw_Moc no why?
05:28.39footnoteI've got a Ford & a Chevy!
05:28.39jetssup gurls
05:28.44setjmpNo $$ for the marshalls I'd want
05:28.45bkw_I watched harley and taylor
05:28.48Mocbkw_, my collegue is at von
05:28.51setjmpAhh.. Confused eh?
05:28.51footnote(Les Paul & Strat)
05:28.55JunK-Cmoc: on SJ? they should be lost :P
05:29.15setjmpWell catchya another time.. Jam well brotha..
05:29.57jetswhat's up bitches
05:36.35`SauronHum
05:36.42`Sauronthe BV patch didn't make it into HEAD today
05:36.43`SauronBummer
05:37.05ta[i]ntedwhat BV patch
05:37.22`SauronThe new INVITE authentication patch
05:37.31`Sauronthey changed some stuff on the fly w/o telling anyone
05:37.45ta[i]ntedi thought u just had to add authuser = <accountnum>
05:39.04`Sauronapparently not
05:40.09Eightso does that break * with BV for now?
05:40.56*** join/#asterisk cero64 (ruiner@fantab.ulo.us)
05:41.08*** join/#asterisk file (~file@251.134.218.209.transedge.com)
05:41.46`SauronEight: unless you apply the patch, yes
05:42.17ta[i]ntedi don't have a patch
05:42.19ta[i]ntedand BV works fine
05:43.05`Sauronfor incoming our outbound calls?
05:43.14`Saurons/our/or
05:43.18ta[i]ntedoutgoing
05:43.27`SauronHum.
05:43.43`Sauronmine's giving the auth on reinvite error that they claim the patch fixes
05:45.16`Sauronsomeone was going to clean it up and submit it to -head
05:45.19`Sauronbut nothing yet
05:45.25ta[i]nteddid u try adding authuser
05:45.29`SauronI did
05:45.35`Sauronis it your account id, or your phone number?
05:45.37ta[i]ntedwhich proxy u using
05:45.41newpersI wonder if I could install asterisk on my dell axim x5
05:45.43ta[i]ntedphone number
05:45.53Mocnewpers, funny
05:45.57newpers:)
05:46.06Mocif you got linux on it, you could probably
05:46.07newpersI have no use for it
05:46.13ta[i]ntedgive it to me
05:46.16newpersyeah, that was the plan
05:46.20ta[i]ntedi need a pda
05:46.34newpersI don't have a wireless router, so I really don't use it
05:46.52`SauronMar  8 23:46:18 NOTICE[5773]: chan_sip.c:7930 handle_response: Failed to authenticate on INVITE to '"SPA 1" <sip:....
05:46.57`SauronIs what I get on the console
05:47.19`Sauronlet's see what this patch does
05:47.40ta[i]nted`Sauron what * version
05:48.21Himekoanyone using an aastra 9417cw on an ata?
05:49.09Newbie___guys, some voip offer both SIP and IAX2 to terminate calls, which is prefer ? IAX2 or SIP ?
05:49.17Himekomy MWI lamp doesn't clear properly, yet other phones on the same port clear
05:49.36`Saurontainted: head as of 10 minutes ago
05:49.38mikegrbprobably the phone
05:50.16Himekoprobably, but the other phones are not the same model, so i am interested if anyone has the same model
05:50.42*** join/#asterisk p0lar (~p0lar@69.111.140.189)
05:50.56p0larAnyone at VON today?
05:51.02Mocmy friend were
05:51.07filep0lar: yes
05:51.33p0larWhat's the official standing of sysmaster using * anyway?
05:51.42*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
05:52.11p0larI point-blank asked Krishnan Jayaraman, Regional Sales Manager, about it and he said he had never heard such nonsense
05:52.20*** join/#asterisk tholo (~tholo@g4.sigmasoft.com)
05:54.10p0larAs well, I almost think they should rename Von2005 to SIP-users2005.  I was told all day long that H323 was a dead protocol.
05:54.25JerJer[mobile]p0lar:  he's fucking stupid gthen
05:54.40p0larKrish?
05:54.40JerJer[mobile]we have absolute undenyable prof
05:54.43JerJer[mobile]proof
05:54.51p0larI asked him point-blank, man
05:55.04JerJer[mobile]and i have taken apart 4 of their boxes
05:55.08p0larI also asked him how many channels of g723<->g729 he could do simultaneously
05:55.15JerJer[mobile]all of them has asterisk running on them
05:55.15`SauronUghrrgh.
05:55.17p0larIt was in the thousands of channels
05:55.21p0larI walked away.
05:55.26myconidlol
05:55.40mikegrbhaw haw
05:55.51mikegrbhe is a liar
05:55.58JerJer[mobile]i should have gone  i guess - i could have brought a sysmaster box out to him and shown him
05:56.00mh-guys, what's a stable cheap solution for a interface card?
05:56.08filethey had no hardware to show either
05:56.14fileconspiracy - I think so.
05:56.16p0larNope.. I was looking for it too
05:56.24mh-..an interface..
05:56.30JerJer[mobile]they have been in discussion with Digium
05:56.37p0larI *thought* about approaching mark about it, but decided it wasn't worth it. *shrug*
05:56.45JerJer[mobile]so why would they even discuss anything with DIgium if they weren't usng asterisk?
05:57.02JerJer[mobile]i think its time for another round of noise
05:57.21p0larI can get an official 'quote' from him tomorrow. ;)
05:57.32p0larI have no problem turning up the heat in that booth...
05:57.35JerJer[mobile]find Marcello from voxilla
05:57.41p0larnoted.
05:57.46filebkw, twisted, and I went around with someone from Digium
05:58.15filebut we didn't go to sysmaster ;)
05:58.21p0laranyone else I should visit?
05:58.25Mocfile, damn I wish I were there ..
05:58.32`Sauronyou at VON?
05:58.37fileAT&T is running Asterisk, Grandstream is too, ah who else... I forget
05:58.41JerJer[mobile]damnit i should have gone - got better things going on though
05:58.42fileswing by the Digium booth and I'll be there
05:58.55fileso will zoa, bkw, twisted, whoever...
05:58.56*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr)
05:59.02p0larYou're at the Digium booth?
05:59.04fileyes
05:59.08_Vilefile
05:59.10_Vileur fired
05:59.18fileI think I've declared tomorrow white golf shirt and black pants day
05:59.22JerJer[mobile]file: i bet the pavilion is killer
05:59.24file_Vile: ha
05:59.29Mocfile PICTURE !!
05:59.37fileJerJer[mobile]: we're the most active booth I'd say
05:59.39_Viledont make me pull out the stick
05:59.52fileJerJer[mobile]: did you get the story about the equipment?
05:59.54p0larQuite possibly
06:00.02p0larMarc Spencer is harder to get to than pulver
06:00.11filep0lar: really?
06:00.13_Vilemarks is easy to get to
06:00.14JerJer[mobile]10g's for 3 days is just stupid
06:00.16*** join/#asterisk casterman (~casterman@63.240.97-84.rev.gaoland.net)
06:00.19fileI had breakfast with Pulver, and Mark is right in front of me :)
06:00.20JerJer[mobile]file: equipment?
06:00.22_Vilejust ask cvs
06:00.25fileJerJer[mobile]: it didn't show up
06:00.28p0larYou're connected.. heh
06:00.31_Vilepulver is simple, call him
06:00.33*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
06:00.39JerJer[mobile]yes just dial 1 on fwd  :P
06:00.44Nuggethe must not use his own wisip phone, then.  :)
06:00.51fileJerJer[mobile]: went to Fry's Sunday night, got the hardware - made the dialplan/network stuff Monday morning
06:01.00_Vilebut I know neither of them
06:01.01p0larJer: I just went for the exhibits, most of the rest of it wasn't that interesting to me OTHER than the 'Sip Trunking' bit...
06:01.02JerJer[mobile]file:  uggg  damn i could have loaded up the twin with gear and flown out
06:01.19fileJerJer[mobile]: the Digium staff brought channel banks, some SIP phones and an MGCP phone
06:01.20p0larNugget: heh
06:01.25JerJer[mobile]someone needs to be fired for that fuckup
06:01.38_Vilemmm
06:01.40_VileFILE
06:01.42fileJerJer[mobile]: tried to get GR303 going... but uh, it didn't work
06:01.42_Vileur FIRED
06:01.48`Sauronnugglet
06:02.08p0larI did ask MattF (unsure of his alias here) about DSP cards today
06:02.18_Viledsp on what?
06:02.23JerJer[mobile]daughter
06:02.24p0larexactly.
06:02.29filep0lar: let's see if he remembers you!
06:02.41_Vileintel chassis or are u thinking pc?
06:02.48p0larI doubt he took many questions on it.
06:02.58_Vilehe should've
06:02.59filehe's talking about... stuff... right now
06:03.03filewe're working on the bug tracker
06:03.06p0larheh
06:03.07_Vilefile get him over here
06:04.00*** join/#asterisk paulc (~paulc@251.134.218.209.transedge.com)
06:04.12_Vilehi paul
06:04.15p0larI was looking for codec conversion offloading, he met me half-way with echo-cancellation.. heh
06:04.19paulcHey hey :)
06:04.27fileyeah Malcolm has the yummy echo can stuff
06:04.29_Vilenonono need both offloaded to hardware
06:04.40filep0lar: he remembers you
06:04.50SexyKenIs there anyway to choose the codec polycom phones use.
06:04.54_Vilefile what did he say about dsp HW?
06:05.05fileMatt thought he was soooooo hot
06:05.05p0larNo surprise, there can't be loads of people wanting to do quad E1s using g729.. :(
06:05.12_Vileoh god
06:05.16_Vileanother bkw
06:05.46_Vileuseless
06:05.50filep0lar: PSST, zoa says you can do it on a dual 3.6GHz xeon - you can just do it
06:06.01_Vilew/ 410s
06:06.05_Vilekeep in mind
06:06.12_Vile64 bus
06:06.15_Viledoable
06:06.32_Vilenot w/ 405s or the previous, I doubt it w/o dropped calls
06:06.47filezoa is getting angry
06:06.48p0larConveniently, I have one of those at the office. ;)
06:06.51*** join/#asterisk Vector_ (~Vector@h-67-101-140-109.hstqtx02.covad.net)
06:06.58_Viletell'em to chat if he wants to whine then
06:07.18p0larIt's no joke when I say I've got four of those TE4XX cards on my desk as well.. :9
06:07.57p0larbut sending a dual xeon to africa, middle east, carribean is not my preferred option.. :(
06:08.07filethey're the same cards
06:08.07_Vileto tell you the truth, I've got an amd 1800 running 4 ports of T1s on one of the original cards built by zaptel
06:08.15_Vileand it works
06:08.16filewell, almost
06:08.20_Vilebut I get dropped calls
06:08.30_Vileoccasionally
06:08.34p0larquad T1s with G729??
06:08.42_Vilenono, sip ulaw
06:08.45p0larheh
06:09.01p0laryeah, bandwidth costs us > $14kusd/month in some places, I need every bit I can get
06:09.36_Vilehm
06:09.45_VileI'd go 8 ports on a dell 1850
06:09.49_Vileper box
06:09.53_Vileusing ser as a front runner
06:09.59WilliamKtry libc yet p0lar?
06:10.00WilliamK=)
06:10.02_Vileall 410
06:10.16_Vileand call it a day
06:10.19_Vile$3200 total
06:10.22_Vileper 8 ports
06:10.45_Vilethat's $16.66 NRC per port
06:11.15_Vile$1200 410s on eBay
06:11.18p0larI've got a very specific need and asterisk is only a dsp away from fulfilling it nicely.. until then, I'm stuck using cisco, quintum, et al...
06:11.29_Vile$1200 1850 Dell 1U in the catalog
06:11.50_Vileyeah
06:11.56p0larI use my 410s as paperweights right now.. <duck>
06:11.58_Vile* needs dsp's, echo can HW
06:12.17_Vileand a chassis, maybe minipci
06:12.27filepaulc: how is it on the other side of the road?
06:12.28_Vile2 per box is OK though
06:12.33fileha, room
06:12.35_Vile8 T1s for $3600
06:12.42_Vileat 1U
06:12.45_Vilew/ GR303 support
06:12.50p0larHonestly, if you could offload echo + codec cancellation to a daughter card (or cards, woohoo!) many of the vendors in that room would probably NOT be happy. ;)
06:13.01filep0lar: there's other ideas
06:13.05*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
06:13.33*** join/#asterisk clive- (~pirch@rndf-146-55-139.telkomadsl.co.za)
06:13.35p0larI'm listening..
06:13.36paulcfile: it's not bad.. ish.. my laptop's making my lap hot.. and I don't think the aircon works in here
06:13.46_Vilefile, dont make me fire u again
06:13.55opus___p0lar -- fpga
06:14.51opus___or,  gpgpu
06:14.57SexyKenWhy would this happen with one phone, but not another connected to the same account: Mar 8 23:10:45 NOTICE[6257]: chan_sip.c:7899 handle_request: Unable to create/find channel
06:15.14puppetHave anyone tried replacing there cellphones voicebox with an own? Like forwarding cellphone to your own number and check if call is from cellphone then send to voicebox?
06:15.38p0laropus: perhaps, but I'm the end-user, I reserve the right to whine and complain until Digium satisfies my minority requests!!!
06:15.40_VileI would want to look at * as a long term switch solution to beat the fuck out of nortel and at&t, meta, tekelec, etc... providing finally an open platform, that's my very simple dream
06:15.44p0larhehehe
06:16.00opus___puppet couldn't you just do a lookup of the number and tell if it is from a cellphone from the web and then send to voicemail?
06:16.00p0lartekelec told me that I wasn't using H323.. haha
06:16.14_Vilethey're basically morons reading scripts
06:16.17*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:16.24puppetopus___: nope, when it redirects it keeps first number for some reason
06:16.35opus___hmmm
06:16.37puppetopus___: i thought that would work to but callerid == first number
06:16.47_Vileprobably typos in everything
06:16.51opus___can you set some kind of variable in a script
06:17.01puppetand its kinda stupid that people that call my cellphone get an option to connect to my homephone ;p
06:17.08p0larI told them they could argue with my CDRs and revenue model
06:17.15opus___puppet -- wuh
06:17.15puppet== expensive on cellphonebilll
06:17.16opus___??
06:17.53p0laranyway, I got that all day long at all the booths
06:18.08_Vileto avoid the ilec annoying UNE removals (it'll all be gone one day), we just signed w/ another telco today to offload PRI cost
06:18.35_VileI don't feel like I'm in that much of a crunch now
06:18.44puppetopus___: I got max debug not, and no where is my cellphone number, its just the first number
06:19.42p0larI'm going to stop back by the * booth tomorrow
06:20.00p0larI noticed they had a WRT54GS there, heh... I'm using one here in the hotel as my VPN endpoint. =D
06:20.00_Vilefile, u heard anything about * HW?
06:20.16_Vilebeyond the PC?
06:20.34p0larWRT54GP2 FXS compatibility?
06:21.20p0larLoads of SBC people at the exhibit as well
06:21.42_Vilethat scares me
06:22.00_VileLD companies buying up local companies buying up vendors
06:22.05_Vilechewing the market up
06:22.10_Vileblah
06:22.15*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
06:23.02p0larI bugged Quintum today about their 'packet saver' technology (sic) today as well, heheh
06:23.15_VileI don't know what these moron regulators think that they're doing by giving back LD<->Local allowances and getting rid of UEN
06:23.20JerJer[mobile]lol packet saver
06:23.22p0larTold them about asterisk's IAX2 trunking -- the guy was like, "Really? -- I had no idea, I thought we were the only ones doing that."
06:23.24_Viledo they actually believe that this is good for the consumer?
06:23.31clive-polar they just do trunking on the rtp
06:23.34_Viles/UEN/UNE
06:23.44p0larYeah, I've seen it before
06:23.51p0larbefore the exhibit
06:24.00_Vilenoone cares, I love it.
06:24.01p0larI just had to ask to see what their thoughts were about *...
06:24.05_Vilebbiaf, smoke
06:24.36clive-polar , we are still waiting for trunking in * to work with jitter buffering, its still busted
06:24.54clive-and thats onlt iax, sip is still to follow
06:25.15p0larIs SIP trunking myth or soon-to-be-vapour-ware?
06:25.20opus___what is a WRT54GP2... i have a GS
06:25.28p0larthe P2 has 2 fxs ports on it
06:25.42p0larbut is supposedly 'locked' to vonage (only in the default software I imagine)
06:25.47JerJer[mobile]but the p2 is not based on the broadcom chipset
06:25.54p0larah, I didn't know that
06:25.55opus___you can't ssh to it can you?
06:26.04p0larno idea
06:26.18opus___i looked at the fcc images but didn't see the arm processor, didn't think so
06:26.30p0larI've not followed any discussion on it yet, kind of waiting for someone else to mess with it first due to lack of time. :(
06:26.37opus___GS runs linux 2.4
06:26.41clive-sip trunking =rtp trunking, and I do not belive that is a hapenning thing in asterisk, we are waiting for jitter buffering in sip :)
06:27.12PTG123anyone ever seen this error
06:27.24PTG123handle_request Unabel to create/find channel
06:27.29opus___yeah
06:27.33p0larI looked for a lower-level way to aggregate packets just to save the raw pps persecution that occurs on some routers/vpn links we use. :(
06:27.40opus___but it still works.. or your nat is fucked
06:27.40p0larno dice
06:28.18JerJer[mobile]iax2 trunking should lower pps pretty good
06:28.33JerJer[mobile]the next step would be to stack multiple voice frames per packet
06:28.58clive-yup, like trunk time = 120 ms
06:28.59opus___videolan has broadcast packet capabilites already implemented...
06:29.09opus___dunno if rtp
06:29.13p0larJer; yeah.. that piece is ok with me
06:29.16tholomsg JerJer[mobile] I hate to have to ask this again, but...  What was the AT&T site to check 800# availability?
06:29.35tholoErr... :)
06:29.53JerJer[mobile]http://www.tinyurl.com/i20o
06:30.08tholoThanks.
06:30.57JerJer[mobile]oh waiter - check please
06:31.06opus___jerjer what kind of mobile are you on
06:31.22JerJer[mobile]gprs
06:31.27opus___laptop?
06:31.30JerJer[mobile]yep
06:31.48JerJer[mobile]abusing the hell out of AT&Ts 'unlmited' gprs  :P
06:31.53opus___what kind of latency do you get
06:31.57opus___voicestream > 1800ms !
06:32.07opus___sprint is around ~200 for me...
06:32.12*** join/#asterisk jmhunter (~jmhunter@64.77.199.223)
06:32.12*** mode/#asterisk [+o jmhunter] by ChanServ
06:32.23JerJer[mobile]since i'm in an EDGE area its acceptable, most of the time
06:32.39p0larwhat kind of pps can you do on that jer?
06:32.55JerJer[mobile]voip is quite latent
06:33.09*** join/#asterisk Tarox (someone@pD9E7BADD.dip.t-dialin.net)
06:33.14JerJer[mobile]but speex on its lowest settings does work
06:33.21opus___which phone do you use?
06:33.22JerJer[mobile]when i'm in EDGE
06:33.26JerJer[mobile]6620
06:33.56JerJer[mobile]right now first hop latency is 221ms
06:34.07Moc~RSP
06:34.10opus___nice
06:34.15Mocwhat does rsp mean in the voip world ?
06:34.18JerJer[mobile]and 231ms to switch-1
06:34.21JerJer[mobile]:P
06:34.24opus___curious, whats EDGE?
06:34.33p0larI'm waiting/watching for a good edge-based smartphone/pda-type device to come out...
06:34.42JerJer[mobile]att's higher speed gprs crap
06:34.56Himekog2?
06:35.21JerJer[mobile]i've heard g2.5 but no clue
06:36.07opus___recently i purchased a mpx200 smartphone for gprs, ... but voicestream has mad latency... 2seconds
06:36.20JerJer[mobile]damn
06:36.33p0larew
06:36.38JerJer[mobile]ok - must sleep, for a change
06:36.40Himekoi am thinkign of getting a 6630
06:36.44p0larOk, so who's going to let me bug them tomorrow at the * booth? and how do I identify them?
06:36.48Himekoi have a 3600 right now
06:37.13*** part/#asterisk alakdan (~alakdan@210.213.173.63)
06:37.58*** part/#asterisk JerJer[mobile] (~jj@65.173.197.109)
06:38.05p0larAnd better yet, who wants to walk over to the sysmaster booth to ask more questions?
06:38.17opus___so is anybody hacking the wrt  54gp2?
06:38.27opus___i want to get one now
06:38.41opus___... i guess it could be my fwd gateway:)
06:38.44p0larNot yet...
06:38.52p0larI'd buy one if I knew I wasn't supporting vonage
06:39.01opus___hah y?
06:39.13p0laralthough, we do sell them traffic.. hehe
06:39.14p0lard'oh
06:39.28opus___rate limit'em to hell
06:39.37p0larrofl
06:40.16opus___what conference are you going to?
06:40.32p0larSpring Von 2005?
06:40.50p0larwww.von.com
06:40.51filep0lar: I'm Josh, Josh Colp... you'll find me
06:40.57filethere's Josh Roberson too... that's twisted
06:41.06p0larThat's twisted like.. not right or..
06:41.11p0lartwisted as in.. aka: twisted
06:41.23filehim
06:41.27p0lark
06:41.51paulcStormOffInAStrop(tm)
06:42.17opus___whoah nice con
06:42.20BoRiS<= Josh
06:42.24fileha
06:42.54paulcTouchyFeelyBoRiS(tm)
06:44.13paulcOI! LESS OF THAT!
06:44.59BoRiSlol
06:46.09ta[i]ntedu guys flirting?
06:46.18paulcThat's the way you like it ;-)
06:46.32ta[i]ntedwas about to ask for an INVITE
06:47.10BoRiSfile: "Once a whore?"
06:47.35paulcALWAYS A WHORE
06:47.40paulcand see my blog for a relevant story about that
06:47.48BoRiSurl?
06:49.30jmhunterwhats ur blogs url, i always lose it
06:50.49tuxinator_linuxAny of you at VON right now?
06:50.56jmhunteri was
06:51.01jmhunternow im at home
06:51.03jmhunterpaulc is
06:51.06jmhuntertwisted is
06:51.08BoRiSfile is
06:51.10BoRiSbkw is
06:51.10jmhunterbkw is
06:51.21tuxinator_linuxI am on my way tomorrow night
06:51.31paulchttp://myworld.q2u.net/ = my blog
06:51.41paulctux: coming for the party eh?
06:51.44tuxinator_linuxyep
06:51.53tuxinator_linuxI am staying at the marriot
06:52.07tuxinator_linuxI will also be at Meet *
06:52.33paulcis that Friday?
06:52.37tuxinator_linuxWho do I meet up with when I get to the center Thursday morning
06:52.46tuxinator_linuxyes, Meet * is Friday
06:52.54*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
06:53.24paulcI'm in San Fran then for work :-s
06:53.30paulcand home on Saturday - YAY!
06:53.58tuxinator_linuxAny of you going to Meet *?
06:54.17p0larI've got to leave tomorrow night to be back in the office by thursday morning. :(
06:54.31p0larpulling the all-niighter in airports.. feh
06:54.36tuxinator_linuxeww
06:54.46filekram says 'hi'
06:54.55jmhunterVONners, what was everyones feel on Michael powell?
06:54.55filejmhunter: kram says hi to you espically
06:55.01filejmhunter: and inquires on how you got back so fast
06:55.03paulcp0lar you're at VON right now?
06:55.06p0larYes
06:55.06paulcMichael Powell seems cool..
06:55.13p0larwell.. I'm in the hotel right now, heh
06:55.18filewhich hotel?
06:55.37jmhuntermichael powell is a kok suker
06:55.54p0larhampton on old tully rd.
06:55.55paulcbecause...?
06:55.57fileah
06:56.08p0larNo, I don't know WHY I got booked here.
06:56.24filewe're spread out across a few hotels
06:56.35p0larmust've been tight downtown I guess
06:56.45filewell it's all close together so all is well
06:57.05jmhunterlol old tully
06:57.06p0larYeah, 10 minute cab ride from here
06:57.06jmhunterfuck
06:57.08jmhunterthats far
06:57.10*** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com)
06:57.15tuxinator_linuxDon't get all partied out with out me.  I'm coming... As soon as I can.
06:57.24p0larI debated getting a rental car so I could go pick up some stuff from soekris engineering in Santa Cruz (sp)
06:57.28filewe can just walk to the conference
06:57.32filein fact, we do! every day!
06:57.35p0larheh
06:57.40file:)
06:57.49BoRiSGood old Soekris.
06:58.04jmhuntersanta cruz is spelled correctly
06:58.15p0larI wasn't 100% sure that's where they are..haha
06:58.18p0larwithout looking it up again
06:58.23ClayReiche123Does anyone know why I can't create an IAX trunk on stable?
06:58.45clive-clay as far as I know its still busted
06:58.46ClayReiche123Are there any known issues with that on stable?
06:58.52ClayReiche123ahhhh
06:59.05ClayReiche123Thank you Clive.
06:59.13opus___clay -- dunno, but i did see a -O6 (!) fly by when i built the latest version
06:59.19clive-clay there are some patches in the mantis
06:59.20opus___try modifying it to -O3
06:59.25p0larI honestly only use their crypto cards (HiFn 7955s) but linux support for those is vapourish
06:59.26ClayReiche123I can stop pulling my hair out now.
06:59.31p0larbsd support is perfect
07:00.04clive-clay are you trying trunking together with jitter buffeirng?
07:00.37tuxinator_linuxjitter buffering, sounds like a new dance
07:00.43fileour air conditioning is broken :(
07:00.47tuxinator_linuxlike this one http://www.theembassyvfx.com/citroen.html
07:00.48BoRiSYou broke it!
07:00.51jmhunteri know that room was fucking hot
07:01.05jmhunterfelt like u guys had an orgy just before i showed
07:01.10ClayReiche123Clive I don't think so. I'm just trying to pass a call to a central voicemail server.
07:01.26ClayReiche123Clive: it worked fine when my vmail server was HEAD.
07:01.29jmhunterthat would explain the plastic sheeting on the ground
07:01.34p0larhaha
07:01.51p0larI had to turn on the AC when I first came in just to feel like I was at home. :D
07:02.13opus___where are you guys at
07:02.14p0larthe 70* weather was killer today.. we won't see that in MTL until July
07:02.17p0larSan Jose
07:02.22opus___oh yeah
07:02.30jmhunterwhere u from p0lar
07:02.33opus___its summertime here to in portland
07:02.36opus___:)
07:02.41p0larjmhunter: Montreal, QC
07:02.44SexyKenI live in san Mateo County
07:02.55jmhunterya we get a sneak peek of summer every march, then it comes back in may... plus it is an el nino year
07:02.55ClayReiche123hehehe... kinda chilly today in Florida!
07:02.58jmhunterahhh a canuck
07:03.06jmhuntera true canuck
07:03.12p0larnot true.. :D
07:03.15jmhunterlol
07:03.21p0larI'm an alien, heh
07:03.39jmhunterwell canuck usu refers to those who come from a la qubec or whatever u say
07:03.43*** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com)
07:03.48jmhunteri c
07:03.52p0larah, quebecois, yes...
07:03.54p0larthey're of their own breed
07:04.12p0larI struggle, but I can swear efficiently in quebecois (dirty french)
07:04.23tuxinator_linuxThe Fresh Canadians are very differned from the English Canadians
07:04.28jmhunterhave u seen canadian bacon?
07:04.34tuxinator_linuxum, French
07:04.36opus___hmm
07:04.38opus___bacon
07:04.41jmhunterhence Moc... u listening Moc?
07:04.47p0lartux: you got that right
07:05.10tuxinator_linuxThe English ones are easier to get along with
07:05.13jmhunterhave u seen the michael moore movie "canadian bacon" w/ john candy
07:05.19ClayReiche123clive: can you point me to some documentation on how to ptch/use the mantis? never heard of it before...
07:05.20tuxinator_linuxin my experience so far
07:05.44tuxinator_linuxMicheal Moore is a fat slob
07:05.55tuxinator_linuxhow was the movie?
07:05.56p0lartux: agreed.. I've got a heinous tale about trying to purchase a $7.5k KVM from Avocent here
07:05.58opus___gppd
07:06.06opus___funny movie
07:06.11jmhunterp0lar where u from?
07:06.13p0larMoney in hand and they wouldn't buy it because we hurt their feelings
07:06.15opus___michael moresuqs
07:06.16p0laror something
07:06.18BoRiSAnd John Candy is dead :(
07:06.23p0larjm: I'm a US citizen
07:06.27moonwickmichael moore is teh suck.
07:06.35jmhunterfrom where
07:06.49p0larKind of all over. =(
07:06.50tuxinator_linuxp0lar: Silly people, I would take your money
07:06.58tuxinator_linuxArizona here
07:07.04p0larBut I live/work in Qc
07:07.08p0larfor the last 3 years
07:07.25jmhuntercanadian bacon is a comedy, not a documentary
07:07.35jmhunterits about a made up war with canada
07:07.39p0lartux: common misconception, heh
07:07.42jmhuntercirca 1995
07:07.44tuxinator_linuxNone of this movies are documentaries
07:07.56tuxinator_linuxhis
07:07.59p0larjm: I've not seen it, but I may put it on the list
07:08.39p0larwhoa.. little caeseres!
07:08.46tuxinator_linuxpizza?
07:08.54p0larwow, I've not seen them in.. uh.. years
07:09.18tuxinator_linuxI just ate, so full
07:09.39tuxinator_linuxCalzone from "Streets of New York"
07:10.03jmhunterdan akyroyd is an ontaro trooper... and he stops john candy who is driving a truck with slanderous canadian words, like fuck canada, etc... They try explaining that they think some teens must have tagged their truck.. dan akyroyd shuts them up and says np.. i stopped you because your slurs do not comply with la quebecois... and makes them write all the slurs in french, in addition to english
07:10.20opus___i just watched Colossus: The Forbin Project (1970)
07:10.43tuxinator_linuxYou know what?  I feel like I live in this channel.
07:10.51jmhunteru probably do
07:10.52p0larhahahaha..
07:10.54BoRiSYou do
07:10.56p0larsounds VERY accurate
07:11.02p0laryou wouldn't believe what I"m going through for residency
07:11.08p0larmy work visa, etc.. it's pure nonsense
07:11.16DyOSi'm trying to configure my voicemail.conf file in * to send me the vm to my email when i get them... i have the line for my extension set up properly but how do you configure the smtp server in the voicemail.conf file?
07:11.20jmhunterive seriosuly planned on mving to BC
07:11.28opus___p0lar - lets go to war with canada
07:11.29HjemmeRoyKhm. I keep having interrupt loss, not much, but too much on these digium cards. any idea how to debug that when digium says 'use another server'? I can't...
07:11.39BoRiSjmhunter: Hopefully not vancouver.... Good luck on the job hunt.
07:11.46jmhunterlol
07:11.47jmhunterya
07:11.54p0larwhy?  The only part worth taking is the first 100 miles.. *shrug*
07:11.59jmhunterit would def be vancouver
07:12.01tuxinator_linuxOnline since: Wed Mar 02 15:32:35 2005
07:12.11p0larand 35M people who like to pay loads of taxes.. *shrug*
07:12.12fileugh tired
07:12.16p0lardon't get me started though
07:12.24p0larBC is beautiful
07:12.34opus___p0lar but they got bacon
07:12.35opus___j/k
07:12.39channanjmhutner- bc is nice... wonder if there's hightek job there? I would not mind to live there
07:12.47Mocoh well it night time again...
07:12.59tuxinator_linuxNight Nick Moc
07:13.05tuxinator_linuxNIght
07:13.26jmhuntersorry for slamming la quebecois moc...
07:13.32BoRiSBC is beautiful......but expensive.... And not very many tech jobs......They have lots of jobs in construction if you are into that.
07:13.35p0larsoon, they'll legalize pot and things will REALLY get interesting there.. haha
07:13.57jmhunterya im not in high tech anyway
07:14.03tuxinator_linuxDid you know France has a work limit, 35 hours per week.
07:14.13jmhunteri havent worked in high tech in a long time
07:14.39tuxinator_linuxWhat does jmhunter do then?
07:15.00tuxinator_linuxIs there anything else in life?
07:15.00BoRiSjmhunter: Then vancouver welcomes you....Just dont forget to bring your hard hat.
07:15.27*** join/#asterisk mikegrb (~michael@thegrebs.com) [NETSPLIT VICTIM]
07:15.27tuxinator_linuxwow
07:15.27BoRiSlook what you did
07:15.27DyOScan someone point me in the right direction in how to configre smtp for the voicemail.conf file so it knows which server to use to send emails?
07:15.27tuxinator_linuxsorry
07:15.27BoRiSlol
07:15.31tuxinator_linuxDyOS, have you checked the Wiki?
07:15.44DyOSi'm reading i did a serach for smtp
07:15.46DyOSnothing really
07:15.50SexyKenHey guys -- how do I make sure CDR is turned on in Asterisk?
07:16.05tuxinator_linuxDyOS: Explain your problem a little more
07:16.12Mocit laptop in bed time ..
07:16.25tuxinator_linuxMoc: ??
07:16.32jmhunterur in AB right boris?
07:16.34tuxinator_linuxoh, I get it
07:16.59Mocgoing to sleep it a process...
07:17.00tuxinator_linuxusing your laptop in bed
07:17.00DyOSi want to send an email every time a voicemail is left on a sip extension so i configured the email in the voicemail.conf for that extension...but i dont' think linux knows which smtp server to use to send the mail....i have smtp server set up on my network i just need to know where to specify it at
07:17.00Mocyes...
07:17.13tuxinator_linuxMoc, stay away from the porn, or you won't get any sleep
07:17.15MocDyOS it use his own smtp server
07:17.23*** join/#asterisk cypromis (chuck-the-@62.212.85.27) [NETSPLIT VICTIM]
07:17.33Mochehe I just need not to think about it and im fine
07:17.46*** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) [NETSPLIT VICTIM]
07:17.47Mocanyone have pic of Digium booth at von ?
07:17.48*** join/#asterisk DrKiller (~DrKiller@62.117.85.65)
07:17.49DyOSoh are you sure?  cause it's not sending me mail when i get VM
07:18.02Mocit use your system smtp server
07:18.05tuxinator_linuxthere is a setting, let me see
07:18.23BoRiSare you using sendmail?
07:18.33BoRiSjmhunter: Manitoba
07:18.39jmhuntermy bad
07:18.49jmhunterso its you and 5 others?
07:18.52puppetCOuld anyone help me out with an example? to grab an commandline raw and parse it and use it? f.ex. uptime, parse it grab days hours and output with SayNumbers or so?
07:18.56jmhunteri heard u have a 6th moving in
07:18.57tuxinator_linuxDyOS: check out mailcmd
07:19.21tuxinator_linuxDyOS: in voicemail.conf http://voip-info.org/tiki-index.php?page=Asterisk%20config%20voicemail.conf
07:19.32puppet;mailcmd=/usr/sbin/sendmail -t
07:19.36puppetlinux uses sendmail
07:19.41puppet:)
07:19.52puppetso its kinda right
07:20.01DyOSi guess i need to know how to setup smtp server in sendmail then....thanks i'll look for that
07:20.31puppetdyos if you can use mail command in linux
07:20.33tuxinator_linuxDyOS: What distro?
07:20.52puppetdyos: regular mail xxx@xxx.net
07:20.56puppetdyos: then sendmail often works if someone aint real messy
07:21.01DyOSfc3
07:21.11tuxinator_linuxDid you install sendmail?
07:21.20DyOSit's install i've never used it before
07:21.30tuxinator_linuxThen it is almost ready to use.
07:21.51DyOSso i need to just figure out the commands for sendmail and specify that string in my vm.conf file
07:21.54DyOSright?
07:22.23AgiNamuyey, I found the hack I need to bill call forwarding!
07:22.26AgiNamuLocal channels! :)
07:23.15DOOMDAt once I apologize for my English. Whether it is realized in the program alive packages for radius
07:23.16tuxinator_linuxRead the wiki page
07:23.34tuxinator_linuxDyOS: http://voip-info.org/tiki-index.php?page=Asterisk%20config%20voicemail.conf
07:23.52tuxinator_linuxDyOS: try sendmail mail from the command line
07:27.36tuxinator_linuxDyOS: May this:  echo 'test msg' | sendmail test@anotherdomain.org
07:27.43bkw_sendmail -t
07:28.16tuxinator_linuxDyOS: http://www.freebsddiary.org/virtualmail.php  has some good stuff
07:28.20*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM]
07:28.42*** join/#asterisk Rez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM]
07:28.54SexyKenWhere do I enable cdr_mysql?
07:29.12*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) [NETSPLIT VICTIM]
07:29.36tuxinator_linuxSexyKen: maybe http://voip-info.org/tiki-index.php?page=Asterisk%20config%20cdr_mysql.conf
07:29.42HjemmeRoyKwith that nick?
07:29.44HjemmeRoyK:P
07:30.00*** join/#asterisk paulc (~paulc@251.134.218.209.transedge.com)
07:30.10tuxinator_linuxSexyKen: http://voip-info.org/wiki-Asterisk+cdr+mysql
07:30.45tuxinator_linuxSexyKen: Are you a guy?
07:30.48*** join/#asterisk CpuID (~nathan@dsl-202-173-176-82.qld.westnet.com.au)
07:30.49SexyKentux -> It's already config'd in that file but it's not oading anything.
07:30.58SexyKenIt is sitll storing cdr data in a file rather than db
07:31.10tholoGo get some sleep, bkw.
07:31.11tuxinator_linuxSexyKen: Reload *?
07:31.25SexyKentux - I have reloaded
07:31.31tuxinator_linuxhmm
07:32.15Mocdamn rackspace.com are crazy... 400$ for 100gig/month transfer..
07:32.24jmhunterYOWCH!
07:32.29Mocand 2$ per gig after that ...
07:32.33Mocand that US
07:32.44p0larMoc: I'll beat that! :D
07:32.57MocI get it at 13cent/gig rightnow
07:33.09Mocwell I could get it at 13cent/gig..
07:33.26p0larj/k, we're not in the colo biz
07:33.32MocIm searching for a replacement provider rightnow, that got bandwidth
07:34.08Mocwe get peek of 40mbits+ and over 8000gig/month
07:34.37tuxinator_linuxSexyKen: Did you " Then edit your modules.conf to load cdr_addon_mysql.so and reload asterisk."
07:34.59Mocoh well it nite time.. cya
07:34.59tuxinator_linuxSexyKen: as found on http://voip-info.org/wiki-Asterisk+cdr+mysql
07:35.06tuxinator_linuxNight Night Mox
07:35.08tuxinator_linuxMoc
07:36.33tuxinator_linuxSexyKen: Working on it?
07:37.10AgiNamuUSA Alaska: Makes more sens 0.25 or 0.025?
07:37.29AgiNamutrying to make sure that our rates sheet isn't screwed up
07:37.52tuxinator_linuxAgiNamu: Explain more
07:38.16AgiNamuIs Alaska closer to 0.25 or 0.025?
07:38.30tuxinator_linuxof what?
07:38.46AgiNamu$/min
07:39.11tuxinator_linuxNot sure, 2.5 cents / minute is a good rate
07:39.26AgiNamuyea, but now my new sheet says .25 cents/minute
07:39.30tuxinator_linuxThat is what I get down here in Arizona for Long Distance
07:39.33AgiNamuwhichs seems completley foobard.
07:40.07tuxinator_linux25 cents / minute, back in the 90's
07:40.24tuxinator_linuxHow is SexyKen doing?
07:40.36DyOSok i figured out what my problem is my isp blocks port 25 so sendmail can't send so i need to configure it to use my smtp relay server which is internal...nobody knows the command to set up a relay for smtp in sendmail do they?
07:41.02tuxinator_linuxDyOS: hhmm
07:41.11tuxinator_linuxMost home ISP block 25
07:41.42AgiNamuMost home ISP block port 25?
07:41.45*** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
07:41.57AgiNamuThen why the hell are some of the large ISPs in the USA huge sources of spam? :)
07:41.57DyOSyea but i have an exchange server on my network that is configured to use the isp(cox) smtp server so it lets all mail out to that server...so i just need to figure out how to add a forwarder in sendmail
07:42.01DyOSyea most do
07:42.09AgiNamuComcast doesnt
07:42.15AgiNamucall them up
07:42.20AgiNamuthey can't block it -- it'd cause too many problems.
07:42.30AgiNamuliek... anyone with an email account NOT hosted by them, to start.
07:42.48DyOSyea well cox is pretty big where i'm from and they block 25 and 80
07:42.56DyOSi'm sure there are other ports also like 135 139 probably
07:42.58AgiNamuoh, incoming?
07:43.05AgiNamuoh sure, incoming :P
07:43.12AgiNamuthey prolly say ou can't run a server too :)
07:43.29tuxinator_linuxI have COX, is stinks
07:43.30*** join/#asterisk Mw3 (mw3@daisy.chains.ch)
07:43.33AgiNamuwell, call them and ask them to configure
07:43.48DyOScox is pretty cool i like them
07:43.50DyOSfaster than shit
07:43.57AgiNamuat any rate, if they block 25 incoming
07:43.58tuxinator_linuxDyOS: When it's up
07:44.06AgiNamuhwo does that stop you from SENDING out email?
07:44.45AgiNamuwell im tired
07:44.46AgiNamunight
07:44.46tuxinator_linuxjust did speed test:  4.6 megabits per second
07:45.06AgiNamuI did a speed test the other day
07:45.11AgiNamudownloading monobundle
07:46.25AgiNamuI get abotu 12Mbps
07:46.31AgiNamuand downloading asterisk and so on
07:46.59AgiNamuon my server. here i only have 512k :(
07:47.00AgiNamunight
07:47.03Eightwell that's cute. X-Lite is telling me its logged in, and I don't even have asterisk running =p
07:48.27DyOSthey block 25 incomign and outgoing
07:48.44HjemmeRoyK~lart digium
07:48.46tuxinator_linuxWho was having port 25 problem?
07:49.04puppeteight: haha
07:49.04tuxinator_linuxHjemmeRoyK: What's wrong with Digium?
07:49.37tuxinator_linuxDyOS: What it you with port problems?
07:49.51tuxinator_linuxor was it SexyKen?
07:50.14tuxinator_linuxDyOS: it was you
07:50.22tuxinator_linuxDyOS: http://sunportal.sunmanagers.org/pipermail/summaries/2004-July/005563.html may help
07:50.43HjemmeRoyKtuxinator_linux: their support, their hardware, their arrogance
07:51.17tuxinator_linuxThere're a young company, let em grow a little
07:51.19*** part/#asterisk Vector_ (~Vector@h-67-101-140-109.hstqtx02.covad.net)
07:51.29*** join/#asterisk Vector_ (~Vector@h-67-101-140-109.hstqtx02.covad.net)
07:51.36tuxinator_linuxHjemmeRoyK: Is there something wrong with the hardware?
07:52.25HjemmeRoyK~lart tuxinator_linux
07:52.41tuxinator_linuxWeeee !
07:52.46jmhunter~broadvoice
07:52.47jbotwell, broadvoice is crap
07:52.47tuxinator_linuxAgain ! ! !
07:53.20tuxinator_linuxI don't know Digium that well, so I will have to take your word for it
07:53.23*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
07:53.42tuxinator_linuxHjemmeRoyK: but I would like to know if I shoudl not use their hardware
07:53.51HjemmeRoyKdon't
07:53.59HjemmeRoyKuse sangoma if you're looking for PRI cards
07:54.03HjemmeRoyKthey cost the same and rock
07:54.16HjemmeRoyK~sangoma
07:54.48HjemmeRoyKjbot: Sangoma PRI cards is what Digium's PRI cards were supposed to be in the first place
07:54.50jbotokay, HjemmeRoyK
07:58.15tuxinator_linuxhmm, sangoma cards look nice
07:58.40HjemmeRoyKtuxinator_linux: and they cost the same
07:58.40Juggiesome of their cards need custom drivers tho i believe
07:59.04HjemmeRoyKJuggie: they use an abstraction layer between native drivers and zaptel called wan pipes
07:59.26tuxinator_linuxHjemmeRoyK: My Digium Dev Kit hasn't given me any trouble
07:59.58tuxinator_linuxbut I will be getting a PRI soon, so I need to get a card
08:00.03HjemmeRoyKmy digium cards in ibm servers gives me lots and lots of interrupt loss and digium just blames the servers
08:00.39tuxinator_linuxCould be, my digium card didn't like sharring and IRQ
08:00.51tuxinator_linuxhad to move and disable a few things
08:00.57tuxinator_linuxold MB
08:01.45tuxinator_linuxIs there a downside to using sangoma?
08:03.28tuxinator_linuxWAN pipes, sounds like a sewer system
08:04.29HjemmeRoyKdon't know yet
08:04.36HjemmeRoyKwill try the 1-port later today
08:04.46*** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl)
08:04.48HjemmeRoyKgetting quad port friday or something
08:04.51MuppetMasterHello everyone.
08:04.56tuxinator_linuxHellow MuppetMaster
08:04.59MuppetMasterWhy does Asterisk not support SIMPLE/Presence?
08:05.05tuxinator_linuxWho?
08:05.08HjemmeRoyKI know they don't like interrupt sharing, but I've tried virtuall every fscking thing
08:05.17HjemmeRoyKMuppetMaster: wtf is that?
08:05.45MuppetMasterHjemmeRoyK:  SIMPLE is the presence capability (ie - IM) within the SIP standard.  Similar to XMPP in the Jabber world.
08:05.46tuxinator_linuxHjemmeRoyK: What is your error, I though my HD was failing, but it and IRQ conflict
08:05.50*** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.rr.com)
08:06.09nine76hello all
08:06.33MuppetMasterFor example, the Microsoft Live Communication Server is using SIMPLE for their presence capabilities and for interconnection with other IM platforms.
08:06.37tuxinator_linuxMuppetMaster: http://voip-info.org/tiki-index.php?page=SIP%20simple
08:06.48tuxinator_linux~rtfw
08:06.49jbotit has been said that rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
08:07.05MuppetMastertuxinator_linux:  Exactly.
08:07.05tuxinator_linuxHello 976
08:07.13HjemmeRoyKtuxinator_linux: just attest reporting some .05 and more interrupt loss and terrible sound drops
08:07.23HjemmeRoyKon three similar servers
08:07.30MuppetMasterNow, since Asterisk is all singing all dancing and one of the best softswitches around, why has it not supported something as 'basic' as SIMPLE?
08:07.37tuxinator_linuxHjemmeRoyK: Interesting
08:07.56MuppetMasterAnyone have any ideas?  As this could be a very powerful tool, invoked within dialplans to do follow me routing based on your IM status, etc, etc.
08:08.00puppetmuppetmaster: cause no one have coded it?
08:08.05puppetmuppetmaster: code it then? :)
08:08.07MuppetMasterpuppet:  Most likely.
08:08.08tuxinator_linuxMuppetMaster: Hack away
08:08.13MuppetMasterpuppet:  If only I were a coder.
08:08.15jmhunteri want one of those slick new grandstream phones
08:08.27puppetjmhunter: i want one of thoose nice cisco wlan phones :/
08:08.35MuppetMasterJust curious if there was some reason not too, or just no one cares...
08:08.43puppetmuppetmaster: no one cares i think
08:08.46tuxinator_linuxhttp://www.grandstream.com/y-gxp2000.htm ?
08:08.55jmhuntercisco wlan phones? cool...  $$$$
08:09.01MuppetMasterWould bring Asterisk to the next level, as could be used within the call queuing capabilities as well.
08:09.02puppetjmhunter: mm ;P
08:09.40MuppetMasterYou could send users messages when a call arrives via their IM client, ask them how they want to treat it, etc, etc.  I have already done some of this with a Jabber server/XMPP...  Would be beautiful to fully exploit the SIP standard within Asterisk.
08:10.05puppetjmhunter: CISCO GLOBAL IP TELEFON 7920 + STATIONS LICENS
08:10.08puppetjmhunter: that one
08:10.23MuppetMasterMaybe another reason to have SER there with Asterisk...
08:10.27puppetexploit? exploit aint good ;p
08:10.56jmhuntero the 7920, the one thats been around for ages
08:11.12jmhunternothing special.. and u cant load SIP into it.. unless thats changed
08:11.19jmhunterthere is no sip image
08:11.23puppetjmhunter: ouch ok :/
08:11.50jmhunterthe new grandstreams are a good competitor with the 7940... less flash, lest cost
08:11.54Juggieit will still work with mgcp
08:11.57Juggieor skinny
08:12.00tuxinator_linuxI got a quote today for various Cisco phones with SIP
08:12.05puppeti want a cordless
08:12.14jmhunterwisip if anything
08:12.35jmhunteri still prefer the 7905g
08:12.37jmhunterits sick
08:12.57puppetjmhunter: it is ;D
08:13.05tuxinator_linuxhttp://pastebin.ca/7065  has what I was quoted
08:13.07Juggietuxinator_linux, take a look at the mitel 5215 or 5220
08:13.54jmhuntersounds right, where the quote from
08:14.19puppetjmhunter: 7970G then? ;p
08:14.27tuxinator_linuxJuggie: looks like nice perkless phones
08:14.34tuxinator_linuxno 7970
08:14.42tuxinator_linuxno SIP for 7970
08:14.42jmhunter7970 doesnt do sip either
08:14.53puppetis cisco that after? ;P
08:14.53tuxinator_linuxjmhunter: Insight
08:14.57jmhunterur talking 7960, 40, 05g
08:14.58puppetall the cool phones lack Sip
08:15.03HjemmeRoyK~sangoma?
08:15.20jmhuntercheck with gtsinc.biz... theyre competitive and you can talk the price down
08:15.22Juggietuxinator_linux, perkless phones?
08:15.40tuxinator_linuxno frills
08:15.42HjemmeRoyK~sangoma
08:15.45HjemmeRoyK~Sangoma
08:15.51tuxinator_linuxjust phones, no fancy stuff
08:16.00Juggietuxinator_linux, what do u want that mitel doesnt do
08:16.07tuxinator_linuxHjemmeRoyK sure likes Sangoma
08:16.30tuxinator_linuxDisplay is a little small for a good directory
08:16.36HjemmeRoyKdid someone tell jbot to forget it?
08:16.48tuxinator_linuxDon't think he learned it
08:16.53puppetI am having troubles to figure out one thing :/
08:17.04tuxinator_linuxspit it out puppet
08:17.13HjemmeRoyKjbot: sangoma PRI cards is what Digium's PRI cards were supposed to be in the first place
08:17.14jbotHjemmeRoyK: i already had it that way
08:17.20Juggietuxinator_linux, i dont know if the 5220 links to a remote directory either
08:17.31Juggiei'm still awaiting my shipment of sip enabled phones
08:17.37Juggiei have 5220's but they are minet only
08:17.38puppetI wanne take uptime, split it up, then output it in voice, kinda
08:17.55puppetCant figure out how to do it
08:18.05Juggiepuppet, should be easy, write an agi to do it
08:18.16tuxinator_linuxpuppet: ya, what Juggie said
08:18.27puppet*reads some about agi*
08:18.45Juggiei'd use agiphp split it up no problem and then play it back using say number and some recorded messages
08:18.53HjemmeRoyKjbot: sangoma is a company that makes PRI cards the way Digium should have done it in the first place....
08:18.54jbotokay, HjemmeRoyK
08:19.09tuxinator_linuxhttp://voip-info.org/wiki-Asterisk+AGI
08:19.18puppethttp://www.voip-info.org/wiki-Asterisk+AGI < was there now ;P
08:20.07tuxinator_linuxhttp://voip-info.org/wiki-Asterisk+AGI+php looks good to me
08:20.14tuxinator_linuxI am a PHP kinda guy
08:20.40puppet< php to
08:20.41puppet:)
08:20.50Juggietuxinator_linux, then go get the latest version of phpagi off the sourceforge cvs
08:20.56Juggieand then you'll get it going no problem
08:20.57puppetI can just try the example.php
08:21.12Juggiephpagi makes it easier
08:21.16Juggiedoes all the dirty work for you
08:21.45tuxinator_linuxI am anxious, just need to work on some other projects first
08:22.14tuxinator_linuxI have learned alot living in this channel
08:23.34tuxinator_linuxThere are a lot of people that never say anything, like DOOMD
08:23.40tuxinator_linux~seen DOOMD
08:23.42jbotdoomd <~DrKiller@62.117.85.65> was last seen on IRC in channel #asterisk, 1h 27s ago, saying: 'At once I apologize for my English. Whether it is realized in the program alive packages for radius'.
08:23.43puppetsweet
08:23.45puppetthat was easy ;D
08:23.52puppetafk gonecode nerd2k uptimescript
08:24.30tuxinator_linuxpuppet: ??, oh well
08:24.52puppettuxinator_linux: using it at home ;p
08:24.54tuxinator_linuxpuppet: Need sleep, can't figure things out
08:25.00puppet"press 4 for current uptime"
08:25.01puppet;D
08:25.05tuxinator_linuxahh
08:25.06tuxinator_linuxfun
08:25.07nine76may I ask a question? (yes,I have rtfw,and getting started guide, and several users pages...)
08:25.12tuxinator_linuxYes
08:25.24tuxinator_linuxSpeak your mind nine76
08:25.46tuxinator_linuxDon't take "rtfw" personally
08:27.16tuxinator_linuxjbot: what time is it?
08:27.18jbotI think you lost me on that one, tuxinator_linux
08:27.23tuxinator_linux~time
08:27.24jbot[time] 1 dimensional, or everlasting
08:27.31nine76I am awaiting a x100p card,so I'm playing around till ti arrives. I setup asterisk on a 2.6 kernel (is that a nono?) anyways,i can connect with a couple sip clients (kphone,xlite) I can use the clients to access voicemail,and other demo features. I added FWD and Iaxtel accnts. When I try to dial a 1800 # with either sip client I get "not found". I have tried sending the 800 calls through iaxtel,and fwd,from console of * it dials fine,from sip
08:27.31nine76<PROTECTED>
08:28.03tuxinator_linuxI wrote a wiki for 2.6 using CentOS http://www.voip-info.org/tiki-index.php?page=Asterisk+CentOS-4.0+Zaptel
08:28.45Juggienine76, its your dialplan
08:28.49tuxinator_linuxI haven't played with that stuff yet, sorry
08:28.53puppettuxinator_linux: do I just do PLAYBACK days
08:29.01tuxinator_linuxpost your dialplan to pastebin.ca
08:29.01Juggieyour sip clients come in on a dif context likely perhaps sip? and you dont have the dial stuff in there
08:29.35*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
08:29.39tuxinator_linuxpuppet: Are you asking how to play a sound?
08:29.46puppettuxinator_linux: yeah :)
08:29.46Juggieyou should put your iaxtel/fwd etc... into its own context
08:29.54Juggiethen include it into default and sip
08:31.35tuxinator_linuxpuppet: http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Playback
08:31.52nine76How depressing. I assuemd I would get no good answers and just "go to wiki". I actually got a good answer,and dont understand it :-P sip clients are in context default. unsure of what context iaxtel and fwd are in. I'll get back to reading. Thanks
08:31.53EightI'll just keep tinkering if there's no simple answer, but is there any basic setup I missed where a SIP call into the demo context works fine, but the sip client doesn't hear any audio? (I'm fairly confident SJPhone isn't having local audio config issues).
08:31.59puppettuxinator_linux: i meant from php
08:32.21tuxinator_linuxpuppet: ohhh
08:34.03tuxinator_linuxpuppet: http://pastebin.ca/7067
08:34.11modulus_bleh
08:34.18puppetthanks
08:34.19tuxinator_linuxpuppet: Don't know anything about PHP AGI yet
08:34.59tuxinator_linuxpuppet: but that is a starting point
08:35.12puppetIm uaing regular php
08:35.33tuxinator_linuxoh, I haven't done sound in PHP
08:36.57*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
08:37.39moonwickholy cthulhu, you have to be kidding.
08:37.57tuxinator_linux~cthulhu
08:37.58jbotIa! Ia! Cthulhu ftaghn! or i heard cthulhu was an ancient horror, sleeping below the seas in dread R'lyeh. or a figment of '30s horror author H.P. Lovecraft's imagination..  Ph'nglui Mglw'nafh Cthulhu R'Lyeh Wgah'nagl Fhtagn
08:37.58moonwickPHP is not a general-use language, for chrissake
08:38.14puppettuxinator_linux: works sweet
08:38.22puppettuxinator_linux: write("EXEC PLAYBACK days X");
08:38.28moonwickwhen PHP is the only language you know, every problem looks like a crappy website.
08:38.35tuxinator_linuxmoonwick: I use it for a lot of stuff
08:38.39moonwickyou're nuts.
08:38.50tuxinator_linuxEasy to do stuff
08:38.54tuxinator_linuxI do perl also
08:38.56moonwickPHP is a lousy, lousy language.
08:38.59moonwickthat's good.
08:39.00tuxinator_linuxI am relearning C
08:39.18moonwickperl is at least designed for doing more than just web stuff.
08:39.30tuxinator_linuxHaven't really done C since I was 14
08:39.45tuxinator_linuxPHP works well for shell scripts
08:40.09tuxinator_linuxwhen the syntax is on the top of your mind
08:40.16moonwickso does, um, you know, shell.
08:40.27tuxinator_linuxI know
08:40.27Zeeekyawn
08:40.39tuxinator_linuxZeeek: Tired?
08:40.49Zeeekya
08:40.55Zeeekand alone
08:41.02tuxinator_linuxpoor Zeeek
08:41.34tuxinator_linuxDo commit suicide and try to install * on windows
08:41.38Zeeekbut my assistant will be here shorly, and I turned on the espresso machine, so all is well
08:41.38*** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
08:41.55tuxinator_linuxI mean don't
08:42.07Zeeekshit I forgot to start the tftp server
08:42.25puppettuxinator_linux: this was easy ;p
08:42.29*** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
08:42.32Juggiedoes * even compile/
08:42.41Juggiesomeone actually bothered to make * run on winblows?
08:42.43tuxinator_linuxpuppet: I will have to pick your brain when I get there
08:42.59tuxinator_linuxJuggie: Why I don't know
08:43.00Zeeekthere are versions
08:43.00moonwickclearly the world needs a port of * to java :)
08:43.02puppettuxinator_linux: Ill paste it to you when its done
08:43.11tuxinator_linuxjava is too bulky for me
08:43.14tuxinator_linuxcool
08:43.35Zeeekbecause of the zillions of windows installs out there. WIndows is like China - you can't ignore it
08:44.12Juggiejava
08:44.15Juggiegay
08:44.24Juggiesure if youdlike it to only do 10 calls before crapping out
08:44.35Zeeekheh
08:44.45tuxinator_linuxI'll wait untill Java is no longer controled by Sun
08:44.51JuggieZeeek, windows is never going to be a reputable server os
08:44.54Juggieno matter what happens
08:45.07Juggieeverything mission critical is a *nix job
08:45.19Juggieor *bsd
08:45.23moonwickI dunno, another couple of rewrites and they might make it.  :P
08:45.24modulus_bsd is a unix
08:45.25ZeeekI don't care to add to the literature on this subject. I'll just follow my own course.
08:45.34*** join/#asterisk Martohtar (Martohtar@82.196.218.130)
08:45.39Zeeekiax2 debug
08:45.51Zeeekfscking windows
08:46.14tuxinator_linuxis that polite way to say f'n?
08:46.29Juggieyou wont get any hardware support under winblows
08:47.12tuxinator_linuxno more windows bashing, not going to waste my energy talking about it
08:47.26tuxinator_linuxno one here disputes is crappiness
08:47.35tuxinator_linuxeven jbot
08:47.36tuxinator_linux~windows
08:47.37jbothmm... windows is a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition...
08:48.05tuxinator_linuxjbot is java
08:48.15modulus_jbot java?
08:48.16jboti guess java is at (Link: http://www.blackdown.org)http://www.blackdown.org or at (Link: http://www.javasoft.com)http://www.javasoft.com or at (Link: http://www.savaje.com)http://www.savaje.com or at (Link: http://www.pocketlinux.com)http://www.pocketlinux.com or at (Link: http://java.sun.com)http://java.sun.com or at http://www.insignia.com, or see the ...
08:48.30Juggie~java
08:48.31jbotmethinks java is at (Link: http://www.blackdown.org)http://www.blackdown.org or at (Link: http://www.javasoft.com)http://www.javasoft.com or at (Link: http://www.savaje.com)http://www.savaje.com or at (Link: http://www.pocketlinux.com)http://www.pocketlinux.com or at (Link: http://java.sun.com)http://java.sun.com or at http://www.insignia.com, or see the ...
08:48.34tuxinator_linux~about jbot
08:48.40tuxinator_linux~status
08:48.41jbotSince Tue Mar  8 23:18:31 2005, there have been 13 modifications, 157 questions, 0 dunnos, 0 morons and 152 commands.  I have been awake for 9h 30m 9s this session, and currently reference 108290 factoids.  I'm using about 12516 kB of memory. With 0 active forks. Process time user/system 1136.61/159.16 child 67.63/6.54
08:48.52Zeeek~got religion
08:49.07Juggie~karma zeeek
08:49.07jbotzeeek has neutral karma
08:49.15Juggie~karma juggie
08:49.15jbotjuggie has neutral karma
08:49.17tuxinator_linux~karma tuxinator_linux
08:49.17jbottuxinator_linux has neutral karma
08:53.04Zeeek~seen anything interesting
08:53.06jboti haven't seen 'anything interesting', Zeeek
08:53.12Zeeeknor have I yet
08:53.21ZeeekCOFFEE TIME WHEEE
08:55.02*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:04.29tuxinator_linuxZeeek, you're going to be up all night, oh wait, it's morning for you, isn't it?
09:04.37ZeeekYep!
09:04.56*** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net)
09:05.58Rivalmornin
09:07.10tuxinator_linux2 AM here, early morning Rival
09:07.39Rival=)
09:07.40Zeeek10:07AM
09:10.22tuxinator_linuxI think I am sleepy
09:10.32tuxinator_linuxbut I don't wanna
09:10.40tuxinator_linuxkinda boycotting it this week
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09:20.25MuppetMaster10.20 am here, nice and sunny
09:20.51nine764:20am here...according to kweather its sunny out...its lying.
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09:24.36EightHooray =)
09:24.37ZeeekLOOP!
09:25.23EightNow I get to see if I can make asterisk available through my NAT =p
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09:31.42*** join/#asterisk Spiro (~icechat5@196.7.14.183)
09:31.50SpiroHello everyone
09:32.14SpiroIt took me like a decade to try and register to access this channel, feeling a bit slow this morning
09:32.44SpiroSo, does anyone know is asterisk support conferencing?
09:32.51multrixHi, do anybody here tried to use french company wengo with a IP-10S phone ? the password seems to be too long !
09:33.26EightSpiro: I'm pretty sure asterisk will do just about anything that you might call 'conferencing'.
09:33.34*** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it)
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09:33.56SpiroI want to run asterisk as a conference server having more than 20 people together in a session communicating
09:34.06justinnnnnnasterisk will do that pretty easily
09:34.08Eightspiro: Yup.
09:34.13SpiroBrilliant
09:34.30EightSpiro: I forget what it's called exactly, but there's even a term for that feature.
09:34.41justinnnnnnmeetme ?
09:34.42nine76meetme?
09:34.44EightSpiro: don't look at me for how to implement it though, I'm a newbie to Asterisk myself =)
09:34.50SpiroI think it's called conference bridging, does that ring a bell?
09:34.51justinnnnnnanyone wana fix my txfax for $$ pls :) ??
09:34.53EightYa, that sounds like it =)
09:34.57Eightre: meetme.
09:34.57justinnnnnnsomeone will fix it for me one day :)
09:35.24SpiroWould you say that asterisk is one of the better SIP servers out there?
09:35.26justinnnnnnspiro, u can have passwds as well etc.. its pretty cool the conferance stuff
09:35.34SpiroAnd what protocol would you suggest using
09:35.43hajekdCan you comment the sound quality of Sipura 1001?
09:36.11SpiroCan you use windows messenger with it?
09:36.27justinnnnnni think u can.. but u goto use some wierd non-sip setup ?
09:36.30nine76first the * problem,then toilet overflows,now openbsd has died...kernel fault:-/ not my day people.
09:36.35Spiroi.e does it support the RTC control library, if a client is built using this
09:36.53justinnnnnnspiro, have a look on www.voip-info.org
09:36.55SpiroQuality you are all leegends
09:37.06justinnnnnnhas all the answers ur after :) pls heaps other
09:37.12SpiroJustin what is your experience with asterisk
09:41.14modulus_i saw a booth at a show once
09:41.24modulus_what is an asterisk anyways?
09:42.27SpiroIS there a rpm for asterisk
09:42.35*** join/#asterisk beezly (~beezly@2001:630:63:16:204:75ff:feed:da66)
09:42.49modulus_rpm eww
09:42.49Eightthe only 'packaged' install of Asterisk I know if is a Mac OS X pkg.
09:42.52tzafrirAn asterisk? The character '*', right?
09:42.55modulus_redhat sucks
09:43.00potteranyone here setup h323 ---- > asterisk ------> SIP
09:43.04potterand working
09:43.13Spiroi am new in linux
09:43.16tzafrirEight, Debian has some pretty good Asterisk packages
09:43.21SpiroWhat is the best linux server to use?
09:43.26SpiroWhy does redhat sucl
09:43.35modulus_because it's brain-dead
09:43.38Eighttzafrir: Oh really? Maybe I should have setup a debian-ish box for Asterisk =)
09:43.48Spirothayt's an ntellegent answer
09:43.57SpiroEllaborate
09:43.57MuppetMasterSpira:  I like SuSE v9.2
09:44.03beezlyI've just got an X100P which is showing RED state in zttool - i'm not sure it's been correctly plugged into the line yet. would the x100p show red if that were the case?
09:44.04EightSpiro: bah. Don't bother getting into a distro-war with people =p
09:44.09tzafrirYou can try Rapid, which already comes with pre-compiled everything (zaptel included)
09:44.11MuppetMasterSpiro:  Works great with Asterisk.
09:44.36modulus_spiro, ever used redhat in a high load environment?
09:44.36Eighttzafrir: Rapid is what?
09:44.47tzafrirbeezly, yes, it will show RED if it's not connected
09:44.56tzafrir~rapid
09:44.57jbotmethinks xorcom rapid is at http://www.xorcom.com/rapid.html . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir
09:44.58SpiroI am very new in the linux circle
09:45.03SpiroI don't have a clue
09:45.05beezlytzafrir: excellent - at least I know that's what I should expect then!
09:45.08*** join/#asterisk nextime (~nextime@ns0.nexlab.net)
09:45.25modulus_spiro, try a real os: http://www.freebsd.org
09:45.35SpiroWhat is this
09:45.40modulus_unix
09:45.42modulus_bsd unix
09:45.46tzafrirthe OS of the daemon
09:45.47Eighttzafrir: aha, Cool. How is that different from the Asterisk@Home thing that got on /. recently?
09:45.58tzafrirEight, a different approach
09:46.05Eighttzafrir: in what way?
09:46.06modulus_tzafrir, penguins last forever but daemons never die
09:46.21tzafrirAsterisk@home is a distro to build Asterisk.
09:46.43SpiroHow does asterisk run on Unix
09:46.50tzafrirIt thus includes a complete build system. I'm a strong believer in the packages system of Linux
09:47.11tzafrirSpiro, what Unix exactly? SCO Open Server?
09:47.24Eighttzafrir: Ah, so it's a bare-bones box that installs binaries?
09:47.36modulus_tzafrir, i highly doubt he even heard of SCO
09:47.57SpiroBSD unix?
09:48.03tzafrirEight, yes. If you want a build system, you can "upgrade" it to a complete Debian installation
09:48.13*** join/#asterisk Tommmo (~tps@203.62.181.52)
09:48.13Tommmohi
09:48.15Eighttzafrir: Cool. Thanks for the info.
09:48.23Tommmoi've got a call coming in on number 03 9015 1234
09:48.30Tommmoi want asterisk to redirect this to extension 1234
09:48.32tzafrirAsterisk seems to run well on FreeBSD. It also seems to run on NetBSD
09:48.35Tommmohow can i strip off the 039015 ?
09:48.40modulus_tommo, some of my calls start with +
09:48.45maruzif a queue waiting is too long, can i allow caller to presss e digit to leave a voicemail message?
09:48.48nine76Then my upcoming OpenBSD attempt should go smoothly:)
09:48.59EightAre there any driver issues with running on FreeBSD?
09:49.05Tommmotried something like
09:49.18Tommmoexten => _0390151xxx,1,Goto(${EXTEN:4},1)
09:49.24potterEight havent gone that far yet ... dont have zaptel cards installed right now on the server
09:49.33tzafrirDoes zaptel work with OpenBSD? (no self interest, just for general information)
09:49.35Tommmobut it removes the first
09:49.46nine76wiki claims it does work with obsd
09:50.17TommmoHow can I send calls between contexts?
09:50.17modulus_tzafrir, openbsd has linux binary (file structure) support too
09:50.25Tommmofor example, my calls come in on context default
09:50.32tzafrirmodulus_, that doesn't cover drivers support
09:50.39Tommmoi'm trying to redirect to extension 1234 in context "test"
09:50.46modulus_tzafrir, but it's the same kernel
09:50.49Tommmois there a way to do this?
09:51.04modulus_tommo, do you need Goto()?
09:51.09*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
09:51.17modulus_goto(extension,priority)
09:51.18shaZwazhi all
09:51.22Tommmomodulus_ im using that
09:51.23Tommmoexten => _039015513X,1,Goto(${EXTEN:6},1)
09:51.29Tommmobut how do i tell it to go to another context?
09:51.29modulus_there ya go!
09:51.38*** join/#asterisk Blazeeeeeee (~lol@196.41.19.130)
09:51.42Tommmoi.e. all my incoming calls only appear to work in "default" context
09:51.54Tommmothen i need to take the call into different contexts...
09:52.11Blazeeeeeeehello there, i am looking for anyone who knows asterisk, but live in South Africa
09:52.25clive-balzeee howzit
09:52.39Blazeeeeeeehi clive-
09:52.59Blazeeeeeeeanyone from south africa here?
09:53.02*** part/#asterisk beezly (~beezly@2001:630:63:16:204:75ff:feed:da66)
09:53.03Blazeeeeeeeplease
09:53.14potterwats the prob Blazeeeeeee
09:53.14EightTommo http://www.voip-info.org/wiki-Asterisk+cmd+Goto
09:53.31EightGot(context, extension, priority)
09:53.35Eighterr
09:53.37EightGoto(context, extension, priority)
09:53.52modulus_i still have an openbsd 2.9 box that's been running for 4 years for a client of mine
09:54.10nine76WOW. Thanks juggie. "oh, *that* context* ;)
09:55.10EightJust out of random curiosity: Anyone in the Minneapolis/TC area?
09:55.17modulus_mall of america
09:55.18modulus_wheeeeee
09:55.28Eightmodulus_: =p
09:55.33puppetIm so bored :(
09:55.52modulus_scandanavian descendants adopting korean kids
09:55.54modulus_wheeeeee
09:55.58puppetGive me something usefull to do
09:56.10modulus_puppet: ping -f whitehouse.gov
09:56.18Eightpuppet: then maybe you can tell me what ports I need to forward to the Asterisk box through my NAT for SIP clients to hit me from outside?
09:56.27modulus_eight, udp 5060
09:56.32modulus_jbot sip?
09:56.33jbotX11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/  Session Initiation Protocol (see RFC 3261)
09:57.04puppeteight: RTFW ;p
09:57.21EightHeh, I'm reading I'm reading...
09:57.22puppetmodulus_: i just forwarded hotkey 5 -> 1-800-Freesex
09:57.24Eightbut you said you were bored =p
09:57.33puppeteight: ;D
09:57.43modulus_[root@mirkwood ~]# cd /usr/ports
09:57.43modulus_[root@mirkwood /usr/ports]# make search key=asterisk
09:57.43modulus_Port:   asterisk-1.0.3_1
09:57.46modulus_do i need to cvsup?
09:57.48modulus_anyone?
09:58.34pottercvsup
09:58.43modulus_what should it be?
09:58.47modulus_1.1.x?
09:58.53pottertheres the 1.0.5 already out
09:59.07nine76hmm
09:59.08modulus_stable?
09:59.11justinnnnnnsomeone who wants money pls fix my txfax :) ?
09:59.12potteryep
09:59.29potterusing it now
09:59.36potteror rather since yesterday :D
09:59.48shaZwazjustinnnnnn: what wrong ?
09:59.58modulus_potter, yesterday?
10:00.39modulus_heh
10:00.44Eight1.0.6 is out...
10:00.51modulus_Last database update: 2005-03-09 09:51:05 UTC
10:00.51Eightatleast, I think that's what I just downloaded =)
10:01.08modulus_Wed Mar  9 02:01:11 PST 2005
10:01.23modulus_i'm not too far behind ^_^
10:03.47*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
10:04.17EightWhere do I tell Asterisk what ports it should use for SIP RTP?
10:04.27MuppetMasterEight:  /etc/asterisk/rtp.conf
10:04.33Eightaha =)
10:04.40Eightnew conf file for me =)
10:04.41EightThanks.
10:04.53MuppetMasterEight:  no worries
10:05.12MuppetMasterToo many conf files...reminds me of Genesys in the early days...
10:05.33*** part/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it)
10:05.37EightI kinda like the segregation, actually.
10:05.51MuppetMasterYes, but difficult to get one's mind around at first.
10:06.04EightI just need to learn to skim the files when I'm looking for something.
10:06.12MuppetMasterFor real enteprise management would be great to see some type of configuration server that actively manages all configuration.
10:06.19EightAye.
10:06.21MuppetMastergrep works wonders
10:06.39EightDoesn't seem like it'd be too hard to write a python front end to the config files.
10:06.43Eighteither web based, or whatever.
10:06.52shaZwaz~seen implicit
10:06.53jbotimplicit <~implicit@ip68-5-148-1.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 7d 4h 59s ago, saying: '~seen cmc'.
10:06.56nine76There a php frontend
10:07.21Eightnine76: I saw mention of that somewhere... haven't had a chance to look. Tried it?
10:07.26MuppetMasterEight:  True, but it is all the consistency checking logic to ensure proper configuration to do it right.
10:07.58nine76Eight, I have not. Hoping to fully understand things better by not going the gui route.
10:07.59MuppetMasterThe PHP front end (unless you are talking AMP) is simply a viewer for the raw text files.  At least the one I have seen.
10:10.04EightI don't really mind the text file thing... it feels really natural to me actually. Perhaps it's all the time I've spent in unix flat config files...
10:10.20Eightbut it's certainly not something I'd want to train people in on regularly.
10:10.53*** join/#asterisk huggie (~huggie@the.earth.li)
10:10.54Eightok, so the port forwarding thing is apparently pretty painless.
10:11.05Eight5060 & 10k-20k, by default.
10:11.21Eight(UDP)
10:12.07huggie_9.,1,GotoIf($[${CONTEXT} == "rob"]?100:2) works fine if context is rob but seems to fail otherwise with a parse error.  What would cause that?
10:12.24huggie(on asterisk 1.0.0 - not been tracking releases)
10:13.06MuppetMasterhuggie:  Try 9.,1,GotoIf($["${CONTEXT}" == "rob"]?100:2)
10:13.32puppethuggie: remove ""
10:13.33puppetmaybe
10:13.46huggieMuppetMaster: yeah I did and it failed again.  Hmm, I guess I didn't try removing the "s
10:13.49justinnnnnnhuggie: i get that as well!!!
10:13.56justinnnnnni copy and paste a gotoif and it doesnt work
10:14.06puppethuggie: thats what i did on mine but it looked a bit diff
10:14.21puppetim using s/number,6,
10:14.24puppetnow instead :)
10:15.58*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
10:15.58*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th
10:17.38huggieSo with quotes there's no parse error but I get:  Executing GotoIf("SIP/501-010a", ""staff"=="rob"?100:2") and then it goes to 100.  Um.  staff isn't equal to rob!
10:18.09*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
10:24.07tzafrirEight, what's so wrong with editing config files?
10:24.29puppethuggie: did it work?
10:24.49*** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com)
10:25.09huggiepuppet: no, haven't figured it out yet.
10:25.18puppethuggie: it didnt work as i said?
10:25.54tzafriranyway,  what I don't like about AMP is that it stays too close to the structure of the config files. It has no real notion of an extension with its own line and voicemail box
10:25.56huggiepuppet: removing the quotes?  Didn't seem to.  But spaces or lack of them appear to be important and I'm not sure I've tried all the combinations yet.
10:26.06tzafrirThis is what I like about DeStar
10:27.46Eighttzafrir: Nothing wrong with editing flat text files. But it *is* alot to learn if you want even the simplest of features.
10:27.48*** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net)
10:27.56puppethuggie: huggie aint it single = and not ==
10:28.51huggiepuppet: ooh good call.  I'll check in a mo.
10:29.35CleanerXres_config_odbc.so: undefined symbol: va_copy  anyone seen this before?
10:31.48*** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63)
10:34.31*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
10:35.15huggiepuppet: woo, that's it thanks :)
10:35.55RoyKlol
10:35.56Eightwhat-are-you-wearing.gsm ?!
10:35.59*** part/#asterisk huggie (~huggie@the.earth.li)
10:36.01RoyK:D
10:38.45tuxinator_linuxtalk-to-me.gsm
10:39.14tuxinator_linuxGetting sleepy, slowing down
10:39.37tuxinator_linuxHey guys, I am going to go join my wife for the rest of the night
10:39.50tuxinator_linuxHave a good day
10:39.51tuxinator_linuxNight
10:41.05*** join/#asterisk mcnobody (~laaksola@server.kopteri.net)
10:41.59*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
10:44.52*** join/#asterisk salviadud (~dude@201.133.209.97)
10:45.53*** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com)
10:46.08salviadudim new to this asterisk deal, i work at a company that handles calls from all over the place, im basically an international operator
10:46.21shadebobhi, does asterisk need a sound card for music on hold or my te110p can make music for client?
10:46.26salviadudbut, since we use winblowz, i think i could help out with asterisk
10:47.33*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
10:49.37shaZwazjustinnnnnn: you there ?
10:50.33salviaduddo i need voice over ip to run asterisk?
10:50.42salviadudor can i run it from a normal phone line?
10:50.51Eightsalviadud: the latter.
10:51.13salviadudi need asterisk to run voice over ip eh...
10:51.52Eightnot necessarily, but it does act as a good bridge between plain phones and VoIP
10:52.14Eightbut if all you want is a residential VoIP line there are easier solutions.
10:52.33salviadudi don't want easy...
10:52.47salviadudwhat i want is to start a "call processing server"
10:52.53salviadudif thats what you call it
10:52.58salviadudat my home first
10:53.01EightPBX =)
10:53.01salviadudthen
10:53.05Eightis what you call it.
10:53.14salviadudright, pbx!
10:53.22salviadudthen, if i get good enough skill
10:53.34salviadudchange my company's old pbx to asterisk
10:53.45EightSounds like a plan.
10:53.50salviadudcause i know anything running under windows is just not good enough
10:54.26salviadudbut, i feel like the newest newbie of them all, you know where i can get a good voice over ip phone to start me up?
10:54.43salviadudany recommendations?
10:54.45Eighthardware or software?
10:54.50EightI'm using SJPhone right now.
10:54.55Eightto do my Asterisk testing.
10:55.21salviadudhardware
10:55.29salviadudi need the "phone" basically
10:55.38salviadudso, i need to know what to buy
10:56.07Eighthttp://www.voipsupply.com/index.php?cPath=99_103
10:56.22EightThe IAXy at the top turns any phone into an "IP" phone.
10:57.04Eightand the TDM400P cards also interface with phones (FXS) or interface with phone lines (FXO)
10:57.15clive-i would get a sipura over an iaxy any day
10:57.42Eightclive-: why so?
10:58.53clive-eight, iaxy cant do g729 or g723, or gsm or jitter buffering
10:59.10clive-its only plus is that it can do iax2
10:59.37clive-which is only a plus if you are behind a NAT
10:59.38eipianyone tried a tdm02b on pci 2.1? works?
11:00.03eipii know that website says that needs pci 2.2, but anyone tried if it works?
11:00.54Eightclive-: Ah, that's interesting.
11:01.21EightSo it's a good device to give to 'roamers', but not really anything special on a LAN?
11:01.43*** join/#asterisk cced (~wangxinta@222.33.36.198)
11:01.53clive-lol..if they roam to south africa over here, it will be worthless also
11:02.09EightHow os?
11:02.11Eighterm, How so?
11:02.23clive-g711 uses way too much bandwidth, more than one can get
11:02.27Eightah
11:02.36ccedhow to improve open-source tormentia cards?
11:02.53Eightis that the only codec it'll use?
11:02.54ccedhow to improve  open-source tormentia cards?
11:02.55ccedas if Digium and Varion card improved from tormentia cards.
11:02.55ccedhow to do?
11:02.55ccedas seen from
11:02.55ccedhttp://www.zapatatelephony.com/
11:03.17clive-eight, take your pick g711 or g711
11:03.34clive-lol
11:03.35Eightclive-: thanks. That's good info.
11:04.02clive-if you are behind a NAT then iaxy is good
11:04.47clive-I am hoping to try out the av168ev ata with iax2 firmware
11:05.20Eightgoogle failed me... av168ev?
11:05.49salviadudeight: its better if i start of with the iaxy adapter, and maybe then, i can move on to better hardware?
11:06.06Eightsalviadud: You can start like me, with no hardware =)
11:06.23salviadudno hardware?
11:06.39EightI'm doing all my 'talking' to Asterisk with a software phone on my mac.
11:06.42salviaduddo i just use the modem?
11:07.12EightIf you want to talk to your home phone line, you need something with an FXO port.
11:07.19Zeeekwho uses tftp?
11:07.20EightIf you want to talk to your home phones, you need something with an FXS Port.
11:08.09salviadudi guess i'd go for the fxs
11:08.23salviadudi am serious about this. i could get a raise or something
11:08.23Eightyou can go for both.
11:08.37salviadudasterisk is getting big
11:08.53Eightthe Sipura 3000 has both ports. Or you can use the TDM400P card with a module of each type.
11:09.07salviadudthe sipura then...
11:09.23salviadudim gonna work some over time, hehe
11:10.18nine76g'night all
11:10.43Zeeeknighty night
11:11.09salviadudyeah, rest your back
11:11.42*** join/#asterisk Newbie___ (~some@211.24.146.11)
11:12.00salviadudeight: another thing, i was looking at the asterisk page, and it mentions bison, to build it
11:12.04salviadudwhat is bison?
11:12.11Eightis there some standard conference system I'm not finding that uses the default sound files?
11:12.13salviadudsounds like the guy from street fighter
11:12.50Eightsalviadud: it's a build tool.
11:12.55Newbie___guys, i need help on setting up SPA 2000, i manage to ping SPA IP, but i cant access its config page
11:12.57Eight(short version)
11:13.08Newbie___arh
11:13.12Newbie___never mind is done
11:13.14Newbie___heheh
11:13.31salviadudis bison like GCC?
11:13.39Eightsalviadud: % man bison
11:13.54salviadudthanx
11:14.57*** join/#asterisk djorange (~djorange@68-64-220-119.lmdaca.adelphia.net)
11:16.27djorangehello everyone!!
11:18.36*** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it)
11:19.05Newbie___i am seeing a completely different config page of SPA 2000 mine vs the one at voxilla
11:19.18Newbie___pls help
11:24.04Eightwhat breaks with meetme if I don't have a Zaptel card installed?
11:24.55CleanerXso i've got app_realtime up and running but sip registers fail
11:25.06CleanerXwhere should I start to search
11:25.20CleanerXodbc connection is there, mapping seems correct
11:25.27CleanerXbut sip registers fail
11:25.50CleanerXpostgres has my table
11:26.01salviadudguys, you're all lots of help, good night!
11:26.56salviadudEight: thanx a lot man, thanx a lot for the attention.
11:27.10djorangewell i just setup my pbx and i live it. i'm here to answer any newbie question
11:27.17djorangelike*
11:29.40djorangemsg me with setup or installation question, but this is my first setup. i come from an altigen and telesynergy background
11:31.12modulus_i went to the school of hard knocks
11:31.40*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
11:32.20Zeeekanyone here use tftp-hpa ?
11:32.38tzafrirI used it
11:32.52Zeeektzafrir where's the config file, if any?
11:33.03tzafrirthe command-line
11:33.10Zeeekok
11:33.50ZeeekIt's installed but I can't find any doc at all about it
11:34.02tzafrirIt should have a man page
11:34.40tzafrirman tftpd
11:35.14Zeeekok I forgot the d - obvious now like dhcpd - daemon thx
11:36.31*** join/#asterisk dabba ([U2FsdGVkX@matrix.lgw.ip6net.net)
11:38.29*** join/#asterisk zotz (~zotz@24.231.32.191)
11:39.13*** join/#asterisk UrBaNLeGeNd (~Savage@202.5.145.13)
11:39.46djorangeman theres this fucken fly on my monitor that won't go away
11:39.57djorangei think i'll make it my pet
11:40.48*** join/#asterisk fitzel (~flint@p3EE3974A.dip0.t-ipconnect.de)
11:40.53fitzelMoika
11:41.19fitzelI am about to install a new server, what kernel would you recommend? 2.4 or 2.6?
11:42.08*** part/#asterisk qwerp (~abc@219.93.57.58)
11:42.25djorangeif you dont have a modem in the phone server what kind of error would you get from xlite when you try to make a call to an outside line?
11:45.17CleanerXanyone familiar with realtime app?
11:47.15EightMan, that wiki is a life saver.
11:47.23Eightthese error messages are worthless.
11:48.25djorangewhat wiki ?
11:48.31Eightvoip-info
11:48.33tzafrir~wiki
11:48.48tzafrir~wiki
11:48.57djorangeoh okay hehe
11:49.18djorangei thought u talking about wikipedia
11:49.34tzafrirwill that work?
11:49.39tzafrir~voip-info
11:49.40jbotsomebody said voip-info was the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
11:50.06Eight...to ZTDummy. How Apropos.
11:50.11EightThat's what it just saved my day with =p
11:50.13modulus_hahahhahaaa "all of your questions"
11:50.15modulus_i think not
11:50.28tzafrirWell, almost
11:50.30Eightmodulus_: sure as heck alot of them.
11:50.43EightAnd, if you think a bit longer term, it SHOULD answer all of them.
11:50.51modulus_the only thing the wiki is good for is functions in the dialplan
11:50.54EightIt's just maybe YOU might be the one who ends up answering it =)
11:51.13tzafrirThere is alway: "answer a question that is not answered in the wiki"
11:51.23Eightmodulus_: well, I disagree. There's alot of stuff I've found useful on it.
11:51.57tzafrirEight, what were the misleading error messages?
11:52.48Eighttzafrir: well, something like 'unable to open pseudo channel' got thrown, where % modprobe ztdummy; was the solution.
11:53.15EightMay not have been very misleading, but only because it really wasn't very *leading* at all.
11:53.22tzafrirWhich is a correct error message. No channel to use.
11:53.52Eighttzafrir: "No real time clock available. Zaptel hardware or ZTdummy required"
11:53.53shaZwazanyone successfully used txfax ?
11:55.00*** join/#asterisk Newbie___ (~some@211.24.146.11)
11:55.23Newbie___any one free to help me on SPA 2000 ?
11:55.31modulus_use the web interface
11:55.31*** join/#asterisk Spiro (~icechat5@196.7.14.183)
11:55.55Newbie___i did and i called, the destination phone ring, but i cant hear anything
11:56.22Newbie___not even ringing tone on SPA
11:57.42Eightsounds like the RTP data isn't getting there.
11:57.47EightI had the same symptom initially.
11:57.55EightEverything is on the same LAN?
11:58.53EightYou might want to go through and turn off the NAT-workaround stuff.
11:58.55Newbie___yes
11:59.01Newbie___oh no
11:59.06Newbie___* has static IP
11:59.23Newbie___SPA on different IP behind a router
11:59.48EightRight, Asterisk and the SPA are both on the same private LAN behind a NAT?
12:00.15Newbie___asterisk and SPA not on the same LAN
12:00.38Eightah. Then your problem is different than mine was, and I have no suggestions =/
12:01.15Newbie___ok
12:05.46*** join/#asterisk Ron-Na (~ronald@203.70.36.126)
12:06.29Ron-Namy iaxy stopped working. Does anybody have an idea how to check it?
12:16.14Newbie___ha is working now
12:19.58*** join/#asterisk jontow (~jontow@ws.woflsys.net)
12:22.13fitzelshazaw, I installed it once and got it more or less working.
12:23.34UrBaNLeGeNdwhich softphones known to works fine with asterisk windows/linix?
12:24.02fitzelUrban, I have xlite and iaxcomm sucessfully in use here.
12:24.40fitzelI never got any reliable things done with firefly
12:24.52UrBaNLeGeNdi have setup windows messenger and linphone but it can't works!
12:24.59ManxPower~docs
12:25.00jbotDocumentation 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
12:25.20*** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net)
12:25.23Mother_hi all
12:26.11shadebobhi
12:27.17UrBaNLeGeNdhello Mother_
12:27.46shadebobI have a HP Server with 2 TE11OP and a PCI soundcard. But I have only 2 "small" pci and 3 "long" (pci X or 64 bit I think)... Can I plug TE11OP on a "long" pci slot?
12:28.15Mother_hi there
12:28.15fitzelHas anybody installed a digi datafire 4 channel isdn succesfully?
12:28.27UrBaNLeGeNdcan you help me Mother_?
12:28.31Mother_what with?
12:28.56UrBaNLeGeNdjust little bit!
12:29.01dabba>fitzel only with swyx not asterisk
12:30.10Mother_what is thy bidding?
12:30.48ManxPowershadebob, Does the card fit in the slit?
12:31.15ManxPowershadebob, Does the card fit in the slot
12:31.21Mother_LOL
12:32.00Mother_ManxPower, they told me they're not recruiting right now btw
12:32.13Mother_:(
12:32.25ManxPowerMother_, It's 6am.  I have no idea what you are talking about.
12:32.43Mother_you were looking for a job in EU no?
12:32.48ManxPowerMoAh.  Yes.
12:32.55Mother_and you should either get some sleep or more coffee
12:32.58Mother_hehe
12:33.10ManxPowerMother_, working on drinking the coffee.
12:33.17Mother_OK
12:34.53*** join/#asterisk eye69 (magnus@upcore.net)
12:35.34eye69If I have an s,1 extension, and then in the same context have another s,1 extension, will the latter one override the first one?
12:35.43Mother_no
12:35.48eye69I'm doing timebased stuff, hence the question.
12:35.50ManxPowereye69, the first one will be the one called.
12:37.10*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
12:37.21fitzeldabba, I need to get first the hardwaredrivers running.
12:37.33UrBaNLeGeNdi have setup asterisk with softphones
12:37.52UrBaNLeGeNdwindows messenger and linphone but its not working
12:38.12shadebobManxPower : Yes card fit in slot
12:38.16UrBaNLeGeNdactually working b ut not working in a way i think something missing
12:38.36shadebobManxPower : I had call hp ... It's PCI-X slot...
12:39.10shadebobManxPower : I'm affraid to explode my TE110P ;)
12:39.13eye69Mother_, ManxPower: Ok, thanks.
12:39.39ManxPower~google site:lists.digium.com TE110P PCI-X
12:39.54ManxPower~google site:lists.digium.com PCI-X
12:40.11shadebobok ;)
12:40.41*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
12:41.29Mother_lol
12:41.49Mother_off-topic: suggestions for external firewire drives? I need to get one to carry backups around
12:42.25EightMother_: My sandisk firelite has done well for me.
12:42.40Eightnot exactly a comprehensive comparison, but no complaints.
12:42.43fitzelMother, German Aldi had one today on sale. I have one here. Firewire and usb2.0 250gb for 139 euro.
12:43.44Zeeekyou are in MN?
12:43.50Eightya
12:43.53ZeeekI vas born dere
12:44.00Eight=p
12:44.04Zeeekund I'll be zere in May
12:44.30Zeeekund I buy a Polycom SIP phone dere to take back to de olde country
12:45.17Mother_Eight, fitzel: thanks a lot for those
12:45.27Eightgrr. Their little availability app acts like it has numbers in MN, but then gives an empty list of area codes =p
12:46.06ZeeekEight are you in the twin cities?
12:47.08EightZeeek: aye
12:47.13ZeeekYAY
12:49.08Zeeekare there any exciting shops for voIP stuff in the TC?
12:49.20Eightno idea.
12:49.39EightTrying to find out about local stuff myself.
12:50.02ZeeekThere is a mail order place called ATTCOM or something (don'thave it in front  of me)
12:51.02ZeeekVP has them I think
12:51.06Zeeekor ICH
12:51.25Zeeekwhat do you need exactly
12:51.28Newbie___everytime when i use SPA 2000 to dial, i must enter a # after the last digit , is that a must ?
12:51.44Zeeekthat's in the SPA2000 dialplan, no?
12:51.57Eightoh wow. Who's bright idea was it to put EVERY area code in north america in a single combo box? =p
12:52.01shadebobManxPower : For information : it seem PCI-X slot compatibilty is ok with "older" PCI card  (http://groups.google.fr/groups?hl=fr&lr=&threadm=3FBFCB24.96AE9B24%40sympatico.ca&rnum=1&prev=/groups%3Fq%3Dplug%2BPCI%2Bcard%2Bon%2BPCI-X%26hl%3Dfr%26lr%3D%26selm%3D3FBFCB24.96AE9B24%2540sympatico.ca%26rnum%3D1). I wait for validation by support@digium.
12:52.13Newbie___Zeeek: i do not see any # in SPA dial plan
12:52.16ZeeekEight - did you try VP or ICH?
12:52.34EightZeeek: ICH?
12:52.35ZeeekSPA is Sipura? There is certainly a dialplan
12:52.46Newbie___(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
12:52.47ZeeekIConnectHere does SIP
12:52.50Eightand VP doesn't offer MN either.
12:52.58Newbie___Zeeek : it looks alien to me
12:53.03Zeeekah non? Thought they did
12:53.26ZeeekNewbie there is probably something the can be added so it dials without the #
12:53.46hajekdVoipJet IAX gateway is not reachable..
12:54.01Newbie___so is the culture of SPA to use # terminator then
12:54.16fafnirhey, anyone here a grammer nazi?
12:54.18ZeeekGrandstreams can use it too if thyey wish I think
12:54.24elricI just read on the mailing list about TE410P drivers for FreeBSD being available but no information as to where I can get them. wiki says they are still not implemented. Google doesnt give anything either.
12:54.35Eightheh, ICH's form breaks in my web browser.
12:54.49EightWhy do people have to be *special* about number availability interfaces.
12:55.18ZeeekLet me just say that I have used ICH - they are very consumer (not geek/clued users) but the network is good and quality is high. Price isn't cheap though.
12:55.33Zeeekyou want 612 or 952 ?
12:56.15Eighteither would do.
12:56.30ZeeekI think they've got one at least
12:56.48ZeeekInterestingly enough I just got something from them about reselling
12:56.48Eightoooh, I have to select a plan option first.
12:56.52*** join/#asterisk tzafrir (~tzafrir@62.90.10.53)
12:57.02Zeeekyou want me to look it up - I have an account
12:57.18Eightnah, it's working now.
12:57.35Zeeekyou want this for you or as a service provider/reseller
12:57.37Zeeek?
12:58.14EightThe former first, the latter potentially.
12:58.44EightAha. 612-605-xxxx
12:58.54Eightwith unlimited national calling as an option.
12:59.44Zeeekthey don't do unlmited AFAIK
13:00.02EightAgh. They do unlimited, but not in their "Open access" stuff.
13:00.16ZeeekI have a par per call acct
13:00.21Zeeekpay per call
13:02.32EightThanks for the heads up on ICH.
13:02.52Zeeekk
13:03.08Eightnow if only they had unlimited (local atleast) calling =/
13:04.07Zeeek$30/mo unlimited
13:04.22Zeeekincl DID
13:04.34EightZeeek: but that doesn't work with Asterisk in their 'open access' thing does it?
13:04.41ZeeekI think it does
13:04.49Eighthttp://iconnecthere.com/nonmembers/eng/broadband_phone/index.htm
13:05.01Zeeek<PROTECTED>
13:05.20Zeeek"The iCall Unlimited calling plan is a Broadband Phone Service* monthly package offering unlimited** calls to the US and Canada for only $29.99 per month. This plan also includes a free local US number where you can be reached, free voicemail and advanced telephony features."
13:05.22Eight"* This calling plan is not available for customers on the Open Access Program"
13:05.45Zeeekwhat is open access anyway?
13:05.59*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:06.02Eightlets you use any user agent (like asterisk) instead of their 'locked' SIP boxes.
13:06.09EightThat's my understanding, anyways.
13:06.14Zeeekfunny I got to that page thru the links in openaccess
13:06.44ManxPowerSo it's the usual "you can't get unlimited unless you use their device"
13:07.01Zeeekyeah I do see that part now
13:07.05EightManxPower: yup =/
13:07.15Zeeekhowever I'l take a look at the latest reseller lit
13:07.54Eightthe 10.95 for 1,000 minutes isn't bad though.
13:09.23Eightah, plus 9 bucks for the phone number.
13:12.02*** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com)
13:12.48ClayReiche123Can someone help me with IAX trunking?
13:13.15ClayReiche123I've been trying to get this to work for days now with varying degrees of success.
13:13.32jontowclayreiche123; whats the issue, and how are you trying to set it up?
13:14.06jontowi've been playing with just that for the last few days, with varying degrees of success.. but mostly with success.
13:14.41*** join/#asterisk nirs (~nirs@62.90.49.115)
13:14.47nirshello all
13:14.50nirsanybody home today ?
13:15.26Zeeekya
13:16.03ZeeekManx know anything about DHCP and tftp?
13:16.28ZeeekI just installed tftp-hpa
13:18.40ClayReiche123nevermind.... I'm on CRACK.
13:18.55*** join/#asterisk AldeBaran_ (~allen@196.35.203.121)
13:19.01AldeBaran_hi
13:19.24Zeeeklo
13:20.25shaZwazManxPower: do you use txfax ?
13:20.45ClayReiche123wait... maybe I'm not on crack.... posting to pastbin.... 1 sec
13:20.48jontowCRACK you say
13:20.55jontowlemme get down, its too early :)
13:20.58*** join/#asterisk d00gster (~in_ter@67.71.121.232)
13:21.04Zeeekcall JerJer
13:21.08AldeBaran_this is probably a very newb question... and in the faq but I prefer IRC to web... so... *deep breath*... is it possible to use a number of soft phones and 1 analog line?
13:21.22jontowaldebaran; sure..
13:21.33jontowaldebaran; but keep in mind you're limited to your single channel analog line if you're doing in or outbound calling
13:21.42AldeBaran_naturally :)
13:21.44jontowyou could have 800 soft phones and 1 analog line if you wanted
13:21.46shaZwazunless all of them are trying to use that 1 line :)
13:21.52jontowdoesn't mean it'd be efficient ;) but you could do it.
13:22.20AldeBaran_what hardware would one use to connect the one analog line? the cheaper the better...
13:22.25jontowa digium FXO card
13:22.36jontowcan be had on ebay at the moment.. well, not the digium card; but a clone
13:22.46tzafrirAldeBaran_, But some "tricks" will be required if you want outsiders to be able to call directly to you internal extensions
13:22.48jontowi have had ok luck with my FXO clone card.. think i paid $40 for 3, shipped
13:22.49shaZwazSoftClient - > Analog doesn't sound too convincing tho
13:23.41jontowbest bet is to buy a TDM400P, i believe.. bit pricier, but waaay nicer than the single port FXO cards :)
13:24.01AldeBaran_jon: it has 4 ports?
13:24.21jontowyep
13:24.30jontowwith changeable 'modules' to do either FXS or FXO, iirc
13:24.46AldeBaran_nice.. price? ballpark
13:24.46jontowFXS being "plug an analog phone into it", FXO being "plug an analog line into it"
13:24.52jontowcheck digium.com for pricing.. i remember not.
13:25.02jontowi think it was a couple hundred $$ at least
13:25.03AldeBaran_'k
13:25.18shaZwazaround 340 USD from digium
13:25.32jontowbut the digium products are definitely worth it.. i've worked with 3 T100P cards and had 0 trouble other than my own learning curve ;)
13:25.32shaZwazmay for less at ebay
13:25.41AldeBaran_heh
13:26.05jontowoh, and the bonus: buying directly from digium helps asterisk :)
13:26.16AldeBaran_sure
13:26.51jontowso, for $XXX, you get the card and a warm fuzzy feeling.. who can beat that :D
13:27.15AldeBaran_what is a good softphone to use, windows I guess, free, OSS if possible
13:27.24jontowx-lite works good.. so does sjphone
13:27.28jontowxten.net and sjlabs.com iirc
13:28.31AldeBaran_is it possible to use a modem as a poor mans FX[OS] ?
13:28.35shaZwazalso firefly if you go for IAX
13:28.46AldeBaran_(last question) ;)
13:28.50jontowhmm.. some modems i've heard thats possible with, most not
13:29.02jontowi don't have specific models.. but check http://www.voip-info.org/ for the list.. its there somewhere :)
13:29.10jontowmostly the 'linmodem' type
13:29.49*** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com)
13:30.13johnnybHow do you set up outgoing rollover?  I have everything working perfectly with one line outgoing and multiple lines incoming, but I don't know how to get multiple lines outgoing.
13:30.30jontowjohnnyb; setup a trunk group :)  you use a T100P or similar?
13:30.44jontow.. or FXO cards?
13:30.47dabba>AldeBaran http://www.voip-info.org/wiki-Asterisk+Hardware
13:31.05AldeBaran_thanks for everything :)
13:31.12jontowvery welcome
13:31.16johnnybTDM400p
13:31.33jontowaha, a trunk group should work ok then
13:31.42*** join/#asterisk jcims (~jcims@rrcs-24-172-217-2.central.biz.rr.com)
13:31.56johnnybIs that in extensions.conf or zapata.conf?
13:31.59jontowzapata.conf
13:32.03jontowsomething like:
13:32.05jontowgroup=1
13:32.10jontowchannel => 1-2
13:32.26Zeeekor add group=1 in each channel
13:32.30jontowthat'd make a group of channels 1 and 2.. you'd reference it liek Zap/G1
13:32.34jontowexactly :)
13:32.47ZeeekZAP/g1/0123456789
13:33.53Zeeeksomeone I need a few secnds help on tftp ?
13:34.27jontowwhats up? :)
13:34.51ZeeekI installed tftp-hpa and run dhcpd
13:35.06ClayReiche123OK...man that took a while... http://pastebin.ca/7075 I'm trying (and failing) to set up an IAX trunk.
13:35.09Zeeekq1: do I need to specify next-server if the ip is the same?
13:35.35Zeeekq2: since it didn't work, I spedified next-server and it still doesn't work
13:35.59Zeeeknetstat shows everyone listening ornt he right ports
13:36.13ClayReiche123basically I'm trying to forward unanswered calls to another asterisk server that handles voicemail.
13:37.05*** join/#asterisk boch (~as24@200.59.172.98)
13:37.41*** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com)
13:39.08UrBaNLeGeNd\quit
13:40.37jontowok.. zeeek, got a bsd machine handy?
13:40.48jontow(or a tftp client on linux.. they don't come with it by default)
13:40.56jontowif so.. tftp ip.address.here
13:41.02jontowget WHATEVERFILE.cnf
13:41.08jontowif it receives, it works.. otherwise its busted :)
13:44.51ClayReiche123I got this to work on a different "CALLED SERVER" running cvs HEAD from a couple months ago.
13:45.00ClayReiche123with the same config files.
13:49.26johnnybjontow: thanks, I got it working.
13:49.43ClayReiche123I'm jealous....
13:51.41Darwin35<PROTECTED>
13:51.55johnnybIs there any way to get the budgetone phones with the built-in hub to prioritize its own packets over that of the PC its connected to?
13:52.04epochDarwin35: I think you misspelled "epoch"
13:52.34johnnybWe're having a problem that when a user is doing a download, his own voice cuts out at the other end, but is able to hear just fine.
13:53.17*** join/#asterisk boch (~as24@200.59.172.98)
13:54.38AldeBaran_bbl
13:54.42*** part/#asterisk AldeBaran_ (~allen@196.35.203.121)
14:00.37*** join/#asterisk afe ([WkojmpP0y@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
14:06.31Ron-Namy iaxy stopped working. Does anybody has a hint?
14:07.08sambalbroken power supply?
14:07.40Ron-Nano,
14:08.47Ron-Nait does not register and I canot ping it with the last known IP address
14:10.24*** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
14:13.57ClayReiche123Hi MikeJ.
14:14.07MikeJ[Jayden]hey
14:14.22tzangermorning MikeJ[Jayden]
14:14.28ClayReiche123I'm still wrestling with this IAX trunk... got a minute?
14:15.01MocClayReiche123, Dont forget you need zaptel loaded + timming card also configured or ztdummy started
14:16.00ClayReiche123MikeJ: Don't know if you remember, but I couldn't get it to work with Stable, then tried a HEAD server I had and it worked.
14:16.37MikeJ[Jayden]I don't recall...
14:16.45MikeJ[Jayden]I do vaguely recall
14:17.00ClayReiche123MikeJ: That HEAD server was not beefy enough so I installed HEAD on a new server and am getting no conection. http://pastebin.ca/7075
14:17.20*** join/#asterisk xyztail (~ska@cable-68-114-110-178.sli.la.charter.com)
14:17.25ClayReiche123Same config files though....
14:18.20xyztailis there a debian src package that is more current than the testing version?
14:18.42xyztailfor * of course.
14:19.08*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
14:19.08*** join/#asterisk santiago (~santiago@63.245.86.95)
14:20.03MikeJ[Jayden]socket_read: Immediately destroying 1, having received INVAL
14:20.10MikeJ[Jayden]that seems to be cut off
14:20.16MikeJ[Jayden]are both servers head?
14:20.23ClayReiche123MikeJ:That's the way it displays
14:20.28ClayReiche123no
14:20.45MikeJ[Jayden]which is not head?
14:21.08ClayReiche1231 HEAD and 1 Stable. The same way I got it to work last Thursday. Calling server is stable
14:21.24MikeJ[Jayden]we know tzanger
14:21.37ClayReiche123MikeJ: It's the same Calling server I was using before.
14:21.53ClayReiche123I just changed the Called server.
14:21.59MikeJ[Jayden]and called server is recent head?
14:22.09ClayReiche123MikeJ: different version of HEAD though.....
14:22.17MikeJ[Jayden]date?
14:22.17ClayReiche123Yes this morning.
14:22.20nextime8 days to the timestamp party!
14:22.35ClayReiche12303-09-05
14:22.36nextime( 1111111111 )
14:23.03MikeJ[Jayden]tzanger, did you notice anything going in to chan_iax in the last few days?
14:23.57MikeJ[Jayden]clay, chan_iax2.c:6689
14:24.07MikeJ[Jayden]can you peek and see if there is anything obvious there?
14:24.21MikeJ[Jayden]I have some stuff to do this am... bbiab
14:24.26*** join/#asterisk spackle (~spackle@209.234.83.19)
14:24.52*** join/#asterisk Eight (~blake@12-205-155-39.client.mchsi.com)
14:24.58ClayReiche123I don't know what that means Mike.
14:25.37ClayReiche123You mean like firewall?
14:26.37ClayReiche123oh... in the code...I'll do my best...
14:27.06*** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co)
14:29.05*** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3)
14:30.50tzangerI haven't updated CVS in a while
14:30.52tzangerI've been sick
14:30.54dabbacan anyone suggest a way to transfer a call to Monitor and back to teh transfering source, i.e to record a call when needed rather than recording all
14:32.28CosmicRaygood morning everyone.  after having played with asterisk a bit last night, I must say: *very* nice piece of software.  if any developers are here: you guys kick ass.
14:33.46CosmicRayI am wondering if there is a reference to the extensions.conf anywhere or, better yet, a reference for all the config files
14:34.05CosmicRayI have noticed how to do a few things (like ring multiple phones at once) by finding examples with google...
14:34.11CosmicRayI figure a reference would be better
14:34.17ariel_morning all
14:34.25CosmicRayhello ariel
14:34.51*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
14:36.45shmaltzCosmicRay, check out voip-info.org
14:36.48shmaltz~docs
14:36.51jbotDocumentation 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
14:37.05*** join/#asterisk coldfeet (~c@213.78.240.109)
14:37.28coldfeetmorning /afternoon to one and all
14:38.01Luhiwuanyone knows where can i buy a GSM-to-POTS converter?
14:38.23*** join/#asterisk bobx (~bobx@206.124.165.14)
14:39.01CosmicRaythank you shmaltz.  The digium link I hadn't seen yet, and it may have what I'm after.  I had already checked the wiki and asteriskdocs.org
14:39.08CosmicRayfwiw, it would be nice to link to these from asterisk.org
14:39.21*** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu)
14:40.50coldfeetguys not sure if neone frm last nite is here, but I am having a problem registering one of my xlites with asterisk
14:42.01CosmicRayalso, is there a reference somewhere on implementing comming features like *67 (disable caller id), etc?
14:42.08CosmicRaythe sample extensions.conf doesn't seem to mention it
14:42.26coldfeetmy config/debug etc is on http://pastebin.ca/7053
14:42.52EightI don't suppose anyone knows a handy trick to kick SJPhone into using a specific external IP in its SIP headers?
14:43.15MikeJ[Jayden]yeah.. I managed to make coldfeet's problems worse... anyone else around to give a try?
14:43.34coldfeetMike ur still here :-)
14:43.42coldfeetwow i've been to sleep
14:43.51MikeJ[Jayden]me too
14:43.53MikeJ[Jayden]at work now...
14:44.04coldfeeti hv thought of another way around it, basically what I am aiming to do is...
14:44.52coldfeetroutre 0845 numbers via ser--->asterisk, asterisk is needed to change set callerID, so a callcenter knows what line is being dialed if they support multiple clients, this call then is directed to a IP phone
14:45.07coldfeetthis IP phone is the xlite I am trying to register,
14:45.41coldfeetI read somewhere that xlite has aprob with REFRESH ...whatever that is
14:46.50coldfeetneone wanna try to login to my asterisk with xlite to see if it works frm there end let me know
14:47.21*** join/#asterisk nestAr (nester@makes.all.the.girlies.go.wewt.wewt.net)
14:47.23nestArgah
14:47.34nestArfreenode is annoying sometimes.
14:50.34*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
14:50.34*** mode/#asterisk [+o anthm] by ChanServ
14:54.29*** join/#asterisk ionix (~ionix@209.71.254.100)
14:54.38ionixre :)
14:55.59coldfeetis neone out there :)
14:56.13Eightwell, if by 'refresh' you mean 'totally ignores the SIP proxy' ya.
14:56.35EightThe thing happily reports its logged into the SIP PRoxy...
14:56.40Eighteven if I don't have asterix RUNNING =p
14:57.02clive-does anyone know if the sipura 2100 can support 2 channels of G729 ?
14:58.57EightThis NAT thing shouldn't be this difficult... ::grumble
14:59.43*** join/#asterisk channan (~channan9@66.180.121.185)
14:59.58coldfeetis it possible to telnet to asterisk and send a SIP register request
15:01.02Eighttelnet is tcp.
15:01.08Eightyou'll just get a 'connection refused'.
15:01.13MikeJ[Jayden]moning anthm...
15:01.17coldfeetoh yeah
15:01.17MikeJ[Jayden]morning that is.
15:01.28anthmhi
15:03.58Eightcoldfeet: I'm not sure if it's your problem... but as I mentioned I've had Xlite just all of a sudden refuse to be prodded into talking to the server. I've ended up using SJPhone instead.
15:04.55coldfeeteight gonna try with a IP phone later, same happened here it was okay, and then it went pear shaped
15:05.23EightI was going to trash the prefs file and start again, but then I didn't find the prefs file =p
15:05.37coldfeetprefs in xlite
15:08.31*** join/#asterisk Moc____ (~mochouina@64.235.210.66)
15:08.38Moc____Hail
15:11.05Eightcoldfeet: find wherever xlite saves its config data... trash it, and reenter.
15:11.21EightI did that, and x-lite is working for me again.
15:12.59*** join/#asterisk Inferna (~sasha@194.158.51.171)
15:13.49coldfeetokay let me try
15:13.54Infernacan anybody tell me if asterisk keeps payload info for SIP clients? i can see that asterisk changes payload to 20ms for g711, is there a way to change that?
15:13.55shadebobwww.sineapps.com/news.php will be updated ?
15:15.18coldfeetis it the .dat file
15:17.07Inferna> The quick and dirty way:
15:17.08Inferna> ------------------------
15:17.08Inferna>
15:17.08Inferna> In rtp.c, function "ast_rtp_write", in the "switch" statement,
15:17.08Inferna> "AST_FORMAT_G729A" case, change the smoother creation to something
15:17.08Inferna> larger. E.g.:
15:17.09clive-inferna there is a way, just cant remeber exactly what it was, there is some value you have to change in the code, and the recompile.
15:17.10Inferna>
15:17.12Inferna>     rtp->smoother = ast_smoother_new(40);
15:17.14Inferna>
15:17.16Inferna> Keep in mind that you must set this into something valid
15:17.18Inferna> (45 obviously is not valid). Recompile and you should be fine.
15:17.20Infernais there a way to do it better?
15:17.26Infernao\hh :(
15:18.39Inferna<PROTECTED>
15:18.40Inferna<PROTECTED>
15:18.40Inferna<PROTECTED>
15:18.40Inferna<PROTECTED>
15:18.42Infernafound
15:19.28fitzelHas anybody installed a digi datafire 4 channel isdn succesfully? I try to get it running in debian.
15:19.39eKo1Inferna: pastebin
15:19.41Infernait's good to have this opportunity on per client basis
15:19.49Infernawhen it will be develloped?
15:19.59tzangerstevekstevek: werd?
15:21.44*** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net)
15:22.13Infernaanybody is familliar with SDP?
15:23.02tzangerstevekstevek: wake up man :-)
15:23.14*** part/#asterisk jcims (~jcims@rrcs-24-172-217-2.central.biz.rr.com)
15:23.26tzangerstevekstevek: come on over to -dev
15:23.58ManxPower.
15:26.33Infernamy company can pay 2k$ for this patch
15:26.37Infernaanybody is ineterested?
15:27.07nestAri love the smell of commerce in the morning..
15:27.21tzangerhaha
15:27.26Infernait's already evening here
15:27.29tzangerInferna: post a bounty
15:27.36Infernaand i really need that
15:28.10*** join/#asterisk wildcard0 (~generic@S0106006097e16040.vc.shawcable.net)
15:28.38Infernaok 3k$
15:28.42Infernais it enough?
15:29.03tzangerInferna: post a bounty, as I said
15:29.20*** join/#asterisk loud (~ariel@null0.flapping.net)
15:29.56clive-inferna what do you want this thing to do
15:32.18*** join/#asterisk tih (tih@athene.hamartun.priv.no)
15:32.37anthmhmm you want asterisk to be able to pipe mms into itself ?
15:32.54*** join/#asterisk fishboy1669 (proxyuser@62.69.81.129)
15:33.31anthmor you want asterisk to emit mms to ppl's client
15:33.38fishboy1669anyone here any experience of ip2006 phones from tecom
15:33.39fishboy1669?
15:34.03tzangerwhat's mms?
15:34.10tzangeranthm can probably code this up in his sleep
15:34.21tzangerand as anthm's official manager, I get 15%
15:34.46*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
15:34.46anthmits the media server streaming protocol
15:35.55fishboy1669is that like sms but for meadia?
15:36.39fishboy1669any one heard of teh ip2006 tecom phones?
15:36.43anthmstp will tune into a media stream and record it to a file
15:37.23louddid voipjet go poop ?
15:37.51*** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net)
15:38.19wildcard0loud, seems so...according to their website
15:38.49loudthanks man
15:38.54coldfeetEight where was the config file
15:38.56*** part/#asterisk pcm (~pcm@user-69-73-0-22.knology.net)
15:39.35Eight~/Library/Preferences/com.xten.SIPsomething
15:39.45Eightbut I'm going to guess you're not running Mac OS X =)
15:39.54loudAh, all back to normal now.
15:39.57EightSo, I've got a weird one....
15:40.23EightI've got two people connecting with x-lite via SIP into Asterisk
15:40.43EightWhen we go into the conference, we can hear eachother.
15:40.48spackleAnyone using the citel devices to convert legacy phones to SIP?
15:40.56Eightwhen we dial eachother with Dial(SIP/user) he can hear me, but I can't hear him.
15:41.37Eightregardless of who is doing the dialing.
15:41.52Eightechotest works for both of us as normal, as well.
15:42.01Luhiwuanyone knows who sells termination and accepts paypal payment?
15:42.08*** join/#asterisk Alexi1 (~alexis@www.trim.it)
15:42.09wildcard0spackle, i've played with them
15:42.15Alexi1hi
15:42.31wildcard0Luhiwu, voipjet...who's down right now in nyc.
15:43.03Luhiwuwildcard0, voipjet is the one who i'm trying to replace :-)
15:43.13wildcard0ah
15:43.20spackleWildcard0, what brand of phone did youtest with?  We have Nortel.
15:43.35wildcard0spackle, i think i was using nortel also
15:43.55loudthe new server seems to be ok, i have 15 ms to it.
15:43.56spackleWildcard0, and what would you say about them?  Run away?
15:43.58wildcard0Luhiwu, unhappy with them?  price too high?
15:44.09*** join/#asterisk freat[laptop] (~freat[lap@12.10.34.36)
15:44.21wildcard0spackle, in my exprience .. flakey and high cost per port
15:44.25Alexi1ok i have an * server and i use windows mesenger as a sip client but the * (which is in verbose mode) is still saying : Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.1.113
15:44.26wildcard0YMMV :)
15:45.15spackleYeah,  The price per port is out rageous at 3k+ but they offered better prices when contacted directly.
15:45.21Alexi1and i can't call to my windows messenger, but i can call from it
15:45.57wildcard0spackle, it was still >2k even when we talked to them directly.  though you may have gotten a better deal
15:46.15Luhiwuwildcard0: very nice price, but had some troubles calling to Mexico and now they are down... i've opened my account 3 days ago and already had 2 problems, maybe i'm just unlucky.
15:46.26wildcard0spackle, honestly, i can get ~20 ip phones for that
15:46.52BeirdoLuhiwu: I'm not terribly impressed with voipjet either
15:46.52wildcard0Luhiwu, ah. good to know.  i've never tried them.  i just saw their web page
15:46.56spackleWildcard0, exactly.
15:47.20*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfni3.dialup.mindspring.com)
15:47.20*** part/#asterisk santiago (~santiago@63.245.86.95)
15:47.50spackleWildcard0, how long ago was it you worked with them?  A few firmware updates can make a difference - unless you are Grandstream ;-)
15:48.02*** join/#asterisk jarrod (jarrod@dipole.informationwave.net)
15:48.07LuhiwuBeirdo: did you have many problems with voipjet?
15:48.14Beirdostill do
15:48.18jarrodhey when does SIP estbalish the rtp session
15:48.22jarrodafter the invite?
15:48.27Beirdothey flake out at regular intervals for me
15:48.38wildcard0spackle, hehe.  it was about 3 months ago.  even if they got rid of the flakey, i'd rather recommend ip units to the customer than a hack to make the old stuff work at the same price
15:49.23spackleFor claming they are so well connected VOIPjet is abysmal compared to Oh, say, Nufone ;-)
15:49.25*** join/#asterisk hemant (hemant@220.226.51.31)
15:49.45Alexi1Someone here know what is a freebox ?
15:49.46wildcard0spackle, it really needs to be at the 800-1k price range for me to be looking at it as a serious solution for migrating off nortel
15:50.21wildcard0Alexi1, http://www.as220.org/jb/freebox/ ?
15:50.22Beirdospackle: I so agree :)
15:50.33spacklewildcard0, thanks for the perspective.  For me the savings would be in keeping what is in place, in place until it could be migrated.
15:51.01spackleWildcard0, still not worth it - as you point out.
15:51.23wildcard0spackle, totally depends on your installation.  i imagine there may be instances that it works perfectly.  it's not a -bad- product.
15:51.52Alexi1wildcard0: no, tx anyway
15:52.05spackleNo, certainly a niche for things like Citel.
15:53.32fishboy1669whats the best phone to use for *
15:53.38fishboy1669sip
15:53.51jontow* is good with alot of phones, i wouldn't tie it to one model :)
15:54.00jontowevery phone has its advantages and disadvantages.
15:54.48wildcard0fishboy1669, on a LAN, i'd say look at polycom and pingtel for quality, grandstream and sipura for price
15:55.01wildcard0but there are many
15:55.08JerJer[mobile]7960 accept no less
15:55.14wildcard0those just happen to be popular
15:55.25wildcard0see?  we all have opinions.  i personally hate the cisco phones :)
15:55.27spackleHow are pingtel phones configured?  From www or tftp?
15:55.37*** part/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
15:55.58wildcard0spackle, the one i have is via www.  though i believe there's a tftp provisioning system i've never played with
15:55.59spacklewildcard0, I just hate the ci$co prices.
15:56.09*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
15:56.44fishboy1669im trying ip2006
15:56.53fishboy1669does anyone use them
15:56.53wildcard0ya.  i -know- they want you to use it on the cisco system.  so im afraid (like many other cisco products) it works better using a cisco server
15:56.58fishboy1669or have knowlege of them
15:57.19wildcard0the tecom one?
15:57.27Zeeekyo yo fish
15:58.26EssobiWhat what?
15:59.05*** join/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net)
15:59.17Essobispackle If you have the cisco $$, try buying one of their call managers or voicemail platforms.
15:59.19wildcard0fishboy1669, from what i remember it didn't have a lot of options.  only like...2 codecs?  and the jitterbuffer didn't work too well.  but that way a -long- time ago so it may have improved significantly by now
15:59.19Essobi:)
15:59.44wildcard0heh i dispise cisco call manager
15:59.48*** join/#asterisk djflux (~djflux@207.250.204.185)
16:00.22EssobiJerJer[mobile] Hey umm.. You heard of anyone getting DTMF to work in a H323 <-> SIP bridging scenario with the h323 driver?  Last time I tried, I could get it to work..
16:00.35Essobis/could/coulnd't
16:01.53*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
16:03.00spackleRemember when buying a hardware product meant getting free firmware forever?
16:03.55*** join/#asterisk insomni (~insomni@82.192.163.226)
16:04.03EssobiAlso.. I was thinking about using your suggestion and getting a 33XX cisco router with the IP-to-IP IOs on it, and using it to gateway H323 to sip, but my h323 gateways olny use slow start, and I'm wondering if it'll be a problem getting * to send a proceeding signal to the router when no SDP has been issued yet.
16:07.36fishboy1669hi zeeek
16:08.14fishboy1669zeek hows things
16:08.40fishboy1669wildcard0 i have the full codecs on it there is a code u put in
16:10.01Zeeekeverything is everything
16:10.15Corydon-wAnybody know if the G.729 codec licensed from Digium is Annex A or B?
16:12.43eKo1a
16:14.03Corydon-wIt doesn't negotiate?
16:14.05*** join/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it)
16:14.08GiabboOhi all
16:14.55wildcard0fishboy1669, define "full"
16:15.11junky[work]how can i checkgroup on multi machine?
16:15.46fishboy1669711 u and a 729 723
16:16.11*** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com)
16:16.31Skyskyjust wanna ask if iax is tcp or udp or both
16:16.46wildcard0udp
16:17.27BoRiSudp only
16:17.48Skyskythx~
16:18.17coldfeetEight sorry boss came in, did my head in, u still there, am on win2000
16:18.28Eightstill here.
16:19.02*** join/#asterisk riksta (~rick@81-178-195-88.dsl.pipex.com)
16:19.44*** join/#asterisk gr8nash (~basketoju@mamabear.si-forest.com)
16:20.09coldfeetso eight where was/is the prefs file, itrs not under program files-->xlite
16:22.51*** join/#asterisk Grooby (~Grooby@12.22.232.212)
16:23.07Eightcoldfeet: I have no idea. It's in a diff' place on a Mac than a PC.
16:24.33coldfeetokay let me google a bit
16:24.51*** part/#asterisk Grooby (~Grooby@12.22.232.212)
16:28.10fishboy1669zeek anything exciting happning lately?
16:28.25GiabboOcan anyone help me with asterisk + mysql ?
16:29.41eKo1What is it?
16:29.47*** join/#asterisk amir (~amir@shield.guindehi.ch)
16:30.13tzangerhmm
16:30.16tzangerin the CDRs
16:30.20tzangerthere's no easy way to match up calls
16:30.32tzangerI mean I have two * servers over an IAX2 link
16:31.06tzangerI see a call in A's log as Zap/10-1 to IAX2/B-1, but that same call on B is IAX2/A-1 to Zap/25-1
16:31.10tzangerwhich is fine
16:31.16tzangerbut that A-1 to B-1 map is not always there
16:31.20tzangerantoher call is A-4 to B-8
16:33.30eKo1Well, yeah.
16:33.39*** part/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net)
16:34.42tzangereKo1: there's no easy way to line up CDRs then though
16:34.49tzangerI mean you can try to match on CID/time/duration
16:35.22eKo1Nope. There is no easy way.
16:35.24tzangerI mean on those two boxes I have the same call (lined up with date/time and CID#) but the durations are quite different
16:35.32tzanger81/73s vs 113/105s
16:35.43tzangerthat's a 30s difference!
16:35.52eKo1Why are you trunking calls anyways?
16:36.09tzangerB cannot get out to the world except through A
16:36.29tzangerand besides, notransfer is the only way to have all billing records on A accurate
16:36.42eKo1Then why not have everything on A then?
16:36.45tzangersince IAX has no way to "call back" the original connection and give actual CDR numbers
16:36.51Zeeekhas anyone checked for memory leaks in 1.0.6 ?
16:36.54tzangereKo1: because I may have a dozen Bs
16:37.39tzangereKo1: also, most of the calls on A are terminated to A's PRI
16:37.47tzangerso it doesn't need to drop out since it *is* an endpoint
16:37.53tzangerand that is the exact case with these particular CDRs
16:37.56tzangermost calls line up perfectly
16:38.02tzangeridentical CDRs
16:38.10tzangerbut some, like this one, don't
16:38.13tzangerand it's off by 30s
16:38.53tzangerphysical setup is Norstar KSU -- PRI -- *B -- SDSL IAX2 -- *A -- PRI -- Telco
16:39.56*** join/#asterisk ruiner (ruiner@ruiner.netslacking.net)
16:40.31tzangereKo1: do you know how the numbers are assigned?
16:40.40tzangerI see technology/peer-callno
16:40.54tzangerI have four callno's the same throughout the day
16:41.02ruinercan anyone help me figure out why I can dial into my asterisk box, but when i punch numbers on my phone, nothing happens?  i just have a default setup.  i call in and get the demo sound saying congrats, etc, press 2 for more technicall info, 600 for echo test.  none of it does anything
16:41.21tzangerruiner: what kind of phone
16:41.27ruineri thought it might be my cisco router because the IOS on it didn't have DTMF events through SIP signaling support, so I upgraded IOS but still nothing
16:41.36ruinertzanger: just a regular analong phone
16:41.42tzangerruiner: what interface then
16:42.03ruinerthe line plugs into a cisco 3640 with a 2fxo port card
16:42.08ruinerthen ethernet into my asterisk box
16:42.16tzangerruiner: make sure the cisco and asterisk agree on dtmfmode
16:42.28tzangerand if it's inband DTMF you can only use ulaw/alaw
16:42.44puppetyeah my uptimechoice says in hour,min,second now to ;D
16:43.10ruinerhow would i see what dtmfmode they are using?
16:43.18tzangerruiner: you've got the cisco, no me.  :-)
16:43.31ruinerheh
16:43.35ruinerin asterisk?
16:43.47tzangerruiner: sip.conf
16:43.50tzangerdtmfmode-=
16:44.40*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
16:44.41*** mode/#asterisk [+o bkw_] by ChanServ
16:44.52ruinerit's not defined to anything in sip.conf
16:44.55puppethi bkw_
16:44.58tzangerruiner: what's the default then
16:45.00tzangerinband?
16:45.15tzangersee this is why I hate the fucking wikki
16:45.22tzangerI'm trying to find the CDR csv fields
16:45.29tzangerand it's just fucking leading me around in circles
16:45.42puppetthe wikki is nice but kinda slow and i think its hard to find the complete command listing but i think ill get usede to it :)
16:46.16`Sauron"cmd <commandname>" is the thing to search for
16:46.42tzangerthere we go
16:46.43tzangerjeez
16:46.57ruinerwoot, got it working
16:48.15tzangerI should have uniqueid in the CDRs
16:48.18ruinerthanks tzanger
16:48.19tzangerI wonder why that's not in there by default
16:48.23tzangerruiner: glad to help
16:48.32ruinerjust had to set dtmfmode=inband
16:48.47tzangerruiner: remember you cannot get DTMF with inband if you're using a compressed codec
16:49.37ruinerulaw/alaw are the only uncompressed
16:49.39ruiner?
16:49.59ariel_ruiner, correct
16:50.04*** join/#asterisk file (~file@251.134.218.209.transedge.com)
16:50.09ruinerk
16:50.12*** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com)
16:50.12tzangerg726 might work but inband is always teh suck anyway
16:50.16ruineri'm stilly grean here
16:50.23filesilly inband
16:50.33tzangerruiner: no worries, you're showing a willingness to learn and that will work for you here
16:50.37ruinerwhat would be used instead of inband?
16:50.46ruiners/grean/green/
16:52.09tzangerruiner: rfc2822 or something like that
16:52.23tzangerbasically the cisco "sees" the DTMF and sends a "DTMF digit X" pressed RTP message
16:52.59tzangerhehehe
16:53.01tzangerMar  9 11:50:57 ERROR[1738]: chan_iax2.c:619 jb_error_output: Jitter value of 93053 larger than absolute maximum of 500, limiting!
16:53.05ruineri see
16:53.09tzangerI ran ntpdate JUST as a call came in and was answered
16:53.31tzangerntpdate to get the clock set, then ntpd to keep it right
16:53.31LoRezyou should be running ntp all the time
16:53.32filecan you recreate it?
16:53.37ruinerootherwise, the cisco just passes on every it receives, basically?
16:53.41tzangerfile: can I recreate what
16:53.51filecan you recreate the jitter buffer error? ;)
16:54.06tzangerruiner: no otherwise the cisco doesn't try to detect anything and just sends it as audio
16:54.08GiabboOcan anyone help me with asterisk + mysql ? i'm using asterisk + app_addon_sql_mysql
16:54.12tzangerfile: sure, if I do that again...
16:54.19filehaha cool
16:55.00tzangerand if the clock's off that much again
16:55.18filewe have cooling in our hotel room now via a fan+window, since our air conditioner was broken
16:55.39tzangerheh it's 12 below here, no need for cooling
16:55.41`Sauronfile: how's VON?
16:55.52fileit's fun
16:56.28fileme, bkw, twisted, zoa, and Angela from Digium went around to the booths and looked at stuff yesterday... was nifty
16:56.32filethe new Grandstream phone is amazing
16:56.58eKo1It better be.
16:57.02Alexi1bye all
16:57.06*** part/#asterisk Alexi1 (~alexis@www.trim.it)
16:57.10tzangerfile: yeah?
16:57.19tzangerwhat's nifty about it
16:57.29file11 line appears, 4 separate SIP connections so a total of 44 lines
16:57.34tzangerwow
16:57.35fileplus a port to expand to more
16:57.48puppetIs it possible to do playback(input-pinc-code) ${pincode} = GetInput(#)  ... So pincode get the inputed numbers before #
16:57.57tzangerfile know anything about CDRs?
16:57.58fileULAW, ALAW, G722, G723, G726, G728, G729, GSM, iLBC for codecs
16:58.03filetzanger: yup
16:58.09tzangerquestion
16:58.12Moc____lol digium Asterisk AGI is broken hehe
16:58.13tzangerI have two * boxes, A and B
16:58.24fileMoc____: I swear we didn't break it lastnight
16:58.33tzangerI have a CDR for a call originating on A's PRI to B
16:58.34Moc____file hehe
16:58.44tzangerthe CDRs are completely different
16:58.50Moc____I can't reach support hehe
16:59.05tzanger81/73s on A vs 113/105s on B
16:59.24filewell they shouldn't be exactly the same, but that's rather extreme
16:59.33tzangerfile: the other CDRs look bang-nuts-on
16:59.44filealmost like the PRI indications are not going too well...
16:59.47tzangerit's just the odd one that looks like htat
16:59.48Moc____damnit ..
17:00.01tzangerwell A's CDR is "shorter" than B's by 30 seconds
17:00.06tzangerwell 32s
17:00.07fileexactly 32
17:00.36tzangerA's priindication is set to outofband, B's PRI is not (B's PRI connects to a Norstar MICS, I want to HEAR the tones on that one)
17:00.48eKo1Maybe A travelled back in time for 32s.
17:00.58Moc____I finally found a queue without being hangup
17:00.59tzangereKo1: heh
17:01.19filehave you done some debugging? like calling through it, seeing how long it takes for box B to realize when someone on the PRI has hungup...
17:01.24stevekstevektzanger: yeah, don't reset your clock when you're running * :)
17:01.29tzangerwhoa
17:01.30tzanger<PROTECTED>
17:01.35tzangerhow the hell did I get a dash in the dial?
17:01.43fileit's your dialplan.
17:01.44eKo1Could be some weird relativistic phenomenological time disruption.
17:01.46tzangerPRI came back with HC100 -- invalid IE contents :-)
17:01.57tzangerfile: no it's from a 3rd party, they sent me bad data
17:02.13spackleeKo1:  I don't like those, they itch.
17:02.27Moc____damn wait is long for payed support...
17:02.28*** join/#asterisk [ro]nic3try (~iancu@81.181.199.39)
17:02.38fileMoc____: dunno if anyone will be there
17:02.38spackleHey Moc!
17:02.38[ro]nic3tryre all
17:02.41tzangerfile: so some inconsistency in billing is to be expected?
17:02.49Moc____hj spackle
17:02.49*** join/#asterisk ApEtc (apetc@ip68-99-136-197.ph.ph.cox.net)
17:02.59Moc____Hi I mean
17:02.59filetzanger: a few seconds I would expect... but that, is bad
17:03.11tzangerfile: well that particular call has some extreme jitter on it
17:03.12tzangerwhich is why I singled it out
17:03.15tzangerbut not 30s worth
17:03.18filesilliness
17:03.23tzangermy graphs show 1200ms jitter
17:03.42filenever had CDRs differ like that, 'nor have I had that much jitter
17:03.52tzangerfile: ok, I will investigate some more
17:04.06tzangerI have packet captures on both boxes
17:04.11filegood good
17:04.24tzangerand I'm trying to line up the two dumps to try and see this
17:04.29tzangerbut I can't find the same call on box A
17:04.55tzangerI mean the CDRs are showing IAX2/A-12 and IAX2/B-6
17:05.02tzangerbut the packet dumps on B have no "source call# 6"
17:05.12tzangerI have A's "source call# 12" on both sides
17:05.58ManxPower!docs
17:06.00tzangersource call # 3,4,7,8,11 in the dump going from B->A
17:06.13*** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com)
17:06.16tzangerand source call # 4,6,12,22,28 going from A->B
17:06.27tzanger#12 in A->B is the one I want to match but the CDRs aren't giving me anything
17:06.38Weezeyhow can I make it so that once I exit voicemail, it brings me back to the main menu and I can dial again?
17:07.20Weezeyjust a simple Goto after the voicemailmain?
17:07.30[ro]nic3tryhas anyone tryed call transfer ?
17:07.41tzanger[ro]nic3try: yup it works great for me
17:07.43tzangerattended and everything
17:07.52*** join/#asterisk Damin_Mobile (~pocketirc@171.sub-70-214-10.myvzw.com)
17:07.56tzangerstevekstevek: think you can help me decypher this?
17:08.15spacklehelp
17:08.29*** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net)
17:08.34tzangerthere's 5 IAX2 calls in the dump, and I'm trying to separate out just one...  I have the source call# to one side but can't seem to match it up
17:08.54tzangerall 5 calls are between A and B
17:08.57junky[work]how can i checkgroup on multi machines?
17:09.27fishboy1669bye guys
17:09.29*** join/#asterisk Grooby (~Grooby@12.22.232.212)
17:09.51[ro]nic3trytzanger: how do i do it, any docs ? pls
17:10.10filejunky[work]: you have to write it
17:10.27Damin_Mobilefile: Good morning..
17:10.31filehi Damin
17:10.34tzanger[ro]nic3try: hookflash on Zap, # on everything assuming 't' or 'T' is used in Dial() and you have a default features.conf
17:10.35filewhere are you?
17:11.03Damin_MobileFile: On the lightrail..
17:11.08Damin_MobileWow... I still feel like Complete shit.
17:11.16fileDamin_Mobile: poor you
17:11.42fileIAX2 Simpler Than SIP is today, I'd advise you to go or else...
17:11.50Damin_Mobilefile: did you guys hook up w jmhunter after dinner ?
17:12.06*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
17:12.06Damin_Mobilefile: i'll Be there...
17:12.32fileDamin_Mobile: yeah he swung by, we were attacking bugs though
17:12.42Damin_Mobilefile; you still at the hotel ?
17:12.45Qwellfile: You guys come down south a bit, I'll buy lunch :p
17:13.07fileDamin_Mobile: yes
17:13.20Damin_MobileQwell how about you come up north and still buy us lunch ?
17:13.21Qwelland I'll even let you use my oven to bake muffins
17:13.31QwellDamin_Mobile: umm, that won't work :p
17:13.31*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
17:14.04*** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk)
17:14.11Damin_MobileLunch is provided at the conference
17:15.35Damin_Mobilefile: I'm heading to the conference speeches today...
17:15.40filek
17:16.13Damin_Mobilefile: What room areyou in?
17:16.51ruinerok so, I've got my cisco 3640 with 2 fxo ports, one has an analog line plugged into it that i dial into to get to my asterisk...the other port goes to one of the phones i have on my phone system, should be an fxs port...i'd like to make it so that i can access my phone system's lines via asterisk...can someone point me in the right direction?  i really want to learn this stuff but it all seems a bit overwhelming at first
17:16.55Damin_Mobilefile: And where did twisted end up?
17:17.12ruineraka, i don't want someone to just do it for me...
17:17.43*** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
17:17.46tzangerruiner: "one of the phones on my phone system" ??
17:18.00tzangerruiner: you have an ATA on your phone system somewhere?
17:18.21ruinernot sure what that is
17:18.32tzangerruiner: what are you trying to do with your existing phone system
17:19.26ruinerwell, nothing with it, specifically.  eventually we're looking to roll out VoIP for customers who have broadband so they pay a flat fee per month and can dial people in any town we have a t1 and a router in
17:19.38ruinerright now, i'm just playing around with the asterisk box we have here to try different things
17:19.45tzangerruiner: right but you're still not answering my question
17:19.57ruinermy boss right now wants me to make it so i can dial into the asterisk box, get to the existing phone system, then dial out form it
17:20.05Damin_Mobilefil: ???
17:20.07tzangerfirst things first -- what is the existing phone system
17:20.23ruinerit's a panasonic tvs200, has 12 analong lines
17:20.28tzangerok
17:20.35tzangernow we're getting somewhere
17:20.41tzangerso you have 12 POTS --> Panasonic
17:20.47ruinerright
17:20.52*** join/#asterisk tull (~danka@wwwcache2.livjm.ac.uk)
17:20.53tzangerand you want 12 POTS -> Panasonic <-- Asterisk
17:20.54tullhello
17:21.05ruinerbasically, yeah
17:21.11tullanyone using spa2000 or spa3000?
17:21.22gambolputtyi do
17:21.23tzangerso that you can have asterisk "pick up" a Panasonic extension and dial other extensions or even dial out through the Panasonic
17:21.33tullgambolputty may i pvt?
17:21.39gambolputtyyes
17:21.46ruinertzanger: si!
17:22.24tzangerruiner: well yuou need to get a Panasonic adapter that turns an extension into a regular phone line
17:22.31tzangerin Norstar land they're called ATAs
17:22.40*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
17:22.48tzangerthey allow you to plug things like fax machiens or cordless phones into the phone system and they appear as an extension
17:23.15tzangerit basically provides an FXO port that you can then plug asterisk in to
17:23.26tzangerso when asterisk picks up the line, the phone system gives it a dialtone
17:23.36tzangerand asterisk can dial whatever it needs
17:23.46ruinerhrm
17:25.25coldfeetwell Eight i couldnt find it, deleted the registry entry and reinstalled and voila I got login, now my prob is one way voice, :-)
17:25.33tzangerfor example if the ATA requires you to dial '9' to grab an external line, asterisk would dial 9...  or 225 to dial extension 225, it'd dial 225.
17:25.37tzangerso on and so forth
17:25.59ruinerok
17:26.15ruineryeah, we have to dial 9 to grab an line, then 9 to get outside the system on our normal phones
17:27.13tzangerruiner: anyway there's your next step -- to obtain a device that lets you plug a cordless phone or fax machine into a panasonic system
17:27.32tzangeras I said, norstar calls them ATAs, NEC calls them SLD or something like that
17:27.47ruineryou mean just a regular phone instead of a phone specifically designed for a phone system right?
17:27.57tzangerruiner: exactly
17:28.02ruinerbecause some of these phones allow you to plug a regular device into them, then dial with that device
17:28.10tzangerbecuase asterisk's FXS ports are essentially "regular phones"
17:28.15tzangerer sorry FXO ports
17:28.35tzangerruiner: well if osmeo fyour existing phones have that capbility then you've already go tthat
17:29.14PCadachJerJer[mobile]: Hello! About #3105. Do you uses gatekeeper for tests?
17:29.35ruinerso basically i just need to edit my extensions.conf to setup an extension to get into the phone system right?
17:29.44ruinerto dial 9,9 basically
17:29.48tzangerruiner: or whatever, yes
17:29.56tzangeryou need to make asterisk dial exactly what you'd have to dial yourself
17:30.21ruinerok, but since my asterisk box doesn't have the fxo card in it, it's through a router, do i have to setup something differently as far as asterisk is concerned?
17:30.26*** join/#asterisk multrix (~chatzilla@ALyon-252-1-32-209.w82-122.abo.wanadoo.fr)
17:30.36ruinerlike, exten => 601,1,Dial(SIP/206.222.200.46/s@default)
17:30.47ruineri'm confused on the dialing stuff
17:31.20*** join/#asterisk w0w0 (~w0w0@80-28-166-80.adsl.nuria.telefonica-data.net)
17:31.24*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
17:31.45coldfeetguys I have a strange problem I have ser connected to asterisk
17:32.03coldfeetpstn inbound via ser is transmitted to asterisk which make the call to a sip call
17:32.10coldfeetI mean sip phone
17:32.37coldfeetonce connected the sip phone can hear the pstn line, but not the other way round...now heres the strange part
17:33.00coldfeetIf I dial the pstn from the sip phone the I can hear both ways
17:33.33*** join/#asterisk xkev (kevin@orbit.xmission.com)
17:33.38xkevif I use pridialplan=national, I need to pass at least 10 digits.  fine, np.  how about international calling though?  011... typically ok in US?
17:33.54coldfeetand whats this Ooh SIP/a.b.c.d changed address line which appears
17:34.35xkevor someone got a europe number I can try so I can answer my own question? :)
17:35.29*** join/#asterisk hemant (hemant@220.226.30.66)
17:35.50*** join/#asterisk doughecka (~dheckaman@doughecka.user)
17:36.01doughecka~seen atacomm
17:36.15jbotatacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 34d 15h 39m 22s ago, saying: 'anyone want a IP 3000 conference phone?  looking to replace ours with a IP 4000 model.  Barely been used, in great condition.... looking for around $500'.
17:36.34TheBearHi all, try to configure *, I'm having a prob with zapata and zaptel. (Linux = gentoo) installed using CVS. When I start * I get warnings when on zapata.conf. "modprobe zaptel" gives errors also and lsmod is empty
17:37.45Zeeekwhat errors?
17:37.49Zeeek~pastebin
17:37.51jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
17:40.12*** join/#asterisk CoaxD (coax@shell1.cornernet.com)
17:40.13Darwin35~sex drugs Rockingroll
17:40.16TheBearhttp://pastebin.ca/7084
17:40.48Zeeekwhat modprobe errors?
17:40.58Darwin35ok you have digium card right
17:41.33TheBearyes I have Dev Kit a X100P connected to my phone number and a TDM10B connected to nothing (with a FXS module only)
17:41.42Zeeekwhat does this show? cat /proc/interrupts
17:41.46TheBearI use 3 SIP phones snom 200s
17:41.48Darwin35ok you have not started the zaptel drivers
17:41.59Zeeekyeah looks like no modprobe
17:42.25Zeeekisn't there a coffee stained sheet telling you to modprobe and ztcfg ?
17:42.38TheBearhttp://pastebin.ca/7084  updated with /proc/interrupts
17:42.53TheBearas I said modprobe zaptel gives errors
17:43.21Zeeekwhat errors?
17:43.44Zeeekok, that wasn't the same URL
17:43.54TheBearhttp://pastebin.ca/7084  updated with modprobe errors
17:44.13Zeeekwhere is the modprobe command?
17:44.22TheBearline 29
17:44.53Zeeekwhich says see dmesg
17:45.24Zeeekyou did make a clean zaptel ?
17:45.52Zeeekby the way it isn't 7084 but http://pastebin.ca/7086
17:46.06Zeeekwhich is why no one is seeing what you post
17:46.14TheBearhttp://pastebin.ca/7088
17:46.36TheBearoh
17:46.37Weezeymy asterisk sounds directory has corrupted gsm files, they used to be fine, but now they play loud, high-pitched noises in them. (at and 40)
17:47.16ZeeekTheBear what version and did you do make clean; make in zaptel?
17:47.54TheBearversion CVS as of yesterday.  yes I did a make clean;make install on zaptel, libpri and asterisk
17:48.22Zeeekand yopu /etc/zaptel.conf is clean?
17:48.26Zeeekyour
17:49.00Zeeekalthough it looks like a make zaptel probleù
17:49.16TheBearzaptel.conf  = loadzone=us , defaultzone = us , fxsks=1 , fxoks=2
17:49.52TheBearprobably a make problem but it reported no errors during make install, and will not load with modprobe zaptel
17:49.58ZeeekI asked because it says it might be a bad parameter
17:50.32TheBearzaptel.conf only has those four lines
17:50.48Zeeekyeah that's all there is except when you add params
17:51.21junky[work]ive ive a wav for my ring, how can i use that wav into my ip500?
17:51.23ZeeekDid you look at the mailing list? http://lists.digium.com/pipermail/asterisk-users/2004-October/066542.html
17:51.28junky[work]oups, damn bold
17:51.41Zeeeklooks like you need to recompile the kernel
17:51.57TheBearno I'll try that thanks
17:52.05Zeeeksee the link above
17:52.27Zeeekit seems to give the solution
17:52.27ZeeekI recompiled my kernel with CONFIG_CRC_CCITT=m, and
17:52.27Zeeekall is well.
17:52.28ZeeekI recompiled my kernel with CONFIG_CRC_CCITT=m, and
17:52.57*** join/#asterisk merlincorey (~delta9@pcp01137507pcs.mtmors01.mi.comcast.net)
17:53.52merlincoreyhello?
17:53.58Zeeekyes?
17:54.04merlincoreyheh
17:54.05TheBearwhere would that be in the kernel make menuconfig
17:54.08Zeeekhow can we be of ass-istance?
17:54.16*** join/#asterisk mbranca (~matteo@80.152.73.227)
17:54.17merlincoreyoooh, I love ass-istance
17:54.21ZeeekTheBear no idea
17:54.21Qwellass-is-tense?
17:54.29TheBearI see in .config CONFIG_CRC_CCITT is not set
17:54.47Zeeekread the thread there are a few interesting comments
17:54.56*** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
17:54.59TheBearok found it let me rebuild quick
17:55.00ruinerif i want to setup a voip solution for broadband customers so they can call various cities where we have equipment, would i need an asterisk box in each location and use IAX?  or can it be used with one asterisk box and SIP.  what would be the best solution?
17:55.33Zeeekone box would work
17:56.26*** join/#asterisk mesi (~player@dsl-082-083-145-010.arcor-ip.net)
17:56.43ruinersince the router has the fxo ports and not the asterisk box, if i want to dial an extension that then allows me to dial out, how would i go about doing that?  would i dial SIP/<ip to router>?
17:56.59mesiWho would like to join the sipphone.com conference at 1-222-000-0000 ? I'm there already!
17:57.03ruineror is sip dialing only to dial to a sip phone?
17:57.09Qwell000-0000?
17:57.21mesiQwell: Yes. It is their conference number.
17:57.27merlincoreywell, I don't have an asterisk specific question because I'm not sure if the system is using asterisk...  but I figured if anyone might know you guys might know; I'm trying to call a friend of mine who is working in a call center, I have the number to the line which then randomly chooses a call center, and I have his site code, and his extension...  I also have a number to a call director and I can have them get me to the corre
17:57.29QwellThats a valid US phone number?
17:58.01Zeeekprolly have to call it @sipphone
17:58.03mesiQwell: Well, on internet phone via sipphone.com, yes.
17:58.10Zeeekmuhahahah
17:59.09intrinanyone know of a txt2speach program that can output to a wav?
17:59.26merlincoreymy phone is VoIP if it makes it easier (for some odd reason)
17:59.27merlincorey:P
17:59.34mesiintrin: Doesn't festival do that?
17:59.40*** join/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl)
17:59.46Ro[b]ertHello all...
18:00.15merlincoreyassuming the system was asterisk, how would I manipulate it to get it to send me where I want, generally?
18:00.19Ro[b]ertwhere can i download the iso version of asterisk??
18:00.38G0shenRo: asterisk@home
18:00.46G0shenif you want to let it wipe your system and install on it
18:00.56*** join/#asterisk chrisvv (chrisvv@200.69.167.195)
18:01.31Ro[b]ertYes...
18:01.40merlincoreyany general phone system operation links you could throw at me so I can continue to experiment on my own?
18:01.43Ro[b]erthttp://www.voip-info.org/wiki-Asterisk+at++Home
18:01.47chrisvvhi ... can somebody point me how to get quicknet's phone jack to work ?
18:01.55GiabboOdo I need to uso the CVS version for install res_mysql_config.so ?
18:02.07ariel_Ro[b]ert,  here is the link http://asteriskathome.sourceforge.net/
18:02.38Ro[b]ertariel_: thanks!
18:02.52MikeJ[Jayden]merlincorey... there are no secret back doors... they have a system set up how they have it set up.  Asterisk or not, no one but the people who set up that system or use it can tell you
18:03.26merlincoreyMikeJ[Jayden]: I'm not necessarily looking for a backdoor, just how to use it; but thanks for response :)
18:03.58merlincoreyIn general, if I get directed to HIS call center, and I wanted to enter his extension and have it redirect me, how would you think in general it would be - or is it so variable there is no way to say for sure?
18:04.20MikeJ[Jayden]whatever they did.. no way to know
18:04.31merlincoreyalright, thanks :)
18:04.35merlincoreyit was worth a shot, hey?
18:04.35merlincorey:)
18:04.47Darwin35ok cool my new extensions.conf works
18:04.56Darwin35now to split it into 2 parts
18:05.33ariel_merlincorey, are you able to get to the CLI on the asterisk box?
18:06.05MikeJ[Jayden]ariel_, it's not his box, and he has no idea if it is even *
18:06.10Darwin35move the macros to a seprate file
18:06.24Darwin35and use include =>
18:06.57TheBearZeek: ok modprobe zaptel now loads what else after modprobe zaptel ?
18:06.58*** join/#asterisk bassie (~bas@datarack.xs4all.nl)
18:07.00bassiehello
18:07.05TheBearZeeek: ok modprobe zaptel now loads what else after modprobe zaptel ?
18:07.27Darwin35load the fxs
18:07.30TheBearhow
18:07.33Darwin35and the fxo
18:07.39Darwin35with modprobe
18:07.48bassiedoes anybody know if a cisco7940/60 can show incoming phone no's on the bottom line in the lcd display (using sip image)
18:07.50merlincoreyariel_: it's uh, HP phone support so I hope not though I haven't tried :p
18:08.00GiabboOdoes anybody use res_mysql_config.so ?
18:08.14*** join/#asterisk GMsoft (~r0_ot@gmsoft.developer.gentoo)
18:08.17Darwin35Mar  9 13:04:06 NOTICE[418]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us'
18:08.22GMsofthey gues
18:08.26Darwin35the last of my errors
18:08.27GMsoftmhh by what is the callerid set with h.323 ? the client, the gk or the gw ?
18:08.33ariel_merlincorey, sorry then I don't have any idea what HP uses. Besides hafe the time I call hp the send me to india for support.
18:09.06merlincoreyariel_: *nod* my friend is in a call center in Canada and he's trying to get fired so we're trying to call him and just chat with him for portions of the day ^_^
18:09.27TheBearmodprobe wcfxo worked ok   modprobe wcfxs  returns FATAL: Module wcfxs not found
18:09.38ariel_merlincorey, wow get him Fired???? why on earth do you want to do that?
18:10.14merlincoreyHE wants to get fired...  if he gets fired he can get comp and some other benefits but if he quits he can't.  so he's having fun trying to get fired
18:10.14Darwin35exit
18:10.25merlincoreyplus if he talks to us he won't be meowing at customers
18:10.26merlincoreylol
18:11.35merlincoreyagreed, but it's his life :o
18:11.43TheBearDarwin35: "modprobe wcfxo" worked ok   "modprobe wcfxs"  returns FATAL: Module wcfxs not found ? I have a X100P and TDM10B digium cards
18:11.55merlincoreyI'm trying to save the poor customers who call him and don't know what the fuck is going on
18:12.11*** join/#asterisk jluk (~jluk@pl6.lawrence.org.uk)
18:12.39merlincoreyoh apparently he is only an ass to the customers that are first an ass to him
18:12.45merlincoreywhich comes out to be 50%
18:14.12ariel_merlincorey, just call in a complaint on him.
18:14.58intrinmesi: festival?
18:15.02intrinwin32 prog?
18:15.45roamer323merlin...  I think I know that support center in Canada he works in - it was something like 1-800-BE-AN-ASS  :-)
18:15.48TheBearanyone ? "modprobe wcfxo" worked ok   "modprobe wcfxs"  returns FATAL: Module wcfxs not found ? I have a X100P and TDM10B digium cards
18:16.04TheBearI done't even have a wcfxs file, how is it created?
18:18.24*** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net)
18:18.45ariel_TheBear, I think in head there is a new name for the tdm400b. wctdm i think it's the new name for it.
18:19.12ariel_I remember they changed it due to it now can do both fxs and fxo
18:19.29TheBearyes I have that file should it modprobe it instead
18:19.58coldfeetMike got it working, i deleted the xlite registry and installed xlite again..now how do I rewrite callerID based on extension dialed
18:20.07TheBearok it loaded fine.
18:20.11ariel_TheBear, yes do modprobe zatel, modprobe wcfxs then modprobe wctdm then ztcfg -vvv
18:21.01*** join/#asterisk Smythe (~Smythe@spock.cbcag.edu)
18:21.15TheBearariel_ ok  ztcfg says 2 channels configured  ZT_CHANCONFIG failed on channel 2: No such device or address (6)
18:21.29*** join/#asterisk iceyp (~icepick@max.unix.co.nz)
18:21.39junky[work]TheBear: ur /etc/zaptel.conf is probably invalid.
18:22.37puppetAnyone that can help me get festival up running? getting strange errors :/
18:23.06JerJer[mobile]help yourself by posting the errors to a pastebin
18:23.14puppetok :)
18:23.56TheBearjunky[work] zaptel.conf has 4 lines  loadzone = us  ,  defaultzone = us , fxsks = 1  , fxoks = 1
18:24.18ruinercan someone help me with extensions?  i want to make an extension that will give me a dialtone...basically my asterisk box does not have fxo ports, i have a router with two in it...if i want to dial out, does my dial command just need to say dial(SIP/ip.to.router)? and then do i need to configure some funky crap in the router?
18:24.28JerJer[mobile]TheBear: so you redfine the same channel twice?
18:25.12JerJer[mobile]ruiner:  that's not how it works... you pass the digits to be dialed and it will pick up the line and pulse out those digits
18:25.16puppetjerjer[mobile]: http://pastebin.ca/7089
18:25.24TheBearJerJer[mobile]: not sure? I have the dev kit X100P and TDB10B (fxs module only) I have my sprint phone line connected to my X100P and nothing to the TDM10B
18:25.34JerJer[mobile]your local device will generate dialtone and all that jazz
18:25.39ManxPowerTheBear, Channels are not per card, they are per system.
18:25.43TheBearI want to be able to make and receive calls via the X100P (modem card)
18:25.56ManxPowertherefore you have channel 1 and channel 2
18:26.12TheBearManxPower: ok so what do I need to change, I'm confused already
18:26.17TheBearbetween fxo and fxs
18:26.31ManxPowerwell setting fxoks=2 might be a good start
18:26.33junky[work]ive ive a wav for my ring, how can i use that wav into my ip500 for my ringtoen?
18:26.47ruinerJerJer[mobile]: i can't just make it pass 9,9 to grab a free line and then dial whatever I want?  the other fxo port on my router is connected back into my phone system so preferably i'd like to either be able to dial other extensions in my phone system or dial out..
18:26.59ManxPowerjunky[work], I've heard a rumor that if you fillow the instructions it might work.
18:27.08ruineror, how do i even make it pass the digits
18:27.14ManxPower~docs
18:27.15jbotDocumentation 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
18:27.21TheBearok done that, do I unload zaptel, wcfxo and wcdtm then reload
18:27.22ManxPower~mailinglist
18:27.23jbot[mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.
18:28.27puppetfestival: Festival Speech Synthesis System: 1.4.3:release Jan 2003
18:28.32puppetcan it be to old festivalversion?
18:28.34junky[work]ManxPower: that would help me a lot, thanks :)
18:28.36junky[work]any link?
18:28.46merlincorey*waves*
18:28.48merlincoreythanks
18:28.49merlincorey:)
18:28.51*** part/#asterisk merlincorey (~delta9@pcp01137507pcs.mtmors01.mi.comcast.net)
18:29.03*** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com)
18:30.18PCadachJerJer[mobile]: Hello! A question about #3105. Do you uses gatekeeper for tests? Could you show your h323.conf and h.323 debug output?
18:30.30ManxPowerjunky[work], Like 4 clicks from the main page: http://www.polycom.com/resource_center/1,,pw-492,00.html
18:30.56junky[work]i'll give a watch, thanks.
18:32.04ManxPowerWhen can I expect the sexual favors?
18:32.06JerJer[mobile]PCadach:  yes I have tried gnugk and my own gatekeeper, both work fine
18:32.41*** join/#asterisk maniak-b (~m0ng0@200.150.248.134)
18:32.46PCadachJerJer[mobile]: I don't use GK and call is failed.
18:32.56ManxPowerYou could have just taken care of your "problem" yourself, but instead you had to ask me to find the doument for you.
18:33.04Mw3off, anyone familiar with panasonic tda100 ?
18:33.50Darwin35grr where is the asterisk P% module
18:33.56Darwin35p5
18:34.06*** join/#asterisk jtodd (~jtodd@199.33.155.4)
18:34.39JerJer[mobile]PCadach:  the bug says with gatekeeper the call fails
18:35.11*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
18:38.13*** part/#asterisk maniak-b (~m0ng0@200.150.248.134)
18:38.21*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-200-5.dsl.scarlet.be)
18:39.04TheBearOk http://pastebin.ca/7090 has what I have, any mistakes ?
18:39.07PCadachJerJer[mobile]: I don't see any text about GK usage in bug notes. tdriscoll wrote "My problem is that I don't use a Gatekeeper."...
18:39.20iceypJerJer[mobile] do you use SER at all?
18:40.14TheBearor http://pastebin.ca/7091 with the ztcfg output also
18:41.27SmytheHey, Gurus, is there a way for me to turn on debugging so that I can see what digits Asterisk is outpulsing to the legacy PBX I am connected to?
18:41.40SmytheConnected via a T1
18:41.52TheBearis the zapata.conf signalling correct yes/no ?
18:42.52ManxPowerdrumkilla, Is Digium support even open today or are they all at VON?
18:44.19*** join/#asterisk scrubb (~scrubb@OCI-19-41.onecall.net)
18:45.11*** join/#asterisk afe ([P+Yx+3SWQ@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
18:46.02Zeeekanyone here use voipjet? are they reachable?
18:48.08TheBearok it must be right, as I can now dial out to a local number, just it won't answer a call ?
18:49.23*** part/#asterisk rustyryan (rustyryan@cpe-65-29-77-113.indy.res.rr.com)
18:53.02*** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com)
18:53.10*** join/#asterisk voipjet (~helios@ottawa-hs-64-26-155-97.s-ip.magma.ca)
18:53.46voipjetHi everyone.  Is there any VoipJet customers who can help test the New York server (new IP)?  If so, please come ASAP to #voipjet
18:54.29*** join/#asterisk jmhunter (~jmhunter@64.77.199.223)
18:54.29*** mode/#asterisk [+o jmhunter] by ChanServ
18:55.06jmhunterblah
18:55.08ariel_voipjet, sure can
18:56.08voipjetThanks ariel
18:56.12voipjetCome to the channel
18:56.26ariel_voipjet, got an error unable to authenticate.
18:56.34voipjetariel it is a new IP
18:56.35Zeeekah no wonder my voipjet call didn't go through
18:56.45voipjetNew to test the new Server
18:56.52ariel_jmhunter, hello long time no see.
18:59.39bjohnsonI'm getting user reports when using livevoip of hearing a repeating beep pattern.  Anyone else getting that?
18:59.40CosmicRaydo any of the many different internet voip providers encrypt the data stream?  is asterisk capable of encrypting the data stream as it crosses the net?
19:00.15Groobyoops..i thought the dev conf was today
19:00.16bjohnsonvoipjet: I've got n account with you but am primarily using livevoip becuse of huge ping times with you
19:00.25bjohnsonCosmicRay: no
19:00.27loudmove to west coast.
19:00.27bjohnsonCosmicRay: no
19:00.41CosmicRaybjohnson: that stinks.
19:00.41*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
19:01.26bjohnson$$ has a way of fixing that
19:01.57CosmicRaybjohnson: of making things encrypted?
19:02.32*** join/#asterisk kraeMit (~chatzilla@p50869929.dip0.t-ipconnect.de)
19:03.10EssobiJerJer[mobile] you around?  I got H323 back up and running, but dtmf isn't passing .. I'm trying to remember how to throw h323 into debug..
19:03.35*** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230)
19:03.36Essobih.323 debug ; set verbose 5; set debug 5 isn't yielding anything..
19:03.47AgiNamuMar  9 14:03:16 NOTICE[28738]: chan_iax2.c:4609 register_verify: Host 10.2.33.82 failed MD5 authentication for 'TLF11006' (fffc8ed4df7a63a47ddbfe5f593ad191 != b67529962007dc17bb7f6f24346ab44f)
19:03.55AgiNamuGetting those notices every 60 seconds
19:04.00AgiNamuBut the user still works fine
19:04.08*** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
19:04.12GiabboOdo I need to use the CVS version for install res_mysql_config.so ?
19:04.19Essobisounds like a bad password
19:04.20junky[work]how can i checkgroup on multi machines? cause i want to limit to x apps started on the same time.
19:04.40AgiNamuEssobi, yea, but when a call is made, it works fine
19:04.44AgiNamuI double checked the phone settings
19:04.50AgiNamuExactly the same as the other units I've configured.
19:05.12AgiNamuand I'm not using MD5 auth
19:05.25AgiNamuunless that's the default one built in
19:06.30bjohnsonCosmicRay: yes.  Offer $$ to someone to make * support software encryption
19:06.51AgiNamuBribe whoever wrote the IAX encryption to start off by writing a spec?
19:06.51CosmicRaybjohnson: ah.
19:07.03CosmicRaybjohnson: I wonder if ipsec would be suitable...
19:07.17bjohnsononly if you can find another end point
19:07.21CosmicRayright
19:07.23*** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || 1.0.7 RC - bug #3746 || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th
19:07.29spackleipsec works for me, run the tunnel on a different machine tho.
19:07.33CosmicRayI am surprised that so few people seem to care about this
19:07.34bjohnsonany number of vpns or even ssh tunnelling
19:07.57CosmicRayouch, I would expect udp over ssh tunnels (I didn't even know that was possible) to be horrible
19:07.58bjohnsonCosmicRay: people don't turn on encryption on wifi APs either
19:08.12CosmicRaybjohnson: what can I say, people put XP on networks... :-)
19:08.17AgiNamuCosmicRay, people use POP3 and SMTP without encryption too
19:08.28bjohnsonsilly bastids
19:08.31*** part/#asterisk gr8nash (~basketoju@mamabear.si-forest.com)
19:08.32AgiNamuAnd that's arguably much more dangerous, considering the "email me my password" shit going on
19:08.41CosmicRayright, but encryption has been available for pop and smtp stuff for some time
19:08.48AgiNamuAnd anyways, telephones are so tappable anyways :P
19:08.51bjohnsonjust not used
19:09.27CosmicRayAgiNamu: yeah but now all I need is some teenage punk in the apartment next door with a wifi sniffer to tap in to the other end of the coversation :-)
19:09.39CosmicRaybefore, one woul dhave had to at least physically find some phone lines to tap :-)
19:09.46AgiNamuUnless you use a cordless phone :P
19:10.00MicH323If you have any interconnection problems with ITSPs (ie Vonage, FWD, BroadVoice) join #asterisk-itsp
19:10.14CosmicRayAgiNamu: even my cordless phone uses encryption of some sort :-)
19:10.17AgiNamuWhat's ITSP stand for ? IT Service Provider?
19:10.23AgiNamuwow, that's a nice cordless.
19:10.24*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
19:10.33CosmicRayAgiNamu: standard feature on vtech models
19:10.33AgiNamueven most cellphones just broadcast clear
19:10.35bjohnsonCosmicRay: or bounce a radio beam or laser off a window
19:10.37*** join/#asterisk DARP (~diegoramo@200.71.33.201)
19:10.42DARPhi people
19:10.45AgiNamuHi DARP!
19:10.48AgiNamuHow are you today?
19:10.59harryvvbj, i know what your talking about :)
19:11.01CosmicRayAgiNamu: aren't most cellphones at least digital these days?
19:11.08AgiNamuCosmic, some yea
19:11.09DARPdo you know if i can use my internal modem as sip client to connect to asterisk??
19:11.15harryvvalot are tri mode
19:11.16MicH323ITSP => Internet Telephony Service Provider
19:11.18AgiNamuso you need more than a radio scanner
19:11.21DARPfine Agi thanks, and you?
19:11.27AgiNamuOh, can't complain.
19:11.30CosmicRayAgiNamu: I mean, I don't even know where I'd go to buy a cellphone that is analog anymore.
19:11.44spacklegoodwill
19:11.47AgiNamuyea, but it's still unencrypted most of the time
19:11.48bjohnsonDARP: no
19:12.06DARPthnks bjohnson
19:12.48harryvvAre there any voip to cell analog boxes by chance? I see one thats voip to gsm for localized service to say a campous.
19:13.05harryvvso most cell calls are unencrypted?
19:13.08AgiNamuyea
19:13.17AgiNamuand even the ones with encryption have shitty encryption
19:13.27AgiNamuDesigned by comittees of cipher-novices
19:13.30AgiNamuLike WEP.. sigh.
19:13.40AgiNamuWhen will people learn to hire experts instead of coming up with their own shit?
19:13.42harryvvMy all mode all freq amature radio can do digital am/fm/wfm/usb/lsb
19:14.12AgiNamuSee, Cell phones had security issues. So instead of fixing them, the telcos just got government to pass laws against listening in
19:14.20AgiNamuand made certain radio scanners illegal
19:14.57harryvvagi this goes way back to the early 90s when a us senator was listened on once and his sensitive conversation was released.
19:15.03AgiNamuyea
19:15.12AgiNamuSince then, that's set a bad precedent
19:15.18AgiNamuFor EVERYTHING. shit like DMCA, etc.
19:15.54AgiNamuI'd blame lack of education and intelligence, but even some really well educated and intelligent people go along with crap like that.
19:16.32spackleencryption is like Karate "someone always know more"
19:16.43harryvvI recall trying to find who was using the same freqency as my home phone and it was the landlord. I said to switch or buy a new phone then discovered how nsecure those 49 meg phones really were. Somone was giving out personal credit card info over the phone heard drug deals going over the phones.
19:17.21*** join/#asterisk gr8nash (~andy@mamabear.si-forest.com)
19:17.30harryvvIt was really funny just driving though huge apartment complexes how many of those phones were around in the early 90s.
19:17.34AgiNamuspackle.... no, it's "you dont know enough" until a certain level
19:17.49AgiNamuLike... Netscape's SSL, WEP, for examples
19:18.03AgiNamuTI's "FastPass" or whatever its called
19:18.17AgiNamuusually committee based or proprietary desings
19:18.25AgiNamuhell, MS's PPTP
19:18.37greg_work* 1.0.5.. console is filled with "TDM PCI Master abort", ssh, ftp, * not responding, and I can't get at the console as that message scrolls too fast. this is the 3rd time this has happened (i think .. the last two times, it didnt have a video card so i couldn't check, it seems to be a few days in between. anyone ever seen that before? what can I do to fix it?
19:18.41AgiNamuOffice's encryption (IV reuse)
19:18.45spackleDon't you think the NSA is listening to more than they can ever admit to, and cracked codes we hope somehow they haven't?
19:18.57greg_workoh, and I have a TDM400P with 4 fxo cards in it -- none are even plugged into lines right now
19:19.01spackleNot to sound like a conspiracy freak but. . .
19:19.14AgiNamuspackle, I doubt it. In fact, I'm quite sure the NSA says they've cracked stuff they havent.
19:19.46AgiNamuAlso, the NSA has historically fixed things, even if only they know the flaw
19:19.47AgiNamu(SHA0)
19:19.55AgiNamuAnd the NSA came up with SHA1, and look at that :)
19:20.04AgiNamuso... while they are ahead
19:20.16AgiNamuI believe a sound encryption system is quite strong against them
19:20.17*** join/#asterisk Ahewes (~rsb@209.81.2.58)
19:20.28AgiNamuOf course, they might have really powerful systems to crack AES etc
19:20.31puppethaha im a reak nerd ;p gone install festival now JUST so I can get random bash.org quotes in the phone
19:20.34puppetlol
19:20.35harryvvnsa national security agency?
19:20.48AgiNamuand that's why you should always use much, much, more than you think is needed
19:20.54AgiNamuuse SHA256, 256-bit AES keys, etc.
19:20.59AgiNamuThere's no real benefit in not doing so.
19:21.10AgiNamuthe thing is, encryption is rarely the weakest point
19:21.46spackleAgiNamu, I think there is no real benefit to doing so, kind've a lost cause metality
19:21.46AgiNamuWEP being an exception :P
19:21.55harryvvI have found its the users and owners that dont think twice about security.
19:22.02greg_workAgiNamu: yeah, apparently just offer the person who knows the key a chocolate bar and they'll give it to you
19:22.03AgiNamusure, they dont think twice, until they are hit
19:22.04spackleJust keeping the honest people honest, as they say.
19:22.11AgiNamugreg: yea, stuff like that
19:22.17AgiNamuspackle, um no, that's DRM
19:22.40AgiNamuencrypting you voip calls, for instance, could stop your ISP from snooping the traffic
19:22.46harryvvpeople just dont think about the ramifications of what thay do
19:22.52AgiNamuencrypting my disk stops a theif from getting my data if he steals the comptuer
19:23.02harryvvtrue
19:23.04AgiNamuBRB
19:23.08*** join/#asterisk buddah (~hnic@208.179.86.5)
19:23.34greg_workanyone ever seen "TDM PCI Master abort" before?
19:23.38spackleAnyone who wants something bad enough can find a way to get it.
19:24.09harryvvI would love to crack physical security and get paid for it.
19:24.11*** join/#asterisk twilson (~terry@63.77.68.11)
19:24.16spackleIf you happen to have the resources of a large corporation or even a samll government to back you up it is just that much easier.
19:24.58modulus_jbot automated attendant?
19:25.06modulus_jbot ivr?
19:25.15spacklepeople are usually the wakest link anyway.  It seems its possible to pay people not to care.
19:26.00CosmicRayspackle: corrolary: it's possible to not pay people enough to care.
19:26.49harryvvI find by talking to people get a sence for there security awarness. Obviosly the aloof person is going to get burned :)
19:28.29*** join/#asterisk Katty (~angela@68.112.15.110)
19:28.51Kattyhihi
19:29.09ElsharDo I need to use devfs to use the zaptel devices? Even with a 2.6 kernel?
19:29.50bassieexit
19:30.02EssobiAnyone using Jer's H323 driver to connect to a Cisco router?
19:30.18Kattyso many questions, so little time
19:30.47EssobiKatty Go away. :)
19:31.13CosmicRayis there a way in the dialpan to make asterisk recognize attempts to dial an extension even while it is dialing another?
19:31.20ariel_hello Katty
19:31.31KattyEssobi: hmmmmmmmm, nope (=
19:31.35Kattyhi ariel!
19:31.39EssobiCosmicRay Plain engrish please?
19:31.46EssobiKatty Don't make me be mean again. :)
19:31.48Kattyariel_: how're the kids?
19:31.56KattyEssobi: i'm not making you do a single thing (=
19:32.12CosmicRayEssobi: OK... well there are the Background applications, that will play sound but if you start dialing an extension, it will interrupt the playback and take you to that extension immediately.
19:32.20EssobiSTOP WITH THE BACKWARDS SMILIES! :)
19:32.22CosmicRayEssobi: I want something that can do that for the Dial application
19:32.26ariel_Katty, she is so-so in fact both her and me are really feeling bad. We both have a bad cold.
19:32.31KattyEssobi: make me (=
19:32.40Kattyariel_: ooh. sorry to hear that..
19:32.51CosmicRayEssobi: so that, say, I dial an extension.... it's ringing... and I decide to dial another extension.  the first one stops ringing, and asterisk takes me to the second.
19:33.08EssobiOh.. Nope.
19:33.11ariel_Katty, thanks for asking hope your doing well.
19:33.12EssobiWrite a macro.
19:33.18Kattyariel_: i'm doing fine (=
19:33.21Kattyariel_: other than being confused
19:33.25Essobiand shove all your outbound dials through that macro
19:33.32greg_workCosmicRay: weird. i'm not sure if you can do it in dialplan or not .. may need to make an agi and use the manager interface ...
19:33.40Silik0nanyone have a good dialplan cheat sheet for singapore?
19:33.55EssobiA what what?
19:34.10CosmicRaygreg_work: basically, what I want this for is when somebody calls in on the POTS line, it will ring a default phone unless they know the secret and dial an extension for a different one
19:34.18Silik0nthe numbering plan in singapore
19:35.06CosmicRayanyway, question #2...  where do I find implementations for things like *67 in extensions.conf?
19:35.12CosmicRayI don't see them anywhere
19:35.32gr8nashsome are in the extensions.conf
19:36.12spackleCosmicRay:  You could answer and play a background rining tone while waiting for the secret keypress
19:36.44CosmicRaygr8nash: I don't see any in the sample and couldn't find any examples on doing them there... can you direct me to enlightenment?
19:37.06CosmicRayspackle: yes, but that isn't actually ringing the extension
19:37.42spackletrue.
19:38.12bsdfreakheh
19:38.39spackleEssobi, did you try the OH323 implementation?
19:39.00CosmicRaywhat is a good, yet inexpensive, hardware voip phone?
19:39.08CosmicRayI don't care about most bells&whistles
19:39.27*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
19:39.29spackleCosmicRay, Sipura-841
19:39.53Ahewesspackle I'm talking on an 841 right now - not recommended.
19:40.13spackleWhat issues have you had?
19:40.14AgiNamuCosmicRay: PA168 :)
19:40.19AgiNamuPA168 supports IAX2
19:40.19buddahanyone know what might be causing this error?
19:40.20buddahWARNING[20606]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end)
19:40.25buddahi'm guessing a codec issue
19:40.26Ahewesspackle echo and rubber buttons top the list
19:40.28buddahwith g729?
19:40.39spackleAgiNamu: IAX@ sort of.
19:40.43BoRiSbuddah: Sounds be fixed in latest cvs
19:40.43CosmicRayAgiNamu: pfft
19:40.57AgiNamupfft?
19:40.57buddahso update then?
19:41.04AgiNamusort of? Well, "sort of" .... works fine for me.
19:41.24CosmicRayAgiNamu: "PA168/PA1688: This is chip solution, complete with reference design"...
19:41.31AgiNamuyes
19:41.35AgiNamuSo you buy a phone based on that
19:41.41AgiNamu(obviously)
19:41.56GiabboOcdr_addon_mysql.c:352 my_load_module: MySQL database table not specified.  Assuming "cdr"
19:42.04GiabboOhow can I specify a table ?
19:42.11spackleAhewes, fair enought, I got over the rubber buttons, I think the echo will be gone in another firmware revision or two.  We haven't had much trouble with it.
19:42.13GiabboOdbtable=?
19:42.13AgiNamuthey dont really sell the PA168 kits anymore
19:42.24CosmicRayHow are the Budgetones?
19:42.27Moc____anyone from voiceconduits aroound ?
19:42.41Ahewesspackle one would hope
19:42.45Ahewesbrb
19:42.45spackleCosmicRay, they work too, sort of.
19:42.50CosmicRaysort of?
19:43.09spackleThey are a good phone to learn the ropes on, anyone disagree?
19:43.16AgiNamuBudgetones suck
19:43.23AgiNamuI had a few. All they did was flood my network
19:43.29Moc____spackle: I really suggest the Polycom for real buisness ;)
19:43.30BoRiSMoc!!!!!!
19:43.32AgiNamustay away from grandstream
19:43.34Moc____hi boris
19:43.39BoRiSWhats up?
19:43.44BoRiSSlacking off at work?
19:43.45AgiNamuspackle, learn the ropes of f*d up voip maybe
19:43.45Moc____working,
19:43.46*** join/#asterisk kraeMit (~chatzilla@p5086972F.dip0.t-ipconnect.de)
19:43.49puppetHmm, wakeup-call dont wanne register
19:43.49puppetexten => _*55*XXXX#,1,AGI(wakeup-call.pl) ; Set wake-up call
19:43.57puppetand x-lite says 404 not found
19:44.01CosmicRaythe nice thing about the budgetones is that they are $20 cheaper than anything else.  when I'm thinking of deploying four of these around the house, that adds up
19:44.09BoRiSMoc: uh-huh.....working
19:44.10AgiNamuCosmicRay, how much is the budgetone?
19:44.10mogormanis dev confrence going on?
19:44.13BoRiSheheh
19:44.16AgiNamuoh, just 4 deivceS?
19:44.21AgiNamuwho cares then
19:44.25spackleAgiNamu, I have had the PA168 unregister on me running IAX2  They are a little twitchy.
19:44.33Moc____BoRiS: yea, it my lunch time hehe
19:44.33AgiNamuspackle, what firmware version?
19:44.35CosmicRayAgiNamu: $64 new
19:44.41AgiNamu$64, not bad
19:44.52AgiNamuI can do $64... FOB china though
19:44.59CosmicRayAgiNamu: add $10 for a second 10Mb (!)
19:45.05CosmicRayport
19:45.09AgiNamuwith 2 10mbps ports built in
19:45.11GiabboOhow can I specify a table for cdr_addon_mysql?
19:45.17AgiNamuwell cosmic, your call.... but hey, hope you're lucky with the
19:45.25AgiNamua LOT of people bitch abotu hwo crappy they are
19:45.32spackleAgiNamu, the latest I could find, 1.42.x or something.
19:45.39CosmicRayAgiNamu: thanks, I'll probably avoid them then.  that's why I asked.
19:45.55fitzelis any good documentation from chan_capi available on supressing echo?
19:47.03puppet~docs
19:47.04jbotDocumentation 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
19:47.30greg_workanyone using sphinx ?
19:47.45AgiNamuspackle, 1.41 is the latest. 1.42 is in dev.
19:47.56*** join/#asterisk DevilFish (~me@staff211.qtm.net)
19:48.12AgiNamuspackle, what's your register ttl?
19:48.13*** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
19:48.19CosmicRayhow about the Snom phones?  those any good?
19:48.22DevilFishcan Asterisk connect to a VoIP provider via MGCP?
19:48.31AgiNamuI've not had any serious problems with the PA168
19:48.34AgiNamuusing IAX2
19:49.18DevilFishanyone using MGCP?
19:49.43jontowi tried.. and failed
19:49.48*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
19:49.58jontowusing an innomedia MTA 3328-4 4port MGCP gateway
19:50.14jontowmgcp support is .. odd ;)
19:50.42*** join/#asterisk SquareRt (~genesis@kurta.narnia.gamesgalleon.com)
19:50.43DevilFishI need to get my Asterisk box working with a Metaswitch 3500 using MGCP and just wondering if it is possible
19:50.56*** join/#asterisk Brumle (~brumle@brumle.com)
19:51.07DevilFishI saw somthing on the wiki about CallerAGent and UserAgent and was not sure what it meant
19:51.17*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
19:51.35SquareRt-- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4
19:51.37Darwin35ok fax is working on fbsd
19:51.38SquareRt-- Executing Dial("IAX2/iaxfwd@65.39.205.121:4569/2", "SIP/kphone|20|mrH") in new stack
19:51.39DevilFishif it meant that asterisk can only have mgcp phone connected to it or if it itself can "register" if you will to a voip provider
19:51.46Darwin35spandsp needs a small patch
19:51.51SquareRtOuch ... error while writing audio data: : Broken pipe (then a floating point exception. crash.)
19:51.59Darwin35oops
19:52.10SquareRtany ideas?
19:52.24neopheri found the spandsp 2pre4 works the best
19:52.31Darwin35wow this is nice to have a full bsd build working
19:52.36Darwin35I am using p10
19:52.44Darwin35but I did some patching
19:53.00Darwin35I  will copair it to 4
19:53.08jontowcome to think of it.. i'll need to find a real landline to see if this MGCP gateway will work
19:53.36neophercan't wait till someone gets t.38 working
19:53.41Darwin35next is sphinx and festival
19:53.55Silik0nDarwin35 on FBSD?
19:53.56neophersphinx?
19:53.59Darwin35someday in the distant future
19:54.02Darwin35yep
19:54.10Silik0n5.x or 4.x?
19:54.11Qwellsphing is a speach recog engine, right?
19:54.15Darwin355.x
19:54.19Qwellerm, speech
19:54.20Silik0nkool...
19:54.22greg_workQwell: yes
19:54.23Darwin35yes
19:54.33DARPbye
19:54.35greg_workugh, now i just broke my asterisk
19:54.35*** part/#asterisk DARP (~diegoramo@200.71.33.201)
19:54.55drspermHey, correct me if I am wrong..but if I am not allowing modules to be modprobed in my kernel...ah...asterisk won't work right? :)
19:55.08zigmandrsperm yes
19:55.12zigmanyou need zaptel
19:55.13drsperms/work right/work at all
19:55.20drspermin my kern?
19:55.22neophertring to get chan_sccp working, i think * is blocking packets, anyone know a good wy to check?
19:55.24greg_workjust compilied cvs v1-0 ..   ..........  == Registered application 'SetMusicOnHold' [chan_alsa.so][root@copper:/usr/src/asterisk-addons]# Junk at the beginning 49443303   Warning, flexibel rate not heavily tested!   Ouch ... error while writing audio data: : Broken pipe
19:55.26zigmanit will work
19:55.27greg_workthen it dies
19:55.28zigmanbut not pstn
19:55.30zigmanno isdn
19:55.32zigmanno timing
19:55.32Qwellit would work, if you don't use zap, right?  heh
19:55.54greg_workoh ,its chan_alsa
19:55.54Silik0nok anyone know the the singapore dial prefex for internation calls? (not the country code but what the correct pattern for extensions.conf
19:55.59drspermok..so add zaptel...any other kern based modules?
19:56.09zigmanno
19:56.23drspermthank...I am running hardened sources...so this might be interesting...
19:56.30zigmanhlfs ?
19:56.55drspermshit..I forget...my business partner is a Gentoo kern dev...
19:57.11drspermwe are running 2.6.10-hardened sources with pax/chpax
19:57.20greg_workanyone know offhand what packages are required on debian to get asterisk to compile chan_alsa ?
19:58.33zigmandrsperm i hope grsec 2.1.3
19:58.43zigmanalsa-lib
19:58.44zigmanalsa-util
19:58.47zigmanalsa-dev
19:59.26drspermna...not the grsec-sources...but a spinoff of that kern
20:00.00drspermJay, my partner, is an anal sob when it comes to security...I'm sure it is whater is best.
20:00.49simonideshow does the FXS port on a TDM400P get the 48 volts for ringing phones from the computer's power supply, doesn't it only supply 12 volts?
20:01.36neopheranyone know if there is a network traffic analizer in RH9, if so, what it is called
20:01.50drspermiptraf
20:01.53*** join/#asterisk t3t (~t3t@cust018.mke.attron.net)
20:01.55drspermmrtg
20:01.57drspermrrd-tool
20:02.04ariel_neopher, ethereal
20:02.04drspermtake your pick
20:02.07greg_worksimonides: transformers (or regulators, more likely in the case of the tdm) can convert power either way .. ohms law.. resistance is constant (same wire) .. so if voltage goes up, current goes down
20:02.16tzafrirtcpdump...
20:02.18Essobitethereal pwns jooo
20:02.18neophertnx
20:02.42greg_workin short, you can boost 12 volts up to 48 no problem .. but instead of having 1 amp, you might only get 300 mA
20:03.13drspermok..so where might I find zap_tel?
20:03.19drspermin the config?
20:03.24tzafrirgreg_work, DC? It is clearly easy with AC
20:04.10greg_workyes, you can do it with DC too. its just less efficient
20:04.23simonidesis that what the caps are for on the board?
20:04.47tzafrirdrsperm, what exactly do you mean? the kernel module? its sources?
20:05.10drspermwhere would I find zap_tel in the kern config (make menuconfig)
20:05.31drsperm....or have I missed the boat
20:05.39drspermsorry...package...
20:05.41tzafrirdrsperm, why won't you make it as a separate module outside the tree?
20:05.50drspermyeah...just found it...sorry.
20:06.02drspermI was trying to make it too difficult.
20:06.16tzafrirWould you really want to boot your system just because of an update to the zaptel code?
20:06.38drspermsure...
20:06.55drspermbut I have to reboot at least once...I have to add module support.
20:07.08drspermthe sys I am working with was optimzed for firewall usage.
20:07.56*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
20:08.05terrapenshido, are you around?
20:08.07terrapenor jerjer?
20:08.11drspermthere's trouble....
20:08.18terrapenhey drsperm
20:08.44drspermI got my polycom's in...time to play.
20:09.03terrapencool
20:09.09terrapenas easy as the pcoms are to set up...
20:09.13terrapeni think im gonna go with cisco
20:09.23terrapeni just like the phones better
20:09.40drspermwell, I have never used them...so maybe one day I will pick one up.
20:09.45terrapenand actually the config files are easier
20:09.56drspermhuh...I was told the cisco's were harder...
20:09.59terrapenbut upgrading a really old phone to 7.3 SIP is a bitch
20:10.03buddahi had a cisco one
20:10.07*** join/#asterisk riksta (~rick@81-178-195-88.dsl.pipex.com)
20:10.08buddahbut it fell behind the filing cabinet
20:10.11buddahso there it sits
20:10.13terrapenand i dont care what anybody says about the upgrade. it was a bitch.
20:10.26terrapennah, the cisco config files are straightforward
20:10.26DevilFishcan Asterisk connect to a VoIP provider via MGCP?
20:10.31terrapenpolycoms use XML
20:10.34terrapenwhich makes me ill
20:10.50drspermyeah....they seem a bit slow on the interface...
20:11.00fitzelIs it possible to have a kind of guest access that somebody whereever in the world can call me by sip:1234@myip.dyndns.org
20:11.04terrapenwhat provider uses MGCP?
20:11.21G0shenfitzel: yes
20:11.24terrapenfitz, check out FWD
20:11.28DevilFishI want to register an asterisk box to a metaswitch 3500 using a mgcp channel
20:11.35DevilFishis it possible?
20:12.04fitzelok, thx!
20:12.08CosmicRayDevilFish: did you not use google?
20:12.14CosmicRayfirst hit: http://www.voip-info.org/wiki-Asterisk+MGCP+channels
20:12.19DevilFishdid that
20:12.40terrapendon't think you can register MGCP with Asterisk
20:12.41terrapeni could be wrong
20:12.47GiabboObye all
20:12.48*** part/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it)
20:12.49CosmicRayQ & A
20:12.49CosmicRayCan Asterisk register as a MGCP client on a remote MGCP service?:
20:12.49CosmicRayThe MGCP implementation in Asterisk is a CallAgent, not a UserAgent.
20:12.50terrapeni see no mention of it in the sample mgcp.conf
20:12.56terrapenyeah, there you have it
20:13.05DevilFish"The MGCP implementation in Asterisk is a CallAgent, not a UserAgent"  does that mean it can only have mgcp phones then? and no connecting to a provider?
20:13.06terrapendevil, wiki is your friend
20:13.40Smythedoes anyone have experience with the Mitel SX-2000 light and Asterisk?
20:13.45DevilFishthankyou so much but yes I read that, now I need some interpretation I guess
20:14.00Essobiheh
20:14.09EssobiAnyone using Jer's H323 driver to connect to a Cisco router?
20:14.44tzangerrescan-scsi-bus -l isn't showing anything new either :-(
20:14.54tzangeroops wrong window
20:15.18SmytheWhen I try and dial from Ast -> Mitel, the Mitel shows the channel as "Locked Out"
20:15.30LuhiwuEssobi: i do
20:15.33SmytheMitel -> Ast works great
20:16.30TheBeartrying to access my mailbox I can get the Voicemailmain, when I enter the mailbox number, and then password I get incorrect-mailbox, where could my mistake be ?
20:18.42drspermok..so will * modprobe the zaptel driver or do I need to to it?
20:19.25jontowdrsperm; you gotta do it :)
20:19.39drspermk...I didn't want to step on *'s dick
20:19.44jontowhehe
20:19.52jontowits all good.. it'll step on yours often ;)
20:20.23drspermyeah...probably...but I step on mine enough (length is a bitch)
20:20.25drsperm:)
20:20.31jontow:P
20:20.35jontow(no comment)
20:21.00MikeJ[Jayden]to anyone using stable.. please note:  http://bugs.digium.com/bug_view_page.php?bug_id=0003746
20:21.42puppetHmm, my /outgoing doesnt work
20:21.47puppetit doesnt run the .call files
20:22.06drspermLoading module res_features.so failed!
20:22.41drspermhmm....I think I need to read more...
20:23.33*** join/#asterisk madclicker (~andrewn@primary.pssnet.com)
20:23.45madclickerhola
20:23.54ElsharAnyone else having a problem with the xvoip forums?
20:24.18madclicker<-- looking for skinny update for 7960.....SOS
20:26.07TheBearwhich sip pairs like context=zyx would effect the a snom 200 connecting to Voicemail ?
20:27.55Zeeekdid you get your hardware working?
20:28.44KattyZeeek (=
20:28.53Zeeekhi Katty
20:30.32modulus_[modulus@luthien /kitty/mp3]# mplayer -fs -shuffle -loop 0 * */*
20:30.32modulus_-bash: /usr/local/bin/mplayer: Argument list too long
20:30.34modulus_HAHAH
20:30.38modulus_stupid bash
20:31.17modulus_[modulus@luthien /kitty/mp3]# du -sk .
20:31.17modulus_27215391        .
20:31.17modulus_[modulus@luthien /kitty/mp3]# find . -type f |wc -l
20:31.17modulus_<PROTECTED>
20:31.17modulus_[modulus@luthien /kitty/mp3]#
20:31.19Zeeekkitty and katty - how cute
20:31.29modulus_3604 mpegs
20:31.39modulus_27 gigs
20:31.40bochmodulus_ hey dude
20:31.40modulus_jesus
20:31.46modulus_boch hey dudette
20:32.20tuxinator_linuxHey Katty
20:32.40bochyou promise me an example of your perl agi xD
20:33.10modulus_oh ok
20:33.12modulus_lemme pastebin
20:33.13modulus_hold up
20:33.20bochyea
20:33.38TheBeartrying to access my mailbox  I can get to Voicemailmain, when I enter the mailbox number, and then password I get incorrect-mailbox,  where could my mistake be Is there anyway to check what * is receiving for the mailbox number and password number ?
20:33.46tuxinator_linuxTime for a shower, see you guys later
20:34.12ZeeekTheBear pastbin your stuff
20:34.35Zeeekand look at this: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
20:34.40ariel_TheBear, did you configure your pass word in voicemai.conf
20:36.05modulus_wtf pastebin.com thinks my perl is php
20:36.05modulus_wtf
20:36.09modulus_re-
20:36.11modulus_TARDED
20:37.14TheBearariel_ yes I did
20:37.24TheBearZeeek: what do you need me to post?
20:37.35modulus_boch: http://pastebin.ca/7102
20:37.36Zeeekyour voicemail config for that box
20:37.55ZeeekTheBear post the line of the box your are trying to reach
20:38.03bochmodulus_ thanks
20:38.16*** join/#asterisk bzzz (bzzz@84.217.8.161)
20:38.19Kattyhey tuxinator_linux (=
20:38.26modulus_boch, tell me what you think
20:38.34ZeeekTheBear you can post it here - it's only 1 line
20:38.58EssobiLuhiwu You still around?  I didn't see your response earlier.
20:39.07TheBearhttp://pastebin.ca/7103
20:39.08rikstaanyone know a tool that can scan all your mp3 dirs and search for the CD Covers from the internet?
20:39.23EssobiHaha,, Pirate.
20:39.33rikstawho says I am :)
20:39.49rikstaim ripping a ton of CDs right now, I dont wanna scan them all in too much hassle :P
20:40.08neopherHELP !!, Using chan_sccp and asterisk is not picking up the phone, the server sees the packets comming in from the phone but asterisk isn't, any ideas
20:40.18TheBearI get the email message saying I have a voice mail but I can't dial it from the phone
20:40.28ZeeekTheBear and what voicemail context TheBear and you arrive at voicemail how in the dialplan?
20:40.35bochmodulus_ interesting, im reading
20:40.51ZeeekTheBear and you arrive at voicemail how in the dialplan?
20:41.00drsperm/usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
20:41.02neophernevermind, bind address was wrong
20:41.10drsperm^^anyone know why I might be getting this on startup?
20:41.20johnnybSUCCESS!!!! My office is now fully Asteriskified!!!!!
20:41.34ZeeekGoode JohnnyB
20:41.36bochlol
20:41.36TheBearZeeek: [local]  exten => 2500,1,Voicemailmain
20:41.46drspermCurrently, I have no lines, just getting familiar with inter office comm.
20:41.56Zeeekand you then enter what on the dial? nnn# ?
20:42.05EssobiHaha.
20:42.11TheBear2202
20:42.17ZeeekGo Johnny, Go Go!
20:42.18EssobiJohnnyB good. ;)  Pun intended.
20:42.23Zeeekend with a #
20:42.50ZeeekTheBear "Mailbox?" 1234#
20:42.57Zeeek"Password?" nnnn#
20:43.04TheBearZeek: oh no just 2202, no # I'll try that
20:43.11Zeeekuh, ya goo didea
20:43.18Zeeekand you should also consider reading
20:43.19ZeeekStarter tutorial:
20:43.19Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
20:43.19Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
20:43.19Zeeekhttp://www.automated.it/guidetoasterisk.htm
20:43.19ZeeekTHE reference of the moment:
20:43.22Zeeekhttp://www.asteriskdocs.org
20:43.31terrapenthe wiki is a great reference
20:43.47Zeeekand look at this: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
20:43.55`SauronI think voip-info is probably better than *docs
20:43.57harryvvjohnnyb, and if your asterisk should die?
20:44.05*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
20:44.05terrapenhere's another good sample config:
20:44.08Zeeekexcept where does it say ending with the # ?
20:44.13TheBearZeeek: still get login incorrect even with mbox# password#
20:44.19ZeeekMore likely in the docs
20:44.29Zeeekthen you tones are fucked up
20:44.35harryvvjohnnyb, what kind of hardware is running asterisk
20:44.37Zeeekdtmf
20:44.38terrapenhttp://scottstuff.net/scott/archives/000207.html
20:44.46TheBearZeeek: even if I press nothing at all it timeout to mailbox  password  login incorrect
20:44.50terrapenthat example config helped me a lot
20:45.05ZeeekTheBear normal
20:45.27Zeeeklook for dtmfmode in the docs and READ it
20:45.34johnnybAn old Sony Vaio.
20:45.55ZeeekMagic: http://www.voip-info.org/wiki-Asterisk+sip+dtmfmode
20:46.15spackle"you came in that?  You're braver than I thought"
20:46.46Essobi"Bite my shiny, metal ass."
20:46.48Essobi:)
20:47.40*** join/#asterisk eKo1 (~bernd@207.42.191.67)
20:48.06eKo1Anyone know if the current stable works?
20:48.46eKo1I keep getting errors with the loader.
20:49.07Juggiedid u delete your module dir and recreate it
20:49.12Juggieso u dont have any old ones in there
20:49.26eKo1Nope.
20:49.52ruinerTheBear: do you have dtmfmode set to inband?
20:50.20AgiNamuHey, anyone know what's Signates play? They're selling Asterisk to SGI it looks like
20:50.23drspermwould anyone know  why I am getting thie error upon startup > /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
20:50.30AgiNamuWhat do they do? Just test Asterisk and private label it?
20:50.32*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
20:50.35TheBearruiner: it is set to inband.   I allow ulaw and allow alaw disallow all
20:50.41eKo1drsperm: Did you just download the latest stable?
20:51.00drspermver 1.0.5-r1
20:51.07modulus_boch
20:52.07drspermI see 1.0.6 is out...
20:52.11*** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com)
20:52.28ZeeekTheBear I think you'll find it'll work better with ifo or rfc
20:52.35drspermso which ver do I want?
20:52.38Zeeek1.0.7 is almost out
20:52.44bochmodulus_ yes?
20:52.55drspermyes...
20:52.58eKo11.0.7 is messed up.
20:53.32MikeJ[Jayden]eko1... did you post issues to bug 3746 about issues w. 1.0.7?
20:53.34ZeeekTheBear try dtmfmode=rfc2833 in your sip client
20:53.50Zeeeksip reoad
20:53.58eKo1I haven't done anything yet.
20:54.26MikeJ[Jayden]ok... just know that that is there for these issues, if you don't comment, it will become 1.0.7 with issues and all
20:54.36AgiNamuSo.. SGI is selling Asterisk now.
20:54.38AgiNamuThat's impressive.
20:54.49brc_AgiNamu, eh?
20:54.53brc_details
20:55.00*** join/#asterisk BrianR___ (brianr@h006067091a61.ne.client2.attbi.com)
20:55.01AgiNamuThey are at the Asterisk pavilion
20:55.13brc_AgiNamu, you at astricon?
20:55.13BrianR___Hmm... nufone is busted or something :(
20:55.19modulus_boch, do you know any perl?
20:55.20AgiNamuand here is the Press release: http://www.sgi.com/company_info/newsroom/press_releases/2005/march/von.html
20:55.24AgiNamubrc_, my dad's at VON
20:55.27bochmodulus_ yeap
20:55.28eKo1MikeJ[Jayden]: OK, I'll start posting.
20:55.36brc_cool
20:55.40bochmodulus_ i liked the idea of the timer, to decrease de balance
20:55.50TheBearZeeek: ok rfc2833 gets me access, but the quality is poor crackles quite a bit
20:56.06Zeeekthat has nothing to do with the dtmfmofde
20:56.40ruinerok, so, i have a cisco router with 2 fxo ports on it, it's doing SIP to my asterisk box...if i want to dial out, what should the syntax of my dial statement look like in my extensions.conf?  say i want to dial 601 to get a dialtone through my router...can anyone help me out?
20:57.08ruineri would dial into my asterisk box with an analog phone, hit an extension, then get a dialtone, essentially
20:57.33ruinerone of the fxo ports has a regular phone line plugged in, the other is plugged into a phone on my regular phone system that acts like an fxs port (if i have my terms right)
20:57.39*** join/#asterisk jakepdev (~JakePDev@pool-68-163-51-30.phil.east.verizon.net)
20:57.53ruinerjust Dial(SIP/<ip_to_router>) ?
20:58.10ruineror would i dial using sip at all since i'm not dialing a sip phone?
20:58.15ruineri'm really green
20:58.41*** join/#asterisk antifuchs (~asf@walrus.boinkor.net)
20:59.43jakepdevis anyone running asterisk with CTI?
20:59.52ariel_ruiner, this router you have the fxo ports on is setup in your sip.conf as [name] you would do exten => X.,1,Dial(Sip/Name/${EXTEN})
21:00.13modulus_boch, i liked the database doing the subtraction from the time left
21:01.18boch\offmodulus_ i have done a similar script that interacts with radius
21:01.31boch\offwell, see u
21:01.40modulus_radius dial-in?
21:02.12drspermOk..I am getting this when it hits the features.conf : /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
21:02.20ruinerariel_: thanks, I'll give that a try
21:02.22jakepdevok - that last question was admittedly pretty stupid...  anyone using asterisk in a IVR scenerio where it speaks back digits/currency?
21:02.33ruinershould the router be set to a peer in sip.conf?
21:03.04TheBearZeeek: what would then effect that. When I hear voicemailmain, when dtmfmode=inband then it was clear with =rfc2833 it crackles. the phone and * server are on the same subnet 5 feet apart
21:03.15ariel_ruiner, set it up for your testing as type=friend for now. Then after you get it working read about peers and users on the wiki.
21:03.25ruinerok
21:03.43ruinerright now, i punch in that extension and it ends up just saying it's not a valid extension
21:04.27ruinerin the sip.conf, does that router need much else other than just host= and type=?  does it need context?
21:05.08*** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca)
21:05.32ariel_ruiner, yes there are more things needed.
21:05.42TheBearIs there anyway to change the voicemailmain greeting "welcome to ...."
21:05.56ZeeekTheBear please make an attempt to read docs
21:06.01ZeeekStarter tutorial:
21:06.01Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
21:06.01Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
21:06.01Zeeekhttp://www.automated.it/guidetoasterisk.htm
21:06.01ZeeekTHE reference of the moment:
21:06.01Zeeekhttp://www.asteriskdocs.org
21:06.40ruinerTheBear: probably need to just record a different wav, or use that festival thingy
21:07.24Moc____TheBear: there is alway a way
21:07.39jakepdevis this the proper forum for question about the capabilities of Asterisk?
21:07.52drspermZeeek: thanks for the posting...
21:08.05Zeeekwhich one?
21:08.12Zeeekthe $10,000 paypal?
21:08.15drspermStater stuff...
21:08.20*** join/#asterisk Nugget (nugget@dazed.slacker.com)
21:08.25Zeeeknot me :)
21:08.25NuggetI like cows.
21:08.36ruinerariel_: the voip-info.org sip.conf stuff I read doesn't really give me much info on this, that's why I'm asking all these dumb questions
21:08.43drspermI still have no idea adsi is..
21:09.08Zeeekruiner they're not that dumb, but you are using a setup not many have
21:09.34Zeeekand what is this phone that acts like a FXS?
21:09.39ariel_drsperm, adsi is a type of signal sent via analog phones. You need one to support it. If you dont have an adsi phone you can do without it.
21:09.48ruinerI guess part of the problem is that when I punch the extension it just keeps saying "i'm sorry, that is not a valid extension"
21:09.56Zeeekwel, it isn't!
21:10.15ruinerZeeek: just a panasonic phone used on our phone system
21:10.16drspermariel_: then where is it being called...so I can kill it for now...
21:10.18puppetanyone that knowsome about debian? cc1plus is evul :/
21:10.32Zeeekit pretends to be a phone company?
21:10.52ruinerWe have a panasonic phone system with 12 analong lines going into it, these phones can grab any line, etc, dial out and whatnot....you can plug a normal phone into it and dial from that phone, just hit 9,9,number
21:10.58harryvvgone are the days when theses will not longer be in existance :)http://www.mltelecom.com/OSP_PBX_Frame.jpg
21:11.06Zeeekok
21:11.07ariel_drsperm, was there an error given when you were loading asterisk?
21:11.12ruinerZeeek: someone earlier said that these phones essentially have FXS ports on them
21:11.27*** join/#asterisk heison (~heison@w3.somanetworks.com)
21:11.29Zeeekmakes sense
21:11.31ruinerso one of the FXO ports on my router has a line run into the FXS port on the panasonic phone
21:11.32drspermariel_: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
21:11.42heisonhello people
21:11.54heisondoes anyone here uses Jabber with Asterisk?
21:11.55drspermariel_: for now, I want to get used to asterisk with no in/out lines...
21:12.08eKo1drsperm: delete that module and recompile * again.
21:12.13ariel_drsperm, post your feature.conf in pastebin.ca and what version of asterisk are you using?
21:12.20ruinerbasically what I'm testing right now is trying to get it so that I can dial into my asterisk system and hit an extension that will essentially get me into our phone system, then being able to dial an extension in the system or hit 9,9 and get outside to dial out
21:12.33Zeeekanyone know if Neil Lewis has a pseudo here?
21:13.05ruinerEventually what I'll be doing is rolling out VoIP system to DSL/Cable customers so they can call any of the cities we have a POP in for free
21:13.20drspermpastebin.ca?
21:13.23ruinerbut we won't have an asterisk box in every city, just routers with FXO cards in them
21:13.50ruinerdrsperm: pastebin.ca is site you can paste large amounts of data into and give a URL, that way you don't have to paste it all into the channel
21:13.54Zeeeka valorous and noble undertaking, ruiner
21:13.57*** join/#asterisk PakiPenguin (~uppal@202.176.230.225)
21:14.05drspermah...tks.
21:14.06tzafrirheison, for what exactly?
21:14.08ZeeekPaki, hi
21:14.17ruinerZeeek: yeah, I'm thinking it would be easier to just get FXO cards in my asterisk box and do IAX
21:14.31PakiPenguinhey Zeek
21:14.33jakepdevzeeek - you seem to know a bit about Asterisk.  Do you happen to know if it is capable of sending a web request out to another computer and speak back digits or currency?
21:14.33Zeeekno it soulds like a good plan if thiose suckers work
21:14.34PakiPenguinZeeek :)
21:14.37ruinerlike I said, still new to this all, but the boss wants it done a certain way and has given me the task
21:14.51Zeeekjakepdev it can do that and more
21:14.53Qwelljakepdev: sure it can, with an agi or something
21:15.26Qwelljakepdev: if you can imagine it up, I'm sure * could do it
21:15.26drspermariel_: http://pastebin.com/251591
21:15.40QwellYou could probably write an app to make you coffee, as part of a wakeup call
21:15.48jakepdevdo you have a link to agi?
21:16.02Qwell~google asterisk agi site:voip-info.org
21:16.09jakepdevok
21:16.15jakepdev(sorry bout that)
21:16.19Qwellnp
21:17.18*** part/#asterisk Moc____ (~mochouina@64.235.210.66)
21:17.20Qwellheh, that last one has a link to a wallup call, that forces you to do a match, to make sure you're awake
21:17.59drspermariel_: can I be helped?
21:18.18Qwellerm, math question, I guess
21:18.29jakepdevI did google this though before asking - and couldn't come up with definitive results... does it supprt call xfer (H.450.2) over h.323?
21:20.14ariel_drsperm, looks like your running cvs head.  The file looks ok to me.
21:20.32drspermyeah...I dont' have a clue where this is comming from...
21:20.59drspermI am going to verify that I can load modules real quick...what if...
21:21.15heisontzafrir: i'm trying to send CallerID to my Gaim client
21:21.51heisontzafrir: but jabber.php seems to just hang there... with no udp 5222 traffic leaving the box
21:23.58tzafrirudp 5222? it should be tcp5222, right? Jabber client
21:24.18heisontcp? really?
21:25.07tzafrirMost internet traffic is TCP. VoIP is not normal: it does its own congestion control
21:25.39drspermok..modules work just fine
21:26.55tzafrirAnyway, I'm out
21:26.56iceypUS $152.50 for a cisco 7940G brand new, is this good?
21:26.58Juggietzafrir, its called UDP
21:27.13Qwelliceyp: brand new?  I'm sceptical
21:27.22harryvvyea thats a very very good price
21:27.27iceypand i one
21:27.31iceypwon one
21:27.32iceyplol
21:27.33TheBearanyone using a snom 200 ?
21:27.33Luhiwuanyone knows the country code for Cayman Islands? is it +1 ?
21:27.35ruinerWould someone be willing to look at http://pastebin.ca/7105 and see if they can figure out why my extension 601 doesn't work properly?  i included what I think is the relevant part of extensions.conf and sip.conf
21:27.39harryvvwon one?
21:27.46iceypwell one the bid
21:27.46iceyp:)
21:27.55harryvvon ebay?
21:27.58iceypyeah
21:28.02heisontzafrir: it now works... thanx
21:28.09iceyp<PROTECTED>
21:28.09iceypPositive Feedback: 96.7%
21:28.10mesiiceyp! Hello!
21:28.17iceyp<PROTECTED>
21:28.22*** join/#asterisk santiago (~santiago@201.245.167.104)
21:28.27iceypi'm not at home
21:28.28Qwellmeans ~8 of those people have complained
21:28.29iceypat work here
21:28.37iceypyeah
21:28.39mesiiceyp: Since we've first met I've improved my * setup a lot :-)
21:28.47*** join/#asterisk AsteriskNoob (~blah@207-114-232-10.gen.twtelecom.net)
21:28.53AsteriskNoobgood afternoon!
21:28.58harryvvI have seen people with good feedbacks give me 1. motherboard dead. 2. hamradio that had a crasked battery case and he never notified me the frequencies were unlocked.
21:29.05AsteriskNoobwho here is an expert with call parking?
21:29.05*** join/#asterisk xcoyote (~coyote@dsl-200-95-78-238.prod-infinitum.com.mx)
21:29.33Tall-guynoob: I can park and retrieve... :)
21:29.34Qwellunlocked frequencies is a bad thing?
21:29.36iceypmesi nice one, pitty i cant try it with u, i'm at work when ur awake
21:29.39*** join/#asterisk ionix (ionix@MTL-HSE-ppp184758.qc.sympatico.ca)
21:29.40harryvvI was lucky the dual opteron motherboard had a rma recall on it and now its this system I am talking on.
21:29.43iceypyeah id like that unlocked
21:29.46tzafrirJuggie, I meant Jabber, not SIP or anything. See section 2.3 in http://www.ietf.org/rfc/rfc3920.txt
21:30.19spackleruiner, what is on the SIP side of that extension?
21:30.25AsteriskNooblol@Tall-guy, I am having an issue, I have music on hold, but when I park a call its dead silent, the people think i cut them off, is there a way to put music on park?
21:30.26mesiiceyp: You can always try when you're back if you want. My FWD# is 434240. You may call any time :-)
21:30.54justinnnnnnasterisk
21:31.00Tall-guynooob: probably a contexts issue.....ie: what your musiconhold is set for the certain context....
21:31.00justinnnnnnb4 u do ur parking extension
21:31.08justinnnnnnuse setmusiconhold
21:31.09ruinerspackle: a router, if that's what you mean
21:31.10ionixHey, anyone has an idea on how I can manage multiple calls on the same SIP account in a prepaid environment (Meaning I cut the communication when balance==0)
21:31.21iceypanyone here using SER?
21:31.34TheBearanyone using a snom 200 ? snom 200 has DTMF Payload Type 101,  what is this in * inband, or what ?
21:31.49ruinerspackle: router w/ two fxo ports, one of them going into an fxs port on a phonesystem phone...trying to dial 601 and get into my phone system
21:32.05AsteriskNoobhey, do those snom's have the capability to show when another phone is on the line, my cisco's dont :(
21:32.08ruinersame router that takes incoming analong calls and send them to my asterisk box
21:32.13Zeeekruiner : remove the . from 601
21:32.25Zeeekremove the EXTEN par fromt he end
21:32.49ruinerso, just exten => 601,1,Dial(SIP/voice-gw)
21:32.55Zeeekruiner try this and see if it works:
21:33.03Zeeek601,1,Dial(SIP/voice-gw)
21:33.09Zeeekya
21:33.09spacklecan you use a phone to dial the router by IP and get a dial tone?
21:33.21ruineri have no voip phones
21:33.25mesispackle: I do this!
21:33.31ruinerdoing this all analog at thois point
21:33.31ZeeekI can't stand the tension
21:33.39Zeeekruiner
21:33.43Zeeekhuuurrrry
21:33.44mesispackle: Perhaps I can help you, though I use asterisk for a few days only.
21:34.04ruineri dial 601 and just get dead air
21:34.13ruinerif i dial another digit it says sorry, invalid extension
21:34.23Zeeekdid you SIP RELOAD
21:34.29harryvvhehe
21:34.37mesiruiner: How long is the "dead air"? Have you waited long enough?
21:34.38Zeeeker extensions reload
21:34.58ruinermesi: 5 seconds or so
21:35.02ruinerZeeek: yes
21:35.05Tall-guyAnyone up for a Xlite  (sip) to analog (digium) "how to get better quality" discussion?
21:35.10puppetdoes anyone now if festaival works with newest festival 1,95  or need 1,43?
21:35.10mesiruiner: Perhaps wait a little longer.
21:35.12Qwellless then your digittimeout, or whatever?
21:35.25puppetfestival*
21:35.25ruinerif I wait long enough it just says goodbye
21:35.48ruinerwondering if my sip.conf is good...i included it in the end of that pastebin.ca post
21:35.48Zeeekruiner add a line like 600,1?NoOp(Hey now:!) before the dial and renumber the dial to priority 2
21:35.49spackleruiner, is there a user and password on the router?  How do you address the difference between the FXO ports when you call the router?
21:36.03harryvvbtw, where can i get xlite beta for linux. seems xten is not responding to my email. Or is there another free softphone to download for linux.
21:36.16*** join/#asterisk claint (~claint@195.174.26.218)
21:36.25spackleruiner, no user and pass info in the sip.conf if it is necessary
21:36.32Zeeekruiner there must be a doc with the router that explains this (what spackle just asked)
21:36.47Weezeywhen sending something via SIP, is there a way to set the callerID?
21:36.49CosmicRayharryvv: kphone is working for me
21:37.05ZeeekSetCallerID?
21:37.15Qwellharryvv: I like iaxcomm
21:37.22Weezeyright now, all calls from asterisk are coming with the original name, but "asterisk" as the number
21:37.38Zeeekwhy not check the allpications list ?
21:37.40spackleWeezy, setcidnum
21:38.05Weezeyit's set.
21:38.13ZeeekThe Allplications are your friends :)
21:38.15Weezeyit's just not being passed to my sip phone.
21:38.25Qwellallplications?
21:38.27Zeeekwhat phone?
21:38.31ZeeekGrandStream?
21:38.32Weezeyspa-841
21:38.33ruinerZeeek: after adding the NoOp it just hangs up...yeah, thinking I'll need to check sip config on router or something
21:38.39spackleWeezey, do you have it set in sip.conf?
21:38.57Zeeekruiner the noop was just to prove you are getting there (print on cli)
21:39.00puppetAnyone know why my /outging dir doesnt work ;p it doesnt call the call file ;)
21:39.00Weezeythe callerid for the extension that the call's coming from?
21:39.22ruinerMar  9 16:04:22 WARNING[14438]: pbx.c:1280 pbx_extension_helper: No application '' for extension (default, 601, 1)
21:39.34Zeeekpuppet let's see an ls -l of that dir
21:39.36ruinerthat's what cli printed when hit the extension
21:39.36*** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net)
21:39.54Zeeekruiner ya screwed up a line there
21:39.55*** join/#asterisk rob- (~robbie@haylott.plus.com)
21:40.12Zeeekexten => 600,1,NoOp(Message)
21:40.16ZeeekNo SPACES
21:40.17Weezeyspackle: caller id isn't set anywhere in my sip.conf
21:40.26puppetweezey: http://pastebin.ca/7107
21:40.26spackleruiner did you put ,1,NoOp(Made it to here)
21:40.36drspermSo does anyone have an ideas what would be causing this:   [res_features.so]Mar  9 15:39:52 WARNING[19215]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
21:40.36drspermMar  9 15:39:52 WARNING[19215]: loader.c:440 load_modules: Loading module res_features.so failed!
21:40.57Weezeypuppet: I don't get it.
21:40.58tzangerdrsperm: you've noloaded res_adsi
21:41.01QwellDid you upgrade, and not remove your existing modules first, or something?
21:41.03spacklelike Zeeek says <grin>
21:41.13Zeeekpuppet what time is it there?
21:41.13puppetweezey: wrong :)
21:41.14drspermno...single install.
21:41.20drspermtzafrir: where?
21:41.25ruinerexten => 601,1,NoOp(Blah)
21:41.30puppetzeeek: Wed Mar  9 22:41:25 CET 2005
21:41.37ruinerexten => 601,2,Dial(SIP/voice-gw)
21:41.39drspermtzanger:  where?
21:41.41Zeeekbut the .call file is in the past
21:41.52*** join/#asterisk chris_d (~chris@66.88.142.66.ptr.us.xo.net)
21:41.54puppetzeeek: it didnt call before at that time
21:41.56tzangerdrsperm: I'm guessing modules.conf
21:41.57Zeeek(or is it supposed to be)
21:42.23Zeeekthere's siomething about the mtime on those files
21:42.26chris_dI've set up an Asterisk@Home system for testing--about as difficult as falling down a flight of stairs...
21:42.30puppethmm
21:42.30chris_dI have two questions:
21:42.35Zeeektry touching it to future a few seconds
21:42.39ruinernow CLI says nothing when i hit that extension
21:42.40drspermtzanger: nothing listed about asdi
21:42.43chris_dIn the FOP, all lines are flashing green/red. What does this mean?
21:42.49Zeeekor just touch period
21:43.13*** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net)
21:43.31chris_dSecond, when I try to call from one test extension to another, there is no ring--after a brief period, I get that extensions voice mail.
21:43.58Zeeekruiner not even the NoOp?
21:44.13ruinernope
21:44.20Zeeekyou didn't forget to....
21:44.21ruinerof course, nothing happens when i do my echo test either
21:44.26ruinerextensions reload
21:44.36ZeeekNOTHING?
21:44.39ruinernope
21:44.46ruinerecho test works, but nothing in CLI
21:44.50drspermtzanger: what might the line be ?
21:45.01Zeeekset verbosity 999
21:45.06puppetzeeek: ok
21:45.09Zeeekor is it verbose?
21:45.18Qwelldrsperm: noload => res_asdi.so
21:45.21Qwellor some such
21:45.28Qwellif its there, comment it
21:45.31Zeeekwe're all being so precise
21:45.34Zeeekat the moment
21:45.40puppetzeeek: i put it to 22:45:06 50 sec before
21:45.43ruinerprobably not...should i run asterisk with a different flag?  -vvcc or whatever?
21:45.47puppetzeeek: it didnt call then
21:45.53Zeeekpuppet DAMN
21:45.56drspermnope...no dice.
21:45.59Qwellruiner: set verbosity is a command
21:46.06habakukI'm trying to bill on incoming/outgoing calls. My idea is to add an extension to a cdr_odbc that will update my users credit for each call. Has anyone done anything similar?
21:46.09puppetzeeek: any config I have to put in to use outgoing?
21:46.20ruinerQwell: in the CLI?
21:46.22Zeeeknot that I know of
21:46.23Qwellyes
21:46.29ruinergave me an error when i typed that
21:46.38ruinerNo such command 'set verbosity' (type 'help' for help)
21:46.38Zeeekmaybe it's set verbose
21:46.41Qwell<Zeeek> or is it verbose?
21:46.44ruineroh
21:46.46*** join/#asterisk afe ([VbYfgFzAS@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
21:46.49Zeeekruiner -
21:46.51puppetNo one that knows if Festival 1.95 works wtih Asterisk?
21:46.55Zeeekplease: look
21:47.01Zeeekshow applications
21:47.08Zeeektype that now on CLI
21:47.25ruinerk
21:47.34*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
21:47.35ruinerok
21:47.37QwellZeeek looks like he could use a beer or something
21:47.40Zeeekinteresting, yes?
21:47.41ruinerit was verbose, not verbosity
21:47.46Zeeekya
21:47.48ruinersorry
21:47.53drspermok..well the line was not there...so nothing to comment...still getting [res_features.so]Mar  9 15:45:45 WARNING[19248]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
21:47.53drspermMar  9 15:45:45 WARNING[19248]: loader.c:440 load_modules: Loading module res_features.so failed!
21:47.53Zeeekthat's a nice feature of the CLI
21:48.13Zeeekit has a list of all that shit that no one can remember
21:48.17Qwelldrsperm: and you didn't switch versions at all, at any point?
21:48.18ruinerthere we go
21:48.19Zeeeklike verbosity
21:48.30drspermQwell: no.
21:49.03ruiner<PROTECTED>
21:49.04ruiner<PROTECTED>
21:49.04ruiner<PROTECTED>
21:49.14ZeeekI wish I had a beer now, absolutly - but I will have Champagne on Saturday night
21:49.16ruinerso probably need to check in my router, i'd assume
21:49.25Qwellsaturday is too far away
21:49.25Zeeekruiner YES! Progress!
21:49.39ruinerI could use a rum and coke right about now
21:49.39johnnybDoes anyone know what would cause asterisk to wait 2-4 seconds before picking up a ringing outside line?
21:49.56ZeeekjohnnyB callerid
21:50.07QwellPakiPenguin: you can't make a rum and coke with pepsi!
21:50.17ZeeekQwell sure you can
21:50.18johnnybjust set usecallerid=no or is there something else I need to do?
21:50.20PakiPenguinQwell: didnt notice that :p
21:50.22Zeeek99.99% rum
21:50.23Qwellit'd be a rum and cola :p
21:50.43Zeeekif you don't have callerid that's good
21:50.50Zeeekand possibly a restart
21:51.11Zeeekzapata.conf needs a bigger shove
21:51.37*** join/#asterisk Tili (~Tili@202-133-65-224-dialup.sat.net.pk)
21:51.38ZeeekPaki you didn't answer my last question
21:52.25*** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com)
21:52.56drspermok..so with a virgin install...should asterisk start?
21:52.56PakiPenguinZeeek : an that was? <-- 3am here , so am running at 66mhz
21:53.03PakiPenguinyes drsperm
21:53.11drspermk...I am ripping it out.
21:53.36Qwelldrsperm: is yours all patched up or something?
21:54.01drspermI just emerged it in using portage on Gentoo
21:54.06Qwelleww
21:54.14QwellI never could get the portage one working
21:54.36harryvvwhat is portage
21:54.42Qwellgentoo's build system
21:54.54drspermthe package tree for......yeah ^^^^
21:54.54harryvvyea, we were just talking aobut it on cinelerra
21:55.05Zeeekah, voipjet has mailed the server ips
21:55.09drspermIt has worked well for us for several years...
21:55.16claintuhm, great, i was just about the emerge asterisk......
21:55.17*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-25-189.d4.club-internet.fr)
21:55.18drspermcompiling....
21:55.25PoWeRKiLLhi
21:55.39harryvvThat is one distro be it gentoo i have yet to mess with..and also solaris.
21:55.41PoWeRKiLLhow can I call a vonage subscriber via my * box ?
21:55.52harryvvbondage?
21:55.53harryvv:)
21:55.56QwellPoWeRKiLL: get a fwd account
21:56.15harryvvFricken vonage and there multimillion dollar marketing budget.
21:56.28QwellI fucking hate that song.
21:56.31Qwellin case anyone was wondering
21:56.36neopherCan x100p 's be configured to act as a FXS ?
21:56.41Qwellneopher: no
21:56.54claintQwell: is it just you, or did others got gentoo one working?
21:57.00Zeeekneopher no
21:57.03harryvvit does not have the electronics to send ring voltages detect dtmf and such
21:57.04ZeeekECHO()
21:57.15neopherahh, true
21:57.30Zeeekruiner ?
21:57.42harryvvI have a x100p and a ata. so that works good for me.
21:57.45puppetSince no one knew ;p im gone try Asterisk with Festival 1.95 now ;p
21:58.16Juggiethe patch doesnt work
21:58.17PakiPenguin02:58:23
21:58.18Juggiedo it with an agi
21:58.19Zeeekruiner, this will come in handy: From CLI  type ext <tab> r <tab>
21:58.20PakiPenguini think i should sleep
21:58.25Zeeekme too
21:58.28puppetjuggie: wasnt thinking of patching
21:58.29neopheryea, just a though as the x100p clones are so so cheap
21:58.32Zeeekit's past my bedtime
21:58.37puppetjuggie: was gone try the configfile
21:58.49puppetjuggie: the configfile didnt work in debian default so ill try it in 1,95
21:59.01ruinerZeeek: ?
21:59.16ruinerZeeek: I've been reloading after extensions.conf changes
21:59.24PoWeRKiLLthere is no other way to call a vonage number without going throw fwd ?
21:59.48PakiPenguinnight everyone
21:59.52Qwellclaint: It wouldn't be in portage, if it didn't work for the people who put it there.
22:00.01Qwellclaint: but I've heard issues from others too
22:00.08johnnybZeeek:  That looks like it worked.  Thanks!
22:00.26Zeeekwhat worked?
22:00.27*** join/#asterisk fjoe (~fjoe@samodelkin.net)
22:00.34claintQwell: hmm. so what was the problem?
22:00.53fjoehi
22:01.06fjoeanyone using quadBRI + * on Linux?
22:01.19Zeeekruiner no I just though if you hadn't found the tab completion it would make your life better :)
22:01.26ruinerMy router is saying that I'm making a bad request...I'm assuming I need to define some kind of username/password somewhere in the router and in the extensions.conf, or just make sure my router config is right in the first place
22:01.29ruinerZeeek: ah :)
22:01.54Zeeekruiner is the router registering? sip show peers?
22:01.57puppetjuggie: is festival worth the time? i have no clue how the quality is
22:02.12ruineryes
22:02.18ruinervoice-gw         206.222.200.46              255.255.255.255  5060     Unmonitored
22:02.29puppetruiner: grats
22:02.35puppet;p
22:02.38puppetor something
22:02.42ruinerheh um, ok
22:02.45puppet:)
22:02.59jas_williamsruiner
22:03.05ruineryeah?
22:03.10puppetjas_williams: i tried to be positive! ;P
22:03.12Zeeekah the night shift! hey Jas_
22:03.20Qwellclaint: similar problems
22:03.21drspermok...virgin install...starting...
22:03.38jas_williamsruiner: Just joined what router are you trying to configure ;-) 22:00 here zeeek
22:03.41Zeeekhow many times are we gonna want to type drsperm...
22:03.50ruinerjas_williams cisco 3640
22:03.59jas_williamsUsing Sip ?
22:04.01ruineryeah
22:04.02jas_williamsor H323
22:04.05jas_williamsok
22:04.06Zeeek22:00? I thought you were down under?
22:04.06ruinerthe router has a 2fxo card in it
22:04.16ruinerone port has an analog phone line plugged into it
22:04.23ruineri can call that line and get into my asterisk box
22:04.26drspermZeeek: you know...I never thought of that...
22:04.29Zeeekwhat is this router ruiner?
22:04.30drsperm:0
22:04.36Zeeekheh ya sure
22:04.41drspermHonest!
22:04.46drspermOk..it died.
22:05.01ruinernow i'm trying to setup asterisk so when i hit an extension it goes back to the router's other fxo port and dials out of it (that fxo port is plugged into an fxs port on a phone i have, not ip phone, just one for our phone system)
22:05.03*** join/#asterisk Twister (~jase@216.30.232.106)
22:05.08drspermok..if I change my nick...will you help?
22:05.09ruinerZeeek: cisco 3640
22:05.15*** join/#asterisk lattice (~lattice@d216-232-212-173.bchsia.telus.net)
22:05.17jas_williamsruiner: ok
22:05.38jas_williamsruiner: Is the call making it to the 3640 ?
22:05.39harryvvLooks like alot of the small telco companies are blocking vonage service to its customers.
22:06.05ruineras best as I can tell, yes, but it doesn't like the router likes it...in asterisk's CLI it shows this when i hit 601 (the extension i'm testing)
22:06.09ruiner<PROTECTED>
22:06.09ruiner<PROTECTED>
22:06.09ruiner<PROTECTED>
22:06.14Twistergonna set up one of my offices with asterisk, got 3 lines + a vonage line comming in, gonna do IP phones throughout the building (probably only be 5-6 extensions)
22:06.22WetPustulesruiner at least you're talking to it now!
22:06.25chris_dharryvv: If, by a lot, you mean 1, then yes. Also note that the one who tried got nailed and fined by the FCC...
22:06.32ruinerWetPustules : yes
22:06.38Twisterwhat would be the minimum specs on the machine id need for this?
22:06.47ruinerI think I always was, just didn't know about the verbose command to see what was happening
22:06.59jas_williamsruiner: Can you post the relevant section from you extensions.conf to pastebin.ca
22:07.08drsperm" undefined symbol: adsi_available" has got to mean something is up....
22:07.22ruineryeah, it's at http://pastebin.ca/7105
22:07.33jas_williamsk i'll have a look
22:07.34ruinerguess i could trim it down some
22:07.38ruinerjust search for 601
22:07.42harryvvchris i know. most of there voip to pstn circuits are in nj right?
22:07.55ruineri have my sip.conf at the bottom too
22:08.00WetPustulesruiner reminds of a Cliford Simak short story about a guy who goes to Jupiter with his dog. Finds out the dog could talk. Says "When did you learn to talk" dog says " I always could, you just never understood before"
22:08.18ruinerheh
22:08.19ruinerindeed
22:08.20*** join/#asterisk Possible (Babbel@23.255-136-217.adsl-fix.skynet.be)
22:08.38ZeeekGreat story - good evenig, good luck and $deity bless asterisk
22:08.39claintwhat's the dog name? Idefix?
22:08.44jas_williamsruiner syntax is wrong
22:08.50ruineroh
22:08.50ruiner?
22:08.51ZeeekClifford? Is that You?
22:08.59chris_dharryvv: North Carolina.
22:09.14jas_williamsruiner try exten => 601.,1,Dial(SIP/${EXTEN}@voice-gw)
22:09.21*** join/#asterisk Moc__ (~mochouina@64.235.210.66)
22:09.23Possiblehi I have a question about asterisk and Mac OS X
22:09.25ZeeekNo DOT!
22:09.31ruinerheh, Zeeek told me no period
22:09.37ZeeekDON'T YOU DARE
22:09.42jas_williamsruiner try exten => _601.,1,Dial(SIP/${EXTEN}@voice-gw)
22:09.48Zeeekah, that's different
22:09.57harryvvchris, vonage is even marketing there service here in vancouver british columbia. Do you have a idea what marketing budget thay may have.
22:10.12puppetiik takes time to compile the speach stuff
22:10.14harryvvosx?
22:10.17Possibleanyone know if there is any hardware w/ driver around that works with OSX ?
22:10.18ruinerwhat does the ${EXTEN} do?
22:10.28harryvvOSx=Freebsd
22:10.31ruinerjust passes that in the sip request?
22:10.42Possibleharryvv: yes and no
22:10.46puppetruiner: exten == dialed number
22:11.00harryvvpossible ask pilotmike on #hamradio2 he is a expert on mac systems.
22:11.06puppetruiner: if i dont remember wrong
22:11.17jas_williamsruiner: yes passes the dialled number so trys to connect to that dial-peer in the cisco
22:11.18*** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
22:11.19Mavvieharryvv: now that's what I call "happily confused"
22:11.19Possibleharryvv: he uses asterisk on OSX ?
22:11.31puppetruiner: no its the dialed number
22:11.38*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
22:11.38*** mode/#asterisk [+o bkw_] by ChanServ
22:11.50Qwellbkw_: omg, its you!
22:11.58harryvvpossible no but he knows unix and osx
22:12.18Possibleyou mean #hamradio...not #hamradio2
22:12.29Moc__bkw_:  any pic from von ?
22:12.36Qwell...my sparcstation is blowing out cooler air then my air conditioner
22:12.57Qwellis it at all normal for the AC to blow out hot air?
22:13.08Qwellit hasn't been used all winter(yes, we need AC in the winter here...sad)
22:13.22Possibleseems pilotmike is not online atm..
22:13.30harryvvpossible #hamradio2
22:13.37ruinerjas_williams: so should the extension i'm configuring be the dial-peer in the cisco?
22:13.45Possible#hamradio2 is empty
22:13.46fjoePossible is it possible to compile FreeBSD kernel module under Darwin?
22:13.55ruineror, is this going to connect to the same port that the call is coming in on?
22:13.56harryvvpossible sorry its on efnet
22:14.07Possiblefjoe: Darwin uses a MACH kernel
22:14.31fjoePossible I know, but how hard is it to port FreeBSD kernel module to Darwin?
22:14.31Possibleoh kernel module...I dunno
22:14.38jas_williamsruiner: you can do what you like depends on how you have the cisco configured, Asterisk can pass what you like to the cisco as a pattern for the dial-peer
22:15.38Possibleharryvv: ahh
22:15.38drspermok..is it possible that grsec is borking *
22:15.38fjoePossible zaptel framework was ported to FreeBSD (ports/misc/zaptel) and * works on FreeBSD via chan_zap
22:15.38bkw_VON VON VON
22:15.38bkw_all about VON
22:15.38Possiblefjoe: ic...interesting
22:15.42fjoePossible ports/net/asterisk
22:15.43Moc__<PROTECTED>
22:15.52jas_williamsruiner: the call will go out whichever port has apattern match that matches the XXX@voice-gw
22:16.11ruinerso it's really an issue of what's configured in the router then
22:16.19jas_williamsyep
22:16.24fjoePossible I never saw Darwin source code but I think that it should not be hard to port a device driver from FreeBSD.. Darwin is not pure Mach anyway
22:16.39ruinerknow where i can get a sample router config?
22:16.48ruineror do you know much about configuring ciscos for this?
22:16.53ruineri could paste mine into pastebin
22:17.02harryvvbkw so what vendors showed off there bleading edge of technoligy products? anything fairly cool?
22:17.29jas_williamsruiner ok and I'll take a look see and see if we can make progress for you
22:17.43gr8nashim alittle stuck if anyone can help.. my box registers at broadvoice.. but i cant dialout.. and im not sure howto get it to stop saying "all circuits are busy"
22:17.48puppetWhy am I lisstening to Slay radio threw Asterisk and not threw winamp/xbox?
22:17.50puppetrofl
22:17.52gr8nashi know dialout plan has somehting to do with it
22:17.56Possiblefjoe: lost you on the ports/net/asterisk part...what url ?
22:18.07jas_williamsruiner: My cisco skills arn't that good but I have got these things working before
22:18.14fjoePossible it's the path inside FreeBSD ports collection
22:18.22harryvvgr8, sounds to me all there pstn circuits are used up.
22:18.42Possiblehttp://www.FreeBSD.org/ports/net/asterisk doesn't exist
22:18.47gr8nashthis is a brandnew install with no cards.. im only using broadvoice
22:18.48ruinerhttp://pastebin.ca/7115
22:18.56harryvvgr8 try in the late evening to see if it happens again.
22:18.56ruiner^^ jas_williams
22:19.03jas_williamsruiner ta
22:19.33gr8nashharryvv,  thanks but its my asterisk box telling me that.. i can dail through x-lite just fine
22:19.38fjoePossible http://www.freshports.org/net/asterisk/
22:19.44gr8nashi dont know howto setup a dialplan
22:19.45jas_williamsruiner: which port is connected to the pbx ?
22:19.45Possibleah
22:19.45gr8nash=(
22:19.56fjoePossible but I do not think that it could be ported to darwin without any efforts
22:20.02ruiner1
22:20.03harryvvyou mean dial though softphone to same service and call uncle john ;)
22:20.09ruiner0 is connected to analog
22:20.26drspermSon-Of-A-Bitch...hardened sources seem to have been fucking me.
22:20.35jas_williamsruiner: is a 9 required on the PBX for an outbound call ?
22:20.41ruineryes
22:20.43Possiblefjoe: ic...lets have a look at this
22:20.45gr8nashhardwire,  i mean astrix registers but cant receive/call .. but my softphone has no problem doing either
22:20.48ruinertwo of them, actually
22:20.54gr8nashharry i mean sorry
22:21.04hardwirehey!
22:21.12ruinerbut if possible, i want to be able to just dial my extension 601 and then either hit like "101" to get someone inside the pbx, or 9,9,xxxxxxx to dial out
22:21.20ruineror maybe need a different extension for all that
22:21.57harryvvgr8, you mean you can send/recive calls between softphones on your network but cannot call out to a voip provider?
22:22.06Possiblefjoe: and thanks :)
22:22.45jas_williamsruiner ok if you dial 60199xxxxx where xxxx is a valid number does asterisk connect to the outside world it should
22:23.26ruinerno, just says invalid ext
22:23.49jas_williamsruiner: What says invalid extension ?
22:23.57ruinerasterisk
22:23.58ruiner<PROTECTED>
22:24.03*** join/#asterisk darby_t (~tom@dml252.neoplus.adsl.tpnet.pl)
22:24.07ruinerthat's from CLI
22:24.21jas_williamsruiner: did you dial 60199number
22:24.51ruinerfrom my phone i called into the analog line that routes into the asterisk box, then hit 60199 and then a phone number
22:25.07ruinerby the time i hit the second 9 it was saying in my ear "invalid extension, etc"
22:25.44jas_williamsruiner: do you have an extension connected directly to asterisk this will aid testing
22:26.06ruinerno, i have no fxo ports in my asterisk box
22:26.15ruinerno cards, that is
22:26.24ruinerjust a box w/ two ethernet
22:26.41jas_williamsruiner: How about downloading a SIP Softphone to a PC and connecting that to asterisk
22:26.58jakepdevanyone know how stable FastAGI is?
22:27.17ruineri suppose i could do that
22:27.22ruinerany one in particular you recommend?
22:27.30*** join/#asterisk Zaw (zaw@zaw.subneural.net)
22:27.39jas_williamsX-lite or SJPhone
22:28.05ruinerso i just put that on a machine and it acts like a SIP phone essentially?
22:28.24puppetx-lte rocks for testing
22:28.25puppet;D
22:28.33ruinerjust plug directly into * or same switch work too i assume?
22:28.59godsmokeruiner: more like the other way around -- the sip phones run software to communicate via sip -- you're just using your computer to run software that communicates via cip
22:29.05godsmokesip(
22:29.09ruinerok
22:31.24ruineraight, i have that installed
22:31.47ruinersip proxy would be the * box or the router?
22:31.55puppetspeach_tools forever to compile! ;p
22:31.57fjoe!seen
22:32.04fjoe!seen kapejod
22:32.16Qwell`, not !
22:32.20Qwell~ rather
22:32.27jas_williamsruiner: sip proxy is asterisk
22:32.41fjoeQwell ?
22:32.42ruinerout bound proxy same?
22:32.48Qwell~seen kapejod
22:32.49jbotkapejod <~kapejod@83.137.99.168> was last seen on IRC in channel #asterisk, 121d 9h 22m 38s ago, saying: 'how about mc?'.
22:32.56fjoeQwell ah! thanks a lot!
22:33.04ruineror would that just be if i was using a proxy in general on my machine?
22:33.09jas_williamsruiner: no outbound proxy
22:33.17fjoe121 day...
22:33.20fjoeouch
22:33.22jas_williamsor asterisk should work
22:33.56fjoeby all
22:33.56*** part/#asterisk fjoe (~fjoe@samodelkin.net)
22:35.04ruinercall not approved every time i try to call
22:35.27PoWeRKiLLI try to call a vonage subscriber via a FWD account it's not working :( any other idea ?
22:35.37QwellPoWeRKiLL: it only works sometimes ;/
22:35.59jas_williamsruiner: has the phone registered
22:36.43ruinerwith sip show peers?
22:36.50ruinerif so , no
22:36.51jas_williamsruiner: yes
22:37.03ruinerdo i need to add it to sip.conf?
22:37.03PoWeRKiLLQwell there is no other solution ?
22:37.15G0shenPowerkill: vonage broke their peering with
22:37.21G0shenFWD, and they won't fix it
22:38.00PoWeRKiLLG0shen it's not possible to peer with them directly ?
22:38.07godsmokewhy not just peer with them directly
22:38.08godsmokeheh
22:38.20godsmokeit's kindof silly to go through vonage
22:38.24godsmokeit's higher latency
22:38.27*** part/#asterisk Grooby (~Grooby@12.22.232.212)
22:38.30PoWeRKiLLlike calling 1-XXX-XXX-XXXX@sip.vonage.net ?
22:38.55G0shenpowerkill: people have been talking about it on the FWD mailing list for months
22:39.03G0shenif they could fix it, I am sure they would...
22:39.09G0shenbut Vonage will not respond
22:39.34godsmokeif they could?
22:39.44godsmokeI mean ... it's not very hard to add FWD to your asterisk box
22:40.05G0shenHere is the Jan 25th reply from Jeff Pulver
22:40.06G0shenIn theory connectivity with Vonage will be restored before the end of this
22:40.06G0shenmonth. This is a Vonage issue at the moment.
22:40.45godsmokeyou're just restating the same thing
22:42.09PoWeRKiLLyes but why it's not possible to call them directly
22:42.28PoWeRKiLLjust resolve the user ip then call him
22:42.59spackleruiner, how did you make out with your router?
22:43.31QwellPoWeRKiLL: welcome to a locked service ;/
22:43.44Qwellif they were running *, sure,you could just call the IP
22:43.53ruinerspackle : still working on it
22:44.35*** part/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net)
22:46.16*** part/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
22:46.16*** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc)
22:49.04*** join/#asterisk Marcus_Burge (~mburge@216.229.216.126)
22:50.30*** join/#asterisk zno (~chatzilla@ip-160-79-174-102.autorev.intellispace.net)
22:51.52*** join/#asterisk zpn (~xpn@dhcp-152.digium.com)
22:53.16*** join/#asterisk zotz (~zotz@24.231.32.191)
22:57.47*** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
22:58.15MikeJ[Jayden]~seen anthm
22:58.17jbotanthm <~anthmct@CPE-69-76-83-52.wi.rr.com> was last seen on IRC in channel #asterisk, 7h 21m 34s ago, saying: 'stp will tune into a media stream and record it to a file'.
22:58.45MikeJ[Jayden]~anthm
22:59.32Darwin35grr spandsp is fucked
22:59.35Darwin35grrr
23:01.16puppetfestival already sucks ;p takes 100 years to compile
23:02.43eKo1*cough* cepstral *cough cough*
23:02.54modulus_speech recognition is no small ordeal
23:03.13modulus_hidden markov models intrigue me
23:03.18puppeteko1: cepstral? ;p is that some typeof drug? ;D
23:04.23eKo1I haven't touched any maths. with the word Markov in a while.
23:04.58Pinholecepstral is not free.  (unless you use an offset of say 510000 to skip the demo message) ^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H
23:05.03*** join/#asterisk anthm (~anthm@68.30.2.131)
23:05.03*** mode/#asterisk [+o anthm] by ChanServ
23:06.05znoanyone use nufone?   I just signed up and got a short email with instructions but it's not working
23:06.09eKo1It's cheap though.
23:06.19PinholeI spent several months playing with speech recognition.  It kinda works, but sending it over the net destroys any chances.
23:06.26znothe email didn't provide instructions to register
23:06.36znojust a NuFone context in iax.conf and an extension
23:08.41puppeta question, on inc phonecalls shouldntit just be to do an monitor? to record it without lagging all?
23:09.18*** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
23:12.27*** part/#asterisk Smythe (~Smythe@spock.cbcag.edu)
23:12.49gr8nash=/ anyone want to help a kinda noobie.. ill add you to my christmas card list =0)
23:13.21puppetcristmascard? like we get an analog->digicard? ;P
23:13.23greg_workPinhole: the most important thing .. if someone says "fuck" or any other similar word/phrase, transfer to a person immediately :p
23:13.32JerJer[mobile]gr8nash:  help yourself first by asking a specific question
23:13.32gr8nashWAIT I GOT IT!!!
23:13.41puppetgreg_work: lol
23:14.23greg_workbell's system does it. i laughed my ass off the first time :p
23:14.32puppetgreg_work: ROFL? ;P
23:14.38puppetgreg_work: whats the number to bell?
23:14.42puppet1800 i hope ;p
23:14.47greg_work310bell here
23:15.16puppetiik ugly :/
23:15.21puppetboring answering
23:15.28puppetget a pr0nvoice!
23:15.32*** join/#asterisk Popdog (daniel@edtn014064.hs.telusplanet.net)
23:15.55greg_workit work?
23:16.02puppethavent tried ;p
23:16.15puppetgot to make a recon first
23:16.24puppetputting up the extension now
23:16.50greg_work18006686878
23:17.11puppetlol
23:17.16puppetdamn what they answred long
23:17.19*** join/#asterisk hacim (micah@micha.hampshire.edu)
23:17.20puppetlike 4 sentences
23:17.37puppet"welcome to.... my name is.. we are trying.... hope... we ... can ... be in swer.. how cnai help u?
23:17.40puppet"
23:18.03hacimanyone have any suggestions for good conference room type speaker phones? I've found http://www.metrolinedirect.com/soundpointpc.html for $70, but other models are at minimum $300 which makes me think this one sucks for some reason (compare: http://www.globalmedia.com/index.php?dest=catlist&gmenu=3&area=store&cat=voice)
23:18.54*** join/#asterisk Spongii (Spongepupp@dial-205-113-113-216.megacom.net)
23:19.17*** join/#asterisk sabre (sabre@69.149.209.83)
23:19.29*** join/#asterisk Tr0j4N (b3nz3r@pcp08761169pcs.mtlrel01.nj.comcast.net)
23:19.58Spongii:|
23:20.42Spongiisabre!
23:21.02Spongii:D
23:21.48SpongiiSyncros!!!!
23:21.55hacim?
23:21.56Spongii:)))
23:22.07Spongiiarf :|
23:22.07*** part/#asterisk Syncros (~sysop@noc.routermonkey.net)
23:22.36Spongiisheesh is it forbidden to tell them i lvoe them? pfft
23:22.38*** part/#asterisk Spongii (Spongepupp@dial-205-113-113-216.megacom.net)
23:23.38greg_workhacim: you can get actual IP speakerphones for that price
23:23.44*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
23:24.30*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
23:25.00hacimgreg_work: what do you mean "actual IP speakerphone"?
23:25.24greg_workthose things hook up to pc's
23:25.44hacimgreg_work: isn't that what http://www.metrolinedirect.com/soundpointpc.html is?
23:26.17*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
23:26.23shmaltztzanger, ?
23:26.24madclickercan someone help me with snippy firmware for 7960?
23:26.28shmaltz~seen tzanger
23:26.29jbottzanger is currently on #asterisk.  Has said a total of 142 messages.  Is idling for 1h 44m 33s
23:26.39shmaltzmadclicker, I can try
23:27.02greg_workhacim: it says it plugs into a pc soundcard
23:27.30madclickercan u pm me/
23:27.34*** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu)
23:27.48puppetEasy Setup - Connects directly to your PC sound card inputs
23:28.01greg_worksounds a little iffy though .. $70 for an onboard dsp + speakers + microphone?
23:28.01madclickershmaltz:i am looking for dl for the firmware
23:28.20greg_worki'd be surprised to see a good microphone alone for $70
23:28.29*** join/#asterisk zapa (zapa@200.77.116.158)
23:28.37shmaltzwhat do you mean dl?
23:28.46madclickershmaltz: download
23:28.54puppetmadclicker: u now firmwares cost? :)
23:28.56shmaltzwhere did you buy it?
23:28.59greg_workand dsps are expensive (which is actually the whole reason * was created)
23:29.05hacimgreg_work: this one is even less: http://www.sylvansoftware.com/Microphones.htm
23:29.16madclickereshmaltz:-bay
23:29.25madclickershmaltz:e-bay
23:30.23greg_workhacim: go look at some of polycom's stuff
23:30.39greg_work(unless you're actually looking for something that just connects to a pc)
23:31.03harryvvI wonder what kind of range I can get with a wifi repeater mounted beneth a aerostat for say a function.
23:31.05hacimgreg_work: yes, I want to connect something to a PC so I can do something like meet-me or skype in a room of people
23:31.11*** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net)
23:31.11*** join/#asterisk david (~dcoulson@muffin.davidcoulson.net)
23:31.14davidhello
23:31.30harryvvOr just a wifi router with directional antennas.
23:31.37Matt-E-when i restart my FC3 box  the /etc/init.d/zaptel service fails to start, even when run manually. though if i remake zaptel it then works... how can i fix this?
23:31.59harryvvmatt in usr/src/zaptel do make config
23:32.10Matt-E-okay
23:32.15Kattybuh bye
23:33.14davidwhat is the simplest way to identify a call without any caller ID information in extensions.conf, so I can route it direct to VM rather than dial()?
23:33.33madclickeranyone with firmware for cisco 7960???? SOS need help
23:34.02`SauronYou can't identify a call w/o CID info.
23:34.16shmaltzdavid, what?
23:34.21shmaltzwithout callerid?
23:34.25G0shenroute them through telezapper, then privacy app if they don't have caller id
23:34.27shmaltzbased on what?
23:34.33`SauronUnless you're talking about finding calls with "Unknown number" as callerid
23:34.41harryvvyea, how would asterisk know who it is without a cid? it wont work.
23:34.44david`Sauron, yes, that's what I mean
23:34.56shmaltzoh i got it, use ${EXTEN}/
23:35.02nextimeif i use zaprtc cause i don't have a usb controller and i can't use a 2.6 kernel, i also need ztdummy for timing or is only zaprtc ok?
23:35.08*** part/#asterisk xcoyote (~coyote@dsl-200-95-78-238.prod-infinitum.com.mx)
23:35.11shmaltzthe slash (/) means that if no caller id
23:35.25hacimgreg_work: basically there are things that are $40 and there are things that are several thousands of dollars, and I can't tell what is crap and what is hype and what I need :p
23:36.10davidshmaltz, okay, cool :)
23:36.38davidshmaltz, and I can use a standard asterisk wildcard after the / too, like 888_ and whatnot?
23:36.54harryvv~seen netsurfer
23:36.57jbotnetsurfer <netsurfer@81-6-224-129.dyn.gotadsl.co.uk> was last seen on IRC in channel #asterisk, 20d 10h 6m 37s ago, saying: 'http://www.theregister.co.uk/2005/02/17/spam_gets_vocal_with_voip/ <-- ffs that takes the piss'.
23:36.57shmaltzdavid, nope
23:37.02nextimeok i've found the answer on the asterisk-user ml.
23:37.13shmaltzyou will have to do like this:
23:37.14KalD|Workwhen does meetme.conf get evaluated?  At each call to MeetMe()  or on reload?
23:37.27harryvvwow he must be injured or in the hospital. Netsurfer used to be here every day.
23:37.48`Saurondum di dum
23:37.55Darwin35dim sum
23:38.19`SauronI'm trying to stream audio using netcat... :)
23:38.23shmaltzdavid, I don't remember the proper syntax for the text functions within asterisk but you could look it up on the wiki
23:38.35Darwin35use madplay
23:38.42shmaltzbut the basic idea is that you create a different extension for each one
23:39.07shmaltzlike:
23:39.08shmaltzexten => 101/,1,dosomething
23:39.10shmaltzexten => 101/2,hangup
23:39.52puppeti hate festival :/
23:39.58shmaltzand for 800 numbers you will have to do:
23:40.00shmaltzexten => 101,1,gotoif($[sytax goes here]?101/,1)
23:40.31`Sauronnow I have to hook up audio to see if it works
23:40.33`Saurondum di dum
23:40.39davidshmaltz, but if I want to say 'if it's from a number starting 800', how do I define a wildcard caller id in the exten?
23:41.03shmaltzdavid, thats where the sytax in gotoif comes in
23:41.06hacimjeez these speakerphones are expensive
23:41.35Darwin35no
23:41.48Darwin35gs 101 02 102 65/75 bucks
23:41.49shmaltzI don't remember the text functions on how to do it, but the idea is that you test the first 3 digits of the variable ${CALLERIDNUM} to see if it matches 800
23:41.50Matt-E-upon reboot i get Waiting for zap to come online ...Error: missing /dev/zap!
23:41.53Matt-E-how do i fix?
23:41.54Darwin35they have speaker phone
23:42.02Matt-E-i got a good init.d/zaptel file
23:42.40zapaHi all, is there any way to change the ring sequence , like in Cisco 79XX via SetVar(ALERT_INFO=Bellcore-dr1), for Budgetone an Polycom phones.
23:42.41harryvvMatt its things like that scares me if a system is deployed into a work enviroment.
23:42.48ariel_wow just got a request which I don't know how to do.
23:42.55shmaltzdavid, don't you give up b/c it's doable, just search on both voip-info.org and lists.digium.com
23:43.11hacimDarwin35: eh?
23:43.27Darwin35hacim what speaker phone ?
23:43.36ariel_I have a user that wants us to put in a auto redial feature if the number is busy the server retries it again until connected then sends the call to the extension. anyone know how to do this?
23:43.37Matt-E-harry, how do i fix it?
23:43.47shmaltzdavid, to search either on thru google enter the search term in google followed by site:voip-info.org
23:43.50hacimDarwin35: you are saying that some speakerphones called gs101 02 102 are $65/75?
23:44.07Matt-E-after i recompile zaptel it the init.d file works... why is that?
23:45.04sivanahas anyone used the exten=>fax,1,blahblah before?
23:45.09hacimDarwin35: I am trying to find a decently priced speakerphone that will work for a small room of people, attached to a laptop running something like skype
23:45.18Darwin35the grandstream 101 and 102 sip phones are speker phones
23:45.21hacimDarwin35: or asterisk with meetme or something
23:45.33hacimDarwin35: but you basically have to be leaning over them to be heard, no?
23:45.42Darwin35no
23:45.46hacimDarwin35: I could plug a microphone into the laptop and it would be no different
23:45.55ariel_sivana, i have
23:45.58Darwin35the 101 yes the 102 they changed the mic
23:46.10sivanaariel_: do I need to enable it?
23:46.28sivanait went to IVR but didn't pick it up
23:46.30Darwin35but I moded my phone and wear a mic around my neck
23:46.35ariel_if you are using the normal asterisk yes in the zapata.conf
23:46.45hacimDarwin35: why would I use a SIP phone to do speakerphone?
23:46.57hacimDarwin35: I am wanting to do conference style calls with 5 people in a room
23:47.05sivanaariel_: I'm using CVS Head
23:47.08Darwin35I just saw you commenting about speker phones
23:47.15Darwin35ahh
23:47.57ariel_faxdetect=incoming,
23:48.43Darwin35ok spandsp is pissing me off not compiling
23:49.01sivanagot it.. thank you
23:49.12ariel_sivana, np
23:49.27davidshmaltz, okay, got it figured out - thanks
23:50.16jakepdevwhat is the most reliable distro to run asterisk on?
23:50.57Darwin35Hacim make a cheap one
23:51.01ariel_jakepdev, that is a loaded question. I use CentOS which is a RHEL clone.
23:51.23jakepdevlooking for stablility and ease of installation
23:51.27ariel_jakepdev, what do you want to do
23:51.30Darwin35buy the parts at radioshack
23:51.46ariel_jakepdev, get asterisk@home comes complete including the os and spandsp and we gui.
23:51.49jakepdevIVR (speaking digits/numbers to about 40 clients)
23:52.09*** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1)
23:52.11Matt-E-why does FC3 remove /dev/zap/ when rebooting??
23:52.14puppetwhat is lacking in asterisk@home?
23:52.22shmaltzdavid, np
23:52.39ariel_jakepdev, http://asteriskathome.sourceforge.net/
23:52.52CosmicRayHello.  I am setting up kphone to talk to my asterisk system.  When a call begins, everything is fine.  However, as the call progresses, things start to sound worse and worse.  It seems as if split seconds of sound are missing.
23:52.56jakepdevthanks - i'll try that
23:52.59CosmicRaywould this be a problem with kphone or asterisk?
23:53.05CosmicRayand where would I go to learn how to fix it?
23:53.08sivanaariel_: I guess I need rxfax too :)
23:53.11ariel_puppet, actually not much. But it's up to what you want to do with it. It even has the new meetme2 available on it.
23:53.44ariel_sivana, you need to install spandsp or send the fax to an fax machine on a fxs port. or on a sipura using ulaw.
23:54.00ariel_asterisk@home has it already installed.
23:54.01sivanaI see.. the script I had used rxfax
23:54.47jakepdevariel - you used fastAGI - if so - is it reliable?
23:55.12*** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com)
23:55.31ruineranyone have example config for a cisco 3640 with a 2 port fxo card in it so that you can make asterisk dial through it?
23:55.35ruineri'm really stuck here, heh
23:56.13ariel_jakepdev, it's for when you want to run an agi script on a different server and yes I have used it. But I did not write the agi for it's use.
23:56.35CosmicRayfwiw, I'm getting similar behavior from iaxcomm
23:56.44CosmicRayI've tried google and the wiki and haven't really found any answers
23:56.52jakepdevthanks ariel
23:56.56CosmicRayeven the Echo demo is giving trouble
23:57.19ariel_jakepdev, any time.
23:57.42*** join/#asterisk stustu (~stustu@fluffy.fatburen.org)
23:58.21ariel_CosmicRay, are you using this on the actuall asterisk server with xwindows installed?
23:58.32*** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu)
23:59.16CosmicRayariel_: no, it is on a separate client
23:59.28CosmicRayconnected over 100Mb lan
23:59.37ariel_ruiner, I did not even know about this device. But if it's like normal cisco sip device it's not too hard but most of the time you have to upgrade the firmware.

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