irclog2html for #asterisk on 20050223

00:04.40thieumSdo you think I can handle 120 g729 SIp to zap transcoding with a bi-Xeon 3Ghz ?
00:05.08thieumSin the wiki, the guy succeed with 100 channels
00:07.32hardwireshit
00:07.35hardwireheh
00:07.43hardwirehow big a machine for just 24 channels
00:07.52hardwireI think buying a dual xeon system for 96 channels is just insane
00:08.06GroobyHP IPAQ??!?!
00:08.10thieumSfor transcoding ?
00:08.12Grooby613mhz ARM
00:08.13Grooby:-D
00:08.21stevekstevekTimex Sinclair...
00:08.26thieumSit is said transcoding is a CPU killer
00:08.27hardwirehttp://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm
00:08.29hardwirenow that would be nice
00:08.37*** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu)
00:08.38stevekstevek"near" real-time
00:08.43stevekstevekfor some defintion of "near"
00:08.45hardwireyou could use a 486 with that thing
00:08.52hardwirebehold the power of basic DSPing!
00:09.08hardwirewell
00:09.13hardwire240 channels on a 2.0 ghz celeron
00:09.25hardwireworks with *?
00:10.04Nuggethttp://bash.org/?464385  <-- heh
00:10.35thieumSdo you mean Digium cards are shit ?
00:11.14atmel:)
00:13.16wangsterAnyone know why ilbc -> anything is reasonable but anything -> ilbc is 13ms+ ? (according to "show translation")
00:13.31wangsterIs this due to ilbc's built in jitter buffer?
00:13.44*** part/#asterisk Grooby (~Grooby@66.160.105.186)
00:14.13*** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com)
00:14.18*** join/#asterisk tuxinator_linux (~anonymous@ip68-99-229-29.ph.ph.cox.net)
00:15.11ariel_hardwire, what are you trying to do.
00:15.44hardwirehttp://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm
00:15.45hardwire!!
00:15.50hardwirethats what I am trying to do :)
00:16.32KalD|Workis there any header (cvs etc) that states that libiax is lgpl?
00:16.46mtqhpyrosteve: BV does not want to help you...they don't support you
00:17.16DJ-Pyrohardwire: that's a crazy card
00:17.19ariel_hardwire, I would love to see that board actuall up and running.
00:17.30PyroStevemtqh: why are you speaking for them
00:17.44PyroStevemtqh: i support them, I pay them money everymonth
00:17.55PyroStevemtqh: 30.00 a month for bussiness line
00:18.06PyroStevemtqh: and more to come in the future
00:18.13PyroStevemtqh: what kind of customer service is that
00:18.19mtqhpyrosteve; read there website....THEY DON'T SUPPORT ASTERISK AT ALL
00:18.25mtqhThey give you a gudie
00:18.27mtqh*guide
00:18.29mtqhbut thats it
00:18.33*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
00:18.34yashaxGUYS: Grooby help me to get the * work with BV....... The issue for outgoing was that you HAVE to use sip.broadvoice.com and can not use their doc that says proxy.xxx.broadvoice.com
00:18.54yashaxBTW, BV so far works great. Crystal Clear
00:18.55PyroStevemtqh: I dont need them to support asterisk
00:19.05PyroStevei already have asterisk working
00:19.09PyroStevewith BV
00:19.51ariel_PyroSteve, here is there link for asterisk support: http://www.broadvoice.com/support_install_asterisk.html
00:20.08*** join/#asterisk denon (denon@synapse.subneural.net)
00:20.08*** mode/#asterisk [+o denon] by ChanServ
00:20.12PyroStevei already have asterisk working
00:20.13PyroStevewith BV
00:20.26ariel_so what type of support to you need?
00:21.10PyroStevenahh... if i explain, ill just get shit for it
00:22.22ariel_PyroSteve, so this was just to see if anyone from BV hangs out here:  PyroSteve needs help from Broadvoice ... is there BV admin here ??
00:22.52PyroStevewell almost
00:22.53tuxinator_linuxDoes anyone want to help a newb with a Digium T1 card?
00:23.12PyroStevewho is JerJer ?
00:23.28ariel_tuxinator_linux, ask the question
00:23.41wangsterAll long distance VOIP providers will be out of business within 1.5 years.
00:23.42ariel_~jerjer
00:23.43jbot[jerjer] the guy who runs nufone
00:23.58ariel_wangster, why?
00:24.11*** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de)
00:24.30Beirdoariel_: because wangster said so
00:24.47wangsterariel_: because its a race to zero for long distance pricing. If you rely on long distance revenue then you are going to be in big trouble.
00:25.09JerJerword
00:25.12ariel_wangster, but 1.5 years...
00:25.36wangsterWe already have a local Cable Co. doing residential voice now and offering a $40/mo package which includes unlimited long distance and all calling features.
00:25.46JerJerits not unlimited
00:25.58wangsterjerjer: says who?
00:26.02JerJeras defined by webster:  'without limits, no boundries'
00:26.08JerJerread the terms and conditions
00:26.12iceypi managed to get the prepaid calling card working to an extent, when i dial 1234 ( exten => 1234,2,DeadAGI(astcc.agi) ) i get a BING then it hangs up
00:26.19iceypanyone know anything about this
00:26.38tuxinator_linuxI don't know what service to order for use with a Digium T1 card.  I want to have 6+ voice lines (one number), 1 fax number, and a toll-free number.  My current setup is a 6 lines in a hunt-group, a fax line, and a the toll-free rings to the voice number.
00:26.50JerJerI guaruntee if you ran more than 4,000 minutes in 3 consectutive months, you would not be a customer for the 4th on a so called unlimited program
00:26.52tuxinator_linuxall analog
00:26.53*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
00:26.58JerJerat this point in the ball game
00:27.15iceypJerJer what prepaid system do you use? custom made one?
00:27.22JerJeryes
00:27.28wangsterJerJer: it is true unlimited. But the price is actually $55/mo (my bad).
00:27.38iceypcare to opesource it :)
00:27.50JerJerwangster:  and i'm saying it is not
00:27.54JerJertheir costs are still per minute
00:27.58iceypthis astcc not working properly :(
00:27.59*** join/#asterisk freddy (~jason@martin.pnc.com.au)
00:28.03iceypi just get BING!
00:28.29wangsterJerJer: $55/mo buys a LOT of minutes. So long as everyone isn't maxing it out they still make lots.
00:28.44ariel_tuxinator_linux, if you have not ordered service from you telco yet what's you location?
00:28.48Beirdo~bing
00:28.49jbotextra, extra, read all about it, bing is an Empirical stochastic bandwidth tester
00:28.52JerJerno they don't
00:28.55Beirdohehe
00:28.58JerJerdo the math
00:29.03KalD|Workwangster, 55/mon only gets me 450 min on my cell =)
00:29.10wangsterIn any case, that is beside the point. It won't be long before everyone has free long distance and then selling long distance only won't be a business.
00:29.45Beirdonot likely
00:29.48DJ-Pyro55/mo gets us 5500 minutes of LD
00:30.02wangsterJerJer: I've done the math. Big providers buy bulk LD for fractions of a cent. thats LOTS of minutes.
00:30.13JerJericeyp:  nope, not gonna open-source any part of my billing system
00:30.15ariel_wangster, bellsouth offers a plan now here for adding to your service unlimited ld us calling for 29 dollars but you need there 30 dollar package before you can order it.
00:30.29*** join/#asterisk chaoscon (~ph33r@chaoscon.user)
00:30.32marc32344is there a smaller capacity line than a T1 PRI ?
00:30.36JerJericeyp:  my theory is if you are billing someone there is a resonable expectation that you are making money
00:30.47ariel_wangster, and that is normal pstn service. But you don't see everyone dropping there service with everyone else and going with them.
00:30.54KalD|Workmarc32344, frac t1 =)
00:30.55tuxinator_linuxmarc, partial T1
00:31.05wangsterariel_: because of the limitations.
00:31.05JerJermarc32344:  a single phone line
00:31.05marc32344how do you get partial T1?
00:31.13wangsterariel_: its probably evenings only?
00:31.16iceypjerjer i plan on it in a few months yes
00:31.18tuxinator_linuxask for it
00:31.19marc32344no I want something like 6 lines..
00:31.20DJ-Pyromarc32344: full t1 but they only turn up a few of the channels
00:31.31tuxinator_linuxyou usually have to pay for a full local loop
00:31.33ariel_wangster, no it's not.
00:31.50ariel_tuxinator_linux, do you have internet service?
00:31.53marc32344dj-pyro -- but still have to pay the full T1 price?
00:31.56yashaxguys, is it possible to have our caller id to show up on someone who is receiving our call from NAME instead of a number?
00:31.56tuxinator_linuxyes
00:32.05*** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz)
00:32.15DJ-Pyromarc32344: no, you pay the local loop charge, and your provider will give you a price for a fraction of the 24 channels
00:32.15marc32344do telcos sell fractioanl T1?
00:32.25wangsterariel_: 1.5 years might be too quick but I don't see a good business in it.
00:32.40JerJerin my book Limitations means the same as restrictions which does not mean unlimited
00:32.44*** part/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz)
00:32.46tuxinator_linuxmarc, just about everybody
00:32.46*** join/#asterisk |neuro| (~|neuro|@212.176.51.231)
00:32.46*** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz)
00:32.50ariel_tuxinator_linux, there are other ways to do this you can get many different packages it depens on cost.
00:33.04JerJerread the legal document(s) behind that so-called unlimited service
00:33.06marc32344dj-- whats the local loop chrage?
00:33.08pdracevichRejected connect attempt from 210.54.x.x, request '00441344844717@bob' does not exist <----- help please please??
00:33.12tuxinator_linuxariel, I'm listening
00:33.18DJ-Pyromarc32344: the cost of the physical line, usually around $200/mo
00:33.20Beirdowangster: ma bell has made money for decades on long distance, people will always be able to make money on something people are willing to pay for
00:33.23JerJerpdracevich:  make that extension exist and it will go away
00:34.16JerJer...in context bob
00:34.25ariel_tuxinator_linux, lets go to a our own window.
00:35.19pdracevichJerJer: AAAAA LIGHT BULB!!!!, i am calling from point a to point b, so at point be there will have to have a dial rule that tells it to go out into the worl...
00:35.21marc32344dj-- how the fractional T1 price work?  $x/24 * $(full T1)?
00:35.44JerJermarc32344:  there are two separate charges
00:35.46JerJera loop charge
00:36.06JerJerand the specific feature charge for which you want the T-1 for
00:36.18JerJerthe loop costs will vary based on milage
00:36.26marc32344specific feature charge?
00:36.29marc32344what is that?
00:36.36DJ-Pyrothe fractional voice service you want to order
00:36.42pdracevichJerjer: am i right??
00:36.54KalD|Workmarc32344, the specail feature charge is for things like more DIDs and line features like CID etc
00:37.06JerJerexten => _00X.,1,DoSomethingHere
00:37.23marc32344ok. so it possible to have a T1 but with only 6channels?
00:37.25JerJermarc32344:  or a connection to the internet
00:37.28JerJersure
00:37.30KalD|Workmarc32344, special feature charges are the charges you get for special features =)  classic 3rd grade answer =)
00:37.35DJ-Pyromarc32344: yes, talk to your provider
00:37.46JerJerthe loop you get will be capible of all 24 channels, but only 6 ~can~ be lit up
00:38.10wangsterBeirdo: the catch being "people are willing to pay for". I'm just saying the small guys doing VoIP for long distance only is not a long term viable business model IMHO.
00:38.12pdracevichexten => _00NXXNXXXXXX.,1,Dial(SIP/${EXTEN}@10.10.x.x) <--- how about that?
00:38.27KalD|Worktypically the telco or provider won't do less than half tho - (or they will charge you for half) because they commonly pay for a full T1 from the upstream carrier
00:38.28JerJerif that mask is correct for your method dialing, sure
00:38.38JerJerand if that SIP carrier wants to see the 00
00:38.50marc32344anyone has a list of cheap T1 providers?
00:38.57JerJerand a period at the end?!  how many digits you dialing man?
00:39.11JerJerwhat's wrong with simply  _00X.  ?
00:39.13DJ-Pyromarc32344: the physical line needs to come from your local telco
00:39.20KalD|Workmarc32344, depends on your location...   try eli.net or twtelecom.com or something
00:39.24_Vilepdr, yeah check Jer out
00:39.25pdracevichnothning *blush*
00:39.25_Vileeli sucks
00:39.30_Vileterrible billing
00:39.31*** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net)
00:39.33_Viledont go ELI
00:39.44*** join/#asterisk chaoscon_ (~ph33r@chaoscon.user)
00:39.53_Viletheir billing system simply sucks, their customer service is ass
00:40.04JerJerdamn sounds like you are talking about NuFone
00:40.07KalD|Worki dunno they bill me every month - i'd say that is good billing
00:40.08_Vilealmost
00:40.16_VileKalD, check what you're billed for
00:40.31marc32344whats the lowest T1 price one can expect?
00:40.41JerJerdepends on the provider
00:40.41KalD|Workmarc32344, ~500/mo
00:40.44_Vileat leas they suck for LD, dunno about T1
00:41.03_Vilemarc, I can deliver an LD T1 locally for 179$ for loop only
00:41.10_Vilei'm doing that right now anyway
00:41.11JerJerI can get loops for pocket change
00:41.15KalD|Work_Vile, we pay 590/mo for local and local ld w/ ELI - it is almost as cheap is TW
00:41.15_Vilecentral oregon area
00:41.22PyroStevei cant seem to find chanspy, does anyone have a link >
00:41.23PyroSteve?
00:41.37_VileKalD, make sure you're paying them the right rate for LD
00:41.37*** join/#asterisk km- (~km-@67.105.178.130)
00:41.40km-HI!
00:41.43_VileI push 1.5m minutes a month
00:41.46km-TE405P rocks. :)
00:41.47_Vileand noticed many discrepancies
00:41.51_Vilew/ ELI
00:42.07_Vileas in 20% of their billing was wrong
00:42.09pointer-gaimJerJer: could you define pocket change?
00:42.23DJ-Pyrotwtc came in at $1200/mo for the DS3 loop, I'm excited
00:42.26KalD|WorkJerJer, who are you working with for your telco stuff?
00:42.30km-hey guys, if I wanted to connect a phone system to a te405p, when it was expecting e&m wink on an esf/b8zs t1, do I have to use a crossover cable from the te405p?
00:42.33_VileI price $1099 for a full DS-3
00:42.36_Vileloop
00:42.44marc32344_vile -- with eli?
00:42.45pointer-gaimKalD|Work: jerjer == nufone
00:42.48km-or is it straight-through
00:42.49_Vilemarc, hell no
00:42.54_VileI don't deal w/ them anymore
00:42.55Corydon76-homeDamn, that's cheap
00:42.56km-I cant seem to get this damn red alarm to clear either way
00:42.57pointer-gaimKalD|Work: oh, sorry...misread
00:42.57_Vilejumped carriers
00:43.02DJ-Pyro_Vile: that includes the cross connect from twtc to GC for the LD service
00:43.06KalD|Workpointer-gaim, yeah but doesn't nufone offer pstn? =)
00:43.13_VileDJ, ahh
00:43.27Corydon76-homeAround here, the ILEC charges ~$1200 for a T1
00:43.27_Vilegood deal then DJ
00:43.42_Vilewe charge $1099 and buy it from the ILEC :)
00:43.48pointer-gaimKalD|Work: so find someone with a nufone did and look it up
00:43.54DJ-Pyronow they're trying to get down to 1c/min LD
00:44.00_VileI'
00:44.03Corydon76-home_Vile: yeah, but for a DS3?
00:44.07_Vilem paying less than 1c/min
00:44.12_VileCorydon, yes for DS-3
00:44.14KalD|Workpointer-gaim, why not just ask JerJer? =)
00:44.15DJ-Pyro_Vile: we don't have the volumes right now
00:44.19Corydon76-homedayum
00:44.21_Vilegotcha
00:44.23pointer-gaimKalD|Work: yes, why not ;)
00:44.34DJ-Pyrostarting at 500k/mo, working up to 10mil/mo
00:44.50_VileWhat are you paying?
00:44.50marc32344_vile-- what hardware you run?
00:44.57_Vilemarc, depends, for what?
00:45.08marc32344local loop
00:45.08_VileDJ, I can blend you at 1.5 for outbound
00:45.15Corydon76-home_Vile: we're paying $500 for a T1 of data
00:45.16_Vileif you're doing SIP stuff
00:45.18DJ-Pyro_Vile: we farm it out to another company right now, we're bringing it back in house
00:45.47DJ-Pyroit's all PSTN termination, conference calling apps
00:45.48_VileCorydon, we sell $99 Data Ts here
00:46.11_VileDJ, colo?
00:46.11marc32344_vile-- location?
00:46.12DJ-Pyrough, stupid probability and stats exam
00:46.16_Vilemarc, Central Oregon
00:46.32DJ-Pyro_Vile: datacenter budget was approved last thursday, construction starts in a few weeks
00:46.40*** join/#asterisk Legend (~legend@24.244.142.133)
00:46.41_Vileok so you're doing your own colo
00:46.49marc32344how many users/did can one typically run on a single T1?
00:46.55_Vilewhat are you bringing in facility wise, for voice?
00:47.06_Vilemarc, depends on the type of traffic
00:47.07DJ-Pyrotwtc extending their sonet mesh into the building
00:47.16_VileDJ, cool
00:47.27marc32344local loop in/out.
00:47.32Trionnisyou're pulling upstream over twc sonet?
00:47.43Trionnishave fun with that
00:47.44Trionnislol
00:47.52DJ-Pyrothey're delivering tdm and ethernet over it
00:48.12_Vilemarc, we charge $99 delivering 100GB/month max, $299 for 200GB, $450 for 300GB T1s
00:48.14Trionnisthey can claim to deliver pizza over it too.... still doesn't matter when it goes down :)
00:48.15Trionnislol
00:48.29_Vileor you can go $139 local loop w/ per MB 95th percentile billing
00:48.36Trionnismaybe they're better in your area
00:48.37_Viles/MB/Mbps
00:48.41Trionnistheir dc here sucks
00:48.48_Vilenever had a problem w/ tw
00:48.53_Vilealways had problems w/ eli
00:48.55Trionnisgood... you're lucky
00:48.58Trionnis:)
00:49.03DJ-Pyrotw is good around here
00:49.08coppicewhy? the pizza will get cold, and the cheese al congealed and nasty
00:49.16_Vileyeah I hate that
00:49.16Trionnishahaha
00:49.22Trionnisnice coppice
00:49.23Trionnis;)
00:50.06Beirdo~insult mikegrb
00:52.15rvhihas anyone implemented any paging function?
00:52.32DJ-Pyrowhat type of pages?
00:53.01rvhicall a number, it rings all phones
00:53.06DJ-Pyroyeah, we did that
00:53.08rvhiall phones have auto answer configed
00:53.21rvhiso all answered
00:53.40rvhihow was it done?
00:53.45DJ-Pyrocreate a meetme conference and join everyone to it
00:54.01DJ-Pyrowe have an agi script that reads out our listing of extentions to join and it forces them all in
00:54.28rvhiis it in open source?
00:54.37DJ-Pyrono, but it's simple to do
00:55.23rvhihow do you join everyone?
00:55.32rvhicall them and auto answer?
00:55.43DJ-Pyroyes
00:56.16*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
00:56.21rvhido i need agi?
00:56.27DJ-Pyrothat's how we do it
00:57.49rvhiis it very similar concept? http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html
00:58.05DJ-Pyroyup, very close
00:58.12DJ-Pyrothat's where we got the idea to do it
00:58.33rvhicool, thx
00:59.20*** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
00:59.25*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
00:59.29BrixiusHello
00:59.42BrixiusIs there anyone here from voipjet?
00:59.56BuckRogershas anyone incorparated automated billing with automated creditcard processing
00:59.57greg_workrvhi: i do it just with computer speakers hooked up to audio out on my soundcard, then call Console/DSP
00:59.59tzangerthere's a voipjet user (that's the nick) who is in here from time to time
01:00.03tzangerI think he got kicked out eventually
01:00.12BuckRogersthis is more for the service providers
01:00.58greg_workrvhi: the spa-841's also have a SIP header they'll respond to specifically for paging, but you need the SipAddHeader() application (only in cvs head, afaik) to use it
01:01.08BrixiusDang, I wanted to talk to someone from there, just had some weirdness I thought they's like to know about
01:01.29tclarkBrixius: like randon call drops on voipjet :)
01:01.40Brixiusno, hearing someone elses call
01:01.47tclarkheh even better
01:01.52ManxPowerSomeone claiming to be from VoipJet has been less than professional in public forums.  That's why I won't use them.
01:02.42Brixiushmmm, I may have to find a replacement, overall I havn't had too many problems with them, but now I'm a little worried who's hearing my call.
01:03.34Brixiushaha
01:03.35tclarkyah its best to have a stable of iax providers they all have issue from time to time, ...
01:04.40*** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.omnilec.com)
01:04.44BuckRogersyeah learn as u go
01:04.46tzangeryup I have two main ones, nufone and then I drop back to sixtel if nufone's down or I can't reach them
01:04.50tzangerbut it's never come to htat :-)
01:04.51BuckRogersat the customers expenese
01:05.07tzangerand failing everything else I can always dial out my Bell Canada PRI at $0.05/min national
01:05.26greg_worktzanger: out of curiousity, how much does the PRI cost you?
01:05.30tzangertoo much
01:05.48tzangerI am in a Tier 4 price group.  That's the offical "bend you over and we won't use lube" group
01:05.51mikes2277app_meetme seems to need a zaptel board, does anyone know of a conference server app that doesn't need a zaptel board?
01:05.53greg_workits a hard thing to find pricing on
01:05.54greg_workheh
01:05.57tzangerno it's not
01:06.06tzangeryou call Bell, Telus, Primus, Allstream and so on and get quotes
01:06.28greg_workwe only have 4 co's.. i'm just curious what the point is where it becomes cost effective
01:06.36ManxPowerThen take the lowest quote, call all the others and tell them what your lowest qoute was.  Rinse.  Repeat.
01:06.38tzangerfour central offices?
01:06.40greg_worki think they'd laugh if i asked for 4 channels :p
01:06.46_Vileoh 4 co lines
01:06.50greg_workno, sorry, i meant 4 Co lines :p
01:06.59_Vileyeah I'd laugh at you
01:07.00Brixiusdoes nufone allow you to set ani information on outgoing calls?
01:07.02tzangergreg_work: yeah PRI is generally only cost effective about the 8-12 line mark
01:07.07_Vileyep 8 or so
01:07.09tzangerBrixius: why would I need to set ANI?
01:07.13_Viledepending on price
01:07.22greg_workwhat about BRI?
01:07.32tzangerI can set outgoing caller ID which does me just fine
01:07.33ManxPowerBrianR___, I don't believe they do, but you can set Caller*ID
01:07.33_Vilewhy bother, more expensive
01:07.37greg_worki mean, all i really want is the call handling capabilites
01:07.41Brixiusthat's what I was wondering
01:07.48_Vileunless you want to set caller ID
01:07.52Brixiusif I can set outgoing caller id
01:07.59ManxPowerBrixius, Caller*ID is NOT NOT NOT the same as ANI.
01:08.03_Vilewhich you probably can't do unless your carrier lets you
01:08.18*** join/#asterisk sysdef (~sysdef@pD9560C7A.dip.t-dialin.net)
01:08.21_Vileand even then you have to give a list of ANI's/Caller ID for you to be able to set
01:08.21ManxPowerIf you use the term ANI when you mean Caller*ID someone is going to smack you.
01:08.30_Vileunless they aren't hard asses
01:08.32ManxPowerThat's like saying "gasoline" when you mean "water"
01:08.35mikes2277most carriers do, we let them set ANI on our PRI's
01:08.35tzangerManxPower: with the mint from your own julip?
01:08.43tzangerManxPower: hahahhaa
01:08.45_VileWe set ANI on our PRI
01:08.47_Viles
01:08.51_Vileand Caller ID
01:08.54nestArlol.. i set mine to whatever i want..
01:08.54mikes2277us too
01:09.03tzangerwhat does being able to set ANI on a VOIP provider get you?  I've not understood that
01:09.04_Vilebut we have 2 DS-3s
01:09.10tzanger911 perhaps but that's a corner case
01:09.26greg_workit kinda sucks theres no viable in-between option. i want to be able to set callerid (only so when you make calls, it appears to come from the main number, as opposed to just random numbers from our hunt group), and just have it answer quicker .. the POTS line rings once before * picks up
01:09.35_Vile911 is a problem, I always advise someone keeping a land line in the place for 911
01:09.38tzangergreg_work: you can get your telco to do that on POTS
01:09.46tzanger_Vile: me too
01:10.12greg_worktzanger: do what? callerid, or sending the callerid info faster so * can answer immediately?
01:10.14tuxinator_linuxWhats E911 then?
01:10.18JerJer911 isn't really that big of a deal
01:10.20_Vileother than that, I just give my carrier 911 information just in case someone makes a call
01:10.29marc32344can you legally operate without 911 capability?
01:10.30_Vilelocation details etc
01:10.32tzangergreg_work: you can ask your telco to set CID to your main # on all your POTS lines
01:10.56_Vilemarc, no but you can tell the customer to use another line for 911
01:10.57tuxinator_linuxtzanger, I tried that, they said they couldn't
01:11.12nestArthey're fibbing
01:11.14_Vileand give your carrier 911 location details for that #
01:11.14tuxinator_linuxputting the main num on all POTs
01:11.15greg_worktzanger: ah ok. next time i need to do something maybe i'll get them to do that. they charged me a $60 admin fee to add a line to our hunt group (was previously a modem line)
01:11.16nestArcan't = won't
01:11.17_Vilejust in case
01:11.44tzangerJerJer: I've heard rumour (nobody seems to know at BCE) that when you're small potatoes (under a DS3 worth of channels) you can't set your DID 911 addresses to anything other than addresses that will be terminated at your local PSAP
01:11.56tzangertuxinator_linux: that sucks, bell will do it for us IIRC
01:12.05tzangergreg_work: yes, I'd do the same thing :-)
01:12.25greg_workcharge me you mean? :)
01:12.30tuxinator_linuxI am changing providers ASAP, XO is killing me
01:12.39tzangergreg_work: yes
01:12.44greg_workbastard ;)
01:13.07tzangeryou make it a nominal fee like $50 or something for any number of changes -- it keeps the dicks who call you every few days wanting a little tweak here and there from calling
01:13.25tzangerand if you save up a dozen things it's better for everyone
01:13.35greg_workfair enough
01:13.51greg_workbut it also costs a lot to people that rarely need to change things
01:13.52*** join/#asterisk yasha (~yasha_x@69.15.218.218)
01:14.08tzangergreg_work: nonsense.. if you rarely need to change things it costs you $50 what once or twice a year
01:14.12greg_workthe sensible thing would be to allow one free change every x days (60 or 90 or something) and beyond that, charge
01:14.27tzangergreg_work: actually that's precisely how I do it
01:14.39tuxinator_linuxgreg, makes sense t me
01:14.40tzangerBell Canada charges us $250 for any change to the PRI
01:14.57greg_worki guess it's more knowing the work they do. i'd guess it literally takes 2 minutes
01:14.57JunK-Ytzanger: like us.
01:14.58nestArbell canada sounds like a bunch of pigfuckers
01:15.00nestAr;)
01:15.00*** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com)
01:15.05nestAroh wait.. it's a bell
01:15.07nestArnvm
01:15.11greg_workwell, thats why you don't use bell canada ;)
01:15.18nestArlol
01:15.22nestArthat and not being in canada
01:15.26BrixiusIsn't ani the # that is displayed as the caller id #.
01:15.30tzangerwell when you're in a town of 5300 and not even allstream will return your calls about quoting a PRI...
01:15.41tzangerBrixius: no, ANI and CID are different
01:15.46tzangeror rather they can be
01:15.48greg_worktzanger: where are you from?
01:15.52tzangerListowel, ON
01:16.11greg_worki used to be with allstream, but they couldn't get us service here. we have some centrex-style lines with primus now
01:16.14km-hey guys, if I have a RED/REC state, I see data on RxA,B,C, and D, but I can't get a green light
01:16.16km-what does that indicate?
01:16.25km-it's trying to recover but its not happening
01:16.25tzangergreg_work: where're you at
01:16.27greg_worksounds familliar, but i'm not sure where it is. western somewhere?  i'm in kingston
01:16.38tzangerkm-: bad framing?
01:16.52km-esf/b8zs is coming from the CO, that's what I'm sending to the remote side
01:16.53greg_workor odessa, more specifically (though i live in kingston)
01:17.00tzangeryou're in RA which means you're transmitting YAI
01:17.04_VileBrixius, ANI is how the call is rated.. what the original calling ID *is*
01:17.10_VileCaller ID is what appears on the phone
01:17.10tzangerthe other side shouldn't be sending you any normal data
01:17.13km-tzanger: I have it setup like this CO --->Asterisk----->Legfacy PBX
01:17.27km-tzanger: I have both spans setup as esf,b8zs
01:17.28BrixiusSo if I want the # I call to display the did # I'm calling from, don't I set the ANI prior to executing the Dial command.
01:17.36km-tazanger: are you saying span2 should be a different framing?
01:17.38BuckRogersi have tear1 and tear2 pricing
01:17.40tzangerkm-: ok
01:17.41_Vilejust use SetCallerID
01:17.42tzangerno
01:17.46_Vileit sets both ANI and Caller ID
01:17.49tzangerAsterisk has two spans
01:17.58_Vileprior to the Dial command
01:17.59BuckRogersspaning tree protocol?
01:18.00tzangerlet's say span 1 is to Telco and span 1 is to PBX
01:18.14km-right, thats how I have it configured, span1=telco span2=pbx
01:18.16BuckRogersgot some nortel baystack switches with that running
01:18.19tzangerset span=1,1,0,esf,b8zs and span=2,0,esf,b8zs
01:18.31tzangerer span=2,0,0,esf,b8zs
01:18.47tzangeryou should get green light if all the wires are correct
01:18.47BuckRogersooo nice
01:18.50km-span=1,1,0,esf,b8zs
01:18.50km-e&m=1-24
01:18.50km-span=2,0,0,esf,b8zs
01:18.50km-e&m=25-48
01:18.54km-that's what I have now
01:18.59tzangerok
01:18.59km-ok
01:19.00BuckRogersgot ot love that green light
01:19.08tzangerkm-: you're using E&M with the telco?
01:19.08BuckRogerslooking smooth
01:19.10km-tzanger: should I be using a crossover cable from asterisk to the PBX?
01:19.18tzangerkm-: generally that is what you need
01:19.19BuckRogerscross over
01:19.21km-yeah, the signal comes in as e&m wink
01:19.21tzanger1&4 -> 2&5
01:19.24km-I tried using the crossover
01:19.25_Vilee&m is cool when you're not going PRI
01:19.27BuckRogersether to ether, yeah
01:19.32_Vilemore features
01:19.37BuckRogerscomp to comp no router
01:19.40_Vilethan standard DSS
01:19.42tzanger_Vile: yup our AS5248s were all E&M Wink
01:19.43km-tzanger: I trued using crossover, but the light stays permanent red in that case
01:19.46tzangerbefore we went to PRI
01:19.51BuckRogersroll over?
01:19.54tzangerkm-: take a loopback cable and plug it in
01:20.04tzangerif it don't go green you have other problems
01:20.10km-loopback cable?  I.e., a crossover?
01:20.13tzangerkm-: no
01:20.18_Vileno, loopback
01:20.19tzangertake an RJ48 end
01:20.22tzangertake two wires
01:20.22BuckRogerswhen you went to pri is the zap configs simialar
01:20.31_Vileyou need to take RX->TX TX->RX
01:20.32tzangerconnect pin 1 to pin 4 and to to pin 5
01:20.46_Vilepin 3 to pin 5
01:20.50*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
01:20.56tzanger_Vile: no not pin 3
01:20.58km-hmm
01:21.06_Vileahh skip pin 3, pin 2
01:21.10km-can I test this loopback cable by using spans 3 and 4 in a particular configuration?
01:21.26tzanger4&5 is pair #1, 1&2  is pair 2, 3&6 is pair 3 and 7&8 is pair 4
01:21.34tzangerethernet uses pairs 2 and 3
01:21.39tzangerT1 uses pairs 1 and 2
01:21.51tzangerkm-: it's not a loopback CABLE
01:21.54tzangerit's a loopback PLUG
01:21.56neophertzanger: yep thats right
01:21.59_Vileyep
01:21.59tzangerit doesn't conenct to anything else
01:22.08km-sorry, I'm getting my nomenclature confused
01:22.17km-I created this crossover cable, I want to prove the crossover is wired right
01:22.20iceypI got a problem using agi's when the call is connected it disconnects within a few seconds, sometimes it'll work for about 15 seconds, but hardly, i get the following error: Feb 23 14:19:19 WARNING[60894]: file.c:1058 ast_waitstream_full: Wait failed (Resource temporarily unavailable)
01:22.23greg_worktzanger: i thought 3&6 was pair 2?
01:22.31tzangerkm-: do you have pair 1 and pair 2 crossed?  If so, it's good
01:22.35_Vilehttp://www.cisco.com/warp/public/471/hard_loopback.html#lb_png
01:22.36tzangergreg_work: nope that's pair 3
01:22.37neopheron ethernet yes
01:22.47tzangerblue is pair 1, orange is pair 2 green is pair 3 brown is pair 4
01:22.50km-so, if I plug this crossover cable into spans 3 and 4, configure 3 and 4 both to have esf/b8zs
01:22.53_Vilekm -> http://www.cisco.com/warp/public/471/hard_loopback.html#lb_png
01:22.54km-and set them both to e&m
01:22.57km-the line should go green
01:22.58km-right
01:23.04tzangerkm-: technically yes but try the plug first
01:23.08km-ok
01:23.09greg_worktzanger: using 568B
01:23.13tzangerdon't create your own experiments, this is basic troubleshooting
01:23.13greg_work:)
01:23.21tzangergreg_work: I use standard telco colouring
01:23.24km-tzanger: ok, make loopback plug, and plug it into span2 of the asterisk box?
01:23.33tzangerkm-: yes, it should go yellow then green
01:23.47km-do I have to set the card in loopback mode?
01:23.48greg_worki used to use B, then i learned that only the US and old telco techs use B, so i switched to A :)
01:23.49_VileI always screw up on using 3 instead of 2, I shouldn't.
01:24.11*** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com)
01:24.14tzanger_Vile: you're just a network weenie instead of a telco weenie
01:24.17tzangerthat's all
01:24.39_Vileim telco, too... I just wire T-1s once a year
01:24.50km-I think my damn crossover cable is wrong, but lemme get this loopback plug fixed up
01:24.54tzanger_Vile: yeah but if you do any trunk wiring you know the color code
01:24.57_VileI get to get my hands on a metaswitch in about 6 months
01:25.06_Viletzang, I don't do enough you're right
01:25.10tzangerI'm also an electronics guy so I know that bad boys rape our young girls but violet goes willingly
01:25.12_Vileim more software
01:25.13greg_worktzanger: oh, T1 cabling is different from ethernet?
01:25.18_Vilegreg, yes
01:25.22tzangergreg_work: yes and no
01:25.37tzangerthe pairs are the same, it's just that ethernet uses pairs 2 and 3 and T1 uses pairs 1 and 2
01:25.38greg_worki just assumed they'd use the same cabling .. on ethernet i was right, 3&6 is pair 2
01:25.47tzangerso an ethernet crossover's totally different from a T1 crossover
01:25.53greg_workoh sorry, for 568A
01:26.03neopheryou could use a straight through ethernet, but the tx and rx pins are different on T1 vs ethernet
01:26.11ManxPowerBut an Ethernet straight-thru cable can work for Ethernet, T-1, and POTS
01:26.14_Vileblue blue white red orange yellow green...?
01:26.18greg_work568B it's 1&2 .. but orange on both
01:26.21tzangerneopher: yeah but the pairs are still right :-)
01:26.35ManxPowerBut, as someone pointed out you'll break the pairs and add potential problems
01:26.50tzangersplit pairs are the cause of global warming
01:26.53tzangerand the iraq war
01:27.00neophercorrect, but if you are making a loopback plug, you need to know the correct pins to jump and tx and rx are diff
01:27.08tzangerneopher: yes
01:27.10_Vilethat and those damn cars
01:27.18BuckRogerso man i wish i had some cronic
01:27.18tzangerI've never seen a useful ethernet loopback plug :-)
01:27.42BuckRogersoooo thank god, (hands shaking)
01:27.51_Vilewhiner
01:27.53_Vile:)
01:28.02neophertzanger: very true
01:28.10tzangerBrixius: so what are they
01:28.14BuckRogerscrazy day of agi perl scripting
01:28.19tzangercid is for us wee plebs, ANI is for billing :-)
01:28.20_VileBrix, you can just use SetCallerID and call it a day
01:28.27_Viletzang is right
01:28.34km-ok
01:28.37km-loopback plug complete
01:28.40_Vilejust remember the BTN will be the SetCallerID #
01:28.41km-and we have a green
01:28.47tzangerkm-: so your crossover's not
01:28.49_Vilewhen you're dialing 800 #s etc
01:28.50BrixiusFrom what I can tell ani is for billing, but you answered the q before I could.
01:29.01tzangerANI can't be (easily) fucked with
01:29.02tzangerCID can
01:29.04km-tzanger: yeah, I had the plug in the wrong orientation, was counting from pin 8 as opposed to pin 1
01:29.12km-tzanger: the cisco document proved that to me
01:29.21tzangerkm-: always treat your RJ48 right
01:29.23neopherANI.L
01:29.26tzangerpins up, facing you
01:29.33tzanger1-8 is left-to-right
01:29.59_Viletzang, I don't know if that's true or not, I can set ANI and Caller ID ??
01:30.06*** join/#asterisk trym (~trym@linux.debian.us)
01:30.07tzanger_Vile: depends on the telco
01:30.08_Vileand I'm on Override CPN
01:30.11_Vilew/ MCI
01:30.13_Vileoutbound LD
01:30.20_Vilethat's LD though, I guess and not 800
01:30.23tzangerasterisk does allow you to set ANI too with another IE but the telco may just reject it
01:30.25_Vileso it doesn't matter
01:30.41km-ok
01:30.43km-for one more time
01:30.46_VileI can call you right now w/ 666 as my phone number and the ANI will be 666 :)
01:30.47km-1-4 2-5 right?
01:31.00tzangerkm-: yup
01:31.34tzangerjust look at it this way
01:31.34_Vilein fact your cdr's will show 666 :)
01:31.36tzangertake your cat5
01:31.40tzangerbring out the orange pair
01:31.48tzangerorange-white and orange go right up the middle
01:31.57tzangernow take the blue pair
01:32.02tzangerbluewhite and blue go on the far left
01:32.06_VileBrixius, but yes, tzanger is right, and you're right, ANI is for billing, CDR is for presentation
01:32.10tzangernow the othe rside is just the opposite
01:32.25tzangertake the greenwhite/green pair and "straddle" the middle pair
01:32.32tzangertake the brownwhite/brown pair and put them on the far right
01:32.40_Viletzang, to correct myself, it'd be 000-000-0666
01:32.53tzangerI don't even count anymore, it's just "right up the middle" and "to the left" or "straddle the middle" and "tot eh left"
01:33.08tzanger_Vile: my CDRs will?
01:33.18_Vilenot LD, just inbound * CDRs
01:33.51_Vilespeaking of which
01:34.05_Vileare standard csv CDRs for Asterisk reporting ANI or CID?
01:34.09BuckRogersdont forget to bring a towel
01:34.11_VileI need to know that
01:34.20tzangerI'm confused
01:34.27tzangerwhy will my CDRs be 0000000666?
01:34.49_VileIf I call you from 666, it will show up in Asterisk as the number dialing being 666
01:34.55tzangerahhhhhh
01:34.56_VileI can show you
01:34.58tzangerokay
01:35.07tzangerI understand now
01:35.07_Vilebut I don't know if that's Caller ID or ANI
01:35.16_Vilein the Asterisk CDRs
01:35.35_VileI think I'm supposed to be getting ANI reporting from my carriers and not CID
01:35.36tzangerI imagine asterisk gets it right although I haven't the need to worry about it
01:35.50tzangerand unless you'e got a PRI you'll never know if it's different
01:35.56_Vileit's all PRI
01:36.24*** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
01:36.26tzanger_Vile: pri debug span 1 and place a few calls where you change the CID -- call yourself and see if you get ANI info
01:36.27_VileI've asked for ANI reporting, but I dunno, I guess I could check by calling myself :)
01:36.32tzangerI'm doubting it but could be wrong
01:36.41*** join/#asterisk sysdef (~sysdef@pD9560C7A.dip.t-dialin.net)
01:36.48km-if the line is in yellow alarm
01:36.50km-whats that mean?
01:37.02km-with my new crossover, the line is now in yellow as opposed to red
01:37.04_Vilesignalling maybe?
01:37.06JunK-Ykm: on PRI?
01:37.10jaydenwassup...
01:37.10km-should I reboot the phone system?
01:37.14tzangeryellow alarm means that one end sees the other
01:37.15_Vilewon't help
01:37.15km-em_w esf/b8zs
01:37.22tzangerbasically take you and me.
01:37.28JunK-YYEL means remote end is unplugged i think
01:37.36JunK-Yor the dchan is not alive.
01:37.38_Vileno red would be remote is unplugged
01:37.40tzangerI can't see you, so I will transmit red alarm.  I CANNOT HEAR YOU
01:37.56_Vileyellow means there's signalling problems
01:37.57tzangeryou hear me screaming at you, so you send back "YOU ARE SHOUTING IN MY EAR"
01:38.00km-ok, so, maybe one of the pins on my crossover isn't fully crimped?
01:38.01_Vilewrong setup, not esf when it should be
01:38.05tzanger_Vile: not quite
01:38.09*** join/#asterisk convey (~test@208-216-127-234.cust.gti.net)
01:38.09_Vilenot b8zs but is ami, when it should be reversed
01:38.10_Vileetc
01:38.15tzangerred alarm means I can't see the other side.  it's an unframed all-1's pattern
01:38.38_VileI've always experienced yellow when my signalling is not right on Digium cards
01:38.45tzangeryellow alarm means I can see a signal but it is unintelligible
01:38.49tzangerthis is "lower" than signaling
01:38.53conveyanyone using asterisk as an application server for Ser?
01:39.00greg_worktesting with untested test equipment == fun
01:39.06km-tzanger: so, do you think my crossover may have a broken wire or something?
01:39.06_Vileso, if one side was setup for d4
01:39.11_Vileand the other side was setup for esf
01:39.13tzangerkm-: looks to be that way
01:39.16jaydenyellow alarm... kinda like what usually goes on in #asterisk...
01:39.18_Vilethat would be unintelligible, correct?
01:39.19km-tzanger: ok, I'll redo it again
01:39.23tzanger:-)
01:39.27jaydenred alarm, like your network cable is unplugged
01:39.29tzanger_Vile: not necessarily
01:39.30km-hey, better I make progress
01:39.40jayden:)
01:39.43_Vilethat's the cases where I've experienced a yellow, check it out
01:39.49jaydenI'm gunna be good tonight :)
01:39.56_VileI'm not :)
01:39.58tzangerD4/SF framing is perfectly legal ESF framing, but the extra bits in ESF can confuse SF
01:40.08_Vilethus causing a yellow
01:40.28tzanger_Vile: incorrect signalling *can* cause a yellow but doesn't ensure it
01:40.41pr0manyone here have experience with the linksys pap2 ata?
01:40.46tzangerpr0m: not me
01:40.48km-besides vile
01:40.59km-in this case we've already verified I'm using the right signalling
01:41.00pr0mtzanger: ok.
01:41.09Himekoi use one, but that is about it
01:41.18_Vilekm, ok, catch me up, what signalling are you using on each side?
01:41.31_Vileand have you restarted asterisk and re-ran ztcfg?
01:41.36Himekoa pap2 that is
01:41.42pr0mhow about decrypting an rc4 encrypted file?
01:41.42eKo1Dang it, why doesn't the src field in the cdr table have the proper source. Damn it to hell!
01:41.57km-_vile: esf/b8zs coming from the CO, esf/b8zs I'm sending to legacy pbx
01:41.59jaydenkm:  are you using all 23 (or 24 if t1) channels?  are you using external csu, if it is a smart csu, does it have the same settings
01:42.00pr0mHimeko: do you use it with asterisk?
01:42.11tzangerkm-: that's framing and line coding, not signaling
01:42.12jaydenor no csu
01:42.16*** join/#asterisk Newbie___ (some@211.24.146.10)
01:42.19_VileCO is sending esf/b8zs?
01:42.23jaydenPRI or T1 or E1?
01:42.38km-yep
01:42.38tzanger_Vile: I've never seen a T1 in the last 10 years that was SF/AMI
01:42.45tzanger_Vile: not "in the wild" anyway
01:42.46_VileI have 22 of them
01:42.49_Vileis it PRI or DSS, and what does your zapata.conf look like?
01:42.50tzanger_Vile: wow
01:42.55Himekopr0m it is connected to * yes
01:42.58Nuggetms word uses rc4.
01:43.16ManxPowerI didn't know you could run PRI over AMI
01:43.23pr0mi've got a pap2 that has vonage as the only voip provider.... and i'd like to "unlock" the device so i can point it towards an asterisk server.
01:43.23_Vileyou can't
01:43.28_VileI have 22 AMI/D4
01:43.33tzangerI didn't thnk so either... technically feasable but why?
01:43.35Himekooh, min eis a pap2-na
01:43.41tzanger_Vile: PRI or CAS T1
01:43.42_Vileand 6 ESF/B8ZS PRI on the rest of the DS-3
01:43.47_Vilecas
01:43.48jaydenpr0m:  I think vonage will unlock forr $15
01:43.49pr0mHimeko:  ok.  hhhmmmm
01:43.50jaydencall them
01:44.02pr0mjayden: really?  huh.
01:44.15JerJeryeah 'unlock'  ok
01:44.17jaydenyes, they will for the cisco and motorolla ones
01:44.23_Vileit's only outbound stuff, calling card application w/ no caller id
01:44.33pr0mhimeko can you download your firmware from the pap2-na?  backup?
01:44.38_Vilegives me an extra channel per T, though I could option the whole thing w/ 6 signalling channels
01:44.40_Vilebut im lazy
01:44.45jaydenthey reflash with a diff firmware I beleive
01:44.45_Vileand use all PRI
01:44.48_Vilebut I'm lazy
01:44.52jaydenthat is the word on the street
01:44.59JerJerjayden:  no
01:45.08jaydenok
01:45.22jaydenhey, I am just repeating the bad info I got... :)
01:45.25JerJerthey simply reconfigure the device not to ask for a password when going to the admin interface
01:45.36pr0mi've successfully upgraded the pap2 with a local tftpserver and vonage provided firmware package.  ....
01:45.40jayden^^ what he said
01:45.56JerJerpr0m: then you must pay vonage now
01:46.01JerJerthat was your mistake
01:46.05ManxPowerJerJer, Could you take over torturing users while I'm busy? 8-)
01:46.11tzangerI just discovered the uselessness of the smarthome X10 stuff that "auto-configures" their address
01:46.19*** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
01:46.20jaydenJerJer says, use Nuphone :)
01:46.23tzangerpower outage...everything is on the address the first command is sent to
01:46.24pr0mJerJer: i've reset the device several times.
01:46.25*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
01:46.40JerJerits locked down hard
01:46.45BuckRogersanyone use simpleconnect?
01:46.49ariel_ManxPower, your busy....
01:46.50pr0mthat's not the problem.  it's uploading firmware that will allow me admin access or get the passwd.
01:46.57JerJernow that you've let it communicate with vonage
01:47.10greg_worktzanger: which ones are those? i just started installing a bunch of the switchlinc dimmers at our cottage .. :p
01:47.33pr0mJerJer: i have access to the factory default web interface.  just no passwd.
01:47.34greg_workthey don't autoconfigure though (neither do the keypad 8's)...
01:47.49tzangerhttp://ericzorn.com/extra/dibs/
01:47.50tzangerwhat the fuck
01:47.55Beirdopr0m: welcome to the club
01:47.56tzangeryou guys put trash in parking spaces to reserve them?
01:48.07pr0mBeirdo: what's your experience with the pap2?
01:48.10tzangergreg_work: these aren't dimmable but anything that autolearns will have this problem
01:48.17Beirdoit's locked
01:48.34pr0mhmmm. no success, eh?
01:48.38JerJerits not locked
01:48.39cbachman<PROTECTED>
01:48.46JerJersimply pre-configured to vonage
01:48.48greg_worktzanger: i had to send some weird sequence to program them
01:48.48tzangercbachman: unreal
01:48.55tzangergreg_work: these auto-learn
01:49.00JerJerthen vonage really locks it the first time it communicates with their TFTP server
01:49.04tzangerjust press the button and send a command
01:49.08tzangeron or off
01:49.13greg_worktzanger: oh sorry, that was to do scenes.. to program all the ones i have you hold the set button for 3 seconds, then send the address
01:49.14Beirdowell...  it's configured to do some funky MD5 auth on the admin interface too apparently
01:49.18pr0mBeirdo: do you know how to reset it to factory defaults?
01:49.37greg_workthey come factory set at A1
01:49.40Beirdothat info's out there, but I don't think it will get you very far
01:49.45cbachmantzanger: http://www.policyguy.com/pubs/ChicagoSnow.html
01:49.46tzangerpr0m: use the JTAG port and reflash that way :-)
01:49.51JerJerpr0m:  it is never going to happen without involving vonage, since you let it communicate with them
01:49.55sysdefeh.. no u* -users here? lol
01:50.12pr0mjtag.  i'll give that a googling.
01:50.28pr0mJerJer: i've heard otherwise on newsgroups.  i
01:50.37greg_workwhich is a bit crazy at first .. i only put a few in so far (most of the elec box isn't wired up yet) and when i first tured it on, one button on the keypad would control EVERYTHING ;p
01:50.38pr0mi'll keep my hopes up for now.
01:50.47tzangercbachman: that's unreal
01:50.52tzangerpr0m: I'm joking
01:50.57tzangeryou won't flash it with a jtag easily
01:51.08km-OK
01:51.09km-that's odd
01:51.12km-green
01:51.15pr0mtzanger: ok then.  thanks for... um nothing.  ;-)
01:51.15JerJerthe pap2's are trivial to reconfigure provided they never communicate with vonage's TFTP server
01:51.18km-but,  I dont get a dialtone
01:51.27tzangerpr0m: sorry man, you let it talk to the world, go sell it on ebay
01:51.32km-the legacy pbx isn't getting a dialtone when I hit the t1
01:51.38BeirdoJerJer: and if they have old enough firmware
01:51.45km-oh.
01:51.46km-fuck.
01:51.49tzangerkm-: have you set your E&M signaling properly in zapata.conf (i.e. winktimes, etc.)
01:52.08*** part/#asterisk sysdef (~sysdef@pD9560C7A.dip.t-dialin.net)
01:52.21km-asterisk wasnt running
01:52.22km-heh
01:52.25km-I'm such a moron
01:52.34tzangerkm-: :-)
01:52.36jaydengreen is good
01:52.44pr0mback to genetic disorders.
01:52.49Nuggetbut it's not easy being green
01:53.06Nuggetgreeno shot first, anyway.
01:53.17Himekoafter the pap2 has talked with vonage can't you fake their provisioning server to provide it with your own config?
01:53.49JerJersure, provided you can provide the proper answer to the provided challenge
01:53.50tzangerHimeko: if you break the password
01:53.58*** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au)
01:54.10*** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com)
01:54.40Beirdomine's virgin and is still password locked.  I think they got wise
01:54.54km-ok, the greens are flowin
01:54.56km-awesome
01:55.03Himekoif it's provosion tftp
01:55.06Himekoiwups
01:55.07JerJerBeirdo: then u don't have the right file
01:55.14Beirdocorrect :)
01:55.26BeirdoI think mine's for an older version of firmware
01:55.54Himekoif it's provisioned tftp wouldn't the challange be plain text anyway
01:56.19Himekothey prolly use http provisioning though
01:56.36Beirdoif anyone has a clue, feel free to msg me :)
01:56.37Beirdoheh
01:56.51*** join/#asterisk stilexip (~wow@pc-24-151-59-224.newt1.ct.charter.com)
01:56.54stilexiphello all
01:57.15Franticguys- anyone has a problem that the show sip peers suddenly becomes empty- no sip reg is possible? (CVS from yesterday)?
01:57.36Franticissueing stop now- doesn't do anything
01:57.43Franticonly way is to restart the machine
01:59.08km-tzanger: ok, I got the t1 up, now I've got a weird problem
01:59.32km-tzanger: when I try dialing a number, for instance, 14848759460, it only sees the 1 before it starts dialing
01:59.36stilexipcan someone tell me why asterisk core dumps  config.c:507 ast_internal_load: Loading Config extensions.conf via mysql engine
01:59.44tzangerwhat only sees the one
01:59.44km-tzanger: this is on a t1 -- shouldn't it be waiting for 3 seconds or something?
01:59.49km-tzanger: asterisk
01:59.55tzangerkm-: what's your dialplan look like
01:59.59stilexiprealtime works with dynmic extensions not the static file
02:00.18km-[incomingpbx]
02:00.18km-exten => _9XXX,1,Congestion
02:00.18km-exten => _1XXXXXXXXXX,1,Dial(Zap/g1/${EXTEN})
02:00.18km-exten => _011XXXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN})
02:00.27km-calls from the legacy pbx go into incomingpbx context
02:00.41tzangerkm-: ok, so no '.' use
02:00.55km-should immediate=no or immediate=yes be in the zapata.conf file for the legacy pbx's t1 line?
02:00.56tzangerthat's all you have there?
02:00.59tzangerimmediate=no
02:01.02tzangeryou never want immediate
02:01.08tzangerunless you're wiring up a phone for BatMan
02:01.17km-group=2
02:01.17km-immediate=no
02:01.17km-signalling=em_w
02:01.17km-context=incomingpbx
02:01.17km-channel=>33-48
02:01.19km-that's from zapata.conf
02:01.24jaydenI want immediate gratification
02:01.47jaydenok... I see an issue
02:01.48tzangerkm-: that's fine, I meant what's show dialplan incomingpbx show fromthe CLI
02:01.54km-ok
02:01.59jaydendialing group 1, config you just sent was group 2
02:02.08tzangeryeah this is true too
02:02.12km-jayden: I'm calling from g2 to g1
02:02.17jaydenahhh
02:02.17jaydenok
02:02.19km-g1 is the CO T1
02:02.22km-g2 is the PBX T1
02:02.31km-*CLI> show dialplan incomingpbx
02:02.31km-[ Context 'incomingpbx' created by 'pbx_config' ]
02:02.31km-<PROTECTED>
02:02.31km-<PROTECTED>
02:02.31km-<PROTECTED>
02:02.33km-actually
02:02.36km-if I dial fast enough
02:02.41*** join/#asterisk lilneon (~tj_r3@200.108.20.38)
02:02.46lilneonhi and good nigth everyone
02:02.47km-I'll get more chars
02:02.51tzangerkm-: set a dial timeout
02:02.55km-on t1?
02:02.59tzangernow how to do it cleanly I'm not so sure
02:03.00jaydenso, if you set somthing on incoming pbx s to just play a file, does that work (is it getting in ok?)
02:04.11*** join/#asterisk xachen (xachen@edtntnt3-port-262.dial.telus.net)
02:04.14lilneonhey guys, anyone can point me in the direction of an installation for freetds with asterisk to use mssql server 2000?
02:04.15jaydenthen, noop(${EXTEN}) and see what your getting from pbx
02:04.18km-although I love listening to allison smith's voice say "the number you have diald is not in service"
02:04.23km-heh
02:04.31jaydenso that works?
02:04.48xachenhmm
02:04.53xachen-> IAX2/guest@66.250.68.194/996
02:05.02xachenhow does a person setup direct SIP and IAX2 calls?
02:05.07jaydenwhat about just a static dial back out zap/g2/some extn that works?
02:05.09km-*CLI>     -- Starting simple switch on 'Zap/48-1'
02:05.09km-<PROTECTED>
02:05.10xachenI'm just familiar to using the PSTN
02:05.18xachenwith IAX2 peer for VoIP
02:05.20JerJerlilneon: you are on your own for that one.... i have to believe not many here are going to even consider something like that much less do it
02:05.22km-that's with extension 1 setup
02:05.38km-jerjer: hey, do you know how to set a longer timeout on a dialplan?
02:05.47km-for instance, if you have a really slow typist
02:05.49JerJerwhich one?
02:05.49km-on the phone
02:05.50tzangerkm-: DigitTimeout or something like that
02:05.52JerJerResponseTimeout
02:05.56JerJeror DigitTimeout
02:06.04tzangerbut I don't know how to do that for every single extension
02:06.05tzangerthat seems wrong
02:06.10jaydenbkw likes sql
02:06.16km-jerjer: is there a global responsetimeout or digittimeout?
02:06.25jaydenI like sql, but have not played with it to * yet
02:06.35shido6dialplan logic is what jerjer will learn you
02:06.50tzangerkm-: I'm not aware of one
02:06.51JerJerkm-:  i have not seen one
02:06.53tzangerthere should be one though
02:06.59km-ok
02:06.59km-so
02:07.01tzangerI mean it's absolute insanity to think you need that
02:07.05km-if I were to set it up for a particular constant
02:07.06tzangerfor every single extension defined
02:07.07km-err context
02:07.10tzangerin fact you can't
02:07.18JerJercould be as simple as checking for the existance of a global variable being set
02:07.22shido6can u rethink the dialplan to make that happen?
02:07.23lilneonjerjer: i did it before... doing the same exact thing but it wont wrk
02:07.26tzangerexten => _1NXXNXXXXXX,1,SetTimeout(10) own't work
02:07.28JerJerif so, use that value
02:07.34tzangeractually
02:07.38tzangeruse immediat
02:07.39tzangeree
02:07.39km-its gotta be defined in the source somewhere, this timeout
02:07.43tzangerimmedate=yes
02:07.45JerJerlilneon:  perhaps there is a reason
02:07.45km-immediate=yes?
02:07.49stilexipcan you load a static config using the mysql res_mysql.so or is res_odbc required
02:07.50tzangerand s,1,SetTimeout(3)
02:07.55tzangers,2,Read(DEST)
02:08.05tzangers,3,Goto(dialout,${DEST},1)
02:08.16tzangerTO THE BATMOBILE
02:08.40km-theres got to be a place where that timeout is defined
02:08.41jaydeneveryone needs a batphone,
02:08.43km-...
02:08.44km-I mean
02:08.44lilneonJerJer: care to share??
02:08.50km-it's not just something the computer randomly decides
02:08.54tzangerkm-: I just gave you the solution
02:09.02km-somewhere in a .c file there is "#define REALLY_SHORT_TIMEOUT"
02:09.10km-tzanger: but that's not the clean way to do it, wouldn't you agree?
02:09.18lilneonJerJer: is it that microsoft being stupid again?
02:09.20jaydentimeout for what?  I'm lost
02:09.25tzangerkm-: actually it's the cleanest
02:09.35km-tzanger: you think so?  Ok, I'll go with it then...
02:09.36tzangerand the only way I can forsee it working
02:10.31*** join/#asterisk bkw_ (nobody@bkw.developer.and.friend.of.asterisk)
02:10.31*** mode/#asterisk [+o bkw_] by ChanServ
02:11.14*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
02:11.25jaydenbkw:  somone was jsut looking to get mssql working w/ *, do you have it working, or were you just saying that ou like sql
02:11.28hermieperfect for circumventing spam filters!
02:11.56km-that Read(DEST) doesnt work
02:12.36lilneonbkw:yeah I think you wrote freetds? could you help me get it wrking?
02:13.27lilneonI keep getting a server is unavailable or does not exists when i try tsql.. and well asterisk just hangs an says unregistered mssql backend
02:14.17km-thats so weird
02:14.22km-I wish I could debug chan_zap
02:14.24tzangerkm-: Read(DEST) followed by NoOp(${DEST}) -- what does the noop say
02:14.28km-matchdigit is 3000
02:14.51BuckRogerszaptool?
02:15.40km-<PROTECTED>
02:15.40km-<PROTECTED>
02:15.40km-<PROTECTED>
02:15.40km-<PROTECTED>
02:15.40km-<PROTECTED>
02:16.03tzanger... user disconnected?
02:16.14km-yeah
02:16.18km-the line was active the whole time
02:16.24km-but it said user disconnected after the 4 seconds
02:16.40tzangershow application read
02:16.44tzangeryou might have to hit #
02:16.49tzangeror specify a max# of digits
02:17.21tzangeryou may also need ||noanswer
02:17.23km-haha, I'm not going to make a whole company dial # when they've complteed the number
02:17.31BuckRogershaha
02:17.35tzangerwell yeah I don't blame you
02:17.41km-shiesse
02:17.42tzangeryou mught have to use multiple reads
02:17.44BuckRogershow many users
02:17.45tzangerread 3
02:17.45BuckRogers?
02:17.48tzangerthen read 4
02:17.50tzangerthen read 3
02:17.56km-buckroger: more than I want to have hit # :)
02:18.03tzangerthen read maybe 10.  :-)
02:18.07tzangerfor international
02:18.10BuckRogers20 -40
02:18.11km-tzanger: oh my god, you cant possibly be suggesting that that's the easy way :P
02:18.17km-buckroger: yeah
02:18.29km-closer to 30
02:18.30tzangerthe timeout though should do it
02:18.32BuckRogersyeah doesnt look professonal
02:18.50km-someone else must have run into this problem before
02:18.50tzangerkm-: I don't know -- there's gotta be an easier way but I'm not seeing it
02:19.57km-thats weird
02:20.10km-I just tried to put a static Dial(Zap/g1/14848759460) in there and I just got a dialtone
02:21.06*** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com)
02:21.48km-this shouldnt be this hard
02:22.32tzangerwhat about
02:22.35tzanger(and I cringe)
02:22.44km-well, if I have a group setup
02:22.46tzanger_.,1,DigitTimeout(4)
02:22.57km-tzanger: hmm.
02:23.09ManxPowerNOOOOOOOOO!!!!!! Not _.!  Anything but _.!
02:23.11tzangerand then have everything else as _NXXXXXX,**2**,Dial()
02:23.18tzangerManxPower: well help us out here
02:23.28km-manx: do you have any ideas?
02:23.31ManxPowertzanger, What's his problem?
02:23.34km-manx: do you know the situation I'm in right now?
02:23.35km-ok
02:23.38km-lemme break it down
02:23.43km-I have a T1 from the CO going to Asterisk
02:23.49km-I have a T1 from Asterisk going to legacy PBX
02:23.49ManxPowerkm-, No, I've been cleaning the apartment.
02:24.04km-when I pick up the phone and dial 9 on the legacy pbx to open up a line on the t1
02:24.16km-the timeout is set too high and the first digit I hit spawns an extension
02:24.17ManxPowerkm-, Loopstart?  Groundstart? E&M?  E&M Wink?
02:24.20km-instead of 14848759460
02:24.22km-I get 1
02:24.25km-as asterisk sees it
02:24.31km-em_w on both sides
02:24.42*** part/#asterisk JoaoCorreia (~JoaoCorre@81.193.116.63)
02:24.44ManxPowerkm-, you didn't do something stupid like immediate=yes, did you?
02:24.49km-well
02:24.51km-we're trying that now
02:24.56km-immediate=no makes the 1 fire off
02:24.59ManxPowerdon't do immediate=yes
02:25.02tzangerManxPower: we're trying that to read in the digits
02:25.07tzangerso we can set a digittimeout
02:25.21ManxPowerI suspect you have wink problems.
02:25.30km-maybe the line isn't wink?
02:25.43ManxPowerkm-, What happens when you call into Asterisk?
02:25.45km-I tried em and it didnt work
02:25.49km-I tried em_w and it worjked
02:25.57km-calling into the Asterisk system from my cellphone works perfectly
02:26.03km-the IVR pops on, you can surf the directory, etc
02:26.07ManxPowerkm-, Have you confirmed with the telco that the line is E&M/Wink?
02:26.13km-I have not
02:26.20ManxPowerkm-, You don't have any DIDs?
02:26.23km-I do
02:26.31km-I tried dialing the did and I saw the correct extension
02:26.37ManxPowerAnd do the DIDs work?  i.e. do the DIDs come in as an extension?
02:26.40km-14848759462 produced a call g1/462
02:26.42km-yep
02:26.47ManxPowerGood.
02:26.55km-I got what I was expecting on the DID dialing
02:27.06ManxPowerWhy isn't your PBX just collecting all the digits, stripping off the 9 and then sending them all to Asterisk?
02:27.15km-good question
02:27.18km-I can't ask the pbx to find out
02:27.22ManxPowerThat's the way you would want to do it with E&M/Wink.
02:27.24km-$700 worth of equipment to answer that question
02:27.27tzangerkm-: if you plug the pbx directly inot the telco it works though
02:27.30km-yeah
02:27.31ManxPowerkm-, The answer is "lazy PBX admin"
02:27.36km-it works fine directly to the telco
02:27.40km-no
02:27.54km-this NEC electra system requires you to pay like $700 worth of manuals/software/cards so you can configure it yourself
02:27.57km-otherwise it's a service call
02:28.05marc32344any other boards based on te410?
02:28.19km-ok, so, how come it works fine to the telco
02:28.21tzangerkm-: does the electra have ATAs -- things you plug in that let you plug ordinary phones into it?
02:28.22km-but not to asterisk?
02:28.34km-tzanger: there are ata adapters for it, yes
02:28.43tzangerkm-: got a part number or something I can go on in google?
02:28.45km-tzanger: but they're to feed extensions
02:28.50tzangerkm-: yes exactly
02:28.52km-tzanger: the phone system?
02:28.59tzangerlets you use an ordinary cordless phone as an extension type of thing
02:29.00km-I think it's an Electra elite IPK 192
02:29.02km-right
02:29.27km-but they're both in use
02:29.27km-what are you thinking?
02:29.28tzangerkm-: totally separate problem, I'm just wondering what they're called (part number or name) in NEC land
02:29.41scythelxanyone know the cmd to perform an extensions reload through the manger interface
02:29.50km-manxpower: the weird thing is, the digittimeout in chan_zap is 3000ms, but, it's like it's only waiting 1000ms
02:32.11scythelxis it possible to issue an extensions reload command thru the manager interface
02:34.33km-thats weird
02:34.39km-I cant dial a 1 first
02:35.08tzangerkm-: sounds like your phone system has a dialplan of its own doing
02:35.20tzangerkm-: got a part number or model number of those NEC ATAs?
02:35.33km-just a sec, lemme see if I can rip one off the wall
02:35.38km-I think I may have mistakenly dialed 911
02:35.42km-that'd be just great
02:35.43km-heh
02:36.28tzangerkm-: hahaha
02:37.36km-SLTA-F-20 unit
02:37.37km-I think
02:37.45km-thats the only thing that looks like a part number on the PCB
02:37.52BuckRogersany one use the SBC t3 Cards
02:38.13*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon....
02:38.38km-SLT1-U-10ADP
02:38.40km-err
02:38.48km-SLT(1)-u10 ADP
02:39.24tzangerthanks km-
02:40.21km-thats really curious
02:40.29*** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.omnilec.com)
02:40.37km-exten => s,1,Dial(Zap/24/4849191400)
02:40.38km-with that
02:40.46km-if I hit 9 on the legacy pbx, I get
02:41.01km-<PROTECTED>
02:41.01km-<PROTECTED>
02:41.04mikes2277Does anyone know of a free activex or java phone for use in Internet Explorer?
02:41.09km-dialtone continues on the legacy pbx
02:41.12km-but my cellphone does ring
02:41.30tzangerkm-: uh
02:41.38km-oh man that's odd
02:41.45km-if I dial a digit to get rid of the dialtone
02:41.48km-I can then hear the ringing
02:42.17tzangeris it the PBX dialtone or asterisk's dialtone
02:42.26km-I think it's the pbx's dialtone
02:42.39ta[i]ntedyou guys ever use DISA?
02:43.04km-[ Context 'incomingpbx' created by 'pbx_config' ]
02:43.04km-<PROTECTED>
02:43.14km-that should say that any invalid digit should dial that right
02:43.24km-I dialed 1, I got
02:43.28km-<PROTECTED>
02:43.28km-<PROTECTED>
02:43.28km-<PROTECTED>
02:43.32ManxPowerkm-, I belive Asterisk expects a DID on E&M/Wink
02:43.46km-hrm?
02:43.59ManxPowerkm-, That's what it looks like to me.
02:44.08ManxPowerYou may need to use DISA
02:44.12marc32344any other cards that does the same as digium te410p?
02:44.23JunK-YTE405P?
02:44.29lilneonalrighty then.. has anyone gotten the mbrola voices to wrk with festival?
02:44.35ta[i]ntedsometimes i can't get DISA to return dialtone.. was wondering if carriers block it
02:44.50tzangermarc32344: what, *exactly* is your question or what are you trying ot do
02:45.07tzangeryou are spitting out odd little questions and not getting any decent results, so how about some fresh tactics
02:45.30ta[i]ntedtzanger it's a super secret squirrel project
02:45.39tzangerta[i]nted: :-)
02:45.40BuckRogersyeah like repeating your self
02:45.54BuckRogersso any one use those sbe t3 cards that work with linux
02:46.11tzangerBuckRogers: IIRC they wont' work with *
02:46.13tzangerthey're unchannelized
02:46.17km-manx: is it possible I need feature D or something?
02:46.27km-manx: feature D is em_w with goodies?
02:46.31BuckRogersthey say they do
02:46.43tzangerthey say they work with asterisk?
02:46.51BuckRogersyeah i was researching it
02:46.56tzangerBuckRogers: interesting
02:47.03km-uh.
02:47.08BuckRogersyeah i read it on a google news articel
02:47.23BuckRogersbut ive yet to meet anyone using them
02:47.34km-holy shit dudes
02:47.35km-get this
02:47.38BuckRogersso what do u use for asterisk if you want t3
02:47.59km-<PROTECTED>
02:47.59km-Feb 22 21:18:08 WARNING[1882]: chan_zap.c:4748 ss_thread: Got a non-Feature Group D input on channel 48.  Assuming E&M Wink instead
02:47.59km-<PROTECTED>
02:47.59km-<PROTECTED>
02:47.59km-<PROTECTED>
02:48.00BuckRogerschannelized t3
02:48.01km-<PROTECTED>
02:48.03km-<PROTECTED>
02:48.05km-<PROTECTED>
02:48.07km-I switched it to feature d
02:48.09km-and it whines and switches to em_w
02:48.11km-but then it works fine
02:48.20km-if I say em_w specifically
02:48.22km-nothing seems to work
02:48.38tzangerBuckRogers: M13 and a bunch of systems with TE410?
02:48.40BuckRogersright on (km)
02:49.01BuckRogerste410 isnt that quad span t1
02:49.06tzangeryes
02:49.17BuckRogersand split the t3 up
02:49.24BuckRogersthats alot of t1's
02:49.25tzangerBuckRogers: I did just say M13
02:49.30BuckRogersahh
02:49.40tzangerfour decent systems should be able to handle it
02:49.42BuckRogersrefresh my memory M13
02:49.42km-tzanger: whatcha think about that?
02:49.46tzanger4x8 = 32
02:49.47km-tzanger: that's kinda weird, huh?
02:49.51tzangerkm-: very
02:51.24BuckRogersmy company is considering stepping up to t3 but their not sure what type of hard ware to get the engineers
02:53.28BuckRogersthere talking dual 64 bit g5's unix systems
02:53.42BuckRogerspossible quads
02:53.47*** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
02:53.49*** join/#asterisk lilneon_ (~tj_r3@200.108.20.38)
02:54.01lilneon_hi again.. got booted
02:54.20Firestrmhi
02:54.44lilneon_Firestrm: any experience getting festival and mbrola voices up and running?
02:54.59Firestrmnone whatsoever..
02:56.10Firestrmmy struggles revolve around fighting with sipura for tech support on their product
02:56.32Firestrmi'll play with festival after i get hardware running..
02:56.41*** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
02:56.46BuckRogerssipura i emailed them over 2 weeks ago
02:57.20BuckRogersno response
02:58.11FirestrmBuckRogers, im glad i bought this one to play with before commiting it to my project.. i would NEVER use sipura gear for any commercial install..
02:58.49km-tzanger: did you find any data on that NEC system?
02:58.52BuckRogersyeah really
02:58.57km-tzanger: it's doing more weird shit
02:59.03tzangerkm-: not much but some
02:59.03BuckRogersthere gsm is not there and they advertise it
02:59.12tzangerI'm looking for something that'll work with an electra elite 48
02:59.24km-tzanger: that's the same hardware but smaller chasis I think
02:59.29km-I may have a 48, who knows
02:59.36marc32344how many call mins can be terminated on a T1?
02:59.37FirestrmBuckRogers, i would be happy if i could get rid of the echo on the pstn line..
02:59.38km-I know that the electra elite 48 sounds familiar
02:59.51tzangerbaby brother to hte IPK I think
03:00.01BuckRogersdid you go into the html interface
03:00.10BuckRogersi think there is echo cancelation
03:00.16BuckRogerson the ata
03:00.56FirestrmBuckRogers, yes.. and upgraded firmware... and read every article i can get on the matter... it seems that the echo cancelation on the s3k is just plain broken..
03:01.17tzangerkm-: hmm
03:01.20BuckRogersyeah they are really not reliable
03:01.41FirestrmBuckRogers, they are ok for residential toys.. but not for business..
03:01.48|Vulture|has anyone tried using the TDM120?
03:01.55tzangerkm-: you don't happen to have a manual for those ADPs do you?  I'm trying to see if I can notify an extension of mail
03:01.58*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
03:02.03tzangerwith Norstar you can flash, then *1+exten
03:02.23BuckRogersno they designed ciscos old ata and vonage had huge probs with it
03:03.08BuckRogersthen they switched to motorala
03:03.41FirestrmBuckRogers, i didnt realize motorola was in the spa market..
03:04.31BuckRogersyeah the vt1000
03:04.37BuckRogersis their voip ata
03:05.04*** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com)
03:05.10lilneon_hey guys, other than ebay anywhere someone can find a cheap channel bank?
03:05.24lilneon_which works with asterisk
03:05.48tzangerlilneon_: ebay.  :-)
03:05.51km-hmm
03:05.59Firestrmlilneon_, i saw one somewhere for 1000.00
03:06.05km-when I try to dial the DID to the legacy pbx, I get congestion tone from the PBX
03:06.06lilneon_tzanger: other than ebay?
03:06.48*** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com)
03:07.11vergecan someone help me with a problem I am experiencing with caller-id?
03:07.52km-oik
03:07.58km-I can make the phone ring, but only if I have the line in em_w
03:07.59km-hahaha
03:08.02km-BALLS
03:08.33km-back to square one on this
03:08.34tzangerkm-: that sounds one one amazingly frustrating problem
03:08.36km-it is definitely em_w
03:08.46km-because em_w is the only way I can get the extensions to ring
03:08.56km-but, that brings back the dial timeout problem
03:08.58toppingFirestrm: http://www.channelbanks.com//pages/specifications.html ?
03:09.34Firestrmtopping, ya i thin it was a rhino unit..
03:09.53toppingstill pretty expensive at $1495... such a little turd of a box inside i'm sure
03:10.07toppingprobably costs $75 to make
03:10.23Firestrmtopping, i know.. im tempted to devel an open hardware design..
03:10.25toppingmost of that in connectors
03:10.40km-hmm, I've got 20 mins before my window is over
03:10.46toppingthere's probably a hundred people thinking the same thing tho lol
03:11.09toppingi'm getting my dad to bring asterisk into his uni
03:11.12Firestrmtopping, yes.. but i have the technology to do it... its just time and money i lack..
03:11.20toppingwow, that's nice
03:11.50tzangerthose are nice
03:12.02toppingwhoooo
03:12.09km-tzanger: you think I need to tweak prewink/postwink times or anything like that?
03:12.16*** join/#asterisk Sedorox (~Sed@Neptune.client.wlgrv.pa.Sed6.net)
03:12.20Firestrmcany you say.. multilayer PCB's in 1 hour?
03:12.31toppingdamn sam
03:12.39tzangerkm-: you can try it but I'm at a total loss as to which direction to do it in
03:12.47km-damn
03:12.48tzangerlike I said it almost sounds like your phone system is not doing its job
03:12.50km-I need kram here
03:13.10km-he'll probably know how to work around this without having to configure the pbx differently
03:13.11toppingit's like that eyeglass place... "bifocals, in about an hour"
03:13.34shido6luv Toonie Tuesday
03:13.43Firestrmtopping, i can go right from protel to board, without even leaving my chair.. assembly requires that i walk the finished board over to the pick&place machine and then to the reflow oven
03:13.58shido62 pcs of (say it with me.. ) Shicken and Fries for 2 dollars.
03:14.12shido62 cdn dollars
03:14.19toppingFirestrm: that's nuts... that's so nice
03:14.24toppingno access any more?
03:14.31toppinghas now!
03:14.33toppingwow
03:14.34Firestrmshido6 2 northen paso's
03:14.45toppingcan you use it for your own stuff?
03:15.07tzangershido6: I am a fan of the KFC tuesday too
03:15.09Firestrmtopping, im %49 partner in the company.... ummmm... yes..
03:15.14toppinghehehe
03:15.16mikegrb~insult Beirdo
03:15.33tzangershido6: it's $2.22 though is it not
03:15.40shido6yeah yeah
03:15.42shido6:)
03:15.44tzangerI get the 3 pcs of chicken + fries with gravy + drink for like $4 something
03:16.12tzangerFirestrm: ehat does your company do that it has its own SMT P&P line
03:16.21vergecan anyone assist me with my callerid issue?
03:16.38tzangerverge: just spit it out and we'll do ewhat we can
03:16.50*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
03:17.13Firestrmtzanger, yes and we are probbly going to add VPR for bga an a month
03:17.20km-tzanger: did you find anything about configuring through the handset how the dialplan works?
03:17.35km-tzanger: for the electra elite systems
03:17.40tzangerFirestrm: nice, what are you, contract builder?
03:17.41km-there's a setup menu you can access from any phone
03:17.48km-where I can fiddle with DID's and shit
03:17.51Firestrmtzanger, aerospace r&d
03:17.52tzangerkm-: uh, no...  heh
03:17.58tzangerFirestrm: *nice*
03:18.04km-yeah, that is some cool shit
03:18.10km-make us some te405p's cheap! :)
03:18.19toppinghey, so i'm wondering about cheap commercial-grade voip handsets
03:18.19BrianR___Does anyone know if callprogress=yes can detect dialtones?
03:18.26toppingany favorites?
03:18.29Firestrmkm- im thinking about doing a run.. have the files.. need the time
03:18.36km-firestrm: hehe
03:18.38*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
03:18.44tzangeruh
03:18.44km-firestrm: if you're charging $50, I'm there!
03:18.52tzangerthe TE405P is not an open design
03:19.05Firestrmkm, how about parts and free beer at the next von?
03:19.34km-firestrm: hehe, I'd be buying it for my own fiddling, so, I don't think Mark would mind.   Parts are expensive for it though I believe
03:20.09Firestrmkm, i think the dsp is the biggie.. but we get very cut rate deals on parts..
03:20.32tzanger... I'm almost certain there is no DSP on TE405
03:20.34tzangeror any Zapata
03:20.39tzangerit's the design of the beast
03:21.16Firestrmtzanger, whats the big bga chip on the board then? i was sure it was a dsp..
03:21.22tzanger...
03:21.25tzangerthere isno BGA on the TE405P
03:21.34kpflemingthere is no DSP on those boards
03:21.44tzangerhttp://www.mixdown.ca/~asterisk/
03:21.45Firestrmtzanger, i must have different design docs..
03:21.46tzangerthat's the TE405P
03:22.02km-heh
03:22.33tzangerthere's only two main chips on there, the quad framer and the PCI bridge/FPGA
03:22.40tzangerthere's a configuration EEPROM too but that's it
03:23.22*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
03:23.22*** mode/#asterisk [+o bkw_] by ChanServ
03:23.36BrianR___poor jtn
03:23.42km-bkw_: dude!
03:23.54km-BrianR: are you the same brianr from #linuxos?
03:24.07BrianR___km-: Yes. Haven't been there for many ages though./
03:24.16Firestrmmy mistake.. i mean the FPGA part..
03:24.18toppinghey, so i'm wondering about cheap commercial-grade voip handsets... any ideas?  there's a hundred of them on voip-info, isn't there a new one that is really solid for about $100
03:24.34topping(less is fine too :-) )
03:24.45tzangerFirestrm: well as I said I'm fairly certain that the design is not open
03:24.55tzangerit's not the same card as on zapatatelephony.org
03:25.07km-brianr: I'm veneficus
03:25.11kpflemingit's not, but it would not be hard to replicate if you wanted to badly enough... but the parts are not going to be easily available
03:25.33tzangerkpfleming: the design between the T400P and TE405/TE410P are siginficantly different
03:25.36BrianR___Considering the amount of time I spend on #linuxos, it's suprising I don't remember any of the people from there :(
03:25.41kpflemingyes, very very much so
03:25.48km-bkw_: dude, you gotta wake the hell up, I need your excellent tutiledge
03:25.58tzangerMOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOSE PENIS
03:26.01km-well
03:26.02tzangeris he awake yet?
03:26.05km-hahahaha
03:26.08km-the emergency wake up call
03:26.14BrianR___km-: I've been doing some asterisk hacking lately though.. Thus my presence on this channel lately.
03:26.20km-MOOSE PENIS!MOOSE PENIS!MOOSE PENIS!MOOSE PENIS!
03:26.26km-brianr___: hehe, splaspood is here too
03:26.29Trionnismoose penis?
03:26.31km-brianr: he idles all the time though
03:26.38km-trionnis: it's the emergency bkw_ call
03:26.38Trionnis...
03:26.41Trionnisahhh
03:26.45BrianR___km-: Heh. I remember that nick..
03:26.46km-trionnis: if he's around, he'll respond to that :)
03:26.52Trionnis;)
03:26.58km-well
03:27.00BrianR___km-: I still idle on efnet... Usually in #978.
03:27.02km-I do have one thing on my side
03:27.05BrianR___km-: RevThresh hangs there also.
03:27.06km-brianr: haha
03:27.22km-brianr: I saw a guy named "Thresh" on my server on world of warcraft, was wondering if it was the same guy
03:27.32km-brianr: my ex gf used to hang out in #978 too
03:27.33BrianR___No idea if he plays warcraft...
03:27.34Trionnisanyone have an spa-1001 they want to sell cheap to a fellow * goon? :D
03:27.41BrianR___km-: Who's that?
03:27.45*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
03:27.48km-tzanger: at least there's no echo on the T1
03:27.52km-brianr: a chick named Meridith
03:27.58km-brianr: she was friends with another dude in #978
03:27.59tzangerkm-: give it a week
03:28.02BrianR___km-: Heh. I remember her..
03:28.05tzangerour echo is VERY destination dependent
03:28.07BrianR___She has the biggest cans I've ever seen.
03:28.08BrianR___Ever.
03:28.14km-yeah, she had huge boobs
03:28.23tzangerwho
03:28.24Firestrmtzanger, not open design.. but also not hard to cleanroom reverse engineer either.. the boards im thinking of are the single t1/e1 tormenta
03:28.26km-my ex g/f
03:28.29km-brianr was aware of her
03:28.37tzangerFirestrm: ahh well yes that is true enough
03:28.40km-she was a damn slut though
03:28.50tzangeryou can take apart a 6 layer board and make gerbers from them pretty easily
03:28.52km-although good in bed
03:28.53km-heh
03:28.56Trionnisyou dated my ex wife?
03:28.56km-anyway
03:28.58Trionnis??
03:29.01km-trionnis: hahahaha
03:29.03Trionnis;)
03:29.17km-damn
03:29.19km-there goes my window
03:29.26tzangeranyway
03:29.28tzangerI gotta get to bed
03:29.30Trionnisseriously tho
03:29.34Firestrmtzanger, no you dont need to copy the artwork, its usually easier to redesign based off of lessons learnt from the origial
03:29.34tzangerlater all, sorry I couldn't be more help km-
03:29.40Trionnisanyone point me toward a cheapie sipura?
03:29.44Trionnis:)
03:29.53tzangerFirestrm: seems like too much of a pain in the ass
03:29.56km-thanks tzanger
03:30.00km-I'll hit kram up later I guess
03:30.06tzangerFirestrm: at the very least make the damn thing 3.3V/5V compatible
03:30.20Firestrmtzanger, hence the time and money problem between me and dsigning an open hardware design
03:30.34*** join/#asterisk syslod (~yurplsl@65.114.0.198)
03:30.34BrianR___km-: I'm doing a legacy pbx integration and the damn ATA's for the legacy PBX don't have disconnect supervision :(
03:30.41tzangerFirestrm: that and the fact that you're going out of your way to hurt asterisk and digium
03:30.49km-brianr: that sucks, I think  I heard another person having that problem
03:30.52syslodHello
03:30.55ManxPowerBrianR___, That's pretty standard.
03:30.59tzangerfrom a business sense it might make sense but from a good karma sense it stinks
03:31.03ManxPowerPlug an Asterisk FXS port into the PBX CO port.
03:31.11tzangerespecially since you're not pushing the tech forward at all
03:31.15km-ok guys, going home, it's been 14 hours now I've been at this hell
03:31.19km-hehe
03:31.21Firestrmtzanger, i make sure to set 3.3/5v operation via a binary jumper set :)
03:31.25Trionnisonly 14?
03:31.26Trionnisn00b
03:31.36km-brianr: good hearing from you again, hope to see ya around
03:31.45km-I'm going to start coming back here now since I'll actually have an active asterisk implementation
03:31.50km-trionnis: haha, you're cruisin buddy!
03:31.55Trionnis;)
03:31.56km-don't make me get out the fishing pole!
03:31.56tzangerhahaha
03:32.01TrionnisOOH!
03:32.05Trionnisdon't tease me like that
03:32.07tzanger"I don't use wearable computers because my tie keeps getting caught in the CPU fan!"
03:32.09Trionnis=|
03:32.09km-ok, that was a world of warcraft reference
03:32.14Trionnisowww
03:32.16Trionnisdamn
03:32.22BrianR___ManxPower: Won't do it for me... There's some special signalling I can put out on the pbx fxs ports that I can't on the fxo ports..
03:32.25km-the fishing pole you can use for fishing, but you can hit people with it for like 5 damage
03:32.30Trionniso
03:32.33km-and I beat the crap out of this guy with the fishing pole and won in a duel
03:32.33Trionnisdidn't know that
03:32.36Trionnishahaha
03:32.37Trionnisnice
03:32.38Trionnis:)
03:32.44Firestrmtzanger, reverse engineering allmost allways brings the design forward.. its just part of the deal.. think of it as stepping on other to get a leg up on the competition.. Standard business practice
03:32.53km-yeah, so whenever I threaten people now I say "don't make me get out the fishing pole!"
03:32.58Trionnisnoted
03:32.59Trionnis;)
03:33.00BoRiS~seen paulc
03:33.02jbotpaulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 22h 36m 1s ago, saying: 'Is it me, or are there a handful of guys in the final 12 who are just fecking awful?'.
03:33.02km-hahaha
03:33.06BrianR___ManxPower: I think I may have to do some hacking on the code surrounding the progress stuff...
03:33.11syslodFirestrm: U building a DS3 card with DSPs?
03:33.13km-ok, later crew
03:33.16*** part/#asterisk km- (~km-@67.105.178.130)
03:33.17Trionnisadios
03:33.17tzangerFirestrm: meh.  I've been reverse engineering for over a decade professionally and probably half that again as a hobby
03:33.34BrianR___km-: Yes.. good seeing some old nicks... Smallw orld..
03:33.38BrianR___hmm.. missed him..
03:33.39tzangerunless there's SERIOUS coin to be made at it it's not generally worth starting lower than ground zero to get the design out
03:33.40syslodmeetme brokey in head???
03:34.00Firestrmsyslod, im concidering designs.. i feel that i may need an open hardware design project to play with...
03:34.05ta[i]ntedtzanger how do u start lower than ground zero? u mean moral grounds?
03:34.10tzangerno
03:34.19tzangerground zero means you know the concepts and base ideas
03:34.32tzangerbelow ground zero you're starting with the card and none of the "higher level" data
03:34.34ta[i]ntedno ground zero means ground zero
03:34.47tzangerta[i]nted: not in design.  :-)
03:34.51BrianR___It can send a DTMF tone to signal disconnect inband. I guess there's some code in CVS to take advantage of that...
03:35.00BrianR___Dialtone detection would make my life easiest, but..
03:35.15ta[i]ntedthen u mean clean slate?
03:35.19tzangerthere's a lot of art and black magic in design that unless you're there for it, you're below ground zero
03:35.23ta[i]ntedsquare one?
03:35.25tzangerta[i]nted: kind of, yeah
03:35.41syslodHell,  Now I'm getting "that is not a vaild conference"
03:35.48ta[i]ntedthere is a lot of cut n paste in design
03:35.56tzangeryes
03:36.02Firestrmtzanger, ground zero kind implies that you tried it and i blew up in your face :)
03:36.23Trionniscan anyone suggest a cheap fxs solution?
03:36.30*** join/#asterisk file (~file@mctn1-8179.nb.aliant.net)
03:36.32Trionnisi.e. ~50usd?
03:36.36tzangerFirestrm: ;-)
03:36.40tzangerwouldn't be the first time
03:36.46ta[i]ntedTrionnis PSTN?
03:36.50BrianR___$10 ebay fx0 card and a battery? :)
03:36.54tzangerI'm actually starting project where I get to parallel up VFD power sectiosn
03:37.00Trionnis?
03:37.02*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
03:37.05Trionnisdon't know that method apparently
03:37.11Firestrmtzanger, its the smoke that makes the chips work.. once you let the smoke out.. they dont work anymore...
03:37.13tzangerthinking of using serdes chipsets to syncrhonize everything
03:37.16greg_workif i have NUM = 1 in [globals], is  SetGroup(OUT_${NUM}) going to make the group OUT_1 ?
03:37.25tzangerWe're talking 250kW and up here, lots of smoke
03:37.32greg_workor do I have to use $[OUT_${NUM}] ?
03:37.34tzangerexcept it's not in the chips per se but moreso in the caps and the IGBTs
03:37.59TrionnisBrianR___: got a link for some info on that "trick" ?
03:38.00tzangerit'll be an interesting project, and I have my OHSA-approved PPE on hand
03:38.07tzangerI have a feeling I'll need it for this
03:38.08*** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net)
03:38.20ta[i]ntedPPE = Power Point Expert?
03:38.22BrianR___Trionnis: search for fxo to fxs adaptor for ready-made solutions
03:38.24ta[i]ntedlol
03:38.30tzangeranyway
03:38.31tzangerbedtime
03:38.32tzangerlater all
03:38.37*** join/#asterisk seawolf_ (~seawolf_@sea.slackwolf.com)
03:38.40Firestrmtzanger, if you want real fun.. try accidentally pointing a weather radar at the com antenna of another aircraft and turning it on for 1/2 hour..
03:38.55FirestrmS.M.O.K.E
03:39.04BrianR___You actually need slightly more than just a battery if you want real 90v style ringing
03:39.08tzangerFirestrm: gimme a job in aerospace and I bet I can find more fun than that with RADAR
03:39.14Firestrmlol
03:39.28Firestrmwhy do i feel warm all of a sudden :)
03:39.39Firestrmive had that happen to me as well..
03:39.47tzangerFirestrm: well yeah...  a system expecting -100dBm getting a few hundred Watts of RF directed at it might not take it so nicely
03:39.48Trionnisthe commercial stuff is putting the price up around the same as a new spa-1001 on ebay
03:39.50Trionnis:(
03:39.53Trionnisoh well
03:40.20*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
03:40.21tzangerespecially if you're emitting at the same wavelength the system's tuned to sniff out of noise
03:40.37ta[i]ntedthat'd break stuff
03:40.44ta[i]ntedexpensive stuff
03:40.48Firestrmtzanger, not at all... also makes a good way of killing Photoradar sets.. just need a traveling wave tube, waveguide, power supply = fun
03:41.06tzangerta[i]nted: yes, but if we knew what we were doing, it wouldn't be called research
03:41.29tzanger<-- R&D manager for a power electronics Mfg
03:41.41ta[i]ntedwhat's that one field of RF.. TEMPEST?
03:41.51Firestrmtzanger, high voltage?
03:41.53tzangerfuck tempest
03:41.57tzangerUWB is where the cool shit's at
03:42.03tzangermedium voltage
03:42.10Trionnisdoh
03:42.11ta[i]ntedUWB is UWGay
03:42.20TrionnisBPL stuff?
03:42.24*** part/#asterisk seawolf_ (~seawolf_@sea.slackwolf.com)
03:42.28tzangerlow/medium is where we play mos to fhte time but we do some 13kV+ stuff on occassion
03:42.42ta[i]ntedtzafrir_home over what medium
03:42.45tzangeryes I have HF HV stories to tell from my youth :-)
03:42.47ta[i]ntedpowerline?
03:42.50ta[i]ntedwireless?
03:42.51*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
03:42.53tzangerta[i]nted: yes this is industrial control
03:43.00tzangermotor starters and VFDs
03:43.05tzangerup to 22000HP
03:43.08jayden~Guatamala
03:43.09jbotmethinks guatamala is where the examples end
03:43.16*** join/#asterisk sricard (sricard@HSE-Montreal-ppp133166.qc.sympatico.ca)
03:43.38BrianR___is there a cvsweb for the asterisk source code?
03:43.46Firestrmtzanger, my two favorite things to play with as youth were,, baloons filled with hydrogen and tesla coils.. hmm i wonder what you could do with those two things? :)
03:43.58jayden~Galapagos
03:44.01Trionniswin a darwin award?
03:44.03Trionnis=>
03:44.08ta[i]ntedlol
03:44.14sricardcan someone help me with tdm400?
03:44.27jaydenjbot, Galapagos is an island, with turtles and stuff
03:44.28jbotokay, jayden
03:44.33FirestrmTrionnis, lol.. some close calls that did allmost result in a nomination :)
03:44.38Trionnislaf
03:44.40Trionnis:)
03:44.58sricardit was working for a while, then installed openldap and other stuff on my gentoo and now, it does not work anymore
03:45.02*** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
03:45.08Firestrmas it was.. our neibours cat was highly tramatized...
03:46.01sricardrefreshed zaptel and asterisk from cvs, recompiled the kernel, zaptel and asterisk and still the same
03:46.13Firestrmthere is NOTHING on this planet that is louder than a garbage bag filled with acetoline and oxygen, ignited by spark..
03:46.41Firestrmat 3:00 am ... im rotfl just remembering it ..
03:46.46sricardi get may errors starting with -> chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address
03:46.50syslodsricard: zttool?
03:47.43Firestrmsricard, i get that on occasion.. lsmod to make sure wcfxo is there...
03:48.03Firestrmfor some reason it occasionaly does not start properly
03:48.31sricardzttool says the card is not configured
03:49.00sricardlsmod show both zaptel and wctdm loaded
03:49.19DaminFirestrm: Try inflating a 4 foot tall Truck intertube to 80 PSI w/ Acete/O2 and throwing it onto a bonfire. ;)
03:49.36Firestrmsricard, mosprobe wcfxo then ztcfg -vvvvv
03:49.37ManxPowerUm, not thanks
03:49.51FirestrmDamin. that work too..
03:50.15sricardChannel map:
03:50.16sricardChannel 01: FXO Kewlstart (Default) (Slaves: 01)
03:50.16sricardChannel 04: FXS Kewlstart (Default) (Slaves: 04)
03:50.16sricard2 channels configured.
03:50.17DaminFirestrm: Better stand back though.. ;) We put an 8 foot wide hole in the ground with that little trick, and completely blew the fire out.
03:50.46FirestrmDamin, with the garbage bag method you can use a model rocket igniter on a timer, and be a safe (read away from arrest) distance away when it goes off
03:51.14sricardall look fine there right?
03:51.42Firestrmsricard, now try zttool
03:52.02Trionnis(read away from arrest) <-- lol
03:52.21Trionnis"who gives a shit if I lose an arm... as long as I don't get arrested!!"
03:52.30sricardFirestrm, it sees it fine
03:52.54sricardit says 4 channels and 2 configured
03:53.13Sedoroxchannel one and four
03:53.14Sedoroxthats two
03:53.27sricardnow asterisk started......   man, i hate those weird one
03:53.29sricards
03:53.53sricardzttool says 4 channels for a tdm400
03:54.02Firestrmsricard, it does that to me once in a while.. for some reason wcfxo doesnt want to start the first time..
03:54.07Sedoroxhow many modules you have on it?
03:54.16sricard2
03:54.26Sedoroxthen you only have two channels.. there are 4 available...
03:54.47sricardunderstand...
03:54.51Sedoroxhehe
03:55.10Sedoroxbecause your not using channels 2 and 3
03:56.08*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
03:56.26*** part/#asterisk lilneon_ (~tj_r3@200.108.20.38)
03:56.28sricardfor a tdm400, do you use wctdm or wcfxo?
03:56.34FirestrmTrionnis, this is what happens when you have neibourhood firecracker wars evolving into an arms race :)
03:56.40SedoroxI guess wctdm
03:57.04Firestrmsricard, wcfxo
03:57.28Firestrmwctdm is t1/e1... i think..
03:57.34sricardSedorox ->  was just reading zttool which says: Span 1: 4 total channels, 2 configured
03:57.49Sedoroxthere ya go ;-)
03:58.19sricardi don't have wcfxo and have wctdm and it is wokring....
03:59.00kpflemingwctdm is the new name for the old wcfxo driver
03:59.18sricardwctdm                  30752   2
03:59.18sricardzaptel                182272   8  [wctdm]
03:59.36sricardkpfleming, so i am ok then?
03:59.40Firestrmtime to make sup... b.b.l8tr
03:59.48Sedoroxlooks like it
04:00.00sricardthanks folks
04:00.10kpflemingsricard: probably so, yes
04:00.16|Vulture|Anyone purchse a ipVolution T1 card yet?
04:00.41sricardspent about 2-3 hours researching on the net and try stuff and i come here,i get it to work in 5 :-)
04:00.55Sedoroxhehe
04:01.09Sedoroxsricard: depends.. sometimes the other way is better.. sometimes not
04:01.19sricardthanks kpfleming,Sedorox,Firestrm
04:02.44sricardttyl everyone
04:03.24Sedoroxenjoy
04:04.16*** join/#asterisk Inv_arp (junya@adsl-8-230-5.mia.bellsouth.net)
04:07.01*** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net)
04:07.16Firestrmnp sricard
04:08.53conveyAnyone have a sample config for a Sipura 841 phone?
04:09.24sricardconvey: for the sip channel?
04:09.46conveylol
04:09.49sricardconvey: 841 are configured in asterisk as sip channels
04:09.55conveyI am a serious newbie
04:10.08*** join/#asterisk astermex (~mmg@einstein.transtelco.com.mx)
04:10.19conveyI was following the quick start guide.
04:10.28conveyI defined the phone inthe sip.conf file
04:10.49conveythen added the dial plan for the estension in the extensions.conf file
04:11.21sricardconvey: and configured the 841 by using its web interface?
04:11.26conveywhen I type sip show peers in the CLI it shows the phone as offline
04:11.43astermexAfter upgrading kernel to support data in zaptel and recompiling asterisk I always get a seg fault when I try to start asterisk. Zaptel module is loading witout any problems. Do I need to recompile something else in linux. I am using RH ES 3
04:11.51sricardconvey:: did you configure the phone itself via it's web interface
04:12.03JunK-Yconvey: past all ur output at pastebin.ca
04:12.30conveysricard: the web interface did not reveal any configurable settings.  ie sip server/ proxy, user name etc.
04:12.55sricardconvey:  i configured mine that way in 2 minutes
04:13.30conveysricard: all I see are netwokr settings, DNS, gateway, etc.
04:13.43sricardgo to ext 1 tax
04:13.46sricardtab
04:13.53conveysricard: ok
04:13.57sricardin proxy, put the ip of asterisk
04:14.28sricardin user id, put the section/user id define in sip channel
04:14.36*** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
04:14.52conveysricard: found it
04:14.56sricardin password  put the password defined in the sip channel
04:15.14sricardall you need to get it to work and register
04:15.19sricardto asterisk
04:15.41conveynow here is where we have a disconnect.  When you say sip channel, which config file is that information located?
04:15.51sricardsip.conf
04:16.08sricardi can paste my sip.conf section for my 841 here
04:16.12sricardif you want
04:16.22conveysricard: that would be great
04:16.50sricard[220]
04:16.50sricardtype=friend
04:16.50sricardhost=dynamic
04:16.50sricardcontext=toll-access
04:16.50sricardsecret=qazwsx20
04:16.51sricardmailbox=220
04:16.53sricarddtmfmode=rfc2833
04:16.55sricarddisallow=all
04:16.57sricardallow=ulaw
04:16.59sricardthat's it
04:17.19sricardit is extension 220 in my house
04:17.27|Vulture|pastebin.ca pls
04:17.46sricardwhat is pastebin.ca?
04:18.08conveythe [220] is equivilant to my [phone1] correct?
04:18.45sricardif you put a name there, you'll need alias in your extensions.conf
04:19.18sricardbut it should still work
04:19.20Inv_arpsricard: pastebin.ca is a site to paste stuff
04:19.42sricardInv_arp: never used that one
04:20.30sricardInv_arp: i get it now, thx will do next time
04:21.14conveysricard: [phone1] is not aliased
04:21.52conveysricard: I am just using phone1 as a device
04:22.22sricardconvey: you could still use it but you will have to define an extension with a number and use DIAL(SIP/Phone1) to reach it
04:22.51sricardconvey: its fine, it should work
04:23.03*** join/#asterisk jayden (~ircatjerr@adsl-69-209-134-225.dsl.sfldmi.ameritech.net)
04:23.15sricardconvey: i always use numbers (right or wrong)
04:23.41conveysricard: yes the example you have stated DIAL(SIP/Phone1) is exactly waht I am using
04:24.38sricardconvey: so if you put phone1 as the user id and what ever password you define in sip.conf, your 841 should register with asterisk
04:25.14*** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu)
04:25.32sricardconvey: all in the Ext 1: tab on the Web interface (Advance mode) of the SPA-841
04:25.50*** join/#asterisk lildivil (user@ool-18bc24d7.dyn.optonline.net)
04:26.36conveysricard: we have a registration
04:26.45conveysricard: THanks for the guidance
04:27.19sricardconvey: np my friend, happy to help someone else for once :-)
04:27.32sricardbye for now
04:27.43conveysricard: thanks again, bye
04:28.20lildivilgood evening all...
04:31.34*** join/#asterisk jayden (~ircatjerr@adsl-69-209-134-225.dsl.sfldmi.ameritech.net)
04:31.35*** join/#asterisk techie (gus@asterisk.horizonte.us)
04:32.22*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
04:46.02*** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com)
04:48.10jaydenManx
04:50.01ManxPowerNobody has any problems?  Cool!  My work here is done!
04:50.24jaydennice
04:50.39mrgobyManx: my asterisk dont work none, what's wrong with it ?
04:50.41jaydenI am fighting with screwed up header files
04:50.53toppingOT: http://www.bash.org/?301963
04:51.03ManxPowerjayden, You are using RedHat FC3, huh?
04:51.05jaydenand a guy in #asterisk-dev that can't figure out how to msg nickserve
04:51.10jaydenummmmm
04:51.16jaydennot on this box
04:51.16ManxPowerjayden, That's pretty sad.
04:51.22ManxPowerjayden, I don't really do developer questions.
04:51.35ManxPowermrgoby, First you need to find a goat.
04:51.47jaydenhehe
04:52.04jayden~Gatamala
04:52.23jaydensigh
04:52.27jayden~mbtt
04:52.28jbotwell, mbtt is Mavis Beacons Teaches Typing
04:52.31ManxPowermrgoby, The next step is kind of complicated.  It's best to contact your local Satanist group for the proper prodecure.
04:52.33jayden~Guatamala
04:55.22*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
04:55.23*** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon....
04:55.32ManxPower~clec
04:55.33jbotextra, extra, read all about it, clec is Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's ...
04:56.12ManxPowerjbot needs a bigger buffer.
04:56.46mrgobyjayden: was that 'sure' directed at me ?
04:56.50jaydeny
04:57.10iceypManxPower how do you add it for your own extensions?
04:57.13mrgobyhints ??
04:57.22jaydenummmm
04:57.25ManxPoweriCEBrkr, I don't use CDRs.
04:57.30ManxPowerI don't bill users for calls.
04:57.31iceypI've added username, callerid, fromuser and none of them do it
04:57.38ManxPowerThe LD company does that.
04:57.41jaydenI can't think of an app that will do that....
04:57.42iceypI'm trying to build a web interface
04:57.44*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
04:58.07bjohnsonhow can I get * to not use chan_oss (and not use my /dev/dsp)
04:58.20PyroStevehey guys
04:58.20PyroStevei just got my iaxy
04:58.20PyroSteveits just simply got a fxs module
04:58.20PyroStevecan I insert an fxo moudule instead
04:58.22ManxPowerbjohnson, noload => chan_oss.so in /etc/asterisk/modules.conf.
05:16.38*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
05:16.38*** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon....
05:16.44shepherdi've heard soooo many stories
05:16.47shepherd:D
05:16.51bkw_shepherd, about me?
05:16.58shepherdmmmhmm
05:17.05bkw_their is a video out there of me drunk as hell at astricon
05:17.08shepherd<-- matt
05:17.11bkw_i tend to repeat myself when i'm drunk
05:17.14bkw_shepherd, you ninny
05:17.16brc_*there*
05:17.18mikegrbbkw_: no, they grow right by my porch, just for you
05:17.18bkw_haha
05:17.19mrgobyhttp://www.voip-info.org/wiki-Asterisk+cmd+Rpt
05:17.21mikegrband maybe Beirdo
05:17.29bkw_haha
05:17.45mikegrbwhere is next astricon going to be and what month
05:17.47mrgoby~cluebat jayden
05:17.49jbotACTION pulls out a ClueBat (tm) and thwaps jayden.
05:17.50bkw_oct 2005
05:17.55mikegrbmay be able to get new employer to foot the bill
05:17.56shepherdchan_ham ?
05:17.56bkw_cluecon is june 2005
05:17.57shepherdhaha
05:18.00mikegrbthat would be most excelent
05:18.00mrgoby:-)   that's more like it
05:18.09bkw_shepherd, you have to write about cluecon when we get it all lined up fully
05:18.24bkw_thats our task this week to get it more filled in and lined up
05:18.38jaydenhey bkw, who from asterlink is in detroit... I saw the address was here
05:18.42mikegrbwhere are each going to be?
05:18.48mrgobyi'm in ann arbor
05:18.51bkw_jayden, the payroll lady
05:18.56bkw_the one that PAYS ME
05:18.58bkw_;)
05:19.10mrgobyBIG UPS TO DA U.P. !
05:19.18mikegrbhmm california
05:19.20bkw_UPS?
05:19.22bkw_I hate UPS
05:19.28bkw_they play football with my loot all the time
05:19.33shepherdwhat is cluecon?
05:19.41bkw_shepherd, a more hardcore dev conf
05:19.56mrgobythey played flippy floppy with my servers
05:20.02bkw_ya
05:20.04bkw_I hate that
05:20.06bkw_pisses me off
05:20.14mishehuI can't stand UPS.  even if I order for 10am delivery, they'll still show up at 18:00 and tell me that I am in a residential area, thus the 10:00 delivery doesn't apply
05:20.14mrgobylove to see them rack ears all bent
05:20.17bkw_they left 600 bucks worth of shit on my porch one friday
05:20.19bkw_I was out of town
05:20.24bkw_came home on sunday it was gone
05:20.24mishehufedex on the other hand tends to arrive on time.
05:20.25bkw_the best part is
05:20.29bkw_they said I signed for it
05:20.32Beirdo<PROTECTED>
05:20.32bkw_funny part is
05:20.40bkw_they infact did have "Brian West" on the sig
05:20.47mikegrbBeirdo: dunno, what?
05:20.48bkw_but I never sign UPS with my name
05:20.50shepherdfedex loses a lot of packages though
05:20.54Beirdoheh
05:20.55bkw_I always sign "bugs bunny"
05:20.57shepherdTRUST ME, I KNOW
05:21.03mrgobythat is funny as hell
05:21.13bkw_so I told them I didn't sign for that
05:21.16bkw_they were like we have your sig
05:21.18mrgobyso your only defense is that it was wrong because it was right
05:21.19mrgobyhaha
05:21.20bkw_i'm like HELL YOU DO
05:21.29mishehubkw_: oooh you wascilly wabbit
05:21.39bkw_well I sign all credit card stuff with looney toon charaters
05:21.47bkw_so if someone has my real sig
05:21.49bkw_I know its fake
05:21.51bkw_;)
05:21.52mrgobymy servers are named after muppets
05:21.59mrgoby:-)
05:22.05bkw_infact the bugs bunny thing with UPS
05:22.07bkw_saved my ass
05:22.14mrgobyyeah
05:22.17bkw_I pulled up the last 10 tracking numbers on shit I ordered
05:22.19shepherdthat's better than choochies
05:22.20bkw_all "Bugs Bunny"
05:22.22*** join/#asterisk t3t (~t3t@bar.pangalacticgargleblaster.com)
05:22.42mishehumrgoby: which one did you name as "swedishchef" or "borkborkbork" ?
05:22.45bkw_t3t so when are we gonna go out for sone Pan Galactic Gargle Blasters?
05:22.47bkw_you buying or me?
05:22.54bkw_s/sone/somme/
05:22.57bkw_fuck I  can't type
05:23.01bkw_I think I had too many already
05:23.21t3thave another http://pangalacticgargleblaster.com/
05:23.27mrgobyneither of them exist yet :0)  but soon to come
05:23.29bkw_LOVE IT
05:23.35bkw_42 42's
05:23.35bkw_haha
05:23.39bkw_LOVE IT
05:23.42bkw_can't wait for the movie
05:24.01t3tI was bored a couple of years ago...
05:24.14bkw_3 letters baby
05:24.16bkw_THREE
05:24.22t3tthat's hard to come by
05:24.23bkw_check the expire date baby
05:24.28bkw_had it since 98
05:24.30t3ti had to settle for a 3-letter .us name
05:24.36Firestrmmikegrb, http://www.vrl.ca/pic19.jpg that was taken over my shoulder as i was running from the fire
05:24.39bkw_didn't think they alloed 3 letters in .us
05:24.44bkw_thought it was a 4 letter min
05:24.49bkw_I want fuck.us
05:24.53t3tI'm an idiot for not getting more domains in '94 when they were free
05:24.58shepherdbkw_: i have hijacked.us
05:25.05bkw_eww debian
05:25.06bkw_get out
05:25.08trymdebian owns
05:25.15bkw_no asterlinux ownz
05:25.15trymso.. what is the thing.. redhat? haha
05:25.18t3tThe rule may be 4, but i have a 3...
05:25.21bkw_roothat
05:25.21mikegrbFirestrm: wow
05:25.29bkw_or root that
05:25.29mrgobyyou a gentoo man, bkw ?
05:25.29bkw_haha
05:25.32trymi also own whiteho.us
05:25.34t3tshepherd: nice catch
05:25.37mrgobyor woman ?
05:25.40bkw_mrgoby, not exclusivly
05:25.43bkw_but I do like it
05:25.47bkw_I like solaris and freebsd more
05:25.53t3ttrym: you missed the 'e' :)
05:25.57mrgobyyikes ...   solaris ?
05:26.01bkw_ya
05:26.10mrgobyi guess i cant speak on it... i only ever used it in school
05:26.18t3tslowlaris, bkw?
05:26.18bkw_as a server its great
05:26.19Firestrmmikegrb, and this http://www.vrl.ca/pic13.jpg is what it looked like at night.. not a trick photo.. this is real.. very eery
05:26.20bkw_as a desktop it sucks
05:26.23loudHINT: do not type reset on the CISCO 7960 cli .. thinking its "reboot" ..
05:26.24mrgobyyes
05:26.24bkw_it might be slow but tell ya
05:26.35bkw_it rocks
05:26.38bkw_and is solid
05:26.52mrgobyand expensive
05:27.00bkw_shit I have seen solaris boxes sail along on shit that would make a linux box blow up
05:27.04bkw_linux will not scale
05:27.11bkw_neither will asterisk
05:27.14Firestrmmikegrb, its very disconcerting to have a mushroom cloud over your home town
05:27.15bkw_unless you throw more boxes at it
05:27.25t3tbkw: I agree... most of my boxes are fbsd
05:27.32mrgobyhehe
05:27.52Firestrmi tried fbsd once..... once..
05:27.53bkw_haha
05:27.56bkw_my address is on my whois
05:27.59bkw_ship away baby
05:28.02bkw_Firestrm, you gave up too soon
05:28.03mrgobyUPS
05:28.07bkw_asterlinux is very BSDish
05:28.07mrgobysend em UPS
05:28.08t3tslowlaris is too expensive for what you get... besides it took them almost 15 years just to get most of the init scripts to work
05:28.25Firestrmbkw_, i felt like a fish out of water..
05:28.40bkw_Firestrm, you will for about a week
05:28.53bkw_I switched from windows to OSX and guess what i didn't miss a beat
05:29.04shepherdassturdlinux
05:29.08shepherd:D
05:29.09t3tlooking at making the leap soon...
05:29.10bkw_shepherd, be nice
05:29.16bkw_t3t, you won't regret it
05:29.16Firestrmbkw_, it was like trying to drive a right hand steer car for the first time
05:29.28t3tthat's what people keep telling me
05:29.42t3tI used to admin about 15 macs at a newspaper
05:29.45bkw_t3t it took  me less than a day to get used ot it
05:29.47freddyFirestrm: all the cars here are right hand steer!
05:29.52mrgobyi'm out....    power to the people !
05:29.54t3tos[7-9] suck
05:29.56*** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net)
05:29.57bkw_yes
05:30.01bkw_OS X is the only reason I own a mac
05:30.03freddyFirestrm: its not that hard really
05:30.04bkw_if it were not for that
05:30.07bkw_i wouldn't even think of it
05:30.08shepherdyeah
05:30.11bkw_but Jobs did it right
05:30.14shepherdi refuse to run x11 on my mac
05:30.18bkw_haha
05:30.19shepherdthough
05:30.22bkw_its actually cool
05:30.30shepherdNO!
05:30.31bkw_rootless X
05:30.33t3tI just wish that apple didn't mess with the fbsd core as much
05:30.43bkw_t3t, well its all good
05:30.47bkw_atleast they have jordan hubbard
05:30.47Firestrmfreddy, when i visited uk, i rented a motorcycle, at least then all i had to do is get out of the way of ppl driving towards me in my lane :)
05:31.12bkw_when he left the freebsd project it kinda isn't as good as it used to be
05:31.12t3tI guess it's just about getting used to it
05:31.42bkw_but i'm sooooooo geeky I painted a white apple on my luggage
05:31.49bkw_ya know
05:31.58bkw_dont wanna loose it when he goes to VON in two weeks
05:32.15shepherdheh
05:32.16t3tbetter than a beastie... never know what dhs would do with that...
05:32.25*** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg)
05:32.37shepherdbkw_: do you have a powerbook?
05:32.46bkw_ibook and imac
05:32.58shepherdk
05:33.01letherglovhahha
05:33.02bkw_in all honesty the system profiler says this is a powerbook
05:33.11letherglovvon's in san jose yeah?
05:33.15bkw_yep
05:33.18letherglovgoodtimes
05:33.22t3tI wish that I could justify the $$ for VON... it should be a good show this year
05:33.31letherglovI volunteered at the tech museum (across the street) for a couple years
05:33.36bkw_you'll love it
05:33.39shepherdm$ is going security crazy
05:33.44bkw_haha
05:33.45t3tFirestrm: they lost their mind 5 years ago when they were contemplating how to do updates
05:33.47shepherdbut they are doing it all about the wrong way
05:33.52bkw_the fact that OS X gets the fuck out of my way
05:33.56bkw_and lets me do what I wanna get done
05:34.04letherglovbkw_, enough mac talk
05:34.08bkw_no no
05:34.08shepherdi like their creative top down approach
05:34.11letherglov:-P
05:34.12bkw_MORE MAC talk
05:34.25bkw_I have apples
05:34.27bkw_you ninny
05:34.30bkw_a pair of them
05:34.34bkw_and you know you wanna lick em
05:34.41*** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net)
05:34.48Firestrmt3t, sadly i have to agree, but the latest, were not going to patch unauthorized boxes stupidity had put a nail in the M$ coffin..
05:35.03t3tFirestrm: they'll change their tune... again
05:35.16t3tThis idea was kicked around two years ago
05:35.42t3tI was one of a few people who convinced those in charge not to do it
05:35.44Firestrmt3t, only after there is a sea of rootkited unpatched machines
05:35.48shepherdwell.. i heard somewhere that m$ signed away their rights to use any type of unix based system
05:35.51t3tsounds like they changed the guard again...
05:36.44t3tFirestrm: then is now... the sea is churning and bubbling as we speak
05:37.13Firestrmt3t, yes, but if you can even imagine it... its going to get worse..
05:37.22t3tshepherd: haven't heard that... they did stop development of the POSIX subsystem of NT a few years back though
05:38.12t3tFirestrm: I don't think that there is currently an avenue for them to get better... if MS gives up, the consumer network providers are are only hope at containing the problem
05:39.19|Vulture|anyone seen any info on the new ipVolution TDMs?
05:39.34*** join/#asterisk Tough_Nuts (~Tough_Nut@m19105e42.tmodns.net)
05:39.36t3tNope, what do they do?
05:39.46DJ-Pyro|Vulture|: someone brought it up earlier tonight
05:40.29|Vulture|they seem pretty good... in concept
05:40.29|Vulture|http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm
05:40.50shepherdwe'll see :)
05:40.58|Vulture|onboard echo cancel and codec DSP
05:40.59Firestrmt3t, the only way to really fix this problem, is to remove the right of software companies to disclaim liability for damages due to poor coding.. Software is the only place that one can currently get away with this. If an auto manufacturer built a car with a flaw that caused it to randomly swerve for example, they would have to recall it and fix it.
05:41.06*** join/#asterisk clive- (~pirch@myw-stp-66-18-86-218.sentechsa.net)
05:41.37t3tFirestrm: I thought that the insurance companies would have had that changed a long time ago...
05:41.45DJ-Pyrowith an add-on daughter card that adds another 4 T1 ports
05:41.52t3t... clearly my crystal ball is broken.
05:42.28|Vulture|DJ-Pyro: doesn't that seem too good to be true?
05:43.01Firestrmt3t, personal injury lawyers prevented that.. problem is that software bugs do cause injury in the form of time and money, but because it isnt a broken leg or somthing physical, judges dont see that..
05:43.07ManxPowerI don't know if those cards have shipped yet or not.
05:43.10|Vulture|wonder if it will run off the same zaptel drivers
05:43.21t3t|Vulture|: It looks like a rendering of a pcb with some rj11 connectors on it... I wonder what the finished product will actually do
05:43.23shepherdFirestrm: it's not the car manufacture's fault if your car gets stolen
05:43.31|Vulture|ManxPower: it says the duals were suppose to ship in Jan.. but they are still in Pre-Order
05:43.50greg_workFirestrm: i've always thought that would be done well with with engineering.. have a policy where you require software to be approved by a P.Eng.  If it fails, the P.Eng. is held liable, and depending on the circumstances may face harsh punishments (not unlike what happens to hte engineer that approves a bridge that collapses under its own weight)
05:43.54|Vulture|t3t: yea the picture is a joke, I was talking about the actual stats
05:44.04ManxPowerHardware design, as Digiumn found out, is much harder than it looks.
05:44.05`SauronMurf.
05:44.10t3tFirestrm: maybe that's a good thing... have you ever seen a federal judge try to figure out what a trojan horse controlling a pc can do
05:44.24shepherdthey would have to rewrite most of asterisk to get it to work too
05:44.29greg_workof course, the chances of finding a P.Eng to sign off on windows are about as good as microsoft licencing it as GPL
05:44.38Firestrmshepherd, yes and no. If they design a lock that is so flimsey that you can break it with a plastic knife (see toyota) i think they should at least be partially responsible.. its all about due diligence
05:44.51bkw_windows is missing something
05:44.53`SauronAnyone know if there's integer versions of most voice codecs out there?
05:45.00Firestrmt3t , now that would be amusing :)
05:45.00shepherdi can't use windows anymore
05:45.05shepherdi tried the other day
05:45.06`SauronMy sparse google attempts earlier today didn't find much information
05:45.22shepherdi get pissed waiting on it
05:45.22shepherdhaha
05:45.33t3t|Vulture|: what actual stats? all i see is some bullet points beneath a picture and some marketing-speak.  Until someone unrelated to the company gets to pound on it, it's vaporware for me...
05:45.50|Vulture|t3t: good point
05:46.11`Saurongreg
05:46.13`Sauronwhat up
05:47.32greg_work|Vulture|: you know what's funny about that? asterisk was originally created (once CPUs were fast enough) so you didn't require onboard DSPs (because they're expensive) :)
05:47.45greg_work`Sauron: not much. about to go home
05:48.07`SauronYou just work there, or you own the place?
05:48.09shepherdbut now.. we're gonna have to move towards them again
05:48.18`Sauronlate work hours is generally a sign of owning a shop. :)
05:48.30greg_workmy dad owns it
05:48.48Firestrmgreg_work, even worse... slave labour..
05:48.49`SauronEr, probably nevermind.. I think I remember.. yeah. Your lastname ~ company name
05:48.53`Saurongrin
05:49.20greg_worktonight i was mostly working on AMP stuff though .. so it guess you could consider i'm doing it partially on company time, partly personal
05:49.55greg_worki didnt even get doing ANY programming till 4 though :p i hate days like that
05:51.09greg_work(actually if any of you use AMP, go read my last couple posts on amportal-users and give me suggestions ;) )
05:51.43t3tDoes anyone know how 800# calls from payphones are billed.  Is there still a ~$.25 surcharge in the US?
05:52.54*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:52.54greg_work(if you don't use amp, basically i added some stuff to make pattern-based routing that can use various trunks with a configurable priority .. ie, for local calls, use ZAP lines then try voip, and for long distance, use voip then zap)
05:53.29shepherdsweet
05:54.22greg_workt3t: by law, toll-free, operator, and 911 calls from payphones are free. (i dunno if the payphone owner pays or not though)
05:55.03`SauronI'd imagine receiving 800 owner pays
05:55.05t3tthanks greg, I meant to ask if I (as the 800# owner) get charged extra for the calls
05:55.32*** join/#asterisk MarkK (~got@65-100-56-164.ptld.qwest.net)
05:55.35t3t`Sauron: that makes sense, but how does a provider like iaxtel or nufone pass it on?
05:55.57`SauronHum, dunno :)
05:56.04t3tI'll just have to wait until jj is on and ask him
06:03.19clive-does anyone have any pointers on iax2 registrations with iaxclient?  I get "Inappropriate authentication received
06:03.20clive-"
06:04.37t3tclive-: is your password  blank?
06:05.51*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
06:08.41t3tclive-: what version of * are you running?
06:09.52clive-t3t I have a password,  I am thinking I should have auth=rsa rather than auth=plaintext
06:10.03t3tgive it a try
06:10.24t3tfrom the  source it looks like the md5 password or the plaintext password is 0 length
06:11.33clive-my client is iaxclient
06:11.55clive-rsa gives the same eroror
06:12.07t3tclive-: I haven't used it in a while.. I use firefly currently
06:12.17t3twhat version of asterisk do you have?
06:12.26clive-I woudl expect firefly works the saem
06:12.32clive-verison..hmm, old,,,lets check
06:13.02t3tprobably over a year, right?
06:14.01clive-eek
06:14.02clive-CVS-06/11/04-03
06:14.19t3tyou may want to think about updating :)
06:14.31clive-let me do it quick
06:14.32clive-:)
06:14.44clive-will that fix my login trouble?
06:14.54t3tthis is gonna be good... a quick update of * from 2003 to present :)
06:15.14t3tI assume... or at least you'll get a more descriptive error message
06:15.21clive-lol
06:15.53clive-let me quickly figure out how to co version 1.0.2 or something
06:16.25`Sauronjust co cvs-head
06:16.27`Sauron;)
06:16.38clive-is head stable (ish)
06:16.40t3thttp://www.voip-info.org/tiki-index.php?page=Asterisk+Download
06:17.00`Sauronstable enough for me at home
06:17.24t3tclive-: It's been stable for me at home and work (2/11/05 and 12/25/04)
06:17.51t3tclive-: The bad part of head is you don't know exactly what you'll get until you try it
06:18.39djinWhat's the difference between 'friend' and 'peer' in sip.conf??
06:19.07t3ta friend is someone you trust, a peer is just someone the same age as you... oops. wrong context again...
06:19.16t3tfriend: both user and peer
06:19.36t3tpeer: outgoing connection to another server
06:19.37shepherdoh peer me!
06:20.05t3tuser: incoming connection
06:20.34Beirdo~httpdtype www22.verizon.com
06:20.40t3tdjin: be careful with the names in [].  Some services expect them to be an EXACT string (capitolization is important)
06:20.46Beirdoahhh, that explains a lot
06:21.11djinwhat setting is used to link two * servers to each other?
06:21.26t3tyou could use friend on both
06:21.51djinok, thanks.
06:21.55`SauroncapitAlization is important ;)
06:22.24clive-oh boy I am rusty at this...since 2003:)
06:22.56t3t`Sauron: Nothing like interactive live spell checking
06:23.00t3tthanks
06:23.10`Sauronyou're welcome :)
06:23.25t3thow do you know I wasn't talking about a governmental unit?
06:23.34bkw_t3t talkin about Mac's again?
06:23.40t3tsure
06:23.52bkw_live spell checking happens in just about every application
06:23.54t3tmacworld is only a few months away
06:23.54bkw_its in the API
06:24.00t3tno way
06:24.03bkw_yes way
06:24.07bkw_my xchat spell checks on me
06:24.09bkw_its nice
06:24.20bkw_everything thats cocoa baseed
06:24.23t3twhat does it do if you mess up a word?
06:24.28bkw_red underline
06:24.39bkw_you use emacs?
06:24.41t3tthat's scary-useful
06:24.52t3tnever got into it, unfortunately
06:25.12t3tI haven't had anyone able to articulate its advantages clearly enough
06:25.26bkw_I would have to show you
06:25.30bkw_I wouldn't knwo if someone didn't show me
06:25.37bkw_I thought emacs was evil
06:25.41bkw_till i was showed the light
06:25.44shepherdyahoo needs to make a decient port of messenger for the mac :(
06:25.47t3tI just don't know what it can do
06:26.05bkw_hehe
06:26.05clive-t3t haa, got it, auth needs to be md5  meanwhile I have a new asterisk versuion
06:26.08t3tsomeone must have created a 'this is why you need emacs' page for the laxy
06:26.22t3tda###it.. ^lazy
06:26.28bkw_haha no it turns in to an editor war
06:26.39bkw_even with live spell checking I still fuckup because i'm lazy
06:26.39bkw_haha
06:26.44bkw_fuck sdfow
06:26.46bkw_haha
06:26.53shepherdvi4life!
06:28.11t3tOther than context-highlighting, regex find/replace, and line numbering what does it do that's so special?
06:28.26bkw_its just nice
06:28.33t3tsimple
06:28.35bkw_I use maybe a tiny bit of it
06:28.39bkw_but its easy
06:28.46t3tnot very informative, but simple
06:28.59bkw_well simple to me is hard as hell for others it seems
06:29.00bkw_haha
06:29.12bkw_when I first meet anthm  before I worked here for him
06:29.15bkw_I was using pico
06:29.17bkw_eww ewww eww
06:29.22t3tdidn't you learn c to mess with *?
06:29.26bkw_he showed me emacs.. and I LOVE IT
06:29.29bkw_t3t yes
06:29.42bkw_jumped right in
06:29.44drumkillawell damn, for code, almost anything would be better than pico
06:29.47t3tnot many out there like you, bkw_...
06:29.57bkw_I still have stuff I dont know in C
06:29.59bkw_but I learn
06:30.05t3tI generally use joe or vi for farting around
06:30.21drumkillai'm a fan of vim ...
06:30.21bkw_I like emacs becuase I can do emacs /usr/src/asterisk/
06:30.36bkw_it shows me a directory.. or emacso /usr/src/asterisk/*.c
06:30.40bkw_and it will open every C file
06:30.46bkw_er emacso.. haha
06:30.48bkw_funny me
06:31.02t3tyou had me.. I thought it was a special emacs command
06:31.11drumkillasounds like a super hero
06:31.16bkw_type'o
06:31.25bkw_emacso the super editor
06:31.37shepherdbkw_: you should irc from emacs
06:31.44t3tI use homesite for most moderate lifting
06:31.44bkw_you can play tetris in emacs
06:31.48bkw_meta-x
06:31.50bkw_tetris
06:31.52bkw_it starts
06:32.00DJ-Pyrosomeone was running an irc bot in emacs on another channel
06:32.02shepherdemacs should replace hurd in a few years
06:32.31*** join/#asterisk rvhi (~rv@66.175.65.89)
06:32.46rvhipolycom uses standard sip call park
06:33.04t3tat any rate, I shoudl check it out... care to put up a screenshot of how you have it laid out, bkw?
06:33.04rvhianyone is working on making this work?
06:33.06*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
06:33.12kpflemingbkw_: i have been using since before there was GNU... before Linux was even a dream, before there were PCs... it's a wonderful editor in innumerable ways
06:33.16t3t^ should check it out that is...
06:33.18DJ-Pyrorvhi: you refering to the park soft key not actually parking?
06:33.29rvhiyes
06:33.51DJ-Pyrook, that means that I can stop messing with the configs trying to find out why it doesn't work :)
06:35.28bkw_kpfleming, its nice isn't it
06:35.37kpflemingyep
06:35.44bkw_i'm glad tony made me learn to use it
06:35.49bkw_he said it was a requirement for me to use
06:35.53bkw_hehe
06:36.12bkw_kpfleming, we need to get you, tony, mark, me and twisted in the conf and smack these patches around on the bug trakcer
06:36.18bkw_before mark closes the darn thing down
06:36.20kpflemingused to work entirely in emacs on a green screen terminal all day, never left it for any reason... even had a forum (bbs) interface in an emacs buffer, along with email and other cool stuff
06:36.39kpflemingthe conf calls do not seem to accomplish much at all
06:37.25bkw_well we have in the past
06:37.32bkw_we have had 8 hour conf calls before
06:37.36kpflemingi wish we could figure out how to get the conference calls to work better, the delays in conversation are just so long
06:37.37bkw_and cleared out 30-40 bucks
06:37.40kpflemingyeah, i know, i've been there
06:37.44bkw_ya we are working on that too
06:37.49bkw_with stevekstevek
06:37.53bkw_we might use his conf thingy next week
06:37.54t3tbucks?
06:37.56kpflemingbut these are not simple patches we are talking about here, they require a lot more thought and discussion
06:38.01bkw_er bugs
06:38.13t3tneed a word context checker
06:38.15bkw_kpfleming, exactly maybe we can line out a few of the major ones next week
06:38.40bkw_kpfleming, we could hog tie mark at VON
06:38.52DJ-Pyronew conference app?
06:39.06kpflemingi hope so, we need a very significant shift in the way things get done if we are going to accomplish what we are all trying to do
06:39.17t3tassuming that you could touch him.. I have a feeling that mark will be a little busy with the minions at VON
06:39.26bkw_t3t, RIIIGHT
06:39.27bkw_haha
06:39.28kpflemingi could spend the next three months cleaning and optimizing code, but i don't, because it would take _forever_ to get the patches merged
06:39.35bkw_kpfleming, yep
06:39.41bkw_we have got to speed this along
06:39.48bkw_that is what caused asterisk to fork in the first place
06:40.17kpflemingyeah, we've talked about all this before, including two more people have commit privs for stuff that they didn't write, and nothing has changed yet :-(
06:43.25`Sauronj/k
06:44.16t3tok, it's late and I'm getting giddy.  Y'all have a good night now.
06:44.21`SauronI understand their licensing thingy before you can submit code, but my understanding is, that even after going through all the paperwork, code doesn't get added even after it was submitted.
06:44.47`SauronAnd I really need to go to bed, even though I just got some coding inspiration.
06:44.49`SauronUgh.
06:51.20*** join/#asterisk akrall (~akrall@201.128.92.118)
06:51.35akrallGuys.. anybody using festival on asterisk? and using the text2wave method?
06:53.26BeirdoJeez, US local calling is so fucked up
06:55.13goatmilklanguage, language...
06:57.05*** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net)
06:58.02odie_floconanybody use IAXtalk.com products?
06:58.09odie_floconie the AT-320ED
06:58.15Beirdogoatmilk: what you disagree?
06:59.19akrallGuys.. anybody using festival on asterisk? and using the text2wave method?
06:59.19*** join/#asterisk tandrews (~tandrews@mail.grok.co.za)
06:59.27goatmilkagree or disagree i'm not going to talk like that in here... not very respectful at all.
06:59.34odie_floconI have never used it.
06:59.50goatmilkakrall: we saw you, and it's late that's why no one is responding.  I have heard of people using festival
07:00.50tandrewsmorning
07:00.53Beirdouhhh, and who is it I'm supposed to be respecting in the US telco market?  the local calling areas are borked beyond belief
07:01.21goatmilkBeirdo: i'm talking about cursing, not your opinions on telephone service.
07:01.27akrallgoatmilk: sorry ofr repeating... too late and lack of cofee
07:01.29akrall:)
07:01.36goatmilkakrall: np buddy
07:02.06Beirdomeh, whatever
07:02.23Beirdocursing is not disrespectful.  Crude and stupid perhaps.
07:02.27goatmilkakrall: there is a script out there that does weather info over the phone.. you dial your zip code and it'll talk it out to you.  maybe this will help you?
07:02.43Beirdobut anyways, I apologize if it offended
07:03.11odie_floconinteresting
07:03.23goatmilkBeirdo: I do myself, but not in this channel.
07:04.16odie_floconis there any good Voip Providers in the States?
07:04.16tandrewsquick h323 question - What's the difference between the h323 implementation on inaccessnetworks and the one included in asterisk in channels/h323 ? (And which is the most sensible to use?)
07:04.24odie_floconI need a Utah number.
07:05.04odie_floconwho is inaccessnetowrks?
07:05.33tandrewshttp://www.inaccessnetworks.com/projects/asterisk-oh323
07:05.49tandrews"H.323 support for ASTERISK PBX using the OpenH323 library"
07:06.37tandrewsThey seem to provide a module which is in effect a wrapper for the C++ stuff
07:07.25odie_floconthe h323 implementation should be close to the same.
07:07.43odie_floconsince H.323 is a standard.
07:08.24odie_floconimho you should use *'s H.323
07:09.16odie_flocontandrews are you in NZ?
07:09.32tandrewsthe documentation on inaccess networks on how to install looked much better though ;)
07:09.43tandrewsno odie_flocon, I'm in ZA
07:09.52odie_floconohh sorry.
07:10.11tandrewssimilar accent though, easy mistake ;)
07:10.32odie_floconmy friend used to work for Siemens in ZA
07:10.58tandrewsk
07:11.44odie_floconthe H.323 implementation is already built into *
07:11.54odie_floconso you should need much to install. just configuration.
07:11.58*** part/#asterisk akrall (~akrall@201.128.92.118)
07:13.05tandrewswell, the README says "You must run Open H.323 v1.15.1 and PWLib v1.8.1. All other versions are not supported."
07:13.11tandrewsyuk
07:13.22odie_floconahh
07:13.39odie_floconhmmm.
07:13.48odie_floconI honestly can't tell you
07:14.03odie_floconI have only used SIP, and IAX.
07:14.50tandrewsThey're simple in comparison by the looks of things
07:14.56*** join/#asterisk TheEmperor (~mattn@203.121.47.100)
07:16.43clive-tandrews howzit
07:16.58tandrewshi clive-
07:18.14tandrewsJerJer are you here ?
07:18.14clive-nice to find anothetr african in here:)
07:18.21tandrews:)
07:19.29*** join/#asterisk schurig (~schurig@p54B2818C.dip0.t-ipconnect.de)
07:21.52odie_floconso tandrews? do you have to pay by the minute to be online?
07:22.15tandrewsno odie_flocon I have a leased line at home
07:22.27odie_floconyou have DSL?
07:22.41tandrewsnah, 28.8 modems :)
07:22.49tandrewsdsl next month
07:22.57odie_floconyeah.
07:23.05odie_floconhow many modems u got?
07:23.08tandrewsbut it's expensive here compared to say the UK
07:23.26odie_floconI'm in Canada.
07:23.53tandrewsah, k
07:24.18odie_floconit kinda sux
07:24.46tandrewsjust the one modem odie_flocon
07:24.47odie_floconcuz here we have T1's, and in Europe etc. they use E1's.
07:25.09odie_floconI wish they used E1's here in Canada.
07:28.47`SauronWhee.
07:28.56`SauronI ordered a full set of embedded linux toys.
07:28.59`Sauronwww.gumstix.com
07:37.08odie_floconso Sauron, you gonna install * on these boxes?
07:37.15`SauronHehe
07:37.45`SauronDoubtful. But I do plan to play around with some audio stuff. We were talking here earlier about making an iax2 phone device or something.
07:37.50schurig`Sauron: please note that those devices have an ARM based chip without FPU, so some voice codecs won't work
07:38.01`SauronI know :)
07:38.49`Sauronas far as I saw, there's integer versions of 711, 723 and gsm
07:38.52`Sauronpossibly others
07:39.01schurig`Sauron: do they have an sound interface?  The XScale has AC97 and SPI, but without some proper chip (e.g. Wolfson Micro, UCB1x00) you won't have any audio sound at all ...
07:39.20`Sauronthe audiostix add-on board has an UCB1400 on it
07:40.26`Sauronhttp://www.gumstix.org/tikiwiki/tiki-index.php?page=Schematics
07:40.35`Sauronhttp://www.gumstix.org/tikiwiki/tiki-index.php?page=Audiostix+Schematics
07:41.22`SauronI'm mainly picking up the stuff for a different project. But I went ahead and added the audiostix add-on board so I could play around with some audio stuff.
07:41.25odie_floconwould be interesting.
07:42.01odie_floconI'm looking for a small footprint * box.
07:42.30odie_floconwas thinking about using an Xbox.
07:42.40odie_floconand calling it the "pbXbox
07:42.43odie_floconand calling it the "pbXbox"
07:42.46`Sauronhehe
07:42.54`Sauronhow about the *box?
07:42.55`Sauron;)
07:43.18kpflemingbe extremely careful what you call anything... Asterisk is a registered trademark
07:43.21*** join/#asterisk dg1_work (~schulte@gate.sympat.de)
07:43.33odie_floconso it the Xbox. :D
07:43.57*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
07:44.03`SauronMmm.
07:44.09`SauronLook, there's an IAXclient library.
07:44.22odie_flocon???
07:44.28odie_floconwhat do you mean?
07:44.46`SauronEarlier, there was talk about there not being any iax2 phones.
07:44.52`SauronJust the IAXy
07:45.36`SauronSince there's an iaxclient library, it should be (relatively) trivial to get an app compiled on the gumstix that connects to * using iax, and turns it into audio, etc.
07:46.27odie_floconhmm
07:46.32`Sauronhttp://www.voip-info.org/wiki-IAXClient
07:46.46`SauronAnd, it looks like someone's done all the hard work to verify iax runs on arm processors:
07:46.47odie_floconwhy don't you just get a hold of an existing phone. and use it.
07:46.50`Sauronhttp://www.kauss.org/Stephan/ziaxphone/
07:47.00`SauronARM softphone
07:47.15`Sauronthe only existing phones, are all SIP
07:48.08odie_floconnot true.
07:48.15`Sauronokay
07:48.28`Sauronso there's sccp, mgcp and h.323 phones
07:48.43`Sauronblah blah blah
07:49.15`SauronIt's a challenge. Something interesting to do.
07:50.53odie_floconhttp://www.iaxtalk.com/
07:50.54kpflemingall the PA1688 phones support IAX2 now as well
07:50.56odie_floconnot true
07:51.16odie_floconI'm sure you could write code for the snom phones as well.
07:51.20*** join/#asterisk eivindtr (~Eivind@193.91.146.34)
07:51.26odie_floconsince they are linux based.
07:52.16odie_floconand the A320AD is still less then 100.00
07:52.51`Saurongoogle can't find anything about a320ad
07:53.07odie_floconlook at www.iaxtalk.com
07:53.08*** part/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
07:54.57`SauronHehn. I stand corrected.
07:55.02`SauronOh well
07:55.22`SauronI'll find something to do with my audio board...
07:55.24`Sauron:)
07:57.31odie_floconalthough I can't verify this..... becuase I've never used it.
07:59.05odie_floconif I use an X-box, I can use Sip phones internally, and use IAX to go outside of the network.
07:59.53odie_floconor if I get something small enough with Dual nic's.
08:00.09odie_floconI can eliminate the router.
08:00.50odie_floconthe thing that I liked about the gumstix. is that they can be battery powered.
08:01.58odie_floconwhich would be nice for power outage.
08:02.16*** join/#asterisk zoa (~zoa@pirus.securax.be)
08:16.07*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
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08:21.23shaZwazhi room
08:23.59*** join/#asterisk qiu (~andrei@home-073519.b.astral.ro)
08:24.00*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
08:26.13*** join/#asterisk inticonnet (~nick@118.68.233.220.exetel.com.au)
08:26.20inticonnetHey Peoples
08:28.40shaZwazhi inticonnet
08:28.43*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
08:30.47inticonnetHey Manx if ur around :) Im not in the mood for asking questions so you should feel safe for a while. I do have 2morow off work tho sooo :P
08:38.52*** join/#asterisk Firebird_ (~xxx@130.40.39-62.rev.gaoland.net)
08:41.08Firebird_Hi, can someone tell me why I have poor audi quality on a wav file coming from the monitoring option ?
08:42.17JerJercuz its a german car?
08:43.22zoahaha
08:43.26zoagerman cars are the best
08:43.31zoaJJ go to bed
08:43.33*** join/#asterisk djin (~marius@62.58.40.196)
08:43.41zoaits not good for your attitude to stay up this long :p
08:44.08Firebird_sorry, I wanted to say audio quality...
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08:45.38*** join/#asterisk Shrink (~tgb@cpc1-cwma1-6-0-cust233.swan.cable.ntl.com)
08:46.49Firebird_euh...did I say something wrong or nobody has knowledge of the monitoring option ?
08:52.49zoanobody is awake
08:52.55zoai have perfect quality from the wav file
08:53.01zoaso it should work normally
08:55.22JerJerbed?
08:55.33JerJerdoes that have something to do with that sleep thing I keep hearing about?
08:56.33shaZwazwhere can I find a good comparison of Yate and * ?
08:58.10inticonnetSleep is over rated. Then again it is only 8pm here :)
08:58.41*** join/#asterisk r1 (~erwan@www.thiscow.com)
08:58.51shaZwaz~sleep
08:58.52jboti heard sleep is overrated, and a poor substitute for caffeine
08:59.14inticonnetJbot is so smart :P
08:59.22inticonnet~jbot
08:59.23jbotwell, jbot is the shipboard computer, but you may call me eddie if it helps you relax
09:02.33*** join/#asterisk meppl (~mephisto@p3E9E220E.dip.t-dialin.net)
09:04.01*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
09:04.37mepplguten morgen
09:05.50djinohne sorgen
09:09.32*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
09:11.59Firebird_zoa : when using monitoring, I have two wav files, both have the same problem...it's like the background sound is on foreground and the conversation is background.... I can't hear what it is being said...  Any idea of what I should check ?
09:13.37zoano sorry
09:13.40zoatry recording to gsm
09:13.43zoai never had that problem
09:17.57zoaoh i did around 1 million recordings so far :)
09:18.16*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
09:18.23Firebird_ok, I will have a try but it's very strange....
09:22.49shaZwazhi Zeeek
09:27.02*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
09:29.40scythelxis regexten the same as mailbox in sip.conf
09:30.29Zeeekhi shaZwaz
09:31.45*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
09:31.56neophergmorning everyone
09:33.43neopheri'm looking for a peice of hardware that will allow you to connect old tdma phones to a PBX
09:33.52neopherany idea
09:35.25*** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com)
09:38.26Zeeek.
09:55.20*** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it)
09:55.34*** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it)
10:02.35*** join/#asterisk Newbie___ (some@211.24.146.10)
10:02.59Newbie___hi, can cisco 186 work with * ?
10:03.00*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
10:06.29*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl)
10:10.22Firebird_cisco 186 works great !
10:10.40Newbie___tks, Firebird_
10:11.54*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
10:15.23Zeeek.
10:17.45JerJer,
10:17.56Newbie___-
10:18.01JerJer`
10:18.18Newbie___:D
10:19.54Newbie___can * reroute calls to my codes? ie. all UK fixed lines to broadvoice and UK cell to another provider?
10:20.13shaZwazwill there be any jitter buffer settings in upcoming chan_sip module ?
10:23.42*** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net)
10:24.09*** join/#asterisk datareactor (datareacto@203.81.192.33)
10:24.48*** join/#asterisk Delvar (~irc@83.146.53.34)
10:24.48datareactorhow can i implement fax over ip though asterisk ?
10:25.48Newbie___i dont think FoIP is any good, datareactor
10:26.04ZeeekFsck over IP ?
10:26.17JerJerfax over ip is very possible
10:26.30JerJerwe use app_tx and rx fax like they are going out of style
10:26.54JerJerit is all in how you implement the solution
10:27.39Newbie___foip is heavily dependent of isp bandwidth, am i right ?
10:27.47JerJernot necessarily
10:27.57Newbie___hrm
10:28.16JerJerdon't attempt to actually make a fax VoIP call over the 'net and you will be fine
10:28.23JerJerfind a better way
10:28.23datareactorwhat is best solution for it . we have plenty of bandwith
10:28.48Newbie___my last experience with T.38 failed, we are using fax - email instead
10:29.15JerJernotice I didn't say T.38
10:29.28Newbie___oh, ok
10:29.30Newbie___sorry
10:30.09Newbie___i was told that T.38 is out now, need to buy some gadget
10:30.16JerJerbleh
10:30.19datareactorJerJer what do you recommend for running FAXOverip
10:30.19JerJernot necessary
10:30.28JerJerapp_txfax and app_rxfax
10:30.38JerJerand don't do it over ip
10:30.57ZeeekJerJer don't you ever have problems with certain fax machines?
10:31.14JerJernot so far
10:31.30ZeeekI have and a lot of other people do
10:31.38Newbie___yeah, i also tried * fax some 8 mths ago, it only work on certain fax machine
10:31.49JerJerdon't do it over IP
10:31.51Zeeekit has been working better recently
10:32.05JerJergoing out our PRIs it has been 100% solid
10:32.06ZeeekI'm talking FXO spandsp
10:32.11JerJerhell no
10:32.13JerJernot gonna happen
10:32.22Newbie___care to elaborate more on 'dont do it over ip'
10:32.34JerJerdon't make a call over IP and expect fax to work
10:32.58ZeeekI'm talking PSTN
10:33.04datareactorJerJEr how you billing for foip services
10:33.13JerJerdatareactor:  per call
10:33.21JerJernothing different whatsoever
10:33.29Newbie___Zeeek: my experince was PSTN -> * but failed on most fax machines
10:33.34*** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au)
10:33.36JerJerZeeek:  FXO is not going to be reliable
10:33.37datareactorJerJer OK
10:33.44JerJerPRI
10:33.46JerJerdigital
10:33.49JerJerall the way
10:33.57ZeeekJerJer why not?
10:34.29Zeeekoh, ok, digital
10:35.01JerJerno DAC going on
10:35.02Zeeekanyway, fax isn't much to us, better to have a $50 fax machine than dick around for days with spandsp
10:35.04datareactorJerJer what services you acquire for international faxes
10:35.13JerJeracquire ?
10:35.26JerJerwe place a call out of our PRIs
10:35.30JerJerif a fax answers, it works
10:35.40shaZwaz:)
10:35.42datareactoruse i mean
10:36.05JerJera PRI, TE410P, a ds-3 mux and a telica switch
10:40.00ZeeekJerJer you receive faxes with spandsp?
10:40.13zoagoddamn jj
10:40.14zoago to bed
10:40.37sambal:D
10:40.56datareactorJerJer how you do  international fax ?
10:41.24JerJermake a phone call
10:41.30JerJerusing app_txfax
10:41.35JerJernext question
10:41.50datareactorJerJer but i think it will be costly
10:41.54JerJerZeeek:  app_rxfax
10:42.00zoajj, but how can your users do a fax ?
10:42.01zoa:)
10:42.02JerJerdatareactor:  how do you figure?
10:42.03Zeeekand it works?
10:42.10JerJerZeeek:  sure
10:42.23zoaanyone feels like sponsoring the sip jitter buffer ? :(
10:42.43*** join/#asterisk teemu-x (~tnurmine@tuomi.oulu.fi)
10:42.45datareactorJerJEr it will all going though Pstn and Telco will charge you international call
10:42.51shaZwazJJ Is there a way that is cheaper than that :)
10:42.52JerJerand this is a problem how?
10:43.11JerJerwe are all about the PSTN here
10:43.18zoayeah
10:43.23zoadont trust a carrier using voip
10:43.25JerJerits cheaper for us to call Russia than my parents up the road
10:43.28zoavoila
10:43.30zoasame here
10:43.30ZeeekI have had several cases where rx_fax refuses to work where windows shareware does on the same line and same fax sender
10:43.44JerJer?
10:43.49shaZwazsame here Zeeek
10:43.49JerJerdon't use analog
10:44.17JerJercoppice's DSP code is only so good
10:44.20Zeeekyou aren't addressing the problem which is why does it work on the same line, same sender with different software?
10:44.20JerJerthus far
10:44.32JerJerum better DSP code
10:44.53ZeeekI'd give him $25 for spandsp if it worked perfectly :)
10:45.05Zeeekthat's what the shareware cost
10:45.15JerJeradd a few decimal places and i'm sure he would be interested
10:45.27Zeeeknah, we don't get that many faxes
10:45.27JerJerthen run the goddamn shareware
10:45.32Zeeekwe do
10:45.35JerJernext
10:45.43zoa:)
10:45.44zoahehe
10:45.58Zeeeknext waht, I'm just sharing my experience
10:47.08shaZwazjerjer  how about a jitter buffer  in Chan_sip ?
10:47.22datareactorwhat is other choices if i dont want to use spandsp
10:47.34JerJersee bug 2532 and implement accordingly
10:47.44JerJerdatareactor: buy a fax machine
10:47.56shaZwaz:)
10:48.00JerJerand plug it in to a telephone line
10:48.17zoashaZwaz: we have a jitter buffer in chan_sip
10:48.21zoabut its not finished yet
10:48.32zoaand we are spending way too much money onit
10:48.36shaZwazJerJer sh'd I send you my faxes and you forward them :)
10:48.49JerJeri guess that means chan_h323 will need to be updated soon as well
10:49.08zoayes probably
10:49.19zoawe could also do that
10:49.28zoabut dunno if we will want to
10:49.32zoaas we no longer use it
10:49.40zoawe already spent 2 months on jitter buffer crap
10:49.48JerJermkay?  why
10:50.06zoareading books, trying different approaches etc :(
10:50.07*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-22-254.d4.club-internet.fr)
10:50.11JerJerum
10:50.23JerJerstevekstevek has a jitterbuffer and plc code that works
10:50.38zoawell the jitter buffer in chan_sip is using his now
10:50.49zoawe are also doublechecking his work
10:50.53JerJernot the newest i don't thikn
10:51.03JerJeri'm only patching chan_iax2 on the devel box
10:51.17zoayeah its too soon to do it anywhere else
10:51.22zoawe found some bugs in there so far
10:51.30zoai think he already patched them now
10:51.56zoawe first were going for a different approach, but then we dediced to team up
10:52.00zoaand started over again
10:52.23zoaanyway the quality is a huge improvement
10:52.53zoai really want it in v1.2
10:53.01zoabut dunno if we will make it
10:53.14JerJerlite a fire under some asses
10:53.26Zeeekthat wouldn't smell so good
10:53.34zoahehe
10:53.37JerJerWant me to bring in Sargent Hartman?
10:55.23shaZwazthat works with SIP ?
10:55.27shaZwaz2532
10:55.41JerJerwhen u implement it into chan_sip, it will
10:55.42JerJersure
10:57.01*** join/#asterisk dg1nsw (~schulte@gate.sympat.de)
10:57.09teemu-xHi. If user B has redirected his incoming calls from number B1 to B2, and user A calls B1, then A is charged for A -> B1 and B for B1 -> B2, right?
10:57.34shaZwazzoa: does't work with 729 ?
10:58.02JerJernot until mark/digium updates their shared object
10:58.38shaZwazand when will that happen ?
10:58.57shaZwazthis thing must be on top priority
10:59.10JerJerand the code is ready
10:59.13JerJerand?
10:59.17JerJerwhen the code is ready
11:00.00shaZwazso one can patch it safely
11:00.59JerJerthere is nothing to patch
11:01.09JerJercodec_g729a.c is not available
11:01.09zoai think mark should have it ready very soon
11:04.04JerJerReal Soon Now(tm)
11:04.13zoahehe yeah
11:04.21zoa:)
11:04.33zoait will be ready as soon as someone nagged enough
11:04.46zoaits like the overflow of nagging to his ears + 2 minutes
11:06.27PoWeRKiLLHello :)
11:07.02PoWeRKiLLI got cvs on sunday on a production server it's very very unstable
11:07.15JerJermake clean install
11:07.19zoawait till you add a jitter buffer to it :p
11:07.31JerJeri am running cvs -head as of a few hours ago and it is fine
11:07.53zoaive also seen no issues with non patched -head
11:07.58PoWeRKiLLI already make a clean install I see that some function completely change I got to update asterisk-addons also
11:08.12PoWeRKiLLJerJer on a production server ?
11:08.21JerJerabsolutely
11:09.06PoWeRKiLLHow many peer do you have on it
11:09.12JerJerall of them
11:09.21*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
11:09.21*** mode/#asterisk [+o bkw_] by ChanServ
11:09.28zoalook
11:09.30PoWeRKiLLI think since I upgrade I loose about 30% of peers
11:09.30zoaits the other guy
11:09.39JerJerthen you have bigger problems
11:10.33*** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar)
11:10.45zoanewsflash of the day
11:12.02zoasome bastard broke the heating and hot water supply for all houses in my region
11:12.22zoathey have like hot water systems in the city, people dont need to have a boiler
11:12.35zoaluckily i have a backup system for hot water
11:12.40zoawhich is also not working :((((((((
11:12.48zoabrian
11:12.50zoayou bastard
11:12.52zoareply to me
11:15.08JerJerugg
11:15.24JerJeryou let a municipality provide you HOT water?
11:15.28JerJerhow evil
11:16.05PoWeRKiLLis it possible that my problem come from asterisk-head and asterisk-stable discussing over IAX ?
11:16.52shaZwazis that due to the jitter buffer ?
11:17.09zoathats how they do it here
11:17.14zoaalso never saw it beforfe
11:17.14JerJerzoa: that's tough
11:17.16zoabefore
11:17.18zoayeah
11:17.21zoaand its freezing here
11:17.42zoaand every year they start cleaning these pipes
11:17.44zoaduring the summer
11:17.55zoano hot water for 2 months or so
11:19.02Newbie___why would u need hot water in summer?
11:19.10ZeeekJerJer out of curiosity, what countries have you visited?
11:19.33JerJerUSA
11:19.35JerJerCanada
11:19.52ZeeekMexico?
11:21.04PoWeRKiLLwhy I got lot of this error Got SIP response 415 "" back from ?
11:21.20zoabecause thats what the phone sent
11:21.27zoaor whatever device you are calling to
11:22.06PoWeRKiLLI think it's a GS device I don't understand I got this error since I got last *-head
11:22.23zoawhat does 415 mean ?
11:23.09zoaunsupported media type
11:23.12zoa= 415
11:23.33PoWeRKiLLI don't know what that mean
11:23.51zoait means get a packet dump :p
11:23.57PoWeRKiLLI'm not trying to insert a floppy disk in the GS :)
11:24.00zoaprobably the codec is incompatible
11:24.25zoaand asterisk fallsback to some codec
11:24.28zoaand gs doesnt accept it
11:24.36zoathats my guess anyway
11:30.42*** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com)
11:33.53shaZwazPoWeRKiLL: are u using the jitter buffer in SIP ?
11:36.05modulus_got some thc
11:36.08modulus_now time to code..
11:37.48modulus_i wish i had some beef jerkey
11:38.48JerJerahhh Vitamin T, H and C
11:38.51Delvari wish i could jerk off a cow
11:39.28modulus_daddy would you like some sausage?
11:42.04bkw_haha
11:42.06bkw_naughty boi
11:42.17modulus_hi bkw
11:42.24bkw_hi modulus_
11:42.29bkw_so you gonna come to cluecon?
11:42.49modulus_only if i can get stoned and drink dangerous amounts of alcohol
11:42.55bkw_beginners corse... look app_skel.c.. look make it say hello world..
11:42.59bkw_haha
11:43.08bkw_modulus_, haha you can do what you wish
11:43.08teemu-xn
11:43.14modulus_which means 'increased chance of getting laid'
11:43.22bkw_for who?
11:43.24bkw_me or you?
11:43.37modulus_me duh
11:43.43bkw_you sure about that
11:43.49modulus_everyone around me gets ass except me
11:43.57bkw_can't really get laid when you're face down in a pillow
11:44.19Zeeekwhy not?
11:44.33bkw_you have to ask?
11:44.39bkw_Have I not warped you enought?
11:44.45modulus_bkw how old are you?
11:44.49bkw_28
11:44.53modulus_cool
11:44.57modulus_let's party together
11:45.03bkw_ok now i'm scared.
11:45.06modulus_and make fools of ourselves in front of women
11:45.08bkw_:P
11:45.11modulus_hot women
11:45.16JerJerLOL
11:45.17bkw_Oh hell I don't need to be drunk to do that
11:45.23modulus_i do
11:45.26modulus_it's much more entertaining
11:45.35bkw_I'm like a broken record when drunk
11:45.37bkw_I repeat
11:45.38modulus_after the fact too
11:45.46Zeeekonly when drunk?
11:45.47modulus_i grow gimongous testicles
11:45.50modulus_when drunk
11:45.56bkw_haha
11:46.00bkw_someone has issues then
11:46.03modulus_yeah
11:46.18*** join/#asterisk jluk (~jluk@pl6.lawrence.org.uk)
11:46.20bkw_so getting drunk automagically straps on a huge pair of balls?
11:46.32modulus_for me it does
11:46.40bkw_haha
11:46.51bkw_I don't have to be drunk for that
11:46.58modulus_i'm a regular don juan under the influence
11:46.59bkw_I'll do all kinds of stupid shit without it
11:47.07bkw_say anything
11:47.09bkw_do anything
11:47.10modulus_usually the girls look better too
11:47.13bkw_make a fool of myself
11:47.17bkw_modulus_, or guys
11:47.19bkw_har har har
11:47.25modulus_i'm hetero
11:47.44Zeeeknot a crime
11:47.57Zeeekyet
11:48.25modulus_jbot cluecon?
11:49.44JerJerClueCon Talk:  How not to create a VoIP signalling method:  H.323 and SIP     :)
11:50.17*** join/#asterisk clive- (~pirch@myw-stp-66-18-86-218.sentechsa.net)
11:50.26modulus_hi jerjer
11:50.37*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:51.11clive-does anyone know what VNACK's are ?
11:51.34clive-something to do with chan iax
11:51.37*** join/#asterisk smurfix (~smurf@smurfix.developer.debian)
11:52.55modulus_jbot vnack?
11:53.19modulus_jbot hug clive-
11:53.21jbotACTION hugs clive-
11:53.25modulus_awwwww
11:55.13clive-thanks guys:), now I have a hug and thousands of VNAcks too :)
11:55.21clive-lol
11:55.27modulus_clive-, where'd you see that?
11:55.47clive-chan_iax2.c:5837 socket_read: Sending VNAK
11:56.04modulus_vnack is kinda different than vnak
11:56.19clive-my typing needs work:)
11:56.34codebreakerhello i have language=de  and exten => _cause_0,3,Playback(demo-congrats,skip)   but asterisk still plays t he sound in english. from cli  Executing Playback("SIP/7304910-69a6", "demo-congrats|skip") in new stack Playing 'demo-congrats' (language 'en')  what have i done wrong?
11:56.52JerJerclive-:  smells like your asterisk box(s) need to be upgraded
11:57.08clive-Jerjer , hi, I just upgraded
11:58.09JerJermake clean install
11:58.16JerJerrun cvs code
11:58.21JerJer-head
11:58.22bkw_also
11:58.28bkw_check your modules
11:58.31bkw_for zaptel
11:58.47bkw_their is a issue if you have say a 410 card loaded but ZERO spans configured
11:58.51bkw_asterisk can't play sound files
11:58.55bkw_have you seen that one jerjer?
11:59.02clive-Jerjer thanks, I'll try that
12:00.54*** join/#asterisk RoyK (~roy@80.239.107.80)
12:01.24RoyKwhat is "Distinctive Ringing"
12:02.10Delvara ring that is distinctive?
12:02.45Delvarisnt it a cisco thing?, the ringtone sounds sligtly different depending on what value you set?
12:03.05RoyKnfi
12:03.57JerJerbkw_:  hmm
12:04.29JerJeri'll have to fire up another devel box and try that (all of the ones i have online are being utilized)
12:05.12RoyKanyone that knows how I can simulate packet loss?
12:05.39modulus_man ping
12:05.50RoyKmodulus_: ?
12:05.55JerJerRoyK:  i think u can do it with iptables
12:06.03JerJerand the tc stuff
12:06.09modulus_royk, have someone ping flood you
12:06.20RoyKJerJer: there's a "-m random" and "-m nth" but those aren't supported in 2.6
12:06.28RoyKmodulus_: I said simulate......
12:06.31JerJerahh sucky
12:06.39tzangerJerJer: what're you building?
12:06.42RoyKalso, it's on a gigabit link to the norwegian internet hub.....
12:06.57modulus_damn
12:07.07modulus_straight into the machine's nic?
12:07.13*** join/#asterisk Tili (~Tili@202-133-65-33-dialup.sat.net.pk)
12:07.21RoyKhm
12:07.24RoyKnth is ported :)
12:08.22JerJerthere ya go
12:08.44*** join/#asterisk __pbx__ (~strace@ADSL-F49-S197-critical-coi.nortenet.pt)
12:09.44__pbx__http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold&comments_threshold=0&comments_offset=0&comments_sort_mode=commentDate_desc&comments_maxComments=10&comments_parentId=202#threadId352
12:09.48__pbx__any comments on this? :(
12:10.12__pbx__manual? :>
12:10.28tzangerJerJer: using my rc.tc script are ya :-)
12:10.35*** join/#asterisk micos (~micosat@host217-44-194-118.range217-44.btcentralplus.com)
12:11.06micosHi All
12:11.36micosJust compiled Asterix on Solaris and did make install
12:11.50micosAnd make samples
12:12.00micosis there a guide somewhere?
12:12.44JerJeroh god make samples is so evil
12:13.00tzangerJerJer: yes it is
12:13.08JerJertzanger:  nope...Royk asked about simulating packet loss
12:13.35tzangerJerJer: with the new jitter buffer there's the set losspct command
12:13.45JerJerLets play the how we spell A S T E R I S K game
12:13.57clive-tzanger not for sip, only iax
12:14.08JerJertzanger:  killer
12:14.27tzangerclive-: true enough but you can adapt that very easily
12:14.51*** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net)
12:14.52Mother_hi all
12:15.21Mother_is there a way to return the call to the original pickup SIP phone after a blind transfer ends on a busy phone?
12:15.58Mother_i.e. instead of the call going to voicemail
12:15.58JerJerdon't beat me Mother_
12:16.04Mother_lol
12:16.23RoyKhow should the new jitterbuffer be configured?
12:16.27RoyKautomagically?
12:16.47Mother_right now these people are complaining that they transfer calls but would like to talk to the caller if the extension to which it was transfered is busy
12:16.54Mother_rather than sticking them into VM
12:18.05*** join/#asterisk marcel_ (~marcel@cpc1-shep4-3-0-cust235.leic.cable.ntl.com)
12:18.06JerJerAsterisk made slashdot again
12:18.52*** join/#asterisk libpcp (libpcp@210.16.20.5)
12:21.14*** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr)
12:22.11pbxjunkieguys, does anyone know how to stop asterisk from flashing my grandstream phones whenever there is voicemail pending?
12:22.25pbxjunkieI'm pretty happy with just e-mailing the messages as attachments
12:22.37Mavviepbxjunkie: how to get asterisk down it would also be interesting :-)
12:22.54*** join/#asterisk phreak (~phreak@ua-83-227-137-86.cust.bredbandsbolaget.se)
12:22.58Mavvieor how to see it.
12:23.07*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)
12:24.29modulus_asterisk can down voicemail?
12:24.42JerJerdown?
12:24.43RoyKdown?
12:24.58modulus_i must be stoned
12:25.10modulus_i'm eating rice
12:25.11JerJerpbxjunkie:  don't specify a mailbox in the sip.conf stanza for that fone
12:25.15modulus_with this korean meat dish
12:25.28JerJermodulus_:  you mean unhatched maggots?
12:25.40*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
12:25.42modulus_jerjer, you tried those before?
12:26.11JerJerum no
12:26.11pbxjunkieJerJer: thank I got it.
12:26.21JerJerThank you, drive-thru
12:27.07JerJeromg sipx is pure evil
12:27.11JerJerjboss ?!
12:27.31JerJerWTF were they smoking when they wrote that crap
12:27.36JerJerI want some
12:27.45*** join/#asterisk Blackvel (~blackvel@dsl-213-023-033-142.arcor-ip.net)
12:28.01Blackvelhiya, anyone using latest bristuff RC7f? DISA still works?
12:28.14BlackvelAccepting voice call from '90' to 's' on channel 0/2, span 1
12:28.37Blackvelexten => s/90,1,DISA,no-password|ctx_blackvel
12:28.49Blackveldoes that still work with asterisk 1.0.5?
12:29.20Blackveli forward to the correct context, but then I get a hangup, not a dail tone
12:29.40JerJerhow about keeping it simple?
12:29.54Blackvelit is simple
12:29.54JerJerdon't try to match callerid when testing
12:30.00Blackveland worked all the time
12:30.08Blackvelbut not with new version :(
12:30.35*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-218-37.dsl.scarlet.be)
12:30.39BlackvelI used bristuff RC2 with asterisk 1.0.2 and it run really fine
12:30.49Blackvelmaybe there are some big changes in asterisk
12:31.14JerJerum yeah, it is called progress
12:31.23Blackvelhehhe
12:31.53Blackvelare there any informations on voip-info.org if there is something I would have to change?
12:32.03Blackveldoes anything sound familar to you, JerJer?
12:33.16JerJernope
12:33.25JerJerdon't run 1.0.2 or bristuff
12:33.33Gh0stywhich type of isdn cards (chipset?) which support NT mode do i need for an asterisk ?
12:34.03Gh0styi just read an article that says the hfc card driver is still beta? :s
12:35.04Blackvelhm
12:35.04Blackvelno
12:35.07Blackvelits 1.0.5 now
12:35.20Blackveland does not work anymore, but with 1.0.2 everything have been fine
12:35.41BlackvelGh0sty: how many lines? 2?
12:35.51Gh0sty?
12:35.55Mother_any good examples on forwarding a call on busy?
12:36.02Blackvelhow many NT lines do you want to connect?
12:36.12Gh0stynot sure, start with 1 :)
12:36.29Mother_hmmm
12:36.33Gh0styi start next week with an asterisk setup
12:36.44Blackvelhfc isdn card (NT mode) can work with 2 lines
12:36.57Blackvelotherwise you would have to buy a quadbri or some other cards
12:37.12Gh0styits a first test with asterisk for a company, its my school working experience ...
12:37.31Gh0stywell, all i know is: the company got 3 bri lines
12:37.33clive-ghosty I never had success wth hfc-s cards, too wacky for a production situation, so I went with eicons
12:37.42Gh0stywhich is connected to an existing pbx
12:38.35Gh0styall the phones are isdn phones
12:38.35ManxPower"the fax machine is nothing but a waffle iron with a phone attached to it." - Grandpa Simpson
12:38.35clive-oh NT, you need the quad bri then
12:38.35ManxPowerBlackvel, What is your problem?
12:38.36Gh0styand i have to fix an asterisk in trial to replace the existing pbx over time
12:39.02Gh0stybut first stage i should need to show em how all of it works on 1 line with 1 phone ...
12:39.19Gh0styand if possible link the asterix up to the existing pbx for now
12:39.50Gh0styso they can start buying voip phones and use the existing setup for a while
12:40.06Gh0styto come to full asterisk over time
12:40.30*** join/#asterisk benno2 (~benno2@host153-15.pool80182.interbusiness.it)
12:41.05Gh0styperhaps i don't even need nt-mode, if they can afford 1 voip phone to start with :)
12:41.08*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
12:41.32Gh0styif i can get the asterisk linked up to the existing pbx in a descent way too ...
12:41.54rajoGh0sty: does the existing pbx have an S0?
12:42.07Gh0stynot sure, i'll look into that next week
12:42.12Gh0stypossibly
12:42.21Gh0styits an expesive and large thing
12:42.35rajoif so, you can hook asterisk up with probably any linux-supported isdn card for first trials
12:42.37Gh0styi believe they have around 30 phones and with 3 bri lines ...
12:42.46benno2modem connection over an ATA. Impossible even with g711 ? http://forums.speedguide.net/archive/index.php/t-161277.html
12:43.17Gh0stywell, the most important stuff would be that they can make a call from the asterisk to the internal phones on the existing pbx
12:43.53rajoGh0sty: should be rather easy then :)
12:43.57Gh0stycause the asterisk would also serve for a daughter company to connect stuff trough vpn connection :)
12:44.06Gh0styto make it a simple setup ...
12:44.45Gh0sty(still wondering if i made the right choice for my school work assignement :s )
12:45.03ManxPowerCan anyone think of a reason I would get a red alarm only on TWO channels, and it only happens ocasionally.
12:47.51Mother_anyone know what happens to a blind transfered call that goes to a busy SIP extension?
12:48.03Mother_i.e. how can I get it to transfer/forward back to the caller
12:48.15Mother_there are some really vague comments on this on list archives
12:49.20Mother_someone said "I got it working" but didn't really say how
12:49.37*** join/#asterisk sysdef (~sysdef@pD9561FF1.dip.t-dialin.net)
12:49.42*** join/#asterisk soulz- (~Soulz-@cm252.sigma237.maxonline.com.sg)
12:49.48soulz-hello all
12:56.23*** join/#asterisk zotz (~zotz@24.231.32.191)
12:57.00*** join/#asterisk sysdef (~sysdef@pD9561FF1.dip.t-dialin.net)
13:02.19*** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com)
13:02.41*** join/#asterisk kamranahmad (~root@mbl-82-51-9.dsl.net.pk)
13:03.51kamranahmadhello
13:06.33kamranahmadhello i am in
13:06.41kamranahmadany developer
13:07.23*** join/#asterisk didz_ (didz_@200.218.192.52)
13:10.00ZeeekMother_ doesn't it ring back to the phone you tried to transfer from?
13:10.13Zeeekooops that was a whaile ago
13:21.49*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
13:24.55BrianR___benno2: Not at any reasonable speed.
13:27.51brc_http://slashdot.org/article.pl?sid=05/02/23/0246246&tid=215&tid=95
13:28.13*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
13:29.58*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
13:33.34benno2BrianR___: 9600baud would be ok for me
13:37.14*** join/#asterisk ^login^ (~avl@star.ukr.net)
13:39.35*** join/#asterisk montoya (~montoya@200.195.90.104)
13:40.06*** join/#asterisk clive- (~pirch@myw-stp-66-18-86-218.sentechsa.net)
13:43.02*** join/#asterisk B0ngFrOg (~wsmith@c-24-9-253-203.client.comcast.net)
13:49.43*** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca)
13:49.46*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-218-37.dsl.scarlet.be)
13:58.41*** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net)
13:59.26*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
14:00.03lyroyIS there a good guide where I can find information about Netmeeting and Asterisk for videoconferencing..
14:00.26bjohnsonno
14:00.38*** join/#asterisk znoG (gs@200.115.216.109)
14:00.45bjohnsonMS doesn't want to admit anyone but MS exists
14:00.57bjohnsonasterisk doesn't do videoconferencing
14:01.03Moclyroy, netmeeting use h323 :( that in itself is a problem
14:01.16bjohnsonI don't think it will even pass it through .. but I haven't tried
14:01.28Mocit should pass video packet between phone
14:01.37bjohnsonI thought there was a version that did SIP
14:04.42*** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net)
14:06.38pointer-gaimanyone found a clean fix for the VM "low volume when played on a pc" issue?
14:07.09pointer-gaimit seems like scaling the volume up before saving the file would make it really loud for people listening to it over the phone
14:08.03*** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar)
14:08.45*** join/#asterisk [ro]nic3try (~iancu@81.181.199.39)
14:08.50[ro]nic3tryre all
14:11.53*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
14:15.28[ro]nic3tryi can't connect mysql cdr to mysql.. i have an 30420 error
14:15.36[ro]nic3tryhelp !
14:15.54Blackvelsorry have been away
14:16.14BlackvelManxPower: I have upgraded bristuff to latest (RC7f)
14:16.20Blackvelwhich is based on asterisk 1.0.5
14:16.47BlackvelDISA accepts the call but does not give me a dail tone (which worked before wtih asterisk 1.0.2)
14:16.50Blackvelthis is the message:
14:16.51BlackvelAccepting voice call from '90' to 's' on channel 0/2, span 1
14:17.14Blackvelbut then I get a busy, instead of a dial tone (well, I use DISA for analog pbx inward dailing)
14:20.10[ro]nic3tryi have installed asterisk addons, and i'm tring to use cdr_mysql.. but, when i start asterisk cannot connect to mysql database cdr
14:21.12[ro]nic3trywhere should i look for the problem ?
14:21.23*** join/#asterisk lyroy_ (~lyroy@picachou.csaffluents.qc.ca)
14:27.00*** join/#asterisk Guyo (~chatzilla@www.paneura.com)
14:27.46Guyoi have a problem with configuring zaptel.conf for an hfc bri card and a tdm400p. I am using bristuff... is this the right channel ?
14:31.02tzangeryes it is the right channel,but I know nothing about the BRI stuff
14:31.33Blackvelwhat problem guyo?
14:32.13Guyoi'm using zaptel and zaphfc at the same time, but can't get zaptel.conf right to make them work at the same time, always getting ZT_SPANCONFIG failed on span 1: Invalid argument (22)
14:32.20*** join/#asterisk thefallen (PolarBear@thefallen.user)
14:32.22Blackvelmaybe to configure both cards? :)
14:32.31Guyospan #1 is the HFC card
14:32.43Blackvelhad you tried each card alone?
14:32.45*** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63)
14:32.48Guyoyep
14:32.50Guyoboth work
14:33.04Blackvelah, so patching libpri package with bristuff does not fail working for tdm400p then
14:33.26*** join/#asterisk cbachman (~cbachman@129.105.7.250)
14:33.32Guyo:-| I need to get it work :-|
14:33.38Blackveltoo bad, but I have not yet the experience about zaptel.conf and configuring both cards at the same time :(
14:33.59Blackvelwhat bristuff btw?
14:34.02BlackvelRC7f?
14:34.04Guyorc7f
14:34.14Guyoyes, just got it... i'm gonna try rc5
14:34.15BlackvelI even can not make it work with DISA anymore :)
14:34.32Guyo....
14:35.55Hmmhesayshmm I'm having trouble getting "read" to work from an agi
14:35.57jobihi all
14:36.48jobiI'm trying to configure an PRI/E1 between an TE410P and a E1 over IP equipment
14:36.51*** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk)
14:37.05Hmmhesaysnevermind I figured it out
14:37.21jobibut whatever I try the equipment reports a LOF alarm, and zaptel reports a yellow alarm
14:37.38Hmmhesays("EXEC READ var|soundfile") for anyone that cares
14:38.04GuyoBlackvel: Thanks, i solved that
14:38.21Guyoseems it's new of rc7f
14:38.23jobiis there a way I can ask zaptel why it raised a yellow alarm?
14:38.31Guyoi'll notify the author
14:38.42BlackvelGuyo: what is new? the problem?
14:38.51Blackvelrc5 works?
14:38.58Guyoyes, using rc5 it works, must be the rework of zaphfc code
14:39.08didz_jobi you configured the flag yellow for this span in /etc/zaptel.conf
14:39.43jobididz_: no
14:39.56jobispan=1,0,0,ccs,hdb3
14:40.54*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
14:41.32Blackvelhmmm RC7f uses new zaptel.conf and zapata.conf?
14:42.20Guyodon't think so
14:42.37*** join/#asterisk djin (~marius@62.58.40.196)
14:43.04HitTopI just got a polycom ip500~ I want to set it to register to my asterisk server. but under Sip Conf., there's no field for inputing userid and password.  Can anybody help me?
14:44.48|Vulture|HitTop: you talking about the web interface?
14:44.54|Vulture|its under Register or something
14:45.02|Vulture|far right link
14:45.26_BrianHitTop: in your sip.conf you will define the server to register to....in the phone.conf file, you define what username/password to utilize
14:49.54*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
14:50.53sivanaanyone using a channel bacnk with an MICS?
14:52.06ManxPowersivana, I think so.
14:53.47clive-does anyone know what this means : Received iseqno 10 not within window 11->11
14:58.25stevekstevekclive-: probably means you got a duplicate IAX2 full frame.
14:59.09*** join/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net)
15:00.43*** part/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net)
15:05.47*** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net)
15:06.19Frantichi all- anyone had a problem with current CVS that the sip peers list is suddenly empty?
15:09.14*** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp)
15:09.17isamarHi folks
15:09.30isamarhaving problems with oh323 and a Cisco 2600... anybody using oh323?
15:09.49isamarQ.931 error cause 24
15:10.39clive-stevek thanks...its on a very high jitter connection, I thought maybe I was like way off on a late packet
15:12.56stevekstevekclive-: no, it's really not a problem, I think..
15:13.26stevekstevekclive-: basically, IAX2 will retransmit full (reliably-sent) frames if it doesn't receive an ACK in 2*RTT.
15:14.04*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
15:14.04*** mode/#asterisk [+o anthm] by ChanServ
15:14.09stevekstevekclive-: so, in this case, the receiver actually got it, but either the ACK got lost, or the ACK took longer than expected to get back to the sender.
15:14.22stevekstevekclive-: so it retransmitted.  The receiver will just ignore the duplicate.
15:15.10bjohnsonsivana: I'm using fxo/fxs devices with a CICS
15:15.55Franticanyone had a problem with current CVS that the sip peers list is suddenly empty?
15:16.11*** join/#asterisk |Barcode (~uid@h-68-165-204-41.chcgilgm.covad.net)
15:16.34*** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com)
15:16.42sivanabjohnson: I'm working on it.  Issue with the channel bank or MICS not hanging up
15:17.09*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
15:18.22|BarcodeI have two TDM400P cards with FXO ports. I can not figure out how to determine which line a call came in on in my extensions.conf.
15:18.52bjohnsonsivana: same here.  the fxo reinitiates a call into the CICS .. I call it a ghost call
15:19.06*** join/#asterisk jsolares (~jsolares@200.30.141.85)
15:19.26bjohnsonsivana: I've been playing with the delay before the fxo gets answered on my SPA 3000 units .. hoping to fix that problem
15:19.27drumkilla|Barcode: an easy way is to put each FXO in a different context
15:19.27sivanabjohnson: my issue is that they have 19 lines.  We've moved line 19 to port 1 on the channel bank
15:19.38sivanaand forwarded that number to a temp number we assigned
15:20.26sivana<PROTECTED>
15:20.48dsmouseI saw the licensplate NXX7823 on the way to work this morning. I though it was a odd patern
15:20.49|Barcodedrumkilla: Do you mean in the zapata.conf file, or in extensions.conf?
15:21.30drumkilla|Barcode: well, in zapata.conf you say a different "context=blah" for each channel, and that corresponds to different contexts in extensions.conf
15:21.46drumkilla|Barcode: or, if you want the channel in a variable, it's available as ${CHANNEL}
15:22.03drumkillabut the context thing is probably what you want
15:23.09|BarcodeOk, let me give this a shot. Different contexts does seem what I want.
15:28.39|Vulture|whats the "show database" command?
15:28.49|Vulture|brain fart...
15:29.05bjohnsonhahahaha - Toronto1-Suncall is launching a Canada-wide service (only in selected cities)
15:29.26|Vulture|oh "database show" lol
15:30.05*** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com)
15:36.17bkw_WHATS UP PEOPLE!!!!!!!
15:36.43bkw_sivana, well stop calling yourself :P
15:36.49mikegrbbkw_: http://thegrebs.com/~michael/rasterbate.jpg <-- I will put your picture next to it, OK?
15:38.49*** join/#asterisk sabre (~urfos@69.149.209.83)
15:39.54sivanabkw_: kewlstart solved the problem
15:40.17*** join/#asterisk km- (~km-@brdgw1.rttx.com)
15:40.31km-Howdy!
15:42.40goatmilkkm-: lo.
15:42.49km-goatmilk: how's it goin?
15:42.53km-~seen bkw_
15:42.59jbotbkw_ is currently on #asterisk (4h 33m 38s).  Has said a total of 38 messages.  Is idling for 6m 16s
15:43.02goatmilkkm-: peachy.  yourself?
15:43.25km-goatmilk: it's 20 of 11:00 and I'm at work..  That means, 1:20 till lunch, so, I'm rather happy :)
15:43.31km-~seen kram
15:43.32jbotkram is currently on #asterisk
15:43.58mmlj4um, kram is mark spencer?
15:44.11km-yeah, why?
15:44.36mmlj4because I met him last weekend in mobile, and wanted to say "hi"
15:44.36goatmilkkm-: class for me in 30 min :(
15:44.49km-hehe, yeah, he's mark spencer
15:44.56km-or rather, "The Mark Spencer"
15:44.57goatmilkmmlj4: he's a nice guy, isn't he?
15:45.04mmlj4yes, very
15:45.07km-kram is awesome
15:45.08ariel_morning all
15:45.20*** join/#asterisk Cresl1n (~matt@216.207.245.23)
15:45.54goatmilkkm-: agreed :)
15:47.03km-ah shoot, he's at a conference?
15:47.11goatmilkariel_: now I am jealous
15:47.37km-maybe twisted knows
15:47.47goatmilkyeah km-
15:47.51goatmilkhe's down there for something
15:48.16km-I've got a T1 problem that I fear only Mark has the knowledge to solve
15:48.28km-because I think it's a bug in chan_zap
15:48.32goatmilkyou callin everypne stupid ;-)
15:48.36km-not at all
15:48.38ariel_It's the km- what is the problem.
15:48.41*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
15:48.48km-I'll be happy to go over what the problem is in case anyone else wants to take a shot at it
15:48.48goatmilki know i am teasin'
15:49.09twisted[work]Maybe I know what?
15:49.13km-I'm having some issues with e&m wink on a NEC PBX, talking to Asterisk.
15:49.14km-The NEC system is sending the digits to asterisk as the user dials them, and Asterisk
15:49.14km-is timing out within 1 second of silence, as opposed to the nominal 3 that you usually
15:49.14km-get with a normal zap channel.  Is there a way to up that timeout?
15:49.16goatmilktwisted[work]: ho hum
15:49.18km-twisted: ^^^^
15:49.42twisted[work]what does your dialplan look like?
15:49.52twisted[work]If you're using a catchall, the timeout will be much shorter than if you burst the digits
15:49.53km-lemme log in and do some cuttin and pastin
15:50.01dsmouseDON'T PASTE THE DIALPLAN
15:50.07km-hahahaha
15:50.08km-dude
15:50.12km-I've been using asterisk for like 2 years now
15:50.14km-I know better
15:50.16dsmouse:)
15:50.16twisted[work]dsmouse, calm down.. I'm the one that will deal with that.
15:50.18km-I know what he's asking for
15:50.41twisted[work]see that pretty @ symbol?  If someone gets out of hand, I have the authoritah
15:50.44codebreakerquestion: if i use  PSTN(ISDN)--CApicard/zaptelcard-->[asterisk]-->othercapicard/zaptelcard--->ISDN phone. can i still see the number whois calling me from PSTN?
15:50.47twisted[work];)
15:50.48km-here's the entry from zapata.conf
15:50.54km-group=2
15:50.54km-immediate=no
15:50.54km-signalling=em_w
15:50.54km-context=incomingpbx
15:50.54km-channel=>36-48
15:51.01*** join/#asterisk __Sparks_ (ringding@bb-194-6-118-37.ukonline.co.uk)
15:51.01twisted[work]mmkay
15:51.05twisted[work]i meant in the dialplan
15:51.12*** join/#asterisk thieumS (~darkmind@nanterre-7-82-229-210-142.fbx.proxad.net)
15:51.14km-just covering all the bases
15:51.16km-just a second
15:51.21twisted[work]ie, what's the first 3-4 lines of [incomingpbx]
15:51.34__Sparks_Am i correct in thinking, any US number beginning with an 8 is toll-free?
15:51.38km-[incomingpbx]
15:51.38km-exten => _9XXX,1,Congestion
15:51.38km-exten => _1XXXXXXXXXX,1,Dial(Zap/g1/${EXTEN})
15:51.38km-exten => _011XXXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN})
15:51.39twisted[work]__Sparks_, no.
15:51.43twisted[work]hmm
15:51.48km-_Sparks: 814 is an area code in PA
15:51.50twisted[work]that SHOULD be okay..
15:52.08twisted[work]you might try using immediate=yes
15:52.08km-twisted: yeah, it's really weird, if I dial the number quickly, I can get it all out before the timeout occurs
15:52.11__Sparks_okay, is there a list somwhere! - or is it just 800 and 888?
15:52.13km-twisted: but, if I dial like a user
15:52.17jsolares__Sparks_: afaik 888, 877, 866 and 800 are toll free
15:52.21km-_Sparks: 800, 888, 877, 866
15:52.27__Sparks_thanks!
15:52.36twisted[work]and using s,1,DISA(no-password|myrealcontext)
15:52.42km-twisted; if I dial 1(pause)484(pause) like most people do, I'll just get the 1
15:52.47twisted[work]right
15:52.48km-OMG
15:52.49twisted[work]see my suggestion
15:53.00ariel_km-, you have to add a longer wait on the wink side. manxpower hd something similar there is a setting for that.
15:53.01km-Why the hell didnt tzanger and I think of that last night
15:53.10twisted[work]km-, *shrug*.. this is what I get paid to think of :)
15:53.14km-ariel: you mean postwink and prewink?
15:53.36twisted[work]ariel_, that may work too, although, the winking is a line state change, irrc.
15:53.39ariel_km-, don't remember which one he did but he added =270 to it. I think.
15:53.39km-hmm, Manx was in the conversation last night, maybe he just didnt relate the problems
15:53.40twisted[work]iirc, rather.
15:53.56bjohnson__Sparks_: in NA
15:54.03*** join/#asterisk nix000 (~nixman@CPE0006256d190c-CM0011aeff5db6.cpe.net.cable.rogers.com)
15:54.04twisted[work]or, it could be that the winktime is actaully like I said - line state change
15:54.10twisted[work]dtmf != wink
15:54.14mtqhAnyone use agent groups?
15:54.25twisted[work]although It may work.. *shrug*
15:54.28km-twisted: you know what's weird, if I set the line to featured, the dialing works fine.
15:54.34twisted[work]ahhhhh
15:54.35twisted[work]well
15:54.36twisted[work]there ya go then
15:54.37km-twisted: but, asterisk complains that the line really isn't feature d
15:54.43km-twisted: and switches back to em_w
15:54.44twisted[work]e&m w/feat_d
15:54.48km-and if I set it to featd, the DID's dont work
15:54.53km-so, it's really em_w
15:54.54twisted[work]oh
15:54.55twisted[work]hmm.
15:55.01*** join/#asterisk ManxPower (~eric@24-116-82-96.cpe.cableone.net)
15:55.05km-but for some reason, switching to featd improves the problem
15:55.05goatmilkproblem solved..
15:55.08km-Manx!
15:55.10twisted[work]I know e&m wink is meant to get it's digits in a burst
15:55.12km-but causes another
15:55.21twisted[work]*CLID*NUMBER*
15:55.38km-Manx: ariel says that you might have had a problem similar to the one I was explaining last night?  Something about upping one of the wink times to improve the timeout?
15:55.47twisted[work]anywho, i'm off to my task list.
15:56.01km-twisted: thanks for the info, I think the DISA hack will work beautifully with immediate=yes
15:56.13*** join/#asterisk Derkommissar (~Loving@fl-southhub-u1-c6-0a-41.miamfl.adelphia.net)
15:56.13twisted[work]it should
15:56.25Derkommissarwhen i type show translation i get this.
15:56.26km-that may be the answer I finalize on, but I have all day before my next maintenance window
15:56.33Derkommissar<PROTECTED>
15:56.33Derkommissar<PROTECTED>
15:56.43km-so I'll keep asking in case some people know the "Right Way" to fix it
15:56.55Derkommissardoes that mean that i can translate from g729 to ulaw ?
15:57.01jsolaresyes
15:57.39mtqhI can't get agent groups to work.....It will only call the first member of the group
15:57.48Derkommissarwhat is the 3 suposed to mean ?
15:58.15km-[ Context 'preincomingpbx' created by 'pbx_config' ]
15:58.15km-<PROTECTED>
15:58.17km-word
15:58.17km-hehe
15:58.34*** join/#asterisk gonzo- (~gonzo@SIRIUS-ats227-UTC.ukrtel.net)
15:58.57jsolaresDerkommissar: the lag in ms the transcoding adds
15:59.01jsolaresi think
15:59.18*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
15:59.21Hmmhesaysgood lord I hate people that think static == public and dynamic == private in regards to ip addresses
15:59.51jsolareshehe
16:00.04*** part/#asterisk djin (~marius@62.58.40.196)
16:00.05goatmilkoff to class.  good luck km- !
16:00.06twisted[work]Derkommissar, the 3 is the amount of time it takes to translate between the two
16:00.08Damasceneidon't follow.  static is just static nat, one to one mapping.
16:00.23Damasceneso what do most people think about static nats?
16:00.23twisted[work]3ms is pretty damn good.   What kind of system is this you have?
16:00.45Damascenethat is 'has' to be public->private mappings?
16:01.03twisted[work]you're thinking of PAT, right?
16:01.16Hmmhesaysheh, most of the people i work are idiots
16:01.23Hmmhesaysand they think exactly what I said
16:01.45twisted[work]Hmmhesays, work at an ISP do ya?
16:01.49*** join/#asterisk bill522 (~bill522@182-30.201-68.swfla.res.rr.com)
16:01.55Hmmhesaysunfortunately not
16:02.08twisted[work]ah... most ISP's i've dealt with think that way, believe it or not
16:02.31twisted[work]in fact, my ISP told me I had a "randomly changing static address" once.
16:02.35DamasceneOH, they think static nat == PAT (many to one?)
16:02.57Hmmhesaysno... they think static addresses are always public
16:03.03km-goat: thanks man!
16:03.04Hmmhesaysand dynamic addresses are always private
16:03.06twisted[work]Damascene, no, i was making a joke :P
16:03.07DamasceneHmmhesays:  OH MY
16:03.20twisted[work]unfortunately humor is lost on irc
16:03.29Hmmhesaysthen I point them to a guide of ip addressing... and it melts their brain
16:03.38twisted[work]anywho
16:03.42Hmmhesayslol
16:03.58DamasceneHmmhesays:  melt them more by asking them to subnet anythin beyond a /24 or... as they say two-five-five dot two-five-five dot two-five-five dot zero
16:04.21Hmmhesaysheh... i'd rather reach through the screen and strangle them
16:04.35twisted[work]nah, be nice to them, they control the intarweb
16:04.43twisted[work]:P
16:04.46Hmmhesaysthe intardweb?
16:04.52twisted[work]that too
16:04.55Hmmhesayslol
16:05.09Hmmhesaysdid you renew your tfark account this month?
16:05.14twisted[work]yep
16:05.17twisted[work]it's on automatic renewal now
16:05.22twisted[work];)
16:05.29Hmmhesayshaha, it keeps me from going insane at work
16:05.34twisted[work]same here
16:06.17twisted[work]speaking of which
16:15.43*** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp)
16:15.46isamarhi folks
16:15.59isamarhaving problems to compile latest chan_oh323
16:16.01isamarchan_oh323.c:5192: warning: passing arg 4 of `ast_channel_register' from incompa
16:16.02isamartible pointer type
16:16.57*** join/#asterisk gr0mit (~gr0mit@router1.txrx.org.uk)
16:24.29bill522can anyone tell me how I can call into asterisk by outside PSTN and transfer to SIP?
16:24.44*** join/#asterisk mutilator (~animenodv@65.111.201.79)
16:24.47mutilatorhey all
16:25.03Beirdoheh
16:25.29mutilatorhaving a problem, i assume related to bandwidth or maybe latency? when this person calls, you can hear him great, but after ~3 minutes of talking all he hears is breaking up and sounds like crap
16:25.39mutilatorand i can still hear him perfectly fine
16:26.07gr0mitbill522, we need a bit more info on your configuration before we can answer your question
16:26.16mutilatorgoing from ata -> g729 -> asterisk -> ulaw -> as5350 -> pstn
16:27.06bill522gr0mit, okay, I just have a X100P with my analog line, I would like to dial in on it, and transfer to my FWD or something like that
16:27.12mutilatorwas limited to 384kbit, where the 729 codec was used, i just bumped to 512kbit to see if it solved problem
16:27.39gr0mitk.  do u have a SIP account already with fwd?
16:27.46km-hrm
16:28.00bill522yes gr0mit, but I would like to choose, I have many SIP accounts
16:28.03km-I don't think I can transfer calls from the nec pbx to the asterisk system
16:28.42gr0mitcos then you can take your inbound call from your X100p and dump it into a context which has a Dial(mysipprovider) line
16:28.54gr0mitwith whatever dial command options you want
16:29.24bill522yes, preferably my voip phone dialplan
16:29.27*** join/#asterisk Slainte (~Slainte@66.55.112.13.ppp.northrock.bm)
16:29.38gr0mityou can send the call to multiple SIP channesl, plus your voip phone.
16:29.42*** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk)
16:29.53kamranhello
16:30.02bill522yes gr0mit, but need to autheenticate first, not just transfer every1
16:30.25SlainteAnyone know how to setup the dialplan on a polycom IP600.  W=users need to be able to dial the missed call list, but cant because there is no 9 in front of the number.
16:30.57*** join/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com)
16:31.06gr0mitplease describe exactly what you are trying to do, bill522
16:31.14MicH323Hi All
16:32.10bill522okay, call into * with pstn, maybe press 5, get prompt for pin, then dialtone to use current sip dialplan like dial 8{FWD}, 7{IAXTEL}, etc
16:32.42mutilatormeh
16:32.54mutilatorsearching cisco's website sucks
16:32.56MicH323Just setup Asterisk on Solaris. Everything running. Configured to FWD via IAX, Now trying to attach an Cisco ATA to it, its failing with registration failed
16:32.57__Sparks_I am after a little help with my sip.conf file! - this line in sip.cong "exten => _8.,1,Dial(SIP/${EXTEN:1}@sipgate1)" seems to make calls prefixed with a 8 go to sipgate- stipiing off the 8 - what bit of this line stripps off the 8?
16:33.02mutilatorcan't find anythin
16:33.23km-__Sparks: the $EXTEN:1 part strips off the 8
16:33.27bill522the EXTEN:1 _Sparks_
16:33.28km-__Sparks: chnage it to ${EXTEN}
16:33.45kamranany one know how to call Dial or HangUp application from one application (maybe changing priority)
16:33.51km-I need to get me a 7960 again
16:33.51km-hehe
16:34.00__Sparks_km- thanks!
16:34.06km-no problem
16:34.08bill522gr0mit?
16:34.29bjohnsonbill522: look at the user authentication wiki page
16:34.42bjohnsonlinked to from the tips and tricks page
16:34.43MicH323Simle question. Where do you configure users?
16:35.04bjohnsonI think it has examples for what you want to do
16:35.10bjohnsonMicH323: voicemail.conf
16:35.15bill522ty bjohnson, will that help with what I am trying to do?
16:35.35MicH323will voicemail.conf have ATA registration passwords? ? ?
16:35.41bjohnsonMicH323: it's about the only place where the concept of 'users' is used
16:35.50bjohnsonMicH323: you want the same page that bill wants
16:35.54gr0mitok, i see what you are trying to do...not sure i undersatnd why.
16:36.04bjohnsonbill522: I think so, dial in, authenticate, dial out
16:36.18gr0mitif you tell me why you are trying to do it there may be a more elegant sln we can suggest
16:36.23kamranhi developres any one know how to call Dial or HangUp application from one application (maybe changing priority)
16:36.43MicH323So voicemail.conf has phonenumbers mapped to password?
16:37.23bjohnsonMicH323: no
16:38.04bjohnsongr0mit: I use it to allow cell phone users to dial in, authenticate, and dial out = 3c/minute for LD instead of 25c/min
16:38.12bjohnson(using a toll free number)
16:38.32MicH323ok, which file does the Authenticate commands go into?
16:38.42bjohnsonMicH323: extensions.conf
16:39.20bjohnsonMicH323: So voicemail.conf has phonenumbers mapped to password? .. nothing is that direct .. you have to config you dialplan to do something like that
16:39.46MicH323lolz! Thans bjohnson
16:40.40bjohnsongr0mit: I use the same concept for all incoming calls to authenticate who gets direct access to dial internal extensions
16:40.42BuckRogershello hello
16:41.02MicH323So something like: USERA=SIP/6601 and then exten => 6601,1,Dial(SIP/6601)
16:41.14bjohnsonMicH323: no
16:41.19MicH323ops!
16:41.22greg_workcan someone read this: http://sourceforge.net/mailarchive/forum.php?thread_id=6634469&forum_id=42627  and suggest the best way to implement it? (really, just the part at the bottom about replacing Dial() ) .. can i do a macro, or would a goto work?
16:41.28bill522ty gr0mit lol searching for info now
16:41.31BuckRogerswe also do a simialar dial in dail out method]
16:42.05bjohnsongreg_work: you want the superdial macro from the wiki I think
16:42.12greg_workno
16:42.34greg_workbasically i'm trying to "fix" a dialstring
16:42.38tzangerugh
16:42.49tzangerI fucking hate automated dialers with AUTOMATED FUCKING MESSAGES!!!
16:43.09greg_workif i dial 5551234, i want it to dial "5551234" on my ZAP trunk, but "16135551234" on my voip trunks
16:43.20bjohnsongreg_work: yes
16:43.26Nuggetgreg_work: that's easy enough.
16:43.36greg_worksimilarly, if i dial 16135551234, i want it to dial 5551234 on my ZAP trunk, but 16135551234 on my voip trunk
16:43.36bjohnsonuse the superdial macro to try different channels in sequence
16:43.50bjohnsonthen use 1613${EXETN} for the voip one
16:43.53greg_worki already have done the sequence thing, and its more powerful and easier to use than superdial
16:44.00bjohnsoninstead of just ${EXTEN}
16:44.12greg_workand i'm writing this into AMP so it's actually easy to manage
16:44.14Derkommissarcan someone tell me whats wrong with this invit
16:44.17Derkommissarinvite
16:44.18Derkommissarhttp://www.pastebin.com/245447
16:44.32Derkommissarits not sending the audio request
16:44.42Derkommissarin sip.conf i configured it for ulaw
16:44.50bjohnsongreg_work: append the 1 and area code to the local number depending on what channel it goes out on
16:45.02greg_workthat's what i'm trying to do ;)
16:45.13bjohnsongreg_work: don't change the pattern match
16:45.23bjohnsongreg_work: I'm trying to tell you how I have it working
16:45.28bjohnsonWORKING
16:45.50greg_worki know its easy to do by just hardcoding into the dialplan
16:46.29greg_worki'm trying to write it so you can put "1613+NXXXXX" in a web-based interface, then it makes  exten=>_NXXXXXX,1,Dial(1613${EXTEN})
16:46.54greg_workor 1613|NXXXXX  makes exten=>_1613NXXXXXX,1,Dial({$EXTEN:4})
16:47.04bjohnsonok..then have the user enter the 1613 somewhere are their local area code
16:47.11bjohnsoncheck for it, and remove it
16:47.19bjohnson(for local pstn calls)
16:47.19greg_workok but here's the complicated part
16:47.26Juggiebj, why not just check and see if the call is local
16:47.38bjohnsonJuggie: how?
16:47.40greg_workmy area code is 613. i can dial 613-544-xxxx locally.. but 613-789-xxxx is long-distance
16:47.51Beirdogreg_work, and what about 819, any of it local?
16:47.54greg_workno
16:48.01Beirdoahh, so not Ottawa.
16:48.06Beirdothat makes life simpler
16:48.06bjohnsongreg_work: yes .. easier to go other way and append rather than remove
16:48.08gr0mitbill522, why don't you use caller id on the inbound call to send you into a special context
16:48.16greg_work1613+789XXXX
16:48.17gr0mitand then use app_DISA
16:48.18Derkommissarcan someone take a look at this?
16:48.22greg_work1613|544XXXX
16:48.36Juggiegreg, i had to solve this issue just the other day
16:48.51codebreakerhow can i check if asterisk has sucesfully registered at iaxtel.com?
16:48.56Juggiei am in ottawa however so we have local dialing across two area codes
16:49.04Juggieand u dial them like they are in the same exchange.
16:49.06bjohnsongreg_work: I do a pattern match on the 7 digit number and append the 1613 to outgoing if through a voip channel
16:49.19bjohnsonI do not append it if going out pstn directly
16:49.22gr0mitexten => s/yourcellphonecallerid,1,Goto(cellphone,s,1)
16:49.24greg_workyeah thats fine
16:49.31greg_workhere, hold on, i'll show  you my implementation of superdial
16:49.45greg_workand EXACTLY what i'm trying to do here
16:49.53Juggiegreg_work, use http://members.dandy.net/~czg/search.html to implement a database of local areacodes/exchanges
16:50.14Juggiethen use some agi be it perl/php etc. to do a lookup and see if its a local call or not...
16:50.19greg_workhttp://pastebin.ca/6340
16:51.06*** part/#asterisk sysdef (~sysdef@pD9561FF1.dip.t-dialin.net)
16:51.28*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
16:51.45bkw_Juggie, so does telcodata.us
16:52.37Juggiebkw, thats all i did, we needed 10 digit dialing for a conference app.... so that was all i did to implement that.... just lookup see if its local or no and then handle it in extensions.conf
16:52.38bjohnsongreg_work: looks like it does the same thing but hides all the macro args in external arrays .. I guess it depends what you mean by easier to use when comparing to superdial .. but you do whatever works for you
16:52.41Juggieworks perfect.
16:52.50codebreakerhow can i call from PSTN in Germany to an iaxtelaccount/number
16:53.03greg_workbjohnson: well, thats more suited to my use where it's built automatically from the web interface
16:53.15bjohnsoncodebreaker: make a gateway or check iaxtel or fwd to see if they have one
16:53.25bjohnsoncodebreaker: might be other ways in as well
16:53.31greg_workJuggie: thanks... how do i use this thing though? :p
16:53.55greg_workoh nm, ok .. very non intuitive interface ;P
16:54.29codebreakerbjohnson: the talk on the website about voicepulse. i dondt know what this is. the numbers there i ve tried but fails for me
16:54.50bjohnsonvoicepulse is a voip provider
16:54.56km-anyone here have an international dialing extension that they'd be willing to share with me?
16:54.59Juggiegreg_work, are you in ottawa?
16:55.05greg_workJuggie: kingston
16:55.15codebreakerah. ok. thanks
16:55.30greg_workhm.. this is very interesting. i wonder if i could build this into the interface so it could automatically populate....
16:55.34bjohnsongreg_work: you don't like the idea of allowing the dialing user to dial the 7 digit number?
16:55.42Derkommissarwhat does m=audio 10840 RTP/AVP
16:55.48*** join/#asterisk devel (~devel@wiggum.digitalcoven.com)
16:55.56Juggiegreg, all you want to do is decide if a num is local or not, if it is dial on a zap chan, and if its not, dial on iax/sip or whatever?
16:56.07greg_workbjohnson: hm? yes, thats what i'm trying to allow. if you dial 7 digits now, and it tries to dial out on voip, it will fail
16:56.08bjohnsonwhy decide?
16:56.42bjohnsongreg_work: so append the 1613 to ${EXTEN} when going out through voip?
16:56.54greg_workJuggie: no.. i have a routing screen, where you can set patterns and then a sequence of trunks to try
16:57.05bjohnsonadd the option to append the 1613
16:57.10Juggiegreg, thats a little holey :)
16:57.16Juggiewhat if someone dials 16135551212
16:57.20Juggiebut the call is local
16:57.29bjohnsonwhat does Bell do?
16:57.30*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
16:57.30greg_workthats the exact problem i'm trying to fix ;)
16:57.34Juggieits gonna go through your ld when it didnt have to
16:57.34greg_workbjohnson: complain ;)
16:57.44bjohnsonif it goes out pstn and it's local, Bell tells you it's a local call
16:57.56Juggiegreg, all you have to do is this.
16:57.58bjohnsonif it goes out voip and it's a local call, it works
16:58.00nestArmatch your local AREA codes
16:58.15nestArexten => _613NXXNXXX
16:58.36Juggiegreg, take this http://members.dandy.net/~czg/lprefix.php?exch=155000&dir=1 make a database out of it.
16:58.48greg_workok the routing is like (we'll do this for the general case of all of 613 being local) ..  NXXXXX is the pattern, trunk priority is:  ZAP/g0, IAX2/voip      second route,  1NXXNXXXXXX, trunks: IAX2/voip, ZAP/g0
16:58.59Juggiethen write an agi script, to look up the number dialed against that database, if it returns a record the number is local
16:59.05bjohnsonall of this to avoid a dialing user getting Bell telling them that it's really a local call?
16:59.06bjohnsonwow
16:59.07Juggieif it doesnt, then its long distance.
16:59.15greg_worksigh
16:59.22greg_worki'm making a general-case web-interface here
16:59.28Juggiebj, its called proper routing, maby they dont know the call is local
16:59.35Juggiewhy waste money on LD if they dont have to.
16:59.35greg_workif it was just hardcoded, it wuold be very simple
16:59.45bjohnsonJuggie: ? waste money on LD?
16:59.59bjohnsonthe call won't be placed if it's a local call!!
17:00.02Juggiebj, if someone dials 16135551212 and the number is local
17:00.08Juggieit will go through iax/sip when it shoudnt have
17:00.09bjohnsonBell gives a message
17:00.10Juggiesmarten up.
17:00.42greg_workJuggie: that would be fixed by making a route like this: patterns: 1613NXXXXXX, NXXXXXX  and then trunks: ZAP/g0, IAX2/voip
17:00.57greg_work16135551234 will match the pattern, and try to dial on ZAP/g0 first
17:01.14greg_workhowever.. before ZAP/g0 can dial that number, it needs to drop the 1613
17:01.20greg_workthat's the part i'm working on now
17:01.20Juggiegreg_work, thats fine for pri, but not analog as they will hear the bell message.
17:01.26greg_workyes
17:01.37greg_workif you had a PRI trunk, you wouldn't need to put any patterns in for the "fixing" part
17:01.49greg_workit would just dial as-is
17:02.19codebreakergreg_work: what du you mean with drop? cut off and dial the number after the 1613
17:02.49greg_workREALLY, myquestion is how can i do pattern matching from within that macro i posted (http://pastebin.ca/6340) .. i want to replace exten => s,8 with something that can fix the number
17:03.06greg_workcodebreaker: yes
17:03.28codebreakergreg_work: do ${EXTEN:4}  the 4 will cut off the 4 digits and dial the rest
17:04.49codebreakergreg_work: at home i do  exten => _99.,1,Dial(CAPI/9420576:${EXTEN:2},30,r)  so i call 99andtherestofthenumber to get calls routet via PSTN and dial everything after the 99
17:05.06greg_workcodebreaker: yes. i actually have code that lets you enter "1613|NXXXXXX" that creates exten=>_1613NXXXXX,1,Dial(...${EXTEN:4})
17:05.16JohnnyCAnyone whats a good board to buy to use with Asterisk, I just need a conection to a S0 line with 10 numbers
17:05.23bjohnsongreg_work: http://www.voip-info.org/wiki-Asterisk+E164+Call+Routing ??
17:05.50codebreakerJohnnyC: multiplex or only 2 channels?
17:06.00JohnnyCcodebreaker: whats the diference ?
17:06.16greg_workbjohnson: hm, maybe useful but i'm not sure thats simplifying things ;)
17:06.21greg_workoh crap
17:06.25greg_workmaybe i can just use chan_local
17:06.46JohnnyCits a ISDN line , now I have 3 numbers but I'll have 10 very soon
17:06.55codebreakerJohnnyC: multiplex is 10 lines can be used simultanous and the other is only 2 users can dial simoultanous
17:07.37gr0mitjohhnyc - best cheap card is an hfc chipset card
17:07.53gr0mitMade by Billion or Asustek
17:07.55codebreakerfull ack.
17:08.02gr0mitcosts eur 15-20
17:08.13gr0miti have 3 of them in my home * box
17:08.48JohnnyCwell can you advise me in these two cases
17:09.04JohnnyCbecause now only 2 users can call simultaneous using that ISDN line
17:09.12gr0mityup
17:09.14Derkommissarwhy when i allow = ulaw it doesnt send a=rtpmap:0 PCMU/8000
17:09.17JohnnyCbut with the 10 lines it should be diferent
17:09.28codebreakerJohnnyC: or buy Acer ISDN 128 Surf PCI (in germany so called)
17:09.30Derkommissarbut when i allow all it sends all and one of them is a=rtpmap:0 PCMU/8000
17:09.41gr0mitno.  if you get 10 numbers on a BRI you will stil only get 2 voice channels
17:10.04Derkommissaris the setting allow=PCMU
17:10.19codebreakerJohnnyC: then a "normal" isdn card is enough
17:10.25*** join/#asterisk devel (~devel@wiggum.digitalcoven.com)
17:10.48JohnnyChmm ok and in the multiplex case where I can use the 10 lines simultaneous ?
17:11.03gr0miton a single BRI? not possible.
17:11.21codebreakerJohnnyC: if you order a multiplex anschluss from your pstn provider :)
17:11.30gr0mityou would need 5 x bri to make 10 simul calls
17:11.54gr0mitconfigured in ptp mode
17:12.02gr0mitas a single group
17:12.06codebreakerand the right hardware. if you have euroISDN
17:12.07JohnnyCso I can use the 10 lines simultaneous if I ask the provider ? I dont need another card ?
17:12.15*** join/#asterisk christo (~chris@office.enovi.com)
17:12.34*** join/#asterisk marc_c (~marc32344@69-28-224-214.dsl.teksavvy.com)
17:12.40gr0mitif u want 10 calls you need 5 x BRI or a single PRI
17:12.52JohnnyCwhats 5 x BRI ?
17:12.57gr0mitdon't confuse 'numbers' with 'channels'
17:13.06JohnnyCok
17:13.16gr0mit5x bri is 5 seperarate Basic Rate ISDN lines.
17:13.30JohnnyChmm ok
17:13.40marc_cpartial T1?
17:13.44codebreakerJohnnyC: for 10 lines simultaneous you need 1. a multiplexline from your PSTN provider 2. 5 BRIcards o a quadbricard and a singlebricard
17:14.26JohnnyChmm so to have 10 lines I need at least 5 cards ( ACER ISDN ......)
17:14.28JohnnyC?
17:14.42codebreakerJohnnyC: not so expensive. i thnk a quadbri now costs about 600¤
17:14.55gr0mitbut in some countries, e.g. for 8 or more chans you can use a sub-provisioned PRI
17:14.59JohnnyCso each 2 lines / BRI Card only ?
17:15.18codebreakerJohnnyC: you have a mainboard with >5 pcislots?
17:15.23gr0mitone BRI from your telco will give you 2 voice channels
17:15.40JohnnyCno I dont
17:15.47gr0mittwo bri will give you 4 channels
17:16.10*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
17:16.14gr0mitin UK, if you want more than 8 voice channels an E1 PRI is cheaper
17:16.37codebreakerin germany i think too
17:16.57gr0mityou can get a subequpped E1
17:17.32gr0mitwhich country are you in, JohnnyC ?
17:17.37outtoluncsounds like something that has bubbles and a bath tub <G>
17:17.45coppicehere its cheaper to get 24 analogue pairs than a T1 :-\
17:18.10JohnnyCPortugal ?
17:18.21gr0mitare you not certain?!
17:19.53gr0mitcu codebreaker
17:20.31*** part/#asterisk Moc____ (~mochouina@64.235.210.66)
17:20.32Blackvelwho is using DISA, bristuff RC7f and asterisk 1.0.5? for some reason it doesn't work anymore
17:21.38gr0mitjohhnyC what is your application? do you really need 10 simultaneuos calls?
17:21.46*** join/#asterisk Moc____ (~mochouina@64.235.210.66)
17:21.48JohnnyCno
17:21.50JohnnyCI dont
17:21.52JohnnyC:)
17:21.59JohnnyCmaybe I just need 2
17:22.05JohnnyCso a BRI Card is enought
17:22.09gr0mityup.
17:22.10Moc____cluecon /
17:22.26gr0mitGet a zaphfc card for 15 or 20 euros
17:22.32thieumSis it possible to use libpri and zaptel CSV with * 1.0.5 ?
17:22.35JohnnyCbut when I ask more BRI to a telco , they give me the same or another cable ?
17:22.41thieumSCVS sorry
17:22.54gr0mitthey will run another pair into your house.
17:23.01gr0mit(if you are lucky)
17:23.16gr0mitif they have no spare copper you are unlucky.
17:23.16JohnnyChmm oki
17:23.29*** join/#asterisk adjacent (~scott@office.bftwave.com)
17:23.42gr0mitin our office in Lisbon Portugal Telecom took AGES
17:23.53JohnnyCAGES ?
17:23.57gr0mit2 months
17:24.13JohnnyCages = time
17:24.22gr0mityes. ages = a very long time
17:24.35JohnnyChehe I tought AGES was some acronim !
17:24.39gr0mithehe!
17:24.43JohnnyChehe Im really traumatized
17:24.54JohnnyCBut I was able to put Asterisk working with IP phones
17:24.58JohnnyCyesterday night
17:25.07gr0mitso, get 2 zaphfc cards
17:25.25JohnnyCso this means that if you have 100 lines you have 50 cables ?
17:25.37gr0mitconfigure one in NT mode, the other in TE mode, and place your Asteroisk box between them
17:25.38JohnnyCcan they pass signal into the same cable ?
17:25.39gr0mityup.
17:25.42gr0mitno.
17:25.48gr0mitit is called PRI
17:26.07JohnnyCPRI ?
17:26.09gr0mityou get 30 voice channels on an E1 PRI EuroISDN.
17:26.15gr0mitPrimary Rate ISDN
17:26.22JohnnyChmm ok a PRI
17:26.32JohnnyCand then I supose you have a PRI card also
17:26.37gr0mityup.
17:26.49JohnnyCthat suports 30 voice channels
17:26.57gr0mitDigium make a few for about 1000 dollars
17:27.20tzanger??
17:27.24tzangerTE110P is $499
17:27.55gr0mitin EU after import and tax, about 1000 dollars
17:28.06outtoluncouch
17:28.58*** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net)
17:29.08thieumSi just bough TE405P for 1100 euros in France
17:29.09__Sparks_in this line I have in sip.conf, "exten => _001800.,1,Dial(SIP/*${EXTEN:2}@fwd-outgoing)"  what is the first _ for?
17:29.09gr0mitGBP359 + VAT.
17:29.55gr0mit'about' means give or take 20-50% !!!!
17:30.05*** join/#asterisk zeek (~zeekk@gw.dhivehinet.net.mv)
17:30.07gr0mitand the dollar is getting weaker by the hour....
17:30.21SuPrSluG_Sparks_:pattern match
17:30.29gr0mitbbl.
17:30.50__Sparks_thanks! - is there a document somewhere that explains it all....well!?
17:30.54coppiceactually the dollar strengthened a little recently
17:31.39SuPrSluG_Sparks_:voip-info.org  look up extensions and dial plans
17:31.42zeekI have a TDM40B with rj45 sockets. Can anyone tell the the pinouts?
17:32.47bjohnsongreg_work: put it on the wiki if you get it to work
17:32.56*** join/#asterisk zno (~zeno@ip-160-79-174-99.autorev.intellispace.net)
17:34.44zeekI have a TDM40B with rj45 sockets. Can anyone tell the the pinouts?
17:35.00SuPrSluGi just got dundi working thru openvpn. anyone piping sip thru a vpn?
17:35.26Blackvelhm
17:35.38znois it possible to do use exten => XXX,hint for queues?
17:36.04Blackvelexternip=xxx.dyndns.org still works with asterisk 1.0.5?
17:36.55*** part/#asterisk Moc____ (~mochouina@64.235.210.66)
17:38.58*** join/#asterisk ^HeLL^ (~admin@217.11.115.168)
17:39.06__Sparks_Another question! - this line from sip.conf routes all calls to the POTS line when prefixed with a 9, removing the 9. - if I wanted it to prefix the number with somthing, then pause, then dial the number origanally dialed, how do T do it? - Do i use commas for pauses?
17:39.13^HeLL^hello all...
17:39.16__Sparks_"exten => _9.,1,Macro(dialout,${TRUNK},${EXTEN:1})"
17:40.48*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
17:41.33Blackvelis it true you can set with asterisk 1.0.5 only externip = xxx.xxx.xxx.xx instead of xxx.dyndns.org? externhost has just been added to CVS head? what should/can I do now? I need 1.0.5 but with dyndns addy
17:41.38__Sparks_Blackvel - it does for me! (Not DynDNS, but another DNS Name in there)
17:41.42znocan you define an agent without a password?
17:42.10Blackvel__Sparks_: ah great
17:42.14Blackveldo you use DISA too?
17:42.32__Sparks_DISA?
17:42.37__Sparks_(I guess not then!)
17:42.38Blackvelyeah
17:42.41Blackvelm
17:42.42Blackvelhm
17:42.44bjohnson__Sparks_: that line isn't from sip.conf
17:43.00bjohnsonBlackvel: have you looked at the user authentication page in the wiki?
17:43.43__Sparks_bjohnson, You are quite correct. I mean extensions.conf
17:44.14bjohnson__Sparks_: read the extensions.conf wiki page .. and the info about the dial command .. they will help you understand it (and yes commas are usually pauses)
17:44.33__Sparks_bjohnson, Okay, I will sit and read!
17:44.35*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
17:44.49Blackvelbjohnson: not for 1.0.5, I upgraded and DISA does not work anymore. what should I look for?
17:45.56__Sparks_is this error bad? -  chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
17:46.55SuPrSluG__Sparks_:yes. if your using a cisco phone and using skinny protcol
17:47.13__Sparks_okay, then I will ignore it!
17:48.12__Sparks_If I have more than one Sip registration in sip.conf, i seem to be getting errors - do I need to define different ports for different SIP profiders to stop this?
17:49.55__Sparks_For example -  NOTICE[1833]: chan_sip.c:6801 handle_response: Failed to authenticate on REGISTER to '<sip:265532@fwd.pulver.com>
17:50.04zeekI have a TDM40B with rj45 sockets. Can anyone tell the the pinouts?
17:50.42roamer323__Sparks_  I have 5 registrations , no problem
17:50.47Delvarzeek: what you pluging it into?
17:51.15roamer323__Sparks_ check for tyeps
17:51.23roamer323typos :-D
17:51.27zeekDelvar: telephone line
17:51.33Delvarzeek: if its into the wall socket then its a standard stright cable
17:51.36*** join/#asterisk rontecxt44 (~rontecxt4@dsl9-173.rb.comporium.net)
17:52.13zeekDelvar: its a rj11 thats going to the socket
17:52.18__Sparks_roamer323 - the account info is correct, as if I only have one in there , I dont get errors (And I dont always get the errors on the same accounts!
17:52.55zeekDelvar: TDM04B bord has RJ45 sockets
17:53.04Delvarzeek: ah.. ill see if i can find one of my links with the exatct pinouts
17:53.14tzangerDelvar: no
17:53.19*** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com)
17:53.20tzangeryou just plug RJ11 into the RJ45
17:53.30tzangerit fits, I think the RJ specification demands that actually
17:53.39Delvarcool
17:53.44Delvarnever tried that
17:53.59MicH323Help Please... getting Feb 23 17:36:07 NOTICE[7062]: chan_sip.c:8404 handle_request: Registration from '<sip:6601@asterisk.itsp.net;user=phone>' failed for 'xxx.xx.xxx.xxx'
17:54.11Delvari usualy use an adapter, rj45 one end rj11 the other
17:54.13greg_workmy office is wired only with rj45 and cat5e .. but we still use regular phones on them
17:54.16MicH323Trying to setup user on ATA
17:54.22tzangerMicH323: well there's your problem right there...  xxx.xx.xxx.xxx isn't a valid domain name.  :-)
17:54.44roamer323__Spraks_ : the only registrar I get problem with is iaxtel, but that server is hopelessly overloaded  ; others may choke once a while, but never consistent
17:54.46MicH323Thats the IP of my ATA
17:54.58*** join/#asterisk Gerrath (Gerrath@shanev.lifecor.com)
17:55.07zeekDelvar: I just want to find out the 2 pins that is used in the TDM04B board
17:55.12__Sparks_roamer323 - I am behind a router - would that possibly be the problem?
17:55.33tzangerzeek: the middle two
17:55.34tzangerpair 1
17:55.40__Sparks_roamer323 - I tried putting my asterisk box inot the DMZ zone, but I still get the errors
17:55.43Delvarits the middle ones
17:56.24roamer323__Sparks_ is it consistent with a fixed sip.conf ?  i.e. the same provider will give the same error given a fixed sip.conf?
17:56.41zeekDelvar: you sure?
17:56.47Delvarzeek: 100%
17:56.59zeekDelvar: I trust you
17:57.01Delvari just dont know the exact pinouts
17:57.16Delvarjust remember to expect a loud band when you plug it in
17:57.21Delvarbang*
17:57.51__Sparks_roamer323 - The error is always the same, but it isn't always the same SIP account that throws the errors
17:58.09__Sparks_roamer323 - I am also seeing this - WARNING[1833]: chan_sip.c:6786 handle_response: Got 200 OK on REGISTER that isn't a register
17:58.42*** part/#asterisk Cresl1n (~matt@216.207.245.23)
17:59.29rontecxt44hi...has anyone dealt with error "Avalible sdfsdf
17:59.29rontecxt44The Spirit of Tulsa
17:59.29rontecxt44Learn about Avenue One
17:59.32rontecxt44Leasing News
17:59.36rontecxt44Request Info
17:59.48rontecxt44hi...has anyone dealt with kernel error  "TDM PCI Master abort"
17:59.50rontecxt44sorry
17:59.52rontecxt44bad past
17:59.54*** join/#asterisk okieplaya (~okieplaya@ip68-229-252-53.ok.ok.cox.net)
17:59.54rontecxt44:(
18:00.01rontecxt44paste
18:00.10*** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk)
18:00.29roamer323__Sparks_ that's a strange one... you're not running any other SIP thing on the same box, are you (i.e. a softphone) ?
18:00.46*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
18:01.38__Sparks_roamer323 - nope, I have 1 asterisk box, just doing asterisk, and two hard SIP Phonbes on the LAN connected to it
18:01.41okieplayahey someone have some time to help me setup my new S100I im sorry i just dont under stand the pdf info ?
18:01.57__Sparks_the router is connected to this same LAN
18:02.03adjacentanyone used asterisk@home ?
18:02.06rontecxt44let me try again without the annoying extra crap
18:02.22rontecxt44anyone run into "TDM PCI Master abort" ?
18:02.30adjacentmore precisely, anyone willing to give it a thumbs up or thumbs down =)
18:02.33rontecxt44then the card shuts down
18:02.48roamer323__Sparks : and you have taken at least 1 incoming call from each of your registration in the past?
18:03.41roamer323adjacent - one thumb down
18:03.46*** join/#asterisk didz_ (didz_@200.218.192.52)
18:03.46SuPrSluG__Sparks_:you need  to port forward
18:04.10SuPrSluG__Sparks_:better off w/ iax
18:04.44adjacentroamer323: reason?
18:05.05__Sparks_SuPrSluG - I have the followinf forwarded UDP ports - 10000->20000  4569   5060->5070
18:05.37__Sparks_SuPrSluG - I tried putting the asterisk box in the DMZ zone, and still got the same errors
18:05.45roamer323adjacent - you get 10+ users coming on here with the same questions on the same bugs every single day - enough reasons
18:06.04__Sparks_I have just turned off all the accounts except one that wes erroring, and now the errors have stopped
18:06.09adjacenti hear ya.
18:06.27SuPrSluG__Sparks_: i think fwd allows iax. use that to register. examples on wiki
18:06.29adjacentbugs is a good reason for a thumbs down
18:06.55outtolunc..
18:07.00okieplayais there somewhere i can read more on how to setup the digium s100i other than the pdf they have ?
18:07.08adjacentbut probably 9/10 users coming here with problems are too dumb to use the product, or read the manual
18:07.14okieplayaor there i tech number i can call
18:07.15__Sparks_SuPrSluG, it isnt only FWD I use - i have Sipgate x3 - FWD and voiptalk
18:07.36adjacentits downloading now. ill give it a look anyway
18:08.45roamer323adjacent - good luck :-)
18:09.02adjacentheh. thanks =)
18:09.11*** join/#asterisk zpn (~xpn@dhcp-152.digium.com)
18:09.26*** part/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com)
18:09.29adjacentimho, its going to be better than VOCAL any way you look at it
18:09.59roamer323__Sparks_ you're best off turning on "sip debug" and do a message by message trace to see what's going on
18:11.57zpni'm having problems dialing out on my sip.  i get a 'Everyone is busy/congested at this time' with a warning that 'No channel type is registerd for 'Zap', then "Unable to create channel of type Zap'  any ideas on how to fix this?
18:12.07SuPrSluG__Sparks_: which one works?
18:13.34__Sparks_<SuPrSluG> - whick what works? - SIP account?
18:14.20SuPrSluG__Sparks_: voiptalk has asterisk setup
18:15.21Blackveloh my dear
18:15.30Blackvelnow I even understand my DISA 1.0.5 problem
18:15.42__Sparks_<SuPrSluG> - all the accounts seem to work ok, just the errors are shown - and when it is erroring there seems to be big delays in the system when dialing out (including if I am using a POTS line
18:16.08Blackvelmy analog pbx is dailing too fast, DISA does not pick up fast enough so the rest of the extensions don't get used for extensions dialplan
18:16.27Blackvelthat can be everyething
18:16.31Blackvelnew DISA code in 1.0.5
18:16.35Blackvelnew zaphfc code
18:16.42Blackveltoo bad
18:17.07*** part/#asterisk zpn (~xpn@dhcp-152.digium.com)
18:17.14SuPrSluGzpn:check lsmod ? is it loaded?
18:17.16Blackvelhow can I find that out?
18:17.31*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
18:21.41*** part/#asterisk rontecxt44 (~rontecxt4@dsl9-173.rb.comporium.net)
18:22.44*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
18:24.11*** join/#asterisk trym (~trym@linux.debian.us)
18:24.46SlainteHow do I get * to have a cold beer ready for me by the time I get home from work?
18:25.22znoTry exten => _XXX,s,Beer(${num_cans{)
18:25.31Slainteits about time someone asks a question with some importance
18:25.43PinholeSlainte, I'm pretty sure you'll need some AGI code for that.
18:26.09*** join/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net)
18:26.13SlaintePinhole,  I think so.  I wonder what variable I use for the temp of the beer.
18:27.09PinholeYou should be able to have it order over the net when supplies are low.  Some low level robotics could also help.  I think that's in the home automation stuff.
18:27.35*** part/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net)
18:27.49SlainteWhile thirsty=yes;do pour_beer
18:28.18Slaintesimple loopk, yet so powerfull :)
18:28.37PinholeI thought it would be useful to have sphinx2 and festival create a gateway to this IRC channel, but sphinx2 does not perform well enough.
18:29.22BeirdoPinhole: then we could all curse you out for the fun of it
18:29.45Beirdocurse you in German and see how well festival handles it, etc :)
18:30.22Pinholesphinx: THE CAN PHONE FIT
18:30.34adjacentwhile (thristy) { if (glass_empty) pour_beer(); else drink_beer(); }
18:30.34adjacentotherwise you would have a big mess to clean up
18:30.57Slainteyipppy  I just won the porn lottery!!,  What a day of good news.  First I qualify for a home loan,  then I win the porn loterry, and then I get an offer to help some african government funnel out money
18:30.59SuPrSluGgrandstream has a new firmware update.
18:31.03CoaxDwhee
18:31.14*** join/#asterisk [cc]smart (~smart@62.65.149.158)
18:31.21Slaintewho says the internet is not full of good stuff
18:31.56Pinholewhile(thirsy) { if(glass_empty && !drinking_beer) pour_beer(); if(!beer_pouring && ! drinking_beer) drink_beer() }
18:32.05tzangerSlainte: heh
18:32.17__Sparks_anyone here know about Xorcom Rapid?
18:32.26Slainte__Sparks  yes
18:32.29CoaxDpinhole: that would be a scarey robot. *lol*
18:32.36SlaintePinhole,  nice code
18:32.49adjacentnow write a peripheral driver to control a device that interfaces with a 12 pack, and use it in pour_beer. then set an extention in * to start it up ;)
18:32.55tzangerdoesn't work worth a shit since ther's no flags to control the pouring
18:33.00Beirdochange it to "while(!passed_out)"
18:33.04*** join/#asterisk Fanguin (~Fanguin@p508187ED.dip0.t-ipconnect.de)
18:33.19*** join/#asterisk trym (trym@linux.debian.us)
18:33.25Pinholewhich is more complicated, call routing or pouring beer?
18:33.29Beirdotzanger: that's all in the pour_beer routine
18:33.35SlainteSparks  what is your question about Rapid
18:33.35adjacenttzanger: use mcdonalds "idiot-proof" style pouring
18:33.38CoaxDpinhole: Hmmm.  Pouring beer would be far more complex
18:33.43tzangerBeirdo: that only gets executed if the glass is empty
18:33.51CoaxDpinhole: (to a robot who thinks in digital terms, doing an analog action.)
18:33.53tzangeronce you start pouring the glass is no longer empty even though there's only a little beer in it
18:34.13adjacenttzanger: output 12 oz and receck.
18:34.17Beirdobut the routine doesn't return immediately :)
18:34.17__Sparks_Slainte - I keep trying to do the Maintanance - Update Software Inventory, but it keeps failing
18:34.17CoaxDtzanger: Of course, you'd have to write a glass_is_half_empty_or_glass_is_half_full() routine
18:34.18Beirdohehe
18:34.32tzangerCoaxD: yeah but you'd get bogged down in that routine with semantics
18:34.40SlainteSparks  throw me a bone,  what does "it" do
18:34.42CoaxDtzanger: Sadly enough, you're correct :/
18:34.49__Sparks_Slainte - I just tired pinging updates.xorcom.com, then doing the updates worked!?
18:34.54adjacent#include <philosohy.h>
18:35.09km-coaxd: sup.
18:35.11Pinholewhile(true) { offer_beer_bottle(); while(!beer_taken); }
18:35.12CoaxDadjacent: #error i cant parse this data so i'll stop compiling
18:35.16__Sparks_Slainte - usually sticks at - 40% [Connecting to updates.xorcom.com (1.0.0.0)]
18:35.21CoaxDkm: Nada, pete. what you up to? :)
18:35.26*** join/#asterisk ScarletCrusader (~GMMiller@wsip-66-210-74-254.mc.at.cox.net)
18:35.35km-coaxd: we just got a te405p in the office, I'm 20 mins from my changeover :)
18:35.45km-asterisk++
18:35.45CoaxDkm: OOOH!!!
18:35.52CoaxD^5
18:36.05km-Got a T1 from the CO running to asterisk, then another T1 running to the NEC system
18:36.10CoaxDkm; We're running * inhouse here on a single POTS line, and also, i have a voip install in the same building.  I like *. :)
18:36.13km-gonna transition off the NEC gradually
18:36.16SlainteSparks,  check your /var/log/messages
18:36.22SlainteWhat are you trying to update?
18:36.23km-yeah, I've got some work ahead of me here :)
18:36.57adjacenthttp://www.telegraph.co.uk/news/main.jhtml?xml=/news/2005/02/18/wpill18.xml&sSheet=/news/2005/02/18/ixworld.html
18:37.01km-luckily I've already got experience running asterisk in full voip and POTS/voip configs so this is just adding T1 knowledge to the pile!
18:37.30*** join/#asterisk mud (~mud@bestekdsl.customer.sentex.ca)
18:37.37km-slainte: Amen.  That's like, what you do when you're a hobbyist.  Can't do that in a business setting!
18:37.38__Sparks_Slainte - I am just checking for avalable updates
18:37.52km-if you're a hobbyist, who cares if it breaks, it's a learning experience
18:37.53__Sparks_Slainte - what am I looking for in /var/log/messages?
18:38.01SlainteSparks,  problems :)
18:38.23SlainteSparks,  do you have much *nix experience?
18:38.32__Sparks_Slainte - the last load of messages are all Feb 23 18:24:05 localhost -- MARK --
18:38.44Slainteok try the /var/log/syslog
18:38.44__Sparks_Slainte - no, cant you geuess :-
18:38.57SlainteSparks,  heheh no biggy,  a few things you need to know about unix.
18:39.09Slainte1.  You rarely need to reboot it
18:39.24Slainte2.  dont listen to km-
18:39.26Slainte:)
18:39.31km-HAHAHAHA
18:39.32__Sparks_lol
18:39.34tzangerkm-: I do that too except it's
18:39.37tzanger*.* -/var/log/all
18:39.41km-tzanger: it fills up a drive quickly :)
18:39.42Slainte3.  Logging will set you free.
18:39.45tzangerkm-: nonsense
18:39.53tzangeronly if you've got something spewing so badly
18:40.00__Sparks_in syslog, the last three messages are
18:40.00__Sparks_Feb 23 17:44:05 localhost -- MARK --
18:40.00__Sparks_Feb 23 18:04:05 localhost -- MARK --
18:40.00__Sparks_Feb 23 18:17:01 localhost /USR/SBIN/CRON[1910]: (root) CMD (   run-parts --report /etc/cron.hourly)
18:40.09km-tzanger: run snmpd with a monitoring system like Nagios, and you'll see! :)
18:40.10__Sparks_there is a lot like that
18:40.11SlainteDONT PASTE to #*
18:40.12adjacent3a: Remote logging will keep you free =)
18:40.21__Sparks_sorry!
18:40.29Slaintepastebin.ca is your friend
18:40.31Slainte:)
18:40.39km-__Sparks: don't worry about Slainte, he's just trying to sound intelligent and in control :P
18:40.41trymor pastebot.org
18:40.42trym;)
18:41.01Slaintehahaha  km-
18:41.16SlainteI get very few chances a week to do this, so dont piss on my parade please :)
18:41.24km-hahahaha
18:41.25km-word :)
18:41.36goatmilkif this is your parade you should get out more often
18:42.07km-hmm
18:42.21__Sparks_anyway, the apt-get update seems to work after i did a ping to updates.xorcom.com
18:42.35Slaintethats wierd
18:42.45Slaintenameservers are in /etc/resolv.conf
18:42.56Slaintecheck to see if anything is listed
18:43.00__Sparks_...he says, as it is now stuck on "40% [Connecting to updates.xorcom.com (1.0.0.0)]"
18:43.16__Sparks_is the 1.0.0.0 the IP it is trying to get to!?!
18:43.45__Sparks_DNS is fine, i can ping anything i have tried
18:44.28*** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca)
18:44.35Slainte1.0.0.0 is not the IP no.
18:44.47Slaintedo a netstat -a | grep xorcom and see where it is connected
18:44.57__Sparks_in /etc/resolv.conf i have the IP of my router and  212.135.1.36
18:45.17lyroyDoes someone can tell me how can i change the tftp server of my phone in telnet...
18:45.42__Sparks_I did a "netstat -a | grep xorcom" but it just returned a prompt
18:45.56Slaintenetstat -a | more
18:46.06km-to pastebin please
18:46.09km-dont want to see that netstat
18:46.10km-:P
18:46.20Slaintehehehe  very true
18:46.39goatmilkhow about we just take all of our chit chat to pastebin
18:47.15SlainteI am still waiting for you spin tops to sort out my beer code for *
18:47.26__Sparks_http://pastebin.ca/6342
18:47.40marc_cis redund abs nec?
18:48.11BeirdoWTF?
18:48.39SlainteSparks.  its not running anymore.  Does it freeze at 40% or did you close it?
18:48.52__Sparks_I colosed it :)
18:48.58__Sparks_Closed it
18:49.16*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
18:49.19Slainteput your game hat on.
18:49.27__Sparks_I will do it again, sorry!!
18:50.14SlainteAn Irishman and Englashman and a Welshman walk into a pub.  The bartender looks up at the three of them, and says..  is this a joke?
18:50.59__Sparks_done, and updated pastebin
18:51.03SlainteI will be here all week.  Please dont put your smokes out on the floor, and please tip your waitresses
18:51.59SlainteSparks  same pastebin number?
18:52.15__Sparks_yep
18:52.21SlainteI dont see any whanges
18:52.40__Sparks_sorry - http://pastebin.ca/6343  I thought it just updated it
18:52.53SlainteI am getting frustrated with you Sparks
18:53.08__Sparks_sorry, I'm new to all this :-S
18:53.34SlainteI allready told you to put your gamehat on.
18:53.39SlainteLine 7 is a problem.
18:53.46Slaintepastebin your /etc/hosts file
18:54.40__Sparks_127.0.0.1       localhost.localdomain   localhost       tsbserver02
18:54.40__Sparks_# The following lines are desirable for IPv6 capable hosts
18:54.40__Sparks_::1     ip6-localhost ip6-loopback
18:54.40__Sparks_fe00::0 ip6-localnet
18:54.40__Sparks_ff00::0 ip6-mcastprefix
18:54.41__Sparks_ff02::1 ip6-allnodes
18:54.43__Sparks_ff02::2 ip6-allrouters
18:54.45__Sparks_ff02::3 ip6-allhosts
18:54.47__Sparks_shit, sorry!
18:55.17km-5 mins to changeover!
18:55.18km-hehe
18:55.55__Sparks_http://pastebin.ca/6344 - not that it matters now
18:56.43*** part/#asterisk Fanguin (~Fanguin@p508187ED.dip0.t-ipconnect.de)
18:56.54__Sparks_...someone at the door now, BRB
18:57.55Slaintedont let it hit you in the ass on the way out
19:00.36km-thats weird
19:00.40km-where's twisted
19:00.40km-hehe
19:02.46km-how do I store a digit?
19:03.08km-I want to save ${EXTEN} as a variable
19:03.10km-how do I do that?
19:03.37km-I'll use dbget/dbput
19:05.18km-shit
19:05.35bjohnsonsetvar
19:06.00RoyKshow application setvar
19:06.20km-theres something wrong with this fucker
19:06.20km-dammit
19:06.42RoyKcopulator?
19:07.01km-there goes my window again
19:07.09km-its this darned NEC system, it just doesnt play right
19:07.19RoyKrunning windows?
19:07.22km-no
19:07.46*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
19:08.07shido6anyone have a serial ata poweredge box and needed to install redhat 9?
19:10.04trelanenope
19:10.22RoyKshido6: use gentoo, or debian, or slackware, or even SuSE, but forget redhat 9
19:11.30km-~seen twisted[work]
19:11.31jbottwisted[work] is currently on #asterisk.  Has said a total of 49 messages.  Is idling for 3h 5m 14s
19:12.07km-Asterisk just doesnt want me to do this
19:12.08km-hehe
19:12.34Delvarnn all
19:13.27shido6RoyK thanks
19:16.04*** join/#asterisk XeNoSiS (user@216.234.145.18)
19:16.11XeNoSiSHello Hello
19:16.21XeNoSiSJust the people I was looking for.
19:16.58RoyKshido6: gentoo will be my first choice
19:16.59RoyKbtw
19:17.02XeNoSiSI am trying to setup Asterisk with a Cisco 2600 as the PSTN to the gateway. Inbound / Outbound calls appear to setup properly but there is no audio either way.
19:17.32XeNoSiSCisco2600 gateway to the PSTN that is. Sorry. I am tired.
19:19.09XeNoSiSSoftphone (Xten Xlite) <---> Asterisk <---> Cisco 2600 <---> Nortel PBX <----> Phone
19:19.38XeNoSiSany sample configs for Cisco 2600 Gateway configuration? or ideas on why i get no audio?
19:20.01*** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com)
19:20.49*** join/#asterisk stepcut (~user@207.67.194.2)
19:21.18*** join/#asterisk sonic_br (~asimoes@ip-64-32-179-115.dsl.nyc.megapath.net)
19:21.24yashaI have seen some references of integrating * with Microsoft CRM. Does anyone have any info on that or can point me in the right direction? Thank you.
19:26.21XeNoSiSI don't think anyone is really here. :)
19:26.34km-eh, there are people here
19:26.35km-hehe
19:27.09XeNoSiSSo no one has done anything with using cisco for the PSTN gateway off an asterisk system?
19:27.17XeNoSiSOr just don't feel like helping right now?
19:28.48*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
19:28.48*** mode/#asterisk [+o bkw_] by ChanServ
19:29.05nestArheh
19:29.17*** join/#asterisk Graphikos (~Graphikos@71-32-6-49.spkn.qwest.net)
19:29.28RoyKyasha: there's something like asterisk tapi support
19:29.33RoyKyasha: that'll be all I know
19:29.38bjohnsonsomehow I've got colours showing now on my cli and it's leaving codes in my message log .. making it harder to read.  How can I correct what goes into the logs?
19:29.53nestArcolors are fun!
19:29.59Graphikosanyone brave enough to take a complete asterisk newbie under thier wing a bit?
19:30.06*** join/#asterisk ACiDV (~joel@69.156.197.246)
19:30.06RoyKnestAr: colours?
19:30.12GraphikosI could use a bit of direction...
19:30.20nestArRoyK: depending on your locale.. yes.
19:30.21nestAr:)
19:30.30RoyKnestAr: what about them? CoVoIP?
19:30.32RoyK:P
19:30.38nestArlol
19:30.57nestAri sometime use colourful language oIP
19:31.00MicH323Anyone explain what this means: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible with our capability 0xff03
19:31.17bjohnsonlooks like ansi colours on the cli
19:31.18RoyKMicH323: codec problems
19:31.22ACiDV=) Does it's possible to get the "original phone context" ? I have sip/zap phones that have context=default, others have context=full, etc... in a macro, does it's possible to get the context of the caller ?
19:31.29MicH323Ah, Thans Roy
19:31.33bjohnsonthey save codes into the messages log file
19:32.15MicH323Graphikos: I am a total newbie myself... Just up today! :)
19:32.31*** join/#asterisk km- (~km-@67.105.178.130)
19:32.50nestAranyone know how to supress the channel reset messages from the CLI?
19:33.20XeNoSiSDoes the gateway have to be SIP? or can i setup an h323 cisco gateway?
19:33.55ACiDVTo create a macro like: [macro-lastredial] exten=> s,1,DBGet(last=...)    s,2,Dial(Local/${last}@${ORIGINALCONTEXT} ?
19:34.03MicH323XeNoSiS: You can setup Cisco gateway to do SIP, but it cant register
19:34.17ACiDVhmmm not sure if I self answer... I think I've see a variable for this... go to wiki :|
19:34.40XeNoSiSWe setup the gateway as SIP and calls work in / out but there is no audio.
19:34.49MicH323XeNoSiS: You have to compile the H323 into Asterisk
19:35.25MicH323Are you behind NAT?
19:35.32XeNoSiSno
19:35.44MicH323What GW do you have?
19:36.17*** join/#asterisk tufone_ny (~asimoes@ip-64-32-179-115.dsl.nyc.megapath.net)
19:36.28XeNoSiSCisco 2600 with an NM-HDV-T1 PRI connected to a Nortel PBX.
19:36.47MicH323So you are using sip-ua?
19:37.09MicH323And dial-peer xx voip?
19:37.41tclarkok who knows how to interface these FRS GMRS walkie talkie to * ? what h/w to wee nned to make chan_gmrs ?
19:37.48*** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net)
19:38.23harryvvanyone here running a spa1k? need to ask a quick config question.
19:38.49XeNoSiSyeah I am using sip-ua and dial-peer voip
19:39.04XeNoSiScan you send me a /msg with your email and I will send the config
19:40.15*** join/#asterisk dstevens_ (~dstevens@cpc3-ches1-4-0-cust87.lutn.cable.ntl.com)
19:41.00PinholeIs there an automated way of testing the call quality through asterisk?
19:41.12nix000anyone tried running asterisk with linksys ata ?
19:42.21RoyKPinhole: no
19:42.30RoyKPinhole: ethereal can do some, though
19:42.55harryvvAre there free bandwith measuring tools on the market?
19:43.11bjohnsonfor?
19:43.26bjohnsondsl speed?
19:43.44harryvvSay I want to see what a customers bandwith on there network before sugesting the idea of voip
19:44.03harryvvand measure the use dsl speed to.
19:44.13*** join/#asterisk cp5 (~samy@dsl081-232-019.lax1.dsl.speakeasy.net)
19:44.15cp5hola
19:44.15bjohnsonI get mrtg graphs on my ipcop box
19:44.23cp5anyone know how to make hints work for outgoing calls?
19:44.27t3tharryvv: It's more about latency and jitter than bandwidth... you need to measure the connection over time
19:44.35bjohnsonmrtg just makes the graphs though .. I think they get the data from iptables
19:44.44harryvvt3t i know over time.
19:45.05t3tharryvv: like over a few weeks, not just a b/w test once or twice
19:45.27PinholeWhat I'm really trying to do is detect * crashes.  Some crashes leave * working, but badly distorting calls.
19:45.50ionixuse ser *cough* for sip server
19:45.50bjohnsonnix000: yes
19:46.01harryvvwell, If a voip consultant walks into the biz he wants to keep the delay of measuing and providing a package service to a min or the company president might change his mind.
19:46.02t3tharryvv: With dsl/cable type services, unless there is an SLA in place for prioritized voice traffic, the connection probably won't always be voice-clean
19:46.28harryvvSLA?
19:46.39t3tService Level Agreement
19:46.46PinholeIt's *not* a bandwidth issue.  It did this on the local network while we were testing.
19:46.51harryvvohh i see yea, set aside some bandwith for voice only traffic.
19:47.11t3tpart of the contract that specifies how they will handle the connection specifics and what they pay you when they don't
19:47.41yashaRoyK: Thank You....... I was hoping that someone would have already looked at it...
19:48.10t3tharryvv: tcptraceroute is a good tool to do external latency monitoring
19:48.15harryvvwhat POe router would prioritize network traffic based on what port number?
19:48.24bjohnsonany linux one
19:48.29t3tPOe?
19:48.32loudi cant believe broadvoice only supports g711.
19:48.36nix000bjohnson, was it working out of the box ?
19:48.36bjohnsonoh .. power over ethernet?
19:48.37PatrickDKharryvv, any real l3 router
19:48.43harryvvlevel3
19:48.44*** join/#asterisk greendisease (~jack@greendisease.fedora)
19:49.26t3tLevel3 is a communications company, L3 is ISO network model layer 3
19:49.35bjohnsonnix000: 1. I don't have one 2. you will of course have to configure it 3. only the PAPs marked -NA will not be locked and can therefore be used
19:49.36harryvvyea I know
19:50.28*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
19:50.28*** mode/#asterisk [+o bkw_] by ChanServ
19:50.29dstevens_Hi all, has anyone tryed compiling asterisk on debian/ubuntu as its failing on ssl, even though " dpkg --get-selections | grep openssl " shows openssl installed.
19:50.29t3tharryvv: what do you mean by a 'POe router'
19:50.37t3twhat's up, bkw
19:50.48bkw_not much
19:50.59t3tjust wake up ;)
19:51.09DJ-Pyrodstevens_: openssl-dev
19:51.11bjohnsondstevens_: a third of the people here run debian (I do not)
19:51.35dstevens_DJ-Pyro, Will check that out thanks back in a bit.
19:51.38cp5anyone use hints in asterisk? any docs on using them for outgoing calls?
19:51.50DJ-Pyroerr, nevermind
19:51.52marc_cany cnd here?
19:51.57stevekstevekhmm, outgoing calls..
19:51.59DJ-Pyroyou need the development libraries, I foret what they are
19:52.00stevekstevekcan it do that?
19:52.13DJ-Pyrodstevens_: libssl-dev
19:52.14DJ-Pyrosorry
19:52.39t3tcp5: are you referring to registration hints?
19:52.51*** part/#asterisk ACiDV (~joel@69.156.197.246)
19:53.04dstevens_bjohnson, Hi ubuntu is debian based, i use deian but prefer a humane interface
19:54.04shido6dstevens_
19:54.07shido6how IS ubuntu
19:54.10shido6Ive seen it
19:54.13shido6never used it tho
19:54.56dstevens_Very nice, gnome only based only at the moment.
19:56.18dstevens_Although Kubuntu is on its way and looking good if your a kde' a
19:56.34cp5t3t, well, call hints. so when someone calls an extension, a special phone configured (such as a secretary's phone) can see that line is getting a call
19:58.16*** join/#asterisk rephorm (~rephorm@ip67-95-13-62.z13-95-67.customer.algx.net)
19:59.24*** join/#asterisk Grooby (~Grooby@12.22.232.212)
20:01.02rephormhello, i'm setting up call recording (via monitor) in asterisk, and was wondering if it were possible to initiate it while in a conversation from the phone keypad (without transfering or switching to a conference)
20:02.04*** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org)
20:02.19`Sauronls
20:02.41*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-22-254.d4.club-internet.fr)
20:03.27rephormi.e. can i have an agi script sit an wait for dtmf during a call?
20:03.39*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
20:09.00bjohnsonhow is ubuntu different from the hundred other distros?
20:09.24tzangerbjohnson: it's got a cool name?  <rolls eyes>
20:09.43Sedoroxbjohnson: its a love CD that actually runs Gnome instead of KDE
20:09.49Sedoroxotherwise its like knoppix
20:09.54tzangera love CD?
20:09.55tzangeroh live
20:09.56tzangerhaha
20:09.57bjohnsonhehe
20:10.09tzangerI'm like "whoa they're REALLY taking this tribal theme too far"
20:10.18Sedoroxyea live.. sorry
20:10.21bjohnsonlet's vote them off
20:10.29Sedoroxif you really wanna use a cd for love... ummm... thats your thing
20:10.57*** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz)
20:11.18aminorexthere was a pakistan-made IAX phone in the works.  what's the link?
20:11.42*** join/#asterisk lorion (~van@63.115.106.66)
20:12.07gr0mitwww.farfon.com
20:12.16lorionAre there any good docs regarding NAT?
20:12.21bjohnsonyes
20:12.41LoganHas anyone here had any problems with Sipura's phones?
20:12.42bjohnsonbut basically it's a system to reuse IP addresses
20:12.42aminorexanyone seen the faron product in the wild yet?
20:12.54gr0mitlogan, yes
20:13.03bjohnsonaminorex: zeeek is waiting for one
20:13.28*** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com)
20:13.29gr0mitsipura 841 has strange behaviour with tx audio
20:13.32dsmouse'lo y'all
20:13.49harryvvgr0mit im not impressed with the looks of the sipura 841
20:14.03gr0mitsticky buttons
20:14.13gr0mitrx audio is very good
20:14.34gr0mitbut tx seems to have strange muting if the audio is too quiet.
20:15.00lorionHow would you configure a phone that was nat'd.. For example the Asterisk server is on one network and the phone is nat'd on another.
20:15.18lorionIt is a remote sales agent config.
20:15.28gr0mitdepends on the phone, lorion!
20:15.31modulus_anyone from broadvoice here?
20:15.50lorionI am testing with X-lite
20:15.59modulus_broadvoice is down again
20:16.09gr0mitx-lite just worked for me
20:16.11modulus_this is like number 3
20:16.23modulus_when receiving calls, they hear nothing
20:16.24*** join/#asterisk Zaw (zaw@zaw.subneural.net)
20:16.24gr0mitseems to have its own stun server
20:16.31pdracevichaaaaaaaaarrrrrrrrrgggggggg!!!!!!!!1 can some one have a look and give me some advice! http://pastebin.ca/6345
20:16.46lorionare there ports I need to open?
20:16.53bjohnsonlorion: add nat=yes to sip.conf
20:16.54gr0mithave you set nat=yes in your sip.conf file?
20:17.15bjohnsonlorion: is the * server behind nat?
20:17.15tzangerpdracevich: interesting
20:17.19tzangerhow are you getting that
20:17.20lorionmy xlite connects, I have nat=yes, but if I try to dial an extension it give me a 404
20:17.34lorionmy x-lite can pick up vm
20:17.56pdracevichtzanger: I had the two sip boxes talking and rules donw so the calls are being placed went home and now this
20:18.04*** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com)
20:18.12pdracevichtzanger: iax2 debug
20:19.19*** join/#asterisk visik7 (~ciao@visik7.user)
20:21.13*** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
20:22.48ScarletCrusaderis there a newbi cnannel I can ask easy question that i cannot figure out?
20:22.57pdracevichtzanger: any idea?
20:23.38nestAranyone know how i can supress these " B-channel 0/2 successfully restarted on span 2" messages?
20:23.44modulus_OMG broadvoice sucks
20:23.58*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-199-86.dsl.scarlet.be)
20:24.31[Outcast]modulus_: what is wrong with broadvoice?
20:24.40BrianR___Mmm... 12648430
20:24.58modulus_outcast, all calls are answered by *
20:24.58PatrickDKwrong? broadvoice? heh
20:25.07modulus_but the caller can't hear anything
20:25.07PatrickDKbroadvoice won't follow sip standards correctly
20:25.16modulus_patrickdk, i'm starting to realise that too
20:25.24modulus_nothing changed in my asterisk box
20:25.28modulus_it just stopped working
20:25.50modulus_and their number on their site just has you hold for 15 minutes then says "your call cannot be transfered. thankyou"
20:26.04[Outcast]modulus_: I have been using broadvoice for over six month without a problem
20:26.17modulus_outcast, this is the 3rd time broadvoice is down
20:26.36modulus_i'm on the west coast
20:26.37[Outcast]modulus_: mine is working fine right now.
20:26.58[Outcast]modulus_: i am using the lax server as well
20:27.14modulus_outcast, what version of *?
20:27.28ScarletCrusaderis there a documentation site which goes through the procedure of adding a sip phone to asterisk?
20:27.41dsmouse~rtfw
20:27.42jbotfrom memory, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
20:27.48[Outcast]modulus_: 1.0.5
20:28.01[Outcast]modulus_: without the broadvoice patch
20:28.06modulus_outcast, yesterday no problems
20:28.12modulus_i don't use the broadvoice patch either
20:28.21[Outcast]modulus_: no problems yesterday either
20:28.27modulus_this morning, all calls coming in via broadvoice can't hear anything
20:28.30modulus_it's just silence
20:28.32modulus_dead silence
20:28.38modulus_i watch the cli and asterisk picks up fine
20:30.08modulus_on hold... (again)
20:31.41tzangerpdracevich: not offhand
20:31.45tzangerget a packet cap and post it
20:32.43dsmousemodulus_: what did it say was the codec it was useing?
20:32.49modulus_ulaw
20:33.00modulus_did they stop using ulaw?
20:33.12jaigertzanger, have you got the echo canceller yet?
20:33.15tzangertcpdump -s0 -w blah.dump host 1.2.3.4 and host 5.6.7.8 and port 4569
20:33.25dsmousebeats me, but that sure could be a codec mismatch
20:33.27tzangerjaiger: yes but I need pinouts :-)
20:33.28terrapencan anyone recommend a good German voip provider?
20:33.29[Outcast]broadvoice will stop using ulaw
20:33.38[Outcast]never
20:33.46terrapeni'm going to visit a friend in Germany and i'd like to set his business up with an asterisk pbx
20:33.53jaigertzanger, I upgraded mine with a daughter card today - sounds pretty good now
20:34.03jaigertzanger, didn't I get that to you?
20:34.11dsmouseterrapen: you've written a app_ before, right?
20:34.43tzangerdaughter card?
20:34.47tzangerjaiger: I have a different echo can
20:34.58jaigertzanger, ahh, what'd you get?
20:35.18tzangeruh
20:35.19*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
20:35.22jaigertzanger, yeah the 2572 can be upgraded with 2 daughter cards
20:35.35tzangertrying to think what it is now
20:35.39tzangerI can't remember offhand
20:36.23shido6shido6
20:36.41jaigertzanger, one card adds time - up to 128ms tail - and the other adds some other dsp features like audio level adjustment
20:36.43cp5has anyone used extension 'hint's and noticed that they stop working after an asterisk reload?
20:36.46cp5for outbound alls
20:36.47cp5calls
20:36.47tzangerjaiger: ahhhhh
20:37.10*** join/#asterisk ReVoK (ReVoK@82.224.60.46)
20:37.12ReVoKhi
20:37.18tzangerthis is bugging me now
20:37.19tzangerwtf was it
20:37.22jaigerand there are multiple versions of the 'feature' card with more/less features
20:38.28tzangersounds hideous
20:38.36tzangerso what is the total cost of your echo can now?
20:38.45tzangerand did you get the upgrade cards off ebay or from tellabs themselves?
20:38.56*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
20:39.11km-tzanger: hey dude
20:39.15tzangerkm-:
20:39.21km-tzanger: twisted had an excellent idea for my problem, till it didnt work
20:39.29jaigertzanger, I got the upgrade card from ebay but also found some vendors as backup
20:39.33tzangerkm-: hahaha what was his idea?
20:39.39km-tzanger: he said do incoming=yes and then set extension s,1 as DISA
20:39.43km-tzanger: with no password
20:39.47tzangeryou mean immediate=yes
20:39.50tzanger?
20:39.54km-yeah, immediate=yes
20:40.04jaigertzanger, the upgrade card came mounted on another 2572 for ~$80 shipped
20:40.06tzangerkm-: why didn't that work?
20:40.11tzangerjaiger: nice
20:40.13km-tzanger: but for some reason, the digits dialed weren't being sent over the T1
20:40.13tzangervery nice
20:40.18tzanger...
20:40.20jaigerand a vendor quoted me $60 for the card alone - not too expensive
20:40.22tzangeryou have some very strange problems
20:40.22km-tzanger: I think that the NEC does have some sort of dialplan problem
20:40.39tzangerkm-: and those fuckers are so hard to configure compared to my norstar
20:40.48km-tzanger: hehe
20:40.50[Outcast]I don't understand why you guys have so much trouble with broadvoice. I use to the asterisk support for them. If there is any way i can help let me know
20:40.58km-I'm thinking of requiring everyone to dial 9 twice before dialing
20:41.03km-but thats not much better than dialing # at the end
20:41.11tzangeryeah?
20:41.26NuggetI think that dialing prefixes are a curious relic from a simpler time.  I think they're avoidable with modern technology and that avoiding them is good.
20:41.39km-because I can always set an exten => 9 to get them in the context waiting to do some magic
20:41.45tzangerNugget: not with his NEC it seems -- not unless he can reprogram it
20:41.58km-yeah, I need the special programming card to fix this problem
20:42.00km-which is $$$
20:42.00tzangermy norstar can't be told to avoid 9 either
20:42.03Nuggetoof
20:42.17tzangersince you can't select a PRI channel like you can an analog or even CAS T1 trunk
20:42.20tzangerso you have to route it
20:42.29tzangerkm-: you do?  You can't do it through a handset?
20:42.33km-ariel was saying that Manx had a problem similar to this that he fixed by changing his wink times
20:42.44km-tzanger: I can't change the dialplan settings without the programming card I think
20:42.48km-tzanger: I can change DID's
20:42.51tzangerowie
20:42.53km-tzanger: but I dont know about dialplan
20:43.51km-it sucks because, Asterisk is working fine, but the PBX is what's not working, but the company will naturally implicate Asterisk, because "it works fine otherwise"
20:44.09terrapeny
20:44.12Nuggetheh, yep.
20:44.37km-unless the wink timing stuff is the magic
20:44.56km-oh.
20:45.04jaigertzanger, I've probably put out < $300 for the echo can hardware I use.  I have purchased another $150 in 'spare' parts while dicking around
20:45.31tzangerjaiger: *nods*
20:45.41harryvvWhen doing a sip show peer on a peer I get this info and mabey it is interfearing with athentication? Expire       : -1
20:45.42harryvv<PROTECTED>
20:45.43km-tzanger: I could setup extensions 1 and 0 in preincomingpbx, then create seperate contexts for "toll dial' and "international dial" and then force everyone to dial the 1 for calls
20:45.45tzangerkm-: have you tried playing with the wink settings?
20:45.59tzangerkm-: you could, but that's painful too
20:46.21km-tzanger: no, I don't have the knowledge to know what I'm doing with that, I'm afraid that if I fiddle with settings I may damage something?
20:46.30modulus_ugh i hate broadvoice
20:46.35km-tzanger: or do you think it's fine to fiddle with those?  Like I said, I'm a T1 n00b
20:46.46modulus_i've never had any problems with any voip providers
20:46.49modulus_broadvoice is the first
20:46.53km-voicepulse
20:46.59km-I've had problems with voicepulse before
20:47.07dsmousemodulus_: I've had problems with vonage.
20:47.09tzangerkm-: damage it: no...  but fuck it up and not know how to fix it: likely.  That's why I was so hesitant about doing anything with the norstar but now I understand it to a very good level and I'm fearless
20:47.10km-only provider I trust anymore is NuFone
20:47.15bjohnsonmodulus_: I rmember you bitching about nufone
20:47.29modulus_bjohnson, they just have shitty bandwidth
20:47.30tzangerkm-: you can fiddle with the settings and you will not break anything, it's all just bits after all
20:47.41tzangeryou can make it stop working but you can comment out your changes and get it back to the level of not working you have today :-)
20:47.42modulus_broadvoice just has shitty services
20:47.43moonwickheh
20:47.45Logangr0mit: We've had three phones fail in a manner that suggests a bad NIC or something.
20:47.49km-yeah, definitely
20:48.26modulus_i still get those "timed out lagged" messages for nufone
20:48.36tzangermodulus_: packet cap
20:48.45modulus_and i'm on a gig-e link upstream via global crossing
20:48.51modulus_tzanger, what?
20:48.52tzangersounds like you just stop receiving data for 2s and asterisk auto-congests
20:49.01tzangermy ADSL line has that problem from time to time
20:49.02km-bbiafm
20:49.03dsmousemodulus_: /me is envious
20:49.29tzangerit's not nufone's issue...  I have some jitter packet caps that make me cry with respect to how deterministic their network is
20:49.47modulus_they could put a proxy in LA
20:49.59modulus_on real bandwidth
20:50.02modulus_somewhere close
20:50.10tzangerI'm talking nominal jitter of 500us (half a millisecond) with peaks of maybe 5-10ms
20:50.31modulus_i offered to put a box on global crossing for them
20:50.32tzangermy ADSL line is nominally about 50ms of jitter with peaks up to 1500ms (1.5s)
20:50.37tzangernow
20:50.39tzangerhaving said that
20:50.39modulus_as a poc
20:50.49tzanger50ms of jitter is *NOTHING* in real-world
20:50.49Nuggetafter 2.5 months of bitching I still cannot get acceptable quality from my area code 512 DID with voicepulse.
20:50.54modulus_tzanger, you're talking about adsl
20:50.57Nuggetit's totally unusable.
20:51.13tzangerNugget: so cancel, have your credit card charges reversed and find someone else :-)
20:51.15modulus_i'm talking about gig-e capacities
20:51.24NuggetI can't find anyone else that offers local DIDs
20:51.27Nuggetany suggestions?
20:51.27harryvvI have checked and rechecked the settings between spa1000 sip.conf and extensions.conf reloaded and still getting a authenticaion message. What else needs to be done.
20:51.27tzangermodulus_: at gig-e you should have no issues unless you have a bad driver
20:51.30*** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz)
20:51.33*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
20:51.34tzangerNugget: iax.cc seems to have DIDs everywhere
20:51.38NuggetI'll try them.
20:51.47modulus_tzanger, nufone always times out still
20:51.49tzangerNugget: they're not perfect but they certainly seem acceptable
20:52.07tzangermodulus_: have you tried using switch-2 in lieu of switch-1.nufone.net?
20:52.13tzanger(it's in Michigan as opposed to Chicago)
20:52.19tzangercould just be a route path problem
20:52.20modulus_tzanger, yeah that's worse
20:52.25shido6who
20:52.27shido6whoa
20:52.31shido6nufone times out?
20:52.35PBXtechcan you wisper page a IP connection?
20:52.37tzangermodulus_: I'd open up a ticket with your provider
20:52.46modulus_global crossing?
20:52.47modulus_hahahaha
20:52.55modulus_i'm the network admin here
20:52.58tzangerI have been using nufone for the past year and any connectivity issues have been on my end
20:53.08modulus_i _promise_ you it's not global crossing's issue
20:53.12ionixbeside the huge long distance charge, nufone is ok
20:53.17tzangermodulus_: can you set up a packet capture at both ends of your network and see what they say when a timeout occurs?
20:53.37modulus_tzanger, i'd love to but jerjer didn't want to even talk to me
20:54.01tzangermodulus_: if you can get that data together and find jerjer I am willing to be he will help you by packet cap'ing at his end (I'm assuming this problem is easy to reproduce)
20:54.03modulus_i guess he's a networking guru as well as a nufone voip guru
20:54.17tzangerjerjer's very curt unless you have some solid data to prove the issue might be around his side of the ship
20:54.26modulus_i've shown traces
20:54.35modulus_where the last hop is his router interface
20:54.45tzangeryou have packet caps on his side of the router?
20:54.50modulus_where the ms times shoot up 150 ms
20:55.01PBXtechcan you wisper page a IP connection?
20:55.04modulus_every link before is ok
20:55.04ionixI had some issues with Nufone too
20:55.09ionixhis routing jumping to 150ms
20:55.09pdracevichcan some one have a look, it is an iax2 connection that was working perfectly, I think it has something to do with lag http://pastebin.ca/6346
20:55.10tzangerPBXtech: not sure
20:55.22GraphikosI'm can't get my IP phone to register... complete newbie here... can someone point out my stupidity?
20:55.40ionixHey, anyone ever used the Nortel Symposium system with asterisk ?
20:55.45ionixTrying to make it work with TAPI here.
20:55.54*** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net)
20:55.54tzangerpdracevich: get me a packet capture of that with -s0 and -w
20:56.14tzangerI'm not sure what "INVAL" means (invalid,yes, but invalid what, auth?)
20:56.20*** join/#asterisk pUmkInhEd (nobody@s142-179-184-59.ab.hsia.telus.net)
20:56.22pdracevichthat was a iax2 debug log
20:56.35tzangerpdracevich: I want a tcpdump or ethereal packet capture with -s0 and -w
20:56.52tzangerpdracevich: in other words, I want a copy of the actual bits on the wire :-)
20:57.08tzangerionix: odd
20:57.13pUmkInhEdhi, hopefully a quick q for you guys, does asterisk provide functionality for bridging two norstar systems?  I checked asteriskdocs.org and found a blank section in the current docs on this area
20:57.26tzangerpUmkInhEd: bridging two norstar systems?
20:57.34tzangerpUmkInhEd: explain, I have done some norstar/asterisk integration
20:57.46pUmkInhEdwell i currently have an FXO/FXS setup
20:57.59dstevens_Does somebody know how to fix a broken package, i do not understand any more.
20:58.07pUmkInhEdand it feeds into an ATA which we access as an outside line (ie line pool b)
20:58.38tzangerpUmkInhEd: eh?
20:58.40tzangerthat's not making sense
20:58.47tzangeryou have a norstar system with analog trunk lines, ok
20:59.09dstevens_{sorry wrong tab ignore previous}
20:59.11*** join/#asterisk Ayano (~erik_leee@209.143.187.254)
20:59.14pUmkInhEdtzanger: sorry if I am confusing
20:59.14tzangerit's feeding into asterisk over an FXS interface (TDM400 or channel bank + T1)
20:59.17tzangerright?
20:59.30pUmkInhEdtzanger: neg, i currently do not use asterisk software
20:59.35tzangerok
20:59.43PatrickDKtzanger, using like a sipura device he means
20:59.45Ayanowhat is the best way to achieve load balance and failover with *?
20:59.54pUmkInhEdPatrickDK exactly
20:59.57tzangerso you have a norstar system with analog trunk lines that plug into the telco jacks then?
21:00.49shido6ionix
21:00.52shido6what issues?
21:01.09pUmkInhEdyes but the analog jacks plug into another device which digitizes the sound, sends it to another unit connected to another norstar system
21:01.13*** join/#asterisk _Raptor_ (RaptorX@dsl-082-083-172-119.arcor-ip.net)
21:01.16_Raptor_hi
21:01.35tzangerpUmkInhEd: hmm what does that get you right now?
21:01.40pUmkInhEdthe other unit decodes and outputs the audio to the norstar
21:01.48tzangernormal way to join to norstar systems is to use a DTI with MCDN keys
21:01.48modulus_OMG
21:01.58pUmkInhEdtzanger I have a bridge between two norstars, if I press 9 I get local dial tone as expected
21:01.59modulus_Peer 'broadvoice' is now UNREACHABLE! Last qualify: 6
21:02.13pUmkInhEdif I press 8 I get dialtone from the other norstar
21:02.16modulus_WTF ARE THEY DOING?
21:02.21pUmkInhEdbut its feature set is somewhat limited
21:02.23tzangerpUmkInhEd: ok
21:02.40tzangerand your question with asterisk integration is what now
21:03.55tzangerMoc: holy disclaimer signature line, batman!
21:03.55harryvvI have a question about sip.conf and sip show peer. If for example a username and password is mismatched on either a spa1000 or sip.conf would this interfear with the * getting a Addr-> IP address ?
21:03.59tzangerquelle fromage!
21:04.27PatrickDKharryvv, hmm, yes, that is the point of username/password
21:04.32modulus_OMG BROADVOICE IS A PEICE OF SHIT
21:04.35modulus_fuckin' eh
21:04.44modulus_someone report them to bbb
21:04.53modulus_someone start a hate/fan site
21:05.02djinplease take that somewhere else
21:05.07lorionwhat is an alternative to Broadvoice?
21:05.08CoaxDmodulus: Um, you might want to reconsider that statement
21:05.19CoaxDmodulus: Already, you have lorion, here, who is actually stupid enough to listen to you
21:05.24harryvvPatrick, the user/pass are the same on sip.conf and the spa1000. doing  a sip show peer spa_01 I get a  Addr->IP     : (Unspecified) Port 0
21:05.25harryvv<PROTECTED>
21:05.28modulus_ok sorry lorion
21:05.37pUmkInhEdtzanger: my intention is to have a larger feature set than the ata can provide
21:05.45PatrickDKharryvv, do you have host=dynamic?
21:05.48modulus_lorion, they all kinda suck
21:05.57modulus_i haven't found 1 voip provider that i like yet
21:05.58CoaxDmodulus: Everybody bitches about non-regulated telco services
21:06.01modulus_and i've tried about 10
21:06.08CoaxDmodulus: er
21:06.15CoaxDmodulus: Everybody bitches about regulated telco services
21:06.21tzangerpUmkInhEd: ok
21:06.21harryvv:)
21:06.21CoaxDmodulus: They always cost 10x that of a non-regulated
21:06.46CoaxDmodulus: But you know what?
21:06.46tzangerpUmkInhEd: that's an awfully broad statement
21:06.46CoaxDmodulus:  Regulated telephone service is WORTH IT sometimes.
21:06.48modulus_yeah you're telling me
21:07.05tzangerCoaxD: I happen to agree with modulus_ ...  just from watching interaction with people with broadvoice
21:07.05modulus_even with all the 7 plus taxes per channel
21:07.12CoaxDtzanger: Heh :)
21:07.20*** join/#asterisk TrevMeister (~thammonds@ip68-4-223-70.oc.oc.cox.net)
21:07.20CoaxDtzanger: (So do I. But thats not the point.)
21:07.29tzanger:-p
21:07.44modulus_i've been on hold for over an hour
21:07.53WGFreewillbroadvoice has always worked great for me
21:07.57WGFreewilli just made a call
21:08.00WGFreewillworked great
21:08.40WGFreewill(its only been a few months that its been up)
21:08.52bjohnsonlorion: there are about 200 alternatives to broadvoice listed on the wiki
21:09.12shido6modulus_
21:09.16shido6u need to make a call?
21:09.33CoaxDthe thing is, broadvoice DOESNT suck.  And if you actually have a reaosn to call their technical support, you obviously didnt read what was on the website
21:09.41CoaxDwhen broadvoice goes DOWN, however, yeah, they SUCK
21:09.53CoaxDthe thing is, regulated telcos CANT go down.  Thats the whole point of a regulated telco.
21:09.55modulus_shido, i have a DID that's half down
21:10.02modulus_my asterisk box picks up for the DID
21:10.06modulus_but the caller hears nothing
21:10.12shido6ouch
21:10.14modulus_and i see my menu system exec playbacks()
21:10.15CoaxDmodulus: Sure sounds like a firewall proiblem to me
21:10.21modulus_omg
21:10.26WGFreewillor maybe some hung asterisk
21:10.32WGFreewillhave you restarted asterisk?
21:10.48modulus_haha coaxd and wgfreewill are so cute trying to help me
21:10.50modulus_but uh
21:10.54CoaxDmodulus: lameriferous
21:11.06pcmcoaxD: who's regulated telco ?
21:11.10modulus_nothing changed network-wise nor system-wise
21:11.15WGFreewillrboc
21:11.22modulus_incoming calls just stopped hearing anything
21:11.28Ayanowhat is the best way to achieve load balance and failover with *?
21:11.30CoaxDpcm: Any pstn telco in the united states is regulated. :)
21:11.44CoaxDpcm: voip, on the other hand, is not regulated
21:11.49WGFreewillAyano: thats an interesting question
21:11.55WGFreewillPSTN failover
21:11.57WGFreewillSIP failover
21:11.58CoaxDpcm; If I am a voip telco and i sell you an 800 number, that 800 number belongs to ME.
21:12.17CoaxDpcm: If my business goes away, so does your 800 number - unless I decide to transfer it to you
21:12.22CoaxDpcm: Same goes with any DID
21:12.34CoaxDpcm: regulated telco service?  That phone number belongs to YOU.
21:12.34pcmCoaxD: so any voip service can go down ... since they won't get any penalties for that ....
21:12.41modulus_jesus the last time i remember talking to broadvoice, they had to replace the whole LAX proxy
21:12.42CoaxDpcm: You got it
21:12.48AyanoWGFreewill: either.
21:12.50CoaxDpcm: It isnt about penalties, btw.  its about law
21:12.56modulus_they sure don't sound like they got a clue
21:13.02pcmCoaxD: law is penalties when you brake it
21:13.07CoaxDpcm: For sure
21:13.20bjohnsonhow can I tell what extension a call came in on?  I'm trying to get a voip call to come in on an extension that matches the DID called .. but I can't tell from the messages whether it is working or just coming in on the s,1 in that context
21:13.31CoaxDbjohnson: set verbose 5
21:13.47pcmCoaxD: but redundancy costs x 2 or even more ... :)
21:13.55CoaxDpcm: *far* more than x2
21:13.59plappyany ideas as to why I keep getting "Maximum retries exceeded on call..." after 5-10 seconds on local sip to sip calls?
21:14.01pUmkInhEdgah, must fix printer, bbiab, srry tzanger
21:14.04CoaxDpcm: (For GOOD redundancy)
21:14.17CoaxDplappy: set verbose 5
21:14.24pcmCoaxD: so how much redundant is the regualar telco ? 3 x ?
21:14.34CoaxDpcm: Well, keep this in mind:  Most telcos only have 1 switch
21:14.43CoaxDpcm: Although the switch can provide redundancy within itself, chances are, it doesn't
21:15.04CoaxDpcm: The redundancy in question needs to be things like. power, extra patch panels, wiring, etc
21:15.41lorionwhat would cause one asterisk extension from being able to dial another?
21:16.00Ayanoextensions.conf?
21:16.31PatrickDKcontext :)
21:16.37loriondo I have to state in extensions.conf whether extension1 can call extension2?
21:16.49bjohnsonlorion: yes
21:17.00bjohnsonnot directly though
21:17.00loriondo you have an example?
21:17.12bjohnsonlorion: thousands on the wiki
21:17.19bjohnson~docs
21:17.20jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
21:17.29bjohnsonstart on the extensions.conf page
21:17.54harryvvbj, I made some changes to make the auth simpler. Well that did not work.
21:17.55bjohnsoneach device has to be configured with an extension to ring that device
21:18.04bjohnsonusually done with a mcro
21:18.26bjohnsonlorion: try making some extensions that do not call another device
21:18.32bjohnsontry from all devices
21:18.39bjohnsonthen you know incoming is working
21:18.41lorionI can call VM
21:18.47bjohnsonok
21:18.52lorionI can cal my extension and it rings on line 2
21:19.00bjohnsonmake an exten definition for each of you devices
21:19.00AyanoOkay guys, I see your point about redundancy.  So how do you do load balancing?
21:19.04lorionwhen I call another extension i get 404
21:19.37harryvvbj, are you talking to me?
21:19.43bjohnsonharryvv: no
21:19.44Ayanolorion: make sure that both are authenticated too.
21:19.46WGFreewillkeepalived
21:19.46harryvvk
21:19.51WGFreewillvrrp
21:19.54pcmCoaxD: so you can realize redundancy using 2 or 3 voip providers
21:20.01WGFreewillbut its failover not active-active
21:20.02sungtards.
21:20.09WGFreewillfor inbound
21:20.38WGFreewillexten => s,1,ChanIsAvail(SIP/Provider1&SIP/Provider2)
21:20.42WGFreewillfor outbound
21:21.09bjohnsonWGFreewill: the superdial macro is a little more straght forward
21:21.18bjohnsonand easier to add/remove voip providers
21:21.25tzangerbjohnson: what are you collecting
21:21.27WGFreewillwhat is superdial macro
21:21.33tzangerbjohnson: and are you going to torastricon on friday?
21:21.47Nuggethttp://www.google.com/search?q=asterisk+superdial+macro
21:21.49bjohnsonfor inbound I've been asking voip providers to forward to my telco pstn if my * is not answering them
21:22.05bjohnsontzanger: I've been collecting voip provider accounts
21:22.11bjohnsonnew hobby I guess
21:22.27Nuggetthe superdial macro ought to do enumlookup.
21:22.28bjohnsontzanger: I'm not going
21:22.31WGFreewill(Miami VON, on now)
21:22.46bjohnsonNugget: the enumlookup is done and then fed to superdial
21:22.54_Raptor_cu
21:23.00Nuggetyou could do it that way, yes.
21:23.09tzangerbjohnson: I'll trade you a VPC for a livevoip
21:23.09tzangerheh
21:23.18Nuggetbut without enum it's not as "super" as it could be.  :)
21:23.20bjohnsonexten => s,4,EnumLookup(${tfnumber})
21:23.20bjohnsonexten => s,5,GotoIf($[${ENUM:0:3} = IAX]?6:7)   ; SIP behind NAT not working
21:23.20bjohnsonexten => s,6,Macro(superdial,${ENUM},,,,voip,${MAXVOIPCALLS},Johnson Engineering Consultants,519-271-9923,e164)
21:23.40bjohnsonthe absolute nicest thing about superdial ...
21:23.48bjohnsonis that it returns to the context that called it
21:24.00bjohnson(if the call is not connected)
21:24.03Nuggetthat's spiffy.
21:24.21NuggetI don't use superdial, but it doesn't look very different from what I built myself, in ignorance of superdial's existence.
21:24.21bjohnsonrather than a speghetti string of goto's all over the place
21:24.35bjohnsonit's based off of examples on the wiki
21:24.39modulus_bjohnson that's almost as nice as having one of those reliable broadvoice DIDs
21:24.43Nugget${DIALSTATUS} is tasty.
21:24.47*** join/#asterisk Alejandriax26 (~nurbina23@proxy.more.cl)
21:25.07*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
21:25.12*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
21:25.14bjohnsonso to try another channel out ..
21:25.19bjohnsonexten => s,7,Macro(superdial,IAX2/iaxnboom/1519${tfnumber},,,,voip,${MAXVOIPCALLS},Johnson Engineering Consultants,519-271-9923,nboom)
21:25.23WGFreewillthe dialparties.agi
21:25.25WGFreewillin the AMP package
21:25.29WGFreewillis the superdial marco
21:25.31WGFreewillto the next level
21:25.38*** part/#asterisk XeNoSiS (user@216.234.145.18)
21:25.44*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
21:25.50mishehuwhat is AMP?  asterisk made painful ?
21:25.51mishehuheh
21:25.54ariel_argh people will never learn.  Get this a person wants to get a sipura working via not one but 3 NATs over a sat link and do faxing. ARgh
21:25.58tzangermishehu: actually that's not far from the truth :-)
21:25.59WGFreewillasterisk more pretty
21:26.15roamer323whatz the detail on the torastrcon?
21:26.23tzangerroamer323: opensource.meetup.com/42
21:26.29loudariel_, wait until they ask for 4 simultaneous calls through sat link.
21:26.38bjohnsonariel_: faxing is tricky .. but I have mine working through 3 nats
21:26.40WGFreewillif your not scared to go edit a few .conf files, it works great, but its lacking a few pieces
21:26.44ariel_bjohnson, I dont' use amp for there conf I use them for the reports.
21:26.47bjohnsonariel_: (not with fax though)
21:26.48roamer323thx tzanger
21:27.07[Outcast]does anyone have some good res_perl examples?
21:27.25ariel_bjohnson, I have already gotten voice calls working for them. Now they want faxing too. ARgh this is not going to work.
21:27.45tzangerariel_: use some app_rxfax/txfax and scp magic
21:28.02ariel_bjohnson, actually what I do is install asterisk@home and change the conf files to my use.
21:28.05modulus_broadvoice could use a little "magic"
21:28.07loudariel_, which sat platform, gilat, ipsat ?
21:28.56WGFreewillyep I use amp and @home pieces
21:28.56bjohnsonariel_: vpn?
21:28.56ariel_loud, don't know trying to find out now.
21:29.09WGFreewillfaxing needs 0% packet loss g711, and latency isnt going to help any
21:29.21modulus_i think broadvoice just power cycled their lax proxy
21:29.32ariel_WGFreewill, yes sir your correct.
21:29.36modulus_but the calls coming in still can't hear anything
21:29.40WGFreewillrxfax at the edge
21:29.49loudbroadvoice service last night like at 11 pm pst was awful.
21:30.02ariel_tzanger, spandsp is not working for it either.
21:30.09modulus_loud, it hasn't worked for me since then
21:30.14tzangerariel_: damn that's not good then
21:30.23loudit works now, althought i have one way audio though
21:30.24tzangerperhaps a pair of t.38-aware ATAs
21:30.58[Outcast]modulus_: what do you have your externip set to in your general section?
21:31.10modulus_the public ip address
21:31.14fearnorwg: even with 0% packet loss, 1ms latency, 1ms jitter, you aren't guaranteed 100% working fax ;)
21:31.17modulus_of their lax server
21:31.29modulus_x.x.8.x
21:31.35fearnorit'll be 90% or so.
21:31.50bjohnsonariel_: what about just digitizing faxes locally, emailing them as an attachment elsewhere, and reprocessing them?
21:31.54fearnori had moderate luck with fax over TDMoE though.
21:31.58[Outcast]are you be hind nat?
21:32.03tzangerfearnor: me too
21:32.05modulus_ooh broadvoice is now UNREACHABLE again!
21:32.05fearnorbjohnson: bad for the end-to-end.
21:32.11tzangeroh tdmoe
21:32.13tzangersorry I haven't tried that
21:32.17modulus_no outcast
21:32.31[Outcast]modulus_ : turn qualify off for broadvoice, just causes problems
21:32.32fearnorbj: imagine if phone number is busy etc, you need to implement redelivery logic etc
21:32.43Blackvelwho has working FWD with asterisk behind NAT? does that work?
21:32.43[Outcast]modulus are you using srvlookup?
21:32.48ariel_bjohnson, it's a customer that wants to be able to use a fax machine in his office.  There not too up on the computer stuff there yet.
21:32.56loudmodulus_, do they support g729 ? bv i mean
21:33.08modulus_loud, i kinda doubt it
21:33.09[Outcast]Blackvel: i do you need to but in the DMZ
21:33.13*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
21:33.19modulus_loud, they can't even support sip
21:33.19shmaltzhi everybody
21:33.20[Outcast]loud: no they do not
21:33.23loudso just g711 :/
21:33.32Blackvelloud: i think g711
21:33.33modulus_ulaw
21:33.34[Outcast]loud: yes
21:33.42loudi think i screwed it up buying an international plus blah plan
21:33.43WGFreewillwish efax was cheaper
21:33.45Blackvelmodules: they do not support SIP? hehehhe....are you sure?
21:33.48modulus_registrations are timing out
21:33.55modulus_blackvel, they try to
21:33.57EssobiUmm. I've got a problem.. My 5400's send SIP calls to my * box with a random port.. and specifingy port= or port=0 on my peers list won't match them.. anyone know how to rectify that?
21:34.00modulus_but they fail
21:34.12shmaltzdoes anybody know if it is possible (and how) to show the status of anohter extension (SIP device) on the Cisco 7960 or Cisco 7914?
21:34.25loudyes
21:34.26modulus_wow their lax proxy is completely down now
21:34.27modulus_neat
21:34.33[Outcast]modulus_: do you want my config
21:34.36Blackvelmodulus_: my sip show registry shows: "Registered". but i am unable to call echo
21:34.38Beirdomikegrb: http://homokaasu.org/rasterbator/galleryimage.gas?2232
21:34.39modulus_maybe they got a tech there punching the copper back down
21:34.51Beirdowhat some people will put on their walls
21:34.59loudshmaltz: show reg through CLI, on the 7960
21:35.03modulus_outcast, thank you for your samaritan antics but i know it's nothing on my end.
21:35.15modulus_outcast, nothing has changed on my end for over a week
21:35.20modulus_and last night it stopped working
21:35.27tzangerand he's sitting on gige, the bastard
21:35.35modulus_yes i'm a bastard
21:35.42modulus_actually i have 3 gig-e uplinks
21:35.42WGFreewillmodulus_: maybe the tech set his crack pipe on the copper your DID comes in on
21:35.44tzangersee, he doesn't know who his father is
21:35.48modulus_global crossing/level3/cogent
21:35.51shmaltzloud, this will show on a Cisco 7960 the status of another cisco 7960? using one of the line buttons?
21:36.01[Outcast]modulus_: i have heard that many time when i did support for them, i sure it just a simple thing.
21:36.03tzangermodulus_: you have a link to cogent, switch-2 should be amazingly close to you
21:36.10modulus_omg lax sip proxy is down HARD for broadvoice
21:36.15tzangercogent -- the Wal-Mart of bandwidth providers
21:36.18loudno, this will show the reg status of an extension on another voip gateway.
21:36.31modulus_tzanger, but for the other voip providers, using cogent would suck
21:36.50tzangermodulus_: true, but you said you had pauses going to switch-2 too (actually you said they were worse)
21:37.07shmaltzloud, somthing like the hint priority
21:37.12tzangerwhat's a traceroute to switch-2.nufone.net look like to you
21:38.15shmaltzis there anyway to show on a sip phone that a call is parked?
21:38.17modulus_they both get around 60 ms pings
21:38.23modulus_but they both spike and lagg out
21:38.25modulus_a lot!
21:38.39*** join/#asterisk kuonSama (kuon@alragore.goyman.com)
21:38.44kuonSamahello everybody
21:38.54Blackvelhmm with 1.0.5 DISA only accepts one dial in, but not the extensions from my analog pbx behind the inward number (no extension can be executed)
21:38.59WGFreewillmodulus: try another BV proxy
21:39.03*** join/#asterisk zapa (zapa@200.92.147.148)
21:39.12Blackvelwhat is going on? is that a problem of DISA or bristuff?
21:39.13tzangermodulus_: I said traceroute not ping, I was just curious as to the path
21:39.16modulus_wgfreewill, the only one that authenticates my sipuser seems to be the lax one
21:39.25terrapenhttp://www.foxnews.com/images/154561/0_25_jackson_michael_legal.jpg
21:39.31terrapenthat spot on his chin
21:39.38terrapeni think that's where his soul drained out
21:39.59*** join/#asterisk Umaro (~umaro@209.140.74.64)
21:40.03modulus_tzanger, gblx->alter.net->garage-webhosting-company-> nufone.net
21:40.09WGFreewillhmm i dont seem to have provlems using chicago or lax
21:40.20WGFreewilli am on chicago now
21:40.37modulus_when i use other sip proxys, authentication fails
21:40.44modulus_lax is the only one that seems to work for me
21:40.48UmaroHey guys, anyone setup their spa-2000 with broadvoice manually? I have asterisk connecting to broadvoice, but I want to try configuring my sipura directly to broadvoice to see if the quality problems I am having are related to asterisk or just broadvoice
21:41.02loudexactly.
21:41.03Umaroif I auto config it with broadvoice, they'll lock me out of my own sipura
21:41.08modulus_umaro, it's broadvoice
21:41.17bjohnsonUmaro: you just missed a gang rape on bv
21:41.21*** join/#asterisk Rith (nobody@66.142.28.35)
21:41.23modulus_i'd bet my 2nd nutsack on it
21:41.27Umarobjohnson: oic
21:41.28terrapenhaha
21:41.28tzangergarage-webhosting-company?
21:41.45modulus_tzanger, did you look at who nufone hosts with?
21:41.54bjohnsonit's better use the the crap I throw into my garage
21:41.55tzangermodulus_: that's through global crossing, I wanted to see what your path to switch-2 was like through cogent
21:42.02modulus_hold up
21:42.30tzangermodulus_: switch-2 is through 123.net I think
21:42.41tzangerbut switch-1's through scnet
21:43.10tzangerI'm 36 and 25ms to him (switch-2,switch-1) and I'm in Canada through Ikano
21:43.41*** join/#asterisk heison (~heison@ns.somanetworks.com)
21:43.42*** join/#asterisk bsenicar (~bsenicar@BSN-77-155-238.dsl.siol.net)
21:44.00zapahi all, i have e1 with digial trunk to pstn, i have 10 DID, is there any way to change
21:44.00zapathe caller id to specific one, when i make call to pst.
21:44.41marc_cikano??
21:44.43[Outcast]There is nothing wrong with BV as long as you configure your box rightg
21:44.43tzangerzapa: if your provider lets you set CID on outgoing calls, sure
21:44.46tzangermarc_c: yeah
21:45.17modulus_tzanger, cogent->switch-2
21:45.19marc_cT1's?
21:45.20Blackvelwhats the best flat provider? broadvoice, nufone, something else?
21:45.24modulus_they use cogent as an upstream
21:45.26tzangermodulus_: so a few hops in cogent and that's it?
21:45.33modulus_few?
21:45.33Umaro[Outcast]: yes.. their hold time sucks though
21:45.34tzangerand you still have shitty connectivity to them??
21:45.35bjohnsonBlackvel: why flat?
21:45.36modulus_more like GANG
21:45.36loudnufone = flat ? international ?
21:45.48modulus_14 hops to switch-2 only going through cogent
21:45.57modulus_cogent needs help
21:45.59PBXtechcan asterisk do whisper paging?
21:46.06Blackvelmy german providers provides a german flat
21:46.09Blackvelyou pay 20EUR
21:46.11[Outcast]Umaro: that is because it don't work there any more
21:46.16Blackveland you can call germans for free
21:46.26bjohnsonBlackvel: no .. you call them for 20EUR
21:46.37bjohnsonis that per month?
21:46.47Blackvelbut i think there are also providers offering for 20$ germany,uk,usa,etc?
21:46.50RithCan anyone point me in the right direction for what equipment I'd need (and what costs I'd run, USD) to set up a small office w/ a few phones and voicemail boxes. the docs on the asterisk.org leave me with a lot of questions
21:46.50Blackvelyeah per month
21:46.59tzangermodulus_: I have 4 hops in cogent, 4 in peer1 and 2 in ikano
21:47.01Blackvelits great if your call volume is always 30EUR and above
21:47.11modulus_12  Internet123.demarc.cogentco.com (66.250.4.86)  68.221 ms  68.642 ms  68.254 ms
21:47.11modulus_13  vl119.lodden.sfld-mi.123.net (216.234.104.114)  68.023 ms  68.077 ms  67.555 ms
21:47.11modulus_14  198.22.67.70 (198.22.67.70)  70.605 ms  68.745 ms  69.363 ms
21:47.27modulus_hop 13-14 used to show 200ms increase
21:47.27tzangerRith: your post leaves us with just as many.
21:47.29heisonanyone here got a Canadian toll free number ?
21:47.30modulus_a month ago
21:47.38tzangerheison: yes
21:47.42tzangermodulus_: hmm
21:47.46zapatzanger: where i change CID ? in zapata.conf?
21:48.01tzangerzapa: there (on a channel basis) or using SetCIDNum() in the dialplan
21:48.26Rithtzanger: understandably. would like to take POTS incoming lines and run them to VoIP phones w/ voicemail and such... mostly confused as to what equipment I'd need
21:48.44shmaltzdoes anybody here use a Cisco 79xx?
21:48.45bjohnsonother than a server you don't need any hardware
21:48.57tzangerRith: what VOIP phones are you interested in
21:48.57bjohnsonyou may "choose" to add hardware
21:49.07tzangerbjohnson: he will need FXO interfaces
21:49.09ionixHey, anyone has ANY IDEA on how I can fill in a name when I have a phone number ? Trying to find a way to access the RBOC database
21:49.14ionixor query it at least
21:49.21bjohnsontzanger: not with voip pstn DIDs
21:49.23Rithdon't know yet, that's part of what i'm trying to determine
21:49.38shmaltzionix, use a script to query anywho reversi lookup
21:49.43tzangerhe said "POTS incoming lines" -- that indicates the need for FXO interfaces :-)
21:50.07tzangerRith: well you don't need any hardware aside from the netwroking infrastructure to connect SIP phones with asterisk
21:50.07bjohnsonhe likely doesn't know an alternative exists
21:50.12tzangeryou want decent network infrastructure though
21:51.32bjohnsonRith: there isn't a magic formula .. you likely want a fxo for each pstn line you want and a voip phone or a fxs+analog phone for each phone you want
21:51.37heison~seen sivana
21:51.40jbotsivana is currently on #asterisk.  Has said a total of 6 messages.  Is idling for 6h 11m 46s
21:52.00bjohnsonRith: however, that is not required .. just what I think you will want
21:52.22dsmouse~seen jbot
21:52.23jbotjbot is currently on #ipaq (16h 33m 55s) #how (16h 33m 55s) #bzleague (16h 33m 55s) #storm (16h 33m 55s) #orkut (16h 33m 55s) #asterisk-doc (16h 33m 55s) #uphpu (16h 33m 55s) #va (16h 33m 55s) #asterisk (16h 33m 55s) #nslu2-linux (16h 33m 55s) #magnia (16h 33m 55s) #aegis (16h 33m 55s) #ol ...
21:52.41tzangerno wonder he's so slow
21:53.24shmaltzanybody using a cisco 79xx?
21:53.35tzangershmaltz: not I
21:53.41Rithso just a few FXO cards on incoming POTS and that's it for specialized hardware?
21:53.50shmaltztzanger, realy?
21:54.01tzangerI don't do SIP
21:54.12shmaltztzanger, so what do you do?
21:54.13tzangerRith: for what you just described, yes
21:54.19tzangershmaltz: IAX2 all the way
21:54.22Rithalright, thanks
21:54.28tzangerI have a TDM430P at home for 3 analog phones
21:54.36shmaltztzanger,what hard phones support IAX2?
21:54.43tzangershmaltz: farfon does if you can find one
21:54.55tzangershmaltz: there are a few others too based on some weird chipset but I've no experience with them
21:55.04tzangerPA1688 or something like that
21:55.22shmaltzwhy souldn't i be able to fine one farfon? tzanger.
21:55.34tzangershmaltz: because wasim's still working on it
21:55.34bjohnsonnot available to the public yet
21:55.46tzangercitats is supposed to be the north american contact but he's been gone for a dog's age
21:55.49tzanger~seen citats
21:55.51jbotcitats <~james@duff.gnuinter.net> was last seen on IRC in channel #asterisk, 72d 22h 24m 47s ago, saying: 'and i gotta go back, so i'll catch ya'll later'.
21:55.58bjohnsonhaha
21:56.00bjohnsonmuch later
21:56.04tzangerok
21:56.06tzanger10 dog's ages
21:56.09shmaltzso tzanger, meanwhile I can't talk to you unless i'm your girlfriend or using IRC?
21:56.24tzangershmaltz: I just said I use a TDM430P and regular old analog phones
21:56.26bjohnsontzanger: <- needs to teach jbot how to count in dog ages
21:56.46bjohnsonshmaltz: he has * server at home
21:56.47tzangera panasonic cordless, a cheapo but pretty one in my bedroom and a PT450 in the kitchen
21:57.20bjohnsonshmaltz: and a PCI card with fxs ports
21:57.24Juggieanyone know of a good call back script? like if the pri is full, on the next chan, call me then give me that channel, etc... or if someone is busy with no call waiting/vm then offer callback?
21:57.40shmaltzI understand but if he doens't use sip and only iax2 and iax is not available he has no phones (he didn't mention analog)
21:57.41tzangerJuggie: huh?
21:57.49tzangershmaltz: yes I did
21:57.57Juggieeg, i make a call, the pri is full...
21:57.58shmaltzok, giving up
21:58.06plappyokay, this makes no sense at all... Both of my clients can use the demo and dont have any issues, client to client through * gives me the "maximum retries exceeded..." error after a few seconds.  is it possible its just not solid on FBSD yet?
21:58.13Juggiewhen theres a free channel, asterisk calls me back, then calls the number from before
21:58.13tzanger17:02 < tzanger> shmaltz: I just said I use a TDM430P and regular old analog phones
21:58.27tzangerplappy: yes, very possible
21:58.33Juggieits one of our requirements here for work... i'm evaluating * against our RFP.
21:58.38bjohnsonshmaltz: anyway .. people are using cisco 79xx phones
21:58.42plappybugga me.
21:58.42plappyheh
21:58.45bjohnsonbut not tzanger
21:58.53tzangerbjohnson: precisely
21:59.00bjohnsonand not me :(
21:59.16shmaltzI know bjohonson, and I'm trying to figure out what is the best way using the multiple lines, using multiple registrations? or just one registration?
21:59.23tzangerJuggie: I don't have a script handy but it shouldn't be too difficult
21:59.26tzangerasterisk has all the info
21:59.41Juggieyeah, but how to loop until theres a channel free
21:59.45Juggiethats the issue...
21:59.45tzangershmaltz: bkw_ might have the low-down, he's a know cisco sympathizer.  :-)
21:59.51tzangerJuggie: you don't
21:59.55tzangeryou put it in a 5min cron job
21:59.59tzangerso it just checks every 5 min
21:59.59shmaltzbkw_, you around?
22:00.00WGFreewillshmaltz
22:00.07WGFreewilllike the cisco multiple lines
22:00.08tzangerI do that with my voicemail callback
22:00.09tzangers
22:00.17Juggietzanger, the requirement is instant call back when the line is free
22:00.17tzangersince the callback as it's currently coded sucks
22:00.23shmaltzWGFeewill, yes
22:00.30tzangerJuggie: IIRC there is also a way to loop inside of asterisk
22:00.33tzangerin the dialplan
22:00.37*** part/#asterisk ReVoK (ReVoK@82.224.60.46)
22:00.38WGFreewillif you are using the AMP packages
22:00.41WGFreewilldialparties.agi
22:00.41tzangeranyway
22:00.45tzangerI gotta go get the kids
22:00.46tzangerttyl
22:00.50roamer323anyone here from the tmc miami show?
22:00.51shmaltzWGFreewill, I'm not.
22:01.03WGFreewilleh good to have a look
22:01.09WGFreewillyou can register all cisco lines
22:01.12harryvvanyone here running a spa1000
22:01.15WGFreewillto one sip entry
22:01.41WGFreewillchange the 0 in that agi
22:01.42WGFreewill<PROTECTED>
22:01.57WGFreewillto the number of lines you want to support system wide (unfortunately)
22:02.08Mavviesometimes, when I call overseas, it looks like asterisk doesn't bridge two channels until I have really spoken with a loud voice. (this bridge is on two FXO cards)
22:02.12shmaltzI know, but I'm trying to figure out if this is realy the best way. Currently I'm using six different sip accounts, it is better b/c it allows one to pick up any line to make a call.
22:02.15Mavvieis that normal?
22:02.19WGFreewilland your snom and cisco phones will ring through each available line
22:02.33WGFreewillsame with this
22:02.59WGFreewillMWI only comes on the first line
22:03.33WGFreewillone extension per phone
22:03.38WGFreewillone user / pass
22:03.45WGFreewillbut maybe there is a better way
22:04.08*** join/#asterisk file (~file@mctn1-8179.nb.aliant.net)
22:04.24WGFreewillif there is maybe we'll hear about it right now
22:05.58shmaltzThanks, WGFreewill, I just tested it and it makes much more sense.
22:06.16*** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar)
22:06.36*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
22:06.57shmaltzWOW:
22:06.58shmaltzhttp://www.surfsantamonica.com/ssm_site/the_lookout/news/News-2005/Feb-2005/02_23_05_Cell_Phone_Virus_Strikes_Santa_Monica.htm
22:07.08*** join/#asterisk bsenicar (~bsenicar@BSN-77-155-238.dsl.siol.net)
22:08.40shmaltzJust wondering when the first Cisco phone virus will enter the internet
22:08.49Blackvelwhat isn't DISA accepting in asterisk 1.0.5 anymore the numbers after the dailin? that worked before with bristuff and asterisk 1.0.2? my analog pbx does not support blockmode, so I have to use DISA to send extensions behind the inward-number into asterisk
22:08.51shmaltzor maybe Polycom
22:08.58terrapencisco phone virus, heh
22:09.53WGFreewillpeople are going to realize one day that talking on the phone sucks anyways
22:10.03filetelepathy, that's where it's going
22:10.03WGFreewillthe cisco virus will help us get there
22:10.12terrapenmy girlfriend will never realize that.
22:10.42Rithok, coming in from pots lines, we can use a wildcards + FX0 bundle into the PBX. coming out to VoIP phones, we just plug straight into NICs? and if so, can we put them through a switch for more ports, or do they need to go straight into each NIC port?
22:11.05WGFreewillthe new voice communication platform will be ale at the pub
22:11.06*** part/#asterisk gr0mit (~gr0mit@router1.txrx.org.uk)
22:11.13WGFreewillRith: no switch the ethernet is fine
22:11.19terrapeni love that i can set my cisco handset off the hook
22:11.20Mavviecalltone detection is really a POS when you try it outside your progzone.
22:11.22terrapenand the phone stays on the hook
22:11.23WGFreewillRith: just like always
22:11.29Rithalright, great, thanks
22:11.32terrapeni just wish there was a little button on the handset to pick up the hook
22:12.34terrapenneed to figure out how you set the 7960 to do not disturb
22:14.06Nuggetunplug it.  :)
22:14.27heison~seen Jerjer
22:14.31jbotjerjer <~JerJer@dsl-107-24.che.centurytel.net> was last seen on IRC in channel #asterisk, 9h 41m 6s ago, saying: 'don't run 1.0.2 or bristuff'.
22:14.52heison~seen shido6
22:14.54jbotshido6 is currently on #asterisk (3h 7m 8s).  Has said a total of 15 messages.  Is idling for 1h 4m 42s
22:15.34terrapenAsterisk CVS-v1-0-01/10/05-03:06:46
22:15.37terrapenis that way old?
22:15.55Nugget10-Jan-2005 was about a month and 10 days ago.
22:16.01Nuggetwas that a trick question?
22:16.04terrapen:P
22:16.17terrapenthis day has turned out bad ass
22:16.21terrapeni should go outside and enjoy this
22:16.25terrapenim sick of being in this office
22:16.31MavvieI don't want to do FSX/FXO anymore....
22:17.56Juggieanyone have an example of a looping dialplan? one where if all the channels where busy on zap, i could call the user back and connect their call when one became available.
22:18.09terrapenmuch less stuck here coding a fucking shopping cart
22:18.37harryvvman... this is truely wierd. I have not been able to configure my ata because of a sinfigure mismatch and now after not touching it for 30 min it works? i have all valid info in sip show peer and have a dialtone.
22:18.52harryvvI was ready to pull my hair out.
22:18.53*** join/#asterisk jero (~SFLphone@199.243.85.90)
22:18.58jerohello
22:18.59terrapenharry, its the gypsies
22:19.56harryvvterrapen, I dont know but I even simplified the authentication and that did not work. Gave up and did more searching on google to find out what was going on and subconciosly picked up the reciver.
22:22.04harryvvSo what can ya say ;)
22:25.33*** join/#asterisk xeet2 (~xeet2@es.jsci.net)
22:26.43EssobiAnyone know why I can't have a dynamic src port on a sip peer calling into me?
22:27.17shmaltzWGFreewill, I'm getting 403s in the log if I do it this way, what Can I do to avoid this?
22:27.35EssobiI got a cisco 5400 that's picking random ports and they won't land in a peer context, only a default context.
22:27.53Trionniswell there's your first problem
22:27.55Essobidue to the ports src ports changing every call
22:27.56Trionnisyou're using a Cisco
22:28.08EssobiUhh. Yea.
22:28.25TrionnisI'm just yankin' yer chain man ;)
22:28.30EssobiI worked fine in 1.0 but in -head it's broke.
22:28.30WGFreewilldoing it using the dialparties.agi from the AMP package?
22:28.34Trionniswe've all had to put up with 'em
22:28.37Trionnis:)
22:29.10EssobiIt's agravating me.
22:29.30xeet2essobi:  what protocol?
22:29.59harryvvEssobi well persistance pays off eventually. I am a bit relieved a little happier this problem this problem of mine is over.
22:31.08*** part/#asterisk Grooby (~Grooby@12.22.232.212)
22:31.35Trionnisyou could diff the sources for the sip stuff, and find out what was changed I suppose
22:31.39Trionnisthat sounds like a PITA tho
22:33.36*** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com)
22:37.57jalsothi
22:38.32jalsotdoes anybody use HFC-PCI ISDN card in NT mode here?
22:38.38km-is there an application Dialtone()?
22:39.13km-...
22:40.30BrianR___Is there anything in the zaptel package I might use to figure out why I'm not getting caller id from my X100P?
22:41.18km-hmm
22:41.20km-I dont think so
22:41.26tclarkclidtest
22:41.38BrianR___clidtest?
22:41.49*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
22:41.51BrianR___Which zaptel version has this program?
22:41.54xeet2does the x100p have to wait for 2 rings to receive all the callerid info?  or can it answer right away?
22:42.10BrianR___xeet2: No idea. Asterisk is having it wait two rings.
22:42.15tclarkhmm looks like from the digium ftp site
22:42.41BrianR___The PBX I have it connected to actually sends the full callerid in two bursts before the first ring, then again .5 seconds after each ring.
22:42.46tclarkhell is just a few line i could put it in pastebin.ca if ya want
22:42.55BrianR___tclark: Ok.
22:43.30BrianR___Aah. Found it.  ftp://ftp.linux-support.net/pub/zaptel/clidtest.tgz
22:43.44BrianR___hmm.. No.. It's not there anymore.. :(
22:43.54|Vulture|What is a good LD rate on a PRI?
22:44.03tclarkhttp://pastebin.ca/6348
22:44.07|Vulture|all the places keep quoting me ~5cents
22:44.13shido6how do u make a single module instead of make modules?
22:44.56Essobithis is pissing me off
22:44.57xeet2vulture:  if you're looking at a pri for just long distance, you would be better off getting a t1 to a good isp and using voip
22:45.08Essobino matter what I do.. my sip peer lands in [default]
22:45.13|Vulture|xeet2: oky
22:46.10WGFreewillEssobi
22:46.18WGFreewillyou need to match
22:46.20WGFreewillIP
22:46.21WGFreewilland port
22:46.26WGFreewillcheck sip debugs
22:47.51xeet2vulture:  I only ever use a pri or a voice t1 from a lec for local calls and inbound
22:47.52BrianR___clidtest exits right away with "Error getting Caller*ID..."
22:47.55*** join/#asterisk zno (~zeno@ip-160-79-174-101.autorev.intellispace.net)
22:48.06shmaltzWGFreewill, I'm getting my logs full with "Registration From ..... Failed for ....
22:48.45tclarkthink you are supposed to run it w/o * up
22:49.16BrianR___same deal
22:49.25BrianR___as soon as the line rings it exits...
22:51.20|Vulture|xeet2: oky thanx
22:51.36|Vulture|xeet2: on a PRI you can get DIDs in different areacodes other than yours.. right?
22:52.30xeet2vulture: yes, depending on your provider
22:53.06|Vulture|xeet2: oky thanx
22:53.12km-man this problem just evades capture all the time
22:53.14xeet2surprisingly intermedia/allegiance/whatever other name they want  seem to be pretty good about that in the US
22:53.33xeet2we're in baltimore and we have dids from every major city in the us, and local calling to them too
22:53.44|Vulture|Any comments about Xspedius?
22:54.27|Vulture|xeet2: how much do you pay for a PRI with multiple local calling areas?
22:55.06xeet2we pay a set fee for each lata we have access too, I don't remember exactly what that fee is
22:56.32xeet2and for anything else outside of those areas we send the calls to a number of sip and iax providers
22:56.39|Vulture|xeet2: which provider do you use?
22:57.15xeet2vulture: for what?  voip?  quite a few, no single one meets all our requirements in all areas
22:57.16km-ah
22:57.20km-so that what prewink does
22:58.09tzangerkm-: what's it do
22:58.22km-tzanger: if I kick everything up to 3000
22:58.32km-tzanger: the t1 takes 3 seconds to realize the phone is off hook
22:58.42km-doesn't change anything about how long I have to hit digits
22:59.00km-i.e., as soon as I hit 1
22:59.05km-it's there
22:59.16xeet2hmmmm
23:00.50tzangerkm-:
23:00.51tzanger;    emdigitwait: Time to wait for DID digits on E&M links (default 250ms) (Increase to 500
23:00.55tzanger;                 or so if you are not getting all DID digits on your E&M link)
23:01.03tzangerset that to 3000
23:01.19tzangerand set it back to 250 on the telco-facing T1
23:01.44km-damn people trying to call at 6oclock
23:01.44km-I did
23:01.47BrianR___hmm.. this is going to be difficult to debug.. a google search for my error finds only one other person and they didn't get it resolved either :(
23:01.56km-emdigitwait?
23:01.58km-I didnt see that
23:02.33km-no change
23:02.38tzanger...??
23:02.43tzangerthat seems odd
23:02.49tzangerI would have thought that did it
23:02.52km-emdigitwait=3000 does not change the digit wait
23:02.56tzangerthat was in CVS HEAD, btw, not sure if that's in stable
23:02.57km-is that a feature only in cvs?
23:03.01km-I downloaded 1.0.5
23:03.11km-eh, lemme pull cvs
23:03.30km-because I obviously love running HEAD in a full-on production environment
23:03.34km-hehee
23:03.45tzangerkm-: look around line 156 in channels/chan_zap.c and see if it's there
23:04.05km-just a min, letting apt-get finish installing \cvs
23:04.11tzangerkm-: -HEAD is often perfectly stable
23:04.17km-I know
23:04.21km-I used to run HEAD before
23:04.25km-but it's just the idea of it
23:04.32tzangerkm-: pish tosh
23:04.33km-I'm forced to think like a narrowsighted IT admin :)
23:05.06km-line 156 of chan_zap.c is a #define
23:05.15tzangerlook around there
23:05.17km-#define DCHAN_UP          (1 << 2)
23:05.17km-#define DCHAN_AVAILABLE (DCHAN_PROVISIONED | DCHAN_NOTINALARM | DCHAN_UP)
23:05.17km-#define zt_close(fd) if(fd > 0) close(fd);
23:05.18tzangeror just search for emwinktime
23:05.45km-negative on emwinktime, emdigittime
23:05.50tzangergood
23:06.41tzangerI think extensions.conf sould look like httpd.conf
23:06.48tzangeror hell everything like that
23:06.51tzangerI really like that setup
23:07.02tzangereasily machine-parsed too
23:07.26terrapenonce you get a hang of extensions.conf, its not hard at all
23:07.35tzangerterrapen: i agree
23:07.40tzangerI'm just saying I like how apache.conf is
23:07.44tzangerer httpd.conf
23:07.44BeirdoI think it should all be in MySQL/Postgres :)
23:07.45Beirdohehe
23:07.49terrapenoh lord no
23:07.51tzangerBeirdo: that too
23:07.55tzangerbt that's a little overkill
23:08.04Beirdomeh
23:08.05terrapenand when i was starting with asterisk, i thought it should be in Apache config format too
23:08.06tzangerso long as the DB is allowed to cache results
23:08.09terrapenbut now that i understand it
23:08.11tzangerand then everything is in realtime
23:08.13terrapeni like it just the way it is
23:08.17tzangerif you need to update several things, use a transaction
23:08.20NuggetI like it just the way it is.
23:08.22Beirdoit works good the way it is
23:08.29terrapenfuck databases
23:08.30Nuggetextensions.conf is a scripting language, not a data store.
23:08.32Beirdobut it would be nice in SQL :)
23:08.37km-its just like the old tired debate about rewriting asterisk in C++
23:08.39terrapenyou should not need a DB to configure a PBX
23:08.40km-it works fine the way it is
23:08.45km-no point in reinventing the wheel
23:08.49Beirdoterrapen: true
23:08.55terrapenDBs are so overused
23:09.02Beirdobe nice to have it as an option though
23:09.13tzangerwhat advantage would a rewrite have?
23:09.19tzangerterrapen: but for a phone system it makes a lot of sense
23:09.23Beirdowhich it seems it soon will be with Asterisk RealTime
23:09.27tzangerespecially for iax.conf, sip.conf and extensions.conf
23:09.29terrapenyou only need a DB if a) you are storing massive amounts of information and need to access a small peice quickly   and b) you need to run specialized queries against that database
23:09.32terrapenerr data
23:09.40terrapentzanger, no, it doesn't
23:09.43terrapenit does not make sense
23:09.57Beirdoif you say so :)
23:10.00terrapena phone system is a critical peice of infrastructure
23:10.03Nuggetit makes more sense for iax.conf and sip.conf, but I don't see how it makes much sense at all for extensions.conf
23:10.11terrapencomplexity decreases stability
23:10.13Beirdofair enough
23:10.17tzangerterrapen: true
23:10.17Nuggetthe benefits sure wouldn't outweigh the disadvantages
23:10.18terrapenand adding a DB increases complexity
23:10.23Beirdoextensions.conf is good as it is
23:10.35terrapenand the DB bits are there, if you want them
23:10.45Beirdoiax.conf, sip.conf, and especially voicemail.conf.  nice to have in DB
23:10.48terrapenbut they should not be required, as most people do not (and should not) use them
23:10.51Nuggetfundamentally, extensions.conf is not a database.  to make it fit in a database would require compromises to functionality.
23:11.03km-yeah
23:11.04Beirdoagreed
23:11.06tzangerI've changed my mind about extensions.conf
23:11.07km-extensions.conf is fine the way it is
23:11.08tzangeryou're right
23:11.13km-hahaha
23:11.17Beirdoor a complete rewrite, and that would be silly
23:11.19km-"I dont want to have this conversation anymore!"
23:11.21tzangerjust need to get the "reload" command working better
23:11.23Beirdo:)
23:11.24terrapenbeirdo, any novice perl programmer can generate any one of those config files based on information pulled from a database
23:11.31tzangerit has on more than one occassion hung the box for a few seonds
23:11.41*** join/#asterisk svantuil (~svantuil@054.209-89-66-0.interbaun.com)
23:11.45km-these mp3's need to get the hell out of CVS
23:11.52Beirdohehe
23:11.55tzangerkm-: or get moved to asterisk-sounds with the rest of 'em
23:11.59dsmousebut why isn't it called "dialplan.conf"?
23:12.11*** part/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com)
23:12.13Beirdobecause it isn't
23:12.15tzangerdsmouse: why is 'signaling' spelled wrong in zapata.conf
23:12.25dsmouseahhh
23:12.44dsmousetzanger: it's a industry term
23:12.46dsmouseSHHHH
23:12.52tzangerheh
23:13.19terrapen"to keep the noobs away"
23:14.52*** join/#asterisk agave-txlink (phanop@216.81.43.75)
23:15.01dsmouseterracon: it didn't work. I was a noob sunday a weekago
23:16.11*** join/#asterisk buddah (~hnic@67.110.253.129)
23:16.11svantuilnoob question: Is there any decent softphones available for XP that will interact directly with asterisk? My boss is pro MS, and pro opensource (oxymoron).
23:16.12*** part/#asterisk bsenicar (~bsenicar@BSN-77-155-238.dsl.siol.net)
23:16.40dsmousesvantuil: almost all of them talk SIP, which can work with asterick
23:16.49znosvantuil: avoid softphones altogether
23:16.51km-mmm one of the ladies at work bought a box of cow tails
23:16.51km-you know, those caramel candies with cream in the middle
23:16.51km-dinner is served!
23:16.58buddahquestion, is there anyway to assign 2 DIDs to 1 sip phone, and manipulate the sip configuration so that when either # is called it rings the same phone?
23:17.02dsmousebe wary of NAT with with sip too
23:17.07buddahor would i have to get a 2 port ATA for that?
23:17.21terrapengoddamn you, internet explorer
23:17.27*** join/#asterisk file (~file@mctn1-8179.nb.aliant.net)
23:17.28dsmousebuddah: that should be easy
23:17.29terrapeni hate coding around your bugs
23:17.32buddahi figured
23:17.38km-hahahaha
23:17.40dca[laptop]buddah: the DID provider should be able to send both to the same device
23:17.42km-terrapen: amen to that
23:17.53km-terrapen: you should try programming for Pocket PC's, they're a hoot.
23:17.54dsmousebuddah: when you have a spot for the extension that recives the line, make it dial the phone's extension
23:17.59dca[laptop]buddah: should be invisible to you
23:18.03buddahso like say 5142718929,1,DIAL(SIP/9378322738) to get the 514 to dial same phone as the 937?
23:18.10km-terrapen: I set the z-order through a wm_message and the device went nuts
23:18.33greg_workwhats the proper dial pattern for an area code? ZXX ?
23:18.41terrapenits hard to make things look nice in IE and make them work and do it without JavaScript
23:18.43km-NXX?
23:18.51buddahthink its N
23:19.27terrapenok im bored
23:19.31terrapenim going downstairs
23:19.36km-asterisk compiles slowly on a P2 266
23:19.45greg_workyeah i guess N would make more sense. thanks
23:19.49km-amazingly though, it runs pretty snappy on the box
23:19.49buddahok, next question. my boss does the DID thing, like he orders them and what not, and i dont know how providers work, when he orders another batch, can he get like 1 in 937 and 1 in 514, or do they usually make you get blocks?
23:19.54dsmouseterracon: don't type and walk down stairs as the same time...
23:19.55svantuilzno: it's just for testing anyways. So I can convince him to let me buy some digium hw and interface (attempt) to our analog ksu.
23:19.58*** join/#asterisk r1_ (~erwan@www.thiscow.com)
23:20.07Juggieits N, Z includes 1
23:20.27buddahthen it should be Z, some area codes are 1xx right?
23:20.32km-are there any good manager gui apps available to the open source community?
23:20.32buddahsome east coast
23:20.33buddahor no?
23:20.35km-buddah: negative
23:20.37buddahoh
23:20.42Juggieno, because long distance starts in 1.
23:20.42km-buddah: never heard of a 1xx areacode
23:20.44buddaherr i'm thinking of zip codes sorry
23:20.56dsmousemmmm
23:21.00dsmousedialing a zip code.
23:21.01km-buddah: yeah, 1xxxx is in PA (zip code)
23:21.05buddahyup
23:21.10km-sounds like a rainbow box to me
23:21.12buddah15044 was mine in pittsburgh
23:21.14stevekstevekhuh?
23:21.17buddahthats why i was thiking about that
23:21.22tzangerkm-: did it work?
23:21.23km-hehe, I live in philadelphia
23:21.25stevekstevek10001 is New York City..
23:21.30buddahso is 10101
23:21.37buddahbrooklyn i believe
23:21.38km-tzanger: asterisk isn't even close to finished compiling yet
23:21.40buddahor part of
23:21.40stevekstevekI think all of 10xxx is manhattan.
23:21.44km-tzanger: I'm running asterisk on a p2 266
23:21.45tzangerkm-: heh
23:21.47tzangerno worries
23:21.52tzangerI ran it on a P90 (no MMX)
23:21.57km-only to indications.c
23:21.58stevekstevek112 = brooklyn, etc.  PA doesn't get all of 1XXXX :)
23:22.03*** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca)
23:22.03buddahyeah
23:22.11km-ok, I know 17xxx 18xxx and 19xxx are in PA
23:22.17km-so I stand corrected
23:22.20buddahisnt some ny 0xxxx? or is that NJ?
23:22.29dsmousemaine?
23:22.30km-NJ has some 0xxxx
23:22.30buddah15xxx is PA
23:23.04tzanger17xxx?
23:23.12tzangeroh zip codes
23:23.17buddahyeah
23:23.18tzanger15xxx is PA too
23:23.20buddahyup
23:23.24buddahpittsburgh area
23:23.26Beirdopah
23:23.26buddahor north of
23:23.27tzangeryup
23:23.29Beirdo:)
23:23.33buddahi miss pittsburgh
23:23.34tzangerglenshaw, etc
23:23.36lyroyI'm in montreal and I want to know if there is any 514 or 450 DID provider, does someone can help me?
23:23.37buddahyeah
23:23.45buddahmy dad worked in glenshaw
23:23.54buddahlyroy: lemme know if you find any, i need a 514
23:24.08buddahi feel stupid looking for just 1 DID
23:24.18tzangerbuddah: our parent company's right on rte 8
23:24.22buddahnice
23:24.29buddahi lived right off rte 8 in gibsonia
23:24.32Beirdosixtel maybe?
23:24.33lyroywell I'm looking for a couples of 514 DID
23:24.34Beirdoiax.cc
23:24.40Beirdothey might offer something
23:24.54lyroyalright thanx ill check this out
23:25.03Beirdothey might also suck rocks, but worth looking at
23:25.08mikegrblyroy: check the asterisk-biz mailing list
23:25.10buddahdo profiders typically allow a request of like just 1 DID?
23:25.19Beirdothere's a Canadian VOIP providers page on the wiki
23:25.19buddahor is there a min.?
23:25.21tzangerthey're not the greatest but they're not bad
23:25.24Beirdomin 1 :)
23:25.42buddahawesome
23:25.45Beirdofor VOIP providers, I'd expect 1 is a common request
23:26.19buddahok, now i dont feel bad
23:26.25km-we buy our DID's in blocks of 20, but we do have some 1-off numbers which lead me to believe you can get them one at a time
23:26.28Nuggetlike good.
23:26.37buddahi need one for montreal, and one for dayton, so people can call me without long distance
23:26.39*** join/#asterisk hcclNoodles (~hcclnoodl@hcclnoodles.plus.com)
23:26.42Beirdofor any home geek users, 1 would be the norm
23:26.50buddahfigured i might as well since the boss is paying for my usage
23:26.52tzangerVOIP providers allow onesie-twosies
23:26.57tzangertelcos generally sell in blocks
23:27.02tzangerBell Canada, for example, sells in blocks of 30
23:27.13BeirdoBell Canada sucks :)
23:27.14Blackvelwhat place in asterisk code (zaptel, libpri) can I change, so DISA picks up some extensions faster?
23:27.24km-tzanger: chan_oss
23:27.32km-tzanger: hehe, this thing sure does cook!  woo doggie!
23:28.00tzangerBlackvel: what do you mean
23:28.22lyroyyou can buy DID from Bell Canada?
23:28.26km-so, my question remains, anyone know of a good graphical manager interface?  One that preferably works in windows?
23:28.53Blackveltzanger: i mean i type this in my analog pbx telephone: #91309110000
23:28.58tzangerlyroy: if you have a PRI, yes
23:29.04Blackveland asterisk does not get 9110000 as extension
23:29.05tzangerkm-: not offhand no
23:29.10Blackvelbut that had worked some time ago
23:29.25BlackvelBUT, if I call #9130
23:29.35BlackvelDISA picks up, gives me dail tone, and I can dail 9110000
23:29.45tzangerBlackvel: interesting
23:29.50tzangerbrain's not functioning well enough to help yo uhere though
23:30.00Blackvelwell its not blockmode, but I used DISA all the time to get that #91309110000 scenario working
23:30.01Blackvelhehehe
23:30.04Blackvelneither mine :(
23:30.13km-ut oh
23:30.16km-pbx_dundi died
23:30.19Beirdowhy would you want a number starting with 911?
23:30.19km-ah I need to get libzlip
23:30.39Blackvelbut asterisk 1.0.2 and 1.0.5 app_disa.c has not changed, so its some zaphfc code
23:30.49Blackvel_91 is sipgate, _92 nikotel, etc
23:30.58Beirdobad idea
23:31.04tzangeryeah I agree
23:31.14Beirdodon't use something that will start with 911
23:31.15km-so, can somoene explain in 100 words or less what DUNDi is, and what one might use it for?
23:31.22Beirdonot in North America
23:31.26Blackvelhehe
23:31.28Beirdothat's just asking for trouble later
23:31.29Blackveli am in germany
23:31.31Blackveli dont care
23:31.38Blackvelits my home system
23:31.39tzanger:-)
23:31.40BeirdoAh, don't use 112 then :)
23:31.45Blackvelhehe
23:31.49drumkillakm-: check out the whitepaper on dundi.com
23:31.51mutilatormy phone is like instant dial soon as 911 is hit dtmf timeout is like 2ms  or something
23:32.11Blackvelwell this * is not connected to pstn, its parallel to my pbx
23:32.12BeirdoI'd still suggest avoiding 911
23:32.15*** join/#asterisk cbachman (~cbachman@129.105.7.250)
23:32.26Blackveltell me why i cant dail in, thats more important! :)
23:32.28Beirdoit could well have special code somewhere that you aren't expecting
23:32.41Beirdothere's likely a reason
23:32.43*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
23:32.46Beirdoanyways, home time
23:32.50km-ooh, WAMI
23:32.55Blackvelyeah, it is, new bristuff version :(
23:33.42buddahgood lord. 2.50 a month and 1.1c a min
23:33.44buddahand thats in canadian
23:33.47buddahthats like nothing in US
23:34.00*** join/#asterisk grailink (~grailink@adsl-66-143-140-135.dsl.stlsmo.swbell.net)
23:34.03km-I need to get some asterisk stickers for this box
23:34.04km-hehe
23:34.16grailinkhey guys i have a quick question...
23:34.20Blackveli am off
23:34.21Blackvelcu
23:34.45grailinkwith meetme the ztdummy is causing serious pausing/delay issues... is there anything that I need to adjust to fix this?
23:34.47Juggieanyone know of any free SMS services?
23:35.09grailinkanyone? anyone?
23:35.11grailink:)
23:35.17*** join/#asterisk Grooby (~Grooby@12.22.232.212)
23:35.32*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
23:35.40Groobyanyone here w/ experience setting up speex?
23:35.42buddahanyone know a provider with cheap 937 (ohio) did's?
23:35.46tzangerok I'm an idiot
23:35.56tzangerI figured it'd be time-saving to skip the 'l' diskset in slackware
23:35.59km-tzanger: hmm?
23:36.06grailinkanyone here have meetme working?
23:36.07tzangerof course that includes skipping all the development libs
23:36.11tzanger*rolls eyes*
23:36.12km-oops
23:36.16Trionnisbuddah: check www.myphonecompany.com... they look kinda low rent, but they have a ton of did's for 4.95/mo
23:36.21buddahk
23:36.29km-oooh codecs
23:36.30Trionnisuse level3 for dids
23:36.31km-we're almost there!
23:36.35Trionnisso they have a lot
23:36.58tzangernow to see if I can remember how to do globbing iwth wget so I can get *.tgz in the /l/ directory
23:37.06Groobyi compiled speex from speex.org
23:37.08*** join/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com)
23:37.09Groobyand recompile my asterisk
23:37.27Groobybut it's crashing when trying to load codec_speex.so
23:37.29wankellevel3 is great.  they just don't talk to you for under like $15k/mo :P
23:37.37hermieanybody here ever lease dark fiber
23:37.37MicH323Anyone using Ast* with BroadVoice?
23:37.39NuggetTrionnis: is there a voip termination firm that doesn't look low rent?  :)
23:37.40Trionnisanyone have some real-life asterisk + broadvoice stories they'd like to share? thinking about a did from them, but I'd like to hear from the guys that use it :)
23:37.48Trionnislol @ Nugget
23:37.51GroobyMich, I use * w/ BV
23:37.52Trionnisthis is true
23:38.37km-trionnnis: someone earlier was saying broadvoice was the only company they ever complained about
23:38.46*** join/#asterisk GodThor (~ninja@212.110.67.6)
23:38.48MicH323I am having registration problems, tried BroadVoice Support mages. but get message -- Called 17182500199@sip.broadvoice.com
23:38.48MicH323<PROTECTED>
23:38.48MicH323<PROTECTED>
23:38.48MicH323<PROTECTED>
23:38.55TrionnisI've heard their support really just doesn't exists
23:38.59Trionniser
23:39.00Trionniss
23:39.03Trionnis-s
23:39.09Groobymich, change your host to sip.broadvoice.com
23:39.14Groobydon't use proxy.dca.broadvoice.com
23:39.17Groobysee if that helps
23:39.21MicH323ok
23:39.33Groobyno one has problem with speex huh?
23:39.44loudsuffered those broadvoice problems last night ..
23:39.55km-almost done!  yay
23:39.57grailinkare the any alternatives to meetme for conferences that work well?
23:39.58Trionniswell, I'm mostly just wanting them for EU calls
23:40.04km-ok, moment of truth
23:40.07grailinkthat don't need the zaptel timer
23:40.14Poincarehmmm, there was a page somewhere with the 'costs' for transcoding from one to another codec... anyone knows where to look for it?
23:40.16TrionnisI call .de a lot, so 20/mo ain't bad really
23:40.21loudTrionnis, only g711
23:40.25Trionnisewww
23:40.27loudyeah, i do .ar and .br all the time
23:40.27Trionnis:(
23:41.28tzangermyphonecompany.com doesn't have any canadian DIDs, at least that's my impression since you can't select a province
23:41.29grailinkpoincare: i saw a page on the wiki that mentioned that transcoding is pretty expensive. do a search on it and it should come up. i'm not sure if it came up with specifics
23:41.41Trionnisthey might not tz
23:41.42tzangergrailink: depends on the codecs
23:41.46Trionnishe asked for ohio tho
23:41.47Trionnis;)
23:41.50tzangeroh
23:41.54tzangerI thought he was looking for 514
23:41.56tzangerwhich is Montreal
23:42.06grailinktzanger: gsm
23:42.12Trionnis<buddah> anyone know a provider with cheap 937 (ohio) did's?
23:42.15Trionnis;)
23:42.21km-tzanger: that did it!  YAY
23:42.25tzangergsm is pretty cheap
23:42.28tzangerkm-: excellent :-)
23:42.52MicH323On the broadvoice. sip.broadvoice.com did the trick!!! Many Thans
23:43.00Groobyyup yup
23:43.03km-tzanger has saved my voip project!
23:43.05GodThorwho knows hot to installed h323 on asterisk
23:43.08km-all hail tzanger!
23:43.10tzangerkm-: hahaha
23:43.14mishehugsm is blah.
23:43.23tzangermishehu: I disagree, it sounds *great* to me
23:43.29grailinktzanger: i need a dynamic conference room that you can setup by calling an xtension. i got it working up till the point where the conf starts and now it just stutters like an idiot. I think its the zaptel timer ztdummy
23:43.33tzangermishehu: I want to like ilbc but every time I use it I get complaints about the quality
23:43.34km-tzanger: dude, ya need some back patting for that :)
23:43.39grailinkbut I can't find an alternative thats worthadamn
23:43.41Trionniswhat stutters?
23:43.48*** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com)
23:43.48hcclNoodleshi there, new to the channel, is there anybody from digium on here, i have a question re TDM400p in the UK
23:43.50TrionnisI'm using meetme with ztdummy
23:43.52Trionnisworks ok
23:44.01tzangergrailink: it could be -- grab yourself an x101P or a clone and try it
23:44.10Trionnis2.6.10 kernel
23:44.15tzangerkm-: hey I am just glad it is working for you :-)
23:44.21MicH323I will be attempting to compile h323 on Asterisk tonite :)
23:44.34grailinktrionnis: what hardwaer?
23:44.38tzangerwhen I get it in my head that something should work, I generally find a way to make it work.  This time it was very easy :-)
23:44.42Trionnis;)
23:44.42km-YES!
23:44.43MicH323Need to sort out the libs!#
23:44.45km-that is so awesome
23:44.51Groobyok...i compiled speex and make clean; make; make install asterisk
23:44.53zapathanks to all
23:44.55Trionnisgrailink: ??
23:44.57Trionnisno hardware
23:45.08Trionnis2.6 kernel has a high-res timer built in
23:45.10grailinkthe hardware you're running asterisk on
23:45.13Trionnisah
23:45.21Groobynow i am getting [codec_speex.so] ouch: error while writing audio data:: broken pipe
23:45.25TrionnisAthlon 2400xp, 512mb
23:45.28tzangerkm-: now install the patches from bug 2532
23:45.33Groobycan someone point me to the right place to start debugging?
23:45.40tzangergive them the highly experimental (but very stable and VERY GOOD) jitter buffer
23:45.53tzangerI've been running it for the last two or three weeks
23:45.59tzangerand switch-3.nufone.net is running it too
23:46.05tzangerso you can have end-to-end groovy jitter buffer
23:46.14grailinkhmm... this thing should work. well i'm running 2.6.1 and when I try to use meetme without ztdummy it gives me the /proc error but with it it stutters.
23:46.15tzangerIAX2 only at the moment but zoa's getting it to work on SIP
23:46.23*** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net)
23:46.25tzanger"/proc error" ??
23:46.29shmaltzanybody here using a polycom phone?
23:46.39*** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no)
23:46.44tzangershmaltz: no, but I hear they're very good
23:46.44Trionnisdid you do "make linux26" when you compiled ztdummy?
23:46.58Trionnisthat's what hooks it into the kernel timer instead of the usb one
23:47.04hcclNoodlesanybody regarding TDM400P ????
23:47.10shmaltzwow, tzanger. Do the polycom's support IAX2? :)
23:47.12tzangerhcclNoodles: what's up
23:47.15tzangershmaltz: no :-)
23:47.27grailinkcorrection Unable to open '/dev/zap/pseudo': No such device
23:47.37Trionnisinsmod ztdummy
23:47.39Trionniser
23:47.42Trionnismodprobe, rather
23:47.50*** join/#asterisk guugmember (~nachoramo@168.234.226.39)
23:48.01guugmemberhello, can we edit the code that writes the CDR´s so we can save a new field? after creating it in the table in mysql of course
23:48.17guugmemberif yes, where?
23:48.20mishehuguugmember: you got the source...
23:48.22grailinkUnable to open pseudo channel for timing...  Sound may be choppy.
23:48.23grailinkwhat's that?
23:48.23Trionnisof course you can, it's open source
23:48.24km-ugh, that sucks
23:48.26*** join/#asterisk Ahewes (~rsb@adsl-69-107-39-45.dsl.pltn13.pacbell.net)
23:48.30Trionnis./fanboy
23:48.33Trionnis;)
23:48.41km-tzanger: I can't transfer phone calls across the two pbx's
23:48.42grailinki did that and this is the new error: Unable to open pseudo channel for timing...  Sound may be choppy.
23:48.47grailinkinsmod that is
23:48.47Trionnisok
23:48.53hcclNoodlesin the uk , the card doesnt detect the british telecom hangup tone, and as such when in IVR /voicemail (when the line is not quiet) the card does not end the call
23:49.10Trionnisgrailink: take this to pm? easier to keep track of
23:49.21grailinksure
23:49.46km-small price to pay I guess
23:50.10hcclNoodleseverybody in the UK with this card is getting this apparantly, and we dont know what to do, digium support have suggested we try a few settings but they do not work,
23:50.10tzangerkm-: could you before?
23:50.21km-tzanger: I hadn't gotten far enough to test it
23:50.37km-tzanger: I can still transfer intra-legacy-pbx, but I can't transfer from legacy pbx out to asterisk
23:50.40tzangerhcclNoodles: can you get disconnect supervision on UK phone lines?  i.e. battery drop or battery reversal?
23:50.42km-I can transfer, naturally, from asterisk into the pbx
23:50.54tzangerkm-: well no, the PBX sees * as trunk lines
23:50.56|Vulture|Anyone know of a Firwall/Router for ~$500 that has an ISP failover feature?
23:51.01tzangerkm-: and it's absurd to transfer a call to a trunk line
23:51.05guugmemberTrionnis, so I can, what file does that?
23:51.12km-kinda sucks because I can't park people on asterisk from the pbx
23:51.21km-I bet I could come up with a zapbarge app or something
23:51.25Trionnishaven't a clue
23:51.26km-there's a ZapRedirect type thing?
23:51.29Grooby:(
23:51.32guugmemberkm-, what are the two pbxs? both *?
23:51.39Groobyno one have any experience with speex?!?!
23:51.45hcclNoodlesi dont know, can you elaborate tzanger
23:51.50tzangerkm-: yeah -- that's one of the reasons I was looking at the ADPs for the NEC
23:51.54Trionnisfind the code that handles CDR and start digging
23:52.00tzangerkm-: it's also why I used PRI -- I *can* do that to a limited degree with the norstar
23:52.00Trionnisonly thing I know to tell ya
23:52.02km-guugmember: one's a NEC system, the other's a Asterisk
23:52.14tzangerkm-: and I could *own* the Norstar if I could get MCDN reverse-engineered
23:52.15shmaltzanybody (but tzanger) using polycom phones?
23:52.16guugmemberTrionnis, ok, thnks
23:52.22km-tzanger: I suppose I could steel the adp for an FXO
23:52.32km-but that'd mean at max only one call could be transferred back to the system at once
23:52.45guugmemberkm- have you tried to pass calls from * to Avaya? I will have to do that soon
23:53.18tzangerkm-: you wouldn't happen to have a manual for those ADPs would you?  SPecifically flash codes or *-codes to do things like transfer or call forard and stuff?
23:53.20hcclNoodlesi know that in the US the hang up signal is a voltage change or polarity reversal, but in the UK it is continuous tone from the teco for a period between 15-30 seconds
23:53.42Graphikosdoes a "sip reload" from CLI reload everthing including extensions.conf?
23:53.43km-tzanger: hmm, no, I dont believe they ever left us a manual on it
23:53.48*** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com)
23:53.56km-tzanger: I'll dig around
23:53.57tzangerkm-: damn, ok
23:54.02hcclNoodles*/TDM400P doesnt detect this and waits for silence on the line
23:54.22*** join/#asterisk |neuro| (~|neuro|@212.176.51.231)
23:54.45EssobiMmm.
23:54.49guugmemberwhat do you think of varion, V400P-E 4 Port E1 Digital Interface Card
23:55.02EssobiGraphikos no
23:55.08Essobithat's what "reload" does
23:55.09guugmemberwhere can I see the diff between it and TDM400P
23:55.13Essobisip reload only reloads sip.conf
23:55.27Graphikosok.. so reload by itself...
23:55.32EssobiWhat?
23:55.35Essobiyea
23:55.44Essobijust reload loads all the "supported" configs
23:55.50km-first phone call
23:55.52km-worked good
23:55.56km-tzanger: I've got local echo problems
23:56.00Essobinice
23:56.06Essobilove the echos
23:56.21km-I kicked rxgain up a bit to make the audio coming in a bit hotter
23:56.21*** join/#asterisk ChrisRouse (~crouse@67.131.247.187)
23:56.28km-is it possible that I kicked it up too far?
23:56.31Essobimake sweeeet sweeet love to them, they'll leave you eventually.
23:56.40Essobikm- maybe
23:56.42ChrisRouseGood afternoon.
23:57.33tzangerkm-: meaning you hear your own voice
23:57.35tzanger?
23:57.58ChrisRouseI have a question about Cisco Call Manager integration if anyone is available...
23:58.02km-tzanger: yeah
23:58.20km-I just shut off the rxgain and the problem appears to have gone away
23:58.21tzangerkm-: you can try killing txgain too
23:58.26KalD|WorkChrisRouse, we've done that at my place - what's the q?
23:58.28tzangerkm-: also what do you have for echocancel= settings?
23:58.34*** join/#asterisk Cresl1n (~matt@216.207.245.23)
23:58.38km-no echocancel settings
23:59.02*** join/#asterisk dsfr (~dsfr@216.207.244.183)
23:59.10tzangerkm-: try echocancel=32 and restart (not reload) asterisk
23:59.10km-should I set echocancel=yes and echotraining=yes?
23:59.16tzangerkm-: don't use echotraining yet
23:59.24tzangerand dont' use yes, it's 128 and that's awfully long
23:59.28ChrisRouseKald: I am working on integrating Asterisk with all manager. I am having a problem associating Cisco extensions with Asterisk. For instance when I attempt to login as an Agent and type in my extension asterisk tells me that it is an invalid extension.

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