irclog2html for #asterisk on 20051005

03:05.42*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
03:05.42*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now!
03:06.01file[laptop]jbot: wb
03:06.02jbotIt's great to be back!
03:06.14theblueHi.
03:06.28file[laptop]jbot: uptime
03:06.57file[laptop]neat
03:07.24Drukenwtf happened to the search on the wiki?
03:07.36file[laptop]they changed software
03:07.40Drukeni hate using google for that... opens a new god damn window everytime!
03:08.43Drukenand apparently the wiki has no information about sending calls over a snom sip proxy
03:08.58hardwireheh
03:09.38Drukenkeeps bitching about an invalid argument...
03:11.04ManxPowerThe mayor of new orleans just announced that 1/2 of all city employees are layed off effective immediatly.
03:11.15ManxPowerI expect him to lay off 1/2 of the rest of them soon.
03:11.30theblueOy.
03:12.23*** join/#asterisk RickTick (n=rpulido_@c-67-191-89-108.hsd1.fl.comcast.net)
03:12.29*** part/#asterisk RickTick (n=rpulido_@c-67-191-89-108.hsd1.fl.comcast.net)
03:12.32PBXtechwhy with a softphone i get 1way audio with 1.0.9 but 1.2 i get 2way?
03:12.42*** join/#asterisk RickTick (n=rpulido_@c-67-191-89-108.hsd1.fl.comcast.net)
03:13.11RickTickanyone got a Grandstream HT488 working with the fxo  port?
03:13.20hardwirefuck no
03:13.27RickTickha ha ha
03:13.31thebluePBXtech: Simple, 1.2 likely has support for the softphone.
03:13.57PBXtecha SIP softphone... uhh
03:14.28RickTickhardwire: are you serious? or were you responding to someone else
03:14.34hardwireRickTick: fuck no
03:14.48RickTickha ha ha ... you cracking me up ...dude
03:15.26RickTickso you mean , I just wasted my fucking $300  for those 4 fuckers
03:15.32RickTickdamn
03:15.34hardwireRickTick: that is pretty neat
03:15.39hardwiredoes that have a speaker on it
03:15.46hardwireor is that just venting
03:15.49RickTicknope
03:16.16RickTickhow can they sell bullshit like that?
03:16.45Drukenschweet!
03:16.51Drukengot it worken....
03:16.59RickTickdude ... I brought 4 of that p.o.s ... and it driving me up a  wall
03:17.22hardwireRickTick: are you configuring them right?
03:17.32RickTickI think so ...
03:17.50RickTickI get the fxs part working ( not reliably though
03:18.15RickTickI have no  idea how to get the fxo part to work yet
03:18.28ManxPowerRickTick, Um, you didn't buy one to test it first.
03:18.34RickTickI thought I would come here ,,,and get some "expert" advice
03:18.46ManxPowerRickTick, check the mailing list archives.  Grandstream makes crap, but it seems to mostly work.
03:18.57ManxPower~mailinglist
03:18.59jboti heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
03:19.01RickTickManx: I should of ... but my friend want to take some abroad with him
03:19.21ManxPowerRickTick, I'm a big fan of the Sipura SPA-2100
03:19.25*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
03:19.30ManxPowerBut it doesn't have FXO.
03:19.56RickTickwell, I am going to take the 15% restocking hit tomorrow ..and get Sipura 3000 ...  fxs/fxo
03:20.23theblueRickTick: I recommend the Cisco ATA 186, if you want an ATA>
03:20.34ManxPowerRickTick, many VoIP adapters with FXO only support FXO passthru when power is lost.  Check the mailing list archives, I seem to recall someone saying that the GS device supported real FXO support, but I can't be sure.
03:20.35theblueRickTick: Though it locks up the ports.
03:20.40RickTicktheblue:  I need an fxo port also
03:20.47theblueRickTick: Ah
03:20.53ManxPowertheblue, the Ciscos are expensive and not as full featured as the Sipura
03:21.03theblueManxPower: Oh/.
03:21.07ManxPowerRickTick, remember the SPA-3000 does not do NAT routing
03:21.11JerJerwhich is why Cisco promptly re-bought Sipura
03:21.31ManxPowertheblue, Also the guy that designed the Cisco ATA is the same guy that designed the SIPURA.
03:21.35JerJerManxPower:  it doesn't?   why does it have the nat options in the config then?
03:21.36RickTickManX:   the the Sipura 3000 is my best best ... for reliablility?
03:22.00ManxPowerRickTick, The FXO isn't the easiest to make work on it, but it's still your best bet.
03:22.22ManxPowerI have an SPA-2100, SPA-3000, and SPA-2000
03:23.20RickTickManX: Can I contact you ...when I get the Sipura's ...  for configuration help?
03:23.21JerJerand a partridge in a pear tree
03:23.35JerJerRickTick:  can you?
03:23.53RickTickJer: SureI can
03:24.04JerJerits May i, smart guy
03:24.05RickTickoh ...May I?
03:24.06JunK-YRickTick: if you pay him a damn good beer maybe :)
03:25.01RickTickdude ... my english is lacking .. so are the grandstream p.o.s
03:25.16*** join/#asterisk casio_ (n=Cris@200-126-118-183.bk8-dsl.surnet.cl)
03:26.10*** join/#asterisk albi (n=albi@dslb-084-056-128-161.pools.arcor-ip.net)
03:27.23hardwirehmm
03:27.26opus__shit
03:27.27hardwirehow do you make a t1 talk to itself
03:27.30opus__i got pizza burns on my hand!
03:27.36denonhardwire: plug in a loopback
03:27.42hardwiredenon: yeh..
03:27.47opus__hardwire you hardwire it
03:27.49opus__hehe
03:27.55hardwirebut.. do you configure the first half of the channels as fxo and the second as fxs?
03:28.09hardwirecause that just doesn't seem right.
03:28.12denonsame way you'd do it if it had a channel bank on it..
03:28.17*** part/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net)
03:28.18denonthough ..
03:28.22denonsince you're on both ends ..
03:28.22hardwirewell what happens when I dial out?
03:28.24denonthat's kinda ugly
03:28.38denonI guess it would work fine though
03:28.51hardwirefor instance.. I set it up as all fxsks
03:28.57hardwireif I dial out of one channel what happens?
03:29.53file[laptop]loopback?
03:30.14*** join/#asterisk simoncion (n=avatar@user-24-236-84-62.knology.net)
03:31.27hardwirefile[laptop]: yeh
03:31.31hardwireI put my adtran loopback on it
03:31.33hardwireno alarms now
03:31.38hardwirejust wondering what happens if I dial out :)
03:32.43file[laptop]it explodes
03:32.45file[laptop]obviously!
03:34.14Pete_Largonothing happens, you get a fast busy
03:34.23Pete_Largoor whatever
03:34.47Pete_Largoa hard loop plug doesn't rearrange the channels, so channel 1 is looped to channel 1 and so on
03:35.02hardwirePete_Largo: makes sense
03:35.42hardwirein pri_cpe it gives me lots of warnings w/ loopback :)
03:36.00Pete_Largobecause the D channel does not have d-chan sync
03:36.11Pete_Largoit is not receiving happy signals from the 'slave' so to speak
03:37.22*** join/#asterisk Craziman2 (n=Craziman@63.108.128.250)
03:39.25bkw_ya it goes "we're CPE but they are CPE also.." or the reverse will be displayed
03:39.46twistedbeverly hills
03:39.48twistedthat's where I want to be
03:40.01blitzrageWeezer
03:40.12file[laptop]and the locals come alive...
03:40.20twistedblitzrage, meh?
03:40.29blitzragetwisted: isn't that a line from a Weezer song?
03:40.34twistedblitzrage, oh, yeah, that too
03:40.37blitzragethat's what I thought you were quoting :)
03:41.06twistedblitzrage, we'll have to rent a escalade and roll through beverly hills
03:41.38blitzragetwisted: you obviously don't know about my Egg-a-Hummer crusade
03:41.46Corydon76-homeOh, no, you can't sleep now, file
03:41.51twistedyeah, but a hummer is not an escalade
03:41.59blitzragetwisted: its possibly worse
03:42.10twistedblitzrage, no way man
03:42.11Corydon76-homefile[laptop]: Alex wants you to call him...
03:42.21blitzragetwisted: I hate gas guzzling vehicles
03:42.35twistedblitzrage, okay, you can rent a scooter
03:42.40twistedblitzrage, i'll rent an escalade
03:42.43Craziman2Any of you guys got a PAP2 from Vonage working with Asterisk?
03:42.56blitzragetwisted: cool :)
03:43.11twistedblitzrage, yeah, and we'll see who gets some hot ass first.
03:43.27twisted;)
03:43.27blitzragetwisted: well, I look better than you... so I'll go with... me :)
03:43.31twistedHAHAH
03:43.53Corydon76-homepretty blue eyes
03:43.58blitzrageCorydon76-home: you've got the wrong plumbing :)
03:44.02twistedlol
03:44.10blitzragetwisted: actually... you might win... I'm not as hot with a beard :)
03:44.18blitzrageits my handicap... :)
03:44.18Corydon76-homeI thought the competition was for some hot ass..
03:44.26twisteddudes, chicks dig big shiny things
03:44.31twistedat least the ones out there do
03:44.45twistedyou can look like ron jeremy and still get some
03:44.48twisted(obviously)
03:45.08Corydon76-homepretty pretty blue eyes
03:45.21Corydon76-homeI could get lost in those pretty blue eyes
03:45.24blitzragewow... SanFran is 400 miles from Anaheim :)
03:45.33twistedhahahaha
03:45.34twistedhttp://www.thewgalchannel.com/news/5056040/detail.html
03:45.49blitzrageI really didn't plan this trip very well... :)
03:46.12Corydon76-homeHitchhike?
03:46.16twistedblitzrage, scooter?
03:46.26blitzragelol
03:46.31*** join/#asterisk dos000 (n=dos000@i216-58-27-16.cybersurf.com)
03:46.33blitzragetwisted: you can drive me up in your Escalade
03:46.36dos000howdy
03:46.40blitzragetwisted: and I'll even let you pay for the gas
03:46.41Juggieblitzrage, flying into lax?
03:46.46twistedHAH... i'll be too busy wiping the champagne off the seats
03:46.46Juggieahh nm
03:47.38twistediunno
03:47.47twistedmebbe i should just pick up women in east la
03:47.51twistedi hear they're feisty
03:48.01theblueLA?
03:48.02dos000is there a uk provider for all these companies offering unlimited uk/europ plans lately ? i am wondering if they are putting gateways in all these countries
03:48.14hardwirehow is cdr_custom.c different from cdr_csv.c ?
03:48.14thebluedos000: www.voipbuster.com
03:48.23hardwireis it in addition too.. or are there conflicts
03:48.28thebluetwisted: LA, which one?
03:48.36twistedtheblue, the one i'm talking about
03:48.40*** join/#asterisk marshall (n=test@S0106000f3dcdd088.wp.shawcable.net)
03:48.42thebluetwisted: Los Angeles or Lower Arkansas?
03:48.42twistedblitzrage, when you getting in?
03:49.00twistedtheblue, if that was supposed to be funny, you failed horribly
03:49.09thebluetwisted: I'm entirely serious.
03:49.26twistedtheblue, only people in arkansas refer to south arkansas as LA
03:49.37thebluetwisted: And.....
03:49.37twistedand I'm not quite in arkansas
03:49.42wunderkintheblue: louisana does still exist.. in some form or another..
03:50.05theblueAs New Atlantis.
03:50.14dos000theblue, how they doing it ? where do they get the wholesale prices from
03:50.21thebluedos000: I have no clue.
03:50.36thebluedos000: You pay 5 euros once, and then you get unlimited calls to landlines in the US and most of Europe.
03:50.36*** join/#asterisk bmg505 (n=leon@rndf-146-7-165.telkomadsl.co.za)
03:50.39twistedsheesh. new orleans is NOT louisiana.. it is IN louisiana
03:50.46thebluetwisted: I know./
03:50.52blitzragetwisted: I arrive this Thursday (Oct. 6)
03:50.58twistedamericans suck at geography
03:51.01wunderkintwisted: theres more?
03:51.05wunderkinlol
03:51.12dos000theblue, dotcom called .. they want their flawed bus plans back
03:51.13twistedblitzrage, where will you be on monday the 10th at about 7pm PDT?
03:51.20blitzragetwisted: not sure...
03:51.24thebluedos000: ...
03:51.24blitzragetwisted: at the hotel?
03:51.27twistedblitzrage, do i have your cell #?
03:51.28blitzragetwisted: drinking? :)
03:51.40blitzragetwisted: I don't know... you'd be able to answer that question better than I :)
03:51.42thebluedos000: That wasn't funny at all.
03:51.55theblueWhat?
03:51.57twistedblitzrage, i do not have your cell
03:52.04dos000theblue, it makes no bus sense at all .. serious
03:52.11thebluedos000: I know.
03:52.15thebluedos000: But it works, somehow.
03:52.30thebluedos000: Like liquid soap.
03:52.41dos000theblue, now i need wholesale tho.
03:52.49thebluedos000: To resell?
03:53.14dos000theblue, yes. I figured in NA already. but europe is total darkness for me
03:53.25thebluedos000: Ah.
03:53.50dos000theblue, do you get better link/rates in uk or france etc ...
03:53.58thebluedos000: Exactly the same.
03:54.15theblueActually, no.
03:54.25theblueI don't get free calls to France.
03:54.39theblueIt is 1.1 UK pence a minute to France.
03:55.21*** join/#asterisk PBXtech (i=nik@136.sub-70-218-100.myvzw.com)
03:55.27dos000france, uk, spain .. which one has the closest and is the best place to host a gateway. in case i do not stumble on a good wholeseller
03:55.36theblueNo idea.
03:55.46dos000theblue, are you in uk ?
03:55.52thebluedos000: No, I'm in the US>
03:56.03dos000tow!
03:56.09thebluetow?
03:56.17twistedi see london
03:56.18twistedi see france
03:56.24dos000homer quotes
03:56.25thebluei see dos000's underpants.
03:56.27theblueOh.
03:56.31theblueYou mean "D'oh!"
03:56.39dos000nop
03:56.42theblueOh.
03:57.14Corydon76-homeNOP, as in the mnemonic for "No Operation" in 6502 assembly
03:57.29theblue6502 assembly is scary.
03:57.45Corydon76-homeNah... multiplication was fun...
03:58.00dos000i never did 65002  by the time i got in they were teaching 8086
03:58.01Corydon76-homedivision, less so.
03:58.20Corydon76-home6502 is more advanced than 8086
03:58.28theblueDespite the number.
03:58.37dos000but nop was still there if my memory is correct
03:58.51Corydon76-homeYeah, the number doesn't have anything to do with it
03:58.57theblueCorydon76-home: Duh.
04:00.36dos000anyone know uk colocation and pri costs ?
04:06.32blitzragehrmmmm... either I take a plane from Anaheim => San Fran, layover in Las Vegas, NV (4.5 hour trip) and $139, Anaheim => San Fran direct (1.25 hour trip) for $307, or Greyhound (10 hour trip) for $37
04:08.32wunderkinblitz did you check prices from lax?
04:08.38wunderkinand the area
04:08.50blitzragejust used travelocity....
04:09.10blitzragemight look at some other sites... but flying is going to be a bitch I think
04:09.21blitzragebus might take longer... but probably a hells of a lot easier
04:10.16*** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net)
04:10.27blitzrageQwell[]: zup zup
04:10.35Qwell[]blitzrage: y0
04:11.39Corydon76-homeblitzrage: what's in SF, anyway, other than gay bathhouses?
04:11.54Corydon76-homeAh!  So that's why you're going...
04:12.12hardwireis G722 what you would see on some E&M lines?
04:12.18Qwell[]Corydon76-home: same thing as Hollywood
04:12.29hardwireI hear for some weird reason the audio on some PRI is 56k w/ signallign above that
04:12.47*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
04:12.48Corydon76-homeTrue... we could put blitzrage in West Hollywood...
04:12.53Corydon76-homeconsiderably closer
04:13.09Qwell[]tons to do in Hollywood
04:13.15Qwell[]gay or otherwise, heh
04:13.23Corydon76-homeLots of teenage hustlers...
04:14.30blitzrageCorydon76-home: I'm going to see a female friend of mine
04:14.40Qwell[]in SF?
04:14.42Qwell[]ask for ID
04:14.49blitzrageQwell[]: I already know her from previous :)
04:14.57blitzrageQwell[]: she's of age :)
04:15.05Qwell[]in SF, it isn't age you need to worry about
04:15.15wunderkinheheh
04:15.17Corydon76-homerofl
04:15.19blitzrageQwell[]: well... I know her from before... so I think I'm pretty safe :)
04:15.26Qwell[]blitzrage: okay :)
04:15.28wunderkinit looks real at least
04:15.39blitzrageQwell[]: oh, and she's hot... so that helps :)
04:15.42Corydon76-homeYeah, and she had a pussy then... and... well, she might still...
04:15.48Qwell[]I know of quite a fe...nevermind
04:16.00blitzragehypa7ia: hey oh!
04:17.25MikeJ[Laptop]hmmm
04:17.38blitzragehypa7ia: I was on the phone tonight with a guy from the states... and he could tell I was Canadian due to my accent :)
04:17.43twistedwhoa
04:17.47twistedpeople
04:17.50twistedholy hell
04:17.55*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
04:17.56simoncion[away]*listens*
04:17.58twistedhypa7ia, what's up :)
04:18.01blitzragesince when is hell holy?
04:18.07blitzrageand since when does it exist? :)
04:18.18Corydon76-homesince twisted took over
04:18.20twistedblitzrage, since technically lucifer is fallen angel from god's little shack
04:18.44twistedblitzrage, caltrain
04:18.53Corydon76-homeWhat, on Highway 1?
04:19.08twistedwhoa
04:19.12twistedsimoncion[away], you who i think you are?
04:20.07blitzragetwisted: caltrain?
04:20.13hypa7iablitzrage: this chick in a meeting today said aboot repeatedly
04:20.22hypa7iatwisted: namooch
04:20.27twistedhypa7ia, ;)
04:20.28blitzragehypa7ia: lol
04:20.31knight_heh
04:20.32Qwell[]blitzrage: yeah, amtrak
04:20.32twistedhaha
04:20.37twistedi've heard blitzrage say aboot quite a few times
04:20.38Qwell[]I think
04:20.48knight_eh?
04:20.53blitzragetwisted: its true
04:20.57twisted'tis
04:21.02blitzrageI don't deny my Canuckizm's
04:21.04wunderkinat least he admits it
04:21.08twistedand yes, blitzrage, caltrain
04:21.14hypa7iaaccording tosome peopel i have a valley accent
04:21.15blitzragetwisted: looking into it now
04:21.23hypa7iaan /ottawa/ valley accent :-)
04:21.48blitzragehypa7ia: thats pretty specific :)
04:21.49twistedblitzrage, looks like i might me mistaken aboot the caltrain
04:21.59shido6ewww ottawa
04:22.04hypa7ialol
04:22.06twistedi thought it ran along the coast, but apparently it only runs between SJC and SFO
04:22.11hardwirehttp://www.bluetooth.com/products/prods.details.asp?CPID=1057&CAT=19
04:22.14hardwireperfect for BTP
04:22.46blitzragetwisted: burn... :(
04:23.00Qwell[]How close is Oakland to SF?
04:23.14blitzrageother side of a bridge me thinks
04:23.20Qwell[]http://www.amtrak.com/servlet/ContentServer?pagename=Amtrak/am2Route/Vertical_Route_Page&c=am2Route&cid=1081256321841&ssid=135
04:23.22Qwell[]perhaps
04:23.22Pete_Largoanyone have a sample config file to a Mediatrix 2102?
04:23.33blitzrageQwell[]: looking
04:23.37Corydon76-homeOakland is east of SF
04:23.43Qwell[]thats not entirely coastal, but...
04:23.44Corydon76-homeJust across the water
04:23.52twistedblitzrage,
04:23.56twistedcashuttlebus.com
04:24.06twistedruns between LA and SF for $45
04:24.10blitzrageQwell[]: even part way would be cool
04:24.34Qwell[]blitzrage: father-in-law took that trip, all the way up to Seattle, said it was a nice trip
04:24.52twistedblitzrage, did you see the link?  you can thank me with beer
04:25.20blitzragetwisted / Qwell[]: checking those links
04:25.23blitzragemuch obliged!
04:25.33twistedQwell[], $45 trumps amtrak
04:25.40Qwell[]yeah, probably
04:25.46twisted;)
04:25.46*** join/#asterisk lehel (n=asd@82.79.20.17)
04:25.51twistedbut amtrak is cool
04:26.02hypa7iayay trains!
04:26.18*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
04:26.22blitzrageI like the trains
04:26.34hypa7iavia rail == <3
04:26.40twistedyeah
04:26.41blitzragewish Streetsville didn't suck so hard core in terms of its lack of trains
04:26.47twistedbut i bet the train == $$$$$$$$
04:26.49blitzragehypa7ia: I agree
04:27.37blitzrageI'll look into it more tomorrow... I"m tired... :(
04:27.42blitzragetoo much programming...
04:27.48blitzrageand I'm not a programmer -- so it hurts me :)
04:28.04twistedhahahaha
04:28.05twistednevermind
04:28.09twisted$8.50 for a train ride
04:28.13twistedniiice.
04:28.21twisteder wait
04:28.24twistednevermind
04:28.30twistedthat's anaheim to las angeles
04:28.33Qwell[]heh
04:28.35hardwireis it wrong that I am planting bluetooth trackers on my employees?
04:28.36blitzrageheh :)
04:28.38lehelhello
04:28.45blitzragehardwire: not at all
04:28.48theblue?\
04:28.48theblueWoah.
04:28.48theblueThat was dumb of me.
04:28.48theblue?\
04:28.56Qwell[]trains are nice.  I used to take a train to work for about 6 months
04:29.17hardwire"Sir he is on the desk having sex .. please call back another time"
04:29.18twistedokay
04:29.23twistedit's $45 for a bus ride
04:29.29twistedand $52 for a train ride
04:29.43blitzragenot bad
04:29.48twistedbut the bus is one bus
04:29.48blitzrageeither one is better than the plane
04:29.49Qwell[]train is probably non-stop
04:29.54twistedthe train is 4 different trains
04:29.59blitzrageewwww
04:30.00theblue?
04:30.05Qwell[]twisted: Really?
04:30.06twistedwith layovers
04:30.12twistedQwell[], yep
04:30.32twistedactually
04:30.37twistedamtrak does part of this by bus too
04:30.37blitzrageOpening balance: -0.88   Transfer: 0.12   Overdraft Interest: 0.01   Closing balance: 0.11
04:30.38twistedso actually
04:30.40blitzragelol
04:30.42twistedit's one train
04:30.43twistedone bus
04:30.44twistedanother train
04:30.44*** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net)
04:30.45twistedanother bus
04:30.46theblueHi all.
04:30.55Qwell[]twisted: You looking at the 14?
04:31.05twistedthe 14?
04:31.10blitzrageif you need a trip planned... GO TO #asterisk!
04:31.11Qwell[]the coast starlight
04:31.17twistedno
04:31.19Qwell[]heh
04:31.26twistedi'm looking at the two routes they have for ANA to SFW
04:31.33blitzrageI leave on Sat. Oct. 15th to San Fran btw
04:31.36blitzrageif that makes a difference
04:31.36twistedyeah
04:31.37hypa7ia#asterisk: the irc travel agency
04:31.39twistedi picked sat the 15th
04:31.45blitzragetwisted: good call
04:31.50twisted:)
04:32.38blitzragedarn... no GST cheque :(
04:33.07*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
04:33.16twistedhaha
04:33.20twistedoverdraft interest?
04:33.21twistedwtf?
04:33.25twistedi thought US banks sucked
04:34.09blitzragehence why I rarely use that bank account :)
04:34.38blitzrageits really just a remnant of my student account from elementary school - I mostly just keep it because it has a wicked bank number
04:34.54twistedwhat's the bank number?
04:34.55twistedlol
04:35.02Qwell[]its only funny if you can share it, heh
04:35.10hypa7ia13371337
04:35.16Qwell[]hypa7ia: :p
04:35.19blitzragehypa7ia: shhhhh! I told you that in private!
04:35.20twisted80081351337
04:35.31Qwell[]twisted: nice
04:35.37twistedQwell[], ;)
04:35.57Qwell[]there was this loan number we always used to use for testing at work...
04:36.06Qwell[]but it was a real valid loan...damn easy to remember
04:36.12Qwell[]but I can't say what it was. :(
04:36.19twisted6969
04:36.30Qwell[]lets pretend it was 10204080
04:36.35Qwell[]erm, too short
04:36.39Qwell[]1020408099
04:36.43twistedlol
04:36.53Qwell[]quite similar format
04:37.15wunderkinQwell[], routing?
04:37.29Qwell[]wunderkin: nah, home loan number
04:37.55wunderkinoh yeah not check acct ;(
04:37.56Qwell[]home mortgage*
04:38.10theblueNight, all.
04:38.24wunderkinbye blue and buy bonds
04:40.55twistedbah
04:41.49blitzragehehe
04:42.03twistedooh look
04:42.04twistedi make smokey
04:42.05blitzragetwisted: that CA shuttle bus has a destination point of 2.3 mi from where I need to go!
04:42.12*** join/#asterisk popvoxdave (i=user@dave2.toad.net)
04:42.28blitzragetwisted: now if I can convince them to pick me up from the hotel... that'd be sweet (its not on their list)
04:42.46twistedah well
04:42.57twistedi'm sure you can drag your shit up the street
04:43.28*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
04:48.40*** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net)
04:49.30*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
04:49.49argos73any ideas what would keep a sip phone from authenticating to *, other than sip.conf ?
04:50.02*** part/#asterisk popvoxdave (i=user@dave2.toad.net)
04:50.05*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
04:51.48blitzragetwisted: hrmm... wonder how I would get 25 mi up the road to the other hotel which is the closest one...
04:51.57blitzragetaxi would be way too expensive I'm sure...
04:52.31ManxPowerblitzrage, Astricon?
04:52.37blitzrageManxPower: aye
04:52.45argos73blitzrage: hoof it?
04:52.46argos73:)
04:52.53blitzrageManxPower: trying to figure out how I'm getting from Anaheim to San Fran
04:52.54twistedargh
04:52.57twistedmy guitar filters suck
04:53.02twistedneed to make new ones
04:53.07ManxPowerblitzrage, not staying at the astricon hotel, or not staying at the astricon hotel the entire trip?
04:53.25ManxPowerblitzrage, can you rent a car?
04:53.43ManxPowerSince it's the LA area, a car is rhe only viable mode of transport.
04:53.43blitzrageManxPower: I'm at the Astricon hotel until the 15th ... then I'm going from Anaheim to San Fran for a few days
04:53.54ManxPowerblitzrage, Ah.
04:54.00blitzrageManxPower: but I only need to go 25 mi :)
04:54.05blitzrageManxPower: then I can bus it the other 400 mi
04:54.11blitzrageManxPower: for $45.
04:54.14ManxPowerblitzrage, Ah.
04:54.16FuriousGeorgehas anyone run asterisk on colinux?  the zaptel hw should work right?
04:54.29blitzrageFuriousGeorge: uhh.. I think the exact opposite of that
04:54.48ManxPowerblitzrage, 25 miles in Los angels without a car is like 400 miles in most places.
04:55.01blitzragecolinux is that thing that runs on Windows.. ?
04:55.04blitzragewho knows...
04:55.06FuriousGeorgeblitzrage: everyone has run asterisk on colinux and the zaptel hw wont work?
04:55.07ManxPowerblitzrage, take amtrak.  It's a pretty ride, I think.
04:55.08blitzrageManxPower: hrmmm... shitty deal
04:55.32FuriousGeorgeblitzrage: its an open source kernel module that allows you to run the linux kernel alongside windows
04:55.33blitzrageManxPower: hrmmm... apparently its a train -> bus -> train -> bus via AmTrak
04:55.34ManxPowerblitzrage, convince Olle to drive you to the train/bus station 8-)
04:55.45blitzrageManxPower: that's a great idea...
04:55.52blitzrageManxPower: not sure why I didn't think of that :)
04:56.01FuriousGeorgenow that i think about it the zaptel hw probably wouldnt work since the even the nic is "virtual"
04:56.04blitzrageManxPower: just need to get dropped off at another hotel :)
04:56.10ManxPowerblitzrage, Um, Amtrak goes direct from Anaheim to Untion STation/LA and you can leave from there via train to SF.
04:56.17ManxPowerOR really, just spend the money and fly.
04:56.19blitzrageFuriousGeorge: if its virtual, it definately won't work
04:56.41blitzrageManxPower: I'd rather not spend money I don't have
04:56.59blitzrageand I'm starting to hate flying
04:57.28blitzragewelp... I"m off to try and find a snack... stomach is hungry
04:59.16tessierAnyone know how many inches in 1u of rackspace?
04:59.24blitzrage4 ?
04:59.26wunderkintessier 1.75
04:59.28blitzragedoh
04:59.32blitzragebad guess :)
04:59.36tessierI put "1.75 inches to u" into google and it didn't give me a conversion
04:59.42tessierThe first link was amusing though.
05:00.10tessierwunderkin: 1.75" is 1u? Cool, so this is a 1u server. I was thinking it was 2 for some reason.
05:00.17jarrodguys im receiving DTMF tones in the middle of conversations using MGCP to various endpoints
05:00.29tessierjarrod: Stop pushing buttons on your phone
05:00.35JerJerjarrod: turn off inband detection
05:00.38jarrodMGCP debug shows it coming from the remote side
05:00.39JerJeron your device
05:00.49JerJerthen tell the far end to stop sending them
05:00.55jarrodthey arent
05:01.05jarrodit just happens randomly
05:02.16tessierDoes your friend have a particularly high pitched voice?
05:02.39jarrodit occurs on both men and women, low and high
05:02.40tessierDo they whistle at you over the phone?
05:02.42*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
05:02.46tessierAre your male friends gay?
05:02.49jarrodheh.. no..
05:02.59tessierI was sure they were gay.
05:05.43*** part/#asterisk techie (i=gus@70.86.133.66)
05:06.03ManxPowerMGCP should not support inband DTMF anyway
05:06.14*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
05:06.34Qwell[]tessier: it might work if two people are talking
05:06.36jarroddtmfmode=none
05:07.20redder86IAXmodem 0.0.3 is released... https://sourceforge.net/projects/iaxmodem/  please beat it up  :-)
05:07.53ManxPowerjarrod, It's called "talkoff" and happens at the point where PSTN is converted to VoIP.
05:08.21dos000jarrod, howdy
05:08.43hardwirewhy use res_mysql vs res_odbc?
05:08.53ManxPowerrelaxdtmf=yes in Asterisk causes the same problem.
05:09.01Qwell[]hardwire: odbc is usually slower
05:09.14hardwirethats all I could think of
05:09.21hardwireand where are the sql CDR tables?
05:09.27hardwirein asterisk cvs..
05:09.29jarrodi have tried yes and dno on zapata
05:09.32hardwireits not terribly documented.
05:09.38Qwell[]jarrod: try no instead of dno
05:09.53jarrodha thats what i meant
05:10.04ManxPowerjarrod, you want it set to no.  Of course the option has to be BEFORE the channel line you want it applied to
05:10.11ManxPowerthen stop and start asterisk
05:10.20ManxPoweror unload chan_zap.so and load chan_zap.so
05:10.26jarrodyea its on the pri
05:10.32*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
05:10.37ManxPowerif using CVS-HEAD or 1.2bets you should be able to just do a reload chan_zap.so
05:11.22jarrodasterisk defaults to no
05:13.10*** join/#asterisk CoiL (n=bah@pcp0010655888pcs.huntsv01.al.comcast.net)
05:13.48*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
05:16.29argos73grrrr....  several sip devices register with one * server just fine, but they all fail on a different server with the same sip.conf
05:18.27hardwireso astcc needs asterisk-perl
05:18.28Qwell[]argos73: firewall?
05:18.31hardwirebut atserisk-perl isn't in cvs
05:18.41Qwell[]hardwire: I didn't need to install that
05:18.48Qwell[]unless it came with *@h...dunno
05:18.49hardwireit requires AGI.pm
05:18.56Qwell[]oh
05:19.38*** join/#asterisk kks (n=kks@202.73.8.130)
05:19.49hardwireyeh
05:19.49hardwiredoh
05:19.57argos73Qwell[]: nope...  they talk both ways, but on on server, the registration just stops after packet #3
05:20.07argos73(first "Unauthorized" message)
05:21.58jarrod"prevent clipping of DTMF tones in an established call"
05:22.05jarrodI wonder what that is referring to in chan_mgcp.c
05:22.45argos73really annoying, since it's the server at work that's giving me problems.  box at home works just fine, but I'm trying to set up an "asterisk demonstration" system to convince the company prez that this "asterisk thing" works. :)
05:25.04Inv_arphmm... why is it every so often sip registry fails to my provider,  sip reload fixes it tho
05:26.10*** join/#asterisk PBXtech (i=nik@85.sub-70-218-35.myvzw.com)
05:26.37ManxPowerInv_arp, switch to using an IP in sip.conf instead of a hostname.
05:26.50ManxPowerInv_arp, using 1.0.x?
05:27.14PBXtechwill the quad fxo work without power?
05:27.25ManxPowerPBXtech, that's the rumpr.
05:27.29ManxPowerrumor too
05:27.41PBXtechthey seem to
05:27.50PBXtechjust cant see the fxs mods
05:29.04argos73i'd be a little cautious, though...  the card might try pulling a little too much current from the pci bus, depending on how it's designed...
05:29.19Qwell[]fxo shouldn't need power
05:29.31PBXtechok
05:29.37argos73Qwell[]: no power?  at all?   :)
05:29.42argos73cool!
05:29.52PBXtechit seems to work
05:31.04Inv_arpManxPower: yea 1.0.x
05:31.30ManxPowerInv_arp, 1.0.x 's DNS support REALLY sucks.  if it fails once you have to reload chan_sip
05:31.49Inv_arpManxPower: ahh k
05:33.32PBXtech2nd Q.. why with 1.09 i get 1way audio on xtenlite and 1.2 i get 2way audio
05:38.25argos73ok - with sip debug turned on, packet #2 - working server responds with "Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY", and the non-working one doesn't include "NOTIFY" at the end...  any ideas?
05:42.47*** join/#asterisk spackle (n=spackle@209.234.83.19)
05:43.00ManxPowerPBXtech, you prolly have some NAT option enabled in X-Lite.  Don't do that.
05:43.33ManxPowerargos73, NOTIFY is typically associated with a mailbox= option in that SIP.conf entry
05:43.53twistedlol
05:44.13twistedleave it to me to get distracted by garage band and start making dance club music rather than with my guitar
05:45.03twistedi was simply going to practice my guitar, then hit a few beats and went "hrmmm"... guitar got put down, headphones on, and off to mix land i went..
05:45.07argos73ManxPower: makes sense
05:45.08twistedi hate having ADD
05:46.56argos73chosen context shouldn't affect registration, correct?
05:48.55*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
05:51.50*** join/#asterisk fugitivo (n=ajf@201.255.103.240)
05:52.24blitzragetwisted: I went and jammed on some drums :)
05:54.29twistedheh
05:55.26twistedbeat them drums, sucka
05:55.37argos73dammit....  just copied ehtire * config directory verbatim from working to broken server, and it does the same thing.
05:58.04*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
05:58.25jontowto what end do we use editline in asterisk's CLI?
05:59.06twisteduse the source and see :P
05:59.11jontowi've been looking :)
05:59.17*** part/#asterisk spackle (n=spackle@209.234.83.19)
06:04.49*** join/#asterisk oej (n=Olle@apollo.webway.se)
06:06.23*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
06:12.08*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
06:12.26jontowyeah, i got halfway there.. im looking at command line editing vi-style
06:22.56*** join/#asterisk sneak (n=sneak@64.220.234.21.ptr.us.xo.net)
06:31.00argos73arrgh, fsck, kill...  just set both machines to identical software, identical configs.  one works, one doesn't...
06:32.47Dr_Rayhow much is CCM for 10 phones?
06:34.27argos73asked my cisco rep the same thing - the smallest they do is for 100 sets - around $5K
06:35.08argos73going over 100 sets is cheap, relatively.
06:35.33opus__how much?
06:35.41opus__i bet CCM is probably around $20k
06:36.07Dr_Rayscrew that
06:36.09Dr_Rayer
06:36.11argos73CCM + Unity VM for 80 sets was $12K with their hardware
06:36.22Dr_Raynot including the phones
06:36.23opus__whoah Thats a deal
06:36.28opus__Oh not including phones hmm
06:36.39opus__phones can be $35k of the deal ! :)
06:36.40Dr_Raygeebus
06:37.24argos73CCM, Unity, 60 x 7960, Call Attendant - total quote around $83K
06:37.36opus__hmmm.
06:37.40opus__thats more then $1k per user
06:37.53argos73yea, but it's cool... :)
06:37.58opus__shoretel 50 users $55k
06:39.58Dr_Rayand don't forget the smartnet account
06:43.40*** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net)
06:46.51*** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org)
06:47.57AmbroseI'm having a problem getting my FXO card to "work". I'm getting a "No channel type registered for 'Zap'" error in the CLI. Can anyone offer any assistance? thx
06:48.48JamesDotComload chan_zap.so?
06:49.24wunderkinholy shit
06:49.40wunderkin<delayed reaction>
06:50.59af_I a wondering: if I put a Dial, priority 1 when a call arrive on pstn, people calling consumes some money?
06:51.15AmbroseWhere would I go to add chan_zap?
06:52.07Ambrosein the cli?
06:53.28Dr_RayAmbrose - is your zaptel.conf setup?
06:54.34AmbroseDr_Ray: Yes, has 3 lines, fxsks=1-2 and loadzone and defaultzone both = us
06:55.43Dr_Rayand ztcfg -vv after you loaded zaptel and your kernel module?
06:56.28AmbroseOh, I installed the zap drivers after I installed Asterisk. I just re-made asterisk with 'make' and now chan_zap.so is in the usr/lib/asterisk/modules. I'm just restarting asterisk now to see if it'll work
06:56.39AmbroseYes I had ran ztcfg and both channels showed up
06:57.47AmbroseOk I rebooted and reloaded the wcfxo and zaptel modules and I still get the same error
06:58.07Ambroseapp_dial.c dial_exec: Unable to create channel of type 'zap'
06:58.57*** join/#asterisk xs-- (n=xs@217.67.230.185)
06:59.06*** part/#asterisk xs-- (n=xs@217.67.230.185)
06:59.39JerJerand people say asterisk sucks at faxing
06:59.55argos73Ambrose: channels defined in zapata.conf ?
07:00.32Inv_arpquick ques on a MASQ *nix box with 2 dsl links, need all internal ip's from 192.168.0.3-10 (phones) to go thru eth0 and the rest thru eth1,  is that a job for iptables?
07:01.11Ambroseargos73 Ah! I had commented out both the channel => 1 and channel => 2 lines
07:01.13AmbroseThanks!
07:01.21argos73heh - no prob
07:01.26AmbroseThanks so much! :-)
07:05.12tmiragedoes anybody here maybe understand or has AMP running on asterisk? and especially with gentoo
07:05.25leheli have "echocancel[whenbridged] = yes" but there is still echo, i hear double myself < this happens when i'm talking with an IAX user from a zap (tdm400p), and after a while (like 10-15secs) interrupts
07:05.25tmiragecause i keep getting the same error when i want to start compiling
07:05.33tmiragehad something to do with php
07:07.27argos73time to snooze..  later
07:08.35*** join/#asterisk denon (i=denon@synapse.subneural.net)
07:08.35*** mode/#asterisk [+o denon] by ChanServ
07:09.15lehelany ideeas?
07:09.22*** join/#asterisk Tangent (n=Arc_Tang@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk)
07:10.16tmirageis echotraining also on?
07:10.57Dr_Rayand does this echo go away after a few seconds
07:12.05lehelDr_Ray nope.. tmirage: echotraining? in iax.conf?
07:12.35*** join/#asterisk _proleone (i=jjmtrev@203-173-26-187.dyn.iinet.net.au)
07:13.07_proleonehey, im just trying to register and use asterisk with a fwd address
07:13.12_proleonewhen im dialing the address i get
07:13.12_proleone<PROTECTED>
07:13.12_proleone<PROTECTED>
07:13.22_proleonecan i test to see if im registering properly?
07:13.27tmirageno you have to put that in your zapata.conf cause you are using the zaptel right or am i missing something in this story
07:13.51_proleoneme? im ztdummy
07:13.55tmirageno lehel
07:14.36kaldemar_proleone: give 'sip debug' in cli and see if you're getting an answer from fwd.pulver.com.
07:15.18lehelright tmirage zaptel.., ok.. i'll try that too;
07:16.47_proleonewhat am i looking for kalademar
07:17.09_proleoneon my dialer im getting service unavailable
07:17.22kaldemargive the sip debug and make the call. then you should be looking at the SIP packet flow.
07:18.15kaldemaryou see all the incoming and outgoing packets. if there are no incoming ones, you've clearly got some problem in the connection to pulver.
07:18.49_proleonei dont see any from pulver
07:19.21kaldemaris your network connection ok to pulver?
07:19.21_proleoneregister => youngpro:password@fwd.pulver.com/1001
07:19.23_proleoneis this the right syntax?
07:19.27_proleoneyeah
07:20.55_proleoneoh does
07:21.07kaldemarthat is the right syntax.
07:21.13_proleonepulver need to communicate to port 5060, do i need to open my firewall
07:21.19_proleoneto allow the incoming connection on port 5060
07:21.40kaldemarwhat does 'sip show registry' say?
07:21.46*** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it)
07:21.58_proleonenothing
07:22.16_proleonefwd.pulver.com/youngpro    69.90.155.70                255.255.255.255  5060     Unmonitored
07:22.21_proleonebut sip show peers does show some sort of connection
07:23.09*** join/#asterisk folsson (n=filip@h82n1fls32o985.telia.com)
07:23.10*** join/#asterisk psycodad_ (n=obiwan@2001:4060:4419:b1:0:0:0:2)
07:23.29kaldemari think you should see the connection in 'sip show registry' also.
07:23.44_proleonehm
07:25.16_proleonedoes the register command need to be at the top of sip.conf
07:25.19_proleoneunder [general]
07:25.41kaldemaryes, under [general].
07:26.02_proleoneah NOW
07:26.05_proleoneive got a sip show registry
07:26.47_proleoneOct  5 16:53:58 NOTICE[19943]: chan_sip.c:9451 handle_response_register: Failed to authenticate on REGISTER to 'youngpro@fwd.pulver.com' (Tries 3)
07:27.49kaldemaris your userid/password correct?
07:27.53_proleonei believe so
07:30.40_proleoneSIP/2.0 200 Received non-routable address,  use public IP next time
07:32.58kaldemarare you behind a nat?
07:33.07_proleoneyes a dynamic ip nat
07:33.12_proleoneand im just looking at that setting now
07:33.17_proleonebut how do i work with the dynamic ip?
07:34.13kaldemarhttp://www.voip-info.org/wiki/index.php?page=Asterisk%20config%20sip.conf
07:34.17kaldemarhave you taken a look at that?
07:34.37kaldemarthere's a externip=xxx option.
07:34.42_proleoneno i havnt seen this actually
07:35.16kaldemaryou could set up dyndns for example.
07:36.20*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
07:38.00*** part/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
07:39.18_proleoneso i need to do something like tha
07:39.20_proleoneokay
07:40.05kaldemarif you're going to take inbound calls.
07:41.04_proleoneokay kaldemar, ill set that up
07:41.26_proleoneone more question, if i put a fxo card in, can i spoof caller id over pstn networks
07:41.58wunderkinno
07:42.13kaldemardoubt it.
07:42.48wunderkinyou would need voip or pri that would let you do that
07:43.31_proleoneokay cool\
07:44.27*** join/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za)
07:44.34musical_Ducklo
07:51.10*** join/#asterisk rainman (n=bla@139.63.241.162)
07:51.33*** join/#asterisk Attila_Kovacs (n=kovacsat@dsl51B79142.pool.t-online.hu)
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07:58.02opus__<_proleone> SIP/2.0 200 Received non-routable address,  use public IP next time
07:58.11opus__localnet=192.168.0.0/255.255.0.0 is missing
08:01.43_proleonehow does that work
08:05.41_proleonethats all i need opus?
08:05.42*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
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08:18.30Attila_KovacsI have a HFC and a Winbond chip ISDN modem. Which channel do you suggest to use with them? Thanx.
08:25.04*** join/#asterisk Thoran (n=Thoran@p54A5A80F.dip0.t-ipconnect.de)
08:25.44*** join/#asterisk pr0 (n=LANmower@ndn-165-146-90.telkomadsl.co.za)
08:25.49pr0lo all
08:26.09pr0wonder if someone can help me with this...
08:26.33pr0I want to make a zaptel tarball that I can just extract to root...
08:26.36*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
08:27.11pr0the problem is the device connections /dev/zap...
08:31.24*** join/#asterisk flenders (n=fserto@61.8.29.101)
08:31.57flendersdoes anyone know a good VoIP provider in the UK?
08:33.23musical_DuckSay anyone know what could cause func_uri to fail because I can't get ast_uri_encode, sounds like module trouble but I don't know which one has ast_uri_encode :(
08:33.34musical_DuckGrr that face is sad :P
08:33.50pr0dont know if theire uk based, but diamondcard and mci are great providers for worldwide ip->pstn termination.
08:34.38Zeeekvoiptalk.org is in the uk, so is sipgate.co.uk
08:34.58flendersZeeek, thanks mate!
08:35.43iDunnosipgate appears to work quite well, too.
08:36.41Zeeekflenders voiptalk is decent but too expensive. Haven't trie sipgate uk check the wiki for more
08:37.24*** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net)
08:37.25*** join/#asterisk zoa (n=kkk@pirus.securax.be)
08:37.54pr0where can I find updated currency stats? I wanna know how many pounds a south african rand is.
08:38.06Zeeektry google
08:38.23pr0what would I search on, currency rates?
08:38.27pr0exchange rates!
08:38.28pr0aha
08:38.38Zeeekcurrency exchange will bring the first entry
08:39.00Zeeekxe.com I can never remember that URL
08:39.23flenderstry the currency calculator on bloomberg
08:39.31Zeeekthat's a good one too
08:39.37pr0hmm voiptalk is sorta expensive
08:39.44flendershttp://www.bloomberg.com/analysis/calculators/currency.html
08:39.44ZeeekI said that
08:39.56Zeeekare you in eu or usa?
08:40.06pr0who, me?
08:40.06Zeeek*SA?
08:40.09Zeeekyea
08:40.18pr0.za, south africa :)
08:40.22flendersI'm in AU
08:40.27pr0I did find the currency rates thx :)
08:40.36Zeeekyeah you might be better off with nufone, voicepulse connect or ICH
08:40.39musical_DuckWho is from za?
08:40.42Zeeekor voiptalk
08:40.48Zeeekshit, voipjet I mean
08:41.15ZeeekI use voiptalk occasionally because it pings at <30ms
08:41.24iDunnoor just put south african rand to GBP in google.
08:41.26Zeeekbut it's at least 2x more than most providers
08:41.26pr0thats a good ping
08:41.32iDunnothat'll give you the answer too.
08:41.47pr0thanx Idunno :)
08:42.36musical_Duckpr0 are you from ZA?
08:42.41pr0yup
08:42.45musical_DuckMe 2
08:42.51pr0<-- lower south coast
08:42.51musical_Duckwhich part
08:42.59musical_DuckNear CT?
08:43.15iDunno(googles currency convertor is neat :)
08:43.30*** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk)
08:43.58*** join/#asterisk paravoid (i=paravoid@147.102.26.135)
08:44.53pr0100km down the coast from durban, Shelley beach and margate.
08:45.18*** join/#asterisk devonst17 (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net)
08:46.12musical_DuckI am near CT, which company?
08:46.38*** join/#asterisk newl (n=newlook@203-59-91-217.dyn.iinet.net.au)
08:47.57pr0openvoice->govoip
08:48.12pr0I work for govoip, wich owns a goodly amount of openvoice
08:48.45*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
08:49.13musical_DuckRedOryx
08:49.21pr0Dont like my job though, I'm an artist
08:49.47pr0just happened to know linux real well
08:49.59musical_Duckme 2 got me into the thing real quick
08:50.27pr0I have an issue now
08:50.37musical_DuckBut, If you look at the error I posted above, you'll see I am having a crappy day as well grrr
08:50.55pr0<-- very crappy day
08:51.07musical_DuckSay openvoice contacted our company a while back
08:51.20musical_Duckwanted to work together or something
08:51.25pr0oh :)
08:51.28musical_DuckWhats happing
08:51.35pr0sounds like the johburg guys if you ask me
08:51.44*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
08:51.44musical_DuckThey clueless?
08:52.07musical_DuckWhat kinda problem do you have with asterisk?
08:52.30pr0I have to make an installable package out of bristuff
08:52.42pr0the zaptel drivers are the issue
08:52.44musical_Duckgrrr, on what distro?
08:53.00pr0slax, slackware based
08:53.16musical_DuckIts much easier and faster on gentoo
08:53.20musical_DuckI use it
08:53.32musical_Duckwell slightly faster at least
08:53.52pr0sure, but theres other reasons for us using slax here... anyway, its a livecd that you can add packages to, wich gets loaded on boot.
08:54.02musical_DuckCatalyst?
08:54.17pr0?
08:54.34musical_DuckLive cd creator for gentoo, dunno if its out for slack
08:54.57musical_DuckI mainly use gentoo for portage
08:55.17pr0yeah yeah, the creator isnt the problem, its making a package that unzips, runs a script, and you have zaptel.
08:55.57musical_DuckOh, you have to build own kernel too in most cases, sounds like fun
08:56.18pr0go tthat done, and all the systems its going on has the same os
08:56.28pr0the /dev/zap devices is the issue, they dont survive tarring and still work
08:56.44pr0not that I expected them to.
08:56.58musical_DuckI have never had to fiddle with devices, thats wierd, your doing some evil stuff
08:57.10pr0:) bwahahahaaa!
08:57.17musical_DuckThere are some switches in tar for devices i think
08:57.35pr0the other option is tar/zipping the source and having it compile each time the system boots, that way it will make them for me.
08:57.55pr0but the bristuff source and linux source together makes a hefty 170mb package.
08:59.04musical_DuckI had to get the device thing working for a modem setup a while back, silly faxing etc, lemme check where i put the script
08:59.20pr0hmm, that would be kind
08:59.21pr0:)
08:59.22pr0thx
09:01.43*** join/#asterisk inspired (i=mikael@213.197.167.61)
09:02.32*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
09:02.48inspiredis it possible to use SIP members in a queue and mark the ones that are on a call as busy? right now when call number two in the queue comes in, all members in the queue, even those that are on a call, ring
09:03.46*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
09:05.02*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
09:07.19musical_Duckpro udev right?
09:08.18pr0yeah I think so...
09:08.38pr0if not I can change it to be ;)
09:08.39musical_Duckkk hold on
09:09.09musical_DuckI actualy have patched mine slightly
09:13.43pr0thxies :)
09:13.54pr0never really screwed around with devices much before...
09:17.14musical_DuckGrr, the script only worked for char devices tho, still looking for it, somewhere on one of my boxes, please hold
09:17.44pr0oh... well, I might have a way of cutting most of the source-based install out
09:17.49pr0looking into it now
09:18.27pr0I think I just brought 170mb down to 30
09:18.38musical_DuckRPM's I shudder :)
09:18.51pr0rpms? tgz's
09:19.29pr0theire only tarballs with an install script added really
09:19.43musical_DuckActually, if it's source use bzip
09:19.59musical_Ducktar -cjf i think
09:20.49pr0theres two things about that though... 1 I dont know if slacware tgz's can be bzip files and 2 I dont know if slax's module maker can use bzipped tgz packages.
09:21.20musical_Duckmmm, it realy should, bzip is very common
09:21.50pr0yeah, but the little module maker is a alpha version windows utility ;)
09:23.04musical_DuckOh the horror, can see y your having a bad day
09:23.14pr0lolz :)
09:23.41pr0works very well, it just not clever at all.
09:25.36musical_Duckgrr, silly script seems to have wander of, I am running search in background but check this out meanwhile http://www.atnf.csiro.au/people/rgooch/linux/docs/devfs.html
09:25.53musical_Duckespecially bit about Permissions persistence across reboots
09:26.16*** join/#asterisk oej (n=Olle@apollo.webway.se)
09:26.22musical_Ducktho you'll have to muck about with devfsd etc grr
09:27.36pr0hmm
09:27.57pr0its so silly, there should really be a way to make a pre-compiled zaptel driver.
09:28.31musical_Duckwell, I you are using the same kernel you could just use the pre-compile lib file
09:28.39musical_Duckthats what all rpm distro's do
09:28.50pr0what do you mean...
09:29.35musical_DuckLike fedora k, you download wifi rpm for kernel, which is just pre-compiled so file, which is loaded with modprobe
09:29.46pr0yeah
09:29.53musical_Duck+ config of course
09:30.09pr0but the zaptel drivers need the /dev/zap, thats where the issue comes in I guess.
09:30.21musical_DuckSo If your using the same arch and kerne go for it
09:31.11pr0you see, theire not just a module, they actually use zapcfg as a module. wich requires /dev/zap to be there.
09:31.18limbiquecan anybody say what the extension: h means in the manager connection?
09:31.27pr0hangup
09:31.35limbiquecool, tnx
09:31.39pr0np
09:32.33pr0mmore like "upon hangup" do something.
09:32.48limbiqueok, like free add. data
09:32.54limbiqueor, write to db
09:33.02pr0its weird though, I compile zaptel on one os with the same kernel/arch, hop it over to another, try and install it and it wants to compile again...
09:33.29musical_Duckmmm, my /dev/zap is detected on bootup, maybe you can split install, evil windows installer thoughts grr
09:33.59pr0um, your /dev/zap is detected at boot?
09:34.05musical_Duckyup
09:34.16musical_Ducklet me make sure
09:34.17pr0um
09:34.43pr0if you plug out your card and reboot, they wont be there?, thats odd cause just after a make install theire present.
09:35.10musical_Duckmaybe I forgot an autoload somewhere
09:35.17musical_Duckbox is rebooting lemme check
09:35.26pr0thx :)
09:36.08musical_Duckyup it's there
09:36.32musical_Duckjust autoloaded the zaptel thingy tho
09:36.43*** join/#asterisk michael1234 (i=michael1@pavas-a715.racsa.co.cr)
09:36.54michael1234I have gotten this how do I debug
09:37.01michael1234T4XXP (PCI) Card 0 Span 1                OK                  0          0
09:37.08pr0hmm...
09:37.09musical_Duckso it seems that if you pre-compile a zaptel driver and auto-load then /dev/zap should be there, pr0
09:37.32pr0with auto-load you mean...?
09:37.39pr0sorry i'm a bit slow today :)
09:37.47michael1234I have no alarms on my channels
09:37.49musical_DuckYou have a script to autoload linux drivers?
09:37.53michael1234zaptel how can I test
09:38.08jonathhHey peeps.. i got the RxFax thing installed using the spandsp thingy.. asterisk runs.. no complaints that i can see... and when i run the RXFax command it does so without complaining... but for some reason it dont work
09:38.11jonathhany suggestions?
09:38.16musical_Ducktry ztcfg michael
09:38.33michael1234musical_Duck: and what will that do
09:38.40pr0I have hotplug... but I normally just add it to startup. the thing is just after a zaptel make install, /dev/zap is there.
09:38.45musical_Duckncurses config for alarms
09:38.51pr0that means that they are created and stay there.
09:39.12michael1234musical_Duck: ok when I run ztcfg -vvvvvv
09:39.17pr0if you for instance remove your hardware, they would still be there.
09:39.17michael1234what am I looking for
09:39.20jonathhis there anyway to get the spandsp stuff to dump some good debug stuff?
09:40.02musical_Ducksoz that should be zttool
09:40.14pr0jonathh, got fax working?
09:40.14musical_Duck:)
09:40.25musical_DuckI have fax problems as well
09:40.27musical_Duckgrr
09:40.38musical_Duckwhat is wrong with yours
09:41.47michael1234musical_Duck: zttool says no alarms
09:41.55jonathhno.. fax isn't working
09:42.08jonathhi added all the good stuff... and it runs without complaint
09:42.12jonathhbut doesn't seem to work
09:42.12pr0jonathh, sorry to hear
09:42.13musical_Duckhaven't used zttool yet, just thout you could make sense of it soz
09:42.29michael1234musical_Duck: how can I debug incoming on the zap channel
09:42.30jonathhwhen i dial into it.. negotiation doesn't seem to take place.. and eventually the line drops
09:42.44pr0there is something like zap show debug I think
09:42.55pr0or show zap debug or something.
09:42.55michael1234no there isnt
09:43.23michael1234nope
09:43.29michael1234any other guesses
09:43.58musical_Ducknope soz
09:44.01tmiragepri debug span?
09:44.40pr0debug channel
09:44.49jonathhis there anyway to get more debug on the fax stufs?
09:45.10pr0jonathh, I suggest the spandsp documentation... never used it myself
09:45.24jonathhok
09:45.34jonathhmaybe debug channel will reveal something?
09:45.41musical_Duckpr0 so if you can load the pre-compiled drive before you start your other stuffses everything should work or am I misunderstanding?
09:46.29pr0yup :), It basically adds the modules before anything else happens on the livecd
09:46.50pr0I tried zipping up the dev/zap but it doesnt keep the hardware data, just the files.
09:47.07pr0tar/gzipping that is
09:47.12michael1234can anyone help me with zap channels
09:47.31pr0michael did debug channel <channel name> work?
09:48.36musical_Duckwoot! Got a lead on my problem, silly CVS version leaves some evil files in strange places! :)
09:48.43leheli have echo in my zap channel when i call IAX.. why could this cause? could be becouse of a firewall.. router?
09:49.02pr0yeah head isnt always too great
09:49.28musical_DuckMmm now how scrub it away
09:49.33musical_Duckto that is
09:50.07musical_Duckso pr0 if the module is loaded and /dev/zap is present what else do you need, is it that zaptel tools?
09:50.37pr0all the other things arent a problem, checkinstall picked them up, /dev/zap is a problem though.
09:51.10pr0I got asterisk, zaptel and a lot of other things packaged correctly in one step with checkinstall, except for /dev/zap
09:51.14michael1234pr0: what is the zap channel
09:51.27michael1234eg
09:51.37michael1234debug channel xxxx. what is x
09:52.17pr0the same one you enter to dial with I guess
09:52.20pr0Zap/something
09:52.42musical_Duckwoot fixed my problem, CVS HEAD leaves an evil func_uri.so behind, you have been warned!! :)
09:52.54lehelzz
09:53.06michael1234pr0. Show do I find out what is available
09:53.12pr0musical_Duck, when going back from head, always delete the modules :)
09:53.13musical_DuckZAP/<channel number>c/<number too dial>
09:53.19pr0show channels
09:53.35pr0or zap show channels
09:53.50musical_Duckwhat is silly tho is that asterisk seems to just load all modules in a directory, which is bad form
09:53.54michael1234Channel              Location             State   Application(Data)
09:53.54michael12340 active channels
09:53.54michael12340 active calls
09:54.13pr0musical_duck you can disable that
09:54.23pr0then you have to tell it specifically each file to load
09:54.24michael1234pr0. does that mean it is not working
09:54.26pr0see modules.conf
09:54.41pr0michael show channels will show open channels
09:54.55musical_Duckgood, I am gonna do that, have keep the performace thing running through the whole show
09:54.56pr0zap show channels will show the zaptel ones if my memory servs me
09:55.12michael1234yep nothing
09:55.35musical_Duckare you using realtime mike?
09:55.58michael1234at just installed asterisk at home
09:56.05michael1234sorry I just installed
09:56.39michael1234my hassle is when I ring my pri I get engaged, I see no debug but I have verbose at 10.
09:56.52michael1234and it shows no alarms on the pris
09:56.53*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
09:56.59puzzledmorning
09:57.05pr0and if you plug it out, does it ring the same?
09:57.20musical_Duckmmm, sometimes channels don't hangup with silly dailplan but I don't know asterisk@home's setup
09:57.22pr0if so, you may want to call the telecom company.
09:57.26michael1234it is a pr0 if i unplug it wont ring
09:57.48pr0oh yeah
09:57.48musical_Duckesp. with voicemail
09:58.08michael1234I think the cahnnels havent created is what the problem is. I just dont know how to create them
09:58.20jonathhis ther anyone in the room that has spandsp working for receiving faxes? i need to know how it is suppose to behave when answering the line
09:59.20pr0does your card have lights on? theire handy sometimes.
09:59.38michael1234the alarms clear when I plug the cable in
09:59.49jonathhthing is.. i am trying to send the fax to a voipuser account...
09:59.50musical_DuckSo you are sure that driver dor zap is loaded etc
10:00.01michael1234how do I check that
10:00.07musical_Ducklsmod
10:00.15musical_Duckalso check last bit of dmesg
10:00.30musical_Duckshows it nicely
10:00.52rainmanI'm trying to use asterisk-im with the Jabber plugin for Trillian. I can dial from Trillian, but all I see in asterisk is "Manager received command 'Originate'", and nothing happens...
10:00.55michael1234Registered tone zone 1 (Australia)
10:00.55michael1234wct4xxp: Clearing yellow alarm on span 1
10:00.55michael1234Zaptel: Master changed to TE4/0/1
10:00.55michael1234wct4xxp: Clearing yellow alarm on span 2
10:01.36musical_Duckmmm
10:01.49rainmanWhen I try to use the Action: Originate manually on a telnet connection to port 5038, I get a "Permission Denied" on the telnet, and exactly the same as with trillian in the debug log...
10:01.56michael1234is that ehat you meant
10:02.45rainmanTrillian sucks monkeyballs, but that's not my point ;)
10:02.46musical_Duckyup, but not what I expected
10:03.01rainmanWell, it's a nice app, but bloated too much now
10:03.36pr0its nice if youre into msn and aol :)
10:03.43pr0and those sort of things
10:04.01michael1234any help
10:04.09musical_DuckSoz got nothing, grr
10:04.20rainmanWell, they also have a Jabber plugin, and it's the only jabber client atm that supports asterisk-im... Unfortunately, I think my asterisk config is not so good.
10:07.51rainmanAny ideas on why I get the Permission Denied?
10:13.24*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
10:16.51musical_Duckfor my next trick some faxing, can't wait...
10:17.30*** join/#asterisk trimi` (i=Pharrel@62.162.243.72)
10:19.50*** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129)
10:19.54rainmanResponse: Error
10:19.54rainmanMessage: Permission denied
10:19.54rainmanResponse: Error
10:19.54rainmanMessage: Invalid/unknown command
10:20.18rainmanThat's what I get when doing Originate, or Monitor, or whatever. I can't figure out why. Anyone have any ideas?
10:20.48*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
10:20.53MuppetMasterHello everyone
10:20.58musical_Duckpr0, have you got for /dev/zap thing working yet?
10:21.30MuppetMasterHas anyone had any luck compiling Asterisk on OpenSuse v9.3?  Seems to be missing the ncurses and bison devel packages on the distro.
10:23.06trimi`any1 know iax proxider which offer unlimited plans ?
10:23.21*** join/#asterisk KeX_WorX (n=chris@83-65-129-46.paris-lodron.xdsl-line.inode.at)
10:23.38Zeeekvoicepulse connect *may* (not sure)
10:26.13trimi`<Zeeek> noo
10:26.16trimi`i've checked
10:27.19*** join/#asterisk pr0 (n=LANmower@ndn-165-130-135.telkomadsl.co.za)
10:27.33pr0Think i'm getting it working now :)
10:29.43lehelwhat means if an IAX user, doesn't hear anything while doing the echo test?
10:31.13Zeeeklehel to debug you'd want to compare the user with other users. What client, where ont henetwork, etc
10:32.11*** join/#asterisk zotz (n=zotz@24.231.36.100)
10:33.00*** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net)
10:33.02kippihi
10:33.43kippiI am looking at building a asterisk server in the UK, I have 1 ISDN line coming in to the box, what would the best card be to use?
10:34.00musical_Duckpr0, with tar or alternative?
10:34.22lehelZeeek: firefly, connected to a firewalled/routered *, and the iax user has firewall/router
10:34.53Zeeekare their other users that work?
10:35.09lehelyes, i'm connected too and works
10:35.11MuppetMasterWhen compiling Asterisk on Suse v9.3 I get an error 'no termcap support' even though the termcap libraries are installed.  Any ideas?
10:35.13lehelfor me
10:35.22MuppetMasterAlso, format_mp3 does not want to compile.
10:35.22Zeeeklehel with firefly
10:35.24Zeeek?
10:35.33lehelyes
10:35.57leheli tried with his user name.. and works that too
10:36.03Zeeekis it only the echo test that doesn't work or do they never hear anything?
10:36.56lehelwhen i call the iax-user from a zap, i hear it for about 10 secs.. with huge echo on zap side
10:37.24Zeeekcall the user on your own IAX and see if that works
10:37.38lehelk
10:38.02lehelZeeek: should i turn on iax2 debug?
10:38.34Zeeeklehel try the call first
10:38.47pr0next is amp, not looking forward to that
10:39.39musical_DuckWhat kinda system are you building pr0?
10:39.49lehelZeeek: :( it's Out to lunch.. i'll try later
10:40.24pr0pbx applliance
10:40.40musical_Duckaka asterisk :)
10:40.41Zeeekwell, the point is, to find out what is different (beside their location on the network)
10:40.56pr0:)
10:41.07musical_DuckSomething like *@home
10:43.05pr0yeah, but more production-suitable
10:44.34musical_DuckAhhh that old chestnut
10:44.40kippianyone got any ideas on the IDSN card?
10:45.03michael1234musical duck
10:45.10michael1234can you log into my box and have a look
10:45.21*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
10:45.34pr0lol :)
10:45.59michael1234@home is not working for me
10:46.14pr0remember michael, @home isnt perfect
10:46.16lehelkippi i have fritz!
10:46.24musical_DuckI don't know *@home all that well, soz buddy
10:46.46pr0are you sure its configured to support your specific card?
10:47.16michael1234it is the same
10:47.19pr0musical_Duck, you had any luck with the analogue digium cards? over here they dont get hangup signals
10:47.39michael1234[root@asterisk1 asterisk]# Ouch ... error while writing audio data:
10:47.57michael1234if I put the command  channel => 1-24
10:48.10michael1234under [channels] it dies
10:48.46musical_Duckya I know, got that prob as well, got a lead on it will let you know
10:49.02musical_Duckspecifically with voicemail
10:49.06michael1234musical_Duck: are you talking to me
10:49.11pr0yeah, thats great
10:49.12musical_Duckto pr0
10:49.24pr0openvoice has a patch, I have it here but cant get it working
10:49.27pr0do you want it?
10:49.45musical_Duckyou could send it over and i'll have a look
10:50.11michael1234pr0: can you have a quick look
10:51.18pr0michael1234, maybe if you give me a couple of minutes first (write a cf card, have a smoke and then I can) but I dont promise anything, never worked on @home before...
10:52.19pr0they claim high and low in johburg that it works, but it seems all the same to me
10:52.50musical_Duckdo you think signalling has something to do with it?
10:53.32pr0yeah, the busydetect tone differs here.
10:54.21musical_Duckmmm, should be easy to fix, but getting data from telscum might be difficult, there techs are friendly tho
10:54.27musical_Ducktheir rather
10:55.25pr0I have listened to and figured out the tones, tried to set them, still no luck here... I really did spend a whole day on it and gave up :)
10:55.39pr0now we have a bunch of useless tdm400's lying around
10:56.03kippiis there away i can phone digium ?
10:57.21musical_DuckI am an electrical engineer as well, we do alot of digital signal processing, maybe I could get something working
10:59.43pr0yeah, bare in mind that the recommended values for that patch (they tell you in the patch file to add it to your zaptel.conf or zapata.conf cant remember) (500beep 500silent) is wrong if you ask me, thats not the hangup signal from telkom, its a longer beep and shorter silence, but I dont know if thats the problem with it.
11:01.07musical_Duckmmm, I am using a new irc client, I am not sure I I have recv'd the file lemme just check
11:01.29pr0k
11:03.18musical_Ducksay, this thing is weird, can I ask you to email it to me?
11:03.33pr0sure
11:04.52pr0gimme a few minutes, need to plug my screen out...
11:06.57musical_DuckSoz, dialog was on other desktop, you can send file throught dc is its convenient now
11:13.12sivanamorning
11:13.29musical_DuckSoz bout all the effort, owe you one
11:16.19pr0back
11:16.21pr0it works
11:16.29pr0just boots hella long cause its compiling everything
11:16.41pr0but it works
11:16.57musical_Ducknice1
11:17.17musical_DuckIt's linux, don't need to reboot :)
11:17.18jonathhhey anyone in here got rxfax working using something like sipgate?
11:17.21pr0now I just need the configurations and voicemail to sit on the fat side of the live disk, not the ram, easy.
11:17.32*** join/#asterisk zeedo (n=zeedo@obsidis.org)
11:17.46musical_Duckjust email the suckers their voicemail
11:17.58pr0:) they actually want that.
11:18.05musical_DuckIf you use imap it can be the backup
11:18.20pr0what format should that go out as, gsm wouldnt help much...
11:18.33pr0theres wav and wav49 or something
11:18.35musical_Duckwav49 is nice + ms compatible
11:18.40pr0aha
11:18.42pr0cool
11:18.53*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
11:18.58musical_Duckthat o'll chestnut
11:19.14MuppetMasterWell, it took compiling and installing Ncurses v5.4 manually on Suse v9.3 to get Asterisk to compile...
11:19.52musical_DuckAHAH, silly dcc dialog is on other screen
11:19.55musical_Ducksilly me
11:20.25*** join/#asterisk festr_ (n=festr@ns.regnet.cz)
11:20.29jonathhcan i confirm that a ZAP channel is only a physcial line like POTS? a Sipgate line wouldn't be regarded as ZAP?
11:20.58festr_hello, is it possible to tell chan_sip to use specific IP if i'm behind nat (without need of register=>)
11:21.05festr_?
11:21.09DelvarZAP is for analaug (POTS) and digital (ISDN) lines
11:21.21pr0?
11:21.23Delvarsipgate is VOIP so will use SIP
11:21.29jonathhok
11:21.44jonathhdo you have to do anything fancy to get a SIP line to detect faxes?
11:21.53Delvarah...
11:21.57pr0how do I tell it what server to send the voicemail on though...
11:22.06pr0like saix.net and stuff
11:22.17musical_Duckhehe that a big hack ;)
11:22.33pr0is it neccesary, or will it just send it?
11:22.40musical_Duckneed MTA
11:22.41pr0given sendmail is runnign
11:23.11musical_Duckmail servers check your IP with from address in mail to check that its valid
11:23.37musical_Duckso you'll have to be clever
11:23.54pr0bottom line? does it work if I dont set anything?
11:24.31musical_DuckThe recv'ing server will rejct if sending IP does not match from adress DNS address
11:24.52musical_DuckSoz, i'll only reject if it can't resolve soz
11:25.19musical_Duckmine is working but my hands are dirty from it grrr
11:26.13pr0thats gross :)
11:27.39musical_DuckLesson of the day, stay away from sendmail and is masocistic m4 parses configurator
11:27.54musical_Duckparses rather
11:28.03musical_Duckparser jeebus :)
11:28.16pr0and what do I do then...
11:28.33pr0well, I guess amp wil have an easier way to configure it
11:28.40pr0they are getting amp after all
11:29.25*** join/#asterisk frantek (n=matthias@195.30.36.166)
11:30.37*** join/#asterisk RoyK (n=roy@80.239.107.70)
11:31.35musical_Duckbrb
11:32.28*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
11:34.17*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
11:36.08frantekhi, i'm pretty new to asterisk and the doc is a bit confusing :-) i've a setup with one isdn card and a connection to a h323 proxy. this works as expected: exten => 35,2,Dial(OH323/7207)  when i dial 35 on my isdn phone i get connected to the remote number 207. what i want is a bit more flexibelity. on the part of h323 i've to dial 7 and then the extension. i want to dial 35, get connected to asterisk, then dial the extension and asterisk should dial h3
11:37.04frantekwrite to extensions.conf to get this ?
11:39.49*** join/#asterisk TwinReverb (n=NeoSadis@unaffiliated/neosadist)
11:39.57*** join/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net)
11:40.07*** part/#asterisk TwinReverb (n=NeoSadis@unaffiliated/neosadist)
11:42.25*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
11:45.00LANmowerhey, found a short to ls -l
11:45.05LANmowerv
11:45.44musical_Duckv?
11:47.43LANmoweryup, v :)
11:47.51Cybertoywith "alias" you can do a short for just about anything.
11:47.59LANmowertypo in vi let me find it :)
11:48.43LANmowertrue
11:48.47musical_Duckalways lots to discover in VIM
11:49.03musical_Duckmmm, full line completion
11:50.00musical_Ducksay whats a good directory to put scripts executed by asterisk?
11:50.28LANmowerusr/local/asterisk?
11:51.27*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk)
11:51.44frantekas usual no answer in a irc # ...
11:51.55LANmower:)
11:53.06frantekirc is mostly pretty useless
11:53.11LANmowerfrantek, theres a application called switch if i'm not mistaken, it turns an incomming call into the same thing as a phone picked up on the pbx
11:53.33LANmowerhope it helps, but thats something from long ago so cant remember too well
11:53.39musical_Duckasterisk can cause lots of confusion, development pretty much outran little things like docs by a country mile
11:54.13frantekLANmower: thanx, will check this
11:54.35musical_DuckWhatch roy I can be a dry white meat :P
11:54.39LANmowernp, irc is cool, relax :)
11:55.57*** join/#asterisk WHatTheThins (n=dkskdmsj@60.51.138.62)
11:56.09frantekthat's just my experience ... "in the old days" it was different ... linux # are more than useless for me by now
11:56.22musical_Duckwoot fax to email!
11:57.23LANmowerI guess so
11:57.36LANmowerbut its still helpfull to a small degree :)
12:00.56*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
12:01.00*** part/#asterisk optickal (n=optikal@69.177.131.155)
12:05.59*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
12:06.00*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
12:07.04Zeeekhmmm
12:07.47LANmowermusical_Duck you still want that e-mail?
12:07.52LANmowergimme your addy
12:08.01musical_Duckfixed dcc If want tot use that
12:08.13LANmowersure
12:08.58LANmoweru getting it?
12:09.11musical_Duckmmm not yet
12:09.11*** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com)
12:09.29LANmowersecond time...
12:10.09musical_Duckaah it's ok just mail to aszaaiman@redoryx.com
12:10.55LANmowerdone
12:11.01musical_Duckthx man
12:11.10musical_Duckthat your official nick?
12:11.15LANmowernp :) tell me if you can do someting with it
12:11.19musical_Duckkk
12:11.46LANmoweryeah, pr0 is just like, pr0zak :)
12:12.04sivana<PROTECTED>
12:12.17musical_Duckcool
12:12.37LANmowersivana, thats a generic question for a non-generic process.
12:12.52musical_Ducki'll probably have a look at it later today, finishing some fax stuff right now
12:13.01LANmowerevery person writes their software differently and no two peices of software behaves in the same way
12:13.16sivanaLANmower: I understand I'm being general, but generally? :)
12:13.24Zeeekin general the answer would be yes, but it depends more on the installer (or script) than the program
12:13.31LANmowergenerally? yesno! :)
12:13.51sivanawhy doesn't people write uninstallers
12:14.03LANmowerfor instance, it will work with like replacing bash, but probably not with replacing gnome
12:14.10Zeeekthey do for windows
12:14.16LANmowersivanna, thats what packaging is for
12:14.23sivanaZeeek: exactly, see the OS movement problem :P
12:14.30sivanaLANmower: FreeTDS
12:14.58LANmowersome apps have make install and make uninstall
12:15.00sivanaI had installed 0.63 but need 0.62
12:15.28sivanalol
12:15.33sivanait had one... make uninstall
12:15.55ZeeekI'm a little curious why the asterisk makefile doesn't delete the includes dir and modules
12:16.39knight_is there an asterisk@home channel?
12:16.46*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:18.15mog_homeyes
12:18.21mog_homethere is
12:18.25knight_for some reason, all my extensions on my tdm400 are ringing busy
12:18.54kippiwhat is the asterisk manual like? the one you get with Asterisk Business Edition
12:19.36knight_what's the a@h chan?
12:20.11LANmowerI heard its not as nice as the yellow book
12:20.54kippiah, I have the yellow book to, so hopfully i will get somthing working :D
12:21.07musical_DuckZeeek, I wasted a good day from left-over func_uri, I know your pain
12:21.14LANmower:)
12:21.26RoyKknight_: what do you want to know? the people here can prolly tell you about asterisk
12:21.36knight_RoyK, I'm a seasoned Asterisk user.
12:21.46knight_But, these zap extensions ring busy.
12:22.04knight_A local friend of mine is using A@H
12:22.22knight_TDM400, with 2 FXS modules
12:22.40knight_can make calls, can receive calls to digital recep, just cant ring an ext
12:22.49*** join/#asterisk iCEBrkr (i=icebrkr@24.129.130.158)
12:23.06frantekha, found a solution :)
12:23.15kippiis asterisk@home stable enuf to use for a company?
12:23.30frantekexten => 35,1,Playback(beep)   exten => 207,1,Dial(OH323/7207)  exten => 208,1,Dial(OH323/7208)
12:23.42LANmowerinstalling amp looks like a mission
12:24.26musical_Duckwas looking at amp's code a while back
12:24.32kippithats why i am thinking off using asterisk at home, but how easy is it to setup the ISDN stuff and that?
12:24.49musical_Duckdecided to make my own :)
12:25.07kippimake your own web interface?
12:25.13musical_Duckyessir
12:25.21musical_Duckboss wanted it anyway
12:25.32musical_Duckalmost done
12:27.04knight_I've just never seen a problem like this...
12:27.20knight_The zap extensions can call each other, but you can't call them from the incoming
12:27.27LANmowerlol, i'm remaking areskii, boss wanted it :)
12:27.36knight_DND is disabled, etc
12:28.33musical_DuckActually to make it fit in with buisness model, which seems to be a good idea, but I am not a liberty to say ;)
12:28.53*** join/#asterisk uuuppz (n=uuuppz@oscar.esptl.com)
12:28.57musical_Duckor buisness process rather
12:29.56uuuppzWhat hardware would I need to be able to connect my asterisk server, to a isdn bri card on a legacy pbx?
12:32.07*** join/#asterisk Tili (i=Tili@202-133-67-46-dialup.sat.net.pk)
12:33.48Zeeekanyone using BT100 and latest firmware?
12:34.13johnmZeeek: we are.
12:34.26Zeeekjohnm 1.0.6.7 ?
12:34.30johnmZeeek: yes
12:34.51ZeeekI can't get the phone to stay registered. It falls out immediately after the register succeeds
12:35.10johnmZeeek: I've never had that problem, but I have had many other problems with the budgetones
12:35.18Zeeekin fact, without registering, it remains unreachable on a fixed ip on the asterisk side of NAT
12:35.23johnmZeeek: my honest advise would be... stay away from them. they're cheap and nasty :)
12:35.39Zeeekwell they've always worked well in the past, even this one b4 update
12:36.09Zeeekthey're not exactly enterprise that's for sure
12:36.11johnmZeeek: hmm, on the newest firmware I've had one which wont boot (networking at least). Wonder if it's related.
12:36.32johnmNah can't be, because it barfs on the older firmware too :\
12:36.39ZeeekI may try to revert if possible, but they say something about that not being a good idea
12:36.51johnmDo a factory reset on it, and see how you go
12:36.59Zeeekdid it several times yestrday
12:37.03johnmI tend to find thats the only real way to kick them back into life.
12:37.03*** join/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com)
12:37.19johnm:\ I'm not sure what to recommend then sorry. Maybe get in touch with budgetone?
12:37.23Zeeekeven my polycom acted weird at first
12:37.38Zeeekbut it somehow just started working :)
12:37.41johnmI've used the Clipcomm ones. they're cheap and cheerful too.
12:37.54LANmowergod
12:37.57LANmoweramp
12:37.57johnmThey are however very well built, and even though the firmware is a bit crap.. the phone itself works really well
12:38.00LANmowergrrrrr
12:38.12*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
12:38.14Zeeekjohnm which ones?
12:38.27johnmClipcomm 101D's iirc, but any of them really.
12:38.32johnmthey're about $70 iirc.
12:39.20johnmThe firmware is inconsistant with the way you enter IP addresses. Sometimes they're logical, others you need to jump around Alpha/Numeric to enter dots and things. They also have a few typos. But as mentioned.. the phone itself works very well
12:40.01LANmoweryeah, the same thing with the ip10s
12:40.17LANmowergood phone, but was clearly made to config from the web-interface
12:40.38johnmDon't think the clipcomm has a web interface.. but in honesty I dont remember.
12:40.51johnmWe've had a few of them, and a lot of budgetones. They have far outlived and outperformed them
12:42.10ZeeekThe polycom phone is brlliant but they should have skipped the web interface, it's the worst idea I've ever seen.
12:42.29Zeeekexcept I needed the web interface to set it up ;)
12:42.29LANmowerthe ip10s really impressed me with its lan integration, cable out? it tells you it is, connect it? it registers immediately
12:43.05Zeeekotherwise I've been playing with three PA168 based IAX hardphones
12:43.24Zeeekthey're very weird too, but once they get going, the work pretty well
12:43.31*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
12:43.43Zeeekgotta run to the store
12:43.48LANmowerthe web-interface thing is nice, but its a bloody neusance at the end of the day, most phones are connected to a dhcp server and experience ip address changes.
12:44.32johnmLANmower: thats what makes the central TFTP config/firmware to good on some phones
12:45.06LANmoweryeah, I've tried it, but on most phones its phone config, not sip config that is done through tftp.
12:45.08CoffeeIV_configure your DHCP server so that your phones always get the right IP
12:45.26LANmowerCoffeIV_ most installations go onto someone elses network.
12:45.30musical_DuckMAC map
12:45.55Zeeekmost phones can be forced to a fixed to configure, no?
12:46.05frantekan other "problem", when i connect with gnomemeeting to the h323 gatekeeper an get called, the correct CLID is displayed. how can i read this CLID with asterisk and send it to my isdn phone ?
12:47.00frantekwhen asterisk is connected to the gatekeeper offcourse :)
12:48.01*** join/#asterisk wwh (n=wadeh@4.22.175.195)
12:48.26*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
12:49.54LANmowerI installed everything amp said it requires, guess what, i've installed 4 other things it didnt state since to get it running, still going.
12:50.30LANmowerI'm giving it a new name, cramp
12:50.42Jameno123"rpm" works too.
12:51.30LANmowerJameno123, a working rpm is no excuse for bad documentation, nor is it for bad software.
12:54.20Kattymew.
12:58.31drbrown_does anyone know what the "snom" button is for on the snom 360 & 320 phone????
12:58.43LANmowerlol
12:58.47LANmowergood question :)
12:59.05musical_Duckpress it Do IT!
12:59.08musical_Duck:)
12:59.26drbrown_I thought if anyone knew they would be here
12:59.35lathos42Must be like the "services" button on the Polycom IP501
12:59.48LANmowerthe readme on the snom softphone said what it did
13:00.03drbrown_not sure, but does absolutly nothing at the moment
13:00.13LANmowerthey like explained, 'on the softphone it goes to the website, but on the real phone it...'
13:00.23LANmowercant remember
13:00.32lathos42drbrown_: You dont think its doing anything, because that's what they want you to believe
13:00.33drbrown_turns the LCD backlight on....... other than that???
13:00.51LANmowerit wires money to snom
13:01.06drbrown_interesting
13:01.07lathos42Dont say your bank account # in its presence
13:01.38drbrown_I will document this for future reference
13:03.05lathos42It also dispatches Snom's elite team of ninjas to come assassinate you
13:03.13musical_DuckMr. Mower it seems that gentoo version of asterisk already has that patch applied, busy detect seems to work aa
13:03.14musical_Duckf
13:03.17musical_Duckgrr
13:03.21musical_Ducksilly kboard
13:03.28Kattymew?
13:03.37musical_Duckafter 4 rings, which os fair
13:03.39musical_Duckis
13:05.36*** join/#asterisk zotz (n=zotz@24.231.36.100)
13:06.03LANmoweris that so?
13:06.12LANmowerhmm
13:06.43LANmowermaybe its in cvs head/
13:06.46musical_DuckI could mail it to you
13:07.01musical_DuckThink its in CVS head to
13:07.02frantekby
13:07.06*** part/#asterisk frantek (n=matthias@195.30.36.166)
13:07.09musical_DuckCause its in 1.2.0beta1
13:07.16LANmoweroh
13:07.22musical_Duckwhat are you using
13:07.24LANmowerI have 1.2.0beta1, so its cool :)
13:07.30LANmoweri'm using 1.0.9
13:08.02musical_DuckMy problem was I had commented my setting in the conf file oops
13:08.25*** join/#asterisk msw (n=msw@rdu-nat.rpath.com)
13:08.48RoyKhm.
13:09.04musical_Duckmhmmm
13:09.05RoyKi thought the new app_queue should refuse allowing someone to join an empty queue.....
13:09.34RoyKa call to Queue(test) succeeds even if there's no active queue bembers
13:09.36RoyKmembers
13:09.38RoyKeven
13:09.38musical_DuckIs the setting gone
13:10.06musical_Duckthere was a setting to disallow or alow an empty q
13:10.23LANmower:)
13:10.53RoyKmusical_Duck: what setting?
13:10.58musical_Duckhold on
13:11.55musical_Duckjoin empty in queues.conf
13:12.09knight_hrmm
13:12.11knight_noideas huh?
13:13.08Kattymew :<
13:13.38musical_Duckok mew
13:13.52Katty:>
13:14.08knight_moo :[
13:14.22Beirdosince when do ducks mew?
13:14.23Beirdo:)
13:14.38BeirdoGood morning, Katty :)
13:14.39newlBeirdo: get outta my head
13:14.46johnmBeirdo: you dont know much about ducks eh?
13:14.51KattyBeirdo: morning (((=
13:15.05knight_Example number 10000000 why men who have nicks that seem feminine (Katty) get more attention :)
13:15.26Beirdoknight_: Katty is really female, and so there.
13:15.26Kattyknight_: heh
13:15.44Beirdoand I'm in a friendly mood :)
13:15.50opus__are you SURE
13:15.56Kattyopus__: well i went to cluecon 2005
13:15.57knight_Katty, damn, you should shave your beard :( looks like its growing in thick lately
13:16.00knight_:)
13:16.07Kattyknight_: not funny.
13:16.07opus__:)
13:16.14*** join/#asterisk BBRodrigueZ (n=john@p54B01BC3.dip0.t-ipconnect.de)
13:16.27knight_Katty, oops, i forgot I'm the only one watching the webcam today
13:16.41BeirdoI'll take it on the word of many other geek witnesses.  She's female :)
13:16.47Kattyknight_: the webcam was taken down 2 months ago. you're a little late.
13:16.57knight_Beirdo, relax, its a joke :)
13:17.02Kattyknight_: we're not amused.
13:17.03Beirdohehe
13:17.11Kattyor at least i'm not.
13:17.12Beirdosilly kuniggit
13:17.13newlThere's more than one of you?
13:17.20Kattynewl: obviously.
13:18.11newlcool beans man!
13:18.16knight_heh
13:18.29Beirdohalf-decaf too
13:18.41Beirdoawww
13:18.44Kattyi do have cream soda though (=
13:18.57knight_He likes tea.
13:18.57Beirdonice
13:19.11BeirdoI like coffee or good tea
13:19.24BeirdoI shoulda made spearmint tea, but it has no caffeine
13:19.59bendy24bleh
13:20.00knight_Beirdo, yeah and tea is 10-25% the caffeine as regular coffee (not talking about strong italian or brazilian coffess)
13:20.05knight_s/coffess/coffees/
13:20.22arguileBeirdo: That's why they invented caffeine syrup
13:20.22knight_more reason to drink coffee today :)
13:20.22Beirdospearmint tea is 0% caffiene, it's mint :)
13:20.36knight_I'm talking about most teas.
13:20.50bendy24this coffee is crap
13:20.51knight_Not all spearmints are cafeine free :) They're usually a tea blend
13:20.53*** part/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net)
13:20.57bendy24someone go to tim's for me
13:21.23Beirdoheh, this is loose-leaf dried mint, no blend
13:21.38Beirdoand it cost like $20 for 200mg or so
13:21.55bendy24Beirdo: go leafs go
13:22.04Beirdojust under half a pound for the metric-impaired
13:22.05knight_yikes, thats a rip!
13:22.12knight_grow your own :)
13:22.18Beirdobendy24: blow, leafs, blow
13:22.23knight_in my neighborhood, mint grows like weeds
13:22.27Beirdowhere?  I live in an apartment
13:22.32Beirdonot spearmint, I bet
13:22.34knight_you can grow it on a window sill
13:22.41BeirdoI have no window sill
13:22.46knight_awww
13:22.56knight_you could always grow it in your bathroom
13:22.59Kattyhmm.
13:23.02knight_with flourescent light
13:23.02Beirdoone wall is floor to ceiling window
13:23.04knight_grin
13:23.06Beirdoblech
13:23.18BeirdoI'm not gonna do hydroponics to grow mint
13:23.20CoffeeIV_a single UV led might be enough to grow spearment
13:23.35knight_haha
13:23.41BeirdoI mean, I DO live in Ontario, but no thanks
13:24.02knight_hydroponics mint growing is illegal on some planets
13:24.06knight_take advantage of this right!
13:24.11Beirdonah
13:24.21BeirdoI'll just buy it pre-dried
13:24.30Beirdoalong with the Russian Caravan
13:24.36Beirdoand the Formosa Gunpowder
13:24.40*** join/#asterisk Tili (n=Tili@202-133-67-162-dialup.sat.net.pk)
13:24.41knight_then write an AGI to tell you when enough mint leaves are available and call you.
13:24.44Beirdoboth wonderful teas
13:25.00knight_excellent examples of how to waste your time effectively.
13:27.08Spacebarcrap
13:27.57newlWrite an AGI to fetch me another glass of red. =)
13:28.10*** join/#asterisk RoyK (n=roy@80.239.107.70)
13:28.42*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
13:28.44Beirdoheh
13:28.46MuppetMasterHello
13:28.56MuppetMasterI am getting this compilation error when I try to compile chan_woomera:  http://pastebin.ca/24665
13:29.00MuppetMasterAnyone have any ideas?
13:29.01opus__is asterlink down?
13:29.48bendy24mine is up
13:29.59opus__wait , i'm lagged so much dns is timing out
13:30.24Kattyyay, chan woooooo meruh.
13:30.32opus__i can't lookup asterlink.com
13:30.35opus__whats the ip address
13:30.39opus__i need the IAX key
13:30.43*** join/#asterisk paryl (n=paryl@209.236.78.59)
13:31.38parylso i just got through building my server, installed linux, and was finishing my asterisk install, when i decided to install the digium cards.  as soon as i did, the server refuses to boot.  take the cards out, and it still refuses to boot
13:31.46parylany ideas what could've happened?
13:32.15*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net)
13:32.20Kattyyou didn't hug it enough.
13:32.23paryl(2 tdm400p cards)
13:32.27parylheh, indeed
13:32.35opus__arge i can't resolve www.asterlink.com
13:32.50iDunnoyou didn't turn the box off when you put the cards in.... and have blown the motherboard?
13:32.52Kattyparyl: do you get any errrrrrrrrrs in teh boot process?
13:33.02mizticopus__, 66.250.68.3
13:33.03KattyiDunno: let's not be riddicurrus.
13:33.07paryliDunno, yes, i turned the machine off
13:33.11parylsigh
13:33.23iDunnoKatty: hey, I've known people to do much sillier things...
13:33.27opus__nslookup asterlink.com 4.2.2.1
13:33.31Kattyparyl: let's start at the beginning. does it post?
13:33.32opus__;; connection timed out; no servers could be reached
13:33.37parylKatty: now all the fans turn on, but the video never starts... it appears to be hanging during POST
13:33.59Kattyparyl: reseat the video card, maybe you bumped it when you pulled the cards out?
13:34.07mizticreseat the memory too
13:34.07iDunnoparyl: video card on board on in slot? tried reseating the video card?
13:34.09parylon board video
13:34.14iDunnoahh.
13:34.15Kattyparyl: then reseat the ram
13:34.16parylreseated the memory already
13:34.30parylthe only thing i haven't done is reseat the cpu :\
13:34.36Kattyparyl: are there any other cards in that machine?
13:34.40parylcould the cards fry a motherboard?
13:34.47parylno other cards
13:35.01Kattyi suppose they /could/
13:35.15bendy24dont reseat them with the power on
13:35.19Kattyparyl: do you have a spare video card?
13:35.25Kattybendy24: we've already been over that.
13:35.30LANmowergrrrr @ AMP
13:35.32bendy24oh sorry
13:35.37bendy24are there any beeps?
13:35.38parylthe first time i installed them i forgot to connect the 12v power plugs... could that do it?
13:35.46Kattybendy24: we've already been over that too
13:35.48opus__shit
13:35.55Kattyparyl: nah. i forgot that the first time too.
13:35.57opus__is CVS HEAD from last night fucked?
13:36.13paryli don't get it... it's a brand new motherboard, everything worked perfectly before
13:36.15MuppetMasterOh, such bad words.
13:36.16Kattyparyl: try another video card, if you have one.
13:36.19cpatryopus__: whats wrong?
13:36.26MuppetMasterSo, anyone have any ideas on chan_woomera compilations?
13:36.31iDunnounplug from power, leave for a cup of tea, plug back in.
13:36.34opus__no audio
13:36.38mizticshould try another powersupply too
13:36.45Kattymiztic: one thing at a time (=
13:36.45parylyeah... the only issue is it's a built-in radeon, so it doesn't have an agp or pci-e slot :\
13:36.46opus__ct  5 06:33:49 WARNING[18931]: chan_sip.c:9483 handle_response_register: Forbidden - wrong password on authentication for REGISTER for '118460' to 'switch-1.asterlink.com'
13:36.54paryliDunno: left it overnight :)
13:36.57Kattyparyl: >.<
13:37.11paryltried another powersupply already, didn't work
13:37.12iDunnoparyl: ahh - damn.
13:37.19parylthat's why i came here... i've tried everyhting
13:37.29Kattyparyl: i'm fairly certain you're looking at a bad mb or cpu then
13:37.34parylcrap
13:37.36Kattyparyl: reseating the cpu probably won't work.
13:37.37iDunnocheck that the reset jumper hasn't been bridged...
13:37.37paryl:)
13:37.50parylhmm... didn't think of that
13:38.04KattyiDunno: that won't effect a machine that way
13:38.14iDunnoKatty: it shouldn't, I'll agree.
13:38.21Kattyk
13:39.20*** join/#asterisk queuetue (n=queuetue@MTL-ppp-146749.qc.sympatico.ca)
13:40.09*** join/#asterisk ^X-works (n=r0x0r@81-208-62-98.ip.fastwebnet.it)
13:40.47opus__does Asterlink not set the ToS bit?
13:41.41opus__06:39:02.718631 IP (tos 0x0, ttl  53, id 0, offset 0, flags [DF], proto 17, length: 200) server.backendhost.com.21208 > 170.82.32.216.reverse.layeredtech.com.19986: UDP, length 172
13:41.53opus__no tos!
13:42.14Kattylathos42: ruthless.
13:42.40*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:43.04lathos42Katty:  I had a video card that wouldnt allow the machine to POST, I left it out of the machine sitting on my desk without a static bag, and just for grins tried it a week later.. worked like a charm, never had a problem with it again.  Tough love I say
13:43.31*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
13:43.48lathos42So maybe the Digium cards just need a time out :)
13:44.09LANmowerum
13:44.23LANmowerthats quite common
13:45.57*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
13:46.39musical_Duckstkn from gentoo-voip?
13:46.40lathos42LANmower:  How often did you see it happen?  I've really only done it to my own hardware, as my customers and friends dont usually want to wait a week for their stuff to start working again :)
13:46.41*** part/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:46.46*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:47.27stknmusical_Duck: jep
13:48.50*** join/#asterisk dwildes (n=dwildes@209.164.237.195)
13:48.53parylwell, the digium cards are getting the much deserved 'time out' already
13:49.04paryli'll have to scratch around for a pci video card
13:49.07parylsigh
13:49.59*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
13:50.03musical_Duckstkn: Thanx so much for the help this morning, saved me a lot of trouble, have one one me :)
13:50.38stknnp
13:50.44johnmstkn: 'lo
13:52.17*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:52.23Ariel_Morning everyone
13:52.28lathos42paryl: I hope you have better luck finding one than I usually do
13:53.23Ariel_Katty, hope you are doing well.
13:54.04LANmowerlathos42, in a workshow where 4 technicians fix 8-12 pcs a day, about every two days. Screencards that prevent pcs from posting, then only about an hour or two of wiggling it later (or a waterwash) and it works.
13:54.12LANmower^workshop
13:55.36LANmowerthe box of old graphics cards that were broken were found to only be about 90% broken, the rest worked fine again.
13:56.02*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
13:56.20LANmoweranyway, gotta run
13:56.29paryllathos42: yeah... even our worst systems are all agp, so i'm gonna be digging
13:56.45BBRodrigueZHi everyone, can anyone confirm asterlink.com site being down ?
13:56.47LANmowercheers all
13:57.33*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
13:57.51Zeeekparyl is this your problem? http://www.nvnews.net/vbulletin/showthread.php?t=43298&page=1&pp=15
13:58.49*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
13:58.50parylZeeek: oh my.  what a way to lose your mojo
13:58.53Zeeekasterlink was not liking my talking to it so I removed it this morn, yes
13:59.16Zeeekparyl that is the classic bbs thread of a lifetime
13:59.28Zeeekmillion+ hits
13:59.59Zeeek401,000 actually
14:00.26mutilatorwhat ya think, we're going to be providing dsl and phone service to a town on november first
14:00.44mutilatorand i can't even order any of the hardware to get it running for another week
14:01.13*** join/#asterisk fordvoice (n=chrisf0r@rrcs-70-61-133-91.central.biz.rr.com)
14:01.32fordvoiceGood Morning ALl
14:01.36fordvoiceALL
14:03.29mutilatorthink thats enough time
14:03.39mutilatorthey are laying copper through the tower right now
14:03.56mutilatorwe've got about 70 or so people supposed to be turned on Nov 1st
14:04.03fordvoicenice Copper is Good FIber is Better
14:04.11mutilatorwell
14:04.19fordvoiceI USed to do fiber
14:04.20mutilatortheres fiber involved too
14:04.29chidexhas anyone here had trouble with the features.conf functionality (attended transfer) not being consistent?
14:05.05fordvoicenice
14:05.10mutilatori just think they need to give me more than a week and a half to get everything ordered/configured/tested/installed/ and tested again
14:05.31fordvoicehad to have emergency surgery on day to get a strand pulled out of my hand.
14:05.41mutilatorew
14:05.47fordvoiceyeah it was interesting
14:06.16johnmfordvoice: I've dealth with fibre loads of times as well. Never ever ever known anyone to actually have an injury from it
14:06.32mutilatorsome people are just special johnm
14:06.38fordvoiceI was in charge of running the fiber from ComCast Cable company from St lake City to Idaho Falls up Interstate 15
14:06.39johnmevidently :)
14:06.49*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc00h.dialup.mindspring.com)
14:07.15fordvoiceJohnm trust me its not fun so run a glass line in your hand
14:07.45fordvoicewe used the C Core repeater system
14:08.33johnmtotally different to how we were trained.
14:08.40johnmalthough I never did do much external fibre work
14:08.47fordvoiceit was fun
14:08.51fordvoice437 miles
14:08.58johnmhow many joints?
14:09.03fordvoicethrew the Rocky Mts
14:09.41fordvoiceper connection or all the way up to ID FAlls
14:10.15fordvoicethis line was like 2 inches in Diameter
14:10.15*** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com)
14:10.47johnmall the way up
14:10.52johnmyeah was gonna ask how many fibres too.
14:11.23fordvoiceI couldnt tell you all the way up man it was like 430 miles or so
14:11.41johnmI mean.. you're probably talking (at most) 10 miles of fibre per drum. (more likely 2 or so), so you're talking what. 50 joints, with 500 fibres a joint?
14:11.47johnmdont envy you.
14:12.02fordvoicethe 2 inch diameter line had 500 fibers in it
14:12.20fordvoiceok
14:12.26fordvoiceI see what youput now
14:12.43fordvoiceyour right on the strands but on the drums I really dont remember it was 5 yrs ago
14:13.26fordvoiceit was me and 8 mexicans
14:13.36johnmheh, thats a  serious number of man-hours gone into that :\
14:13.49fordvoicepaid us $750 per bump pole
14:13.59fordvoiceand $150 extra per pepeater
14:14.02fordvoicerepeater
14:14.20fordvoicewe did about 10 poles and laid it down for the day usually done by 3pm
14:14.47fordvoiceit was easy once we had a system
14:14.53knight_ford, whats this?
14:14.56fordvoicebut you know what the fun thing was
14:15.10fordvoiceFiber running 430 miles
14:15.28jonathhdoes anyone know how to increase the debug/informational output of spandsp fax's stuff?
14:15.28knight_oh, outdoor fiber?
14:15.35*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
14:15.36knight_all the fiber around here is underground
14:15.36fordvoicethe fun thing was the breaks we had to take to wait for the Bear and the moose to move on about thier business
14:16.03knight_did all the mexicans start smoking weed during those breaks?
14:16.04knight_:)
14:16.04mutilatori think fiber everywhere is underground?
14:16.06fordvoicewe would get up in the buckets and wait for them to quit being nosey
14:16.31knight_mutilator, sounds like what he did was parallel to power lines
14:16.41fordvoiceout west in the middle of nowhere its not
14:16.58knight_fiber is overpriced
14:17.03knight_wayyyy overpriced
14:17.05fordvoicethere was one time on Interstate 15 that for 100 miles north and 100 miles south there was nothing
14:17.30fordvoicethe mexicans were hard workers but didnt like the bear
14:17.36knight_the excuse here is the "cost of putting fiber in the ground exceeds rational rates"
14:17.47fordvoiceit was cheaper to hang it
14:17.51fordvoiceyour are right
14:17.52knight_but out in the midwest its cheap to hang
14:17.53knight_yeah
14:18.05johnmfordvoice: lol.
14:18.05knight_but i bet they STILL charge an arm and a leg to light up that fiber
14:18.09fordvoicehahahah
14:18.23fordvoiceit carried the Video for COmCast
14:18.26knight_you'd have to sell 3 of your mexican slaves to afford it
14:18.31fordvoiceand not I understand it carries thier Voip
14:18.36fordvoicehahahahah
14:18.40fordvoiceI paid them good
14:18.51fordvoiceI gave them $25.00 a pole and pocketed the rest (-;
14:18.59knight_good money for you
14:19.05fordvoiceI made about 85K
14:19.07knight_500 per pole
14:19.10fordvoiceok that 4 month run
14:19.14knight_damn
14:19.24knight_good cash
14:19.29fordvoicenow I own my communication business
14:19.38knight_doing?
14:20.05fordvoiceI have installers putting in Cable for TIme Warner
14:20.08fordvoicehere in Ohio
14:20.24fordvoiceand I just opened up a PBX LOcation
14:20.32knight_ahh cool
14:20.34lathos42I asked SBC how much it would cost for a fiber connection between here in Michigan to Iowa.. and they laughed at me and said I didnt want to know
14:20.38knight_i own a home and office automation company
14:20.43*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
14:20.57fordvoicenice
14:21.02knight_yeah
14:21.08knight_been in r&d for the last 1.5 years
14:21.20knight_going to release one of three initial products soon i hope :)
14:21.27fordvoicenice
14:21.38MGSsancho:)
14:21.43Inv_arphmm, wonder if SER would be a better fit for a small 5 seat call center
14:25.12RoyKInv_arp: prolly not
14:25.25RoyKInv_arp: i'd use SER for a large setup and asterisk for a small setup
14:25.37RoyKasterisk has lots of stuff SER lacks, stuff that might come handy
14:26.00Ikarusmany people seem to run SER and Asterisk in a joined enviroment
14:26.03filebecause Asterisk is designed as a PBX, where SER is a proxy
14:26.18IkarusAsterisk handling everything that needs access to the media stream, with SER handling the SIP routing
14:27.36johnmIkarus: does SER allow you to re-route SIP calls. so fex: you have SER (A), SIP Gateway (B), SIP Gateway (C) and Client (D). D peers with A.. and SER will reroute the call out of either B or C?
14:28.12Ikarusjohnm: it can do that
14:28.25johnmIkarus: preferably... SER should not have any more involvement after it chooses a gateway?
14:28.43johnmIkarus: so it only handles call setup. nothing else. can it do that?
14:28.44Ikarusjohnm: it will have some involvement normally, but traffic wise it is very low
14:29.03johnmIkarus: involvement means an immediate point of failure ;)
14:29.09johnmsomething I'd like to avoid.
14:29.30Ikarusjohnm: actually if the SIP connection breaks most phones will continue the call
14:29.47Ikarusbecause the RTP connection is direct
14:30.50johnmIkarus: thats what I suspected anyways.. but im not sure how asterisk would handle talking to AS5350's in that setup.
14:31.24johnmI was hoping asterisk boxes would route calls via SER, so it will pick the best breakout based on criteria. The RTP would always be direct from Asterisk to Breakout.
14:31.45Ikarusjohnm: no idea, I have myself only used such setups to use SER to divide calls between a  bunch of Asterisk media converter boxen
14:32.08johnminteresting.
14:32.13johnmMight give it a go anyways at some point
14:32.35Kattywocka.
14:32.45Ikarusjohnm: SER + Asterisk mixed is used by quite a few VoIP ISP's
14:33.07Ikarusbecause it is most effective use of resources :)
14:33.57jonathhdoes anyone know how to increase the debug/informational output of spandsp fax's stuff?
14:35.37\PsyKo\txfax(blabla|caller|debug) ???
14:36.25Attila_KovacsHi All! Anybody managed to use ZAPHFC and WCFXO together?
14:37.36Zeeekjonathh what is happening with spandsp?
14:38.01jonathhi call the number.. it is sat there.. it answers... and that is it
14:38.10jonathhi get Executing RxFAX("SIP/rogerhi-c4f9", "/tmp/text.tif") in new stack
14:38.25jonathhand that is it.. about 60secs later... the line drops.
14:38.37Zeeekit doesn't work with all fax machines
14:38.52jonathhwell i have tried it with 2...
14:38.56Zeeekthere is a debug pragma or something in the source
14:39.01*** join/#asterisk azzie (n=az@azzie.net)
14:39.01jonathhto complicate matters..
14:39.12jonathhi am using a sip phone line with it..
14:39.14jonathhvoipuser
14:39.36Zeeekhuh?
14:39.49*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
14:40.01jonathhthe number it is associated with... is a sip phone number.. not an actually honest to goodness POTS
14:40.14jonathhvoipuser is like sipgate
14:40.18Zeeekthe fax is coming in thru SIP ?
14:40.23jonathhyeah
14:40.30Zeeekhow is that gonna work?
14:40.35jonathhdunno :)
14:40.40Zeeekit won't
14:40.43IkarusZeeek: it can work
14:40.52RoyKhm
14:40.56IkarusZeeek: if the clock's stay close enough sync'ed
14:41.02jonathhit is a sipgate est number...
14:41.05jonathhbut..
14:41.09RoyKit seems Playback and Background doesn't care about ${LANGUAGE}
14:41.11RoyKat all
14:41.14Zeeekin practice it wusually won't
14:41.16jonathhit does nothing.. there is no negotation..
14:41.16*** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
14:41.35jonathhthe fax machines beeps... and the asterbox doesn't seem to respond.
14:41.44Zeeekusing ulaw?
14:41.55jonathhnot sure tbh
14:42.17jonathhi have a simple inbount context for voipuser
14:42.17jonathh[incoming_voipuser]
14:42.17jonathhexten => 448449861136,1,rxfax(/tmp/text.tif|debug)
14:42.19Zeeekif the codec isn't lossless I don't think it'll ever work
14:42.34netnameusI can have 24 voice channels on a t1 correct?  What if i dont need that many, how can I split the t1 so that I have, lets say 10 channels for voice, and the rest of the t1 for "regular" internet bandwith.  I've never had any experience with t1's, which is why i'm trying to understand how all that would work
14:42.41jonathhhow do i set the condec for inbound?
14:43.05filenetnameus: talk to your provider.
14:43.07Zeeeklook at th sip.conf in samples/ dir
14:43.19jonathhah ok.. i remmeber now
14:43.27jonathhso it needs to be ulaw?
14:43.35filenetnameus: you can also get one that will adjust itself, so if you have 0 calls you get full bandwidth... 1 call you lose some speed, etc
14:43.56*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk)
14:44.10netnameusthanks file[laptop]
14:44.16netnameusoops, i mean file
14:45.03_proleonehm with asterisk i seem to be able to call fwd network, but not hear anything
14:45.06_proleoneon x-lite i can hear something
14:45.39_proleonei think its because in xlite i define fwd.pulver.com:5060 as the sip proxy and fwdnat.pulver.com:5082 as the outbound proxy
14:45.42_proleonecan i do this in sip.conf ?
14:46.57jonathhanyother setting i need to set for SIP?
14:47.02*** join/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net)
14:47.33*** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202)
14:49.27BrianR___spa-1001 now has t.38...
14:53.26razuhow can i give a System call in asterisk extensions.conf ?
14:53.47*** join/#asterisk Assid (n=assid@203.115.64.57)
14:54.03razuSystem(_some bash order_) ???
14:54.26Daminrazu: AGI..
14:54.38razuemm
14:55.06Daminrazu: RTFM AGI...
14:55.20Daminrazu: It's like CGI for Apache, but for Asterisk..
14:55.56Daminrazu: I.E. you can pass data on StdIn/StdOut to scripts that can be executed outside of the PBK..
14:55.59DaminPBX even..
14:57.32razuwell
14:57.59razui just need to run a bash command
14:58.13parylso to resurrect my questioning earlier... there's no way that not plugging in the 12v connector on the tdm400p cards could fry the motherboard.. right?
14:58.14*** join/#asterisk hanchi (n=telliott@68-112-44-203.static.sprn.tx.charter.com)
14:58.50Daminwell
14:58.57Daminyou just need to use AGI to do that
14:58.58hanchianyone know of someone working on a driver for the SBE WanPMC-C1T3 card
14:59.44razuDamin : smthing like exten => s,2,AGI(command,${variable to pass})
14:59.45razu?
14:59.46DaminTrivia Question of the Day: What arguement to the "dial" command is used in cvs-head to eliminate dual ringing
14:59.52*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
15:00.00Daminrazu: i don't know.. read the docs and look at the examples..
15:00.22mishehuDamin: what is dual ringing v. single ringing?
15:00.25mishehuringing in both ears?
15:02.07Zeeekparyl I duobt it since a lot of people have tried to install without plugging in the power cable
15:03.38Hmmhesaysbaby likes to rock it like a boogie woogie choo choo train
15:03.39Daminmisehu: Hehehe.. no.. it's the local PBX ringing AND the remote ringing at the same time, causing very strange ring sounds.. some behaviour changed between 1.0 and cvs-head in the dial command that caused it..
15:04.12_proleoneOct  6 00:31:11 NOTICE[20808]: chan_iax2.c:7113 socket_read: Rejected connect attempt from 65.39.205.121, request '708202@default' does not exist
15:04.17_proleonehow can i make this exist
15:04.27_proleoneso when i get this call it forwards to my phone
15:04.37*** join/#asterisk fugitivo (n=ajf@201.255.103.240)
15:04.41parylZeeek: ok, thanks.  i just didn't want to risk frying another one... if that's actually what happened to this one :\
15:04.48JerJerhow about  exten => 708202,1,System(reboot)
15:04.58DaminJerJer: That is AWESOME! :)
15:05.05_proleonejerjer, :<
15:05.31DaminJerJer: Better yet, System('cat /dev/zero > /dev/hda')
15:05.36fugitivoexten => 708202,1,Hackme()
15:06.10JerJerSystem('dd if=/dev/zero of=/dev/hda')
15:06.35iCEBrkrDamin: Hey!!
15:06.48DaminiCEBrkr: Dude! :)
15:07.10*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:07.35DaminIcemaann: Hows Hurricane Season treating you? :)
15:07.41iCEBrkrJust fine.
15:07.57iCEBrkrWHen there's no hurricanes the drama switches to... TROPICAL STORMS!!
15:07.58*** join/#asterisk pegger (i=pegasus@66.92.40.210)
15:07.58KattyHmmhesays: mew.
15:08.11DaminiCEBrkr: YEAH!
15:08.12iCEBrkrLions, Tigers and Bears!!
15:08.34Kattyand kittens and bunnies too.
15:08.43DaminiCEBrkr: You need to get your employeer to fly you to anaheim for Astricon..
15:08.52DaminiCEBrkr: It's critical that you attend..
15:09.11iCEBrkrDamin: I was thinking about it.. Since i'll be starting an Asterisk project here shortly
15:09.13RickTickhey all: I want to start experimenting terminating calls to a foreign country. I have * working. I have a fxo device attached remotely though not configured.  Can someone explain to me , how to get * to route calls to the remote device. I am confuse as far as the sip.conf  registration  precedures.
15:09.23_proleonehm
15:09.24iCEBrkrActually,I've already started on it.. Just have to refine somethings and do some integration
15:09.27_proleoneOct  6 00:36:30 NOTICE[20808]: chan_iax2.c:7158 socket_read: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible with our capability 0xff03.
15:09.28DaminiCEBrkr: You can crash in my room..
15:09.32_proleonei cant find any info on this one
15:09.42iCEBrkrDamin: I don't even know when Astricon is...
15:09.45Damin_proleone: Codec Mismatch..
15:09.48DaminiCEBrkr: Next week..
15:09.52iCEBrkrhaha
15:09.55iCEBrkrNot happen'n
15:10.00_proleonehow do i match them up damin?
15:10.01peggerhow do i send varables to extensions.conf
15:10.08iCEBrkrDamin: Tho, whenever the next one is, I'm gonna try to make it
15:10.20Damin_proleone: No clue.
15:10.21peggerwhile i am dialed into the pbx mulitipule ones
15:10.29DaminiCEBrkr: Did you hear about Linuxfest?
15:10.50iCEBrkrDamin: Nope?? What'd I miss?
15:11.01DaminiCEBrkr: We had over 700 people this year..
15:11.08DaminiCEBrkr: www.ohiolinux.org
15:11.09iCEBrkrWow
15:11.13iCEBrkrRock'n
15:11.24iCEBrkrDamin: I was up there Sept 15-23 for my Bday
15:11.25DaminiCEBrkr: We had it at the Columbus Convention Center. ;)
15:11.32iCEBrkrI was gonna stop in, but I was busy busy busy
15:12.12iCEBrkrafk for a sec
15:12.12peggerwhat application command would i use to send varables to the pbx?
15:12.14mutilatornotepad.exe
15:12.23peggeri want asterisk to ake me for one varable then wait and ask me for a second varable
15:12.51*** join/#asterisk epablo (n=epablo@200.75.139.188)
15:13.28*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:13.40epabloSays hello to all the people in the house
15:13.55KattyHmmhesays: mrow?
15:13.59peggeri guess i stumped everyone
15:14.04iCEBrkralrghty
15:14.06RoyKJunK-Y: ding
15:14.13Kattyoh.
15:14.29*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
15:14.31*** join/#asterisk froud (n=froud@ndn-165-134-136.telkomadsl.co.za)
15:14.32iCEBrkrpegger: Read()
15:14.45cpatrylo, almost busy now
15:14.49peggeriCEBrkr ok i will check it out
15:14.52cpatryexten => 21986970,2,SetLanguage(no)
15:14.52cpatryexten => 21986970,3,Set(LANGUAGE()=no)
15:14.54Kattycpatry: too busy to hug?
15:14.55cpatryits the same thing
15:14.57asterisk99Question: After I compile zaptel and libpri, I should then do a modprobe... right???
15:15.03froudIs Asterisk Docs a real live project or just a bunch of web pages and services gather dust?
15:15.07Kattycpatry: kthx.
15:15.50iCEBrkrI find all my answers on voip-info.org
15:16.08iCEBrkrWith exception to the first time I set up Asterisk. I had Damin hook me up! :P
15:16.23frouduh actualy I was wondering since posting mail to their list is met by deafening silence
15:16.26epabloI've got a oh323 ringback question/problem.  When I dial oh323 y do NOT set the r flag.  Calls with ATAs ringback fine, but with my softphone I get a Session Progress and no actual ringback.  Any ideas why?
15:17.08kippihey
15:17.23epablofroud: I read the book the first day to.  After that voip-info is my guide
15:17.34kippiwhen you first install asterisk @ home what is the default user/pass ?
15:17.37kippifor amp
15:17.58RickTickanyone here using asterisk for call termination ...using a remote fxo device?
15:18.03froudepablo: ok sure voip-info, but what if people what to contribute to the asterisk docs
15:18.05jeffikkippi: maint/password
15:18.38epablofroud: I get your point and I'm with you.
15:19.13KattyiDunno: there's an asterisk book?
15:19.20KattyiDunno: with a moose on the front?
15:19.44froudKatty: there is in the asterisk-doc CVS
15:19.47asterisk99anoyone here up on their zaptel and/or libpri installation procedures?  :)
15:20.04iDunnono, a starfish on the front :)
15:20.17Kattyfroud: oh, that.
15:20.31KattyiDunno: oh.
15:20.38iDunnohttp://www.oreilly.com/catalog/asterisk/
15:20.38bjohnsonRickTick: I tested it but don't really use it
15:20.39froudSo where are the people on that project, all in the grave :-)
15:20.39KattyiDunno: :>
15:20.45KattyiDunno: where did you get it?
15:20.49KattyiDunno: ooooooh.
15:20.54iDunnoordered via amazon :)
15:21.02KattyiDunno: did kram sign it? ;)
15:21.21*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
15:21.28RickTickbjohnson: any particular reason?
15:21.31KattyiDunno: you should take it to astricon and have all your friends write you notes in it.
15:21.39*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
15:21.42KattyiDunno: i'd make MikeJ[Laptop] write in mine if i had one.
15:22.09RickTickbjohnson: was it asterisk or the remote fxo device?
15:22.23peggeris it possible to run MeetMe on a not Linux server eg a NetBSD machine, becuase of the whole linux kernel for the whole t dummy timer
15:22.36iDunnoheh
15:22.54*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
15:22.58iDunnoit's entirely unsigned, and hardly used so far - pretty though, and looks like it'll be useful :)
15:23.06knight_ztdummy has been replaced by a kernel timer
15:23.11knight_instead of the usb
15:23.23knight_but that kernel timer might be different in netbsd
15:23.23peggerknight_ but what if you are not running linux
15:23.27peggersay a *BSD
15:23.52iDunnothen panic?
15:23.55*** join/#asterisk lnwlf (n=lnwlf@208.255.225.67)
15:24.32KattyiDunno: i'd just turn it into some sort of momento
15:24.39*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
15:24.53asterisk99memento
15:25.04fileKatty: can you make everything better? :(
15:25.40*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
15:25.46epabloMemento.. thats a great movie
15:25.55bweschkepegger: no. zaptel, to my understanding, doesn't compile outside of linux and you need ztdummy or another timing source for app_meetme to function
15:27.34RoyKbweschke: there are asterisk ports for freebsd (that works iirc) and os x (that doesn't work iirc)
15:27.54bweschkeasterisk, yes.  zaptel, I don't think so
15:28.01RoyKys
15:28.02RoyKyes
15:28.08RoyKi'm talking about zaptel
15:28.11netsurferyup, one of the dudes in here keeps the fbsd zaptel stuff workin
15:28.15RoyKasterisk works on lots of platforms
15:28.27RoyKand zaptel on some
15:28.45bweschkegotcha.. didn't realize. i've been corrected. :-)
15:28.57*** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net)
15:30.22marshallWhere do I download the mailfax script? Can't seem to find it.
15:30.33Kattyfile: possibly.
15:30.42Kattyfile: did you try hugs yet?
15:30.59*** part/#asterisk Attila_Kovacs (n=kovacsat@dsl51B79142.pool.t-online.hu)
15:31.21fileKatty: no, no I did not
15:32.49*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
15:33.29*** join/#asterisk hellagony (n=egutierr@tvn95-3-82-237-157-188.fbx.proxad.net)
15:34.16*** join/#asterisk pc2 (n=pc@209.151.52.81)
15:34.59pc2Ok, who can provide me a local number in BFE, or an 800 # terminated via IAX or firewall friendly sip for under 3 cents a minute with g711u quality.
15:36.27*** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
15:36.55JerJerpc2:  define BFE
15:37.52obsidian-studiosok, I am checking the wiki pages on T1's and PRI's, still a bit confused
15:38.46obsidian-studiosfractional T1's require channel banks correct? Is there a way to bring X numbers of channels of a T1 into * and still have a data channel
15:38.56wunderkinJerJer: bum fuck, egypt :)
15:39.07kippiwhat is the best softphone to use?
15:39.24wasimkippi: asterisk
15:39.45kippifor windows
15:40.02JerJercuz bum fuck eygpt isn't listed in my SBC cheat sheet
15:40.07obsidian-studiosseems T1's are used when one needs Data or Voice and Data, and PRI's are for voice only?
15:40.11kippiso that i can connect to my asterisk box
15:40.47wasimkippi: moziax
15:42.24*** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
15:42.31obsidian-studiosok so the digium cards can do split voice and data? I am totally confused on T1's and PRIs with regard to which one to get
15:43.40pc2JerJer - Bum F Egypt :)
15:43.53parylthe readme on the ftp server for addons says: Using res_config_mysql at the same time as res_config_odbc can create
15:43.53parylsystem instability on some systems.
15:43.56pc2JerJer - Sun Valley, Idaho
15:44.06parylis that referring to an option during compile?
15:44.12parylor something else?
15:44.15pc2JerJer  Ketchum and Hailey also works.
15:49.45iDunnohmm. right, now then, as I appear to be entirely failing to find the information, how would pickup groups work...
15:50.19*** join/#asterisk Katty (n=katrina@68.112.15.110)
15:50.44Kattyrehi.
15:50.50bkw_WOOOHO go Level 3
15:50.53*** join/#asterisk TrevorSHarrison (n=trevorsh@24.49.36.218)
15:53.38*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
15:54.15pc2?
15:56.29*** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp)
16:02.56ZeeekiDunno ?
16:04.12iDunnojust trying to get pickup groups to work, so that I can pickup a different ringing phone from over *here*
16:04.29Zeeekand you looked hard on the internet for an explanation?
16:04.46iDunnonot hard, yet...
16:04.55Zeeekbecause using google and asterisk psickupgroup, I found plenty of explanations
16:04.59Zeeeklike
16:05.02Zeeekhttp://www.marko.net/asterisk/archives/0210/0266.html
16:05.09iDunnoI found *8#
16:05.20iDunnobut that was about it, weirdly dialing that doesn't work ;)
16:05.20Zeeekwhy ask here of you haven't even asked google yet? :)
16:05.41iDunnoI asked google, just not as thoroughly as usual ;)
16:05.51iDunnoand I'd seen that!
16:05.54Zeeekwhat the two words were too hatrd to type?
16:06.03*** join/#asterisk fiber0pti (n=johndoe@207.114.199.98)
16:06.05iDunnoand that doesn't magically work ;)
16:06.15Zeeekwell shit, doesn't that explanation from Mark Spencer do the job?
16:06.16fiber0ptiAnyone have good sources to procure polycom equipment?
16:06.21kippiwhen I try and ring my softphone from one softphone to the other I get a forbidden error, any ideas?
16:07.10*** join/#asterisk lemmy_ (n=rkienzle@gw.k-itx.net)
16:07.19iDunno*cough* - of course, it's entirely possible that I entirely failed to set any pickup groups in the sip.conf.
16:07.45Zeeekfor starters you could just add callgroup=1 and pickupgroup=1 to every accoutn
16:07.59Zeeekthen sort it out later if it works
16:08.03lemmy_hi, has anyone here some experience with rtai, realzap and zaphfc?
16:08.29*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
16:09.15*** part/#asterisk epablo (n=epablo@200.75.139.188)
16:10.21iDunnoZeeek: yeah - just did that, now trying to work out why it's not quite working ;)
16:10.59*** part/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp)
16:11.42iDunnonope - still nothing
16:13.34*** join/#asterisk [MrRed] (n=red@64.39.162.113)
16:14.13[MrRed]cheers, all
16:14.28[MrRed]anyone have any experience with zyxel p2000w?
16:15.19[MrRed]i can get it to register, but not make or receive calls
16:17.42mutilatorhttp://www.biddingatauction.com/listings/predetails/index.cfm?itemnum=813015466&ID=
16:17.49mutilatorcan anyone tell what all equipment that is?
16:17.53mutilatorother than the 2600 router..
16:18.02*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
16:18.07mutilatorone looks like a catalyst i'm not sure tho
16:18.29*** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129)
16:21.28*** join/#asterisk queuetue (n=queuetue@MTL-ppp-146749.qc.sympatico.ca)
16:22.17*** join/#asterisk cTiger (n=tron@213.30.180.98)
16:22.40cTigerI is it possible to use asterisk as you would use a cisco AS5300 as IVR ?
16:22.46cTigerhi by the way
16:22.48*** join/#asterisk pbd (n=plancomm@12.144.118.36)
16:23.01shido6yes
16:23.17cTigernot SIP controled, it has to be MGCP
16:23.35shido6no problem.
16:23.39cTigerit is to interface a cisco BTS10200 to asterisk
16:23.41MuppetMasterWhat does one need to do to get Asterisk v1.2 to use ODBC for Realtime?
16:23.51MuppetMasterI have uncommented the ODBC references in app/Makefile. Is that it?
16:24.01iDunnoahh - that *does* work, I was being thick.
16:24.03iDunno*sigh* ;)
16:24.17pbdMorning/Afternoon/Evening, all.
16:24.31pbdAnyone out there have any experiences connecting Sipuras in Brazil?
16:24.34shido6aftermorning, pbd.
16:25.39*** join/#asterisk wolfson (n=hehe@65.174.122.198)
16:29.14MuppetMasterI do have MySQL/unixODBC/MyODBC installed and have tested the ODBC connection with isql and it works fine.  And I compiled and installed Asterisk after this and no res_odbc.so.  So is this true?:  http://pastebin.ca/24683
16:30.32*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)
16:31.44*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
16:32.07FABRIZIOxxxhello .. must asterisk be working to use fxotune?
16:32.43FABRIZIOxxxok .. found the answer ....
16:33.07hypnoxis there a consensus on what is the best way to avoid crackling and cut outs with music on hold? ive had varying success with changing mpg123 settings and using madplay
16:33.11*** join/#asterisk oej (n=Olle@apollo.webway.se)
16:35.33mmlj4heh... from the mailing list: > Anyone using inter Asterisk trunking IAX /IAX2 ? | No - you're the first to think of that.  Congratulations.
16:35.51Nyvari use rawplayer for musiconhold
16:36.27Nyvarthis takes the cpu load of the machine from having to decompress mp3's on-the-fly
16:37.03*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
16:37.09shido6erf?
16:37.23funxiondoes anyone have a mod10 script integrated with *?
16:37.55Nyvarhypnox, you might also have a look at http://www.sineapps.com/news.php?rssid=1019
16:37.56MuppetMasterAnyone on ODBC?
16:38.30jeffikAnyone familira wih H323?
16:38.51Delvaranyone eat cheese?
16:39.04funxionI eat cheese
16:39.06MuppetMasterjeffik a bit
16:39.07jeffikDelvar: what kind?
16:39.12*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)
16:39.28mmlj4i normally dip my cheese in the first first, it improves the flavor
16:39.33*** join/#asterisk syle (n=blah@unaffiliated/syle)
16:39.34Delvarjeffik: anyrealy.. personly i prefer brea
16:39.45MuppetMasterI eat cheese, my favorite is Spanish Manchego
16:39.51MuppetMasterjeffik what do you ened on H323?
16:39.54MuppetMasterneed that is
16:40.12funxionMuppetMaster thats made with sheep's milk
16:40.26jeffikMuppet: tks, i am usign aah and trying to set up H323 ot provider
16:40.39jeffikDelvar: i prefer camambert
16:40.51Delvaroh yeah thats nice too.
16:41.14Delvari eat it with tomatoes.. hmm
16:41.19MuppetMasterjeffik what is 'aah'?
16:41.19jeffikMuppet: i have H323 installed but do not know how to add it, can not do it vfia amp
16:41.27*** join/#asterisk Inkubot (n=inkubot@200.75.4.7)
16:41.31Nyvarfunxion, http://www.voip-info.org/wiki/view/Asterisk+Credit+Card+Validation
16:41.38funxionthnx myvar
16:41.42funxionnyvar
16:41.54jeffikMuppet: asterisk at home
16:42.13MuppetMasterjeffik Ah, okay.  What provider are you trying to connect to that uses H.323?
16:42.24jeffiktelenet7.pl
16:42.38jeffikthey gave me pin and ip
16:42.47jeffikthey said that is all i need
16:42.47MuppetMasterThey do not support SIP?
16:43.00jeffikyes bu tonly for retail customers
16:43.09MuppetMasterAnd H.323 for?
16:43.16MuppetMasterWhat kind of customers?
16:43.23Inkuboti'm trying to configure festival, i think i got all the configurations, but when it connects, doesn't sound anything..
16:43.26jeffikthey are in Poland, most east europe whole salers only dong H323
16:43.37MuppetMasterjeffik too bad for Poland.
16:43.46jeffikMuppet: for whole sale customers
16:43.48MuppetMasterI have used H323 in ooh323 and with chan_woomera.
16:44.00MuppetMasterchan_woomera is nice but ooh323 comes with the distro.
16:44.03Inkubot<PROTECTED>
16:44.04Inkubot<PROTECTED>
16:44.27Inkubotand the hang up the call
16:44.33Inkubotthe*then
16:44.34MuppetMasterDoes Asterisk@Home support H323 in the web interface?
16:44.36jeffikMuppet: i have the H323 set up just need to know where to add the settings they gave me
16:44.47MuppetMasterh323.conf
16:44.50jeffikMuppet: no AMP does not
16:44.51jeffikok
16:44.52MuppetMasterNot sure in the web interface.
16:44.55MuppetMaster/etc/asterisk/h323.conf
16:45.06RickTickanyone here using asterisk for call termination ...using a remote fxo device?  I need some advice on set up...
16:45.38MuppetMasterBueller, Bueller, Bueller.  No one on ODBC?
16:45.55jeffikMuppet: i have oh323.conf will that work?
16:46.17MuppetMasteroh323 is a very finicky driver, would strongly recommend ooh323 over oh323.
16:46.29jeffikhm
16:46.30MuppetMasterI have an example h323.conf file if you want it.
16:46.45jeffikMuppet: can you pm me?
16:49.26shido6chan_h323
16:50.06Juggieis the bug tracker slow or is it me?
16:50.17fileyeah 'tis slow
16:50.21fileback now
16:50.30pc2Ok, who can provide me a local number in BFE, or an 800 # terminated via IAX or firewall friendly sip for under 3 cents a minute with g711u quality.
16:50.30cpatryJuggie: its okay for me now, not that fast, but its okay.
16:51.48MuppetMasterAnyone know how to get res_odbc.so to compile when unixODBC is present?
16:53.49*** join/#asterisk odie_flocon (n=Odie@ptr-64-201-182-211.ptr.terago.ca)
16:54.03*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
16:54.17fileHmmhesays: ooh
16:54.31odie_floconhey yall
16:54.33Hmmhesaysfile will you buy me a boss gt 8?
16:54.38Hmmhesayspretty please
16:54.47filenope
16:55.03IronHelixhi there
16:55.14IronHelixis it just me or is voip-info's logon system screwed?
16:55.30pbdI noted that yesterday, Iron.
16:55.42IronHelixgotcha
16:55.44*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:56.02mmlj4pc2: try teliax, 2 cents/minute, they probably have 800 #s
16:56.08pbdIt's been new and improved- possibly to death.  I couldn't log in, can't seem to find any comments- and the local search appears AWOL.
16:56.18IronHelixi hope they get their issues worked out, although personally i liked the old format better (even just formatting looks worse)
16:56.39IronHelixyeah logons worked for a while after the upgrade, but broke i guess *shrug*
16:57.43IronHelixthanks
16:57.44IronHelixbbl
16:58.51pbdHey- I'll put this one out for general consumption, since none of my usual contacts seem to be panning out.. anyone out there seen the infamous 'Moving call from channel X to channel Y' message in dealing with a PRI trunk group?
16:59.39pbdI have two working theories as to why I've seen it in the logs- both say it's benign (although possibly a provider issue), but I'd love to know the definite why.
17:00.32pbdCode reads aren't helping much, other than pushing me more to say 'it's really not an issue, it's normal, ignore it'.
17:00.34syledid you grep the c source for that message?
17:01.02pbdSyle- I know precisely when it's generated- in response to a PRI event (any of them, actually).
17:01.34sylenot asking when, asking how
17:01.53pbdBut I *think* it's just referencing the actual channel on the second span, and looking like it's referencing the first span- the code also keeps track of the logical channel that the call's been made on, calling it the REAL channel.
17:02.45pbdHow is simple- whenever the PRI generates an event, a bit of logic gets called to clean up the tables and match the channels- prefixed with the routine that could generate that message, based on the channel number being different from the one requested.
17:03.32pbdWhat's not clear (since it's not commented, and full of the usual 'p->tempnam[]' references) is why there is a difference between the requested channel and the one that's in use at the time- this happens as the call is connected.
17:04.42pbdWorking theory one, since the move makes it correspond to the actual channel on the second PRI in the group, is that it's just not being clear about the fact that it's on span 2, not span one.  Working theory two is that it's responding to the provider requesting to move the channel to a different one.
17:05.24pbdTheory two has me on the phone to a provider, since it's illogical- it's an unused span, being brought up for testing so far.  Theory one says- jeezum, I should figure out a better message and release a patch.
17:06.17sylethat show up in latest cvs?
17:06.32pbdWe've got a couple of other issues with the PRI, so I'll probably be on the phone with the provider to work it all out anyway- but I wondered if anyone else had run into it.
17:07.16*** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
17:07.23pbdCode is essentially unchanged in latest CVS- but this system is locked into 1.0.7 for now, thanks to a third party tool being used.
17:07.46sylewhich one?
17:08.38sylecan;t see how modding a Makefile in apps directory for an addon would lock you back
17:09.00pbdNo, it's a CTI callcenter app- that's unimportant.
17:11.24*** join/#asterisk nagl (n=nagl@213.235.241.6)
17:12.11sylewell , get your logs together etc and post on dev mailing list
17:13.38rayvdTimes'o'joy!
17:15.18*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
17:15.25*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
17:17.01*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
17:17.49*** join/#asterisk Inv_Arp (i=junya@adsl-156-145-65.mia.bellsouth.net)
17:19.27syleasterisk is evil
17:19.48rayvdevil as in mr rogers?  or evil as in barney the dinosaur?
17:19.52sylenow i;ve setup asterisk to listen in on my wife and zapbarge her lol
17:20.07syleerr record and zapbarge
17:20.10rayvdwhat's zapbarge?
17:20.20Nivexthat sounds more evil as in homeland security
17:20.32syleto listen in on zap channels
17:21.16rayvdi'm sure there's a clause in your pre-nup that you have the right to zapbarge her at any time :P
17:21.29sylehehe, well they should cut homeland security funding period and deal with hurricane, but no its just funny, i don; actually listen in all the time , i;m just usually testing stuff and leave that stuff in lol
17:22.24*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
17:23.17sylenoticed zapbarge don;t return a dialstatus variable though
17:24.04*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
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17:25.01*** join/#asterisk Katty (n=katrina@68.112.15.110)
17:25.52BrianR___does anyone know if the sipura spa-2100 actually has working t.38?
17:26.06rayvdsupposedly the latest firmware does
17:26.08rayvdhave you tried it?
17:26.11rayvdcurious myself actually
17:26.34BrianR___rayvd: No.. I'm about to order some stuff.. I don't have a t.38 capable media gateway locally, but I figure I can send the t.38 stuff over the Internet to someplace that does.
17:26.38*** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m)
17:26.49crash3mis 66 jitter 'horrible' ?
17:26.57BrianR___not very good...
17:27.06BrianR___Oh... Does 1.2 now do jitter buffering for SIP?
17:27.31rayvdnot sure if it's in HEAD yet, but there are patches for HEAD that enable it.
17:27.39mutilatorso what games are comin out for xbox 360 that are worth purchasing it right now?
17:27.40rayvdthey were discussing pushing it into HEAD for wider testing
17:28.08BrianR___And the IAX jitter buffer stuff now actually works when there's more than one call over the same IAX transport?
17:28.19BrianR___ie, trunk is on
17:28.30rayvddunno on that
17:28.58BrianR___I suppose I could get the spa-2100 and if it doesn't work out use it in G711 passthrough mode...
17:29.31rayvdactually that has worked fairly decently for us if you have a solid connection :)
17:29.39rayvdi'm going to order a 2100 as well though to try it out w/ t38
17:30.07BrianR___rayvd: the media gateway is in the same building, ethernet with QoS.. Usually less than 10ms latency and no jitter.
17:30.12enderwhat is the show line that shows status of echo chancel and jitter and all that?
17:30.32rayvdshould be fine i'd think
17:30.58rayvdour SIP/PSTN gateway is a few hops out on the internet and it generally works
17:31.05BrianR___just want some ATA's that mostly don't suck and aren't a config nightmare.
17:31.06rayvd(decent sized pipe though and not much jitter)
17:31.15hardwireYou sunk my battlesh*t
17:31.20*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
17:31.20*** mode/#asterisk [+o anthm] by ChanServ
17:31.26rayvdthe sipura's have been pretty good.  i like their provisioning stuff.
17:31.29Nyvarstill waiting for t.38 in the sipura 2002 :(... Mediatrix ata's do t.38 well, but are a PAIN to setup
17:31.40Kattymew.
17:31.45rayvdis sipura actually talking about t38 in the 2002?
17:31.50BrianR___can I do a simple dhcp/tftp deal with the SPA?
17:31.56rayvdyep!
17:32.00rayvdwe do it with the 2002/PAP2-NA's
17:32.05rayvdjust have a master config file
17:32.12rayvdand then device-specific ones (based on MAC) if necessary
17:32.18rayvdauto-upgrades firmware, etc
17:32.35BrianR___very cool stuff. The Sipura ata's are used by vonage and all those big services too, right?
17:32.54hardwireI didn't know linksys ought sipura
17:32.57hardwirebought
17:33.06rayvdi think so
17:33.09rayvdyep, well, cisco i guess technically
17:33.27rayvdpap2-na and spa-2002 are identical down to the interface :)
17:33.38BrianR___so it's pretty much fully baked if the big players are using them to provision residential end user accounts...
17:33.45hardwiret.38 is just store and forward faxing right?
17:33.56Nyvarno, t37 is store-and-forward
17:34.00Nyvart.38 is realtime
17:34.02rayvdi think t38 is a little differen than that
17:34.03hardwireok
17:34.14hardwireso t.38 sends data vs voice?
17:34.16rayvdhttp://bugs.digium.com/view.php?id=3854&nbn=19
17:34.20rayvdthere you go brian
17:34.24rayvdno activity on that one for a few days
17:34.24BrianR___hardwire: t.38 is where the modem is done locally, the bits are sent on the 'net deocded, and then re modeulated at the other end.
17:34.36Kattyanthm: exim is driving me insane :
17:34.38Kattyanthm: :<
17:34.38hardwireBrianR___: so there is generation loss :)
17:34.48BrianR___hardwire: Think of it as putting a faxmodem inside of the ATA. No generation loss.
17:34.48anthmwhy are you using it ?
17:35.04BrianR___hardwire: it's a lot like how data/fax dialing works on a CDMA cellphone.
17:35.05Kattyanthm: because i don't know how to use anything else, obviously.
17:35.21Nyvarno, t.38 requires a capable dsp/chip/code on both ends that touch the PSTN, one on the fax side, and one on the PSTN side, and currently asterisk can't sit between them because it can't handle the SDP for t.38 yet
17:35.22hardwireBrianR___: you got me there brain.
17:35.31*** join/#asterisk KranZ (n=user@sme.bestline.net)
17:35.39BrianR___In the case of CDMA phones, the modem is on the other end at the wireless provider and you've basicly got a telnet style data session open to it.
17:35.40anthmI mean as in "what are you using it to accomplish"
17:35.53hardwireBrianR___: thtas how I invisioned it
17:35.57rayvdhopefully asterisk can soon.. :)
17:36.10hardwireok
17:36.11rayvdmy isp's SIP/PSTN gateway tries to use t.38 and refuses to fallback to G711u :(
17:36.19BrianR___Nyvar: check the mailing list - someone has hacked in t.38 passthrough support. Also, no idea what happens in the case of reinvite.
17:36.35hardwireBrianR___: so a t.38 keeps a channel open but switches the stream to go to a fax modem
17:37.01hardwireor rather.. turns on a fax/modem on a specific channel.. then proceeds to not say a word while the channel is open.
17:37.11Nyvarbrian, i think you are talking about t38/spandsp, not passthrough sip-sip t.38
17:38.03hardwireI need to get something working for me soon
17:38.05hardwireanyhoot
17:38.05BrianR___Nyvar: I haven't seen any report of being able to use spandsp to provide the media gateway function for t.38.. That's the next logical progression though.
17:38.08Nyvarif I'm wrong, I'd love to hear that sip-sip t38 has been patched into *
17:38.09hardwirethe potty awaits!
17:38.31hardwireNyvar: re-invite inside of domains
17:38.37hardwirenot really  good solution
17:38.41hardwirea
17:38.46hardwirebut hey.. it works
17:38.53Nyvarahh
17:39.03Nyvarre-invite works for me
17:39.05hardwireor I would assume it would
17:39.19hardwireI saw that you can now limit re-invites to domains.. which I thought was majorly cool
17:39.23hardwirethats going to help me out a bundle
17:39.29Nyvarerr, that is to say that I wouldn't have a problem using re-invite
17:39.47hardwireI was't just trying to shut you up :)
17:39.50hardwirebbl
17:40.13Kattyanthm: every time my machine reboots, i turns my conf file into a backup file...then i have to go delete the original and rename the backup file..... what i want it to do is just send out emails when someone gets a new voicemail. don't really want it using our exchange server though...
17:40.41Kattyanthm: exchange gives me nightmares :<
17:40.57anthmi will give you a perl script
17:41.09hardwirehttp://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-46251073536.htm
17:41.10hardwiresexy
17:41.15anthmthat you can rename to /usr/sbin/sendmail
17:41.24anthmthat will emulate it's behaviour
17:41.39Kattyi gather sendmail is an email server of some sort
17:41.41anthmand you can set it to fwd all the mail to your outboud server
17:41.42anthmif you wan
17:42.11Kattyanthm: i don't know perl.....does that matter?
17:42.22anthmno?
17:42.30Kattynot a perl gerrrl
17:42.31anthmyou do have a real smtp server correct?
17:42.41Kattyyes
17:42.43Kattyexchange
17:42.46Kattybut i don't want use that
17:42.51anthmthis script can accept email from ast and send it all to there for outb
17:42.56crash3mexhange *shudder*
17:43.19Kattyanthm: but it can forward to any email server?
17:43.21hardwirehmm
17:43.34hardwireI can see why anybody coming in here with AudioCodes Mediant 2000 questions is so frantic
17:43.36KranZhardwire: that's only clearchannel
17:43.37Kattyanthm: based on.........ip?
17:43.42hardwirethey just blew $5000 and are pissy its not working
17:43.49anthmsmtp relay
17:43.57hardwireKranZ: you want a ds3 in fx* ?
17:43.59anthmyou pick the ip of an outbound relay server
17:44.05Kattyanthm: l
17:44.07KranZyou can only use it for data
17:44.08Kattyanthm: k
17:44.09rayvdour PSTN provider is using AudioCodes
17:44.20Kattyanthm: you have my gmail account?
17:44.21rayvdi think it sucks at following the SIP RFC
17:44.26hardwireKranZ: my PRI is clear channel
17:44.27Kattyanthm: or are you just going to post it somewhere
17:44.30anthmand the script accepts the email and connects to that ip and sends the mail there to be delivered wherever
17:44.38KranZyour pri has 24 clear channels
17:44.39Kattyanthm: excellent
17:44.51hardwireKranZ: yeh
17:44.52anthmmy network is on the fritz
17:44.57Nyvarhardwire: no it isnt
17:45.01Kattyanthm: i'll bug you for it later.
17:45.14BrianR___http://www.ionidea.ua/oss/asterisk/ (half-baked t.38 media gateway thing)
17:45.25hardwireKranZ: 23 clear channels and one d :)
17:45.31hardwireit says so in /proc/zaptel/1
17:45.48hardwireI am most likely mistaken
17:45.49*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-82-81-5-2.red.bezeqint.net)
17:45.57hardwireand yes.. this doesn't look like it has any voice abilities
17:45.58KranZtrue
17:46.11KranZbut you cant use that ds3 card for what you're thinking
17:46.20hardwirewell you damn well should be able too
17:46.21Nyvar23 DS0 channels isn't clear channel, all 24 DS0's on the T1 being used in one big 1536Mb/s channel is clear channel
17:46.39KranZNyvar has it right
17:46.48hardwireNyvar: whereas its data at that point
17:46.52hardwirevs voice
17:47.01KranZyeah
17:47.22hardwirewhy would they say clear w/ zaptels /proc/ summary?
17:47.26hardwirejust seems odd
17:47.28KranZhardwire: i got excited too when i saw that card the 1st time
17:47.30Nyvarwell, i could use it for voice, if I had software at both ends that pulled the voice from the 1536Mb/s channel instead of a 64Kb/s channel
17:47.36KranZcus each channel is a "clear channel"
17:47.48PoWeRKiLLhi
17:47.50hardwireKranZ: is it that way just to piss me off?
17:47.52Nyvarhowdy
17:47.53KranZheh
17:47.55KranZyeah
17:48.03PoWeRKiLLHow not to pass audio when dialing SIP -> * -> ZAP ?
17:48.19hardwireNyvar: clear channel == one big whopping channel?
17:48.27Nyvaryep
17:48.33*** join/#asterisk misak (n=misak@0x535f6f15.virnxx18.adsl-dhcp.tele.dk)
17:48.39Kattymy firewall is overloaded >.<
17:48.40hardwireyou would have an amazing voice stream over a large clear channe
17:48.41Nyvarwell, treating the ds0's together as one big channel
17:48.59hardwirebut you can split it up
17:49.13hardwireso that you have a 12 channel block for data at the end that acts as one channel
17:49.13Nyvards0's aren't really there, unless your t1 device breaks them out individually
17:49.24KranZif the card supports it
17:49.37KranZthat ds3 card doesnt :(
17:49.42hardwireheh
17:49.56hardwirea 44mbps ss7 link?
17:50.17KranZcould handle the worlds signalling traffic
17:50.19KranZheh
17:50.24KranZprolly not, but its alot
17:51.03Nyvara t1 samples and sends 193 bits 8000 times a second, 192 bits are used for the 'data' part of the t1, and the 193'rd bit is used for t1 signalling.  192*8000 =1536Mb/s, 193*8000=1544Mb/s
17:51.27anthmKatty: http://www.slinear.org/eg/sendmail
17:51.38anthmtry putting that as /usr/sbin/sendmail on the box
17:51.49anthmedit it to point to your smtp server
17:52.00*** join/#asterisk wrzf1 (n=chrisf0r@rrcs-70-61-133-91.central.biz.rr.com)
17:52.06JuggieNyvar, isnt each channel 56kbit with 8bit used to signal, is that how it works?
17:52.11anthmmy $relayhost = "localhost";  change localhost to your relay box
17:52.25Nyvarjuggie, thats a layer ontop of the actaul t1
17:52.34hardwirebut its full duplex
17:52.42Juggietrying to think back to the last t1 course :P
17:52.42hardwireso it sends and recieves 193/sec
17:52.47hardwireerr
17:52.51hardwire193 bits 8000/sec
17:52.53Nyvarthe t1 just carried bits, then you can do whatever "protocol" you want on the t1, clear channel, channelized, PRI, etc
17:52.55*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-82-81-5-2.red.bezeqint.net)
17:53.19Nyvarwhich just means that both ends decide to pull the bits off the t1 in an agreed fashion.
17:53.27hardwireyoink
17:53.34Nyvaryou are talking about 'bit robbed' ds0 signalling
17:53.35hardwire@ 8000/sec
17:54.14*** part/#asterisk pbd (n=plancomm@12.144.118.36)
17:54.18NyvarPRI is a differnt protocol, it breaks the 1536Mb/s into 23 64Kb/s channels, and 1 signalling channel
17:54.30NyvarPRI doesnt bit-rob
17:54.57*** join/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
17:57.04Kattyanthm: thanks.
17:57.11anthmnp
17:57.24Juxthi, can anyone help me figure out why zaptel isn't compiling correctly on 2.6.3.13?
17:57.26PoWeRKiLLHow not to pass audio when dialing SIP -> * -> ZAP ?
17:57.35Kattyanthm: can you fix my firewall with a magic wand too?
17:57.40Nyvards-3 works the same, you can use the whole 44.7Mb/s as clear channel, or fractionalize it into T1's, or run ATM, or whatever other protocol you want.  For voice, you'd probably want a DS-3 card that is capabile of channelizing all the way down to individual DS0's (64Kb/s channels)
17:57.42*** join/#asterisk mut (n=animenod@65.111.201.79)
17:57.54Kattyanthm: it's a big yellow symantec box :<
17:58.05Juxtahah call mr. norton
17:58.07anthm?
17:58.16Kattyanthm: you obviously missed my sarcasm
17:58.18anthmdoubt it
17:58.31anthmyah 1/2 attn atm
17:58.35anthmmega crisis
17:58.43Juxtdon't those synamtec boxes run a stripped version of windows?
17:58.47KattyJuxt: mister norton goes straight to voicemail!
17:58.56KattyJuxt: uhh
17:59.00KattyJuxt: it's a box
17:59.04KattyJuxt: not a computer
17:59.06Juxtit's something along the lines of what io mega nas run
17:59.40hardwireiomega?
17:59.47hardwirecrazy stuff
17:59.48Juxtyeah, iomega makes NAS boxes
17:59.56hardwireI have their rev drive
17:59.58hardwireI don't use it
17:59.58Juxtinside is a board that runs windows for appliances
17:59.58*** join/#asterisk BBRodrigueZ (n=john@p54B01BC3.dip0.t-ipconnect.de)
17:59.59hardwireyet
18:00.05hardwireneed to make backup system w/ it
18:00.44hardwirebecause its worth the shipping
18:00.46*** join/#asterisk Assid (n=assid@203.115.64.57)
18:00.46hardwireand thats it
18:00.49hardwiredonate to a school
18:01.21Assidheyyyyyy
18:01.23Juxthar har
18:01.27Assidhrmm
18:01.29hardwireKatty: its like you are going to have a monday all over again
18:01.32Juxtsun is pickin gup pgsql
18:01.34lathos42Naah, I used to work for a School District and I was always pissed when people donated junk that I was supposed to do something with :)
18:01.36Assidanyone here using voipjet?
18:01.38Kattyhardwire: not my problem.
18:01.46HmmhesaysKatty: I got in the karaoke contest last night
18:01.52KattyHmmhesays: oooooh
18:01.53hardwireKatty: I typically have a monday whenever ms is concerned
18:01.55Hmmhesaysi was selected for the sing off
18:01.55KattyHmmhesays: how'd it go?
18:02.04KattyHmmhesays: !!!
18:02.08KattyHmmhesays: i wanna go!
18:02.18Hmmhesaysthere were more people dancing for my songs than anyone else
18:02.20*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
18:02.20odie_floconbkw_ are you in here?
18:02.22Hmmhesaysit was grand
18:02.31KattyHmmhesays: send a plane to get
18:02.36hardwireHmmhesays: rock on man
18:02.37KattyHmmhesays: s/get/get me/
18:02.44Hmmhesayslol
18:02.56Hmmhesaysi love karaoke night
18:03.04KattyHmmhesays: i've yet to try karaoke night.
18:03.16KattyHmmhesays: i went to one resort where my best friend tried to get me to sing some whitney houston
18:03.31Hmmhesaysits a rush when all kinds of people come out and dance
18:03.46Kattyi imagine i'd make the sing off thingy though
18:03.47Juxti once did bon jovi at karaoke
18:03.54KattyHmmhesays: what do you think?
18:04.02KattyHmmhesays: you've been to lots of karaokes....would i make sing off thing?
18:04.19Hmmhesaysi dunno, they judge you on  singing, stage presence and audience particpation
18:04.24Hmmhesays*participation even
18:04.27Kattyoh
18:04.30*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
18:04.38KattyHmmhesays: you mean i can't sing and hide in the corner at the same time? :/
18:04.43muthire some people to come in and start clapping along with the song
18:04.43Hmmhesaysnope
18:04.45mutand moshing
18:04.47Kattysad.
18:04.47Hmmhesaysgotta play the crowd
18:04.54KattyHmmhesays: i don't do crowds.
18:05.00mutcrowd surfing
18:05.01Hmmhesaysi love it
18:05.05KattyHmmhesays: you would.
18:05.13mutdon't tell my heart, my achy breaky heart
18:05.14KattyHmmhesays: you little social fiend you.
18:05.20muti just don't think it understands
18:05.24Kattymut: he
18:05.29Kattymut: don't think /he/ understands
18:05.42muto
18:05.49muti just don't think he understansds
18:06.01Hmmhesaysi don't need to hire people to clap for me
18:06.03Kattyactually, he'd
18:06.11mutand if you tell my heart, my achy breaky heart, he might blow up and kill this man
18:06.18KattywhoooooooohooooooO!
18:06.22*** join/#asterisk SwK (i=pfvzfm@12-219-144-126.client.mchsi.com)
18:06.29KattyHmmhesays: i'd clap for you.
18:06.30mutheh
18:06.33Hmmhesays:D
18:06.49KattyHmmhesays: i'd claw for you with mittens and kitty ears on!
18:06.53Kattyoh.
18:06.55Kattyi mean clap.
18:07.01Kattybut i'd claw too.
18:07.02Kattyclawing is fun!
18:07.17Hmmhesaysok gotta run
18:07.21Kattykthxbi
18:07.44Kattylathos42: it's almost mitten season :>
18:07.46SwKhmmmm
18:08.01SwKheyya katty
18:08.14*** part/#asterisk dwildes (n=dwildes@209.164.237.195)
18:08.15Kattyhihi
18:08.37*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166117213.nb.aliant.net)
18:15.09*** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
18:15.09*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now!
18:16.54Nivexdamnit... digium.com just locked my browser at 100% CPU usage
18:18.26*** join/#asterisk kippi (n=chrisfro@cpc4-hatf3-6-0-cust228.lutn.cable.ntl.com)
18:18.30kippihi
18:18.40*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
18:18.55Kattyhi.
18:19.09KattySwK: come fix my firewall!
18:19.21Nivexhttp://www.cnn.com/2005/TECH/10/05/katrina.tech.response.ap/index.html
18:19.24kippiI am runing asterisk @ home, I am runing ISDN, how can i get it to work out which number is ringing and point it to the right group/extension
18:19.31*** join/#asterisk implicit (n=implicit@dhcp-211193.mobile.uci.edu)
18:20.23mutah
18:20.32mutnone of that wifi crap works without a telco backhaul
18:20.36mutpish posh
18:21.01rayvdOwls.
18:21.16*** join/#asterisk santiago (n=santiago@208.195.215.98)
18:23.44Hmmhesaysjust like every cowboy sings a sad, sad song
18:23.48Hmmhesaysevery rose has its thorn
18:25.11rayvdtimely words, TIMELY
18:26.12RickTickanyone here using asterisk with  ... a remote fxo/ata grandstream HT488?  I need some advice on set up... the fxo port
18:27.22brettnembarf
18:28.36*** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com)
18:28.50*** join/#asterisk C4-Telecom (n=sales@254.Red-80-33-214.staticIP.rima-tde.net)
18:29.01C4-TelecomHi
18:29.06kippihow do I work out what number is coming in on isdn?
18:29.11Pete_LargoHas anyone thought up a way to send an email message when a call comes in to a DID?
18:29.36C4-Telecomany one know about the Quad GSM card for * ???
18:29.36Pete_Largokippi your provider should have given you a list of numbers
18:30.25kippiPete_Largo: how can i get asterisk to see these numbers and point to the right extension?
18:30.37Pete_Largothat I don't know
18:30.52Pete_Largowhat does the wiki say?
18:31.09kippinot sure what it would be called
18:31.59Pete_Largomy guess, and it's purely a guess, is that you will have a context line in zapata.conf and that is where you assign a context to incoming calls.  then in extensions.conf you assing the did number to ring a phone...
18:32.32kippithey won't be did
18:32.36enderAny users of the Polycom IP-4000 ?  I'm thinking about using it for our conference room.
18:33.11enderkippi: most likely the ISDN passes on 4 digits, the last 4 digits of the telephone number dialed.
18:33.20enderkippi: you have to work from those 4 digits.
18:33.37Pete_Largohttp://lists.digium.com/pipermail/asterisk-users/2003-June/013410.html
18:33.44Pete_Largoand they will always be did on a pri
18:33.56Pete_Largo~docs
18:33.57jbothmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
18:34.35*** join/#asterisk NDT (n=me@cpe-24-195-219-245.nycap.res.rr.com)
18:34.51NDTanyone here use TDM400p card before?
18:36.23NDTI bought the 4 port with 1 fxs and 1 fxo shouldn't the 2 ports both be lit up on the back?
18:36.52Nyvarshould be 1 line to send an email when a call comes in, something like exten => xxxx,1,System(echo SomeText | /bin/mail -s "MySubject" address@domain.com)
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18:38.47MRH2is the  GROUP_COUNT() function intended to be case sensitive?
18:39.24simprixIm trying to make a call to a outside trunk with asterisk. I am using linphone but when I put in the number it doesn't know to dial it. it says cannot resolv address
18:40.01enderNyvar: I would be careful.  If system returns -1 the call gets dropped..
18:40.26Nyvartrue, so make sure that your mail proggy works
18:41.07glm2kNyvar: use the management interface then?
18:41.11Pete_Largokippi : http://www.voip-info.org/wiki/view/Asterisk+PRI
18:41.30Nyvarglm2k, i don't understand your question
18:41.33glm2ki rememberreading about an app that notified you through IM
18:41.59Nyvarneat, haven't seen that
18:42.24Pete_LargoI was just thinking about the same thing nyvar
18:42.58Nyvarthat would be pretty cool, pop up an ICQ message or something when a call comes in
18:43.11enderrisky
18:43.14Pete_Largoemail is fine... motly for tracking purposes
18:43.36Pete_Largokeep track of what calls you've missed while you were in the head or what not
18:44.00Pete_Largoand I could do with the email being sent after the caller hangs up
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18:44.19*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
18:46.17PupenoLWhat is the meaning of 'nonce' in the Asterisk environment ? in such phrases like: "generate fewer 'stale nonce' messages (issue #5381)" ?
18:47.02PupenoLif it was jabber instead of ICQ it'd be cool.
18:47.20enderthere are cli interfaces to jabber clients.
18:47.24Nyvarwhen authenticating, the challenging server sends a 'nonce' value that the caller then uses to encode their password and send back the reply. Nonce's are time-based, making it difficult on replay attacks
18:47.32enderit's just risky becuase you could get a -1 return if the client has become offline.
18:47.51enderyou could however wrap the call to the jabber app in a script that returns 0 no matter what.
18:49.28*** join/#asterisk argos73 (n=mike@adsl-70-228-109-5.dsl.akrnoh.ameritech.net)
18:50.28Nyvar~nonce
18:52.45PupenoLNyvar: thank you.
19:01.35peggeri want dailing in to be able to put a number into a read varable how would i make it so that say i put the pound key to tell the read statement that i am done
19:02.06*** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
19:03.03TripleFFF2sdfi need a answer prob
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19:08.55AssidNyvar : i think you can
19:09.08enderTripleFFF2sdf: ask your question.
19:09.09Assidsomething like uin@icq.com or something
19:10.57Nyvardoesnt look that easy
19:11.44Assidit used to be
19:11.46Assida long time ago
19:11.49Assidsomething like that
19:11.53Assidor icqemail.com or something
19:11.58Drukenhas anyone figure out how to stop voicemail from saying root is sending the email ?
19:13.17Drukenpegger: read uses # to end it as default...
19:13.42TripleFFF2sdflol
19:13.45TripleFFF2sdfi need this
19:14.02TripleFFF2sdfsend SIP invite  with ESRN in Invite field, CBN in From field and ESQK in PAI field
19:14.26TripleFFF2sdfwhere ESRN is emergency service number, CBN is callback number and PAI is P Asserted dentity
19:14.30TripleFFF2sdfwth is PAI ?
19:14.33TripleFFF2sdfin the ivite packet
19:15.04*** part/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
19:15.23Jam1eHow do I enable like, when someone rings, they can press # then an extension number to be put through to that extension direct?
19:16.44*** join/#asterisk file (n=jcolp@mctnnbsa31w-142166117213.nb.aliant.net)
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19:19.09Nyvarjam1e, exten => _#XXX,1,Dial(SIP/${EXTEN}:1)    - this will strip off the # and dial the last 3 digits
19:19.18*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
19:19.31Jam1ethanks
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19:20.46michael1234can someone help me with zaptel
19:21.52afrosheenDruken: uh..isn't that in /etc/asterisk/vm_general.inc
19:22.15Drukenafrosheen: i dunno... i will have to look
19:22.30Druken-bash: /etc/asterisk/vm_general.inc: No such file or directory
19:22.33Drukenapparently not
19:22.47afrosheenyou need to make it or get it and include it, put in a directive called serveremail=
19:23.13Drukenoh, that's in voicemail.conf and it's already set... just don't work
19:26.14*** join/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
19:28.52Nyvardruken, did you try something like: fromstring=Mario <itsameamario@domain.com>
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19:30.49Nebukadnezahi
19:31.04Nebukadnezai got rather big problems with a cologne zaptel / zaphfc card
19:31.12Nebukadnezai loaded the kernel modules ...
19:31.15Juxtdoes anyone have a take on this? http://bugs.digium.com/view.php?id=5236
19:31.24Nebukadnezabut just wont dial in or out from/to that zap channel
19:31.39Nebukadnezadmesg says everything is allright
19:32.07Nebukadneza(just the normal zaphfc[0]: tx sync changed: x,x
19:32.14pygrammer^foodDoes one need anything special to send faxes with Asterisk over the PSTN?
19:32.18pygrammer^foodI know nothing about it.
19:32.37theblueHi all.
19:32.45DrukenNyvar: yep
19:32.54Juxtpygrammer: how are your faxes originating?
19:33.08Nebukadnezaany ideas where to look for more? asterisk itself registers the channel, but when a call is incoming it simply doesnt notice
19:33.12pygrammerpygrammer: Can you just use JPEGs/images?
19:33.21Nyvardruken, nm, do serveremail=mario@domain.com   and fromstring=Mario
19:33.38pygrammererr
19:33.39pygrammerhaha
19:33.41pygrammerJuxt*
19:33.44thebluepygrammer: I do that all the time.
19:33.51pygrammerYeah. :/
19:34.11Juxtyou should look into hylafax
19:34.17pygrammerJuxt: What do you mean by originating? I'm just wondering about sending faxes, not receiving.
19:34.37Juxtlike how do you want the image/text to get to asterisk?
19:34.45Juxtare you trying to build a emial-to-fax gateway?
19:34.46Nebukadnezano ideas?
19:35.04pygrammerJuxt: Oh, not necessarily. Perhaps just an uploaded JPEG right now.
19:35.13pygrammerI'm just wondering if it can be done over the PSTN without any special hardware.
19:35.14Juxtyou should look into http://www.hylafax.org/
19:35.28Nebukadnezawould be really nice if anyone could help me ... it sucks not beeing able to call someone :P
19:35.28pygrammerYeah, I am looking at it, but I was wondering about *'s built-in fax functionality.
19:35.52Juxtasterisk can def. receive faxes and turn them into images
19:36.01Juxtbut even there is relies on 3rd party software
19:36.06Juxtasterisk itself wasn't designed for that
19:36.37pygrammerHow about sending faxes?
19:36.44pygrammerDo standard POTS lines allow data like that to be sent?
19:36.50pygrammerI don't know much about the physical PSTN.
19:36.51Juxtabsolutely
19:37.17Juxtas i said, you just need a fax modem
19:37.19pygrammerSo, theoretically, one could use Asterisk to send faxes over the PSTN with a standard termination service like Voipjet/Voxee.
19:37.23pygrammerOh...
19:37.29Juxtwell! that's a different issue
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19:37.48pygrammerSending faxes via VoIP (SIP/IAX) to a termination provider and out thru the PSTn
19:37.49Juxtyou can send faxes over voip services but it requires wide band codecs and a perfectly clean connection
19:37.49Givurhi all
19:37.50pygrammerPSTN*
19:37.59pygrammerJuxt: Yeah, I've heard there are issues with it.
19:38.07Juxtgenerally possible but very hard
19:38.31Juxtyou need to find a T.38 capable carrier
19:38.34Kattymewmew
19:38.35pygrammerJuxt: So, it's wiser just to hook it up to a fax modem and go through the PSTN analog?
19:38.45Juxtyes, def.
19:38.54Juxtyou'll save yourself a lot of time
19:39.00theblueAnd aggravation.
19:39.06*** join/#asterisk Teeli (i=Tili@202-133-67-133-dialup.sat.net.pk)
19:39.13Juxti have spent about 6 months so far trying to get fax to work
19:39.20pygrammerHeh.
19:39.24afrosheenDruken: well, that's how mine works...
19:39.24Juxti have mostly succeeded but sometimes i have to send a fax 2-3 times
19:40.38Juxti suggest all my business customers that switch to voip to keep 1 pots phone line for alarm/fax
19:40.41pygrammerJuxt: Easier to just send it conventionally. :D
19:41.15GivurQuestion: Do I need for the MusicOnHold (MP3s) a SoundCard in the Asterisk computer?
19:41.19Juxtthis will be taken care of as soon as T.38 support gets into *
19:41.33*** join/#asterisk queuetue (n=queuetue@MTL-ppp-146338.qc.sympatico.ca)
19:41.34JuxtGivur: no but you need a timer
19:41.43GivurJuxt: Timer?
19:42.03queuetueI've set up a new aah box, put it behind my firewall, redirected ports 5060 and 5061 to point at it, and set up a sipura 2000 across the internet. I set up an extension and logged into the server to watch asterisk -r (sip debug).  The sip call comes in, asterisk recognizes it and shows me the guy's caller id settings, but then gives a 403 forbidden error.  What did I miss?
19:42.05Juxthttp://www.voip-info.org/tiki-index.php?page=Asterisk+Zaptel+Installation
19:42.08*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
19:42.16JuxtGivur: look on voip-wiki.org for ztdummy
19:42.18pygrammerGivur: Ztdummy.
19:42.39*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
19:42.55GivurAhh. Thanks
19:43.13pygrammerJuxt: What does the timer really do?
19:43.22pygrammerI know it's essential, but for what?
19:43.45Jameno123"delay"
19:43.54Jameno123id assume :)
19:43.55Juxtit's a reference for the * on how to fast to keep feeding packets to the chanel
19:44.12Nebukadnezayou need some option for the zaphfc module to use it in nt-mode, right?
19:44.20pygrammerJuxt: Ah.
19:44.27Nebukadnezai just forgit which
19:44.31Nebukadnezaplease help me ... i beg you :/
19:44.38Nebukadnezai _really_ need help
19:44.41Jameno123if it wasnt there, then your moh and such, would be all jibbered.
19:44.43Juxtthe best thing you can do is actually convert your mp3s to raw
19:44.52bjohnsonqueuetue: adding him to sip.conf and forwarding all the other sip ports
19:44.59Nebukadnezajust my simple cologne card and the zaphfc (bristuff) module
19:46.20theblueWhat's a cologne card?
19:46.28Juxtclone?
19:46.38theblueOh.
19:46.42Nebukadnezaa nt-mode noname isdn card
19:48.56Nebukadnezazaphfc: Card 0 configured for TE mode
19:48.56Nebukadnezazaphfc: Card 0 configured for master mode
19:48.56Nebukadnezazaphfc: 1 hfc-pci card(s) in this box.
19:48.59Nebukadnezalooks okay to me :/
19:50.17*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
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19:51.32Nebukadnezaah goddamned
19:51.36Nebukadnezawhats so hard bout this?
19:52.00*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:52.34Nebukadnezahm
19:52.42Nebukadnezathis isnt good style but i guess ill reboot my box :/
19:53.55Nebukadnezad'oh
19:53.56NebukadnezaDOH
19:53.56*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
19:54.05Nebukadnezawith make loadNT in the bristuff/zaphfc dir it works :?
19:54.07Nebukadnezadunno why
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20:12.33Givurrehi
20:13.23*** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
20:13.41pc2Ok, who can provide me a local number in BFE, or an 800 # terminated via IAX or firewall friendly sip for under 3 cents a minute with g711u quality.
20:14.29pygrammerIs there a good seller/reseller of virtual DIDs?
20:14.33pygrammeri.e. for VoIP use
20:15.19syleunder 3 cents would be american only
20:15.21syleno canada
20:15.32*** join/#asterisk Micc (n=dotirc@c-24-19-175-112.hsd1.wa.comcast.net)
20:16.29MiccSo I've rebuilt a new linux box and put the latest build of asterisk on it and I still have gaps in my audio after a few minutes when dialing out with nufone.
20:17.10Pete_Largoanyone have a Mediatrix 2102 sample config file?
20:17.21MiccDoes anyone use nufone for long periods of time like > 30 min?
20:17.41MiccI've been having problems with the call breaking up for months now.
20:17.43pc2syle - that's fine
20:17.47BeirdoMicc: all the time
20:17.51Beirdoit works fine for me
20:17.55MiccI'm not sure if its nufone or something with my configuration.
20:18.04pc2Nufone is great... except incoming 1-800
20:18.19meshugaanyone else having teliax problems?
20:18.57meshugawhen i make an outgoing call and i dont hear ringing, the other person can always hear me, but i cant hear anything
20:19.02MiccIt seems that when using my PRI there is no problems but when dialing out using nufone there is something wrong.
20:19.03meshuga1 out of 3 or 4 calls works great
20:19.07pc2meshuga - sounds like a firewall to me
20:19.09MiccI'm not sure where to get help on this.
20:19.15meshugapc2: its not.
20:19.22meshugapc2: its a directly connected ip
20:19.34meshugaand if it was, it'd be consistent wouldnt it?
20:21.07meshugaso is anyone on teliax here?
20:21.16meshugaand if so, whats your gateway?
20:21.23meshugai'm on gw03 right now.. with limited sucess
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20:26.14pbdRegreets, all.
20:27.16pbdAnyone out there have personal experience setting up a Sipura in Brazil?  (no, thanks, I already know how to set one up in the US.. but I'm lost on the appropriate voltages, etc for Brazil.. anyone see 66v as an FXO voltage before?)
20:27.30*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
20:28.50pbdSound of crickets.  Is this thing on?
20:29.05xhelioxYup.
20:29.19pbdGood, now I know it's just cuz no one has done it before. :)
20:29.25xhelioxYup. :)
20:29.31xhelioxNo one here, anyhow.
20:29.44pbd(no sweat, back to google- I was just hoping to short circuit an afternoon of reading portuguese.
20:30.05xhelioxAnd I'm an American, so it's illegal for me to recognize that things are done differently in other countries, especially Brazil. ;)
20:30.21*** join/#asterisk Inv_arp (i=junya@adsl-156-145-65.mia.bellsouth.net)
20:30.46pbdOy.  And you obviously don't work for a multinational company.  I have to understand Japanese AND brazillian portuguese PSTN standards.
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20:31.36Dr_Raypbd.. I think he was joking :)
20:31.54pbdAlong with remembering the US standards, next to my desk.
20:31.57*** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
20:31.59pbdI know, I was just griping.
20:32.01pbdFeels good to gripe.
20:32.05pbdGripe Gripe Gripe.
20:32.06Dr_Rayyes it does
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20:33.32RickTickdoes anyone here know ...how to setup an external FXO port ...so that * can access it... for dialling out... to the pstn?
20:37.27Ariel_RickTick, you mean like a sipura 3000
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20:39.30pbdRckTick- all depends on your hardware.  X100P?  TDM4XX?  Sipura?
20:39.45*** join/#asterisk SwK (i=nhdrbr@12-219-144-126.client.mchsi.com)
20:39.46pbdFunky USB FXO adapter?
20:40.10pbdPhoneJack?
20:40.13Ariel_well external to means on installed on the box. like a sipura 3000 or a grandstream 488
20:40.21pbd(erm.. I think that's LineJack).
20:40.35Ariel_well I have to drive home see you all later
20:40.42pbdLater, Ariel
20:44.14RickTickAriel: Yes
20:44.36bjohnsonthere is info on the wiki about the 3000
20:44.45RickTickpbd:  I have a grandstream HT488 ...it has a fxo and fxs ports
20:45.22bjohnsonbasically you set it up exactly like an fxs, but there might be a few more options to set in the fxo firmware
20:45.46bjohnsonas far as * is concerned, it's a fxs, fxo, abc ... * doesn't care
20:46.38RickTickBjohnson: I have the port registering to * ... but when I send a dial command ... I get  a 482 ...loop detected ... then * dial thru default context
20:47.05Kattymew?
20:48.25*** join/#asterisk acertain (n=acertain@200.119.18.102)
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20:48.56acertainHi all
20:49.26acertainspanish talking going to Astricon? quien que hable español va al Astricon en Anaheim?
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20:51.41afrosheennyet
20:52.10queuetueI've set up a new aah box, put it behind my firewall, redirected ports 5060 and 5061 to point at it, and set up a sipura 2000 across the internet. I set up an extension and logged into the server to watch asterisk -r (sip debug).  The sip call comes in, asterisk recognizes it and shows me the guy's caller id settings, but then gives a 403 forbidden error.  What did I miss?
20:52.38pauldyit is an unknown sip call
20:52.48pauldyso you are getting conjection
20:52.52pauldyerr conjestion
20:53.15pauldyit is in the dial plan you need to either route the incomming call to an extension or setup a did for it
20:53.26acertain403 means that asterisk doesn't know the number extension
20:54.47acertainanyone going to make the dCAP test on Astricon, Anaheim?
20:55.46acertain???
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21:02.21*** part/#asterisk froud (n=froud@ndn-165-134-136.telkomadsl.co.za)
21:03.10*** join/#asterisk x-router (i=richard@uberpussy.net)
21:03.29queuetuepauldy: was that an answer to me?
21:03.37acertainis anybody here?
21:03.44Beirdono
21:03.56Beirdojust 329 nobodies
21:04.37Dr_RayI was here 20 minutes ago, but I am gone now.
21:06.40pbdSorry, guys- I've been off googling with no success.
21:06.42*** join/#asterisk h3x0r (n=h3xor@64.192.116.16) [NETSPLIT VICTIM]
21:07.05pbdSomeone mentioned that they have any Sipura 3000 experience in brazil?  Or knowledge of FXO setup in Brazil?
21:07.20Dr_Raymaybe ask on users?
21:07.40queuetueacertain: OK, you were right - I had set up the wrong extension #.  I deleted that one and added a new one, and now instead of 403 - forbidden, I get 401 - Unauthorized.  The passwords seem to be set correctly...
21:07.42pbd(I know how to set up a Sipura, guys- the tough part is not being able to speak Portuguese and slogging through technical sites looking for the settings)
21:07.49acertainI have 8 FXO in Brazil
21:07.56Dr_Raythere you go
21:08.03pbdDr_Ray- it's a good idea..
21:08.04acertainwhat do you need?
21:08.39pbdAcertain:  I've sent a Sipura down to SP, and when I attempt a dial out- I get nothing.  I think I need to monkey with the FXO settings, but don't know what to monkey with.
21:08.44pbdThe FXS port is working fine.
21:08.49Connor-Does the order in which codec's appear in sip.conf make any difference??
21:09.10pbdThe 3000 is set to US standards.  (although I've adjusted the impedance to Telefonica Europe standards, to no avail)
21:09.24*** join/#asterisk epablo (n=epablo@200.75.139.188)
21:09.38acertainpbd: give me 1 sec... I'll tell you how to setup the FXO for SP
21:09.57pbdYou'd be a lifesaver if you get it, acertain- I'd owe you big.
21:10.07Dr_Raypbd, then maybe add that to the wiki (in english) :)
21:10.24acertainI have several asterisk servers in production at SP
21:11.04peggerwhat is sp
21:11.44acertainSao Paulo, Brazil
21:12.48queuetueThe sipuras inside the firewall (on the local lan) seem to have no problems connecting.  The ones coming from other locations seem to be getting 401 errors.  Is there something I need to do to give them access?
21:15.45acertainpbd: this is the clue, use "signalling = fxs_ls"
21:15.59acertainat least that work perfect with my Digium FXO cards
21:16.08acertainI have 4 PRIs too
21:16.14mishehucan anybody recommend a good opteron board that works well with the te1xxp cards?  I don't want to have a repeat of my last system that chokes and gives PCI PERR# up the rear with digium cards
21:17.12tzangermishehu: contact digium, I think they've found and eliminated that problem.
21:17.14acertainpbd ?
21:17.27pbdAcertain- let me try it (on phone, wait one).
21:17.34acertainok
21:18.13hardwirebtpd and chan_btp rock my nuts off
21:18.16endermishehu: I'm using a Tyan S2882...
21:18.22hardwirebut they need to be re-emplimented thats for sure
21:18.25PoWeRKiLLHow not to pass audio when dialing SIP -> * -> ZAP ?
21:18.39pbdHmm. You're saying that Brazil PSTN uses loop start signalling?
21:19.10pbdAcertain- have you implemented a Sipura down there?
21:19.10funxionweird
21:19.13*** join/#asterisk akimbo64 (n=akimbo64@193.251.169.129)
21:19.23akimbo64Hello world
21:19.35acertainhello
21:19.40funxionanyone know why I would get "No application 'Set' for extension (a" isn't Set included by default?
21:19.42akimbo64Can somone help on installing or compilling chan_spy please
21:19.44*** join/#asterisk likwid-- (n=likwid@nc-71-0-190-14.dyn.sprint-hsd.net) [NETSPLIT VICTIM]
21:20.07mishehutzanger: really?  when did that happen?
21:20.11acertainpbd: no just Digium cards (TDMXX, TE110P, TE410P)
21:20.15mishehuender: single or dual proc?
21:21.06acertainpbd: but I was having problems until use fxs_ls
21:21.23acertainpbd: rest works on defaults
21:21.28endermishehu: dual
21:21.41endermishehu: I have a dual-port T1 card in the PCI.
21:21.46tzangermishehu: not too long ago, I was talking to kp on differnet things and it came up that they'd identified a major source of PCI NMI errors
21:21.54pbdacertain- hmm.  I'm not sure I can beat the Sipura into doing it this way.  What's odd is that a standard POTS line phone in the FXS port works fine, but not the FXO.. which leads me to believe there's something I need to jiggle on the sipura- but I don't think I can change the base signalling.
21:21.58akimbo64help on chan_spy please
21:22.18mishehutzanger: ooh, I'll email somebody then
21:22.21tzangermay not be the exact issue you are talking about but call 'em and see
21:23.42KranZender: you run sata?
21:24.02enderKranZ: I use a 3ware card to do Sata RAID.
21:24.23KranZim running a tyan k8sdpro and had to buy a pci sata card b/c the onboard was on the same bus as the legacy pci slot
21:24.27Nugget3ware is good stuff.
21:24.30KranZit was slippin like crazy
21:25.54enderNugget: for certian things.
21:26.01mishehutzanger: I hope matt fredrickson is still there, as he was previously dealing with my issue...  though it's close to a year ago
21:26.18*** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net)
21:26.20SpaceBasshowdyt
21:26.20mishehuI find that emailing for non-critical issues is better when it comes to dealing with digium.
21:26.24SpaceBassor howdy...
21:26.32acertainpbd- you must, but if not you just have an option... read manuals and ask for tech suport to your dealer
21:26.47SpaceBassi'm having problems with DISA not recognizing tones well... is that common?
21:26.55acertainpbd- sorry if my help was not enough
21:27.08acertainpbd- bye, see you later
21:27.10akimbo64ChanSpy please !!!
21:27.30Nuggetender: what things do you feel 3ware would not be good for?
21:28.00blitzrageper chance, anyone here going to Astricon, who's leaving on Sat. Oct. 15th and heading to Union Station ?
21:28.12enderNugget: Database servers, high random-access servers, very low latency I/O servers.
21:28.20mishehutzanger: need to swap out this old dual proc pIII-800 for something a bit more modern...  I get some skipping and crackling that I doubt are caused by the internet link, and it simply cannot keep up with the codecs and pri...
21:28.24queuetueOther than bad passwords, what causes 401 (unauthorized) SIP errors?
21:28.34NuggetI'm quite happy with my 3ware database servers.  What do you use instead?
21:28.44enderNugget: Scsi.
21:29.04Nugget<PROTECTED>
21:29.04enderNugget: 3ware chokes on the highly random I/O, especially Raid 5 setups.  It couldn't even come close to handling our DB needs.
21:29.11Nugget*nod*
21:29.36Kattymister Nugget
21:29.53NuggetI've never done raid5 on a database.  Just raid1 or raid10
21:30.00Nugget(with 3ware)
21:30.01tzangermishehu: I find that hard to believe.  are you sure it's not just a shitty network card
21:30.05mishehuender: what 3ware cards were you using in your db setup?
21:30.11enderNugget: I used to be the main Linux engineer for Pogo Linux, and we sold shit-tonns of 3ware based systems.  I've seen them used in all sorts of stuff, so I have a pretty good concept of what its good for and where it fails.
21:30.17mishehutzanger: eepro100 ?
21:30.24NuggetI've also never used 3ware with linux.
21:30.32tzangerthat's a good card
21:30.37tzangera dual p3 should have lots of power
21:30.42endermishehu: various cards ranging from 7000 cards through 8000 cards and into the 9000 cards as well.
21:30.47tzangerwhat all are you doing with it under load?
21:30.55endermishehu: don't use eepro100 modlue.  Use the 'e100' module.
21:31.01endereepro100 is shit code.
21:31.05*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
21:31.09mishehuender: I have been using the e100 module for ages
21:31.17enderoh ok.
21:31.28Kattys/let/le
21:31.44mishehutzanger: a dual p3 800 doesn't have as much power as it used to...
21:31.51mishehuthe world moved on...
21:31.52mishehuheh
21:32.32tzangermishehu: :-)
21:33.54mishehuchatty Katty
21:36.57Kattymeeeshawhoo
21:37.36Kattymishehu: you know i'm not chatty in Real Life(tm)
21:40.15blitzrageReal Life (patent pending)
21:41.05Kattyblitzrage: heh.
21:41.44blitzrageKatty: wuz up
21:41.54queuetueThe sipuras inside the firewall (on the local lan) seem to have no problems connecting.  The ones coming from other locations seem to be getting 401 errors.  Is there something I need to do to give them access?  Something related to NAT?
21:44.03*** join/#asterisk akimbo64 (n=akimbo64@193.251.169.129)
21:44.08KranZmishehu: it could be an IO bus problem
21:44.33KranZhow much traffic do you have?
21:44.54*** join/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
21:45.18RickTick<PROTECTED>
21:45.25ManxPower~docs
21:45.28jbothmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
21:45.28ManxPower~mailinglist
21:45.30jbothmm... mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
21:45.43ManxPowerRickTick, the Wiki has the info, as well as a google search
21:46.26RickTickManX: I wanted to know the reliabilty of the Sipura's ?
21:46.57ManxPowerRickTick, they seem reliable.
21:47.04ManxPowerUnless you have them on an unreliable internet connection....
21:48.20toddf~hardwaredocs
21:48.28RickTickManX: kool ... I hope they can withstand ... 95 deg .. temp ..16 hrs a day. Thanks
21:48.40ManxPowerRickTick, don't count on it.
21:48.40KranZdoubt it
21:49.02ManxPowerThey are designed for home/office use, not high temps
21:49.06*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:49.11akimbo64Can someone help on compilling chan_spy please
21:49.25ManxPowerakimbo64, it's inclided with recent versions of Asterisk
21:49.28RickTickhmmm...
21:49.30ManxPowerpost 1.0.x
21:49.46*** join/#asterisk TommyTheKid (n=tommy@dsl-206-53-27-254.den.pcisys.net)
21:49.57KranZdont go dlink either
21:50.02KranZthey idle at 95degrees
21:50.19ManxPowernetgear is the worst
21:50.20akimbo64asterisk 1.2 has some bugs and it is not included with latest stable release
21:50.21toddfany pointers to hardware docs of digium hardware (TDM400P is what I have)
21:50.29*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:50.41KranZvoip-info.org
21:51.24hardwireso realtime sip peers == no MWI?
21:51.29hardwirethats no bueno
21:51.39hardwireand of course no notify.
21:52.14RickTickany idea of a decent price 2 port fxo device ,,, that can withstand 85-95 deg temp ..(in a sheltered room)?
21:52.57*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
21:53.03GivurQuestion, when I convert the MP3s to RAW Files I not use to install this ztdummy stuff?
21:53.08hardwireRickTick: maybe you should buy a fan
21:53.14ManxPowerRickTick, all FXO devices except SIPura are expensive.
21:53.20hardwirebut most sip ata's should work just fine in that temp
21:53.25generalhanwhats going on everyone ?
21:53.28ManxPowerRickTick, Honestly it would be cheaper to put an A/C unit in the room
21:53.33hardwireconcidering your computer processor is 120 most of the time :)
21:53.48KranZwith a fan
21:53.57ManxPowerhardwire, he wants stuff to work without testing it first.
21:55.19generalhanI need some help with my Cisco 7960s, can anyone tell me how to get the time and date to display on the phone? it used to do it the first time i plugged them in, and now none of them will show it.
21:56.06KranZmaybe you've entered some sort of temporal vortex
21:56.15toddfKranZ: voip-info.org contains programming info?  I want to write a bsd licensed driver for the digium hardware I have ..
21:56.15generalhanwow !
21:56.29hardwireManxPower: so do I.. but it doesn't work
21:56.39akimbo64I'v already an asterisk 1.0.9 server running and i want to add chan_spy capabilities, is it possible without reinstalling ??
21:56.43KranZoh, dunno toddf :/
21:56.44argos73kill me.  now.
21:56.56Kattybyebye.
21:56.59Ariel_~kill argos73
21:57.01jbotACTION shoots a magneto-ionized  electron gun at argos73
21:57.28argos73been fighting with this sip registration problem for three days, and it looks like it's debian screwing up the UDP checksums
21:57.35PupenoLSIP is a protocol used for starting an RTP sesion between two ips (and other similar things), right ? RTP is http://www.ietf.org/rfc/rfc1889.txt ?
21:58.03distortiongeneralhan: i have the same issue on my 7940's, you using sip?
21:58.09generalhanyes i am
21:58.30distortionhave you looked through the SIPDefault options?
21:58.33generalhanbut even the first time i updated the firmware for SIP it showed the time and date. now for some reason its doesnt
21:58.43KranZargos73: what are you registering to?
21:59.30distortionyeah, it does that on all my phones after a hard boot
22:00.02argos73KranZ: recent * on new Debian box...  works fine when registering to same * version/config on a slackware machine
22:00.18*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
22:00.39KranZyou know of ngrep right?
22:01.43argos73yup...  something's messed up on the debian box causing it to send out udp sip responses with incorrect checksums in them, which the phones then ignore
22:01.54mishehuKranZ: not much.  the server is currently used mostly just by 3 people for a php app, qmail, dspam, and mysql...  that shouldn't be too much load.
22:02.18KranZmishehu: sounds like you have IO timimg issues
22:02.31*** join/#asterisk fugitivo (n=ajf@201.255.102.89)
22:02.41KranZmake a call and run "hdparm -t /dev/hda"
22:03.00KranZif you get an ass load of slips then that's the problem
22:05.32pbdHey- anyone out there seen problems with jitter kicking up when IAX trunking is enabled?
22:05.36RickTickManx: the Sipuras will be going to a place with ac .... in a house in the caribbean ... so I don't know exactly the conditions. I am just trying ... to set my expectations
22:05.48RickTickManx : without AC
22:05.50*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
22:05.57mishehuKranZ: Timing buffered disk reads:   66 MB in  3.06 seconds =  21.54 MB/sec, which isn't probably too bad for a 4 or so year old machine with an old megaraid raid5 controller on it...
22:06.08ManxPowerRickTick, try it an see.  Check the specs on the sipura web site, they normally list operation temps
22:06.14KranZdid you get any errors in "asterisk -rvvv" ?
22:06.19KranZor hear them
22:07.08mishehuKranZ: no errors showing up on asterisk, but I can't vouch for the static yet
22:07.15mishehuI'd have to call in and rerun it
22:07.20KranZargos73: gentoo
22:07.30KranZmishehu try it
22:07.38*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
22:07.50KranZor watch the asterisk console for "HDLC ABORT"
22:07.57KranZyou're running a pri right?
22:08.01*** part/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net)
22:08.27justinui need to figure out how to get aserisk to not hangup on the calee when the caller hangs up... i want the callflow to continue and play some prompts to the callee after the caller hangs up
22:08.48mishehuKranZ: I get the HDLC aborts every so often when the PRI is idle heh.
22:08.52KranZyeah
22:09.07KranZmove your pri card to a different pci slot
22:09.08KranZi had that on my dual opteron 244
22:09.25KranZhad to buy a pci sata card b/c the onboard was using the IO bus as my PRI card
22:09.37ManxPowerakimbo64, chan_spy was never updated for later versions of 1.0.x.  If you want it to work you'll have to spend some time patching it or re-writing parts of it
22:09.37KranZand i only have 1 legacy pci slot, so moving it wasnt an option
22:09.54peggerjustinu why would you not want it to hang up?
22:10.11mishehuKranZ: no correlation between disk io and static.  no hdlc aborts...
22:10.13justinubecause i want to play some prompts to the caller
22:10.29justinutelling him the other guy hungup or something
22:10.36peggerbut they already hung up so how are they going to hear them
22:10.46KranZyour onboard nic might be on the same bus
22:10.51justinuthe called has hung up
22:10.55ManxPowermishehu, static?
22:10.58justinuthe calle hasn't hungup yet
22:10.59KranZi still think moving the pri card will fix your problem
22:11.07ManxPowermishehu, sounds like frame slips
22:11.14*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-237.cybersurf.com)
22:11.25KranZmove it like 2 spots one direction, sometimes adjacent pci slots are on the same bus
22:11.46ManxPowerif he's not getting HDLC aborts, I doubt moving the card will help.
22:11.50GivurYippie :)
22:11.51KranZhe is
22:11.56GivurThe Music works :)
22:11.58Givur*dances around*
22:12.01mishehuManxPower: sometimes static, mostly choppy sound.  Li--ts be--jammed
22:12.16mishehuand this line shouldn't be dropping packets, it's a frac ds3.
22:12.33KranZyou could also disable VAD on the client side
22:12.33justinupegger: what I'd like to happen is when Asterisk receives the BYE from the CALLED phone, i'd like it to not pass it on to the CALLING phone
22:12.41mishehuKranZ: doubtful it's the same bus anyway.  the e100 is onboard
22:12.55KranZyeah, but you'll have more than one bus
22:13.04DrukenManxPower: feel like helping my dumb ass running asterisk as a common user instead of root?
22:13.10ManxPowermishehu, sounds like your timing source might be wrong
22:13.19mishehuKranZ: asterisk itself support VAD on IAX2?
22:13.34KranZwhat are your callers using?
22:13.34ManxPowermishehu, ASTERISK DOES NOT SUPPORT VAD
22:13.45mishehuManxPower: I didn't think it did.
22:14.07ManxPowermishehu, you have "1" in the timing field of your span= line, right?
22:14.09mishehuManxPower: let me check.
22:14.20KranZVAD will stop sending info when the caller doesnt talk, so Asterisk wont send back b/c it uses the client as the source for RTP timing
22:14.34KranZwhich is annoying
22:14.45ManxPowerKranZ, Not an issue with IAX
22:14.54ManxPowerSince no IAX/IAX2 device supports VAD
22:15.11KranZtrue
22:15.20KranZi havent messed with iax, only sip/mgcp
22:15.22mishehuKranZ: it's just answering a DID, and piping the call to another asterisk box's menu system via IAX anyway.
22:15.27mishehuManxPower: span=1,1,0,esf,b8zs
22:15.35ManxPowermishehu, I know of at LEAST two people that had issues until they turned off their onboard LAN and installed a PCI Ethernet card
22:15.42mishehuManxPower: the problem is intermittent.
22:15.55KranZmishehu: or move the card to a diff slot will have the same effect
22:16.05mishehuManxPower: well, that's easy enough to test, I have spare e100 cards around...  could test it.
22:16.06ManxPowerKranZ, not always
22:17.03KranZsounds like it accomplishes the same thing
22:17.13mishehuKranZ: not on all motherboards.
22:17.23justinuis there any way I can use the "h" extension to play prompts to the CALLING party after the CALLED party has disconnected?
22:17.26ManxPowerKranZ, no.  some cards/drivers lock interrupts for a long time, movig the card will not fix that
22:17.29justinuor some other way to do it?
22:17.44mishehuand it is a digium card we're talking about ;-)
22:17.48ManxPowerjustinu, see the "g" option of "show application dial"
22:17.48KranZManxPower: im talking bus, not irq
22:18.00ManxPowerKranZ, I kow, and the PCI bus has LOTS of bandwidth
22:18.17*** join/#asterisk phpboy (n=shane@c1-118-6.tbnb.isadsl.co.za)
22:18.19justinumanx: thanks
22:18.21KranZManxPower: i just fixed this same problem with pri slipping and timing errors
22:18.21snittvad means voice activated dial?
22:18.27phpboyHey all... I was wondering... I've got 10 numbers assigned to my ISDN line... how do I specify which number or those 10 numbers I dial out onto the pstn from?
22:18.52KranZdisabled the onboard sata b/c it was on the same IO bus as the only pci slot i could use for the digium card
22:18.52ManxPowerKranZ, and the card was not sharing interrupts before you moved it?
22:18.55mishehuphpboy: should be as simple as SetCIDNumber
22:18.55KranZno
22:19.02mishehuat least that's how it works for me on my PRI
22:19.18KranZthe only reason i had to disable was b/c i have 4 64bit slots and 1 32bit
22:19.24KranZso i couldnt just move the digium card
22:19.29peggerphpboy  use the talk  john@lain
22:19.29mishehusnitt: VAD == voice active detect I believe
22:19.40snittmishehu: ok, thx
22:19.42KranZbut is mishehu's case, he might be able to get away with trying a diff slow
22:19.46KranZslot
22:19.54ManxPowerhe should try every suggestion
22:19.59KranZtrue
22:20.13KranZbut adding a new piece of hardware should be option 2
22:20.17phpboymistral: can I set it for each extention?
22:20.51ManxPowerphpboy, callerid=blah <1234567890> in each device's config in asterisk
22:20.53blitzrageany recommendations for a book on javascript for web development with PHP ?
22:21.06KranZheh
22:21.14*** part/#asterisk epablo (n=epablo@200.75.139.188)
22:21.45KranZblitzrage: that sounds like 2 books
22:21.57mishehuManxPower: you don't think it might be a problem with trunkingtimestamps in IAX2?  I am using 1.2.0b1
22:22.27phpboyhow can I specify which channel in a group I dial out of?
22:22.37KranZManxPower: why dont you try eliminating the 2nd * box and add the same ivr to the main one
22:22.44KranZif the slips stop, then it's an iax prob
22:22.49mishehuphpboy: in isdn, it shouldn't matter which channel you dial out
22:22.52KranZif not its a pri prob
22:22.55blitzrageKranZ: not really -- would make a good book on the best way of using a combination of server side and client side scripting
22:23.26blitzrageKranZ: with the focus being on javascript (client side) and just using PHP (server side) to draw the initial screens and develop the forms etc...
22:23.33phpboyok... then how do I link callerid= to an extention?
22:23.58*** part/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
22:23.58mishehuwell, I'll try switching around the cards  on friday.
22:24.00phpboyso if I dial from extention 10 I can specify which number I'm calling from to the pstn...
22:24.05pbdManxPower: The people who wound up changing out the onboard NIC for a card.. care to elaborate on the symptoms they were seeing?  zttest errors?
22:24.07mishehuthankfully the system is still being tested.
22:24.28ManxPowerpbd, I don't recall.
22:24.59ManxPowerpbd, I think it might have been HDLC Abort errors
22:25.38*** join/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net)
22:25.43KranZzttest would should less than 99.965
22:26.05KranZand the * console would shot hdlc abort (6) errors
22:26.09KranZshow
22:26.25phpboyManxPower: how would I go about doing this?
22:26.46pbdManx: I've got a weird case where zttest is clean, no HDLC errors, but some ocasional dropped voice.  Network appears clean, too- so I'm focused on the client ends- but it's a case of an onboard NIC and a TE410P.. so it might be a match.
22:26.58WillySillywhats the Asterisk Manager Interface in 1.2.0-beta1?
22:27.05phpboyI see the example for setting the callerid in zapata.conf... but no example for linking those caller id's to an exten
22:27.10KranZpbd: do a cat /proc/interrupts
22:27.15pbdzttest is generally above 99.975.
22:27.26KranZer
22:27.30KranZcat /proc/zaptel/1
22:27.43KranZdoes it have a "misses"?
22:27.47ManxPowerphpboy, what type of phone?\
22:27.48*** join/#asterisk acidfoo (n=acidfoo@66.11.160.156)
22:28.11phpboyManxPower: I'm using swissvoice SIP phones and X-Lite soft phones
22:28.18Ariel_phpboy, callerID="Joe Blow" <123456789> in the sip.conf for the device or the iax2.conf or zapata.conf
22:28.21mishehubah, now I have to lookup zttest's params, as there's no --help
22:28.23pbdI've confirmed, the boards do not appear to be sharing interrupts with anything.   I'm not in front of the machine, but I'll dig into the proc/zaptel stuff later.
22:28.41KranZmishehu just run zttest
22:28.46KranZ./zttest
22:28.48Ariel_cat /proc/interrupts
22:28.50phpboyAriel_: so I should put the full number in.. not the internal exten?
22:29.03Ariel_phpboy, your choice
22:29.17mishehuKranZ: it opens a virtual zap device...
22:29.33mishehuerr pseudo zap interface
22:29.43phpboyAriel_: I did that... it still dials out from the default number in the number range :<
22:29.48Ariel_in outbound dialing rules you can also set callerID depending if your using a voip provider or pri setup
22:30.04Ariel_dials out what?
22:30.38phpboyAriel_: dial out to the pstn... using pri ofcourse
22:30.46ManxPowerphpboy, then it would be the callerid= setting in sip.conf for each device.
22:31.03KranZmishehu: also, disable onboard usb and "assign irq to vga" in the bios
22:31.25mishehuBest: 100.000000 -- Worst: 99.987793 -- Average: 99.988084
22:31.29WillySillywhats the Asterisk Manager Interface in 1.2.0-beta1?
22:32.07*** join/#asterisk jarnaud (n=jarnaud@216.143.37.5)
22:32.11jarnaudHi all
22:32.30phpboyManxPower: for each user/exten ?
22:32.36jarnaudcalling a AGI and providing 2 args: is it AGI(myphp.php|arg1|arg2) ?
22:32.38ManxPowerphpboy, for each device.
22:32.42ManxPowerextensions are NOT devices
22:32.52KranZphpboy: user
22:33.33justinuhey manx: thanks for the tip w/ the "g" option
22:33.42generalhananyone in here using Cisco 7960s with Asterisk ?
22:33.46KranZphpboy: http://www.voip-info.org/wiki-Asterisk+config+sip.conf
22:33.48justinuone more question: i'm calling a macro when I dial to an extension
22:33.54justinuwith the M(macro) option
22:33.58KranZphpboy: look at the examples
22:34.27justinui'd like the call flow for the CALLING party to be different after the CALLED party hangs up, depending on WHEN the CALLED party hangs up
22:34.32phpboythanks KranZ :D
22:34.53justinufor example, my macro plays a prompt to the CALLED party right after he answers... if he hangs up during that prompt, I'd like to have the CALLED party hear a prompt
22:35.03justinuoops, i'd like the CALLING party to hear a prompt
22:35.26justinuif the CALLED party hangs up after the prompt, but after the call has been bridged, the call can tear down normally.
22:35.35KranZjustinu: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
22:35.40KranZlook at option "g"
22:35.45justinui've got that part down
22:35.58KranZmmk
22:36.06justinuis it as simple as setting VARs in the macro and then using GotoIf in the context that calls the macro?
22:36.11*** part/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net)
22:36.29KranZso under the dial command, put the commands you want to execute after the hangup
22:36.36jarrodhey is there a way to remove the digit from the end of an extension like ${EXTEN:1} removes it from the beginning?
22:36.39ManxPowerjustinu, maybe.  try it.
22:36.55justinualright.
22:37.11KranZS{EXTEN:$LEN(${EXTEN})-1:1}
22:37.11ManxPowerjarnaud, see README.variables
22:37.15jarrodKrazy
22:37.16KranZsomething like that
22:37.18jarrodyou are the man
22:37.39justinugeneralhan: i know someone that uses cisdo 7960s
22:37.44justinuw/ asterisk
22:37.58KranZ${EXTEN:$LEN(${EXTEN})-1:1}
22:38.05pbdJustinu: So do I.  Me. :)
22:38.15generalhanwell i am using 15 of them in my office, but the time and date wont show on them anymore, and i just need to know if i HAVE to have an SNTP server in order to get that to work ?!
22:38.32ManxPowergeneralhan, Yes
22:38.33justinui dunno about SNTP, but I think they need NTP
22:38.37justinuthe cisco phones kinda suck
22:38.43generalhani LOVE these phones
22:38.52jarnaudManxPower:thanx, where can I find README.variables?
22:38.55*** join/#asterisk dos000 (n=dos000@i216-58-27-16.cybersurf.com)
22:38.58ManxPowerWE just point all our phones to our local cisco router that gets it's time from pooo.ntp.org
22:39.00justinucan you make them do early (overlapped) dial?
22:39.02generalhanthey were definately better on the SCCP protocol but they are still pretty nice phones in SIP too
22:39.09ManxPowerjarnaud, asterisk/docs
22:39.17mmlj4hey manx
22:39.28mmlj4i see you got some internet, cool
22:39.30jarnaudthx again
22:39.40generalhanManxPower: what if i dont use a cisco router ! lol
22:39.46pbdI've had the converse true- the things wanted to self-boot all the time in SCCP, they've been nothing but solid under SIP for me.
22:40.00*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
22:40.14pbdThere was a mention, back about a month ago, on -users, about the time dissapearing.  Wasn't the protocol, however- I just can't remember what it was.
22:40.23generalhanhmm
22:40.30NuggetOne of my 7960s needs to be rebooted daily to scare away the periodic lockups
22:40.31pbdIf you google for it, it should be there.
22:40.38Nuggetbut I think it's just a flaky one.  I bought it off ebay
22:40.44pbdNugget: What version of firmware are you running?
22:40.47Nugget7.5
22:40.49generalhanyeah of all the things that employees can bitch to me about, the time and date seems like it should be the easiest for me to fix lol !
22:41.14pbdNugget: so am I- and no lockups.  I'd suspect power issues, too.
22:41.24generalhanNugget: i had the EXACT same problem with 7.5 ... i downgraded to 7-3-00 and havent had one problem !
22:41.32NuggetI tried swapping power bricks with another phone and the problem didn't follow the brick.
22:41.44Nuggetthe phone has had the problem since I got it, circa 7.2
22:42.01generalhanwell thats no good !
22:42.05NuggetI just wrote a script in cron to reboot it at 4am every day and put it in my guest room
22:42.08Nugget:)
22:42.10pbdI've been running 7.5 for about a month now.. no issues.  I think there's various odd issues with hardware revs, myself- I have one that refuses to save it's ringer volume.  Period.
22:42.14generalhanbut does it show you the time and date in the top left corner !? lol
22:42.14KranZwith those phones, can you see if some1 is on another line?
22:42.26justinuhow are the grandstream gxp-2000s?
22:42.32pbdKranZ: Which phones, the 7960?
22:42.38KranZyeah
22:42.49phpboynow I can't connect to asterisk through my windows VPN
22:42.55phpboythat's kinda useless :<
22:42.57KranZand do intercom?
22:43.12generalhani could do line apperences with the SCCP ver. 7960s. i dont know how with the SIP
22:43.37pbdUnder SCCP, yeah.. haven't messed with it under SIP in some months.  Logically, no- unless Asterisk is good at sending out the SIP headers.. I know I can't register the same extension on mulitple 7960's, weirdness abounds.  I could write a service to show it, however...
22:43.44generalhani assume that if you could get the expansion module to work with with asterisk you could show line apps.
22:44.10pbdThe sidecar isn't supported to my knowledge under SIP on the 7960.  And it's only barely there for SCCP.
22:44.13*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)
22:44.17KranZpbd, you tried SER?
22:44.18*** join/#asterisk zotz (n=zotz@24.231.36.100)
22:44.46generalhanpbd: i have used them with SCCP and it worked VERY nice. now that i switched everything over to asterisk i havent gotten it to work
22:44.48pbdKranz-  Not personally, no.  There's a chance that would allow the right headers to go out, true.
22:44.56generalhanits pretty low on my to-do-list though
22:45.04KranZalso does multiple reggies
22:45.13ManxPowerYou don't register extensions, you register devices.  An extension is just a dialable way to call a device.
22:45.20pbdGeneralhan: Under CCM, they are much better under SCCP than SIP, no question.
22:45.46KranZSER you can emulate registering extensions
22:45.46ManxPowerWe name our devices after their MAC address with a -a|b|c|d, etc suffix for multiple "lines" on a device.
22:45.59pbdManx: You've not spent enough time under the Cisco flag yet. : )  In the sip configuration, you tell the phone what extensions it should register back to Asterisk.
22:46.04generalhani would rather just figure out why the time disapeared and how to get it back !
22:46.24ManxPowerpbd, they may call them extensions, but they are not.  They are line appearances
22:46.33pbdMultiple lines is no problem.. the 7960 is good with that.
22:46.49generalhanVery god
22:46.51pbdSemantics.  They are what they are- and they don't work that way.
22:46.55generalhanVerry good rather
22:46.55knight_any suggestions on connecting asterisk behind NAT to remote H323?
22:46.58ManxPowerpbd, yes, the cisco does a trick where it only registers one line appearance no matter how many the phone actually has.
22:46.59KranZi'd like to test a 7960, but $200 is pricey
22:47.07*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
22:47.11ManxPowerPolycoms can do that too, BTW.
22:47.34generalhanwoah where did you find a 7960 for $200 ?
22:47.49pbdKranz: It's worth the bucks- it's a solid phone.  Polycom is very much the same (no surprise, the original DSP code for the 79XX was written by Polycom.. go figure)
22:48.07KranZebay
22:48.07ManxPowerWe have man extensions that have to do things like ring on line appearnace 1 on device X and line appearance 4 on device y and line appearanc 2 on device z
22:48.12generalhanwe paid like $350 for every one of our 15. that was like a year ago, but still
22:48.35ManxPowerKranZ, on ebay don't expect to get the SIP firmware already installed, you have to buy that for > $100, or a power supply, about $45
22:48.38Hogiedamn generalhan... we paid $250 for ours a year ago
22:48.41pbdGeneral: The price hasn't changed much, especially if you don't have cisco prestandard POE handy.
22:48.49KranZManxPower: hmm
22:48.58KranZsome1's selling the 7.5 cd
22:49.02KranZfor $12
22:49.03*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
22:49.20pbdKranz: They are, but it's illegal- and Cisco has a lot of lawyers.  Pay cash.
22:49.23generalhanwell the phone was like $300 when we bought them, then they got us for $45 for the power cube
22:49.50generalhanthen another $35 if i wanted the licenses with each phone !
22:50.01KranZrip
22:50.25pbdTechnically, a phone without a license is grey market- Cisco doesn't want their resellers selling them.  They do, but with a lot of funny looks.
22:50.51generalhanwe just hired about 15 more people to work in the office so i just got the cheapest phones i could, the Aastra 9112i, and they work great !
22:51.06fugitivoa phone with a license is a black market
22:51.13ManxPowerAnd Cisco's firmware license does NOT allow transfer of ownership of the license.
22:51.13fugitivophones shouldn't have licenses
22:51.29generalhanif i knew a year ago that i would be going SIP i would have ogtten those instead of the Ciscos and saved a grand
22:51.34ManxPowerThat is why we went with Polycom.
22:51.57peggeri was reading the documentation for the Read() Application and it said that Max accepted value is 255.  so that means i cant pass it a 10 digit number correct?
22:52.11KranZyes
22:52.24ManxPowerpegger, that's 255 characters, I believe
22:52.39pbdPolycoms are good phones, no doubt.  I'd hate to have to rewrite my services for them (different XML dialect altogether), but the quality is there in either unit.
22:52.59pbdAt the end of the day, supposedly, Cisco will drop their price to match Polycoms- but you have to know your distributor.
22:53.07generalhanyea and ill bet my time and date will show up on the polycom ! lol
22:53.11ManxPowerPolycom only supports XMl on their high end IP 600 phone
22:53.14generalhansorry im really bitter about that right now
22:53.15FuriousGeorgeanyone using gentoo find that from time to time their init.d script for * stops working?
22:53.21KranZRead([variable],[soundfile],[numdigits])
22:53.24fugitivoFuriousGeorge: no
22:53.30fugitivoFuriousGeorge: check permissions
22:53.31ManxPowergeneralhan, only if the phone can contact it's NTP server
22:53.32GivurSo... Good night all
22:53.34Ariel_Polycom includes the Sip license which Cisco does not. you need to pay extra for it.
22:53.40pbdManx: I know.  Cisco supports it on the '40 and on the '60, which is nice.. but the price isn't that different from the IP 600.
22:53.48generalhancan i do some kind of completely free NTP server ?
22:53.58KranZgeneralhan: they're all over
22:54.04KranZmany universities have them
22:54.08ManxPowergeneralhan, you mean like is inclided with every distro of linux
22:54.21pbdgeneral: Absolutely.  If you've got an asterisk server, you have all the software and hardware you need- check out the ntpd package.
22:54.40FuriousGeorgefugitivo: nothing wrong with my perms
22:54.46pbdI've found them to be tricky to set up- but not THAT tricky.
22:55.01*** join/#asterisk zapa (n=zapa@200.52.223.11)
22:55.05KranZhttp://www.polycom.com/products_services/1,,pw-34-182-12072,00.html
22:55.06KranZwow
22:55.12KranZthat's a serious phone
22:55.26*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
22:55.43peggerMaxpowers so it is max value of 255 or max spaces of 255?
22:55.52Cresl1nland lines are such a thing of the past
22:55.54pbdI'd need a sidecar for my desk to run that beast.
22:56.15KranZpegger: are you reading digits from a phone?
22:56.24pbdMy understanding is that the fine print with that picture says it's not available yet.. but coming soon.
22:56.39jontowman, having a hell of a time getting my pri up on freebsd :/
22:56.51KranZpbd: i'd hate to have to man that phone
22:56.53Cresl1nlinux is the answer my friend ;-)
22:56.54*** join/#asterisk marc324 (n=marc3234@206-248-159-4.dsl.teksavvy.com)
22:57.52zapahi all, does any one has trouble with Agents when transfer with #, it's too erratic, sometimes work some times don't and the agent still busy any clue ?
22:58.12kippiwhats the channel for asterisk @ home?
22:58.19pbdThe person who runs the switchboard here can't handle a PC based attendant console- says it's too complex, she'd rather just remember everyone's number (85 people), or look them up on the paper copy of the phone book she prints each month.  She'd have an attack with that phone.
22:58.29*** part/#asterisk opus__ (n=opus@dahphish.org)
22:59.32*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
23:00.57afrosheenhand her a box with FOP on it
23:01.02afrosheenit's moron-proof
23:01.08FuriousGeorgeits odd.  there seem to be two init scripts floating around for gentoo.  one that checks for niceness and takes option and one that just starts asterisk.  eventually the more complicated one stops working and i cant figure out why.
23:01.23FuriousGeorgebut whatever
23:04.34*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
23:04.34*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now!
23:04.37afrosheenour new secretary loves FOP, started using it from day one
23:05.17Nyvaryeah but they are just gonna press 0 anyway
23:05.27marc324is there a way to learn asterisk without purchasing t1 card?
23:05.31afrosheenyeah well, you would think that your average fool could maneuver through an IVR but experience proves otherwise
23:05.40afrosheenthat's why zero is an option
23:05.45Jam1eyou know you can use .wav's for voice menu's etc, do I need to put .wav on the end of the filename, e.g BackGround(mainmenu.wav) as opposed to just mainmenu if it was a .gsm
23:05.49Jam1eor does it run it anyways
23:06.02Nyvarmarc, no need to purchase anything to use asterisk (assuming the computer is free)
23:06.24afrosheenI would recommend some kind of zaptel card at least so meetme is reliable
23:06.28marc324is it possible to emulate a t1 connection?
23:06.36pbdMarc: A $100 Sipura will get you a basic pstn connection.
23:06.44FuriousGeorgedid i hear someone saying that usb headsets were not better than standard analog headsets
23:06.46FuriousGeorgewas that in here
23:06.46pbdmarc: TDMoE looks pretty similar.
23:06.55FuriousGeorgeor was it the opposite
23:06.59Nyvaranyone know if the timer issue was fixed in meetme with the new rtp patch?
23:07.04afrosheenFuriousGeorge: personally I prefer standard over USB
23:07.21afrosheenNyvar: to fix the timing drift? is that still a problem after all this time?
23:07.22pbdI've been running meetme fine on the rtc module forever now.
23:07.40FuriousGeorgeafrosheen: you know what bothers me?  how do you heard the phone ringing?
23:07.44marc324pbd -- what card is it?
23:08.02Nyvarmarc, if you need to plug into a t1, you'll need a t1 device of some sort.  If you just want to try asterisk without any investment in hardware, use 2 sip software phones on pc and set up asterisk in the middle
23:08.03pbdI think the jury is still out on the headsets.  One client I have swears by them, but has had all sorts of problems.  I prefer the standard analog type and a sound card- but it depends on the sound card you've got.
23:08.24Nyvaror iax software phones even
23:08.31afrosheenFuriousGeorge: my speaker controller has mic and headphone jacks, I halfway unplug the headphone jack :)
23:08.41pbdmarc: TDMoE is cardless.  But it's not a T1 interface, it's just a way you can make your dialplan look like it is- if you need to check signalling, etc, you're back to a TE4XX card.
23:08.47FuriousGeorgeafrosheen: thats what we do
23:09.00afrosheenFuriousGeorge: it's an ugly but reliable solution :)
23:09.12Jam1eanyone? when using .wav's in "Background()" do I need to put .wav on the end, or will it work with just filename without the extension?
23:09.15FuriousGeorgeafrosheen: one thing i can say for these headsets, they have a switch to turn the earpart on or off
23:09.26pbd(ok, for the purist, TDMoE is not technically cardless- you need ethernet cards, and two PC's, and a single lan segment, and...)
23:09.29afrosheenFuriousGeorge: odd, most turn the mic on and off
23:09.38Nyvardont put the filename extension
23:09.47FuriousGeorgeafrosheen: i havent tested it yet.  if ur right im still annoyed
23:09.55*** join/#asterisk kippi1 (n=chrisfro@cpc4-hatf3-6-0-cust228.lutn.cable.ntl.com)
23:09.57Jam1ek so it'll be fine even though its a .wav as opposed to a gsm?
23:09.59kippi1hey
23:10.07afrosheenFuriousGeorge: eh..well what do you expect, manufacturers won't get with the trends
23:10.16kippi1is there a good web based SIP/Softphone
23:10.21Nyvaryes, background can use .wav, assuming that it was encoded in a way that asterisk is happy with
23:10.41Jam1ekk
23:10.47Jam1eI'll give this format a try
23:10.55afrosheenkippi1: web based? like you visit a page and it works? not that I know of..dialpad USED to have one
23:10.56*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)
23:11.29kippi1one that i can run on my box, so users can logon and just use the net to connect to there extension?
23:11.33FuriousGeorgedoes anyone know of an "earbud" style USB headset.  not bluetooth.
23:12.01afrosheenkippi1: haven't heard of anything like that but it may exist
23:12.09kippi1ok
23:12.14marc324is a T1 line required for verifying signalling?
23:12.21*** join/#asterisk MnxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
23:12.29afrosheenawesome, it's manx
23:12.40MnxPowerI don't suppose anyone recalls what one long scary beep means when you power on a PC?
23:12.43pbdkippi: it would have to be a java app.. a web page w/o java wouldn't be able to drive the sound card.. but it should be doable somwhere.
23:12.55pc2MnxPower - google "bios error beeps".
23:13.03afrosheenMnxPower: video?
23:13.04pc2MnxPower - it's out there.
23:13.06Nyvarmarc, what are you up to?
23:13.10pc2pdb - I think dialpad was activex
23:13.12pbdmarc: What kind of signalling are you trying to verify?
23:13.23pc2pdb - I'd love a java/activex sip phone.
23:13.24pbdpc2: Point.
23:13.29MnxPoweryou can do anythin in ActiveX, including format a HD
23:13.34pc2true :P
23:13.36pc2spyware too hehe.
23:13.39afrosheenor install lots of spyware
23:13.43afrosheenor nail.exe
23:14.26pc2pbd - Closest I found was http://www.sip-communicator.org/  -- it uses java web start and does require a local install, and isn't leightweight (7mb) but you might be able to rig it up.  IT's not an applet though -- it runs in java.
23:14.48*** join/#asterisk Malthus (n=admin@port0115-abt-adsl.cwjamaica.com)
23:14.50marc324what card is needed for calls from softphone to pstn.
23:14.59pc2marc324 - digium.com will tell you
23:15.01Malthushi
23:15.15pc2It's like $100 for one line.
23:15.28afrosheensounds like a party
23:15.31pbdmarc: Over what sort of PSTN line?  POTS? PRI? ISDN?
23:15.33pc2marc324 - For residential use, search ebay for asterisk.  Do not ever think of using that card for a business.
23:15.51Malthusis there a way to escape '@' symbols in callfiles (or the dialplan)?
23:16.17pbdpc2: Interesting.  I'll have to keep that one in my back pocket.
23:16.20FuriousGeorgeafrosheen: seems to say its for the speekers.  check the last bullet:  http://www.voipsupply.com/product_info.php?products_id=1017
23:16.25justinuhow about an FXO gateway?
23:16.34justinusip/fxo
23:16.41*** part/#asterisk cran (n=cran@vdp4094.ath04.dsl.hol.gr)
23:16.47marc324tdm400p?
23:16.48pbdJustinu: That would be a Sipura SPA-3000
23:16.59marc324a pots
23:17.05Malthusso that I could do like 'Local/user@domain@context' where user@domain is the extension
23:17.11justinuseems like that would be better than a PCI card
23:17.29pc2Anyone else using the vonage adapters unlocked with asterisk?  (linksys pap2)
23:17.45pbdMarc: POTS can be done with Sipuras, or a number of other ATAs.. or the X100P, or a TDM400p with appropriate FXO modules.. for beginners, I'd recommend the sipuras.
23:17.49Malthusiax.conf allows you to register with @ in the username
23:18.02Malthuscan't figure out how to call that user though
23:18.18peggeri am back
23:18.23justinuanyone tried the sipura phone?
23:18.31Nyvari have a sipura 841 here
23:18.36justinuany good?
23:19.19Nyvarno complaints.  It's flimsy, buttons are a bit rubbery to push, but seems to work ok, has speakerphone, costs $80
23:19.25marc324a x100p allows calls from softphone to be sent to a pots line?
23:19.29Nyvarmake that read no major compaints
23:19.32justinuyou think it's better than a budgetone 101?
23:19.38Nyvarhell yea it is
23:19.38*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
23:19.40justinuok
23:19.46justinuthanks
23:19.57enderSipura sound quality is ass.
23:20.04Nyvarender, agreed
23:20.09endercompared to something like a Polycom.
23:20.12Nyvarbut we are talking about an $80 phone
23:20.30justinui have an ancient pingtel xpressa
23:20.31justinuworks tho
23:20.32pbdmarc- Yes, it will.
23:20.32*** join/#asterisk anthonywms (i=anthony@12.108.5.202)
23:20.33enderyeah, but I'd gladly pay more to nto have that poor of sound quality.
23:20.36anthonywmshello
23:20.41Nyvarewwww xpressa
23:20.44anthonywmsi am wondering if anyone here has speakerphone problems with snom 190?
23:20.52justinuit was free
23:20.58Nyvari bet :)
23:21.07justinusound quality seems fine
23:21.29Nyvarsnom190's speakerphone use to have problems, are you running a later firmware?
23:21.32justinuso how's the grandstream gxp-2000? anyone tried it?
23:22.13anthonywmsi am running the latest firmware, and the snom 190 speakerphone goes mute if you talk, some sort of noise cancelation, i have 30 snom 190's deployed and im about to shoot myself they're being a real hedache
23:23.57Inv_arpmarc324: yes
23:26.34Nyvarmarc, there are clone versions of the x100p cards being sold on ebay, by one direct from digium if you need one, the clone cards have issues..
23:26.43shido6how do you view the line number with BBedit?
23:28.31shido6i found it
23:28.37shido6its in preferences ( I feel dumb now)
23:41.03*** part/#asterisk Malthus (n=admin@port0115-abt-adsl.cwjamaica.com)
23:42.45generalhancan some one tell me what this WARNING is all about ? i just got 4 of them in a row
23:42.45generalhanPhoneBox*CLI> Oct  5 16:43:06 WARNING[2399]: app_queue.c:388 play_file: ast_streamfile failed on IAX2/
23:44.18theblueHi all.
23:44.31justinuhello
23:44.36theblueIf I have an external SIP/IAX connection to a provider, does the username have to be a number?
23:44.52justinuno
23:45.00theblueBut the extension does.
23:45.17justinuare you making calls to the provider, or receiving?
23:45.26justinuthe extension doesn't either
23:45.29theblueMaking.
23:45.32theblueIts outgoing only.
23:45.35justinuok
23:46.15peggerwhere do you put sounds files to be played by Backround?
23:46.30justinuvar/lib/asterisk/sounds
23:48.07kippi1is there away to test your network for asterisk? like a test call thing?
23:48.19justinudial 1000, or 600
23:48.38kippi1nope
23:48.47*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
23:49.04FuriousGeorgekippi1: check the wiki for the echo test dialplan context
23:49.09justinuinstall the sample configs
23:49.10kippi1ok cool
23:49.20kippi1i am using asterisk @ home
23:49.30justinuoh, dunno about that
23:49.56generalhani wish i could get my asterisk to work at home. i try to connect to my asterisk box at work and i can receive voice but no one on the other end can hear me
23:50.00FuriousGeorgejustin u is 600 or 1000 some reserved extension i dont know about?
23:50.11pbdAfter fighting with Sipuras and Brazil all day.. I come to realize.. dummy... the IP address of the sipura changed- you know that, because you've been playing with it's web configuration all day.. but you never changed the sip.conf entry for it!
23:50.18justinuno, it's just something in the default extensions.conf
23:50.23pbdSome days you're better off staying in bed.
23:50.24justinuif you type make samples
23:50.33justinugeneralhan: sounds like a NAT issue
23:50.36generalhanpbd: i know what you mean !
23:50.38justinuare you using iax?
23:50.48*** join/#asterisk Hymie (i=hymie@L8R.net)
23:51.02generalhanyes i am
23:51.11justinuodd
23:51.12Hymieman, this channel has grown
23:51.32justinui've got my home box connected to the one at work via IAX, no problems
23:51.44*** join/#asterisk oliverqg (n=oliverqg@dsl081-096-215.den1.dsl.speakeasy.net)
23:52.25generalhanwell i am testing it with X-Lite, and everything works nicely, i can dial a 4 digit extension to get to someone at work. i hear them pick up but they cant hear me. i can make out going calls from home and its the same
23:52.32Hymiecan anyone tell me a way to make sure that a command is executed after a dial command?  I need the command to run no matter what happens, and no matter who hangs up...
23:52.39pbdI still have some disconnect tone stuff to work out, but I think I found those answers earlier today.
23:52.52justinuwait, you can't make outbound calls from the home pbx?
23:52.58pbdNow of course, I discover this 30 seconds after I send off a mailing to -users.
23:53.00justinusounds like xlite may be configured incorrectly.
23:53.00Hymieadding 'g' doesn't seem to cut it...
23:53.10pbdSo you all get to read about my dumb question later. :)
23:53.26oliverqgI'm using Agi GET VARIABLE command, but it always return 200 result 0 even with Predefined Channel Variables like CALLERID...  does anyone have a clue on what is happening?
23:53.29justinuhymie: how about the h extension?
23:53.37Hymiejustinu: sec
23:53.38justinutry putting your commands there
23:53.50justinubut for me, g works
23:54.10justinuyou put some NoOps there, and see if it's really not getting called?
23:54.14Hymieah, like ,h,
23:54.15Hymiehmm
23:54.25Hymieyeah... my noops work until after the dial command...
23:54.31justinuweird.
23:54.33Hymiesec, trying h
23:54.52*** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net)
23:55.32justinugeneralhan: if they can't hear you, the rtp stream you send is not making it to the asterisk box
23:55.42generalhanhmm
23:55.46justinucheck the SDP in your INVITE command
23:55.47generalhanok how do i go about fixing that ?
23:55.55generalhanuhh INVITE command ?
23:56.05justinuif it's sending to an address other than your internal PBX, you've got issues
23:56.11justinuturn on the diagnostic log
23:56.15justinufrom xlite
23:56.15kippi1is there away you can get the news streaming to your asterisk box, for use as hold music?
23:56.28justinuthat's a cool idea
23:56.35justinuprobably illegal tho
23:57.03pbdNot illegal if the news service licenses it to you.
23:57.07justinuyeah
23:57.10justinuobviously
23:57.19generalhanit is illegal. i work for a law firm and we were going to do something like that, but use streaming audio from a Jazz station, but he looked into it and its against copyright laws
23:57.32generalhanwell yeah if its licensed to you ! LOL
23:57.50justinuhan, i think your xlite is using STUN thinking it's talking to an outside proxy
23:57.55pbdIt's the same twist as using the sound card to bring in hold music (see the wiki), or using streaming internet radio for a hold music source (see the mailing list).
23:58.01thebluegeneralhan: Is it illegal to take text input, run it through a text-to-speech reader, and use that on an Asterisk box?
23:58.02*** join/#asterisk toddf (n=toddf@ns0.fries.net)
23:58.02Hymieheh, it's not illegal in Canada.. the US perhaps
23:58.12thebluegeneralhan: In the US?
23:58.24pbdI believe it's illegal everywhere- ASCAP forbids rebroadcast without royalties.
23:58.33generalhanyep
23:58.41pbd(ok, so enforce it in guatemala.. but in principle)
23:58.44theblueASCAP is icky.
23:58.51theblueBut is reading text illegal?
23:59.07Hymiein Canada, you can rebroadcast off air TV and radio signals legally.. our cable companies do not pay to re-broadcast
23:59.08justinuif you read it out verbatim, yes
23:59.22pbdNot really.  Expensive, yes.. but ASCAP is there to protect the rights of the people who make the music, so eventually they die in a house, not on a street corner.
23:59.40theblueTrue, but I think that ASCAP's gone a little too far.
23:59.48justinui'm with theblue
23:59.55*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)

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