irclog2html for #asterisk on 20050914

00:00.04sergiovel_iax.conf I have all codecs
00:00.16sergiovel_I will try that
00:00.27fugitivosergiovel_: you should set that options if you only want ilbc for that user
00:00.28sergiovel_to disallow all codecs and see what happens
00:00.59*** join/#asterisk Tili (i=Tili@202-133-67-164-dialup.sat.net.pk)
00:01.02sergiovel_I allowed most codecs in iax.conf since some users need it
00:01.10sergiovel_but I will try that anyways
00:01.13fugitivosergiovel_: set that options only for that peer
00:01.15sergiovel_thanks for the tip fugitivo
00:01.27fugitivosergiovel_: do you receive my private messages?
00:01.29sergiovel_only for that user
00:01.39Ariel_Blake0PS, if it's like sendmail you need to setup your correct host name in /etc/hosts if your in RH you need to put it in /etc/sysconfig/networks
00:01.39sergiovel_oh ok
00:01.41sergiovel_hango
00:10.06*** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
00:13.34*** join/#asterisk CoderCR (n=creyna@cpe-24-165-14-170.san.res.rr.com)
00:14.07CoderCRwhat is the easier way to hide caller id from extensions.conf
00:14.26*** join/#asterisk bkw__ (n=brian@adsl-69-154-1-104.dsl.tulsok.swbell.net)
00:14.32CoderCRhey bkw
00:14.45bkw__yo
00:14.54fugitivoya
00:15.05Ariel_hay
00:15.12fugitivohey
00:15.34Ariel_buena
00:15.37fugitivoho
00:15.55CoderCRhow do i hide my caller id in extensions.conf?
00:17.31*** join/#asterisk IOscanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net)
00:18.39*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
00:18.39*** mode/#asterisk [+o twisted[asteria]] by ChanServ
00:18.53ManxPowerCoderCR, your expensive search of the asterisk mailing list archives were not helpful?
00:19.00fugitivolol
00:19.03ManxPowerextensive, even
00:19.09mitchelocor expansive
00:19.12Ariel_SetCallID${unknown}
00:19.31ManxPower~docs
00:19.32jbotextra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
00:19.47ManxPowerAriel_, I'll be he's using analog.  They never read the archives.
00:20.10Ariel_ManxPower, ok
00:20.17Ariel_ManxPower, how are you doing today
00:20.27Ariel_is nufone finally working?
00:20.33CoderCRexten => _9NXXXXXX,2,SetCallerID(${unknown})
00:20.58CoderCRlike that?
00:21.08Ariel_CoderCR, not really
00:21.14CoderCRno
00:21.15CoderCR:S
00:21.26Ariel_SetCallerID(UnKnonw)
00:21.38ManxPowerAriel_, they started working sometime before the following morning
00:21.44*** join/#asterisk L|NUX (n=linux@202.5.145.14)
00:21.45Ariel_great
00:21.46ManxPowerAriel_, I made it back to my apartment
00:21.48mitcheloc***** SetCallerID(asterisk) ;)
00:21.51Ariel_well?
00:22.08ManxPowerAriel_, everything I own now fits in 1/2 of a 5'x10' storage room.
00:22.17Ariel_wow
00:22.23Ariel_I know that feeling
00:22.28ManxPowerI don't recall how high the cealing is.
00:22.31mitchelochow did you salavage your computers?
00:22.40ManxPowerbut it's prolly like 1/4 of that area
00:22.53ManxPowermitcheloc, I picked them up and put them in the van.
00:23.10ManxPowerAriel_, You remember the CNN pictures of people picking thru the rubble of their houses?
00:23.13mitcheloc*duh*, i was picturing the wind and rain coming through a broken window for some reason
00:23.22mitchelocwhile you were rushing them to safety...
00:23.26mitchelocespecially the ones running asterisk
00:23.28ManxPowerthat's what most of what Bay St Louis looks like
00:23.40Ariel_ManxPower, we found part of our stuff about a block away.
00:23.49ManxPowerI took pics, but not of the worst hit areas, it just felt wrong
00:23.58ManxPowerMY damage was all water damage.
00:24.21Ariel_ManxPower, yes I do remember. I tend not to look at them. Brings back too many bad memory's
00:26.24ManxPowerAriel_, It's just terribly depressing
00:27.02Blake0PSAriel_ : Isn't /etc/hosts for local IP addresses only?
00:27.06mitchelocmanx are you gong to move back or somewhere else?
00:27.17mitchelochosts is like dns but it overides anything else
00:27.19ManxPowermitcheloc, there isn't enough infrastructure to move back to.
00:27.23Ariel_ManxPower, yes also the next day gets worst when you start to wake up over the lost of things and items like person effects that can't be replaced.
00:27.28mitchelocso where will you move to?
00:27.45ManxPowermitcheloc, Atlanta TX (30 miles from Texarkana) for 1 - 2 monthsn.
00:27.51ManxPowerdon't know where after that.
00:27.59mitchelochow old are you?
00:28.10ManxPower36
00:28.25bkw__ManxPower, you're not that far from me
00:28.25mitchelocoh ok, so you can take care of yourself pretty well then probably
00:28.35Ariel_Blake0PS, yes it is. But you can also put a quailified domain name of your server there.
00:28.58mitcheloci was talking to my mother about offing a job to someone out there, but i get the feeling most will want to move back to new orleans no?
00:29.03ManxPowermitcheloc, I have to keep my significant others in mind
00:29.10mitcheloc*** back home, not necessarily NO
00:29.17Ariel_ManxPower, I got a job offer but I really dont' want to go up to Nebraska.  just too cold for me. If you want I will send then your name?
00:29.18mitchelocof course
00:29.34mitchelocAriel: california is warm ;)
00:29.43Ariel_mitcheloc, it's warmer here
00:29.57dougheckajob doing what?
00:30.01dougheckaasterisk/voip stuff?
00:30.03mitchelocpsst, i've got a job offer here if you know .net and love asterisk ;)
00:30.04ManxPowerAriel_, I have the work, it's the matter of recovering from the temp loss of income and massive increase in expenses
00:30.05dougheckaor computer/network stuff?
00:30.05Ariel_yes
00:30.35ManxPowerI'd consider moving to California, a nudist campground, or europe.
00:30.45dougheckaHAAHA
00:30.46Ariel_doughecka, voip/asterisk stuff. We are considering it due to the income.
00:30.53dougheckaAriel_: too true :)
00:30.58dougheckaa nudist campground?
00:31.08dougheckayou-a settup a webcam no?
00:31.17mitcheloca * sorority nudist campground
00:31.22dougheckawe could vote
00:31.22ManxPowerdoughecka, you'd be suprized at how common they are
00:31.23fugitivoAriel_: i went there
00:31.24Blake0PSAriel_ : Is this where exim4 will grab the Return-Path information from then?
00:31.43fugitivoAriel_: near sunny isles?
00:31.54dougheckaManxPower: you have to admit, its cheaper than buying new threads
00:31.55doughecka:)
00:32.04ManxPowerdoughecka, It saves on laundry
00:32.09dougheckaexactly
00:32.09Ariel_Blake0PS, I know know exim4 but sendmail does from there unless you have rh which is in /etc/sysconfig/networks
00:32.43Ariel_fugitivo, actually it's closer to Orlando
00:32.56CoderCRdoh
00:32.59CoderCRunknown did not work
00:33.08CoderCRit is still showing the same ID
00:33.18fugitivoAriel_: then there are more than one nudist area in florida :)
00:33.29Hmmhesaysnow that is damn weird Set(CALLERID(num)=1234567) works fine but Set(CALLERIDNUM(num)=${myvar}) doesn't
00:33.30Ariel_there are many
00:33.41nextimeok, agi zombies problem solved. pyastre.so is bugged.
00:33.42*** join/#asterisk adker (n=adker@67-136-209-100.dsl1.glv.ny.frontiernet.net)
00:33.53dougheckafugitivo: yea, all around the edge of florida, it borders water usually
00:34.11dougheckaor is this complete nudists?
00:34.18fugitivocomplete nudists
00:34.45CoderCRAriel: i did as you said.. it still shows the same number
00:35.09*** join/#asterisk adker (n=adker@67-136-209-100.dsl1.glv.ny.frontiernet.net)
00:35.09dougheckaoh, not almost-but-not-quite-nudist-because-I-have-1-square-inch-of-fabric-covering-me beaches?
00:35.09Ariel_CoderCR, pastebin.ca your dial setup.
00:36.16dougheckaindeed
00:36.20dougheckait blinds people
00:36.33sivanaairplanes get confused
00:36.40*** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net)
00:36.40*** join/#asterisk L|NUX (i=asad@66.225.200.234)
00:37.09CoderCRhttp://pastebin.ca/22868
00:37.58opus_oh! bad!
00:38.01opus_line 14 : #
00:38.01opus_exten => h,1,Hangup
00:38.21opus_thats a big bug in asterisk, it'all go into an infinite loop
00:38.53hugo-v6hmmm wont compile cleanly :/
00:39.01hugo-v6*patch*
00:39.02CoderCRAriel: http://pastebin.ca/22868
00:39.06Ariel_CoderCR, there are allot of analog lines that don't allow you to change the callerID
00:39.35CoderCRthis is a T1
00:39.36dougheckalike ALL analog lines
00:39.38CoderCRPRI T1
00:39.47Ariel_CoderCR, you need to fix the last few lines
00:39.50dougheckaCoderCR: your telco has to allow you to set CID
00:39.59CoderCRi am allowed to set caller id
00:40.09Ariel_does your provider allow you to send different callerID info?
00:41.07Ariel_change it from unknow to 00000000 numbers instead
00:41.28Ariel_allot of them don't allow names but only numbers
00:41.56dougheckaisnt name pulled from the telco's billing systems?
00:42.01dougheckaI didnt know you could even set it
00:42.09Ariel_doughecka, sometimes depending on the provider
00:42.17dougheckasweet
00:42.19hugo-v6puh. wont rewrite the complete app. use working one
00:42.42Ariel_but most of the names come from the end point not the calling point
00:42.57dougheckaah
00:43.34Ariel_if your sending the call via sip to sip or iax then yes you can set the name and number. But most of the others are just the number.
00:44.01*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw)
00:45.34opus_http://ws.cdyne.com/NotifyWS/phonenotify.asmx?op=NotifyPhoneBasic
00:45.36*** join/#asterisk neowillis (n=neowilli@61.149.11.221)
00:46.04CoderCRthat allowed me to do it
00:46.44Ariel_CoderCR, so numbers it took then
00:48.51*** part/#asterisk neowillis (n=neowilli@61.149.11.221)
00:48.54*** join/#asterisk neowillis (n=neowilli@61.149.11.221)
00:49.10*** part/#asterisk neowillis (n=neowilli@61.149.11.221)
00:49.14CoderCRyes
00:49.38*** part/#asterisk CoderCR (n=creyna@cpe-24-165-14-170.san.res.rr.com)
00:50.21mitchelocoh damn...just thought about something for katrina...it's a chance for us to rebuild their entire telecommunications network and power it all by asterisk
00:50.32mitchelocvolunteer of course
00:50.44mitchelocit'd be great if we could co-ordinate some effort to help them out in that area...
00:50.59Ariel_wow why not get the goverment to pay for some of it.  Since there going to any way.
00:51.03mitchelocthe businesses will all need new phones and things like that..
00:51.49Ariel_Yes your correct. When they start to rebuild there will be a great need for new pbx's.
00:51.52mitcheloci'm up for helping out, is there anything as a community we could do?
00:52.07mitchelocmaybe an e-mail to the mailing list...
00:52.17fugitivomitcheloc: you'll get novel peace prize
00:52.40Kattymitcheloc: beep!
00:52.44mitchelocheh, i'd rather just have clear calls to that area terminated by allison saying "thank you" (after leaving a vm)
00:52.53twistedmitcheloc, while that's a novel idea, most of the telecommunications network is still in tact, just some downed lines and whatnot
00:52.59Kattytwisted: !
00:53.08ManxPowerdoes CVS-HEAD only support kernel 2.6 now?
00:53.09twistedKatty!! :)
00:53.16Ariel_mitcheloc, belive me when I say that the biz will get there pbx and pay for them. The ones that need more of the help will be the people's homes which the major telco's rule
00:53.19Kattytwisted: do you suffer from long term memory loss? o/~
00:53.29twistedKatty, no
00:53.29mitcheloctrue, the main system was probably protected pretty well....the small businesses will need help
00:53.37Kattytwisted: that's not how the song goes.
00:53.41Kattytwisted: try again
00:53.48twistedKatty, what song?
00:54.02Kattytwisted: chumbawamba - amnesia
00:54.05Ariel_ManxPower, no I just finished setting it up on a 2.4 as of yesterday.
00:54.14twistedKatty, oh
00:54.18twistedKatty, i don't know that one
00:54.19*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
00:54.27Kattyi must be getting old
00:54.34twistedKatty, either that, or I don't remember it ;)
00:54.43mitcheloc=/
00:54.46jskcrhy all
00:54.48ManxPowerAriel_, idd.
00:54.51ManxPowerAriel_, odd
00:54.59Kattytwisted: get back to work.
00:55.05mitchelocwell if anyone can come up with a good way to help them with their phone needs...
00:55.22twistedback to work? hahahaha... i just got home from the office
00:55.24Kattytwisted: ooh, do you bark too? ;)
00:55.36twistedKatty, actually, yes, I do
00:55.42Kattyneat.
00:55.55twistedI also meow, whinny, and hoot.
00:55.59ManxPower"Property of Digium"
00:56.00Kattykinky.
00:56.06twistedManxPower, NOT
00:56.06hugo-v6.o(lots of subdoggies here?)
00:56.06jskcranyone use a t1 and mix voice and data with hdlc
00:56.20Ariel_slaves
00:56.20Kattyi can't picture twisted as a puppy
00:56.23twistedKatty, don't ya know it :P
00:56.28twistedlol
00:56.28Kattyhe's just....too big
00:56.34dougheckaKatty: what a twisted mind
00:56.36twistedi couldn't picture myself as a puppy either...
00:56.40ManxPowerUgh.  EVERYTHING smells like mildew
00:56.44dougheckayum
00:56.45twistedManxPower, ack
00:56.50dougheckafree drugs
00:56.56VcoWHERE?
00:56.58Kattydoughecka: i'm a geek and a perv. i can't help it.
00:57.09twistedKatty, they go hand in hand
00:57.15Kattyobviously.
00:57.15twisted(most of the time)
00:57.28Vcoisn't geek perv an oxymoron?
00:57.32Kattythough i admit the anne rice beauty novels repulsed me
00:57.37twistedlol
00:57.41Kattyi want to burn them
00:57.44KattyBURN them.
00:57.50twistedkeep them for when the day after tomorrow happens
00:57.50ManxPowerKatty, They were....odd.
00:57.58twistedyou can stay warm from their burnination
00:58.09KattyManxPower: i sure didn't like them.
00:58.10ManxPowertwisted, better to burn the tax code.
00:58.12dougheckaKatty: female too?
00:58.17twistedManxPower, :P
00:58.20Kattydoughecka: uhh
00:58.24Kattyi'm not answering that.
00:58.33ManxPowerKatty, at least you read them.
00:58.37dougheckawell, I knew a guy named sue
00:58.47KattyManxPower: i have anne rice's entire collection.
00:58.50twistedi knew a boy named suzanne before.
00:58.59KattyManxPower: and 147 star trek novels.
00:59.04KattyManxPower: amongst other things.
00:59.11twistedtrekkies.
00:59.15Kattydoughecka: yes, i'm a female.
00:59.17dougheckaLOL
00:59.23Kattydoughecka: and a natural born one too
00:59.25ManxPowerKatty, I lost all my star trek novels in Katrina: Storm of Doom.
00:59.30twistedbwahahahhaha
00:59.31dougheckaKatty: woah, no way
00:59.47ManxPowerI didn't realize it until now.  *sigh*  They were a birthday present
00:59.49twistedManxPower, not laughing at you, btw.
00:59.51Kattywhat? a girl can't be in the asterisk channel?
00:59.53dougheckaall them grown-in-a-glass-bottle ones get to me after a while
01:00.09mitchelocno Katty, a female can't be on irc, not specifically #asterisk ;)
01:00.15mitchelocwell, they can be, but it's rare
01:00.20Kattymitcheloc: and certainly not screened irssi
01:00.23Kattymitcheloc: oh noes!
01:00.29mitchelocirssi?
01:00.40ManxPowerA female geek is great, but like a gazelle in central park, uncommon enough to cause comment.
01:00.52dougheckaManxPower: did you make that up?
01:00.52twistedManxPower, hahaha
01:01.03Kattywe need more gazelle, obviously
01:01.03ManxPowerdoughecka, Yes, but inspired bt Scott Adams
01:01.04dougheckacause its poetry
01:01.07hugo-v6damn again 3pm. i should go to bed earlier
01:01.15hugo-v6s/pm/am/
01:01.16Ariel_see you all later. It's time to sit and watch tv with my girls...
01:01.16dougheckaah, I knew it sounded familer
01:01.21KattyAriel_: bye bye
01:01.27dougheckaKatty: female gazelle
01:01.27twistedAriel_, toodles
01:01.36dougheckatwisted: toodles?
01:01.45Kattyget off the toodles.
01:01.51twisteddoughecka, yeah, as in, "see ya later"
01:02.01twistedonly stranger.
01:02.02dougheckalol
01:02.14Kattypeachy keen!
01:02.16dougheckaits starts with g
01:02.19twistedgoogle
01:02.20dougheckaand ends in ay
01:02.26Kattygray!
01:02.33twistedgambalay
01:02.34dougheckaKatty: your good
01:02.34mitchelocgay?
01:02.38ManxPowerThe three non-technical books that were not destroyed were: Dilbert: The Way of
01:02.43ManxPowerThe Weseal
01:02.44Kattyi'm always good.
01:02.44jskcrdoughecka:  I would with a very smart femle with a phd :)
01:02.50ManxPower"I HAte Fun"
01:02.57jskcrerr /would/work
01:02.57twistedjskcr, DOH, i forgot your shit
01:02.58dougheckalol
01:03.04ManxPowerand Final Exit for cats: a Feline suicide guide.
01:03.11dougheckatwisted: I prefer to flush it actually
01:03.11twistedjskcr, was gonna do it today, but got all busy and shit at the office again
01:03.19jskcrtwisted:   as long as I get it by friday its all good
01:03.20dougheckaManxPower: woah, scan that in for me
01:03.33twistedjskcr, check your doorstep tomorrow
01:03.38twistedjskcr, and don't forget to stomp it out
01:03.40jskcrtwisted: cool thankx
01:03.41Hmmhesayswell everything seems to be working right now
01:03.48KattyHmmhesays: hi asl
01:03.51opus_damn it. mkisofs doesn't like svn
01:04.03ManxPowermy most commonly referenced tech books were also not damaged
01:04.03jskcrtwisted:  have you used a t1 with channelized data/voice
01:04.05twistedKatty, rofl
01:04.13dougheckaWonka: where abouts?
01:04.21twistedjskcr, used one? no.  set them up? yes.
01:04.22HmmhesaysKatty: lolzomg 87,male(ish),in an out house
01:04.27twistedactually
01:04.31KattyHmmhesays: lolhotttt!!11oneone
01:04.36twistedwe use one at the office, but it goes into an IAD before the box
01:04.55drumkillaanyone have a polycom ip300 handy?
01:04.55Wonkadoughecka: germany, about 54°24'N, 10°E
01:04.59dougheckaasl?
01:05.02twisteddrumkilla, no.
01:05.03jskcrtwisted:  I have a problem with the channel bank im using freaking impedence problems
01:05.05dougheckaWonka: sweet, own a VW?
01:05.17ManxPowerdrumkilla, not handy
01:05.19jskcrtwisted:  Im ripping it out and puting it a te110p
01:05.21Wonkadoughecka: no, a honda motorbike
01:05.26ManxPowermy IP300 was also damaged.
01:05.29twistedjskcr, heh, coo
01:05.44twistedManxPower, water damage or structural failure + water damage?
01:05.53dougheckaWonka: lol
01:05.56drumkillahm, just trying to find out the rating of the power supply
01:06.01Wonkadoughecka: wassup?
01:06.02Kattymm, eagles.
01:06.08twisteddrumkilla, enough to power teh phone
01:06.09ManxPowertwisted, at LEAST the power supply under water.  The handset may or may not have fell into the water.
01:06.09dougheckaoh dunno
01:06.33Kattytwisted: yeah like that's going to work.
01:06.39ManxPowerI have like 10 transformers for various devices that were soaked in water, but the device was not
01:06.46twistedKatty, i can become small when necessary
01:06.48Kattytwisted: you're over a foot taller than me.
01:06.49WonkaKatty: you're too thin to hide behind?
01:06.58opus_did anybody get that TTS call back webservice to work?
01:07.04Hmmhesayspoor lindsay fagerlee, she has no voicemail
01:07.05Wonka<-175cm, 60kg
01:07.10KattyHmmhesays: aww.
01:07.25HmmhesaysI find it amusing to watch my creation run
01:07.33dougheckaopus_: yea, its 11 digits, not 10
01:07.35twistedHmmhesays, hehe. that's always fun
01:07.37dougheckabut I never heard anything
01:07.40Kattydo you feel like a daddy, Hmmhesays?
01:07.49twistedHmmhesays, what's NOT fun however is when some dillhole resets the rack
01:07.50Hmmhesaysa little bit, and ldap dialplan daddy
01:07.57dougheckaKatty: do you feel like my daddy, Hmmhesays?
01:08.09jskcr<-185, 100kg
01:08.13Kattydoughecka: uhh.
01:08.27Kattydoughecka: kthxbi.
01:08.28opus_dougchecka, what do you mean. are you talking about http://ws.cdyne.com/NotifyWS/phonenotify.asmx?op=NotifyPhoneBasic
01:08.42Kattytwisted: :>
01:08.45dougheckaKatty: what is that
01:08.46mitchelocC# ;)
01:08.52dougheckaopus_: yea
01:08.53mitchelocermm well maybe, but .net nonetheless
01:09.02Kattydoughecka: something to annoy you. is it working?
01:09.04Hmmhesayssomeone seriously wants to get ahold of lindsay fagerlee
01:09.08Hmmhesaysthey keep calling and calling
01:09.12twistedhaha
01:09.17Hmmhesayseven though i'm dumping them after a 1 minute
01:09.19twisted37 idle in #asterisk-unregistered not including myself
01:09.20Kattymaybe it's her mommy
01:09.28Hmmhesaysi could call her and ask
01:09.30Hmmhesayslol
01:09.31blitzrageevening all
01:09.34jskcr38
01:09.37dougheckaKatty: no, google helps
01:09.41Kattyhi blitzrage
01:09.50blitzrageKatty: how are you this fine evening?
01:09.59Kattyblitzrage: awake!
01:10.05Kattyblitzrage: miracles /do/ happen.
01:10.07mitchelocopus_: blah thatt thing does too much checking
01:10.11MikeJ__jbot, WAKE UP!
01:10.18jbotUP!: GOOD MORNING!!!
01:10.24doughecka~monkey
01:10.26jbotThis problem, like many others in the computer industry, can be solved by the application of monkeys.
01:10.37jskcrhow long does it take to get a patch reviewed?
01:10.46MikeJ__depends.
01:10.49dougheckawhens the next bluemoon happening?
01:10.58MikeJ__jskcr, a day, or a few months...
01:11.03blitzrageI have a question: I have a Macro doing a Dial(Local/number@context) without a timeout. That context then performs a Dial() as well. When that Dial() times out, I want the context to immediately end, giving control back to the Macro - any ideas how I might do that?
01:11.07ManxPowerAriel_, I think I had a kernel .vs. kernel source mismatch
01:11.17MikeJ__or anywhere inbetween
01:11.25MikeJ__having patches usually helps
01:11.36jskcrI posted one
01:11.38blitzrageonly thing I can think of so far is to set the AbsoluteTimeout to 0 after the dial, then not include a 't' extension - then the call would fail right away (what I want), but it seems a bit messy
01:12.00*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
01:12.14MikeJ__having working, and tested, with community support helps more... but is still no promise somone is available to review
01:12.17Drukengod damn required registration
01:12.23MikeJ__jskcr, which one?
01:12.26twistedblitzrage, you could always use a goto(macro-blahblah,exten,pri)
01:12.33jskcrhttp://bugs.digium.com/view.php?id=3658
01:12.36twistedjust return the call to the macro afterwards
01:12.41twistedit's a bit of a kluge, but it would work
01:12.54blitzragetwisted: problem I see if that the Dial to the local channel never ends then...
01:13.00twistedyeah
01:13.03twistedthis is true
01:13.11Kattytwisted: are you flying to the next convention?
01:13.12blitzrageI need a "break" application :)
01:13.16twistedwhy are you dialing a local channel anyway?
01:13.18twistedKatty, yea
01:13.24blitzragetwisted: find me follow me stuff
01:13.31twistedblitzrage, use macros for that
01:13.32blitzragetwisted: its the only way to do what I want
01:13.32MikeJ__jskcr, and a one liner too.. nice
01:13.33MikeJ__ummm
01:13.34Kattyhmm.
01:13.38blitzragetwisted: macros won't work for this application
01:13.40sivanamaybe someday we can have a real extension logic similar to AEL
01:13.46twistedblitzrage, for find me follow me? sure it will.
01:13.48MikeJ__try to catch oej online
01:13.49Hmmhesaysor SER
01:13.56MikeJ__he is UTC +1 I think..
01:13.57blitzragetwisted: trust me - the way I'm doing it is a better way :)
01:14.05MikeJ__and ask him to peek at it
01:14.07jskcrMikeJ__:  its rfc 3261 compliant  :)
01:14.09twistedblitzrage, if you say so :P
01:14.17blitzragetwisted: what I'm doing I've tried with Macro's - it won't work with them
01:14.28twistedblitzrage, you're a doc writer, not an implementer...
01:14.29jskcrMikeJ__:  Ive tested on links upto 5000ms
01:14.30blitzragetwisted: either way - my problem still stands :)
01:14.34MikeJ__yeah.. oej usually gives his thumbs up on most sip stuff before anyone else will look
01:14.38blitzragetwisted: ouch
01:14.44twistedblitzrage, ;) love ya dawg
01:14.48*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
01:14.52blitzragetwisted: so you're saying I can't implement Asterisk eh?
01:14.54blitzrage:)
01:14.59sivanamaybe some day we'll have a development plan
01:15.17twistedblitzrage, i didn't say that, you did
01:15.20twistedblitzrage, :P
01:15.25blitzragetwisted: you kind of said that though :)
01:15.27twistedsivana, a development plan?
01:15.31blitzrage<@twisted> blitzrage, you're a doc writer, not an implementer...
01:15.31Wonkadev-what?
01:15.38jskcrMikeJL: granted 5000ms turns it into a nasty walky talky :)
01:15.58sivanaya
01:16.01sivanaa development foucs
01:16.03sivanafocus
01:16.13twistedword.
01:16.18dougheckasup
01:16.23dougheckafo shizzal
01:16.27jskcrMy favorite cvs patch for chansip is the accunt fix
01:16.28blitzragedamnit .... still have to figure out how to break this damn Dial()
01:16.33dougheckadawgs with new threads
01:16.34twistedjskcr, thank you :)
01:17.11jskcrSure ya can code in C but can you spel
01:17.15twistedblitzrage, if you can outline what you're trying to do, i can pastebin you a dialplan for it in about 5 minutes.
01:17.16Hmmhesayseverybody was kung foo fighting
01:17.41twistedze wuz fast az litenang
01:17.44dougheckaKUNGGG FOO
01:17.45MikeJ__twisted, jskcr fixed that sat link bug
01:18.03twistedMikeJ__, so did martin pycko, but apparently wrongly.
01:18.17sivanatwisted: does anyone know what Mark's focus is with development?
01:18.31twistedsivana, Mark does
01:18.42Hmmhesaysdamnit cvs-head won't let me exit the console
01:18.51twistedHmmhesays, sigint!
01:18.53twistedsigint!
01:18.57Hmmhesayshahah
01:19.01sivanaheh
01:19.03tzangerHmmhesays: dammit now I want to hear that song
01:19.04*** join/#asterisk Syncros (n=sysop@noc.routermonkey.net)
01:19.12Hmmhesaystzanger I got a dance remix
01:19.18Hmmhesaysall yours if you want it
01:19.31tzangerHmmhesays: sounds disgusting :-)
01:19.34sivanatzanger: you're up past your bed time
01:19.35Hmmhesaysit is
01:19.36sivanaheh
01:19.38tzangeryes I know
01:19.40tzangertwisted: hahahaha
01:19.40Hmmhesaysbut oh so addicting
01:19.43jskcrbtw dont try to run spandsp-0.0.2pre18 with the asterisk-beta it dosn't work :)
01:19.45sivanaI broke down.. bought a blackberry today
01:19.57Hmmhesaystwisted if you alt f4 you can get to some l33t warez sites
01:20.00sivanagoing to VoIP it up.. haha
01:20.12twistedHmmhesays, strangely enough, that doesn't work on OSX :(
01:20.23Kattybye now.
01:20.24Hmmhesaysgrrr asterisk console pisses me off
01:20.26Hmmhesayslater Katty
01:20.29twistedKatty, nini
01:20.30jskcrtwisted:  kill -9 does :P
01:20.31tzangerexten => 2914574/5192912500,1,Goto(call-andrew,${EXTEN},nocid)
01:20.37Hmmhesaysgotta leave it running
01:20.37twistedjskcr, yes, this is true.
01:20.39sivanatzanger: Bell is offering digital voice services now
01:20.44mitchelocdo any of you know how a phone alert might work if someone doesn't answer their phone? i.e. my mothers school would like to send messages to the parents...if they don't pick up, is it possible to leave it on their voicemail?
01:20.47tzangerthat should work shouldn't it (the 'nocid' marker instead of an actual number)
01:21.03tzangersivana: what's that in english
01:21.09twistedmitcheloc, sure. use backgrounddetect() or whatever the fsck it's called now
01:21.13jskcrmitcheloc:  yes
01:21.36sivanatzanger: Bell has VoIP
01:21.49sivanatzanger: http://www.digitalvoice.bell.ca/Overview/
01:21.49twisteduh oh
01:21.49mitcheloccool, but does it detect their voicemail answering? i'm looking now (i'll probably answer my own question)
01:21.49twistedbrb
01:22.05tzangerinteresting
01:22.20tzangerat $40/mo
01:22.27sivanalol.. ya
01:22.40blitzragetwisted: spanish flea - the song returns!
01:22.48blitzragetzanger: yo
01:23.01tzangerblitzrage: yup
01:23.20blitzragetzanger: did you see my question in your scrollback?
01:23.36tzangerblitzrage: which?  (that would be a no)
01:23.52Hmmhesayslater kids
01:23.52blitzrageI have a question: I have a Macro doing a Dial(Local/number@context) without a timeout. That context then performs a Dial() as well. When that Dial() times out, I want the context to immediately end, giving control back to the Macro - any ideas how I might do that?
01:24.04blitzrageonly thing I can think of so far is to set the AbsoluteTimeout to 0 after the dial, then not include a 't' extension - then the call would fail right away (what I want), but it seems a bit messy
01:24.12jskcrblitzrage:  with a goto
01:24.26mitchelochmm....interesting stuff (backgrounddetect)
01:24.40blitzragejskcr: no - if I do that then the Dial() calling the Local channel never ends
01:24.45blitzrageit needs to "break"
01:24.57jskcrblitzrage:  then set a var and use a gotoif
01:25.09twistedblitzrage, congestion()?
01:25.14twistedhangup()
01:25.24*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
01:25.32Kattyyay, bed.
01:25.34blitzragetwisted: hrmmmm - I wonder if Hangup the whole channel... or just the Dial(Local/)
01:25.35ManxPowerI think someone started to loot my apartment, but didn't find anything worth stealing
01:25.38blitzrageKatty: night
01:25.39tzangerblitzrage: wow I have no idea
01:25.39twistedKatty, ?
01:25.44ManxPoweror were interrupted
01:25.50twistedblitzrage, you could call softhangup on the channel
01:25.51blitzragetwisted: let me try that - I wonder if Hangup() will do it :)
01:25.53Kattytwisted: bed is good.
01:25.58twistedKatty, i agree
01:25.59tzangerthis anti-ex-girlfriend feature is unbelievably useful
01:25.59jskcrManxPower:  or shoot by someone
01:26.03Kattyblitzrage: amn't sleeping yet. tis only eight thirty
01:26.09twistedblitzrage, softhangup(${CHANNEL}) might work
01:26.10ManxPowerjskcr, I dunno.
01:26.22blitzrageKatty: oh - was under the impression you were going to bed :)
01:26.28Kattyblitzrage: i did
01:26.33Kattyblitzrage: doesn't mean i'm sleeping
01:26.46jskcrManxPower:  In LA there going into all the apt's and houses to check
01:26.48mitcheloc*arg* i've been in this channel just for today (been a while since the last time) and i don't know how any of you focus on your work!
01:27.06twistedhey ManxPower i might be down in NOLA at the end of the month for a few days
01:27.26mitcheloctzanger: until she blocks her number, or fakes her callerid ;)
01:27.27ManxPowerjskcr, I don't/didn't live in Louisiana
01:27.31ManxPowertwisted, gads, why?
01:27.34tzangermitcheloc: you just do it :-)
01:27.37twistedManxPower, rebuild effort
01:27.42ManxPowertwisted, ah.
01:27.44blitzrageKatty: oh I see :)
01:27.49mitcheloci get distracted easily =X
01:27.51tzangermitcheloc: nope, no CID ends up direct to voicemail too, and calling from a friend's phone just adds that to the list
01:27.51ManxPowerWell, if you make it to Texarkana, look me up LOL!
01:27.58tzangershe'll run out of friends and payphones long before I run out of patience
01:28.07blitzragetwisted: the Hangup() just hangs up the Dial(Local/) portion - perfect!
01:28.09twistedManxPower, moving to there?
01:28.12twistedblitzrage, ;)
01:28.19ManxPowertwisted, for 1 - 2 months
01:28.20mitchelocor.....she fakes her callerid to your mothers phone number, or yours even, or just a random one...
01:28.22twistedblitzrage, i told you. you're a documenter, not an implementer ;)
01:28.30blitzragetwisted: *insert middle finger here*
01:28.34twistedblitzrage, hehehehe
01:28.35mitchelocof course if your gf is that smart...you'd have to ask why she is your "ex"...
01:28.36tzangermitcheloc: she has no concept of faking CID, and then I start looking at ANI
01:28.48sivanahehe
01:28.49tzangersmart women can be just as aggravating as smart men
01:29.01blitzragemore so
01:29.04mitchelocwhat is the difference between ani vs cid?
01:29.16tzangerbesides, my incoming ANI seems to jus tmimic CID... I set my CID to a pittsburgh, PA number and the ANI matches it, which should not be possible
01:29.16twistedmitcheloc, ANI is facility/circuit based
01:29.23twistedCID is simply a label put on the call
01:29.29twistedthat's quite possibly the easiest way to describe it
01:30.05tzangerI recompiled chan-zap with PRI_ANI enabled but from teh looks of it it alters outgoing only not incoming
01:30.33tzangerso it looks like Bell just isn't giving me ANI, although when I call my house from my cell the ANI number is set to some random number which is in the physical location of the cell phone (telus mobility)
01:30.40tzangerso I am getting different IEs from Bell
01:31.10tzangerI wonder ... if I don't get an IE for ANI, maybe * just copies it
01:31.17tzangerand if I get the IE then it uses it as received
01:31.30JerJeri have to believe there is issues with asterisk and ANI
01:31.45JerJermore specifically libpri and ANI
01:31.58JerJeri know my upstream is providing us info digits, but asterisk cannot see them
01:32.17JerJeri just haven't taken the time to track it down
01:32.36JerJerso for now we eat payphone surcharges :(
01:33.15*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
01:33.16sivanaJerJer: any resolution to the scam issue?
01:33.21JerJernope
01:33.28sivanaJerJer: ie, you on hook or still under investigation?
01:33.51JerJerwe are responsible for a half a million dollar phone bill
01:33.57sivanashit
01:34.04JerJerand we've lost one carrier because of it
01:34.05jskcrthe afgan lec's? JerJer?
01:35.44twistedhey JerJer can I ask you if you see any problem with my ani-II mappings?
01:36.06twistedhttp://pastebin.ca/22869
01:36.11tzangertwisted: all I ever get for ANI2 is '0' from Bell Canada
01:36.41jskcrINMATE/PRISON :)
01:36.55tzangertwisted: oh shit I didn't know that's what ANI2 was!
01:37.38blitzragewhat's ANI2?
01:37.41JerJertwisted: all we ever see is 0 or a null field
01:37.46twistedblitzrage, extended ANI
01:37.53blitzragegotcha
01:37.54twistedblitzrage, contains number type
01:37.56JerJereven from known payphone calls
01:38.03twistedJerJer, oh.
01:38.09twistedbut do those codes look correct to you?
01:38.14jskcrwhat about cocotts?
01:38.25twistedcocott is a payphone :)
01:38.30JerJeryet i know my upstream is sending II digits -  we can see them on a 5300 i still have around
01:38.48tzangerwell 0 is right then for POTS
01:38.52tzangerwhich is what 99% of my calls would be
01:38.56tzangerI'd have to call from a payphone now to see
01:38.58JerJertwisted: they look right - but  i haven't played with them in quite a while
01:39.04tzangeralthough I never see '26' which is cell
01:39.13twistedJerJer, ah.  I know i always get 26 when i call from my cell
01:39.20tzangertwisted: what telco?
01:39.26*** join/#asterisk Borgon (n=L3orgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
01:39.27JerJerhttp://www.nanpa.com/number_resource_info/ani_ii_assignments.html
01:39.39*** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net)
01:39.48twistedhmm.
01:39.50twistedJerJer, thanks :)
01:40.59JerJer26 says unassigned according to that link
01:41.10twistedyeah
01:44.46*** part/#asterisk mitcheloc (n=mitchelo@ip67-153-163-202.z163-153-67.customer.algx.net)
01:45.26hugo-v6well... waiting for fax-spam sucks.
01:46.16jskcrTheres a company down here you send you fax-spam to and they go after the spammers
01:47.06hugo-v6jskcr: atm it would be welcome. since its 4am so i cant call someone to send me a fax ;)
01:47.24hugo-v6well go to sleep and see in a few hours if i got a few
01:48.34*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
01:50.01tzangerdamn I just had the ex call from a payphone and ANI2 was still '0'
01:51.08tzangerI changed my dialplan to answer right away now too before dropping to voicemail
01:51.12tzangershe'll be losing the quarter
01:51.26tzangernot sure if voicemailmain answers right away or just before the beep
01:52.14tzangerI hope she doesn't make me change the outgoing voicemail message for her to FUCK OFF ALREADY
01:52.38tzangerI hate being rude but jesus christ I told you in person a thousand times, I told you in email, I told you on MSN and you STILL own't leave me the fuck alone
01:52.50tzangerwhat other choice do I have...  next step would be restraining order I guess :-S
01:54.22hugo-v6tzanger: well... shift happens :>
01:54.24*** join/#asterisk BrianR___ (i=brianr@24.61.206.174)
01:54.31BrianR___hello folks.
01:54.33jskcrtzanger:  stalker?
01:54.38BrianR___I'm working with an asterisk system where calls come in on one PRI and are immediately forwarded (via Dial) out another PRI. Trouble is, indications like busy / congestion / etc. are not passed along in the event the Dial() fails. Any suggestions?
01:54.51tzangerwow the app_voicemail code isn't exactly the most straightforward
01:54.58*** join/#asterisk Delta34 (n=delta34o@198.87.24.253)
01:55.01hugo-v6tzanger: record a msg and let * play it on her phones til she learns it
01:55.18tzangerlooks like it answers right away though...  my Answer() is superfluous
01:55.36tzangerI should have never given her my main DID
01:56.07hugo-v6tzanger: get a number for girlfriends only ;)
01:56.13Nivextzanger: set up an extension that plays the screaming monkeys and just transfer her to that everytime she calls :)
01:56.22hugo-v6thats what i did
01:56.35jskcrSpeaking of funny ringers mine play's mission impossible
01:56.37tzangerhugo-v6: I did have that :-)
01:56.42tzangerNivex: heh
01:56.48tzangerno as I said I do not enjoy being mean
01:56.50Vcoxfer  her off to anpther ex gf
01:56.55tzangerVco: haha
01:56.56Vcoanother ...rather
01:57.04tzangerI think that's a bad idea
01:57.05hugo-v6vcoo: lol
01:57.08tzangerlike matter and antimatter colliding
01:57.13Vcoheh..heh..
01:57.17*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:57.23DarthClueI could calculate your chances of survival, but you won't like it.
01:57.25VcoShit + Fan
01:57.34BrianR___jskcr: i have that ringer too.. grabbed the midi file from diax, a wind0ze IAX softphone...
01:57.50jskcrBrianR___:  mines converted from a wav file
01:58.14BrianR___Anyway.. Anyone have ideas why if one call comes in from a PRI and gets Dial()'d to another PRI without answering, why the signalling info is lost?
01:58.27BrianR___Ie, shouldn't the first caller get busy if the Dial() returns busy?
01:58.51BrianR___It some cases it seems like dial() is answering the call on me so I can no longer send status information down the PRI. :(
01:59.21Delta34any experienced cisco 7960 sip users out there?
01:59.48NuggetDo you really care how experienced we are?
02:00.01NuggetWhat if I don't know a damn thing except the answer to the one question you're avoiding asking?
02:00.10hugo-v6is *cli> restart now like stoping * and starting * again?
02:00.29*** join/#asterisk Derkommissar (n=pocketir@82.sub-70-197-196.myvzw.com)
02:00.31Delta34is there to program the buttons
02:00.40Derkommissarsup people
02:00.53*** join/#asterisk SwK (i=yacjru@12-219-144-126.client.mchsi.com)
02:01.10SwK*yawn*
02:01.38Delta34so i wan to add the transfer button to be on the bottom roll of displayed buttons
02:01.46Delta34is there a way to do that?
02:02.04Derkommissarl have a new pockt pc......  verizon high speed.....   anyone know a good softphone for it?
02:02.48spackle_Derkommisar: there is no such thing as a good softphone.
02:02.57jskcrDerkommissar:  what kind of pocket pc
02:03.02Derkommissaran ok then
02:03.10Derkommissarsamsung
02:03.18jskcrtry skype
02:03.23Derkommissarwith 1xvdo
02:03.34Delta34another question regarding 7960 sip phones, when i dial an extension, can it show who i am calling
02:03.44Derkommissar300kbps at this time
02:04.16Derkommissardont want skype i w
02:04.27Evanrudeanyone in here really farmiliar with the Directory() function?
02:04.41*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
02:04.42Derkommissari want it registerd to my asterisk
02:05.26spackle_Derkommisar: that will be cool.  does the device have a USB port?
02:06.04Derkommissarits  a pocket pc .. wifi an
02:06.07Derkommissarbt
02:06.39Derkommissarand also usb connecrivity
02:06.49hugo-v6kewl. asterisk receives my faxes :)
02:06.57hugo-v6im in love.
02:07.18jskcrhugo-v6:  what version of spandsp are you using?
02:07.28hugo-v60.0.20pre20
02:07.28Derkommissarwelcome <hugo-v6>
02:07.42Derkommissari use it for r2
02:07.46Derkommissar;)
02:07.53jskcrhugo-v6:  your using cvs or the beta?
02:08.02hugo-v6jskcr: tried at first with .3pre1 but it wont compile. and too much to do to fix it ;)
02:08.09hugo-v6jskcr: 1.0.9
02:08.25Derkommissarjskcr. u wrote it?
02:08.30jskcrnoo
02:08.34hugo-v6Derkommissar: for r2? whats that the other part of d2?
02:08.36jskcrI use it quite a bit
02:09.25Derkommissarnever gotten it to work.... but i got r2 for zaptel to work.... it uses the same drivers
02:09.49hugo-v6next step (tomorrow or in a few hours) is to send faxes via windows printer somehow.
02:10.36Derkommissarcups
02:10.40Derkommissar;)
02:10.45hugo-v6well here it works now with mISDN as ISDN < * > ISDN and ISDN < * > sip and ISDN < * > fax
02:10.46jskcrhugo-v6:  ive done it before with cups
02:10.53Derkommissarsee ya ppl later
02:11.20hugo-v6jskcr: well but i need a uhm printer-driver or such stuff for windows which let me etner a fax-number or such shit
02:11.26hugo-v6l8r Derkommissar
02:13.14*** join/#asterisk santiago (n=santiago@63.245.86.203)
02:13.35jskcrhugo-v6:  last time I did it I had it print a cover sheet for the faxes with the info in front of the fax
02:14.31hugo-v6jskcr: googleracle said winprint ;)
02:15.43hugo-v6damn its called winprinthylafax. and its only for hylafax
02:16.01*** join/#asterisk mwgbc (n=junkmail@adsl-71-132-196-196.dsl.pltn13.pacbell.net)
02:18.06mwgbcI get this msg in Asterisk CLI prompt but I cannot find anything on the web that defines the codes i.e. 0,2,3 Here is the line that I get:
02:18.06mwgbcSep 13 22:11:05 NOTICE[20216]: pbx_spool.c:243 attempt_thread: Call failed to go through, reason 3
02:18.07tzangerOk
02:18.14tzangerhow the fuck do I apply anti-ex-gf logic to my house
02:18.15tzangerugh
02:18.35Corydon76-homeUsing which tool?
02:18.51Corydon76-home5055?
02:19.05jskcrignore all callerid blocked fones
02:19.16tzangerjskcr: no I mean the doorways
02:19.28Corydon76-homeHeh
02:19.38Corydon76-homeTeleportation
02:19.58DarthCluetzanger, restraining order and an armed officer?
02:20.01Corydon76-hometeleport the ex-gf who walks in the front door to just outside your back door, facing the other way
02:20.04hugo-v6jskcr: how did u do that? i mean how did * got the informations like faxnumbeR?
02:20.09Corydon76-homeand vice-versa.  ;-)
02:20.16jskcrtzanger:  go to the court house and get a restraining order
02:20.20jskcrtzanger:  she comes within 500 feet go directly to jail
02:20.40tzangerDarthClue: yeah
02:20.46Corydon76-homeActually, she has to be reported, first
02:20.51tzangerjskcr: heh... trying to not let it get to that level
02:21.05jskcrC
02:21.08Corydon76-homeAnd the cops aren't bound to enforce restraining orders; see the recent SCOTUS case
02:22.23jskcryea and cities can take you property and build a hotel, they dont do it too often
02:22.25hugo-v6tzanger: hmmm guess a big wall with barbwire and a trench with crocodiles in it?
02:22.46tzangeryeah but that makes entrance and exit difficult for me too
02:23.59hugo-v6tzanger: well a drawbridge and a door could solve that problem.
02:24.26hugo-v6.o(back to the 16th century
02:24.28hugo-v6)
02:24.30tzangersure, but what about when I forget my keeys?
02:24.31tzangerer keys?
02:24.39hugo-v6tzanger: then ure fscked
02:24.46Himekohow do you reset a pap2-na (or sipura too i guess) to factory defaults?
02:24.51tzangeryeah leaving the basement window open's not enough any more :-)
02:25.02hugo-v6lol
02:25.07hugo-v6indeed that wont help
02:25.09Himekoi forgot my passwords on it
02:25.11mwgbcWhere can I find info about reason codes for * "Call failed to go through, reason __" ?
02:26.23fugitivoHimeko: tools -> system => restore
02:28.04*** join/#asterisk cio (n=na@adsl-068-209-198-242.sip.bhm.bellsouth.net)
02:28.07Himekoya, but i forgot my password
02:29.09*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
02:29.50fugitivouser password and admin passworD?
02:30.14hugo-v6hmmm sending it per email sucks but should work easy
02:30.41hugo-v6well go to bed now. 4h30 am dam early in the morning. have to get some sleep now and think about it.
02:30.54hugo-v6gd n8/day pals
02:30.56cioAnyone here run debian 3.1 and *?  My /etc/init.d/asterisk script doesn't seem to work.  Any suggestions?
02:31.14hugo-v6cio: yep i do but im on the way to my bed ;)
02:31.19fugitivocio: what does it say?
02:31.20hugo-v6*detach*
02:31.22ciog'night
02:31.28hugo-v6thx
02:31.29Himekooh crap
02:31.33Himekonever mind
02:31.36cioDoesn't say anything at all, just dies.  I can start * just fine from the command line.
02:31.53Himekoi am trying to loginto my wrt liek it was my pap2
02:32.02fugitivocio: does it run * as user asterisk?
02:32.18fugitivoHimeko: huh?
02:32.28ciofugitivo; hrm.. dunno.. How do I start a binary as an alternate user?
02:32.28Himekologing into the wrong ip
02:33.00fugitivocio: stop asterisk, chown to asterisk every asterisk file and try the script again
02:33.15Himekohappy there is new pap2-na firmware though
02:33.23Himekosupposed to fix a nasty bug
02:33.28cioAhh.. I see where you're headed..
02:33.34fugitivoHimeko: what bug?
02:33.37cioI'm using the debian package, btw.  1.0.7
02:34.13Himekowhen you have 2 different phone numbers for each port and you connect to the same server there is a registration bug
02:34.24fugitivoreally?
02:34.40cioYea, it's a permissions problem most likely.
02:34.48fugitivoHimeko: what version?
02:35.21fugitivoHimeko: i'm using 2.0.10 and i don't have that bug
02:35.28*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
02:35.56mwgbcSep 13 22:11:05 NOTICE[20216]: pbx_spool.c:243 attempt_thread: Call failed to go through, reason 3
02:36.23mwgbcI need to find the reason codes for lines like this.  Any ideas?
02:36.30cioShould astdb be owned by asterisk?
02:36.31JerJerfix reason 3
02:36.41Himekoslepp or methos know more about it than me
02:37.00Himekodepends on the setup too
02:37.23Himekoit didn;t affect me one their old provisioning but did on the new
02:37.51mwgbcJerJer: Am I suppose to type that at the *CLI prompt?
02:38.53Himekoit was a known bug and supposedly this new 3.1.5(LS) firmware fixes it
02:41.15cioNo luck...
02:41.41cioAnybody have a clue?  My /etc/init.d/asterisk script isn't starting asterisk.  I can start it manually just fine.  Probably a permissions problem of some sort.
02:42.27fugitivocio: try to see what you get in the logs
02:42.43*** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
02:42.46jskcrcio permissions problem
02:43.15ManxPower; There are presently two jitterbuffer implementations available for * and chan_iax2;
02:43.15ManxPower; the classic and the new, channel/application independent implementation.  These
02:43.15ManxPower; are controlled at compile-time.
02:43.19fugitivocio: chown to asterisk /var/log/asterisk too
02:43.29cioWhich log will show init.d stuff?
02:43.30ManxPowerOK, where do you control them at compile time?
02:44.18cioIt's owned by * ... hrm...
02:44.32fugitivocio: /etc/asterisk ?
02:44.50fugitivocio: /var/spool/asterisk ?
02:44.53cioYep...
02:46.53wunderkinManxPower: it looks like you are supposted to remove the NEWJB define in channels/chan_iax2.so if you want to use the old one
02:46.54cioI bet it's zaptel related.
02:47.37fugitivocio: check the logs, the answer is there
02:48.11ciofugitivo; thanks for the help. I've looked in the asterisk logs, nothing...
02:48.15*** join/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
02:48.31fugitivosyslog?
02:48.45*** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
02:48.57cioAhahahahah!!!!!
02:49.05cioThe problem was permissions on the log files themselves...
02:49.24fugitivo<PROTECTED>
02:49.31cioCool. Don't have to run it as root now. ehe
02:49.44cioYep, you were right -
02:49.47cioThanks for the help.
02:49.48fugitivo;)
02:51.40obsidian-studioswith regard to Cisco 7960, is there any way to determine if the # is the local area code and strip it's first three digits, and or if it's not local add a 1 on to it? Or do I need to if, and modify, pad it myself? like with if EXTEN:-3  = area code dial(ZAP/line/${EXTEN:3}), or else dial(ZAP/line/1${EXTEN}).
02:52.12obsidian-studiosbasically trying to come up with a way to return missed calls on the 7960
02:52.34obsidian-studiosis there a wiki page that covers if statements?
02:53.46mswobsidian-studios: sometimes you can complete those calls with 10 digit dialing
02:54.45*** join/#asterisk Supaplex (n=supaplex@shell.aros.net)
02:55.26Jabronianyone knows which would be the best way to be reconnecting a call every 5 minutes??
02:56.08JabroniI was thinking in having an AbosoluteTimeoute(), and when the timeout expires catch the T extension, BUT, when the T extensions gets invoked, the channel is still onhook
02:56.24Jabronitried doing a softhangup(inteface) and still no luck :(
02:56.37obsidian-studiosmsw: well at the moment when I try to redial a 10 digit number the phone does not even talk to * and just gives me a busy signal?
02:56.56mswobsidian-studios: that sounds like a digitmap problem or a extensions.conf problem
02:57.11mswobsidian-studios: if it never contacts *, it's a digitmap problem
02:57.52obsidian-studiosmsw: hmm, wonder if the phone does not like the first digit, I have run into issues with cisco fxs devices capturing digits
02:58.41fugitivoobsidian-studios: what's the problem?
03:01.27obsidian-studiosfugitivo: I would like the ability to return missed calls on a 7960 without having to manually redial #. two problems, the firsts is when I dial a 10 digit number starting with 9, seems I get a busy single from the phone. Seems to be the same if I manually dial the same 10 digit number
03:02.05fugitivoobsidian-studios: extensions.conf problem
03:02.23mswobsidian-studios: have you done anything to the dialplan on the phones?
03:02.25obsidian-studiosnow once I can get the phone to pass the info to *, I was curious if I could make an if statement to check the first 3 digits, if it's the local area code, dial the # with the first 3 digits stripped, if it's not local, pad the # with a 1
03:02.39obsidian-studiosmsw:  dialplan is empty at the moment
03:02.58obsidian-studiosfugitivo: problem with extensions.conf even if nothing shows up on the cli when I enter the #?
03:03.15mswobsidian-studios: so maybe the trick is actually to change the caller id info on the way out, instead of on the redial path
03:03.23obsidian-studiosfugitivo: I thought it might be my generic invalid and exit includes, but I commented them out
03:03.26fugitivoobsidian-studios: did you try ${EXTEN:3} ?
03:03.30mswobsidian-studios: that is, prepend the 9, do any other modifications when the call comes in
03:03.42mswobsidian-studios: that way the phone can just use the number as-is
03:04.10*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
03:04.17fugitivoobsidian-studios: example: _393.,1,Dial(XXX/${EXTEN:3}) ?
03:04.24obsidian-studiosfugitivo: that's what I was thinking and depending on the outcome do a dial(ZAP/1/${EXTEN:-3}) or dial(ZAP/1/1${EXTEN})
03:04.29mswobsidian-studios: because if your dialplan and extensions.conf don't handle dialing arbitrary local numbers (i.e., without 9) this wont't work afaik
03:04.32UberbotHi all.
03:05.05*** join/#asterisk ManxPower (n=eric@ip68-225-97-156.br.no.cox.net)
03:05.24*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
03:05.27fugitivoobsidian-studios:  _393.,1,Dial(XXX/${EXTEN:3}) for 3 digits prefix
03:05.30obsidian-studiosmsw: I can pastbin the extensions, it's very small but I have patterns for 7,10, and 11 digit #'s?
03:06.05*** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
03:06.16obsidian-studiosmsw: I can dial a 7 digit normal number and it will call it just fine
03:06.25obsidian-studiospretty sure I can dial an 11 digit one as well
03:06.34mswobsidian-studios: aah, cool setup
03:06.48obsidian-studiosmsw: yeah, trying to get away from the dial 9 for outside line ;)
03:06.54mswobsidian-studios: not the way we run it here, we use '9' to dial the outside line
03:07.18obsidian-studiosmsw: yeah that's the normal way, and I sort of had to with some other * deployments
03:07.22*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
03:07.23mswobsidian-studios: it's just sometimes tricky to get the local extensions worked in when you don't set aside a prefix to get out
03:07.53obsidian-studiosmsw: when I use Cisco fxs devices I have to use 9, because for some reason they will grab and strip certain #'s before passing the info to *
03:08.44obsidian-studiosmsw: there are not to many local extensions, deff enough to keep them in the  4 digit range, although I am not to sure about 411, and 911, I think I have them mapped properly
03:09.38mswobsidian-studios: you should be able to change that with dialplan templates, I would hope -- but I've not used any cisco ATAs
03:09.42mostywhen we make a SIP call -> asterisk -> E1 -> mobile phone, there's a couple of seconds delay between for asterisk to dial out through the E1. the person i set this up for wants that delay minimized, is there anything we can do about that?
03:10.00mostyugh, nasty grammar in there
03:10.29obsidian-studiosmsw: well it's the fxs devices in 827-4v, and UBR924/UBR925. Also sip does not pass along all info like CID :( via the fxs devices on those cisco routers
03:11.09JerJerhttp://www.naiveamoeba.co.uk/index.asp?dir=2&id=11
03:11.32obsidian-studiosmsw: I have no clue about dialplans and etc with those devices, although I sware I saw a digit strip or something along those lines in the IOS when I was poking around looking for a solution
03:12.54*** join/#asterisk Hmm-home (n=Neg@24-117-213-113.cpe.cableone.net)
03:14.20fugitivoobsidian-studios: use the logic in the asterisk box
03:15.08obsidian-studiosfugitivo: just found the page on expressions, so I can do what I want when I can get the 10 digit #'s to *?
03:15.34fugitivoyes
03:15.51obsidian-studiosfugitivo: here is my extension to handle 10 digit #s in the cli when I extensions reload Added extension '_NXXNXXXXXX' priority 1 to main_out
03:15.52Hmm-home10 digit #'s?
03:16.21obsidian-studiosHmm-home:  well the CID info get's stored in the phone for missed calls I want the user to be able to return those calls without having to enter the #
03:16.21Hmm-homeleave me, right here, cause I don't wanna go,
03:16.36*** join/#asterisk zobia01 (i=zobia@222.212.64.83)
03:16.44zobia01hello everybody
03:17.01zobia01any body knows how to turn of the debug of asterisk CLI ?
03:17.03Hmm-homei for one hate my users and want them to have to go through every little tiny annoying dial pad stroke
03:17.07fileHmm-home: poke
03:17.12Jabronizobia sip no debug
03:17.15Jabronior iax no debug
03:17.23Hmm-homeand if they screw up I want them to be called horrible names
03:17.32Hmm-homeyo file
03:17.38zobia01i tried them , it's not working
03:17.44obsidian-studiosHmm-home:  in that regard, I wonder what the max length and extension # can be :)
03:18.02*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl1k.dialup.mindspring.com)
03:18.04obsidian-studiosHmm-home:  give them all like 128 digit extensions ;)
03:18.28Jabronigrrrr i cant figure out a way to make * redail one number every 5 minutes :(
03:18.32zobia01this debug is not like sip debug i often use. the diguim guy turn it on but forget to turn it off
03:18.39zobia01Sep 13 22:17:48 DEBUG[6974]: manager.c:1210 process_message: Manager received command 'Command'
03:18.45Hmm-homeLoL
03:19.12zobia01any body has seen this kind of debug mod?
03:19.43filehot-n-sexy Hmmhesays@home, how are you?
03:20.25Hmm-homet'aint bad
03:20.26zobia01hello except sip debug, iax debug , what kind of else debug they can use?
03:20.30Hmm-homehowsit
03:20.42filenot bad
03:20.48fileI'm all hyper from being in a coding frenzy
03:21.04Hmm-homesweet, i've come down off my geek high
03:21.30obsidian-studiosok now I am high with file :)
03:21.34fileha
03:21.40fileyou don't want to deal with what I've been doing today
03:21.42Hmm-homedude that was epsom salt
03:21.46fileSIP contact branching :P
03:22.01obsidian-studiosHmm-home: it all does about the same thing :) fks u up ;)
03:22.13zobia01http://pastebin.ca/22876
03:22.33zobia01please take a look of this strange debug. i just want to turn this one off.
03:23.06Hmm-homelogger.conf
03:23.09*** join/#asterisk jimlinux (n=jim@69.49.168.58.swcp.com)
03:23.20filetime to write a cool'n'quick debug module
03:23.38*** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net)
03:24.19zobia01is there any person here work for digium?
03:24.36filejust calling them is easiest...
03:24.44fileduring business hours naturally.
03:24.57Hmm-homeor you can pay me 20 bucks to edit your config
03:25.01zobia01yes. but now it's not business hour. how can i do
03:25.05Hmm-homeso I can go play some pool
03:25.18filezobia01: well depends, why do you want to talk to someone from Digium?
03:25.22MikeJ__zobia01... well.. that probably depends
03:25.24MikeJ__heh
03:25.25bkw__Hmm-home, nice nick
03:25.28MikeJ__STOP THAT!
03:25.34fileMikeJ__: no!
03:25.41Hmm-homewhat is wrong with my nick?
03:25.44ManxPowerI have a choice between SBC Yahoo! DSL service or Charter Cable Internet for broadband where I'll be living for the next few months.
03:25.56MikeJ__CABLE!
03:25.58bkw__ManxPower, SbC has been good
03:25.59ManxPower"I can shoot you or I can kill you with a kinfe"
03:26.09zobia01i can not find digium person at this time.
03:26.10bkw__for 14.95 you can't beat DSL
03:26.19Hmm-homeif you were cool like ahhnold you'd get shot with a knife
03:26.22ManxPowerbkw_, Yeah, but I'll have to pay a $200 cancel fee when I move out of TX in 2 months
03:26.24MikeJ__zobia01, what do you need?
03:26.31bkw__ManxPower, if you're mmoving they don't bill it
03:26.38bkw__just ask before
03:26.42obsidian-studiosManxPower: cable usually no contract, dsl, contract
03:26.46ManxPowerAND who knows how long it will take them to install the service
03:26.46jarrodyea but you can find competitors to get you the higher rates with other options (i.e. static addressing) for cheaper bkw
03:26.52zobia01hello i need to turn off this debug mod http://pastebin.ca/22876
03:26.55filepeople are evil, stop existing!
03:27.01bkw__jarrod, thats stupid
03:27.01jarrodmanx where are ya in texas
03:27.01QwellSBC will also do a month-to-month contract
03:27.06Hmm-homezobia01: pay me 20 dolla so I can go play pool
03:27.07Qwellat least, in some areas
03:27.08bkw__yes SBC will
03:27.09ManxPowerjarrod, I will be in Atlanta TX
03:27.09obsidian-studiosManxPower: then again, dsl static ip, cable usually no chance
03:27.14jarrodbkw: i fail to see how that is stupid
03:27.23MikeJ__zobia01, so turn off debug
03:27.24ManxPowerobsidian-studios, I can live without static IP
03:27.31bkw__I used to think static IP's were good
03:27.33bkw__but not a big deal
03:27.36*** join/#asterisk gopher (n=gopher@d14-69-231-241.try.wideopenwest.com)
03:27.37obsidian-studiosManxPower: if you are in texas see if you qualify for Verizon fiber, if you do I will kill you :)
03:27.42QwellIPs are practically static on cable
03:27.49filehey is there something I can do... you broke your heart into too many pieces... now we gotta put it back together again
03:27.51jarrodbkw: that happens to be personal preference / need
03:27.52zobia01how to trun off debug?
03:27.56ManxPowerDynamic, fastest speed = $24/month, static at same speed $80/month
03:27.56jarrodmaybe you dont.. but some do
03:27.59Qwellzobia01: sip no debug
03:28.02ManxPowerthat's with SBC
03:28.04Qwellor whatever
03:28.05*** part/#asterisk gopher (n=gopher@d14-69-231-241.try.wideopenwest.com)
03:28.05fileQwell: you're funny
03:28.06zobia01i tried , no use
03:28.09obsidian-studiosbkw__: they can be nice if you setup rules and etc to block all other ips except known ones to certain services and etc
03:28.09Qwellfile: What?
03:28.11Hmm-homezobia01: I told you like 2 seconds after you asked before
03:28.16MikeJ__zobia01, modules.conf
03:28.16filezobia01: edit logger.conf and take out debug, Hmm-home said
03:28.21Qwelloh, THAT debug
03:28.23filerm -rf / would work too
03:28.23MikeJ__errr
03:28.26MikeJ__logger.conf
03:28.33bkw__file unlink /
03:28.34zobia01ok, let me try
03:28.34bkw__is much faster
03:28.34Hmm-homeor the power button on your monitor
03:28.36obsidian-studiosQwell: yes, my cable ip has not changed in 2.5 yrs, I did a dns mapping to it off my main network and T1 :)
03:28.53filebkw__: you're worse!
03:29.25ManxPowerI guess with the new IAX2 jitter buffer, I can live with cable
03:29.36wunderkinManxPower: well let me tell you something, im not sure if its still true anymore but i used to work for 2wire and at least they used to do support for sbc too.. if they still work with 2wire, i would definately get a 2wire modem, otherwise for line/modem support problems you're off to their offshore shit, theyre all dumb and they all go off of flow-charts for EVERYTHING
03:29.40ManxPowerbkw_, is the new iax2 jitter buffer enabled by default on CVS-HEAD?
03:30.00ManxPowerwunderkin, define "2wire"
03:30.10wunderkinManxPower: they make dsl modems
03:30.15obsidian-studiosManxPower: manufacture
03:30.25filemy DSL modem is uber-cheap... it's a "Comtrend"
03:30.42Hmm-homesome littler oriental boy probably lost a finger making it
03:30.44ManxPowerI'll go with cable.
03:30.52obsidian-studiosI still have an old Alcatel 1000 in my closet :)
03:30.55*** join/#asterisk Snake-Eyes (n=blog@203.201.98.28)
03:30.56fileit overheats so much that I've actually put it in the freezer before
03:31.09wunderkinsbc outsources all over the place,  bleck
03:31.22filelet your heart do all the talking!
03:31.38obsidian-studiosManxPower: fyi, you can get a Cisco UBR924 for $100 or less and Cisco UBR925's for $300-$400 if you want a cable router with fxs ports on it ;)
03:31.44Hmm-homei'm listening to HOT ACTION COP
03:31.52obsidian-studiosManxPower: if you have a use for fxs ports or etc
03:32.57*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
03:33.07Hmm-homethis chick's trying to pay for things swinging from a pole with her nipple rings and this dudes tryin' to ease the pain, tying off his arm with a nylon string
03:33.57*** join/#asterisk litage (n=nick@203.201.98.28)
03:34.44zobia01thank you all , it's working
03:35.07*** join/#asterisk halogen8 (n=halogen8@ip68-8-18-103.sd.sd.cox.net)
03:35.42ManxPowerobsidian-studios, I had one.  Lost in the flooding.  I did not shed a tear.
03:35.54Hmm-homewith my sexy pants and my blue hair....
03:35.58Hmm-homeI dunno, I got nothin
03:36.12ManxPowerThe flood got rid of lots of the crap equipment I could not bear to throw out.
03:36.14obsidian-studiosManxPower: flooding? hurricane stuff or general flooding?
03:36.16Hmm-homei need to find some karaoke tonight
03:36.21ManxPowerI have pics of me with blue hair.
03:36.27ManxPowerobsidian-studios, Katrina: Storm of Doom
03:36.54Jabronianyone knows why does the extensions T gets invoked even when someone hangups the channel?? isnt the T just when the AbsoluteTimeout is reached??
03:36.56obsidian-studiosManxPower: well the 924's are god awful with syncing, I think the 925's are a bit better, both are EOL
03:37.09ManxPowerI had tghe 924.
03:37.12obsidian-studiosManxPower:  sorry to hear that, glad you are ok
03:37.14ManxPowerDidn't support SIP
03:37.22ManxPowerobsidian-studios, that remains to be seen
03:37.32Hmm-homespent the whole night through, feels so go to be wiiiith you
03:37.37obsidian-studiosManxPower: ophelia was scaring me a bit, here in Jax, Fl but it went north
03:38.15obsidian-studiosManxPower:  where are you know? with friends/family? irc'ing from a shelter :)
03:38.43ManxPowerobsidian-studios, It's complicated.
03:38.55obsidian-studiosManxPower: I bet it is
03:39.39obsidian-studiosManxPower: I could not imagine what I would do, aside from freaking out
03:40.24obsidian-studioswith regard to AEL in *, is there anything special I need to do other than making sure the module is loaded, and I can slap in normal if statements and etc in extensions?
03:40.35jimlinuxAnyone here work with Cisco 7940s?
03:41.05ManxPowerobsidian-studios, README.ael
03:41.23ManxPowerOK, I got Charter Internet ordered.
03:42.00obsidian-studiosManxPower: I am reading the wiki, is there stuff in the readme not on the wiki? I do not have a README.ael?
03:42.02Hmm-homegoing to play some pool
03:42.06Hmm-homelater guys and gals
03:42.18obsidian-studiosHmm-home:  peace
03:43.26obsidian-studiosManxPower:  oh  I am not sure I like ael, I have to redo everything :(, but I guess if I needed allot of ifs and etc it would be come in handy
03:45.33ManxPowerobsidian-studios, /path/to/asterisk/docs is the reference you should turn to first
03:46.42obsidian-studiosManxPower: he he, I do not use the doc use flag on gentoo, so my stuff get's compiled and installed with no docs :)
03:52.50*** join/#asterisk bmg505 (n=leon@rndf-146-12-156.telkomadsl.co.za)
03:55.06andrew`i asked sixtel why it took months to get a response, they wrote back "Not sure about your inquiries. Let us know if there's anything else."   lol
03:56.11file[laptop]...haha
03:56.55kramhi file
03:57.05file[laptop]krammy boy!
03:57.37obsidian-studiosok is this expression correct? exten => _NXXNXXXXXX,1,GotoIf($[${EXTEN:-3} != 904]?3), where priority 2 will dial less the first 3 digits, and priority 3 will pad a 1?
03:58.01file[laptop]kram: how are you?
03:58.16krami'm alright, how about yourself
03:59.07kramyou need covers and snuggles, it sounds like
03:59.20file[laptop]I'm snuggly in bed
03:59.31file[laptop]muahaha!
03:59.55obsidian-studiosfile[laptop]: if you are cold a fat chick usually can help out with that :) lots to snuggle with :o
04:00.06file[laptop]you're evil
04:00.09bkw__file we have to finish that rate plan tommorow :P
04:00.11*** join/#asterisk mwgbc (n=junkmail@adsl-71-132-196-196.dsl.pltn13.pacbell.net)
04:00.20file[laptop]bkw__: which one
04:00.32*** part/#asterisk mwgbc (n=junkmail@adsl-71-132-196-196.dsl.pltn13.pacbell.net)
04:00.40halogen8trying to setup my firewall with NAT and asterisk  behind it.....what ports do I need to open besides 5060?
04:00.41jimlinuxSo, do I REALLY have to buy a maintenance contract with Cisco just to update my friggin' 7940 phone to SIP?
04:01.03file[laptop]halogen8: UDP 5060, UDP 10000, and setup externip and localnet in sip.conf
04:01.04obsidian-studioshalogen8: depends on what protocols you are using, any aside from sip?
04:01.29obsidian-studiosfile[laptop]: what's UDP 10000 used for?
04:01.36file[laptop]er UDP 10000-20000
04:01.46file[laptop]RTP, audio stream...
04:01.58obsidian-studiosjimlinux: that or find a local cisco rep and beg like I did for a client who just bought a 7960 ;)
04:02.04*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
04:02.21jimlinuxI can beg
04:02.24halogen8file[laptop]: can the external IP be my dyndns name?
04:02.24jimlinux(I'm married)
04:02.32obsidian-studiosjimlinux: but yes, from what I have been told per cisco unit or model, you need a smartnet contract or no firmware or support
04:02.37file[laptop]halogen8: yes
04:02.40jimlinuxugh
04:02.52halogen8obsidian-studios: I'm not really sure yet......i'm really new to this and just learning.....right now I'm just setting up IPKall with asterisk
04:02.56obsidian-studiosjimlinux: but with phones all you need is one per model
04:02.59jimlinuxobsidian: I could care less about support.  I just need the stupid bin fine
04:03.19halogen8obsidian-studios: ipkall gives free US DID's, but so far, I can get it to ring my asterisk
04:03.29obsidian-studiosjimlinux: yep I said the same thing to my rep ;) I was like so if someone buys a new phone with old firmware, they have to pay more for current or diff firmware :)
04:03.31jimlinuxobsidian-studios:  I just wanted to avoid having to pay a Cisco tax
04:03.58obsidian-studiosjimlinux: Cisco = $, and once you have the gear, Cisco = $$ :)
04:04.06*** join/#asterisk BoyGenius (n=ldvoipen@ip68-108-88-208.lv.lv.cox.net)
04:04.19obsidian-studiosjimlinux: but it's good stuff for the most part so? they got us by the short and skinny
04:04.39BoyGeniusello all, and good evening
04:04.56obsidian-studioshalogen8: well I was only curious because I was not successfully able to do remote sip, but I did not try to hard. udp port 10000 could have been my problem?
04:05.04jimlinuxobsidian-studios: damn them and their good hardware
04:05.21halogen8obsidian-studios: I thought it was UDP 10000-20000
04:05.39obsidian-studioshalogen8: ? no clue, I only opened up 5060?
04:07.11*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
04:07.34halogen8anyone in here familiar with IPKall and asterisk?
04:08.06VcoShort and skinny?
04:08.24obsidian-studiosVco: pubes :)
04:08.26MikeJ__chan_skinny?
04:08.34BoyGeniusLOL
04:08.37jimlinuxYou mean short and sccp
04:09.35Vcoy'know, i used to think MS SMS sucked .....
04:09.37file[laptop]sleepy time... hrm yes
04:10.07*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:10.10Vcothen i had to use CA Unicenter
04:10.47bkw__THERE SHE BLOWS!!!!
04:10.54bkw__Yet another pointless discussion on -dev
04:11.03halogen8file[laptop]: get my PM?
04:11.08tzangerooh ooh where
04:11.15bkw__asterisk stable one
04:11.18bkw__pointless
04:11.24bkw__why not just spend the time to fix it
04:11.29bkw__NEXT!!!
04:11.32*** join/#asterisk mmlj4 (n=jkelly@redfishnetworks.com)
04:11.47Vcofixing things is for suckers
04:11.53tzangerbkw__: do you not feel that digium is doing exactly that?  I mean look at the last dozen or two dozen cvs commits
04:11.55*** join/#asterisk MJR^ (n=n0c1@61.246.231.50)
04:12.01drumkillabkw__: another pointless comment in #asterisk
04:12.39drumkillayou were in that thread just like the rest of them
04:12.44mmlj4hey, um, in extensions.conf, how does * know to ignore the leaving digit "9" when dialing out?
04:13.15mmlj4s/leaving/leading/
04:13.26obsidian-studioscan someone confirm on my expression ? good bad? exten => _NXXNXXXXXX,1,GotoIf($[${EXTEN:-3} != 904]?3)
04:15.04MJR^somebody plese help me out ..... i get an warning message ....  it goes fine and calls get connected wen i forward it to n2p but now i am using a diff ip to forward and diff user and secret... wen i dial frm sj phon i get forbidden frm te phon and an warning msg in te asterisk
04:15.43Vcodid you just type that by mashing your forehead on the keyboard?
04:15.57*** join/#asterisk Hogie (n=hogie@c-24-0-74-98.hsd1.tx.comcast.net)
04:16.05MJR^Vco: ?
04:16.29Hogiehah
04:16.47Hogiesorry, Im just happy, just finished a 1.9 hour night flight of 3 hours I need for my PPL
04:16.48MJR^Vco: plzz help me i get te msg Sep 14 09:19:35 WARNING[3430]: chan_sip.c:695 retrans_pkt: Maximum retries excee                                              ded on call 2a70681628f969321907617a00608f9e@202.54.117.212 for seqno 102 (Criti                                              cal Request)
04:20.06obsidian-studiosyep, my expression sucks and does not work
04:23.53*** part/#asterisk Kyreeth (n=ashley@aquila.feathers.net)
04:24.16*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
04:26.30BhaalWKHey guys...  I have asterisk setup to talk between my ATA and my sip provider...  Where abouts in this scenerio is the ring tone being generated for an incoming call?
04:26.43*** part/#asterisk Borgon (n=L3orgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
04:27.21mostyon the phone/ata i believe
04:27.31BhaalWKAhhhh okey...
04:31.29obsidian-studiosexcellent, I got my expression squared away, and got the removal of local area code or 1 padded when a 10 digit # is entered :)
04:35.21BhaalWKmosty: I am talking about the ring tone that the person calling me hears...
04:35.35WhiteWolfyBhaalWK: you can't determine that
04:35.46BhaalWKWhiteWolfy: Where is that determined?
04:35.57WhiteWolfywell, probably by their provider
04:36.08WhiteWolfybut it's the device's job to interpert the ring # and make it
04:36.16WhiteWolfymost devices support a primary ring, and an alternate ring.
04:37.07BhaalWKWhiteWolfy: Im not interested in how my fone sounds when it rings...  Im interested in what people hear when they are calling me before I answer the fone...  I am in australia, and I would rather have them hearing an australian tone..
04:37.26BhaalWKThat way they wont be as confused...
04:37.30WhiteWolfyBhaalWK: are you talking about that little briing that it makes?
04:37.48WhiteWolfylike, while transfering or dialing?
04:38.20BhaalWKWhiteWolfy: Okey, in America, when you call someone, and your waiting for them to answer, the tones you hear "long pulse .. short pause .. long pulse .. short pause" etc
04:38.31WhiteWolfyok, i understand
04:38.58BhaalWKWhich end of the setup generates this?  My asterisk server?  My ATA?  Or my providers gear?
04:39.41file[laptop]by default a 180 Ringing (if we're talking SIP) is sent to your provider, and they generate it
04:39.57file[laptop]if progressinband is enabled a 183 Session Progress is sent and Asterisk generates the ringing
04:40.07WhiteWolfyBhaalWK, something relevant: http://voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf
04:40.41*** join/#asterisk Snake-Eyes (n=blog@203.201.98.247)
04:40.43BhaalWKWhiteWolfy: Ive gone over that and adjusted my indications.conf so that an australian ring should be heard by the calling party while I dont answer...
04:41.00WhiteWolfyBhaalWK: you're sending a 180 ringing, it's their client.
04:41.14WhiteWolfyif that had no effect.
04:41.18MJR^somebody plese help me out ..... i get an warning message ....  it goes fine and calls get connected wen i forward it to n2p but now i am using a diff ip to forward and diff user and secret... wen i dial frm sj phon i get forbidden frm te phon and an warning msg in te asterisk
04:41.25MJR^Vco: plzz help me i get te msg Sep 14 09:19:35 WARNING[3430]: chan_sip.c:695 retrans_pkt: Maximum retries excee                                              ded on call 2a70681628f969321907617a00608f9e@202.54.117.212 for seqno 102 (Criti                                              cal Request)
04:42.10BhaalWKWhiteWolfy: right, thanks...
04:42.17mostybhallwk: i think that would be your asterisk machine producing that sound, but the sip provider could also put a ringing sound on the line before it gets that far
04:42.32MJR^<PROTECTED>
04:42.45WhiteWolfyBhaalWK: humm, i've never had anyone complain when dialing my systems.
04:43.12WhiteWolfyi think people ignore ringing.
04:43.18BhaalWKSo I need to lodge a support ticket to get them to adjust their gear to sound more australian :) Can just picture my providers pstn termination saying "here, turn this shrimp over while I get the person your trying to call"
04:43.43BhaalWKWhiteWolfy: I havent had them complain *shrug* I guess Im just unique, thinking it might confuse people...
04:43.44JerJerso does zaptel not like gcc version 3.3.5 (Debian 1:3.3.5-13)?
04:43.54JerJer-head of course
04:43.59WhiteWolfyBhaalWK: i've never had someone say to me (From euro or .au), weird ringing.
04:44.00HogieJERJER, ITS ALL YOUR FAULT
04:44.10WhiteWolfyBhaalWK: because, i think everyone has heard every type of ringing at some point
04:44.22WhiteWolfybe it on a movie, or on tv
04:44.22BhaalWKWhiteWolfy: true true...  Maybe I leave it *shrug*
04:44.46WhiteWolfyif you're really that concerned, you could always instantly answer to MoH :P
04:44.49BhaalWKBut I do need to bug them and find out whats going on with caller id...
04:44.52WhiteWolfystick them in a queue.
04:45.18WhiteWolfyheh, that'd be funny. record a bit of the ring and have it ring over and over :P
04:45.50obsidian-studiosHogie: I still ain't huggin ya
04:46.02WhiteWolfywe love you.
04:46.21Hogieheh
04:47.14HogieI had my first "emergency" tonight in the air Jerjer, lol
04:50.38JerJerit seems zaptel has issues with gcc 3.3.5
04:50.57JerJeror something is seriously broken here
04:51.03JerJerthen again asterisk compiles fine
04:51.07Hogieit was oh so important, the passenger side window latch broke on climb out, and instructor made me turn around
04:52.46*** join/#asterisk trig_hm (n=jb@home.monkeypr0n.org)
04:53.11*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
04:57.42JerJersweet
04:57.58JerJergotta start somewhere
04:58.37HogieI thought it was nice that Tower asked if I needed assistance since I was changing from a south departure to staying in the pattern to do a full stop
04:59.24JerJershoulda declared an emergency - then you could have went thru the NTSB paperwork   :)
05:00.17WhiteWolfyyeah, that's wonderful to do :P
05:00.27Hogiehah, I think what made him ask me that though is that I asked him to repeat, because the wind was hitting her mike, and I hadn't turned up com1 yet, so I couldn't make him out well
05:00.48*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
05:01.28JerJeryeah
05:01.35WhiteWolfyi think i should pick up flying, but i'm too big for a small plane
05:01.45JerJeryou can't be any bigger than me
05:01.46HogieWhiteWolfy: Im 339lbs
05:01.55WhiteWolfyyou're probably not as tall as me.
05:01.55JerJeri'm 6 foot 5 - 350+ lbs
05:02.04HogieIm 6'4
05:02.10WhiteWolfyyeah, i've got you both beat pretty good.
05:02.12JerJerjust train in a C172
05:02.19Hogieahha, a C140;)
05:02.23WhiteWolfy6'11" - 335
05:02.28Hogieoh wow
05:02.29Qwellumm
05:02.33Qwellbeast
05:02.34Hogieyou wouldn't even need to move your seat forward
05:02.35Hogie;)
05:02.36Qwell:p
05:02.37JerJerdunk dunk
05:02.52WhiteWolfyi think the passengers would be freaked out
05:02.58WhiteWolfylike i'm a monster or something
05:03.06Hogiewhat passengers?
05:03.09Hogieyour flight bag?
05:03.10Hogie:P
05:03.26WhiteWolfyprobably right, i'd never fly anyone
05:04.10WhiteWolfywe have these small little planes that fly over our house every day, we have an Exec. Airport near here
05:04.56HogieI wonder if its true that google just bought a new company jet
05:05.04WhiteWolfywho knows?
05:05.10WhiteWolfygoogle has more money than they care to share
05:05.17HogieI heard they bought a 747
05:05.19Hogierecently
05:05.20Hogielol
05:05.57WhiteWolfygoogle could buy a fleet of airplanes
05:06.01*** join/#asterisk zoo (i=nobody@ip-117-16.travedsl.de)
05:06.12*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
05:06.21harryvvevening all
05:06.28Math`Google Airlines?
05:06.29harryvvjust got back from the oregon coast.
05:06.33WhiteWolfyMath`: scary thought
05:06.36Math`yeah
05:06.42WhiteWolfyi think i'm ready for Google Pay though
05:06.45*** join/#asterisk Rowter (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx)
05:06.47Rowterpbx_spool.c:307 scan_service: Unable to
05:06.47Rowteropen /var/spool/asterisk/outgoing/1978.voice.out: Permission denied,
05:06.54Hogieif I ever buy a plane, Im going to try my best to get the registration number changed to Hotel Zero Golf One Echo
05:06.58Rowterwhat it means, I already set permisions to file.. let me see.
05:07.00Math`WhiteWolfy: what about Google Wireless? :P
05:07.09Math`sounds well :P
05:07.11WhiteWolfyMath`: i'm happy with my service.
05:07.18harryvvAnyone following this?
05:07.21harryvvhttp://www.eweek.com/article2/0,1895,1857523,00.asp
05:07.48WhiteWolfyHogie: i'll buy you a B52
05:08.00HogieI dont want a B52
05:08.04HogieI want a Mooney M20
05:08.15harryvvThis is about the vsas that are being assembled to provide voip service to all of the red cross shelters.
05:08.20Hogiesomething I can handle by myself, and cruise at 200kts
05:09.12WhiteWolfyHogie: but, i'll load it for you.
05:09.18BoyGeniusg'night all
05:09.36JerJerRowter: aparently not
05:09.49JerJerapparently
05:10.18RowterJerJer, its chown asterisk:root and chmod 777 strange mmh
05:10.29WhiteWolfyi have a question for you two, how big is the sky, really?
05:10.37JerJeris asterisk running as user 'asterisk' ?
05:10.41*** part/#asterisk BoyGenius (n=ldvoipen@ip68-108-88-208.lv.lv.cox.net)
05:10.58Math`how fast is a VSAT?
05:11.02JerJerand does the directory have write access for the user 'asterisk'
05:11.06JerJerthat fast
05:11.20JerJerhowever, first hop latency is horrid
05:11.36JerJerbest i've seen is around 175ms
05:11.48WhiteWolfynot too bad when considering an emergency situation
05:11.56wasimin reality its closer to 500ms
05:11.57RowterJerJer: yeah
05:11.57WhiteWolfyJerJer: sat. phones aren't too much better either
05:12.08Math`WhiteWolfy: well.... how much traffic can it handle
05:12.08JerJerno they sure aren't
05:12.23JerJerdepends what your provider gives you
05:12.23WhiteWolfyMath`: enough for the voice traffic?
05:12.30WhiteWolfyt-mobile should've stepped it up though
05:12.41WhiteWolfythey had the capacity to increase the available towers... they all ran away though
05:13.04harryvvanyone here might know why my spa 1000 is blinking 2 short blinks then one long one?
05:13.21WhiteWolfyharryvv: the manual will say
05:13.22Math`VSAT can go up to 2mbps full duplex
05:13.35wasimMath`: and beyond
05:13.36WhiteWolfyat 500ms... heh heh
05:13.40harryvvWhite...actuyally the manual would not. need to look it up on there site.
05:13.48Math`at 500ms... heh
05:13.48WhiteWolfyharryvv: yes, it would :P
05:13.49harryvvand that site is long.
05:14.10Math`well vsat-systems(tm) are offering 2mbps full-duplex at 5 000$US/month
05:14.18WhiteWolfythat's about right
05:14.32harryvv5,000 per month?
05:14.45WhiteWolfymhum
05:14.46JerJertalk about corncob batman
05:14.47Math`yeah, but thats the most expensive plan, with 25 gig of up & down
05:14.52Qwellumm
05:14.56pauldyanyof you have experience with fios yet?
05:15.04Math`u can get 1mbps, 5gb down, 5gb up for 1000$/m
05:15.04WhiteWolfypauldy: yeah, it's nice
05:15.25harryvvAnother option would be a large aerostat with a wifi repeater link.
05:15.27pauldyis it? how reliable has it been for you and what service leve do you have
05:15.33QwellMath`: are you in the middle of the sahara or something?
05:15.45WhiteWolfypauldy: hasn't gone down yet where i've used it
05:15.53Hogieno, prob just New Orleans
05:15.55WhiteWolfypauldy: fairly close to the 15mbit/2 that's offered
05:16.00Math`Qwell: no, we're talking about the red cross using VSAT for ip telephony in new orleans
05:16.15pauldyexcelent, the trucks have been outside my house for alomst 6 months now
05:16.21wasimMath`: they'll hate it, just like they hate it in afghanistan
05:16.23QwellYou could buy quite a bit of wifi equipment for that price
05:16.25harryvvmath, I suspect the sats are Leos?
05:16.31WhiteWolfypauldy: 14.5mbit // 1.98 mbit
05:16.42pauldyI've had comcast and just moved over to dsl becuase comcast coulndn't keep their equipment up
05:16.50Math`WhiteWolfy: is that the technology's limit?
05:17.00WhiteWolfyMath`: no, they run full OC3's into the house
05:17.08WhiteWolfyit's limited at the device.
05:17.09harryvvQwell, problem is the large amount of distruction is daunting. wifi is only good for local link ups unless you aim it to say a aerostat.
05:17.37pauldy30Mbs/5Mbs seems to be the fastest they offer
05:17.48Qwellpauldy: thats about $200 a month...
05:17.57Qwell4x as much as the 15/2 plan
05:17.59Math`I mean, how fast can a complete satellite relay traffic
05:18.06pauldyQwell, select markets have shown a rate of 59 or 69
05:18.11WhiteWolfypauldy: i can ask them for more.
05:18.13Qwelloh
05:18.29pauldyI know WhiteWolfy
05:18.36WhiteWolfypauldy: verizon+me = love
05:18.37pauldyjust wondeirng about the experience
05:18.49WhiteWolfypauldy: i asked about how much 100mbit/100mbit is if i coarsed them
05:18.50harryvvMath` do the math
05:18.53WhiteWolfythey said $3,295/mth
05:19.05pauldyhahaha
05:19.15Math`harryvv: I dont have enought data to compute that
05:19.17WhiteWolfywhich isn't bad really
05:19.20WhiteWolfythat's about right
05:19.32pauldyno it isn't bad but I would rather be paying that for a house payment than internet service
05:19.35harryvvMath` whats the max bandwith vstat can up/downlink?
05:19.39Math`yeah
05:19.49Math`that is, the physical limitations
05:20.08pauldyI can't imagine they are pulling all they can out of fiber yet
05:20.18harryvv2 mbps
05:20.22pauldyI mean they don't even use multiple frequencies yet do they?
05:20.29harryvv20 calls at once.
05:20.32WhiteWolfypauldy: hah, i'd probably start a small datacenter
05:20.41WhiteWolfyharryvv: more than 20
05:20.51WhiteWolfymore like 30
05:20.55manybandwith doesnt matter when the latency is poof.
05:21.13pauldyWhiteWolfy, they still sue pppoe with it?
05:21.19Math`Each VSAT contends for time slots on a shared (TDMA) inbound channel.
05:21.20Math`heh
05:21.21pauldys/sue/use
05:21.28wasimMath`: VSAT is a very generic term ... you also need to see what type it is, TDMA, SCPC, MCPC, PAMA/DAMA etc
05:21.29WhiteWolfypauldy: no, it's not required
05:21.33WhiteWolfyonly on legacy equipment
05:21.35Math`oh ok
05:21.36pauldyreally
05:21.41pauldyoh now thats good news
05:21.50pauldyI bet the latency is low low without it
05:22.12WhiteWolfypauldy: <2ms to google
05:22.19Qwell...
05:22.22manyvia satelite?
05:22.23pauldynice
05:22.23Math`well... latency has no choice to be high while youre transmitting radio waves to satellites
05:22.29Math`WhiteWolfy: via sat?
05:22.34WhiteWolfyMath`: no
05:22.42harryvvWhite, thats still good. now no way all 30 lines will be bussy at once so double it.
05:22.58WhiteWolfyMath`, Qwell, many: i've been talking about fios to pauldy
05:23.13pauldyyea fiber internet service
05:23.23manyoh. sorry. :)
05:23.25niZoni want fios
05:23.30Math`yeah the only thing is... you pay the fiber to the NOC of your ISP
05:23.34Math`which is expensive
05:23.39Math`(well thats in here)
05:23.46niZonthere's only one company here that is running fiber
05:23.47WhiteWolfyMath`: verizon digs it to the curb
05:23.49niZonbut they won't use it
05:23.51pauldyI can't wait now,  I wonder how long it will take them to start actually moving it into the townhomes here
05:24.07Hogieverizon fios isn't that great from my experience @ my parent's
05:24.23pauldyHogie, really what problems?
05:24.25WhiteWolfythen they should complain, my internet experience is the best
05:24.30HogieI moved them back to earthlink DSL, it was so bad, couldn't even stay on IRC for over 2 minutes
05:24.34Hogiethe bandwidth was great
05:24.49Hogiebut not long FTP's, or anything like that
05:24.54pauldyHogie, do you know what the problems where exactly?
05:25.03WhiteWolfyi'm surprised
05:25.06WhiteWolfyi had no such issues
05:25.58HogieWell, I have an openbsd server @ my parent's house running as a nat gateway for their machines (my dad has one and my mom has one), and so I just plugged it in and set it up like I needed too, even when I installed their crap to my dad's machine to see if it was the obsd box, it kept doing it
05:26.24Hogiemy dad irc's to undernet to get ebooks...  so he was freaking out when he couldn't stay on irc for over a few minutes
05:26.44Qwellheh, parents who pirate
05:26.46Hogiessh's would get broken, it was just... I dunno
05:27.14WhiteWolfyweird
05:27.17WhiteWolfyi had no such issues at all
05:27.18pauldyso you just lost connection? was the connection dropping and not resuming correctly? Where you using pppoe?
05:27.59Hogieit was static ips, and it showed up as a ping time out on the ircd...
05:27.59*** join/#asterisk MJR^ (n=n0c1@61.246.231.50)
05:28.03MJR^WARNING[3500]: chan_sip.c:1401 create_addr: No such host: vsnlpins9@vsnl.net
05:28.11MJR^some help plzzzzzzzzzz
05:28.33harryvvhttp://www.blackberry.com/products/blackberry7200/blackberry7270.shtml
05:28.38harryvvanyone here try it yet?
05:28.50Hogieheck, Ive seen the VP of Network Ops @ The Planet doing the same thing to machines in their own DC (on irc)
05:28.51twistedMJR^, uh, duh.  that's not a valid host
05:28.55pauldyWierd Hoggie maybe some type of icmp filtering?
05:29.14Hogiepauldy: it wasn't on the obsd box, because I tried removing it and it still happened
05:29.22MJR^that means the forwarding ip is wrong
05:29.27HogieI just know my parents are happier on verizon copper & earthlink dsl
05:29.32twistedMJR^, no, look at the thing.
05:29.42twistedvsnlpins9@vsnl.net is not a valid host
05:29.51twistedvsnl.net is a valid host
05:29.56MJR^k
05:30.00pauldyHogie any issue with normal surfing?
05:30.01twistedvsnlpins9.vsnl.net would be a valid host
05:30.06twistedbut not an email address
05:30.10pauldyOr just with IRC
05:30.12Hogiepauldy: no, not unless you tried a long download
05:30.20Hogiemost http traffic is over in seconds...
05:30.26twistedharryvv, buy me one, and i'll be glad to try it ;)
05:30.48Hogiebut when I'd try to get an iso off say, a centos mirror, it wouldn't come
05:32.20HogieI dont remember if comcast reboots their network on monday or tuesday nights here, grrr, I hope this isnt when they do it
05:32.39Hogieguess not
05:32.40Hogielol
05:32.44pauldyI just left the comcrap network
05:32.59Hogieits been fine for me, plus Im getting it for $20/month
05:33.08Hogieand I dont want a phone line, so it doesn't matter
05:33.11harryvvcomcast is advertising there voip service in either seattle or portland.
05:33.51pauldyI was too but canceled because they have no customer service and however services our area doesn't know anything about what he is doing except how to screw cables together
05:34.19pauldykept leaving cables to other units that did not have cable with their inner pins touching the nice metal grounded case
05:34.24Hogiehah, on New years eve, I came home at about 10pm to find someone had broken into the cable box, and stolen MY port
05:34.39Hogiethey said a tech would be out Sunday (that was a friday night btw), so I was okay, I'll wait
05:34.42Hogiehe never showed up
05:34.43*** part/#asterisk Uberbot (n=Uberbot@69.252.219.76)
05:35.04HogieI ended up going and patching myself back in, and using 2ton epoxy (what I use when building RC airplanes), to close the box back up
05:35.14Hogieomg, was the tech pissed on monday when he showed up
05:35.57WhiteWolfylol
05:35.57WhiteWolfywhy?
05:36.05Himekocouldn't get it open
05:36.06harryvvWe need to orginized topics of discussion concerning voip and have them in seperate channel.
05:36.07Hogiehe couldn't get the box back open
05:36.21WhiteWolfywell, sucks to be him
05:36.24WhiteWolfynot your problem
05:36.26harryvvA month worth of topic of discussion.
05:36.39twistedHogie, i worked for comcast in TN, they never rebooted shit there
05:36.46WhiteWolfyharryvv: the 7270 is so worthless
05:36.52Hogietwisted: its something, it happens EVERY week at the same time
05:36.57HogieI think its monday nights at 12:30
05:37.08Hogieall my connections die at the same time, then it comes right back up
05:37.09twistedHogie, did you call them?
05:37.15Hogietwisted: its down 30 seconds
05:37.15harryvvWhiteWolfy ohh and why is that?
05:37.16twistedor better yet
05:37.18MJR^twisted now a new msg WARNING[3542]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 691d599d1fbf1ad5781bd45d38638993@202.54.117.212 for seqno 102 (Critical Request)
05:37.18MJR^<PROTECTED>
05:37.25WhiteWolfyharryvv: worthless, just buy cell phones
05:37.29twistedMJR^, oops.
05:37.39MJR^what may be reason now
05:37.41twistedHogie, i think comcast's DHCP lease lasts 1 week exactly
05:37.51harryvvWhiteWolfy hehe well cannot sell cell service :)
05:37.52twistedHogie, could be renewal ;)
05:38.04Hogietwisted: Ive had same IP for 6 months now...
05:38.12WhiteWolfyHogie: it still has to renew.
05:38.13twistedHogie, renewal does not mean getting a different ip
05:38.23*** part/#asterisk zobia (n=laura_sh@218.6.242.212)
05:38.25twistedi've  had the same IP for about 8 months now
05:38.30Math`Hogie: the DHCP client tells the server the last ip it had, if the ip is available, your gonna get it
05:38.32Hogiesec, let me see what time dhcpcd has for the lease
05:38.47Hogieright, but if it doesn't change the ip, why should it drop all the traffic?
05:38.51twistedbut my router still renews it every 240 hrs
05:38.56MJR^what may be the reason for those warnings
05:39.03MJR^WARNING[3542]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 691d599d1fbf1ad5781bd45d38638993@202.54.117.212 for seqno 102 (Critical Request)
05:39.04MJR^<PROTECTED>
05:39.11twistedMJR^, many reasons.
05:39.24MJR^can u name them plz
05:39.25twistedMJR^, have you checekd the wiki/google?
05:39.30MJR^no
05:39.32twisted~wiki
05:39.33WhiteWolfythen do it
05:39.39twistedoops
05:39.43twistedhaha, jbot spammed me
05:39.44WhiteWolfyhttp://voip-info.org
05:39.50twistedhttp://www.voip-info.org <- wiki
05:39.56WhiteWolfytoo late :P
05:40.04MJR^k i go there
05:40.08WhiteWolfygood idear
05:40.13Hogie<PROTECTED>
05:40.23WhiteWolfyyou don't choose your own lease time
05:40.27WhiteWolfythey do, heh heh
05:40.31Hogiethat's in the dhclient.leases
05:40.37WhiteWolfyah
05:40.39Math`heh
05:40.54WhiteWolfywell, that sounds about 1 week
05:40.56Hogiethat's 4 days, unless my math is off
05:41.25WhiteWolfyyeah, 4 days
05:41.32Hogieand my expire time is set for 2:26am
05:41.44WhiteWolfyhogie: time warner does the same thing to me
05:42.04pauldyforce a renew
05:42.05WhiteWolfyevery monday i get a very short service interuption
05:42.22twistedyeah, that's 4 days
05:42.23WhiteWolfybut, my nat keeps the connections alive using aggressive tatics
05:42.31twistedi ran the math again too
05:42.32MJR^twisted: i was forwarding calls to n2p and it was working fine now i am forwarding to vsnl and i am not able to place te call ,
05:42.42twistedMJR^, sorry to hear that.
05:42.56WhiteWolfyMJR^: take more time with your questions and notes please
05:42.57MJR^ip is 202.54.117.212
05:43.06twistedgood to know
05:43.12twistedCOMMENCE SYN FLOOD!
05:43.17twistedj/k :)
05:43.23Math`uhu
05:43.23WhiteWolfysyn flood is so 2000 ;)
05:43.24HogieI am gonna try to keep my ip as static as possible, I have ipsec to my office (which carries my sip traffic to the asterisk box), and I dont really want to have to redo isakmpd config files
05:43.47twistedHogie, just use ppp and chap/pap
05:44.04twisted;)
05:44.12twistedoooh
05:44.16twistedfound out somethign cool about my cable here
05:44.17harryvvhogie so use alot of asterisk voip connections at the office?
05:44.20twistedi can ping my set top box
05:44.23HogieI'll stick with ipsec, we have it between offices
05:44.27twistedand nmapping it reveals a few service ports ;)
05:44.55twistedand if i ping flood it, my picture stops
05:44.56twisted;)
05:45.08WhiteWolfywhen did this channel go registered only?
05:45.10harryvvtwisted truing to hack it?
05:45.11Hogieharryvv: we have 2 * boxes, in 2 offices, one office has 30 cisco phones local, plus 2 barbietones remote (ones at my house), while the other office has 6 local cisco's...  nothing great
05:45.19QwellWhiteWolfy: 9 months ago at least
05:45.21twistedharryvv, no way! that would be illegal!
05:45.28WhiteWolfyQwell: how did i sneak in here?
05:45.28WhiteWolfylol
05:45.48twistedWhiteWolfy, I did it last night due to attack of the clone bots part deux
05:45.53WhiteWolfyahh
05:45.55WhiteWolfyroger that
05:45.59WhiteWolfydon't kick me :(
05:46.02harryvvhogie, how long as these been setup for and how reliable is the conenction direct to both those offices?
05:46.04twistedi'm not kicking
05:46.04Qwelltwisted: fone et al?
05:46.15WhiteWolfytwisted: i know, but i dont want to register today
05:46.23WhiteWolfyharryvv: i use VoIP over a VPN every day
05:46.30twistedyou just need to be nickserv registered unless you want to hang out in #asterisk-unregistered with the other folks who can't read
05:46.32harryvvmm
05:46.40Math`lol
05:46.46harryvvWhiteWolfy that is about the most secure way to do it eight?
05:46.53WhiteWolfymost secure? i suppose
05:47.15Hogieharryvv: remote office is on Speakeasy DSL (lowest one), they RDP into our terminal server at the main office too over the link...  They just iax2 to us for intercom, or incase their local lines are down
05:47.18harryvvI have never setup my linux box with vpn but I guess it would not be to hard.
05:47.35HogieI dont think they have ever had a failed call go through over the iax2, except when the main pbx box isn't on at our office
05:47.55harryvvhogie good.
05:47.56harryvv:)
05:48.06Hogielike last wednesday, we had a 6 hour power outage... on my day off, and someone didn't move everything to generator right, so we had 45 minutes of downtime after the UPS's ran out
05:48.08QwellSo...I used a Nortel Meridian system for the first time today
05:48.09Qwellits crap
05:48.12WhiteWolfyOpenVPN is probably the easiest VPN server i've ever setup
05:48.27harryvvhogie, use relays next time
05:48.28harryvv:)
05:48.31Math`I use pptpd
05:48.35Hogierelays?
05:48.37Math`thats on linux right?
05:48.38QwellI actually enjoy Avaya compared to that POS
05:49.01harryvvhogie, if the power went out..then the relays would click over to generator and genset would start up.
05:49.02QwellMath`: "poptop" is cross-platform, isn't it?
05:49.04*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
05:49.16Hogieharryvv: transfer switch;)
05:49.19WhiteWolfyQwell: PPTP is originally MS'es anyway
05:49.22Math`I dont know poptop
05:49.24Qwellis it?
05:49.26Math`PPTP is PPP over TCP
05:49.26harryvvtransfer relays.
05:49.27harryvv:)
05:49.29Hogiewe are getting 2 installed, for our new generators, but they aren't auto
05:49.31QwellMath`: pptp == poptop, I believe
05:49.34Math`oh ok
05:49.35Qwell"poptop"
05:49.38WhiteWolfyQwell: yeah, it is, microsoft made it.
05:49.47twistedHogie, just do what I did
05:49.52twistedHogie, buy an 8hr backup
05:49.59Qwelltwisted runs voip over tin can
05:50.10QwellVoTC
05:50.12Hogieto do that, I'd need 240v run into the server room
05:50.18WhiteWolfyi still use the pony express.
05:50.20Hogiefor a UPS big enough
05:50.22harryvv240 volts is best
05:50.31twistedHogie, why?
05:50.31Qwellouch, what was that for?
05:50.37twistedi have an 8hr backup running off of 110
05:50.44Hogietwisted: for how many machines?
05:50.50twistedHogie, right now only 4
05:51.04twistedbut they're power hogs
05:51.06harryvvI have a 125 amp hour battery. Wonder what a small voip service would last on that one battery.
05:51.14Hogiewe have 15 machines + a cisco cat w/inline pwr
05:51.22WhiteWolfyharryvv is seeing $$'s
05:51.38harryvvWhiteWolfy seeing dollars?
05:51.38twistedput step up transformers in the line and power everything off of a couple copper pots loops
05:51.47Hogiehah
05:51.52WhiteWolfyharryvv: yeah, you seem to only be concerned with selling it.
05:52.14harryvvWhiteWolfy no im interested in calculating amp hours.
05:52.49harryvvI have a trojan super deep cycle battery. Would like to obtain one or two more.
05:53.20WhiteWolfyAH = SomeValue * 20
05:53.54Math`I know how to calculate Watt-Hours, but I never understood the VA thingy in the UPS
05:54.00Math`how the hell do u convert VA to W-h
05:54.02Hogiehttp://gallery.cyberjunky.net/power_outage_sept_7_2005/P0018433 <-- what happens when someone makes the power lines hit your metal building
05:54.22twistedi have a trojan XL that hasn't been used
05:54.25QwellWhat is it, A * V = W?
05:54.28twistedi need to get a few more
05:54.30HogieA * V = W
05:54.31Math`yeah
05:54.37harryvvIm getting tired of this forest fire smoke
05:54.38Math`but VA isnt Watt
05:54.53twistedharryvv, so put it out!
05:55.03Math`well.... its supposed to be an energy quatity, Watt is a power mesurement
05:55.06Math`quantity*
05:55.20harryvvBurns Bog is on fire..5 miles from here and martin water bombers and Sikorsky sky crains and a-star helicopters were putting it out.
05:55.36harryvvits been on fire for 3 days now.
05:55.58andrew`my sister lives right by there
05:56.06twistedHogie, that's pretty
05:56.16*** join/#asterisk greendisease (n=jack@fedora/greendisease)
05:56.35Hogieyeah, happened at 11, at 11:05, I get a call "our powers out, and there's a fire, and it looks like we wont have power for a few hours, and can you come in?"
05:56.39QwellMath`: Are you trying to figure out what VA is, or what?
05:56.40harryvvandrew` where?
05:56.49andrew`the fire
05:57.04WhiteWolfywhich one
05:57.04Math`Qwell: Im trying to figure out, when I buy a UPS with 500VA written on it, how many watt-hours worth of energy can it give
05:57.11harryvvandrew` we are talking about burns bog in delta bc?
05:57.15WhiteWolfy500VA is typically 7AH
05:57.18Qwellwell, VA = Volt Amps, which is V(A)?
05:57.21twistedhey Math` : http://www.gordonengland.co.uk/conversion/power.htm
05:57.22andrew`yes
05:57.26QwellV(A) == V * A == W
05:57.26harryvvohhh
05:57.33Qwellno?
05:57.34pauldyVA is volt amps it takes into acount phase changes with capacitance and inductance of a circuit
05:57.35Math`yeah but W != Wh
05:57.37harryvvandrew` do your canadian also?
05:57.44Hogieat 11:13, our ISP calls my cell phone (which is in my emergency procedure for all our colo servers), and tell me that our router isn't responding
05:57.46andrew`yeah, non-practicing
05:57.48QwellMath`: 1 watt for 1 hour is 1 Wh
05:57.52Qwellright?  heh
05:57.57Math`thats right
05:57.59harryvvandrew` hehe
05:57.59Math`but in a UPS
05:58.02QwellI am most certainly not an electrical engineer
05:58.07WhiteWolfyMath`: 500VA is typically 7AH!
05:58.11harryvvandrew` i have a feeling it was arson.
05:58.14pauldyI took classes to be an EE
05:58.15Math`7AH at which voltage
05:58.18andrew`that's terrible
05:58.20Qwell110?
05:58.21WhiteWolfyMath`: 110
05:58.26Math`and how did you compute 7AH?
05:58.31WhiteWolfyMath`: i just know it.
05:58.37harryvvandrew` read up on it yet? its on www.vancouversun.com
05:58.39WhiteWolfyi have like 10 of them sitting upstairs
05:58.43Math`there must be some kind of formula somewhere
05:58.50QwellYou can't directly convert VA to Wh, can you?
05:58.55WhiteWolfyno
05:58.57QwellYou need to know how many watts your devices use
05:59.04WhiteWolfyyeah.
05:59.05Qwellso, if you're att say 500W
05:59.11QwellThats like...  1Wh?
05:59.12harryvvamp and ohms law
05:59.14andrew`looks terrible
05:59.26pauldyMath the formulas must take into acount the phase angle to compute the VAs you can take their polar form and compute the amount of "power"
05:59.37harryvvE/I*R
05:59.44twistedokay you guys are making my brain asplode
05:59.47*** join/#asterisk grimse (n=grimse@p5481D57F.dip.t-dialin.net)
05:59.49Math`Qwell: Wh is a mesurement of how much energy you took, if you use 5W for 1h, you took 5Wh of energy
05:59.56harryvvP=E*I
05:59.59Qwelltwisted: I just about expended all of my knowledge about electricity...don't feel bad
06:00.01Math`500VA would give 500W, but for how much time?
06:00.11Qwellone hour
06:00.18Math`its always for 1h?
06:00.19WhiteWolfythat's not true
06:00.24Qwelldunno
06:00.27pauldyAC circuits are a little complicated because as they occilate with capacitance and inductance they can build some interesting characteristics
06:00.39WhiteWolfya 500VA battery can't run a 500W load for one hour
06:01.09Math`cuz 7AH * 110V = 770Wh
06:01.35Math`(that doesnt mean it can handle a 770Watts charge at all)
06:03.02*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
06:03.54twistedhahahah
06:03.57twistedfamily guy is on
06:04.01twistedit's they y2k episode
06:04.05twistedcomedy central
06:04.37Math`VA is Vrms * A
06:04.47Math`which is different from W which is W = V * A
06:05.52harryvvwell, the burns bog fire is on seattle king5 news
06:07.10pauldymath VRms is calculated by taking the area under the curve for an alternating wave form
06:07.19pauldyit may not be exactly sinusoidal
06:07.38pauldyso the old 2^.5/2 may not work
06:08.12Math`Vrms is, very approximatively, 0.7*V
06:08.37pauldy.707
06:09.06pauldytake the square root of two and divide it by two and tell me what you get
06:09.12Math`that is why I said very approximatively
06:09.27pauldybut that only works for sinusoidal wave forms
06:09.30Math`0.707 :)
06:09.48pauldyvery often the wave forms from a UPS may be more squared
06:09.54Math`oh
06:10.30harryvvtalking about RMS?
06:10.34harryvvroot mean squared?
06:10.34Math`yeah
06:10.39Math`root mean square yeah
06:10.53pauldyyup
06:11.05harryvvpeople look at me and I say the typical wall outlet puts out over 300 volts Peak to Peak voltage :)
06:11.13harryvvLook at me strange :)
06:11.17Math`lol
06:11.33Math`isnt it 220?
06:11.36harryvv2 years of heavy avoinics teaches alot :)
06:11.45Math`well... depends where u leafve I guess
06:11.47pauldyI was trying to explain to him that much of what you will fidn for introductory to AC circuits does not fully explain the phenomina he is looking at and it might cause problems trying to reverse engineer what is going on there
06:11.53harryvvMath` I dont recall but do the math..i may be off.
06:12.20Math`I like electricity and electronics as a hobby so I may be off too
06:12.21pauldyMath`, Vpp for US to the home is 220
06:12.39Math`ok
06:12.41pauldyto get 110 they tie one wire to ground an the other wire is live from the power company
06:13.08harryvvadam savage got 20kv on myth busters
06:13.09harryvvheheh
06:13.14*** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
06:13.26pauldyreally
06:13.29harryvvyea
06:13.32pauldyI bet that left him tingly
06:14.03harryvvmade a fake ark of the covenenat and charged it with 20k fence transformer.
06:14.23harryvvhe lays his hands on the angles and says "now this thing is not charged is it??" ZAP
06:14.39harryvvhe really jumped and was well to say..shocked
06:14.40harryvv:)
06:14.42Math`lol
06:15.12harryvvim off to bed.
06:15.14pauldywelp gota reboot and see if I can fix my now corrupted FS
06:15.17Math`little * question now, how do I setup a context to be called when a voicemail message is left
06:15.19pauldybaghhh
06:15.24Math`(or even an AGI app)
06:18.24*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
06:19.07*** join/#asterisk razu (n=razu@fw.voicenet.ee)
06:19.42niZonif you're going to call an extension when someone leaves a voicemail, why not just direct the caller there before going to VM?
06:22.38*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
06:25.04*** join/#asterisk oej (n=Olle@212.247.206.4)
06:25.13*** join/#asterisk newl (n=newlook@203-59-184-194.dyn.iinet.net.au)
06:28.56hugo-v6gd morning pals
06:31.43*** join/#asterisk jaike (n=a@203.131.137.76)
06:32.36jaikeis it true that dtmf wont work properly with g729? or is there  a fix for this already
06:33.42wasimjaike: it won't work inband, it does out of it
06:34.13jaikewere using rfc2833
06:34.29wasimthat should be fine
06:34.48wasimthen its no longer a codec issue
06:35.00wasimand your protocol is supposed to handle it
06:35.54jaikeok thanks...just wanted to clear that up
06:36.54Math`uhm DTMF doesnt work well if Im not using rfc2833 with g729
06:36.58Math`well it doesnt work
06:37.04Math`(lets say I use inband or smth)
06:37.21Qwellyou can only use inband with ulaw, I believe
06:37.21Math`but I saw smth refering to dtmf on bugzilla, maybe I should cvs update -dP
06:37.22Qwellmaybe one other
06:37.45*** join/#asterisk Thumann (n=Thumann@217.157.30.66)
06:37.54ThumannGood morning all
06:37.54jaikewell be buying a lot of g729 licenses now
06:38.08jaikepretty satisfied with the quality
06:38.24Math`thats what my SIP-PSTN provider uses
06:38.25Math`its pretty nice
06:38.57jaikeno one can seem to differentiate if were using ulaw or g729
06:39.09QwellWhats a damn cheap provider in AU?  Call quality isn't an issue
06:39.17Thumannhmm.. i need to dial a sip extension on a Samsung OfficeServ 500M from asterisk... anyone know how this can be done ?
06:39.37Math`jaike: except from the bandwidth?
06:40.41jaikewe need it to save bandwidth
06:40.53wasimjaike: have you tried ilbc?
06:41.35movergood morning
06:41.59jaikenot yet
06:42.29wasimjaike: its about as computationally expensive, free and better sounding, especially on lossy networks
06:42.51jaikewer willing to shell out the $10 for the licenses..the least we can do to help digium
06:43.02jaikeill keep that in mind though
06:44.18jaikehehe..were in the third world too
06:44.28Qwellbeing a US citizen, I think I might be worse off
06:44.33Qwellsilly US
06:45.02wasimthird world citizens can't afford $10 per license ... thats like our percapita
06:45.19Qwelldaily, weekly, monthly?  yearly?
06:47.56wasimwell ... i think we're about $2200 per year
06:48.48*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
06:48.55wasimit might have improved 5% this year
06:49.11wasimbut that still sucks, with petrol at $1/liter
06:49.13Qwellwhere is here?
06:49.20wasimQwell: pk
06:49.27Qwellpakistan?
06:49.43Thumannhmm.. can asterisk dial a sip phone like dial(sip/ext@ip)
06:49.49QwellThumann: sure can
06:49.59Thumannhmm
06:50.11ThumannQwell: no additional conf needed ?
06:50.28zzzirkanyone have any experience with quintum equipment and interfacing to it?
06:51.00jaikewasim: where u at?
06:51.45ThumannQwell: i just get: == No one is available to answer at this time
06:51.51wasimjaike: lahore currently
06:52.11ThumannQwell: in the console that is.. when i do a dial(sip/ext@ip)
06:52.47QwellThumann: try something like Dial(SIP/user:password@extension@ip)
06:52.57wasimzzzirk: we did once, connect PRI to a quintum tenor box ... seemed fairly straight forward
06:53.52zzzirkI'm just pretty new to this and I've been tasked with integrating the quintum with asterisk.  I don't think it's a tenor though.
06:54.38Qwellzzzirk: if its got PRI, it should be fairly simple
06:56.22*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
06:56.25ThumannQwell: i have no user info.. but if i just try adding like user:password@ext@ip i get
06:56.49ThumannNo such host: ext@ip
06:56.53zzzirkI guess it is a tenor...  Tenor DX2048 is what it says
06:57.09zzzirksays it has 2 T1/E1/PRI ports so I guess I may be alright
06:57.22wasimzzzirk: ok, they have a winblows based gui admin thingy
06:58.09wasimzzzirk: how do you want to connect the * to the quintum, over ip or tdm?
06:58.39zzzirkIP I believe.  I'm not the one that's making the requirements
06:58.55wasimzzzirk: ok, IP on which protocol then? SIP or H323?
06:59.01zzzirkh.323 is what the telco people were telling me
06:59.08zzzirkis there a preference?
06:59.18wasimzzzirk: yes, and guess which one its not
06:59.20*** join/#asterisk drooth (n=drooth@251.16.79.83.cust.bluewin.ch)
06:59.34zzzirkI would guess h323 the way you said that
06:59.41zzzirkI think we can change if need be though
06:59.43Qwellheh, h323 is crap :p
06:59.52zzzirkheh, thanks!
07:00.14QwellI'd probably do PRI to that box...its all preference though
07:00.18zzzirkI've only ever used asterisk with small potatoes setups and I got thrown into this
07:00.28wasimQwell: quintums won't trunk out on a PRI, iirc
07:00.49Qwellwasim: Didn't you say you went PRI to a quintum?
07:00.58zzzirkbasically what I'm doing is a proof of concept for a friend...
07:01.05wasimQwell: thats to trunk in, and sip out ...
07:01.15Qwelloh
07:01.41wasimzzzirk: do you have digium pri card?
07:02.00zzzirkthe quintum and asterisk working together.  the idea is to have some soft phones on the asterisk side that can dial out via the quintum and then have DIDs on the quintum that route to the asterisk box and the softphones
07:02.20zzzirkwasim: no, I don't.  Is that a requirement?
07:02.22wasimzzzirk: why not just get a pri card for asterisk and fleabay the quintum?
07:02.45zzzirkagain, I'm looking to get a handful of lines going just to prove to these "old telco" guys that this voip stuff works
07:03.12zzzirkwasim: I actually had that same thought, but I don't know that these guys are quite up for doing that quite yet
07:03.15wasimits those "old telco" guys who came up with h323 and then sip you know, just so voip is difficult and they keep their old jobs
07:03.27*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:03.32zzzirkI'll keep that in mind
07:03.43wasimzzzirk: ok, so sip is your best bet
07:03.51wasimzzzirk: it shouldn't be all THAT difficult
07:03.57zzzirkokay, I'll keep that in mind
07:04.15zzzirkit shouldn't be that hard for me to map my DIDs to a given SIP connection then?
07:04.19*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
07:04.28wasimzzzirk: no, it shouldn't ...
07:05.01zzzirkif we can "prove" this works they may be willing to go with the PRI card approach, but I gotta get there first
07:05.11wasimzzzirk: the PRI card would be MUCH simpler
07:05.20zzzirkyeah, I would think so...
07:05.43zzzirkI've worked with a single FXO digium card and I'd think that a PRI would just kind of be a scaled version of that for the most part
07:05.52wasimzzzirk: for the most part, just better
07:05.56zzzirkright
07:06.08*** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl)
07:06.14zzzirkbut then I don't have to worry about what I'm having to deal with in another piece of hardware
07:06.18zzzirkit's all in one box...
07:06.27wasimyes, secure, snug and warm
07:06.31zzzirkhow many connections can I expect to be able to scale in a single box?
07:06.39wasimzzzirk: which codec?
07:06.52zzzirkum...  don't know that there's a requirement
07:06.56wasimzzzirk: i'm testing a dual opteron we want to push to 240 g729
07:07.25zzzirkand if you need more you can just "cluster" multiple boxes?
07:07.31zzzirkI use the term "cluster" loosely
07:07.39wasimyep, we're thinking of getting dual opteron blades
07:07.45zzzirkokay
07:07.51wasimfor just the encoding bits
07:07.58zzzirkis g729 the way to go on codecs then?
07:08.11wasimnah, depends
07:08.12zzzirkI don't know all that much about codecs
07:08.26zzzirkI just know that they seem to be alphabet soup to me
07:08.30wasimg729 is fairly widely supported and costs $$
07:08.30zzzirk;)
07:08.34zzzirkah
07:08.35wasimis abuot 8kbps
07:08.45wasimilbc is free and is about the same
07:08.52wasimjust not widely supported (other than *)
07:09.06wasimgsm is 13 kbps, but has very little cpu overhead
07:09.47wasimlike my p4 3.0 can do 200 gsm channels
07:10.02wasimthe dual opteron 248 can probably push that closer to the 1000 range
07:10.15zzzirkbut it uses mroe bandwidth
07:10.29zzzirkbut is the quality decent?
07:10.33wasimyeh
07:10.53zzzirkso if you can "afford" the bandwidth it's not a bad deal
07:12.01wasimwell, with iax2 trunking yuo can save a lot of bandwidth, so the difference is there, but not that much, it starts becoming a tradeoff on number of calls, bandwidth and cpu cost
07:12.39wasimwe're designing a new network for a telco here, thats using GSM locally (they are doing cell already) with IAX2 and then when you egress the network, they can opt for ilbc or g729 or whatever
07:13.01wasimso they need/buy a lesser number of g729 channels, and the cpu resources are pooled in one farm ... but this requires thoughtful design
07:13.35wasimthoughtful design requires beer ... brb
07:14.18Thumannanyway to force asterisk to use sip 1.0 when dialing a remote sip extension?
07:14.45Qwelldidn't realize SIP was versioned
07:15.29Thumannafaik: there's a 1.0 and 2.0
07:20.09*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
07:28.33*** join/#asterisk Red15 (n=irc@d54C323D1.access.telenet.be)
07:28.50Red15top of the morning to y'all
07:29.01Thumannu2
07:39.54*** join/#asterisk FreezeS (n=gido_b@82.208.156.94)
07:40.07Qwellbed
07:40.43FreezeSdo you guys know any torrent sites with old games ?
07:41.06hugo-v6FreezeS: buy a better pc and get the new ones ;)
07:41.15FreezeSit's not about my PC
07:41.35FreezeSI have a barton 3200
07:41.38hugo-v6well then... get a emu and play c64 or arcade games ;)
07:41.39FreezeSwith 1G ram
07:41.49FreezeSbut I want to play Dark Reign
07:41.56hugo-v6well i dont have a barton but 1,5gb ram ;)
07:42.01FreezeSand can't find it anywhere
07:42.21FreezeSI used to play it on my 486, a long time ago
07:42.37hugo-v6FreezeS: then buy it on the local garage sale ;)
07:42.50FreezeSwe don't have garage sales in Romania :)
07:43.00hugo-v6but no i dont know where to get that
07:43.16hugo-v6then u have no luck ;)
07:43.19FreezeS:(
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07:53.52jaikeanyone every tried converting a 2.5mm terminated headset to rj11?
07:54.07Red15can't say that i have
07:54.33jaikeweve lots of 2.5mm ended headsets, cant use em for polycoms which require rj11
07:54.37Red15you mean a mini-jack ?
07:54.50jaikewonder if you can just cut and crimp em
07:54.53jaikeyup
07:55.03xmingwith RT iax peers, I can register but I can't make any calls, debug says CAUSE           : No authority found
07:55.06xming<PROTECTED>
07:55.06Red15i doubt it, the headset won't be able to hande the voltage
07:55.13xmingany hints?
07:55.21Red15xming: type iax2 show registry
07:55.42xmingit's empty
07:55.50xmingiax2 show peers
07:55.50xmingName/Username    Host                 Mask             Port          Status
07:55.50xmingxming/xming      81.242.138.161  (D)  255.255.255.255  4569          Unmonitored
07:55.52Red15then you have no register => line in the iax.conf
07:55.53xming1 iax2 peers [0 online, 0 offline, 1 unmonitored]
07:56.13xmingbut I don't need the register line when I was using iax.conf
07:56.16jaikecant make outbound? or cant receive inbound
07:56.40xmingcan't make outbound with a iax client connect to * (with RT iax friends)
07:56.50Red15RT ?
07:56.56xmingRealtime
07:57.11Red15ah don't have much experience with that
07:57.12jaikenever tried it
07:57.20xmingbtw I am running 1.2-beta1
07:57.31Red15what does 'iax2 show peers' give you ?
07:57.53xmingfrom the plain simple iax.conf, everything works, sip RT works too, but no IAX RT :(
07:58.10xming09:55 < xming> iax2 show peers
07:58.10xming09:55 < xming> Name/Username    Host                 Mask             Port
07:58.10xming<PROTECTED>
07:58.10xming09:55 < xming> xming/xming      81.242.138.161  (D)  255.255.255.255  4569
07:58.10xming<PROTECTED>
07:58.20*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
07:58.58*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
07:59.22xmingwasim: try me ;)
08:00.40*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
08:01.02*** join/#asterisk ellvis (n=ellvis@adsl-data-253.84-47-65.telecom.sk)
08:01.08ellvishi people
08:01.35xmingand iax2 show peer xming also shows that everything is normal
08:01.42ellvisplease, can anyone help me with the calls picking? (features.conf)
08:02.06*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
08:02.33Red15xming: so you are trying to make a plain connection with IAX to a voice client ?
08:02.41*** join/#asterisk mrtwister (n=mrtwiste@cable-1-32.cgates.lt)
08:02.41Red15xming: like idefisk or what ?
08:03.06*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
08:04.09Red15xming: any nat going on? since your host has external ip ?
08:19.04xmingI am trying to use firefly (behind nat) to connect to *, I can register but cannot even dial 600
08:19.07mrtwisterany idea friends.. i have registered inbound DID at two asterisks, and at one i want process it only in case if other is offline (one asterisk registered as friend at another). how to check, is peer/user online in dialplan (exten => )
08:21.09*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
08:21.10xmingwith the peer/friend config in the iax.conf everything works, but when the peers/friends are in the mysql using RT, I can register but cannot callout
08:21.29xmingso I don't thing that nat is the issue here
08:26.17hugo-v6*stalk*
08:27.53*** join/#asterisk sep (n=sep@217.17.211.40)
08:29.31xmingok I have compared the outputs of "iaz2 show peer xming" in both cases
08:29.43Red15xming: do you have any sign that your context of the iax2 account is correct ?
08:31.44xmingthe differenmces are   * Name       : xming
08:31.44xming<PROTECTED>
08:31.44xming<PROTECTED>
08:31.44xming<PROTECTED>
08:31.44xming<PROTECTED>
08:31.47xming<PROTECTED>
08:31.50xming<PROTECTED>
08:31.52xming<PROTECTED>
08:31.55xming<PROTECTED>
08:31.57xming<PROTECTED>
08:32.00xming<PROTECTED>
08:32.02xming<PROTECTED>
08:32.05xming<PROTECTED>
08:32.07xming<PROTECTED>
08:32.10xmingsorry
08:32.13xmingthe differenmces are callerid and ACL
08:32.29xmingRed15: the context is the same in both case, so is the ip address
08:32.54Red15does you * actually show it gets into the context atm you try to dial ?
08:33.48*** join/#asterisk z1on0110 (n=z1on0110@customer-201-133-103-11.prod-infinitum.com.mx)
08:34.00xmingRed15: how do I see that?
08:34.08z1on0110hi, im new here ...
08:34.17xmingis ther somthing like dialplan debug?
08:34.25Red15xming: does your cli show any reaction when you try to dial from the iax extension ?
08:34.36z1on0110im trying to install asterisk on an RS/6000 ... but got some errors when i try to compile it
08:34.37Red15xming: you have the CLI running ?
08:34.55xmingRed15: it does when the per is in iax.conf, but not when it's configure in the mysql (RT)
08:35.19Red15xming: what verbosity is your CLI running at ? I suggest at least 3x -v
08:35.21*** join/#asterisk jamc (i=p3dmildg@h250n1fls34o969.telia.com)
08:35.52z1on0110does anyone had compiled asterix on a PPC64 ????
08:35.55xmingRed15: set verbose 999 right now
08:36.10xmingand set debug 999
08:37.02Red15so absolutely nothing happens at the cli ?
08:37.18xmingz1on0110: I have done that on sparc64/Linux
08:37.30xmingz1on0110: what kind of compile problems?
08:38.02z1on0110i got this one    "db.c:46: undefined reference to `.dbopen'"
08:38.09xmingz1on0110: basically, if you have gnu tools, you just need some Makefiles hacking
08:38.37z1on0110db.c:46: more undefined references to `.dbopen' follow
08:38.37z1on0110collect2: ld returned 1 exit status
08:38.37z1on0110make: *** [asterisk] Error 1
08:39.16xmingz1on0110: do you have multuply sleepycat db version?
08:39.26xmingz1on0110: is it linux or aix?
08:39.46z1on0110this is for linux, im running gentoo 2.6 on a rs 6000
08:40.19xmingis your userland 32 or 64 bit?
08:40.29z1on011064 bit
08:40.44*** join/#asterisk RoyK (n=roy@80.239.107.80)
08:43.15xmingsep: try this ls -d /var/db/pkg/sys-libs/db*
08:44.43sepxming,  ????
08:44.53Red15wrong nick i gues :P
08:44.56sepohh
08:45.16Red15z1on0110 see " try this ls -d /var/db/pkg/sys-libs/db*"
08:45.25xmingsep?
08:45.37Red15sep ? xming ?
08:45.39Red15rofl
08:45.43xmingxming ???
08:45.53xmingsorry
08:46.34xmingz1on0110: I think it's just trying to link to the wrong db
08:46.42z1on0110xming i got this /var/db/pkg/sys-libs/db-1.85-r1  /var/db/pkg/sys-libs/db-4.1.25_p1-r4
08:46.55opus_hi
08:47.27*** join/#asterisk rajo_ (n=rajo@bfs.cs.uni-sb.de)
08:49.15z1on0110xming: ... and how i can fix that ??  :S
08:49.31Red15hi opus_
08:49.41opus_Hey Red
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08:50.44opus_4 monitors at once kicks ass
08:51.28xmingzloc: I just check the source, * includes it's own db source tree, so what you have shouldn't matter
08:51.45xmingz1on0110: I just check the source, * includes it's own db source tree, so what you have shouldn't matter
08:51.54xminggod I hate nick completion
08:52.40*** join/#asterisk dexteruk (n=dexteruk@217.165.103.194)
08:53.18xmingz1on0110: can you give me more details, like where the error occured (in the directory)
08:53.28dexterukhi can anyone help me with the SIP software from Cisco 7912 as i have ordered the software form CISCO but it will take two weeks for my registration
08:53.43xmingdexteruk: call cisco ;)
08:53.43z1on0110xming: ok ... hold on plz
08:53.46dexterukI live in the UAE and there not very helpful here
08:54.24dexterukI have
08:54.49xmingdexteruk: so why don't you use/try other sip softphones?
08:54.49dexterukmany time i have phone and i cannot use them, please help
08:54.55X-RobWell.
08:55.03X-RobWasn't that thrilling.
08:55.24z1on0110xming: /home/z1on0110/asterix/asterisk-1.0.9/db.c:46: undefined reference to `.dbopen'
08:55.24z1on0110db.o(.text+0xdd0): In function `database_show':
08:55.24z1on0110/home/z1on0110/asterix/asterisk-1.0.9/db.c:46: undefined reference to `.dbopen'
08:55.24z1on0110db.o(.text+0x105c):/home/z1on0110/asterix/asterisk-1.0.9/db.c:46: more undefined references to `.dbopen' follow
08:55.24z1on0110collect2: ld returned 1 exit status
08:55.26z1on0110make: *** [asterisk] Error 1
08:55.36dexterukI have some 7940 already but we purchased some 7912 thinking the software world be the same but is not so we have had to order
08:55.38xmingdexteruk: sorry I really don't have any cisco softphone experience
08:55.41X-Robz1on0110 -
08:55.43X-Rob~pb
08:55.46jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
08:56.10opus_z1on i think means you have berekley db installed, but you don't have berekley-dev header files  installed or the wrong version installed.
08:56.35dexterukDoes anyone have CISCO 7912?
08:57.37z1on0110opus_: u suggest i must install or upgrade my berekley db ?
08:57.55opus_both, ?
08:58.00opus_Its a wild guess
08:58.07z1on0110hehe
08:59.21xmingz1on0110: put more details on http://pastebin.ca/
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08:59.46z1on0110xming: ok
08:59.54xmingand the /msg me
09:00.09xmingso anyone who can help me?
09:00.26X-Robxming - what's your problem?
09:00.31xmingstill fight RT iax with no authority pb
09:00.56X-RobWoah. Sorry. Right out of my league. I haven't played with RT at all yet.
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09:01.47xmingX-Rob: oh, RT is easy for sip, but not for iax
09:02.31dexteruk<PROTECTED>
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09:04.16z1on0110xming: here are more details http://pastebin.ca/22886
09:06.22xminghmm
09:07.44jaxkzexten => _00.,1,Dial(IAX2/${EXTEN}@voipgate,60,tr)
09:07.48jaxkzwould this be correcT?
09:07.55xmingok go to your * src, then cd db1-ast/
09:08.33xmingpaste the out of "nm db.o |grep dbopen" here
09:08.47Delvarjaxkz: no
09:09.06jaxkzwhere is the error?
09:09.10Delvarjaxkz: try > exten => _00.,1,Dial(IAX2/USERNAME@voipgate/${EXTEN},60,tr)
09:09.29xmingz1on0110: ping
09:09.37Delvarjaxkz: requires you have the username...
09:09.51z1on0110xming: im on it
09:10.13jaxkzUSERNAME is my loginname right?
09:10.29xmingomg
09:10.52z1on0110i got this
09:10.55Delvarjaxkz: should be
09:10.58jaxkzok
09:11.10z1on01100000000000000000 T .__dbopen
09:11.15Delvarjaxkz: doesnt your provider offer isntructions?
09:11.25jaxkzno
09:11.32Delvarwhats your provider?
09:11.33z1on01100000000000000000 D __dbopen
09:11.46jaxkzjust some guy with an E1 line at home
09:11.56jaxkzit's dutch
09:11.57z1on01100000000000000000 W dbopen ... this is the last line
09:12.43Delvarheheh
09:12.59xmingjaxkz: take a subscription with me, I will give you instructions ;)
09:13.01dexteruk<PROTECTED>
09:13.44jaxkzwhat do you offer
09:13.46xmingz1on0110: ok you linker is really fsck, have you had any other problems compiling? Did you tried the asterisk ebuild?
09:13.52jaxkzI need cheap calls to south-african mobile
09:14.46xmingjaxkz: I am a guy with E1 at home and who speaks Dutch ;)
09:15.26xmingz1z
09:15.28z1on0110xming: i had no other problems compiling, i have already compiled zaptel 1.0.9
09:16.04jaxkzWat doe jij met een E1 thuis
09:16.13z1on0110i have not tryied ebuild
09:16.34jaxkzWhat do you pay for the E1?
09:16.57RoyKjaxkz: jeg skjønner ikke hollandsk
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09:18.00jaxkzRoyK: yeah sure ;)
09:18.10MimmusI have a problem with recognition of '*' digit incoming from an analog PBX connected by a PRI cable.Any idea?
09:19.07Mimmusonly '*' doesn't work
09:19.18xmingz1on0110: I don't know if zap drivers are going to work wiht ppc/64, but * should, if you trust me give a a user account on a test machine I can try to get it compiled
09:22.01*** join/#asterisk _omer (i=o@203.215.180.250)
09:22.05_omerhi
09:22.17xmingjaxkz: well it's not really at home it's in the colo
09:22.19_omeranyone who have exp with Cisco 5300???
09:23.34xmingwhy is everyone asking cisco questions here? :)
09:24.36Mimmusany experience with SMS+Asterisk?
09:24.37z1on0110bad cisco support ;)
09:24.40*** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com)
09:24.41RoyK~seen zoa
09:24.47jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 1d 16h 12m 11s ago, saying: 'coldfeet: chanspy'.
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09:24.51*** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
09:26.13_omerz1on0110 :   :(  just need to know how to change port from 5060 to 5061
09:26.50jaxkzxming: i'm trying to get a E1 @ redbus (eweka)
09:27.01jaxkzhow much does that cost?
09:27.55z1on0110_omer: sorry _omer, i would like to help u, but i have never used a 5300 :S
09:27.55jaxkzexten => _0031.,1,Dial(IAX2/user@peer/${EXTEN},60,tr), i always get a congestion when i call a number in The netherlands
09:28.13X-RobMimmus - is '*' working when it's _not_ the first digit dialed?
09:28.15nextimeany pyastre user here?
09:28.52_omerz1on0110 :  thnx
09:29.50dexteruk<PROTECTED>
09:29.51MimmusX-Rob: yes, it works (for instance MeetMe conferences 8XXX work)
09:29.57xmingjaxkz: please talk to me in the private channel
09:30.49hugo-v6problem (and no idea, so idea is wanted): phones (sip or isdn) call numbers (ie 012341234 or 1234 which are no local numbers) atm im routing them to ISDN via leading 0 which is anoying. now i want to route them to isdn (w/o leading 0) and also wand to call the internal numbers. possible and how?
09:30.51xming_omer: put a linux box b4 it and use iptables to redirect the port ;) sorry I don't have any cisco ip phone experience
09:30.54*** join/#asterisk vlrk (n=vlrk@59.93.74.167)
09:31.40nextimeanyone from brasil that can resell terminated voip traffic from italy? i need a LOT of traffic
09:31.49dexterukI know im just asking periodicaly just incase any new does
09:32.00X-RobMimmus - you might be catching one of the built-in channel codes
09:32.02X-Robsee http://www.voip-info.org/tiki-index.php?page=Asterisk+ZAP+channels
09:32.24MimmusX-Rob: I'm using a special prefix on the analog PBX to engage outgoing line, then I digit Asterisk exts
09:32.57MimmusX-Rob: I'd like to offer Asterisk services to 'traditional' users
09:33.15X-Robso you're dialling '*4' or similar?
09:33.33X-Rob(does 'I digit' mean 'one digit')?
09:34.01vlrkdoes this ICD  module produces any .so file
09:34.12MimmusX-Rob: no, I'm dialing AsteriskAtHome services (*98, *60 ...)
09:34.17vlrklike all other applications
09:34.32_omerxming :  I need to change  Connection and SIP LISTEN PORT....both
09:34.39_omerxming :  there might be a problem :)
09:34.39X-RobAhha. You want AMP support. See #amportal
09:34.51X-Rob(I'm there, and I'll help you out, but this isn't an 'asterisk' problem, it's an AMP problem)
09:35.04MimmusX-Rob: no no! I have 'full' controll of dialplan!
09:35.39X-RobUh.
09:35.49X-RobI guess that means 'I'm not using ASterisk@Home'?
09:35.49MimmusX-Rob: services work if I use SIP extension or phone calls incoming from PSTN PRI line
09:36.02X-RobMimmus - yes. The Zap trunk has built in codes.
09:36.09X-Robsee the link I pasted earlier.
09:36.19X-RobOh
09:36.21X-RobOOooooh
09:36.29X-RobSo if you bypass the channel bank it works?
09:37.02*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
09:37.05MimmusX-Rob: yes, Alcatel support said me that PBX passes ALL digits
09:37.25MimmusX-Rob: but '*' doesn't work
09:39.19X-RobSo, turn 'pri intense debug span 1' (or whatever span it's on) on.
09:39.22X-RobSee what's being sent.
09:39.31X-Robpossibly you don't even need intense
09:39.40MRH2can u  use standard cat5 patch / networking cable to connect 410 card to to an isdn30e network termination box?
09:40.23X-RobMRH2 - yes. ISDN uses pairs 1/2 and 5/6
09:40.34X-Robin a standard straight-through cable, they'll be pairs.
09:40.44X-Robuh. that's _pins_ 1 and 2 and 5 and 6.
09:40.52X-Robit's _pairs_ 1 and 3.
09:42.26MimmusX-Rob: thanks, I will trry ASAP
09:42.32MRH2cool so i can use standard networking cable without probs
09:42.37*** join/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
09:47.19jaxkzHmm.. When using DISA, not al pressed digits make it to the asterisk box
09:47.41X-Robjaxkz - how are you connecting in? Zap?
09:47.47jaxkzsip
09:47.51MimmusX-Rob: no, debug gives me no useful info
09:48.18X-Robjaxkz - ok. Odds are, your phone is sending DTMF inband once the call is established. That'll suck.
09:48.35jaxkzsome make it it, some don't
09:48.54jaxkz003124722
09:48.55X-Robeasy way to find out is to sip debug the connection and try pushing buttons in a call. If you see sip packets, it's rfc3389, if you don't see anything, that's your problem
09:48.58jaxkzthis is left from 11 digits
09:49.33*** join/#asterisk Snake-Eyes (n=blog@203.201.99.188)
09:49.36jaxkzok
09:49.42X-RobMimmus - it does. I suggest doing a 'pri intense debug', then take a long hard look at the debug file. it is readable.
09:50.02X-Robif you _do_ see packets, that's bad, and it's probably a bug somewhere.
09:50.53MimmusX-Rob: ah, ok, I was looking on the console not in the file
09:51.14X-Robin /var/log/asterisk
09:51.27jaxkzwell
09:51.40jaxkzwhen DISA starts, there is NO debug
09:51.41jaxkz000117417722
09:51.48jaxkzthis is the result again of a long number
09:51.49MimmusX-Rob: sic! it's full of unreadable output!
09:52.03X-Robjaxkz - good. Your problem is you're sending DTMF inband. That won't work reliably.
09:52.19jaxkzok good.
09:52.27jaxkzSo i need to change the type of sending
09:52.34xmingset tit to rfc
09:52.38jaxkzI am sending this from a mobile phone to an asterisk box
09:52.44jaxkzcome again?
09:52.44*** join/#asterisk mmmToop (n=chatzill@c1-66-2.rndf.isadsl.co.za)
09:52.45X-Robyes. You also need to put 'dtmfmode=rfc2833' in your sip.conf file
09:52.56X-Rob..from a mobile phone?
09:52.59jaxkzyes
09:53.02X-Robthat's not SIP
09:53.08jaxkzhear me out
09:53.16xmingmobile -> sip something?
09:53.22jaxkzi am calling an access number in holland, wich calls me back
09:53.31jaxkzusing SIP/mobile number
09:53.36xmingor just mobile -> calling pstn -> pri/bri/fxo
09:53.44jaxkzthen it presents me with DISA
09:53.56jaxkzmobile -> pstn
09:54.02jaxkzsip -> mobile
09:54.07X-RobMimmus - Paypal me AU$80 (about US$60) and you've got me for an hour.
09:54.17X-Robset me up an account and I'll log in and tell you what's going on.
09:54.42xmingwhooaa a real busness man here ;)
09:54.49X-Robjaxkz - OK. The device that's answering the phone and converting it to sip needs to do RFC2833
09:55.01_omerxming: I can do it for nothing ;)
09:55.07*** join/#asterisk gonzo- (n=gonzo@195.140.246.50)
09:55.24jaxkztesting again
09:55.37jaxkzall the testing costs me a shitload of money
09:55.52X-Robjaxkz - OK. The device that's answering the phone and converting it to sip needs to do RFC2833  <------
09:55.56X-Rob^^^ read that
09:56.02X-Robthat's usually set up at the provider
09:56.14MimmusX-Rob: I see a strange message "waitfordigit returned < 0..."
09:56.38jaxkzX-Rob: i cannot setup that up in my mobile phone
09:56.48X-Robyour mobile phone is not doing the sip translation
09:56.56X-Robthe number you're calling from the mobile phone
09:57.02X-Robcall the people who own that.
09:57.09X-Robtell them to turn on rfc2833
09:57.20jaxkzX-Rob: check
09:57.26jaxkzI am calling a number. wich calls me back
09:57.36jaxkzi use an sip carrier to call myself
09:57.58X-RobYes. The sip carrier
09:57.59X-Robcall them
09:58.10X-Robtell them to turn on rfc2833 dtmf sending
09:58.32jaxkzusing voipgate.com
09:58.51xmingping z1on0110
09:59.13z1on0110xming: im here ... everything is going ok ???
09:59.33xmingok ther is somthing really fsck in the toolchain
09:59.45xmingI don't know if this is ppc64 specifiek
09:59.58xmingsee this?
09:59.59xmingnm db.o |grep dbopen
09:59.59xming<PROTECTED>
10:00.14xmingand this on a "normal" boxen
10:00.16xmingnm db.o |grep dbopen
10:00.16xming<PROTECTED>
10:00.43z1on0110i think it might be a problem with ppc64
10:01.06xmingthe .dbopen isn't defined anywhere, and the linker is right
10:01.49xmingI don't know why there are .function everywhere
10:02.22z1on0110are there any libraries missing ?
10:02.50gstdo i need to compile the zaptel module with the same gcc version as the kernel. since recompiling/installing the zaptel modules i experience kernel-panics when i load them. i just noticed that the kernel is compiled with gcc 2.95 and zaptel with gcc 3.3. could this be the reason for the panic?
10:03.26z1on0110i compile openssl, ncurses, and all stuff they request in the handbook
10:04.06X-Robgst - I don't think gcc 2.95 is a supported kernel compiler any more
10:04.09X-Robare you running 2.6 kernel?
10:04.10xmingz1on0110: no ther are no libs missing, but the lib is missing the function we need, because the function name is prefixed with a .
10:04.24jaxkzisn't dftmmode=auto better/
10:04.27*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
10:04.36xmingX-Rob: 295 still is
10:04.38*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
10:04.39gst2.4.30 - i am just upgrading to 2.6.13 to try if it works better with this version
10:04.51gstit worked without a problem on 2.4.30 for several months
10:04.54X-Robif you're using 2.6.13 you _must_ use CVS HEAD
10:05.00X-Robyou can't use asterisk 1.0.x
10:05.20xmingz1on0110: I can keep looking if you want
10:05.23gstthen i tried to upgrade to cvs head and i just got kernel-panics. after recompiling the old cvs version from several months ago this didn't dix the problem.
10:05.38jaxkzxming: check priv msg
10:06.15gstX-Rob: tnx. i hope that it works under 2.6.13 with cvs-head. the strange thing is that exactly the same version which worked for months now leads to a kernel panic. the only thing which (afaik) changed is the gcc version.
10:06.37xmingjaxkz: euh, you haven't anwsered my question yet
10:07.09z1on0110xming: thank u xming ... i was just trying to run my x100p on an old RS i got ... but i'll try on an regular x86
10:07.24xminggst: the kernel and kernel modules ( in this case the zaptel) should be compiled with the same compiler/linker/as ...
10:07.55xmingz1on0110: I don't know if the x100p driver is going to work
10:08.45z1on0110xming: i really appreciate ur time ... its late here in Mexico, and i have to sleep a couple of hours ... Thank you
10:08.58*** part/#asterisk sob0l (n=sobol_@ip-62-69-206-38.internet.v.pl)
10:09.12jaxkzxming: you are dutch right?
10:09.23xmingjaxkz: flemish ;)
10:09.49xmingz1on0110: ok, let me know if you need some help to get that thing compiled
10:09.52jaxkzAlmost the same
10:10.07xmingjaxkz: still big differences ;)
10:10.16xmingjaxkz: /msg
10:10.33jaxkzi am msg'n my ass off
10:10.44xmingto me?
10:10.50jaxkzyes ma'm
10:11.06z1on0110xming: ok ... see you later thnks 4 all !
10:11.08RoyKhm
10:11.15xmingz1on0110: no pb
10:11.18RoyKanyone here using a cisco 7940?
10:11.26z1on0110see you * channel ! :D
10:12.04*** part/#asterisk z1on0110 (n=z1on0110@customer-201-133-103-11.prod-infinitum.com.mx)
10:12.43xmingoh an other cisco question;)
10:13.42hugo-v6i dont know much about that cisco phones. but one thing i know for sure. i wont buy them nor deploy them somewhere
10:14.19jaxkzxming: maybe you should remove me from the ignore list or so :)
10:14.21hugo-v6hilight on cisco eh dude?
10:14.40xmingjaxkz: you are not on my ignore list
10:14.59X-Robjaxkz - are you registered? You can't /msg unless you're registered.
10:15.04xmingjaxkz: you are not a registerd freenode user?
10:15.05X-Robcheck your 'freenode' status window.
10:15.09*** join/#asterisk Drizzt321 (n=drizzt@c-67-189-187-233.hsd1.ma.comcast.net)
10:15.13xmingjaxkz: 11:31 -!- Private messages from unregistered users are currently blocked due to
10:15.16xming<PROTECTED>
10:15.20jaxkz[11:03] -NickServ- Password accepted - you are now recognized
10:15.26X-RobAhha.
10:15.30X-Robjaxkz is a wally.
10:15.40xminggreat !!!\
10:16.18dexterukhugo-v6: what phones would you deploy?
10:16.19jaxkzwally might be?
10:16.30xmingwally == jaxkz ;)
10:16.41jaxkzdefine: wally != jaxkz
10:17.06jaxkzi am an registered user
10:17.06Drizzt321I'm having trouble getting outbound and inbound calls with the xten x-lite softphone to dial through asterisks out to broadvoice. Whenever I try to dial, I get 404 not found, and when I try to dial in from my cell, it says the party you are trying to reach is unavailable. I've followed all the info I can find, including voip-info.org, but it just doesn't seem to be working. any help?
10:17.46xmingjaxkz: yes now you are
10:18.35jaxkzstill can't msg you
10:19.05xmingfreenode is still blocking jaxkz
10:19.25jaxkzyou msg me then
10:19.37xminghttp://freenode.net/faq.shtml#privmsg
10:19.39jaxkzi can respond to incoming
10:21.07RoyK<retry count=1>anyone here using a cisco 7940?</retry>
10:21.45X-Rob<XML><response><retry>No</retry></response></XML>
10:22.09X-Robwhat's it doing, or not doing, royk?
10:22.53RoyKjust hanging on startup, now reporting Protocol Application Invalid. Last time i booted it up, it said Not Provisioned or something
10:23.06vlrkhas any body with ICD experience
10:23.07RoyKand whatever I press, nothing happens
10:23.22X-Robdoesn't the not provisioned thing mean it can't tftp it's config file?
10:23.33RoyKX-Rob: prolly
10:25.47X-Rob<PROTECTED>
10:25.47X-Robhttp://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm
10:25.47RoyKX-Rob: do i need to configure ut using tftp, or can it be configured just using menus?
10:25.48X-RobIt needs to tftp it's configuration file
10:26.18X-RobOoh.
10:26.20X-Robo assist with any further errors or failed upgrade attempts, the RS-232 port on the phone provides console access for troubleshooting and debugging. Refer to the console access documentation located at the following URL:
10:26.20X-Robhttp://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/admin/4_0/790 0con7.htm
10:26.33X-Robdoes the .40 have a RS232 port?
10:27.09RoyKif that is the AUX port, but no 'console' port
10:28.07*** join/#asterisk torisa (i=hak@heceta.db.net)
10:28.15X-Robyeh, aux port would be serial
10:28.45X-Robbut you're gunna need to create all the TFTP files and make sure your DHCP server gives out all the right stuff.
10:28.59X-Robhttp://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960#comments
10:29.01X-Rob^^ some hints there
10:29.21torisaIs there some debugging software that could decode bell 202 / cid from pcm/wave?
10:30.29wasimtorisa: baudline would help
10:31.42torisathank you :)
10:33.15ellvisanyone can help me with pickup the call, *8 in features.conf?
10:36.13ellvisok ok, then nothing ;)
10:49.50X-RobGrubs - Uh-oh.
10:50.03X-RobThis falls into the 'sucks to be you' category.
10:50.22gordonjcphope you've got a laptop ide adaptor
10:50.36gordonjcpah wait, that doesn't work for Windows, does it?
10:51.07Grubsheh - currently pulled the drive and now am scratching for an adapter so I can open up my workstation, plug it in, and format the bitch.
10:51.33gordonjcpcan you stick the drive in another machine and install XP on it?
10:51.47Grubs..then load the cabs - back into the laptop - boot from usb floppy .. what a load of crap.
10:51.52gordonjcpah...
10:51.58X-Robgordonjcp - it does.
10:52.04X-RobGrubs - you're screwed.
10:52.07gordonjcpbut you can bang the cabs across
10:52.08Grubs:)
10:52.24X-Robgive up. do a net-install of debian from a bootable USB drive
10:52.56X-Robgordonjcp - you had to do that with win95. Whenever you changed the damn IP address it wanted the installation media.
10:53.02GrubsX-Rob: .... given this is a P3 600 with maxed out at 300MB RAM thats a bloody good idea!
10:53.21gordonjcpX-Rob: you could just skip all the files it asked for
10:53.40*** join/#asterisk voipguy (n=voipguy@196.200.26.42)
10:53.57GrubsI have a network Debian CD here...but no optical drive!
10:54.29Grubsis there a debian boot floppy for a net install?
10:54.34X-Robhurmmmm
10:54.44X-RobDunno. Possibly.
10:54.48gordonjcpGrubs: put the drive in another machine and install the base package
10:55.08Wonkathere are debian bootfloppies
10:55.10Grubsthe base is transferable across machines?
10:55.32*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
10:55.36X-Robyep
10:55.50Grubs<PROTECTED>
10:56.13WonkaGrubs: http://http.us.debian.org/debian/dists/sarge/main/installer-i386/current//images/floppy/
10:56.52gordonjcpGrubs: yes, of course
10:57.04gordonjcpas long as it's the same architecture
10:57.16Grubslove it.  Ta.  XP was too much for this laptop anyhow.  Thanks for the push in the right direction everyone.
11:05.48*** join/#asterisk zotz (n=zotz@24.231.36.100)
11:08.06*** join/#asterisk MJR^ (n=n0c1@210.212.195.134)
11:13.39*** join/#asterisk christo (n=chris@195.82.114.14)
11:14.22MJR^when i directly dial thru the soft phone the calls hit the destination ip, but  when i use asterisk it doesnt fwd to the destination ip, what may be the reason?
11:14.40MJR^can anybody tell me the reason
11:15.08christothis is really magic - how can asterisk report a file as not existing, when it clearly does?  http://pastebin.ca/22896
11:15.12wasimnot without what happens when * gets the call
11:15.33MJR^what do i do
11:15.39wasimMJR^: look at *CLI
11:15.45MJR^k
11:15.58MJR^it gives a warning
11:21.54MJR^WARNING[4490]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 07fab95068a6d22f7eaedce313519b2f@202.54.117.212 for seqno 102 (Critical Request)
11:22.09MJR^this is the error i get
11:23.18MJR^wasim can u see that
11:23.29MJR^i saw it in the packet analyser
11:23.41MJR^the call is not flowing out of asterisk
11:23.51MJR^what may be the reason
11:24.31wasimnat?
11:24.47MJR^this a live ip
11:25.10*** join/#asterisk Akke (n=akke@082-146-104-111.dyn.adsl.xs4all.be)
11:25.12AkkeHi
11:25.14MJR^from the same mach i can place the call thru the soft phone
11:25.24MJR^my asterix is on a live ip
11:26.00MJR^the ip 202.54.117.212 is of the sip provider
11:26.10*** part/#asterisk Grubs (n=Miranda@c211-28-131-24.eburwd3.vic.optusnet.com.au)
11:26.51wasimMJR^: check your dial string and contexts and sip.conf
11:27.14wasimMJR^: try a different provider also
11:27.24wasimMJR^: or perhaps a different account
11:27.45AkkeThere are providers over here in belgium that do carrier-select over normal phones and they only charge 0.15euro/minute to belgium mobile's. Does anyone know of any SIP-provider that allows me to call using my asterisk at the same or lower price?
11:32.58jaxkzget yourself an bri connection
11:33.30Akkebri connection?
11:33.49jaxkzisdn
11:34.09jaxkzsluit je die aan op je asterisk bak
11:34.31jaxkz0.15 vind ik btw vrij prijzig
11:34.42jaxkzin .nl kan je al voor 0,025 cent per minuut naar mobiel belle
11:34.52Akke0.15 naar GSM is vrij goedkoop denkik in belgie
11:34.58Akkehoeveel betaal jij naar GSM in belgie dan?
11:35.08jaxkzIk woon in nederland ;)
11:35.32Akkeja, maar misschien betaal jij 'internationaal naar belgium mobile' minder? :p
11:35.37jaxkzBelgium Mobile Base  
11:35.37jaxkz0.2471
11:35.37jaxkzBelgium Mobile Mobistar
11:35.37jaxkz0.2092
11:35.37jaxkzBelgium Mobile Proximus
11:35.38jaxkz0.1627
11:35.40jaxkzvrij duur dus
11:35.45Akkeidd
11:35.47*** join/#asterisk RoyK (n=roy@80.239.107.69)
11:36.03Akkemaar een isdn connectie op mijn asterisk aansluiten, zo'n isdn bri connectie zal wss wel vrij duur zijn?
11:36.05RoyKding
11:36.11RoyKhow can o reset a cisco 7940???
11:36.13jaxkzBelgie en nederland zijn zowiezo duur qua mobiel
11:36.19RoyKit only asks for the bloddy password
11:36.25jaxkzin nederland iets van 14 euro per maand
11:36.34jaxkzje kan ook je analoge lijntje eraan hangen
11:36.45RoyKjaxkz: jeg tror det er noe feil med tastaturet ditt. jeg skjønner ihvertfall ikke noe
11:36.54*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
11:37.01jaxkzRoyK: swedish?
11:37.10Akkejaxkz maar dan betaal ik toch ook gewoon de prijs die ik nu betaal met mijn vaste lijn thuis? (wat nog altijd duur is :p)
11:37.12RoyKnah. norwegian
11:37.17jaxkzpreselect toch
11:37.33*** join/#asterisk tuxinator_linuxM (n=spabin@24-53-54-195.ontrca.adelphia.net)
11:37.49jaxkzdone
11:44.07*** join/#asterisk reagent (i=mathias@2002:d4fe:b1a1:0:20d:88ff:fef4:66bb)
11:47.26RoyKhm
11:47.33RoyKanyone here using cisco 7940 phones?
11:48.17sepRoyK, leiker med asterisk ?
11:48.51RoyKsep: leiker og leiker fru blom
11:49.21reagenthi. I have not more than 100kbit/sec of upstream bandwidth. Which codec (if any) do you suggest for SIP?
11:49.28sepRoyK, kva erfaringar har du med asterisk ?
11:50.00RoyKsep: norsk asterisk-prat skjer typisk på #asterisk-no
11:51.00wasimenglais!
11:52.38MJR^omar can u help me
11:55.30_omersure...
11:55.38_omerso is your asterisk at public IP?
12:01.06xmingreagent: ilbc, gsm?
12:02.19reagentxming: Ok. Thank you
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12:03.10*** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-181.sw.biz.rr.com)
12:03.27*** join/#asterisk ManxPower (n=eric@ip68-225-97-156.br.no.cox.net)
12:04.40sivanamorning Manx
12:08.40ManxPower'morning
12:09.15*** join/#asterisk Martohtar (i=Martohta@82.196.218.80)
12:10.46*** join/#asterisk Assid (n=assid@203.115.64.62)
12:12.01ManxPowerLong term extreme stress can do weird things to people.
12:12.16Assidhrmm..
12:12.26MJR^omer r u there
12:12.40ManxPowerWent to the gas station the yesterday, put in my Shell card at the pump, it rejected it, went inside to ask what was wrong.  Turns out it was an Exxon station.
12:12.51sivanaheh
12:13.05Assidhahaha
12:13.07Assidrofl
12:13.19sivanadude... you need a day off :)
12:13.25sivanaat the spa or something
12:13.34Assidspa?
12:13.37sivanawith 4 or 5 masseuses
12:13.38*** join/#asterisk iguy (n=iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
12:13.42ManxPowersivana, I'm a Katrina Refugee.
12:13.54ManxPowerTraveled 2,000 miles in 2 weeks
12:13.59ManxPowerby car.
12:14.04sivanaya, my heart goes out to ya... unbelievalbe
12:15.03Assidhey JerJer, any luck on my DID?
12:15.07ManxPowersivana, I'd handle it much better if EVERYTHING in the region isn't totally fucked up.
12:15.29sivanaya, time to move to Canada :)
12:15.44ManxPowerwhen all the nearby cities double in population in like 2 days it messes things up. sivana get me a canadian green card.
12:16.21ManxPowerIf I can find a campground that I like with broadband access I might go camping for a year.  Would be fun.
12:16.29sivanaprobably wouldn't be very hard
12:16.37sivanato get a green card here
12:16.50ManxPowergetting me to leave the house in january, however, would be hard.
12:16.57ManxPower<-- tropical kitty
12:16.59sivanacamping.. we're heading into fall season with winter not far away.. might be a tad cold
12:17.07tzangerheh
12:17.12ManxPowersivana, Um, in the South it's not all that cold.
12:17.20sivanaask tzanger
12:17.21tzangerI have in my hand a single 1.44MB floppy disk worth over US$8000
12:17.25sivanahe's in the toronto area
12:17.29Ahrimanestzanger: omfg
12:17.30ManxPowerand I was thinking "camper camping", not "tent camping"
12:17.45sivanatzanger: how cold does it get there?
12:18.09Ahrimanestzanger: how can such a crappy piece of hardware be worth that amounT?
12:18.10tzangerman you say camping for a year I think "god that'll be cold in the winter" but I forgot you're down south :-)
12:18.27tzangerthere's nothing wrong with wintertime in Canada
12:18.49tzangerit's the perfect opportunity to really do code development...  so long as you've got food and water and coffee why go out?
12:18.54tzangerI love that about my place
12:19.06Ahrimanesfoodstock for the whole winter?
12:19.08tzangertons of storage, tons of food/water...  and a 50 foot laneway on the highway
12:19.17tzangerAhrimanes: it's not inconcievable...
12:19.36sivanalots of canned beans.. heh
12:19.37Ahrimanestzanger: damn.. my fridge hardly has food for a nightsnack :)
12:19.43tzangerAhrimanes: you can buy a side of beef, have a couple dozen chickens and pork products in the freezer
12:19.52Ahrimanessivana: hehe.. remember to stock up on toilet paper too..
12:20.05*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
12:20.10tzangerAhrimanes: canned beans double to supplement the heating fuel bill
12:20.17sivanaheh
12:20.21Ahrimaneshaha
12:20.44MJR^ahow do i change te codec wen forwarding calls
12:20.46tzangeractually I was really hoping to have a 3kW windmill up before winter
12:20.48sivanatzanger: what kind of temperatures do you get at peak winter?
12:20.55*** join/#asterisk easydone (n=notdone@eksel.demon.nl)
12:21.03tzangersivana: depends... this past winter it was 30 and 32 below without windchill
12:21.03Ahrimanestzanger: oh that would be sweet
12:21.11sivanaya
12:21.35tzangerwindmill will be JUST for heating though... no batteries.  everyone wants to use batteries and as soon as you do that you eat up any savings you might have had with maintenance costs
12:21.39sivanaI got to admit, I like your little farm house
12:21.55MJR^anyone plzz help me out in tis how do i change the codec while forwarding calls
12:22.01tzangerI do too... gotta make some more coin to fix the roof and get the barn fixed up
12:22.14tzangerMJR^: it should happen transparently
12:22.37Ahrimanestzanger: gps coords? i could look at google earth :P
12:22.41xminganyone who can help met with iax and RT?
12:22.47AhrimanesRT?
12:22.51xmingrealtime
12:22.53Ahrimanesah
12:22.55Ahrimaneshm not me
12:23.14Ahrimanesthough i do have to play with 1.2.0 beta soon
12:23.24tzangerAhrimanes: not far from 43° 26' N     80° 30' W
12:23.29xmingit doesn't seem to work
12:23.36xmingRT sip is working fine though
12:24.16ManxPowerIn a fit of insanity I installed CVS-HEAD last night.
12:24.28tzangerManxPower: hahaha
12:24.34sivanayay.. welcome to the darkside
12:24.38*** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org)
12:24.53tzangerManxPower: and...?  did your computer explode?
12:24.59tzangerdid blood ooze out of the TDM ports?
12:25.06ManxPowerWell, I DID say I would install 1.2x when it became beta
12:25.44*** join/#asterisk aneredes (n=hannes@port-212-202-55-34.dynamic.qsc.de)
12:25.50aneredeshi
12:26.09ManxPowerI should have installed it before the Storm of Doom.  That would have made a good story.  "Yeah, I installed CVS-HEAD and it cause a hurricane"
12:26.38aneredesi use a zaphfc driver and when i call my number is not shown on the caller's display. can anyone help?
12:26.57*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
12:26.59*** join/#asterisk lehel (n=asd@82.79.20.17)
12:27.02lehelhello
12:28.08ManxPowerI'm still getting "Sep 14 07:26:57 NOTICE[24005]: chan_iax2.c:3023 iax2_read: I should never be called!" on CVS-HEAD. 8-)
12:28.20Ahrimanestzanger: near saint george?
12:29.03*** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org)
12:30.28*** join/#asterisk newl (n=newlook@203-59-184-194.dyn.iinet.net.au)
12:31.06*** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org)
12:31.34*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:32.02Ariel_morning al
12:32.10_omermorning
12:32.23bjohnsonwho's Al?
12:32.55*** join/#asterisk dave99 (i=David@207.201.200.136)
12:33.00sivanaheh
12:33.01_omerA-rie-L  = al
12:33.10*** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org)
12:33.31dave99Got a caller ID/voicemail issue
12:34.09dave99When we had caller ID set to the internal extension in sip.conf...
12:34.18Ariel_anyone have the lottery numbers for tonights lotter???? hummm...
12:34.48dave99all we had to do was enter 8501 and that executed, VoiceMailMain(${CALLERIDNUM})
12:35.04dave99so we didn't have to enter the mailbox number.
12:35.07sivanaAhrimanes: 46° 18' N     79° 25' W
12:35.18dave99Now we have caller ID set to the DID #.
12:35.22AhrimanesAriel_: yes, but already gave it to like 1.000.000 others..
12:35.29sivanahome of the only other NORAD defense station
12:35.49dave99Anyway to send the DID on external calls and the local extension on internal calls?
12:36.13Ariel_Ahrimanes, I don't mind sharing...
12:36.19AhrimanesAriel_: hehe
12:37.27Ariel_dave99, yes there are ways
12:37.46Ariel_how did you create your users are then by names or by the extension numbers?
12:38.03dave99By the extension
12:38.24Ariel_are the extensions the same as there voicemail boxes?
12:38.37dave99Yes, in sip.conf, you mean right?
12:39.20dave99[657], username=657, mailbox=657@default
12:39.26Ariel_dave99, then use the exten to route the voicemail
12:39.30Assidhey JerJer: you around?
12:39.54dave99Like this:?
12:39.59dave998501,1,VoiceMailMain(${EXTEN})
12:40.08dave99Cool.
12:40.25Ariel_dave99, no that will give you what you dialed
12:41.24dave99Ahh, right.  I guess I don't know what variable it stores the calling extension in then.
12:42.00AhrimanesCallerID?
12:42.15Ariel_Ahrimanes, he has changed the callerID to a number
12:42.44lehelAriel_: so i have that fritz card.. CAPI.. 4 numbers, how do i tell the zap to receive the fax from a number?
12:42.49*** join/#asterisk DrukenWrk (i=Druken@67.69.139.226)
12:43.04leheli would like an "exten" example..
12:43.07dave99IS there a variable?
12:43.14Ariel_lehel, route on the did
12:43.15AhrimanesAriel_: ah
12:43.36Ariel_dave99, looking at some settings to see if I have an Idea for you.
12:43.58DrukenWrkAriel_ == mr fix it
12:44.18DrukenWrk:)
12:44.48ManxPowerIf using CVS-HEAd you can just SetVar=DID=5551212 in each device stanza in sip.conf then SetCIDNum(${DID])
12:45.10dave99${MACRO_EXTEN} - is that it?
12:45.52dave99ManxPower, not sure what CVS-HEAD is (not used CVS much).
12:45.56*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
12:46.02ManxPowerdave99, or 1.2BETA
12:46.25ManxPowerI wrote a perl script to deal with changing the callerid between the extension and DID.
12:46.26MikeJ[Laptop]1.2 beta 1!
12:46.48ManxPowerMikeJ[Laptop], I did say I'd install it when it had a release.
12:46.55MikeJ[Laptop]heh
12:47.07DrukenWrkManxPower: do ya think if i blow up a canadapost truck it'd be considered going postal ?
12:47.10MikeJ[Laptop]yes.. much committing yesterday.. beta 2 should be soon..
12:47.36MikeJ[Laptop]just need kevin to get another good block of time to get the rest of the non-frozen features in
12:47.41ManxPowerMikeJ[Laptop], I did NOT need to hear that.
12:47.59MikeJ[Laptop]what.. the much committing part?
12:48.01ManxPowerMikeJ[Laptop], We have been massive numbers (like most calls) disconnected calls when they are IAX2
12:48.01christois anybody able to solve this asterisk-thinking-a-file-doesnt-exist problem?  http://pastebin.ca/22899
12:48.11ManxPowerSo in desperation I installed CVS-HEAD last night.
12:48.32MikeJ[Laptop]and?
12:48.41cypromisnow he is even more ddesperate
12:48.43cypromis:P
12:48.46ManxPowerMikeJ[Laptop], And I don't like installing right after a bunch of comits.
12:48.47MikeJ[Laptop]heh
12:48.47sivanatzanger: ping
12:49.00ManxPowercypromis, I can't be any more desperate
12:49.13_omerManxPower: how do you run perl script thru Asterisk?
12:49.29ManxPowerchristo, Asterisk puts the extension on for you, don't put it in yourself.
12:49.32MikeJ[Laptop]_omer, AGI, res_perl
12:49.42ManxPowerPlayback(happysounds)  NOT Playback(happysounds.gsm)
12:49.53tzanger43.725N 80.935W is pretty much my house
12:49.57tzanger(dug out the GPS)
12:50.07sivanahehe
12:50.09christoManxPower - doh good point :S :)
12:50.11christothanks
12:50.19MikeJ[Laptop]your house is a bunch of weird numbers and letters?
12:50.20ManxPower_omer, System(/my/perl/script.pl) or AGI(/my/perl/script.agi)
12:50.22MikeJ[Laptop]:P
12:50.31tzangerMikeJ[Laptop]: lat/long
12:50.32sivanatzanger: that headset you showed me once while I was down in TO, was it bluetooth?
12:50.36_omerthanks Mike
12:50.36_omerthanks Manx
12:50.37MikeJ[Laptop]or system.. that'll do it to.. heh
12:50.38tzangercould tlel you the elvation if I could get the GPS to work
12:50.44tzangersivana: yes.  Motorols HS810 I think
12:50.55MikeJ[Laptop]my elevation is about 960
12:50.57sivanawould it work with blackberry?
12:50.59MikeJ[Laptop]roughly
12:51.05tzangersivana: bluetooth should work with bluetooth
12:51.18Ahrimaneshm i dont know how to enter gps coords in g earth it seems
12:51.26sivanaya... n/m :)
12:51.32sivana<PROTECTED>
12:51.36ManxPowertzanger, I finally found a situation where I desperatly wanted to use Monitor
12:51.43tzangerManxPower: yeah?
12:52.22Ahrimanestzanger: near listowel?
12:52.24ManxPowerI called NuFone support, their system asnwered the phone, asked me to please wait for a rep, then after about 30 seconds of ringing told me nobody was available and then said goodbye and hungup on me
12:52.41tzangerManxPower: fun...  ugh
12:52.42AhrimanesManxPower: hehe nice support
12:52.49*** join/#asterisk aneredes (n=hannes@port-212-202-55-34.dynamic.qsc.de)
12:52.54*** part/#asterisk aneredes (n=hannes@port-212-202-55-34.dynamic.qsc.de)
12:52.58ManxPowerI feel the need to say it again.  ALL ITSPS SUCK!!!!!!!!!!!!!!1111
12:53.16ManxPowerTeliax's support has gone way downhill
12:53.27ManxPowerNufone's support was never great.
12:53.39ManxPowerboth provider's service is pretty good.
12:53.59ManxPowerOh!  Anyone have a power supply for a Cisco 1700 power supply.  Mine is...wet.
12:53.59tzangerManxPower: I just always use email support
12:54.20tzangerI get the ticket number and then jerjer smacks them if they don't make it right
12:54.27tzangerManxPower: your power supply needs a power supply?
12:54.35dave99I guess I'm using CVS-HEAD.  I'm assuming setvar only works with that?
12:54.40ManxPowertzanger, Yeah.  I e-mailed them on friday evening about all our DIDs being down.  I got a response at 3am this morning telling me the DIDs were working.
12:54.51ManxPowerOh!  Anyone have a power supply for a Cisco 1700 router?  Mine is....wet.
12:54.51dave99setvar=CID=657 in sip
12:54.55tzangerManxPower: no automatic trouble ticket?
12:55.12*** join/#asterisk MJR^ (n=vikramhe@210.212.195.134)
12:55.14dave99then exten => 8502,1,VoiceMailMain(${CID}) in extensions.conf works.
12:55.19MJR^omar
12:55.20*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
12:55.38dave99Thank Manx and Ariel
12:55.42ManxPowertzanger, I dunno, I was on my way to The Warmart Riot to try to buy food, then went to the campground.
12:55.48dave99*thanks*
12:56.23*** join/#asterisk konrads (n=konrads@out.ctkom.lv)
12:56.25MJR^is there any software to find the codecs used by a sip provider
12:56.47dave99Anybody have trouble with restarting Asterisk?
12:56.49ManxPowertzanger, I picked up the 1700 power brick and water poured out of it.
12:56.51tzangerManxPower: heh
12:56.57tzangerManxPower: that's not good
12:57.00tzangerthis isn't freshwater either is it
12:57.05dave99restart now sometimes works and sometimes just sits there.
12:57.12tzangeroh wait
12:57.13ManxPowertzafrir, It may have been at one time.
12:57.13MJR^is there any software to find the codecs used by a sip provider
12:57.22tzangerit's a cisco 1700... that's the water cooling
12:57.38ManxPowertzanger, the router is fine. 8-)
12:57.38MJR^can anyone help me
12:57.39dave99restart when convenient always just sits there, until eventually the system just freezes.
12:57.50MJR^is there any software to find the codecs used by a sip provider?
12:58.01ManxPowerMJR^, Yes.  "sip debug"
12:58.08tzangerManxPower: that's what I mean... the power supply is so small because it employs water cooling, just submerge it again and it should work
12:58.14tzangerMJR^: nope, you contact them
12:58.23*** join/#asterisk pablix (n=root@lacnic.net.uy)
12:58.28ManxPowertzanger, I think my Humor Module is broken
12:58.46*** join/#asterisk yartelecom (n=no-email@82.211.129.230)
12:59.05pablixi have a problem with my asterisk and the e1 from my pbx. can anyone  help me? i have this error message
12:59.22pablix<PROTECTED>
12:59.56FreezeSpablix: I think it's a driver issue
13:00.06Kattybeep!
13:00.06tzangerthat's okay, my devicenet module segfaults
13:00.13MJR^.how do we change the codec while forwarding the calls in asterisk
13:00.14tzangerkatty
13:00.14pablixi reinstall the last version of zaptel and pri
13:00.15tzangerkatty
13:00.16tzangerkatty
13:00.22tzangerkatty
13:00.24tzangerhad enough beeping yet?
13:00.25Katty...
13:00.31Kattyi'll beep /you/ in a minute.
13:00.36tzangerkatty
13:00.39tzangerkatty
13:00.39Ahrimanes<PROTECTED>
13:00.42tzangerkatty
13:00.43konradsHas anyone done feature-by-feature implementation or analysis of some midrange conventional PBXs? Like alcatel OmniPCX or any other small/midrange gear
13:00.44tzangerkatty
13:00.47tzangerkatty
13:00.50tzangerkatty
13:00.52tzangerkatty
13:00.55tzangeroh
13:00.56konradstzanger, please cut it
13:00.57tzangervirtual beep
13:01.00tzanger*sigh*
13:01.06Katty;)
13:01.16pablixits not a driver problem i think is another problem
13:01.41konradsLike address book in sip phone displays
13:02.17*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
13:04.58MJR^.how do we change the codec while forwarding the calls in asterisk
13:05.16*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
13:05.19MJR^where is the setting for codecs in asterisk
13:06.03Thumannsip.conf
13:06.12Thumannbut it should be done transparent dude..
13:06.16BrianR___Anyone have experience with making sure the hangup status gets propagated when routing calls from PRI to PRI?
13:06.36BrianR___Ie, I want the out-of-band signalling related to a call to get propagated.
13:06.50MJR^allow=ulaw? thumann
13:07.00MJR^thats the setting
13:07.03*** join/#asterisk gonzo- (n=gonzo@195.140.246.50)
13:08.09gonzo-hi. Can anyone clarify me whether "Billion 1 Port S0 Card" and "Billion HFC-S PCI ISDN card  (BiPAC PCI)" are the same?
13:09.28konradsgonzo-, so it seems
13:09.52Duschinger?
13:10.10bjohnsonMJR^: you can control it by device.  Start with disallow=all and then allow the ones you want
13:10.37*** part/#asterisk jaike (n=a@203.131.137.76)
13:10.44bjohnson\/me picks up his fatass tuba
13:10.57mmlj4morning manxpower :-)
13:11.01MJR^where is the setting for codecs in asterisk while forwarding it to some ip
13:11.30bjohnsonMJR^: you've been told
13:12.23tzangerBrianR___: it should just propagate
13:12.41tzangerbjohnson: you have a fat ass?
13:12.45tzangerbjohnson: OH... tuba
13:12.50tzanger:-)
13:13.12MJR^bjohnson: ok sir ... we were using asterisk to for3ward calls to n2p now we r using vsnl  it works fine with n2p but i am not able to plce calls on vsnl plzz help me..
13:13.30tzangerMJR^: please.  do yourself a HUGE favour and read the asterisk handbook draft, then go to www.asteriskdocs.org and read.  If you're feeling masochistic head on over to the wiki afterward
13:13.34tzangerwe're here to help, not hand-hold.
13:13.46ManxPower'morning mmlj4
13:13.49*** join/#asterisk aneredes (n=hannes@port-212-202-55-34.dynamic.qsc.de)
13:14.10ManxPowerBTW, mmlj4 is also a katrina refugee
13:14.18tzangerwow
13:14.35MJR^tzanger": who asked u
13:14.49ManxPowermmlj4, Have you been able to get back to your place yet?
13:14.58bjohnsonMJR^: he gave you the answer \
13:15.01ManxPowerMJR^, Be nice or nobody will help you.
13:15.08bjohnsonMJR^: if it's sip, sip.conf
13:15.27mmlj4yes... no water damage, only little wind damage to the building (which may include the air conditioners)
13:15.29ManxPowerUgh.  I have hives all over my hands
13:15.39gordonjcpmmlj4: not bad
13:15.43ManxPowermmlj4, Cool.  Still no power or broadband?
13:15.47bjohnsonMJR^: I don't know what n2p or vsnl are.  I assume they are voip providers
13:15.59mmlj4dunno, i haven't been in a week
13:16.04MJR^yes it is ..
13:16.04ManxPowermmlj4, Ah.
13:16.06tzangerMJR^: you are asking VERY basic questions that show that you don't have the requisite understanding of how to set this up.  Instead of doing the research and asking intelligent questions you want us to just make it work so you can forget about it until the next problem crops up.  Many of us will do that, but it falls under "they don't care to learn, so we're going to charge."  Pretty simple.
13:16.08mmlj4hopefully will go in again tomorrow
13:16.17MJR^am sorry i was totally frustrated sorry
13:16.20tzangerManxPower: I told you that hooker was trouble
13:16.30tzangerManxPower: be thankful it isn't hives elsewhere
13:16.37MJR^becoz it isint workin its eating ma head
13:16.40ManxPowertzanger, *bap*
13:16.44tzangermaybe you best pee sitting down until the hives subside
13:16.59MJR^ManxPower:am sorry
13:17.00gordonjcpmmlj4: wind damage is a damn sight easier to fix
13:17.08tzangerMJR^: and it will continue to eat your head until you get your head around the basics.  We've all been there, trust me
13:17.09Hmmhesayshave no fear Hmmhesays has arrived
13:18.11mmlj4ManxPower: you wanna try another call?
13:18.19tzangerbjohnson: you have no idea the sheer size of the spiders around my house lately
13:18.32tzangerI mean for garden spiders they are HUGE (2.5" across)
13:18.37gordonjcptzanger: what is it with huge spiders this year?
13:18.39bjohnsonthat IS bug
13:18.43bjohnsonerr big
13:18.45bjohnsonerr big bug
13:18.47gordonjcpit's a bug too
13:18.54tzangerand they make webs across the windows as if to say "if I could figure out how to, you'd be next."
13:19.12bjohnsonwe have a lot too, but not THAT big
13:19.22*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:19.22*** mode/#asterisk [+o anthm] by ChanServ
13:19.22tzangerthey hang in the middle of them in the night because the light from the kitchen draws the bugs to the windows... they aren't stupid that's for sure
13:19.36Hmmhesaysi need to think up a good way of blocking callerid in the dialplan
13:19.49Hmmhesayswhat did I just type
13:19.50gordonjcptzanger: it's probably a lot milder where you are, but last time I was up at my Mum's I found my cat chasing a spider about the size of my thumbnail, excluding the legs
13:19.53*** join/#asterisk zotz (n=zotz@24.231.36.100)
13:19.55tzangerI had a really big one on my satellite dish... he had a web between the LNB and the reflector, 20 feet off the ground.  I have no idea what the hell he was catching up there but he was on to something, that was for sure
13:20.01HmmhesaysI need to think up a good way of blocking call waiting in the dialplan
13:20.21tzangergordonjcp: yes that was the size of this one in the dish...  their abdomen is as big as my thumbnail or a little larger
13:20.28gordonjcpyeah, about that
13:20.32tzangerI absolutely HATE walking down the steps and getting webbed
13:20.57gordonjcpit was funny as hell, the cat would go chasing after it, and get his paws in front of it, trapping it but not actually pouncing on it
13:21.11gordonjcpthen the spider would climb over his paws and the cat would spook
13:21.15ManxPowertzanger, 2.5in or 2.5cm?
13:21.27Ariel_Hmmhesays, like setgroup & group counts?
13:21.28FreezeS2.5 inch is a mouse !
13:21.56Hmmhesaysyeah Ariel_ not a bad idea
13:21.57tzangergordonjcp: :-)  yeah my puppy eats the daddy longlegs spiders (the harvester spiders) and the kitten is too little yet
13:22.02tzangerManxPower: 2.5in not cm
13:22.13ManxPowertzanger, You canadian rebel!
13:22.14tzangerFreezeS: well that's legspan too not just abdomen
13:22.19tzangerabdomen is the size of my thumbnail
13:22.25HmmhesaysAriel_, I got the ldap stuff working pretty sweet, now its on to features
13:22.43tzangerHmmhesays: awesome I'm definitely bugging you for that setup
13:22.43*** join/#asterisk wuwu (n=wolfgang@81.223.6.242)
13:22.46tzangerManxPower: :-)
13:22.46ManxPowertzanger, yellow and black spiders?
13:23.08Ariel_Hmmhesays, great
13:23.13tzangerit's amazing, I really like metric but inches/feet and acres are all "familliar" measurements to me.  oF and miles don't "work" for me
13:23.20wuwuhi all, does anyone here knows how i can change the value of the variable EXTEN in the dialplan ? - Using SET(EXTEN=0293402934) does not work...
13:23.30ManxPowerwuwu, you can't.
13:23.40FreezeSI like spiders... the biggest I saw was only about 4 cm (legspan)
13:23.43ManxPoweruse Goto(0293402934,1)
13:23.47tzangerManxPower: no, they're pale coloured with blackish markings... one's reddish rather than pale
13:23.54tzangerthey're just common garden spiders on steroids I think
13:24.16wuwuManxPower, that would also be possible...
13:24.38KattyManxPower: beep!
13:25.21tzangerhttp://212.84.179.117/i/Garden Spider.jpg
13:25.30tzangerlike that but with longer (50% longer?) legs
13:25.39tzangerand some paler, one redder
13:26.37crash3mwhats the difference between wav and wav49?
13:26.53*** part/#asterisk pablix (n=root@lacnic.net.uy)
13:27.00mmlj4crash3m: wav only does it once
13:27.15anthmwav49 is gsm and the other one is raw pcm
13:27.36crash3mheh
13:27.47crash3manthm: ty
13:27.52anthmnp
13:29.40ManxPowermmlj4, not now
13:30.33Ariel_does anyone know where I can find the auto generator for the tdm400p boards.? I need to set it up on a remote system.
13:31.27RoyKanyone here tried isdn early media with cvs head?
13:32.25tzangerwow I didn't know that there were that many venomous spiders (significantly toxic to humans) in Ontario
13:32.30tzangergoogle and google images rocks
13:32.34tzangerAriel_: "auto generator" ?
13:32.58jake1932<PROTECTED>
13:33.20Ariel_tzanger, yes asterisk@home has one that will detect your fxs and fxo's on the tdm400p
13:33.26Ariel_jake1932, no
13:33.31tzangerAriel_: dmesg output will do that :-)
13:34.52tzangerI'm pretty sure I've seen the hobo spider (Tegenaria agrestis) around my house
13:34.56tzangerthat's not good
13:37.08jaxkzcan someone name a few corporate sizes voip=
13:37.14jaxkzcan someone name a few corporate sizes voip->pots converters?
13:37.29Hmmhesaysyou mean fxo gateways?
13:37.34*** join/#asterisk Snake-Eyes (n=blog@203.201.96.147)
13:37.37ManxPowerCan anyone recommend a cell data service provider with good coverage in the south?
13:37.41bjohnsonor channel banks?
13:37.47jaxkzchannel banks
13:37.57jaxkzwich can be hooked up to a trunk
13:38.01ManxPowerCingular max 56K, Alltell, Verizon, and Sprint are max 2Mbps with 400k typical
13:38.10Hmmhesaysjaxkz, I think you mean gateways
13:38.25Hmmhesaysvoip-->trunk
13:38.26jaxkzerm.. yes sorry
13:38.44jaxkzI would like to hook up some old Avaya pbx to the asterisk pbzx
13:38.45Hmmhesaysaxt2400 for fxo,  1124 for fxs
13:38.54Hmmhesayson the station side or trunk side?
13:39.01jaxkztrunk side
13:39.10jaxkzvoip inbound on the trunk
13:39.25Hmmhesaysthen to the pbx, where it rings
13:39.32Hmmhesaysand vice versa?
13:39.39jaxkzoptional
13:39.58Hmmhesaysso voip-->trunk-->station
13:40.21Hmmhesaysstation aka: user handset
13:40.27jaxkzyep
13:40.32Hmmhesaysmediatrix 1124
13:40.51jaxkzthose are analog
13:41.03Hmmhesaysyeah
13:41.08jaxkzah
13:41.12jaxkzthey also have PRI
13:41.20Hmmhesays<jaxkz> can someone name a few corporate sizes voip->pots converters?
13:41.38jaxkzsorry. i misplaced alot of terms
13:41.59Hmmhesaysfor digital you can get a t1 card for asterisk or you can use an external gateway
13:42.01jaxkzMediatrix Single E1/PRI -> pretty expensive :)
13:42.09Hmmhesaysall single pri gateways are
13:42.35*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:42.35Hmmhesaysbut you get what you pay for
13:42.41jaxkzmost likely
13:43.18Hmmhesaysbut also have their advantages
13:43.23jaxkza digium card would be more cost effective
13:43.37brad_msswHmmhesays: any idea when vonage is going to officially launch their business plus service ??  you mentioned it to me a while back, but they couldn't transfer even their own phone numbers to the business plus stuff at that time
13:43.44*** join/#asterisk focks (n=craigb@nsc66.147.95-93.newsouth.net)
13:43.44Hmmhesaysif you want multipath switching... IE you pick what numbers come and go voip and are able to leave the original PRI in place, then a quintum dx2024 is the way to go
13:43.48jaxkzi could hang it in the trunk and use the voip possibilities
13:44.04Hmmhesaysbraw_mssw let me ask
13:44.08*** join/#asterisk mbranca (n=matteo@81.208.92.210)
13:44.14KattyHmmhesays: i've been invited to a party!
13:44.27HmmhesaysKatty: sweet
13:44.35KattyHmmhesays: it's a girly party though.
13:44.46KattyHmmhesays: i've probably been invited to cook :/
13:44.51Hmmhesayslike college coed girly or tupperware party girly
13:44.59Kattymore like sex toy girly party
13:45.09Hmmhesaysrock
13:45.13focksi have user A who forwards his phone to user B when A is away from his desk. in the event that A is forwarded to B and B does not answer, is there any way to have A's voicemail pickup instead of B's?
13:45.16Hmmhesaysso a mix of the two
13:45.30Hmmhesaysjaxkz I do a lot of legacy integration using that unit
13:45.52Hmmhesaysfocks yeah
13:46.00brad_msswfocks: sure, but you'd have to script it out in your dialplan to do so
13:46.03jaxkzAlso on Avaya INDeX?
13:46.04Hmmhesayschannel variables
13:46.18Kattywith their complete lack of vegan lotions and lots of leather
13:46.19Hmmhesaysjaxkz when you are doing trunk side integration it doesn't matter what you are plugging into
13:46.23Kattyall sorts of things i wouldn't buy anyway
13:46.44Hmmhesaysbeen awhile since i've got my freak on
13:46.45focksHmmhesays or brad_mssw can you point me in the right direction?
13:46.57Kattyheh, and can you imagine me buying it from my best friend?
13:47.03Kattyshe'd freak out if she knew i was bi
13:47.05Kattysilly missouri
13:47.09Hmmhesaysfocks, set a channel variable that says this is a transferred call from "A" use A's voicemail
13:47.10jaxkzHmmhesays: It also doesn't matter on line-side right?
13:47.13brad_msswfocks: just set a variable when you do the forward, and check that variable before you go to voicemail
13:47.25Hmmhesaysline side == station side?
13:47.35jaxkzpoint of entrance
13:47.47focksi see, thanks
13:47.49Hmmhesayspoint of entrance for what
13:47.58jaxkzfor the external pri line or your pbx
13:48.11jaxkztelco line
13:48.25Hmmhesaysfocks, that would be the theory, look at the source in docs for the README.variables
13:48.42Hmmhesaysthat would be the same as the trunk side i'm talking about it seems jaxkz
13:49.13jaxkzI see what you mean
13:49.17focksanother question, does xfersound=beep only work in certain versions of Asterisk? and which would those be?
13:49.32Hmmhesayswhenever they started using features.conf
13:49.56Hmmhesaystrunk side = side the telco is on, station side = the side the users phones are on
13:50.04focksHmmhesays would it be in 1.0.7 stable, or just CVS?
13:50.09*** join/#asterisk MattH (n=MattH@63.174.244.174)
13:50.14jaxkztrue
13:50.35Hmmhesaysif you want to do a seemless voip integration on the trunk side, where the user never knows their call is getting routed out on voip, that is what I do
13:50.44Hmmhesayswell theoretically they never know
13:50.46MattHHi... I'm getting 'unable to negotiate codec - cause code 58' when trying to connect to an IAX terminator I have used for awhile now just fine.   Can anyone enlighten me as to what a cause code 58 is?   I'm running CVS-HEAD from about 2 weeks ago
13:50.47jaxkzyes. That's my goal as well
13:51.44jaxkzTo bad extra trunks for legacy systems cost a lot of money
13:52.02Hmmhesaysjaxkz why you need extra?
13:52.13jaxkzcause none are free atm
13:52.19Hmmhesaysyou put a voip gateway inline and pick out the numbers you want to route voip
13:52.35Hmmhesaysotherwise pass all other numbers to and from the telco
13:53.14jaxkzThe PBX doesn't have any spare PRI connection left. So there is no way of hooking up the gateway without a connecction point
13:53.17*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
13:53.25Hmmhesaysheh, read what I just said dude
13:53.52*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
13:54.00jaxkzi did, doesn't make sense to me
13:54.06jaxkzwhat do you mean by "inline"
13:54.24Hmmhesaystelco pri-->gateway-->telco pri
13:54.46jaxkzHow would you hook up the legacy pbx then
13:54.59Kattyi'd like to know why people get all pissed off at you when you don't want to talk to them.
13:55.12Hmmhesayspri-->gateway-->pri--pbx
13:55.21Hmmhesaysinstead of pri-->pbx
13:55.24Kattylolz urr hott, wanna chat?
13:55.30Hmmhesaysomgz yesssssss
13:55.34jaxkzWe have over 200 lines
13:55.37Kattyno, you shallow minded illiterate freak.
13:55.56Hmmhesaysjaxkz is that a problem?
13:56.01Kattyand then they're all BYE YOU SLUT
13:56.05Kattywhat's up with that?
13:56.07*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
13:56.10jaxkzhmm.
13:56.19tzangerKatty: its simple
13:56.22jaxkzstations have to be able to call over the voip connection
13:56.26tzanger'they're pizza-faced 13 year olds who think they're don juans
13:56.32Kattyah, right. k
13:56.46jaxkzso, telco -> gateway -> telco totally bypasses the stations ability to use the viop unit
13:56.55FreezeSKatty, how old are you ?
13:57.00Mimmusanyone using SMS() app with Asterisk?
13:57.04Hmmhesaystzanger: try 25 year olds who have had any chance of a social life zapped away from the intarwebs
13:57.04Kattytzanger: actually. he was 23
13:57.07bjohnsontzanger: not at all like us 30+ don juans
13:57.10Kattytzanger: or at least, that's what his profile said.
13:57.10tzangerFreezeS: she's old enough to know better than to answer those kinds of questions.  :-p
13:57.20Hmmhesaysjaxkz, have them dial a prefix
13:57.21tzangerbjohnson: shhhhhhhhhhhhhhhhhh
13:57.28tzangerbjohnson: if they know how old we are we'll never get any :-p
13:57.32Hmmhesaysjaxkz or have them do it seamlessly
13:57.34bjohnsonany what?
13:57.40Hmmhesayssammichs
13:57.40KattyFreezeS: twice as old as half.
13:57.43jaxkzHmm-home: yes, i was thinking of that
13:58.05bjohnsonthe only thing I want these days is Applebees riblets
13:58.06Hmmhesaysso you have 8 pri's coming in?
13:58.08bjohnsonmmmmmm
13:58.16bjohnsonnow I'm hungry
13:58.16jaxkzso the gateway needs to pickup the prefix
13:58.23FreezeSKatty: why do you answer people you don't want to talk to ?
13:58.38jaxkz5 PRI's, 1 Data pri and 1 other pri for EuroRings
13:58.44bjohnsonFreezeS: let's see if she answers you.
13:58.52Kattybjohnson: i was just thinking that ;)
13:59.00HmmhesaysI hate to break it to you dude, but thats not 200 lines
13:59.04FreezeSbjohnson: she answered once already :)
13:59.13KattyFreezeS: why are you expecting an answer then?
13:59.17jaxkzcomes close
13:59.20jaxkz180 then
13:59.21KattyFreezeS: and why do people ask questions if they don't expect an answer?
13:59.31bjohnsonmmmm .. riblets
13:59.38bjohnsoncan't get them out of my mind now
13:59.49FreezeSKatty: maybe because they want to be your bf... or something....
13:59.50Kattynah nah nah can't get you out of my head.
13:59.56Katty^_-
14:00.18mutilatoranyone else know of a problem with the internal DB using extension functions and CLI?
14:00.21KattyHmmhesays: geesh, can't you /please/ not do that in the middle of the room
14:00.36KattyHmmhesays: i didn't mind at the strip club..
14:00.42KattyHmmhesays: but that was at least a wall!
14:00.49*** join/#asterisk hassler (n=hassler@r-corp.hcst.com)
14:00.56mutilatorusing the dialplan i get no value, says not found
14:01.05mutilatorbut when i do the query at the CLI i get a value returned
14:01.15Hmmhesayssyntax error
14:01.20mutilator<PROTECTED>
14:01.20mutilator<PROTECTED>
14:01.27mutilatordatabase get SIP Registry/9893332469
14:01.28mutilatorValue: 172.16.0.7:6054:1800:9893332469:sip:9893332469@65.111.201.79:6054
14:01.49Hmmhesaysyou and me gone fishing in the dark
14:02.11mutilatori don't see why it's doin that
14:02.14tzangerHmmhesays: that's a good song
14:02.15*** join/#asterisk spackle (n=spackle@209.234.83.19)
14:02.16hasslerquick dialplan question: I have a few extensions defined as "*43" (for example), but when I dial them, I get a fast busy as if they are not recognized. The standard Zap prefixes (*67 to turn off callerID) are apparently recognized directly by the card though. I'm sure I'm missing something easy
14:02.17Hmmhesaysjaxkz: if you want some info, I do that type of integration all the time and am for hire
14:02.31KattyHmmhesays: you're for hire?
14:02.35Hmmhesaystzanger: yeah it is, we got a little drunk saturday night and I busted out out the guitar
14:02.38HmmhesaysKatty: always
14:02.39KattyHmmhesays: hot dang, i could use an assistant!
14:02.49Hmmhesaysoh can I CAN I!
14:02.57Katty;P
14:03.03tzangerHmmhesays: I am trying to find the name and artist of the song that is like "I'm addicted to/all things that you/make me feel inside" in the chorus -- the country station DJ said it was JD Vature and the name was Addicted but I can't find that artist anywhere and addicted is such a useless word to search on
14:03.28tzangerHmmhesays: yeah I am learning some uncle tupelo (actually the song was originally Bob Dylan) -- I love Moonshiner as Uncle Tupelo does it
14:03.38KattyBob Dylan :<
14:03.40tzangerit's helping me iwth my picking/augmentation
14:03.41Kattycan't stand his voice.
14:03.45jaxkzHmmhesays: i don't think you're dutch ;)
14:03.49Kattygood write though.
14:04.00Kattys/write/writer/
14:04.05dave99Addicted To Your Love - Jagged Edge
14:04.14Hmmhesaysjaxkz: i dunno dude this is the intarwebs I do most of my stuff off site
14:04.33tzangerKatty: well I didn't listen or play the original
14:04.34Hmmhesaystzanger: i've just been brushing up on a lot of rock, thinking about firing up a band again
14:04.39jaxkz... dutch... you know. ppl from The Netherlands? :)
14:04.45mutilatorah
14:04.46jaxkzI think you know if you are dutch or not
14:04.49mutilatori see the error
14:04.50mutilatormeh
14:05.24tzangerthe uncle tupelo stuff is all "off" chords..  7ths and augments/diminished stuff... I have no idea what chords I'm playing since they're not standard but damn it's fun to play
14:05.32Kattytzanger: (=
14:05.55tzangermy pinky is getting a deep cut/blister combo from the E1 string :-)
14:05.55Hmmhesaysjaxkz: I know i'm not dutch, but this is the internet with that wonderful tool it enables people to work together from around the world, while they are sitting in their boxer shorts eating cocoa crispies
14:06.03KattyHmmhesays: i don't have boxer shorts.......or cocoa crispies kthx
14:06.13HmmhesaysKatty: you're not invited then
14:06.20tzangerI prefer jockeys and corn pops or coffee
14:06.23KattyHmmhesays: sad.
14:06.34HmmhesaysKatty: right in the heart kind of sad
14:06.49KattyHmmhesays: i don't have a heart. that's steel.
14:06.56tzangerI really enjoy playing the bluesy/bluegrass kind of music but then I pull out hte mustang and the distortion and just make a lot of noise too sometimes :-)
14:06.57Hmmhesaysnot sad: they let that mentally challenged guy drive a vespa sad
14:07.21spackleLOL
14:07.22jaxkzHmmhesays: that's kinda true ;) I will get you cookies when you are able to fysicly install my hardware over the internet :)
14:07.23KattyWhat's sad is I hold this company together and don't get paid crap
14:07.37tzangerKatty: reminds me of the simpsons "who shot mr burns" where that old guy (cornelius?) gets shot in the leg and he pulls off his fake leg and says "that there's solid oak!"
14:07.40tzangerhahahaha
14:07.53Hmmhesaysjaxkz: I don't install hardwire, that is what interns are for
14:08.03Kattytzanger: that must have gone over my head cause i didn't find it funny.
14:08.07spackleKatty: just go into a coma fro a week and they will see how much they need you.
14:08.11Hmmhesaystzanger I remember that well
14:08.15tzangerthat old guy is one of my favourite characters
14:08.19Hmmhesaysfamily guy so trumps the simpsons though
14:08.19tzanger"who did what in the where now?"
14:08.25jaxkzI wonder what kind of hardware an major telco uses
14:08.33Kattyspackle: uhh, no.
14:08.34tzangerHmmhesays: family guy is good... much "edgier" humour... I wouldn't let the kids watch it
14:08.40tzangerstewie rocks
14:08.48Hmmhesaystzanger: no kids here so no problems
14:08.57Hmmhesaysquagmire rocks... have you seen the movie?
14:09.06tzangerno
14:09.07spacklewhat movie?
14:09.11Hmmhesaysthe family guy movie
14:09.14mutilatorgiggidy giggidy
14:09.15hasslerquick dialplan question: I have a few extensions defined as "*43" (for example), but when I dial them, I get a fast busy as if they are not recognized. The standard Zap prefixes (*67 to turn off callerID) are apparently recognized directly by the card though. I'm sure I'm missing something easy
14:09.20Kattychicky chicky
14:09.23spacklewhat?  There is a Family Guy Movie?
14:09.31tzangerhassler: not recognized by the card.  defined in features.conf IIRC
14:09.50Hmmhesays"stewie griffin: the untold story"
14:10.02spackleI'm on that like Oprah on a canned ham.
14:11.45Hmmhesaysin one part quagmire gets a motorhome and it shows stewie and brian walking up...  quagmire comes into the shot and says "quagmires cross country cruiser" or something like that... then they say "wait isn't their an O in country?" - "no"-quagmire
14:12.09tzangerHmmhesays: heh
14:12.18tzangerreminds me of mclean and mclean skits
14:12.31KattyHmmhesays: you know, i'm quite impressed really
14:12.37KattyHmmhesays: asterisk has only exploded once.
14:12.41HmmhesaysKatty: about?
14:12.45KattyHmmhesays: and all it required was a reboot.
14:12.48Hmmhesaysyeah, it'll do that
14:12.53Hmmhesayswatch your memory usage
14:12.59KattyHmmhesays: do you know how many times i have to reboot that windows server?
14:13.00spackleHmmhesays: reminds me of that episode with wheel of fortune - Go _uck yourself __  -> Go tuck yourself in.
14:13.11tzangerspackle: HAHAHAHAHAHAHAHAHA
14:13.11Hmmhesayslol
14:13.19tzangertoo bad the 'F' was already there
14:13.39Hmmhesaysspackle: i dunno I think you gotta get that movie by questionable means
14:13.40tzangerGO _UCK YOURSEL_ __
14:13.50HmmhesaysKatty: eleventy billion about
14:14.00tzangerI'M _UPID - Alex, I'd like to buy a vowel...
14:14.06*** join/#asterisk dec (n=tom@ppp225-27.lns2.adl4.internode.on.net)
14:14.07KattyHmmhesays: yub yub
14:14.16tzangerer fuck I fucked the joke up
14:14.20tzangerI'M _TUPID - Alex, I'd like to buy a vowel...
14:15.28hasslertzanger -- not finding anything applicable in features.conf
14:16.13Hmmhesaysapplicable to what?
14:16.15dave99tzanger, never mess up tupid jokes.  It makes you look tupid.
14:17.21Hmmhesayswow asterisk is not logging cdrs anymore
14:17.26Katty:<
14:17.27Hmmhesaysfun FUN
14:17.31mutilatoranyone know how GROUP_COUNT() works?
14:17.31spackle"My hairy Aunt"
14:17.38jaxkzdoes someone know an sms inbound service?
14:18.21jake1932jaxkz: clickatell.com?
14:18.27Kattyjake1932: beep!
14:18.30_omerclickatell.com
14:18.35Kattybeep!
14:18.42jaxkzokie
14:18.59jake1932beep?
14:19.06Kattyjake1932: you're setting of my hilight.
14:19.09tzangerdave99: I know..  heh
14:19.10jake1932do i leave a message?
14:19.12Kattys/of/off/
14:19.16*** join/#asterisk elriah (n=bill@adsl-068-209-198-242.sip.bhm.bellsouth.net)
14:19.21*** join/#asterisk valence (n=valence@Quebec-HSE-ppp230300.qc.sympatico.ca)
14:20.25jake1932Katty: i've never set anyones hilight before - is that perverted?
14:20.37elriahHi all.  I'm running a couple remote phones and when I do sip show peers the ports listed for these phones are  in the 38000 range.  I have my rtp.conf set to use 10000-20000.  What determines these sip ports?  My local phones show 5060.
14:20.38Katty...
14:20.57Kattymaybe i'm just unusally grumpy today.
14:21.07Kattyjake1932: i'm going to hush now before i take your head off (=
14:21.25jake1932Katty: did i say something wrong?
14:21.33Kattyjake1932: nope, just generally stupid (=
14:21.41Mimmusanyone using SMS() app?
14:21.58dave99I know, clickatell.com!
14:22.15dave99beep
14:22.28cioHi all.  I'm running a couple remote phones and when I do sip show peers the ports listed for these phones are  in the 38000 range.  I have my rtp.conf set to use 10000-20000.  What determines these sip ports?  My local phones show 5060.
14:23.02jaxkzfunny. It's south-african
14:24.18jake1932Katty: oh - like a game show type thing? I get it
14:24.26Katty^_-
14:24.28Kattyk
14:25.55*** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
14:25.56tzangercio: NAT?
14:26.03ciotzanger: I have nat=yes in my sip.conf, yep.
14:26.26*** join/#asterisk rob314 (n=rob314@207.58.194.2)
14:29.01Hmmhesaysand I want it and I need it i'm addicted to you
14:29.09dave99So, who else has their Asterisk crash regularly when running restart?
14:29.24Hmmhesayswhat is crashing it?
14:29.46Hmmhesaysaha!
14:30.03dave99Good question.  restart now just sits there, and after a bit everything stops working.
14:30.12KattyHmmhesays: how hard is it to set up the thingy where 3+ people can be in the same conversation
14:30.21*** join/#asterisk mjr^^ (n=muhajir_@210.212.195.134)
14:30.33KattyHmmhesays: like oversized conference thingy
14:30.35HmmhesaysKatty: purdy simple
14:30.40KattyHmmhesays: is that meetme?
14:30.44Seyrtakes like 30 seconds
14:30.45Hmmhesayscan be
14:30.45spackleKatty: no fair, you're getting all technical on us.
14:30.58KattyHmmhesays: is there something else that'll do that too?
14:31.12Kattyspackle: oh noes?
14:31.57Hmmhesaysthere is a fair amount of conference bridging software out there,
14:32.03KattyHmmhesays: just like where one person can dial a long distance number, and eleventy billion people can dial extension foo and listen too
14:32.11Hmmhesaysbe easiest for you to use meetme though
14:32.15Kattyk
14:32.30HmmhesaysI had 40+ calls into meetme with transcoding on this laptop
14:33.04file[laptop]FOOD! NEED FOOD
14:33.07file[laptop]yay cookie
14:33.28KattyHmmhesays: ooooh.
14:33.35KattyHmmhesays: so it's just give the extension and password?
14:33.41KattyHmmhesays: and it's all done?
14:33.53mutilatorif a macro1 calls a macro1, goto(${macro_context},200) should still go to macro1,200 priority shouldn't it?
14:34.05KattyHmmhesays: do i need to somehow link it to extensions.conf?
14:34.23mutilatorcause mine is skipping to n+1 in macro1 once i do the goto
14:34.43HmmhesaysKatty: extensions.conf is where you would define that
14:35.04KattyHmmhesays: not meetme.conf?
14:35.12bjohnsonboth
14:35.12Hmmhesaysexten => _XXX,1,Meetme(1|d)
14:35.12KattyHmmhesays: or to include meetme.conf?
14:35.16SeyrKatty: There are examples for extensions.conf and meetme.conf here: http://www.voip-info.org/wiki-Asterisk+cmd+Meetme
14:35.23HmmhesaysKatty: I usually only use dynamic conferences
14:35.33KattyHmmhesays: k
14:36.01spackleHmmhesays: what other conference bridging software is available?
14:36.06Hmmhesaysopen mcu
14:36.08Hmmhesaysfor h323
14:36.34spacklefor sip & zap?
14:36.41KattySeyr: murr, ow?
14:37.35SeyrKatty, sorry, i only speak english :-(
14:37.43KattySeyr: i noticed.
14:37.44Hmmhesaysi'm drawing a blank here
14:37.48KattySeyr: Hmmhesays speaks kat.
14:37.55KattySeyr: wiki rarely speaks kat
14:38.01Seyr:-(
14:38.54Wonka*mrau*?
14:39.54Kattythis is not a valid conference number, please try again!
14:40.04Hmmhesays1|d?
14:40.06tzangerWonka: what's that a french cat?
14:40.08Kattyshe should have said, you screwed up meetme, please don't act like an idjit!
14:40.20spacklekatty: do you have Zap hardware?
14:40.49Kattyaww, look who's trying to be all helpful!
14:40.52Wonkatzanger: no. it's wrong to be french.
14:40.55spackle;P
14:41.13KattyHmmhesays: yay, it works
14:41.17KattyHmmhesays: thanky.
14:41.20Hmmhesaysnp
14:41.44KattyHmmhesays: you can dial my eeeeeks number, but with 500 to join conference (=
14:42.31Hmmhesayswhat did I have set for you
14:42.33Kattyi dunno
14:42.36Hmmhesayshrm
14:42.38Katty2000?
14:42.50file[laptop]woot Emergency Alert System Test!
14:43.03Kattyfile[laptop]: please stand by while we inspect all system.
14:43.18Kattyfile[laptop]: in the event any cookies disappear, please don't panic and leave the building as calmly as possible.
14:43.31Kattyfile[laptop]: thank you and have a great day
14:44.51dave99You know I like the way you make me feel inside
14:44.51dave99Baby I jus can't shake ya, I'm addicted to your love
14:44.59spackle...... and vanishes in a puff of logic
14:46.22Hmmhesays500 doesn't answer
14:46.36file[laptop]Hmmhesays: I forwarded it to the whitehouse situation room
14:47.40file[laptop]ugh I'm hungry
14:48.57dave99Are the 1700 numbers through iaxtel down totally?  Cause I set it up yesterday and haven't been able to get a connection once.  Might be my firewall.
14:49.09file[laptop]iaxtel rarely works... just like yeah
14:49.24dave99Well.  That stinks.
14:50.03dave99How about FWD, is that worth playing with?
14:50.49Assidanyone seen JerJer, shido around?
14:50.49*** join/#asterisk cio (n=bill@adsl-068-209-198-242.sip.bhm.bellsouth.net)
14:50.59jake1932~seen JerJer
14:51.04jbotjerjer is currently on #asterisk (10h 17m 57s).  Has said a total of 163 messages.  Is idling for 9h 36m 18s
14:51.15jake1932beep
14:51.28cioHi all.  I have a couple remote SIP phones (Polycom IP300's and 301's).  The can call out just fine.  Using nat=yes.  They can receive calls for a few minutes and then they become unreachable (sip show peers).  Any suggestions?
14:51.53*** join/#asterisk tguid (n=tguid@user-10bj0jo.cable.mindspring.com)
14:52.05hasslerdoes the ZAP channel pass any numbers beginning with * into the dialplan, or does it intercept them all?
14:52.39Seyrcio: do you have qualify set? is it a low number?
14:52.46cioIt's set to 250 right nwo.
14:52.50cionwo = now
14:52.53tguidhey all i have a digium 4 port t1 card and still trying to make sense of how to configure signalling and such
14:53.03tzangerhassler: works for me..  I use *1/*2/*3 for special functions
14:53.13*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
14:53.14Seyrcio: might just be slow between the phones and they server, increase qualify time and watch
14:53.17PakiPenguinevening
14:53.27tguidany good resources for diagnosing?  zttool gives red, debug=1 option given in modprobe and don't get too much
14:53.29cioSeyr - thanks, will do.
14:53.34dave99tquid, digium offers free install support
14:53.43tguidseems like it doesn't recognize the line at all, verizon t1
14:53.52cioquit
14:54.43tguidbtw using latest zaptel from cvs as this is a te411p card
14:55.05hasslertzanger -- mine still isn't working. I'm sure it WAS, but not sure what broke it!
14:55.19tguidasterisk libpri are 1.0.9 apt-get source and made packages for debian sarge as sarge was still 1.0.7
14:55.49tguiddave99 thanks i may very well call digium up for config clues, docs are scattered and scant
14:56.09mutilatorwhen using setgroup, it decrements the group when the channel is hungup correct?
14:56.18tguidvoip-info.org wiki has some good pieces but no good, detailed install diag docs exist
14:56.23spackletguid, it really is all on the wiki and in the config files.
14:57.06spackletguid, and a little bit on the digium website to0.
14:57.32*** join/#asterisk mmmToop (n=chatzill@c1-66-2.rndf.isadsl.co.za)
14:58.31*** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net)
14:59.08*** join/#asterisk epablo (n=epablo@200.75.139.188)
14:59.18epabloHi people
15:00.08epabloI'm having some problems with perl Asterisk::AGI on FC4.  It just doesn't do anything.  Any ideas on what can be done?
15:00.19epabloat least to debug
15:00.20*** join/#asterisk cio (n=bill@adsl-068-209-198-242.sip.bhm.bellsouth.net)
15:00.22olivier_tquid : are your wt4xxp loaded ? are ztcfg -vvv ok ? what give you pri debug span ? have you try to link span 1 to your span 2 in order to check your conf and your card etc ...
15:00.46cioNo luck with increasing my qualify.  Still calls them UNREACHABLE after 10-15 seconds.  hrm....
15:01.15tguidolivier_ yes wtc4xxp loaded and ztcg -vvv looks good
15:01.39cioI think it's firewall related.
15:01.57tguidthe verizon t1 line is a pri line
15:02.04cioI see incoming udp messages on the nat port being denied ...
15:03.02tguidand i tried the various combos esf instead of d4 and ami instead of b8zs
15:03.31spackletguid, are you in US or other country?  Is it s T1 or E1?
15:04.29*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
15:04.46spacklespan=1,1,0,esf,b8zs
15:05.25*** join/#asterisk convey (n=test@66.55.43.2)
15:07.00tguidus and my span lines look like that
15:07.20cioIs there a way to configure the polycom ip phones to send udp keep-alive packets?
15:07.29*** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
15:07.52*** join/#asterisk Gerriall (n=NonYa@209.42.198.18)
15:07.55spackletguid, what distro are you using?
15:08.25tguiddebian sarge, i apt-get source for asterisk 1.0.9, zaptel driver from cvs and libpri also 1.0.9
15:09.18*** join/#asterisk CMike (i=daemon@c-f54171d5.116-1-64736c10.cust.bredbandsbolaget.se)
15:09.32spackleOK, so how are you loading zaptel and do you get any messages from it?
15:10.45sivanaanyone know of a PRI usage grapher?
15:10.47sivanasoftware
15:11.40tguidSep 14 15:10:04 mojo kernel: TE4XXP: Span 1 configured for ESF/B8ZS
15:11.40tguidSep 14 15:10:04 mojo kernel: wct4xxp: Setting yellow alarm on span 1
15:12.15cioquit
15:13.03tguidSep 14 15:10:04 mojo kernel: VPM: Span 3 U-law mode
15:13.03tguidSep 14 15:10:04 mojo kernel: VPM: DTMF threshold set to 1000
15:13.03tguidSep 14 15:10:04 mojo last message repeated 4 times
15:13.03tguidSep 14 15:10:04 mojo kernel: VPM: Present and operational
15:13.24tguidi have each t1 port of this 4 port card set as a span
15:13.40tguidlog is showing 0 based index instead of 1 of zaptel.conf
15:13.42spackletguid, do you have span lines set for the other channels?  Is the PRI plugged in to port 1?
15:13.59tguidpri might be plugged into port 4 now, was port 1
15:14.10tguidi'll have to get someone with physical access to move
15:14.31spackletguid, wait
15:15.01tguid#port 1
15:15.01tguidspan=1,1,0,esf,b8zs
15:15.01tguidbchan=1-23
15:15.02tguiddchan=24
15:15.12tguidand similar for 2-4 with 4 being:
15:15.19tguid#port 4
15:15.19tguidspan=4,1,0,esf,b8zs
15:15.19tguidbchan=73-95
15:15.19tguiddchan=96
15:15.28olivier_be careful you can only take timing on 1 port
15:15.46olivier_if you take timing on the firts span :
15:16.03olivier_span=1,1,0,esbf,b8zs
15:16.06spackleWhichever port the pri is plugged into will say span=1,1,0,est,b8zs, others will say x,0,0,esf.....
15:16.19olivier_span=1,0,0,esbf,b8zs
15:16.24olivier_span=2,0,0,esbf,b8zs
15:16.30tguidah
15:16.34tzangerolivier_: heh esf,b8zs not esbf :-)
15:17.05tguidso should i have timing on any or none?
15:17.17spackletguid, timing is only from pri
15:17.30spackletguid, timing is only from pri - port with pri plugged in.
15:17.32*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
15:17.38tguidgotcha
15:19.03*** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net)
15:21.00*** part/#asterisk mmlj4 (n=jkelly@redfishnetworks.com)
15:21.10mutilatorbah
15:21.16mutilatorfscking groups don't work correctly
15:21.56tguidpri connected to port 4, unfortunately as i didn't know before if numbering was top to bottom or bottom to top
15:22.26tguidsame deal with span=4,1,0,esf,b8zs and others span=n,0,0,esf,b8zs
15:22.52tguidtrying to get hold of someone who can get phyiscal access to this box as i'm remote
15:22.57mutilatoror..
15:23.40spackletguid, just so we are on same page - where n= port number right?
15:24.20spackletguid, IIRC, ports go 1-4 from top of card to mobo edge.
15:24.50MrCh|ckenX-Rob are you there
15:25.09dave99Getting timeout trying to IAX to FWD too.  I think it's my firewall.
15:25.39mutilatorok they do but i still don't like it :P
15:26.04tguidspackle yes i just didn't want to list out the full numbers for span=1-3
15:26.17MrCh|ckenI need some help with my asterisk box
15:26.29tguidyou are correct wrt port numbering from http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html
15:26.34spackletguid, me either, just wanted to make sure ;-)
15:27.12tguidand i didn't see that until after i changed ports yesterday when i was there, remote now
15:27.18MrCh|ckenthe box was down and after the soport from someone when the box goes up . . the SIP extensions dindt work
15:27.18*** join/#asterisk gclark (n=gclark@66-193-73-162.gen.twtelecom.net)
15:27.53gclarkhello all - I have question in regards to vpb...
15:28.34MrCh|ckenhow can I get the SIP extensiosn working again
15:28.45spacklevpb or vpn?
15:28.49gclarkin the vpb.conf file there is an example of exten => _9XXX,1,Dial(vpb/g1/ww${EXTEN:${TRUNKMSD}}) .... Where is g1 getting defined.
15:29.04MrCh|ckennow . . al the time they are busy but nobodies is using it!
15:29.46olivier_<MrCh|cken> chan_sip.so loaded ?
15:30.12conveyWhat is the best Linux distro for *?
15:30.26spackleconvey, the one you are most familiar with.
15:30.29MrCh|ckenthanks olivier . . how I can reeview that??
15:30.48olivier_show modules
15:30.59conveyspackele: I am most familiar with SuSe but I can't get zaptel compiled on the darn thing...
15:31.02*** join/#asterisk Drizzt321 (n=drizzt@c-67-189-187-233.hsd1.ma.comcast.net)
15:31.02olivier_if not loaded --> load chan_sip.so
15:31.46MrCh|ckenolivier  . . . sorry Im totally newbie, coud you guide me thorough??
15:31.52*** join/#asterisk avizion (i=avizion@amiga500.org)
15:32.28*** join/#asterisk pablix (n=root@lacnic.net.uy)
15:32.30pablixhi all.
15:32.34*** join/#asterisk toddf (n=toddf@net-66-210-104-17.theshop.net)
15:32.34olivier_well. First : asterisk -r to have the cli command
15:32.55MrCh|ckenoliovier OKIS
15:33.19spackleconvey_ there seem to be a lot of references to similar issues on Google, have you gone down that route?
15:33.25pablixi have a question, when i call a extension in my pbx from asterisk i dont have a tone, not free tone, not busy tone. only silence until the people hangup
15:34.00conveyspackle: I think the problem is that my Kernel is different from my Kernel source and Suse does not offer any updated sources.
15:34.11Hmmhesaysi kill you i kill you good
15:34.30twisted[asteria]i kill you dead
15:34.34Qwellconvey: I'm certain there is a kernel source for your kernel, if you installed the suse kernel package
15:34.59olivier_<MrCh|cken> then : show modules
15:35.13tguidconvey or get the source from kernel.org for version of kernel you're running
15:35.16jake1932anyone with a 7960 using the password?  is there an parameter besides 'secret' to set?
15:35.26pablix?
15:35.32spackletguid, whats going on with yours?
15:35.42gclarkpablix: You don't get "ring back" or "dial tone"?
15:35.47conveyspackle: kernel source is installed. Uname kernel 2.6.11.4-21.9-smp. /usr/src Kernel 2.6.11-4-20a
15:35.48*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:35.50Qwelljake1932: besides username and such?
15:36.02*** join/#asterisk jthunder (n=jthunder@S0106000f669104ea.ed.shawcable.net)
15:36.10MrCh|ckenolivier . .could you give me the steps . .ill take note because now I have problem with the pbx pw?
15:36.15Qwellconvey: it must be exactly the same version
15:36.20tguidsame deal, same syslog messages.  probably would be better if pri plugged into port 1 and i haven't located anyone with physical access yet
15:36.27jake1932Qwell: yes - user name take but as soon as I set the 'secret' param to match the pw in the 7960 - reg fails
15:36.31MrCh|ckeni cant gain access to it!
15:36.32gclarkjake1932: all you need is the secret - there is some good examples on the wiki
15:36.35pablixgclark. Not. im only dial the number and wait if the people pickup and talk perfect.... but i dont have a tone tuuu tuuu tuuu tuuu before people pickup
15:37.09olivier_<MrCh|cken> are you sure asterisk is up ? ;-)
15:37.25tguidi've been running asterisk at home with a couple x100 cards fine and trying to set something up at work
15:37.35spackleconvey, are you just not running the newer kernel?
15:37.40MrCh|ckenyes the other extensions diferent to SIP are working but irregular
15:37.45gclarkpablix: so you are not getting ringback tone... you can try adding an "r" to your dial string
15:37.49conveyquell: SuSe is currently offering a difference kernel from the kernel source in it's installation app.
15:37.57tguideventually do sip-pstn and sip-telecon bridge with t1 interfaces
15:38.06pablixgclark in the extension.conf?
15:38.11tguidtrying sip within asterisk for now and eventually go with openser i think
15:38.12olivier_<MrCh|cken> so asterisk -r should work
15:38.13Qwellconvey: then get it from suse.com
15:38.21conveyquell: will try
15:38.39gclarkpablix: example - exten,1,Dial(SIP/2000,20,tr)
15:38.56MrCh|ckenyeah but i cant get now the correct root password
15:39.00tguidso while waiting for someone to move t1 for me here's my other problem :)
15:39.25*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
15:39.30tguidalthough i can see from cli that i can connect to sip on this server (firewall off even) and it is playing sounds in dialplan
15:39.39tguidi get no audio on my local sip client
15:39.45QwellNAT?
15:39.50tguidbut calling other sip phones i get sound
15:39.52olivier_<MrCh|cken> s call your admin and ask him to fix your asterisk
15:40.05tguidserver is not behind nat, public ip
15:40.06pablix<gclark> yep i have this.
15:40.13Qwelltguid: client, I mean
15:40.18tguidkphone
15:40.21QwellNAT?
15:40.42gclarkpablix: what kind of phones?  Are they SIP or Zap?
15:41.00tguidi had same results sitting behind nat with stun set and being on public ip (me client connecting to sip on my asterisk server)
15:41.03MrCh|ckenhe has left the company and cant get him now . . . we are caos totally
15:41.11*** join/#asterisk yartelecom (n=no-email@82.211.129.231)
15:41.25pablixSIP phones to digital phones
15:41.25QwellMrCh|cken: hire a consultant
15:42.11gclarkpablix: all on your network or are you call out the pstn?
15:42.20MrCh|ckeni am in colombia (southamerica) and no consultants in 100 miles around
15:42.36MrCh|ckenthanks qwell
15:42.44pablixgclark, i dont have outside connection is in my netowrk
15:42.55tguidMrCh|cken if you don't have root boot from live cd like knoppix and edit /etc/shadow with crypted password you know
15:43.10*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
15:43.10_omerbye guys....
15:43.17Qwelltguid: booting into single user mode is far easier
15:43.30tguidsome distros require root pwd for that
15:43.35Qwellshouldn't
15:44.00gclarkpablix: you mentioned digital phones, how are these connected to asterisk?  What are the results if you call sip to sip?
15:44.28MrCh|ckenno easy to me to do that
15:44.30*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
15:44.46*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
15:44.56pablixgclarsk SIP to SIP have tone! but when i try to connect to my pbx to digital phones. not TONE
15:45.43gclarkpablix: how are the digital phones connected to your * Box?  Channel Bank?  Another PBX?
15:45.45pauldypablix, had the same problem yesterday oddly enough removing the dtmf settings for the context of the sip provider made it work perfectly
15:46.44pablixgclark i connect the box to the pbx with a te110p (E1) and the digital phones are connect direct to the digital bank in the pbx
15:47.17pablixpauldy i dont have a provider its only me and i dont have a dtmf setting
15:47.21pauldybah my problem was dialing digital lines not digital dialing me
15:48.00gclarkpablix: got run... but it something to do with the connection from boxA to boxB...
15:48.52*** join/#asterisk synthetiq (n=roger@64.201.13.50)
15:48.59MrCh|ckenolivier . . im in and done and i am in the cli comand
15:49.35olivier_<MrCh|cken> oh you finaly find the root password ?
15:49.50olivier_So show modules and check the line chan_sip.so
15:50.06MrCh|ckenyep
15:51.01MrCh|ckeni put show modules but i cant stop the screen . . what must i do to review all the modules like you suggested
15:51.14Qwellscroll up
15:51.31Qwellshift-pgup and shift-pgdn
15:51.43MrCh|ckenbut the information is all the time mixing with the pnx transactions
15:52.35QwellI don't really see a problem there..
15:52.43QwellYou can still scroll up just the same
15:52.56olivier_ok forget it just type : load chan_sip.so
15:53.51Kattyanyone want to help me test my conference?
15:54.00QwellKatty: if its quick, I can
15:54.30Kattyk, i'll get you the info, sec.
15:54.33Kattyanyone else?
15:54.57MrCh|ckenoliver . . done . . .it answers everyone is bussy at this time
15:55.11Qwellolivier_: give up dude
15:55.12Qwell:p
15:55.52lehelKatty, can I?
15:56.10Kattylehel: sure
15:56.25lehelk
15:57.23MrCh|ckenolivier . . . only the SIP extensions are sounding busy all the time
15:57.38*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0ge.dialup.mindspring.com)
15:58.17olivier_<MrCh|cken>sorry to hard to help you. You should take time to read * docs or recruit an admin
15:58.47*** join/#asterisk trimi` (i=Pharrel@62.162.242.254)
15:58.49MrCh|ckenbut what can i do to reload the sip extensions??
15:59.11olivier_sip reload it's in all the docs
16:00.13*** join/#asterisk outtolunc (i=outtolun@adsl-66-218-53-170.dslextreme.com)
16:00.37trimi`<MrCh|cken> asterisk -r
16:01.12MrCh|ckentrimi thanks . .. ive done the asterisk r now
16:01.17MrCh|ckenwhat else?
16:01.47*** part/#asterisk jamc (i=p3dmildg@h250n1fls34o969.telia.com)
16:02.24*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
16:04.13MrCh|ckentrimi whe i use the sip extensions we have this message=
16:05.05*** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net)
16:05.13MrCh|ckenwarning 1060: chn_zap.c:1521zt_set look:zt look fails:device or resource are busy
16:06.15twisted[asteria]Katty, need any more help?
16:06.20*** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
16:06.38MrCh|ckenqwell can you review my last messege??
16:07.28MrCh|ckenwarning 1060: chn_zap.c:1521zt_set look:zt look fails:device or resource are busy
16:08.14Kattytwisted[asteria]: yeah, call the conference.
16:09.22*** join/#asterisk santiago (n=santiago@63.245.86.203)
16:09.28Kattytwisted[asteria]: err, my conference
16:09.57*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
16:10.18twisted[asteria]Katty, no tengo info
16:10.19MrCh|ckensantiago coul we speak in spanish i need some help??
16:10.26Kattyk, moment
16:10.28twisted[asteria]kk
16:10.33Rowterwhen dialing a POT I could get answered status, busy etc? or just on PRI?
16:10.58MrCh|ckenonly on a PRI i think
16:11.11RowterMrCh|cken, ohh
16:14.12twisted[asteria]Rowter, dont' dial POT, smoke it!
16:14.51cpatryhummm
16:14.51cpatryhttp://www.digitalvoice.bell.ca/DigitalVoiceLite/AdditionalFeatures/MeetMeCalling/index
16:14.59Rowterhaha twisted[asteria]
16:15.30MrCh|ckensantiago are you available??
16:15.55MrCh|ckenPOTs days have gone!!
16:16.12Rowtertwisted[asteria], exten => start,2,Goto(s-${DIALSTATUS},1) this will not work with TDM POT's?
16:16.12jake1932Qwell: registration with my 7960 only works with no 'secret' in the sip.conf - are you using 7.5?
16:16.34*** join/#asterisk silug (n=steve@206.80.72.34)
16:16.40Rowterjust with PRI?
16:18.25MrCh|ckenRowter . . .nop . . .you can configuarate your one line PBx with POT . . just go to config examples
16:19.44MrCh|ckenwhat's "busy cause 17"??
16:20.02Rowterhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Variable+HANGUPCAUSE
16:20.22Rowter#define AST_CAUSE_USER_BUSY 17
16:20.46RowterMrCh|cken, you got it calling a sip or with ZAP?
16:20.52*** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
16:21.10MrCh|ckenRowter . . .calling a SIP
16:21.11obsidian-studiosanyone using blacklists here?
16:21.35MrCh|ckenrowter . . .callin a SIP from zap
16:22.16twisted[asteria]Rowter, you can TRY callingprogress in zapata.conf, but it might make weirdness
16:22.50Rowtertwisted[asteria], yea I heard that! let me try tho to one channel
16:23.13MrCh|ckencallingprogress wasn't deprecated???
16:23.29obsidian-studiosare Cisco 7960's supposed to display the sip # instead of say the display name on the phone?
16:23.29santiagohi MrCh|cken
16:23.40MrCh|ckenhi santiago
16:24.35*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
16:24.44jake1932obsidian-studios: it's a 2 line display - should be able to display both
16:24.54MrCh|ckensantiago could we speak in spanish?
16:25.15obsidian-studiosjake1932: well it's got the sip #'s sitting next to the speed dial buttons, and no display name anywhere I can see?
16:25.17santiagoMrCh|cken, i'll try ;=
16:25.21santiago;)
16:25.30RowterColombian >)
16:25.47MrCh|ckensantiago you aren't latino??
16:25.52zzzirkI've got a n00b question: can I accept an h.323 connection and direct it to a sip connection via asterisk?
16:26.04santiagoMrCh|cken, yeap
16:26.17MrCh|ckenRowter . . . colombian is another kind of spanish . . .only regional
16:26.19jake1932obsidian-studios: oh - for your number/name - hmm - haven't tried that
16:26.44RowterMrCh|cken, yep
16:26.47obsidian-studiosjake1932: yes, for the users it would be nice to have a name or something other than a # next to the line, but it's not a big deal
16:26.50santiagoMrCh|cken, lo siento, había leído mal el mensaje
16:27.08MrCh|ckensantiago tengo un gran problema con las extensiones SIP de mi asterisk box
16:27.24MrCh|ckensantiago . . me podrias aytudar a encontrar que les pasa??
16:27.36santiagoMrCh|cken, claro
16:27.50santiagoMrCh|cken, ¿qué problema es?
16:28.11MrCh|ckensantiago . . el viernes pasado el box se enloquecio
16:28.36MrCh|ckeny X-rob me ayudo a recompilar un nuevo asterisk y dejarlo funcionando
16:29.02MrCh|ckenpero las extensiones SIP no subieron y todo el tiempo suenan ocupadas
16:29.26santiagoMrCh|cken, ¿no tienen tono de marcado?
16:29.52MrCh|ckense genera un mensaje: warning 1060
16:30.13MrCh|ckensantiago . . .suena todo el tiempo tono de ocupado
16:31.08*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
16:31.21MrCh|ckensantiago . . .hice un load chan_sip y dice que el modulo existe
16:31.54santiagoMrCh|cken, ¿qué clientes sip estás usando?
16:32.39MrCh|ckensantiago . . .dos extensiones conectasa a un ATA
16:33.08*** join/#asterisk cio (n=bill@adsl-068-209-198-242.sip.bhm.bellsouth.net)
16:33.17*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
16:33.23santiagoMrCh|cken, ¿sólo eso?
16:34.08obsidian-studiosI have a special extensions emails CID info as one of it's functions, the extensions are a range like _[100000-999999], but when I use the ${EXTEN} var in the email all I get is 1? when I dialed 100000?
16:34.09MrCh|ckentengo 2 channel banks de 24 extensiones c/u que funcionan +o- bien
16:34.11cioHi all.  Having a problem with a SIP NAT client.  Seems to be keep-alive related.  I'm using nat=yes, qualify=250.  It's not a lag issue.  It stays qualified for a few minutes then becomes unreachable.  Is there a way to control the qualify=xxx cycle in asterisk? i.e., force asterisk to check every XX seconds?
16:34.29MrCh|ckene inicie con 2 SIPs conectadas al ATA
16:35.11santiagoMrCh|cken, ¿pero el banco de canales no tiene problema ahora o sí? (lo digo porque sólo me estabas hablando de sip)
16:35.53MrCh|ckensantiago . . hay problemas menores con los banks . . pero las SIP suenen todo el tiempo ocupadas
16:36.16MrCh|ckensantiag . . es decir . . son las unicas que no funcionan . . todo el tiempo ocupadas!!
16:36.18santiagoMrCh|cken, puede ser por el ATA, yo he tenido muchiiiiisimo problemas con unas sipura
16:36.26Drizzt321I'm setting up asterisk to connect an internal softphone out to a SIP PSTN provider, however I cannot get incoming calls to ring, and when I try to call out I get 404 Not Found and a busy signal. I've followed all the online guides I can find, but I haven't gotten it to work :(   any help?
16:36.33*** part/#asterisk dave99 (i=David@207.201.200.136)
16:36.35santiagoMrCh|cken, ese mismo problema me ha pasado
16:36.44santiagoMrCh|cken, ¿las extensiones te aparecen registradas?
16:36.55MrCh|ckensantiago . . y en tal caso comopuedo hacer que funcionen otra vez??
16:37.11*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:37.17MrCh|ckensantiago . . .como reviso si aparecen registradas??
16:37.25santiagoMrCh|cken, sip show peers
16:37.58*** join/#asterisk _omer (i=o@203.215.180.250)
16:38.01_omerhi
16:38.04santiagoMrCh|cken, ¿qué ata es?
16:38.27_omerhow to run php in asterisk ?
16:38.31MrCh|ckensantiago . . .acabo de revisar . . aparecen registradas
16:38.53MrCh|ckensantiago . . creo que es un 186 . . esta lejos de mi en este momento
16:39.07cioAnyone have any suggestions?
16:39.27santiagoMrCh|cken, bueno, así mismo aparecían las mías, inclusive, en el CLI se podía ver que las llamadas entraban y timbraban, pero los teléfonos sonaban ocupados
16:39.52MrCh|ckensantiago . . y que hiciste?? para salir de esa
16:40.14santiagoMrCh|cken, prueba apagando y volviendo a iniciar el ATA
16:41.17jake1932obsidian-studios: the only way i see to change what is on the display is 'line1_name' (for line 1) however, that is also your username
16:41.19_omeragi-test.agi source code is different than it's response to AGI(agi-test.agi) ....???
16:41.29MrCh|ckensantiago lo intente ayer pero nada cambio
16:42.02santiagoMrCh|cken, entonces, desconecta el ATA y prueba con un softphone, utilizando una de los canales del ATA
16:42.02obsidian-studiosjake1932:  yeah, and for now I would just assume leave them as #'s instead of names, although it would be nice for names on the phone. What is the point of the display name then?
16:42.29jake1932display name seems to be for outbound calls
16:42.31santiagoMrCh|cken, si el softphone funciona, es problema del ATA; si no, es por asterisk
16:43.19MrCh|ckensantiago . . .si fuese asterisk . . .que se debe hacer??
16:43.38*** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net)
16:44.01santiagoMrCh|cken, podría ser por problema en la configuración, pero lo dudo mucho
16:44.21santiagoMrCh|cken, estoy casi seguro que debe ser el ATA
16:44.44MrCh|ckensantiago . . que puedo hacer??
16:44.53santiagoMrCh|cken, lo que ya te dije
16:45.02MrCh|ckensantiago  . . .para verificar el ATA
16:45.33santiagoMrCh|cken, prueba con un softphone
16:45.54MrCh|ckenhummmmmmmmmmmmm
16:46.16obsidian-studiosjake1932: ah, ty
16:47.33Drizzt321can anyone give me a help with my dialplan? I can't seem to get incoming calls or make outoing calls via SIP.
16:47.59*** join/#asterisk Assid (n=assid@203.115.64.59)
16:48.40MrCh|ckensantiago  . . .gracias por tu tiempo y seguire intentando resolver el asunto!!
16:48.41pauldyhow do you know it is your dialplan and not the registration
16:49.07santiagoMrCh|cken, pues pruebe lo que le digo, es lo más fácil que puede hacer
16:49.16*** join/#asterisk zx225 (n=me@65.183.42.3)
16:49.25Rav1974MrCh|cken: yo no habla espaniol.  Habla engles?
16:49.36Drizzt321pauldy: I'm pretty sure, I can get registered to my SIP-PSTN provider, and my softphone is registered alright to my server...although I can't be 100% sure its not the registration
16:49.38santiagoMrCh|cken, si le queda difícil desconectar el ATA, cree otra extensión sip y llame a alguna de las del banco de canales
16:49.38Rav1974:)
16:50.06MrCh|ckensantiago . . yeppp . . . .aqui es casi medio dias  . . esperare un poco y desconetare el ata y lo reconectare
16:50.09MrCh|ckensantiago
16:50.19MrCh|ckenme puedes dar tu correo??
16:50.33pauldyTengo huevos y el tocino en mis pantalones usted tiene gusto de ensamblarme?
16:50.47MrCh|ckenRav1974 . . .wich idea??
16:50.56santiagoMrCh|cken, ¿sos de colombia?
16:51.17MrCh|ckenyepppppppppppp.... y vos??? Argentina???
16:51.24santiagoMrCh|cken, colombia
16:51.30Drizzt321pauldy: would you like to see my sip.conf and extensions.conf?
16:51.39MrCh|ckeny en que ciudad estas??
16:51.49MrCh|ckensantiago . . yo en bogota
16:52.06santiagoMrCh|cken, en popayán
16:52.15*** join/#asterisk clint_ (n=clint@snap.helixsystems.com)
16:52.15*** join/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net)
16:52.36blitzrageanyone know if 1.0.9 support nat=route ?
16:52.41MrCh|ckensantiago . . .miercoles . . .bien lejos. . . .podrias darme tu telefono??
16:52.55*** join/#asterisk NORANDOMS (n=watchy@adsl-69-152-41-249.dsl.ltrkar.swbell.net)
16:53.07NORANDOMSdoes a 7960 not forget its .cnf on reboot?
16:53.07MrCh|ckensantiago  . . para contarte en vivo y en directo y ver si me puedes hechar una mano???
16:53.24blitzrageNORANDOMS: don't think so - it "caches" it I'm pretty sure
16:53.43NORANDOMSblit: yea thats what its lookin like. know of a quick way to reset it?
16:54.08santiagoMrCh|cken, el problema es que voy a salir de la ciudad, haga lo que le digo, y verá que es el ATA
16:54.32MrCh|ckensantiago . . .okis . . .gracias!!
16:54.33blitzrageNORANDOMS: ummm... Settings > Network Configuration > 28 (Erase Configuration)
16:54.42NORANDOMSwill that erase the ips i've put in etc?
16:54.53blitzrageNORANDOMS: it'll reset the configs - so yah
16:54.55santiagoMrCh|cken, no hay problema
16:55.01NORANDOMSaint that a bitch
16:55.09blitzrageNORANDOMS: well what would you expect a reset to do? :)
16:55.12NORANDOMSbut i guess once the phones setup you dont need to mess with that
16:56.04blitzrageI wonder what that would entail...
16:56.29filebeer.
16:56.33blitzragefile: do you know if 1.0.9 supports nat=route?
16:56.34outtolunc16oz hammer <G>
16:56.39blitzragefile: lol
16:57.06outtoluncfile, that's a 'soft' reset
16:57.08fileblitzrage: I don't do stable :P
16:57.15blitzrage"Co-ffee"   "Bee-er"   "C O"   "B E"
16:57.28blitzragefile: me either :)
16:57.36filehard baby hard!
16:57.42blitzragedamn you drumkilla - where are you?! :)
16:58.15blitzrageI suppose I could always just go and test it...
16:58.20filegood idea
16:58.24fileor read the source
16:58.24blitzragebut thats not the "asterisk" way :D
16:58.35*** part/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net)
16:58.38blitzragethe source looks like swahili to me
16:58.44*** join/#asterisk xaoc|work (n=me@212.113.51.69.utel.net.ua)
16:58.50blitzragenow if it was programmed in PHP.... :)
16:59.05filethen it would be like PHP to you :P
16:59.10wunderkinyeah that'd be 3r33t
16:59.48xaoc|workguys, is there a possibility that asterisk can forward traffic to skype ?
17:01.23*** join/#asterisk wolfson (n=hehe@65.174.122.198)
17:01.33JerJerif you write chan_skype, sure
17:02.01halogen8anyone in here get a free IPKALL DID working with their asterisk?
17:03.21AssidJerJer: pm
17:03.22Nivexhalogen8: I cheated and routed mine through Free World Dialup and had them send the call to me over IAX :)
17:03.43AssidJerJer: sent you an email.. need did
17:03.52xaoc|workJerJer: so no modules for that exists yet ?
17:04.22Assidxaoc|work: if you do write it.. lemme know i would like it too
17:04.25halogen8Nivex: i have been thinking about doing that......but like you, figured it would be cheating.....did you do it that way because you wouldn't figure out how to get it working directly?
17:05.28halogen8IAX seems to work more seemlessly than SIP.....is that true?  espcially when dealing with a asterisk behind a NAT.
17:05.42Assidyep
17:05.48Assidbetter nat support
17:05.53Nivexhalogen8: I did it that way mostly because I already had FWD set up and that's what was in the IPKall dialog box :)
17:05.54Assidsomething about the stream
17:06.04halogen8Assid: do most providers support IAX now?
17:06.16Assidwell.. most providers who run asterisk do
17:06.36halogen8Assid: is there a list of asterisk providers available?
17:06.47Assidnufone does.. but im still trying to ask JerJer for a DID
17:06.49Assidhrmm
17:06.56Assidi think voip-info had a list
17:06.58Assidgoogle for it
17:08.22hardwiremoof
17:09.20halogen8Assid: do you know of another provider giving away free DID's in the USA?
17:09.27halogen8Assid: similar to IPkall?
17:09.40Assidfree did's ?
17:09.43Assidnope:(
17:09.54Assidi gotta tryo ipkall
17:10.21*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
17:11.04halogen8Assid: IPkall gives a free DID, but I can't get it set up for the life of me......its starting to piss me off......but I think its because its SIP and not IAX
17:11.18*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
17:11.22halogen8Assid: although I'm new to asterisk, so it may be a configuration thing on my end
17:11.37halogen8btw....the provider list can be found here:  http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers+B2B
17:11.50*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
17:13.42hardwireI need to jump on the bandwagon
17:13.47hardwirewhich means more sugar.. caffeine
17:13.51hardwireand some solitude
17:17.07*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
17:19.15blitzrageanyone know some good manufacturers of T3 multiplexers?
17:20.23hardwireblitzrage: I have no freaking clue
17:20.31blitzragehaha - me either
17:20.33hardwireMediatrix has a modular digital gateway now.
17:20.39hardwireit may be able to do what you want.
17:20.40*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
17:20.42hardwireerr
17:20.45hardwirenot multiplexer
17:20.54hardwireI suppose you could use VoIP to multiplex it :)
17:20.55blitzrageyah - I don't think Mediatrix does what I want :)
17:21.10blitzrageI just need something to break out a T3 into multiple T1s :)
17:23.03hardwireso you get a t3 gateway.
17:23.11hardwireand map it to a few t1 gateways
17:23.13hardwire:)
17:23.35blitzrage:)
17:24.26blitzrageanyone want to backport nat=route to 1.0.9? (will pay)
17:25.23hardwirescared of 1.2.0.
17:25.26hardwireor too integrated to it?
17:25.32hardwireI have to upport :)
17:25.45hardwiresip notify is going to save my ass :)
17:25.45blitzragecustomer is using 1.0.9 until 1.2.1 or 1.2.2 :)
17:25.52hardwireah
17:25.59hardwire1.2.0 will become stable at some point right?
17:26.02blitzrageI personally haven't used "stable" since it was "head" :)
17:26.07hardwireor will it officially be titled stable at 1.2.2?
17:26.28*** join/#asterisk darkskiez (n=darkskie@host86-132-168-193.range86-132.btcentralplus.com)
17:26.38blitzrageit'll be *titled* stable at 1.2.0 - who knows how stable it'll actually be, but how much software can YOU think of that was actually stable on a .0 release?
17:26.44blitzrageI assume Asterisk will be no different
17:27.11hardwireasterisk is unique
17:27.15hardwireit has good coders :)
17:27.23outtoluncwhy not just name it STABLE(ish)
17:27.33hardwiresemi-stable
17:27.35hardwirevs beta
17:27.39darkskiezthey refuse my patches - it must be good.
17:27.43hardwirehahaha
17:27.59*** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net)
17:28.09obsidian-studioswith sip debug, and cli set to verbosity 9, I should see sip calls no? I am testing a fax from my Cisco UBR924 to * via SIP. I am hearing the Sorry the # you have dialed recording from the telco, but I can't find out what # was passed to * from the URB?
17:28.13blitzragelol
17:31.57*** join/#asterisk IOscanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net)
17:34.45synthetiqi have an idsn pri for e911
17:35.28synthetiqspan=2,0,0,esf,b8zs  \n   bchan=25-47   \n         dchan=48
17:35.35synthetiqis that correct?
17:35.49synthetiqor do i need anyhtign else
17:37.54*** join/#asterisk jontow (i=jontow@ws.woflsys.net)
17:37.57hardwireyou have 24 channels for e911?
17:38.03hardwireyou guys have issues.
17:38.17obsidian-studiosok, got past a cabling issue, and successfully sent a fax via SIP to *, and then out via  a zap channel
17:38.50obsidian-studiosnow I have to try testing faxing using the cisco fxs ports via a wan, instead of a lan
17:38.51hardwireobsidian-studios: hey
17:38.51synthetiqhardiwire: i have 600 users on this system
17:38.55Hmmhesaysyay asterisk runni at 99% cpu
17:39.04hardwiresynthetiq: does E911 spec the ratio?
17:39.29synthetiqwhat do u mean by spec the ratio
17:39.33hardwireobsidian-studios: did the target zap channel have to ring a fax line?
17:39.35spackletguid, are you still around?
17:39.42hardwireI am finding we get faxes on alot of our inbound lines
17:39.50hardwireand I wanted to forward it when tone detected to our fax line
17:39.55spackle~seen tguid
17:39.58jbottguid is currently on #asterisk (2h 48m 5s).  Has said a total of 54 messages.  Is idling for 1h 56m 28s
17:39.58obsidian-studioshardwire: yes it called another remote fax machine
17:40.10hardwirehowever.. that means I have to ring the fax machine after I already answreed the fax
17:40.13hardwireso will that even work?
17:40.28synthetiqno point ringign the fax if it had been answered
17:40.31bzbwhi, anyone know how to fix this error:
17:40.32bzbwSep 14 10:14:02 WARNING[3833]: chan_zap.c:890 zt_open: Unable to specify channel 3: No such device or address
17:40.32bzbwSep 14 10:14:02 ERROR[3833]: chan_zap.c:6650 mkintf: Unable to open channel 3: No such device or address
17:40.32bzbwhere = 0, tmp->channel = 3, channel = 3
17:40.32bzbwSep 14 10:14:02 ERROR[3833]: chan_zap.c:10030 setup_zap: Unable to register channel '1-3'
17:40.43synthetiqfax,1,Goto(${EXTEN},1)
17:40.46hardwiresynthetiq: courious if fax spec allowed for that
17:40.56obsidian-studioshardwire: at another client I have no problem with inbound faxes to * via zap channel to sip to Cisco 827-4v, which one extension for the fax just dials that sip channel. Works great for inbound and outbound faxes, and outbound credit card terminals
17:41.04hardwireand detected a ringtone even after answered
17:41.05synthetiqhave that in your incoming context
17:41.23obsidian-studioshardwire: now I need to see if faxing via a wan is possible, remote Cisco router using sip to * via a wan instead of a lan
17:41.26synthetiqdamnit im having no 911 calls going out
17:41.28hardwiresynthetiq: yeh.. but doesn't that answer a fax?
17:41.41synthetiqno it doesnt
17:41.51hardwiresorry.. maybe I need to know more about how it detects the fax.. as well as what the goto exten needs to be set at for auto-answer
17:41.55synthetiqchan zap answers it
17:42.28*** join/#asterisk ManxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net)
17:42.32synthetiqi have fax working via iax =]
17:42.34bzbwI can see 3 channels by using "zap show status", but the third channel failed to make * start
17:43.04obsidian-studioshardwire: I would assume from what I have done with faxing just dialing the fax line is all that is needed beyond the detection aspect. In my case clients have a dedicated fax line, so anything ringing that line is a fax :)
17:43.27obsidian-studioshardwire: I would be curious on the detecting aspect, because I know someone keeps trying to fax my voice line ;)
17:43.32hardwireyeh
17:43.35hardwireI need that to stop
17:43.37hardwire:)
17:44.07obsidian-studioshardwire: I am about to fax them back, or get something to fax them back continually
17:44.16obsidian-studioshardwire: I hate faxing
17:44.23hardwireI hate it too
17:44.25hardwirebut fax happens.
17:44.37obsidian-studioshardwire: yes for sigs, there is not much alternative at the moment
17:45.11bzbwLet me try again, anyone knows how to fix following error?
17:45.11bzbwSep 14 10:14:02 WARNING[3833]: chan_zap.c:890 zt_open: Unable to specify channel 3: No such device or address
17:45.11bzbwSep 14 10:14:02 ERROR[3833]: chan_zap.c:6650 mkintf: Unable to open channel 3: No such device or address
17:45.37bzbwzap show status does show 3 channels
17:45.41obsidian-studiosbzbw: does the /dev/zap/* stuff exist?
17:45.42hardwireoh poop
17:45.48hardwirethere are a billion people using the phones
17:45.53hardwireI can't restart it tillt hey are all off
17:45.54hardwireheh
17:46.25obsidian-studiosjust kick them off
17:46.28bzbwobsidian-studios: yes
17:46.44obsidian-studiosbzbw: you have permissions and etc for the devices?
17:46.58shmaltzwhat does this mean:
17:46.59shmaltzWARNING[5965]: chan_zap.c:2131 pri_find_dchan: No D-channels available! Using Primary on channel anyway 48!
17:47.02obsidian-studiosbzbw: do the other channels work?
17:47.05bzbwobsidian-studios: yes
17:47.26obsidian-studiosbzbw: hmm
17:47.39jovui have an fxo hooked up to a uk landline and i`m making calls from a cisco 7960g, but for some reason after it`s connected to a number the remote end doesnt recognise any dtmf tones (its one of those press 1 for x, press 2 for y menu systems) any ideas how i can get this to work?
17:47.43bzbwobsidian-studios: other 2 channels working
17:47.54hardwirenow testing fax,1,Dial(Local...
17:47.56obsidian-studiosbzbw: weird
17:48.07hardwireoh man
17:48.09hardwireHendrix
17:48.14hardwireI need to be alone nwo
17:48.40spacklewhat song?
17:49.11obsidian-studiosdamn, irc kicks those in a pm for flooding?
17:49.28obsidian-studiosbzbw: was pming me info, and got kicked :(
17:49.40Hmmhesayswiki wild wild, wiki wiki wild wild
17:52.06hardwireI have here in my hand
17:52.07hardwireTo shane
17:52.09hardwirefrom Lenora
17:52.11hardwirethis is just a test
17:52.12hardwirea fax
17:52.22hardwiredetected on our main company inbound number
17:52.24hardwire:)
17:52.29hardwireso happy
17:52.36hardwireno more ear squelching terror for the operators
17:52.41*** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net)
17:55.12*** join/#asterisk ClayReiche123 (i=fwuser@mail.accxx.com)
17:55.45ClayReiche123Can I authorize sip traffic based on originating IP address?
17:56.12hardwireyup
17:56.17hardwireinsecure=very
17:56.24ClayReiche123I just want to "proxy" sip traffic from 1 asterisk out to another provider
17:56.27ClayReiche123thank you!
17:59.32ManxPowerHas anyone here experienced randomly dropped calls when sending them via IAX2?
18:00.17ClayReiche123hardwire: still getting a 403 Forbidden
18:00.18ManxPowerIf I was any easier to replace I'd have been fired because of the dropped calls.
18:00.23ManxPowerAs it is, it's just a matter of time.
18:00.32ClayReiche123hehee
18:00.35pskis there anyone ho can help me with a strange problem witch asterisk@home? i can't hear voicemail
18:00.45Hmmhesaysthis cvs-head from a few days ago is performing tyte
18:00.57pskbut by console i can see calls answered
18:01.12JerJerpsk: don't use asterisk@home then
18:01.32ManxPowerThis is a problem with BOTH 1.0.9 and CVS-HEAD
18:01.57pskwhy? buggy?
18:03.07ClayReiche123Hardwire: any ideas?
18:03.42*** join/#asterisk barc (n=barc@216.232.66.249)
18:04.07zlocsomeone looking for zotz earlier?
18:04.16ClayReiche123Hardwire: all I have in sip.conf is [sip_proxy]\n type=friend\n context=wholesale\n host=xxx.xxx.xxx.xxx\n insecure=very\n
18:04.22hardwiremoo
18:04.37hardwiretry a different type=
18:04.49mutilatorw00t
18:04.54mutilatormy new dial plan is done!
18:05.02hardwiremutilator: congrats
18:05.22mutilatorohhhh so much prettier than the old one...
18:05.41mutilatorthat was made and hacked at for more than a year now
18:05.42*** join/#asterisk azzie_ (n=az@azzie.net)
18:06.00azzie_good afternoon
18:06.16azzie_i'm having a weird problem... expressions don't seem to work
18:06.27blitzrageA@H - not impressed with the GUI
18:06.32azzie_a like like: exten => s,6,NoOp($[1 + 1])
18:06.34blitzragewas restrictive and inconsistant
18:06.49azzie_debugs to     -- Executing NoOp("SIP/fwd.pulver.com-0010e660", "0") in new stack
18:07.02Hmmhesaysi kind of like amp
18:07.07azzie_shoundn't 1+1 be 2 ?
18:07.30hardwiremutilator: AEL now?
18:07.46*** join/#asterisk mjr^^ (n=muhajir_@210.212.195.134)
18:08.31ManxPowerazzie_, 1.0.x or CVS-HEAD/1.2?
18:08.41*** join/#asterisk mcreedjr (n=mcreedjr@72.240.172.15)
18:08.53*** topic/#asterisk by drumkilla_lab -> Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now!
18:09.10mcreedjrHey, anyone successfully using Asterisk to iconnecthere who is willing to share configs?
18:09.43azzie_ManxPower, 1.2
18:10.07ManxPowerazzie_, And you read UPGRADE.txt (specifically the part about the expression handler changeing?
18:10.25mjr^^mcreedjr:sip std port is 5060 right is it te std for both reception and transmition
18:10.32mutilatorno, normal extensions
18:10.41mutilatori don't like ael to be honest
18:10.54mcreedjrmjr^^: ?
18:11.10*** join/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net)
18:11.39mjr^^mcreedjr: 5060 port is the standard port right?
18:11.41bzbwwhat is fxols(loopstart) versus fxoks(Koolstart)? In my zaptel.conf, it is using fxsks for x100p, is it right? I'm in us
18:12.10azzie_ManxPower, reading now -- can't find anything on expressions... could you quote please?
18:12.12mcreedjrmjr^^: Is this in reference to my question? I couldn't answer taht question anyways without more context.
18:13.07mjr^^mcreedjr:i used tis port to for forwarding calls to n2p now i am using vsnl .. so any chng to be made in port numbers
18:14.37mcreedjrmjr^^: I am totally confused, I think you've got me mixed up with someone else who was helping you. Sorry.
18:14.48mcreedjrHey, anyone successfully using Asterisk to iconnecthere who is willing to share configs?
18:15.07mjr^^sorry to confuse u but i was askin some doubts with u if u dont mind
18:16.55mcreedjrmjr^^: I'm an Asterisk n00b, I'm probably not the one to answer your questions. Sorry again.
18:18.16ManxPower* The dialplan expression parser (which handles $[ ... ] constructs)
18:18.17ManxPower<PROTECTED>
18:18.45ManxPowerless /home/software/asterisk/asterisk/UPGRADE.txt
18:18.50mjr^^mcreedjr: hey u want conf.. for iconnecthere  right?
18:19.20mcreedjrmjr^^: Yeah, if someone has a working one. I can't get the one from the wiki to work right
18:19.27mjr^^mcreedjr: http://lists.digium.com/pipermail/asterisk-users/2003-February/007289.html ,try tis link
18:19.31azzie_ManxPower, oh i've read this. so it says $[1 + 1] should work :)
18:19.37azzie_but it doesn't
18:20.18pablixhi again
18:20.35*** join/#asterisk jsaunders (i=jsaunder@d207-81-167-183.bchsia.telus.net)
18:21.01pablixi have this problem i have a box and a pbx connected via e1... my digital extension in the pbx is 109 when i try to call a asterisk extension give me this error//// Extension '' in context 'from-pstn' from '109' does not exist.  Rejecting call on channel 0/1, span 1
18:21.54*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net)
18:23.05*** join/#asterisk voipguy (n=voipguy@196.200.25.253)
18:23.14syleyou should really pastebin your config pablix, how is anyone going to know how to help you
18:24.30mjr^^mcreedjr: whts ur host?
18:24.30*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
18:26.05pablixsyle waht file do you need?
18:26.21obsidian-studiosbzbw: yes fxsks is for x100p cards, and you want the  wcfxo kernel module
18:26.57sylei would include your logs, your extensions.conf and your sip or iax.conf , whichever your using
18:27.32PupenoLWhere is one supoused to get apps_[rt]xfax from ?
18:28.19sylegoogle
18:28.27mjr^^can we edit the source port and the destination port in asterisk
18:28.54mjr^^can anyone answer this plzzzz
18:29.14sylefor what
18:30.12mcreedjrI'm using iconnecthere and when I try and call out, I get the following: Got SIP response 481 "Call Leg/Transaction Does Not Exist". Any ideas?
18:31.10xhelioxMeh.
18:31.18xhelioxI've ordered 3 TDM400P's..
18:31.20*** join/#asterisk tubes41 (n=tubes41@203-59-179-19.dyn.iinet.net.au)
18:31.24xhelioxI had to return 2.
18:31.30xhelioxand now I'm returning a return
18:31.37tubes41hello
18:31.43xhelioxIs this something I can expect all the time?
18:31.49*** join/#asterisk qWer^ (n=muhajir_@61.246.231.50)
18:32.01qWer^can anyone help me
18:32.20tubes41i was gonna ask the same question...
18:32.23sylepretty much xheliox
18:32.24JerJeronly if you help yourself first
18:32.34qWer^how
18:32.41qWer^?
18:32.51JerJerperhaps, asking a specific question
18:33.19nextimeany pyastre user here?
18:33.20qWer^can i edit the source port and the destination port in asterisk
18:33.30sylefor what?
18:33.33JerJersee rtp.conf.sample
18:33.39qWer^k
18:33.41tubes41I'm getting compile errors...
18:33.43qWer^thanks
18:34.09tubes41can anyone help? I have specifics
18:34.31tubes41gcc -shared -Xlinker -x -o app_curl.so app_curl.o -L/usr/lib -lcurl -L/usr/kerberos/lib -lssl -lcrypto -lgssapi_krb5 -lkrb5 -lcom_err -lk5crypto -lresolv -ldl -lz -lgssapi_krb5 -lkrb5 -lk5crypto -lcom_err -lresolv -L/usr/kerberos/lib -lidn -lssl -lcrypto -lssl -lcrypto -lgssapi_krb5 -lkrb5 -lcom_err -lk5crypto -lresolv -ldl -lz -lz
18:34.31tubes41/usr/bin/ld: cannot find -lidn
18:34.31tubes41collect2: ld returned 1 exit status
18:34.32tubes41make[1]: *** [app_curl.so] Error 1
18:34.32tubes41make[1]: Leaving directory `/usr/src/asterisk/apps'
18:34.34tubes41make: *** [subdirs] Error 1
18:34.37tubes41Hopefully someone here can tell me what i've screwed up...
18:34.49syleyour missing a library
18:35.04tubes41any idea which one?
18:35.06cpatry~pastebin
18:35.07jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
18:35.15tubes41soz
18:35.17*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
18:35.18nextimetubes41 : if you are using debian, apt-get install  libidn11-dev
18:35.45tubes41using fedora core 3
18:35.48halogen8is it possible to have the same extension logged into an * at the same time?
18:36.06JerJerhalogen8:  eh?
18:36.10halogen8so i can have an ata logged in at home and work, and they both ring when someone dials it
18:36.10nextimetubes41 : search a package named something like libidn*-dev
18:36.43halogen8JerJer: have extension 200 logged in twice, at the same time
18:36.50halogen8JerJer: is that possible?
18:37.18JerJerin different contexts, sure
18:37.28halogen8JerJer: i'm trying to setup a few ata to simulate regular phone wiring inside my house.......you know, if you have multiple phones plugged in at home, they all ring at once
18:37.32JerJerdefine 'logged in'
18:37.46JerJerasterisk doesn't deal with multiple SIP locations
18:37.51Ariel_halogen8, no they don't work that way.
18:37.52halogen8JerJer: well.......in the ata, you have to tell it your extension and password
18:38.26Hmmhesayshow much is a t1 fxs channel bank running these days
18:38.35halogen8so how would i make multiple ata's in my house ring when someone calls in? would i have to use groups
18:38.40Ariel_halogen8, each one needs it's own login what you do is you can dial more then one device with sip/100&sip/200&sip/300
18:38.48*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
18:38.53Hmmhesays600 or so?
18:39.16Ariel_Hmmhesays, used about 500 on ebay Adtran 750/850
18:39.25Hmmhesaysnew
18:39.33Ariel_new lots of money
18:39.39syle1k-1500
18:39.46halogen8Ariel_: if someone calls in on my DID, and wants to reach me in my house.......i want all the phones in my house to ring, just as it does with a regular landline.....how is that done?
18:40.02halogen8Ariel_: they call in and get the auto attendant, and his ext 200
18:40.05Ariel_halogen8, read my reply
18:40.46Ariel_halogen8, are you using amp or asterisk@home? or doing this via your own setup and rules?
18:41.09halogen8Ariel_: i'm using asterisk@home which uses AMP
18:41.25Ariel_halogen8, ok it's simple create a ring group
18:41.35halogen8yeah......thats what i'm seeing
18:41.44Ariel_all the devices in that ring group will ring at the same time.
18:41.51*** join/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu)
18:41.54halogen8so when the caller presses 1 at the AA, it will ring all in the ring group 1
18:42.05Hmmhesays24 port fxs gateways are damn expensive
18:42.05halogen8Ariel_: only one can answer though?
18:42.08Ariel_if that is how you setup your rules yes
18:42.12Ariel_correct
18:42.21Ariel_Hmmhesays, yes they are
18:42.25halogen8Ariel_: can you set it up so all extensions can answer?
18:42.33halogen8Ariel_: probly not, but just though i would ask
18:42.35Hmmhesaysbut channel banks are a bitch to work with sometimes
18:42.43Ariel_halogen8, yes but sending them to a meetme but that will take more doing
18:42.55halogen8Ariel_: thank you.....i understand now
18:43.32Ariel_Hmmhesays, that is a loaded question is it not! sometimes is always sometimes....
18:44.12Ariel_Hmmhesays, I have found it very easy to work with channel banks.
18:46.57MrCh|ckenAriel . . hello
18:47.08MrCh|ckencould you give me a hand?
18:48.13MrCh|ckenhahahahahahahah
18:48.33MrCh|ckenlike yours obsidian he?
18:48.54MrCh|ckenthen give me some help?? sound better obsidian??
18:48.59obsidian-studiosMrCh|cken:  nah I need my hands to type ;) I tried one handing it after a skateboarding incident but no go
18:49.23tubes41nextime: Thanx 4 that lib info... works like a charm now :D
18:49.30obsidian-studiosMrCh|cken:  what's your problem, Dr. Phil is still training me ;)
18:49.38Kattyobsidian-studios: beep!
18:49.53MrCh|ckenobsidian . . . thanks
18:50.13MrCh|ckenobsidian  . . .the last friday my asterisk box was crazy
18:50.32HmmhesaysKatty: did you ever listen to "Doom Boom" off the hot action cop album?
18:50.33MrCh|ckenX-Rob helped me to put it up
18:50.50obsidian-studiosMrCh|cken: setup and exten =>s,1,Dial(Dr.Phil) have your * box tell Phil what makes it crazy ;)
18:50.56MrCh|ckenobsidian . . since the SIP extension soun all the time busy
18:51.30KattyHmmhesays: probably
18:51.30obsidian-studiosMrCh|cken: hmm, incoming calls or internal
18:51.35MrCh|ckenobsidian  . . .but the zap are functioning Ok (minor problems)
18:51.48MrCh|ckenobsidian . . internal
18:51.54halogen8what do you all think of the Linksys PAP2 ATA?  is it as good as Cisco ATA-186?
18:52.26MrCh|ckenobsidian . . my SIP phones are connected trough ATA 186
18:52.34HmmhesaysI love that song
18:52.34obsidian-studiosMrCh|cken: ok so you pick up a phone, dial and exten, and get a busy signal? If so more than likely your extensions are not correct. I was having a similar problem last night
18:52.55obsidian-studiosMrCh|cken: can you call in from a zap channel and get to your sip lines, less the busy signal?
18:53.20MrCh|ckenobsidian  . . when I type the SIP extensions both sound busy!
18:53.24*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
18:53.41MrCh|ckenobsidian . . yeeeeppppp
18:53.57MrCh|ckenobsidian  . . i only can use the zap extensions
18:54.07obsidian-studiosMrCh|cken: internal & external = busy ?
18:54.11MrCh|ckenthe internal SIP extensions always sound busy
18:54.18obsidian-studiosMrCh|cken: ok just internal busy, and external rings
18:54.37obsidian-studiosMrCh|cken: sounds like an problem with the extensions, pastebin it
18:55.03obsidian-studiosexcuse me, pastebin the context so I can see the extensions
18:55.31MrCh|ckenobsidian . . if i type the extensions fron outboun the messege says that this extension is aou of service
18:55.41mutilatormy leg keeps twitching@!!@$
18:55.50obsidian-studiosphew, I am glad the qWer dude left :) I think he grabbed my butt
18:56.15obsidian-studiosmutilator:  put down the vibrator ;)
18:56.24Ariel_halogen8, the pap2-na is better and also the sipura 2002
18:56.55obsidian-studiosMrCh|cken: are the sip devices registering and available? Sounds like the sip lines are not working via internal dialing an extension, or external via zap?
18:57.44MrCh|ckenobsidian . .it is a strange thimg . . . because
18:57.45Ariel_MrCh|cken, if you go to the asterisk boxes console you can do sip show peers to see if your ata's are login
18:57.57MrCh|ckenthose extensions sound with its group
18:58.13MrCh|ckenbut don't allow internal switching
18:58.20*** join/#asterisk razu (n=razu@hps713.kodunet.webs.ee)
18:58.23mutilatorwonder if i should throw this new dial plan in my production system mid day
18:58.29Ariel_MrCh|cken, then your context are not setup correctly
18:58.37halogen8Ariel_: the sipura is also better than the cisco?  You wouldn't be able to tell by the price......the linksys is dirt cheap....
18:58.38mutilatortested it out for a while now and it SEEMS to work fine..
18:58.39*** part/#asterisk azzie_ (n=az@azzie.net)
18:59.20MrCh|ckenobsidian . . the sip devices are registering when i done the sip show peers
18:59.21Ariel_halogen8, the cisco's correct cisco's always charges more
18:59.48halogen8Ariel_: which ata do you recomend that has two ports?
18:59.50MrCh|ckenAriel . . .the ata are logged
18:59.57Ariel_sipura
18:59.59obsidian-studiosMrCh|cken: hmm, how are you referring to them, pastebin a context
19:00.07halogen8Ariel_: sipura is better than linksys?
19:00.16Ariel_halogen8, linksys are made by sipura
19:00.26Ariel_there all owned by Cisco
19:00.31MrCh|ckenariel . . i've gone to the ATA and trough ther phone i've confirmed the same IP address
19:00.39Ariel_Linksys, sipura, Cisco owns them
19:00.42*** join/#asterisk bumblefsck (n=bumblefs@69-160-158-193.ontrca.adelphia.net)
19:00.51obsidian-studioswonder is Cisco has eyes on Digium ?
19:01.02Ariel_no please not that
19:01.13bumblefsckwhat is the bot's name?  can i ask it questions?
19:01.19obsidian-studiosMrCh|cken: what about the other way? Can you call out or other extensions from the sip lines?
19:01.45MrCh|ckenobsidian  . . nooooooooooppppp
19:01.47Delta34anybody know how to display the called party name when dialing an extension on a cisco 7960 phone?
19:01.53obsidian-studiosbumblefsck: I am not sure, but it will not tell you the meaning of life ;)
19:02.26Ariel_jbot
19:02.29MrCh|ckenobsidian  . . . sorry . I'm totally newby . . .how can i made the pastebin??
19:02.29obsidian-studiosMrCh|cken: so you can't do crap with your sip lines? pastebin your sip.conf and any contexts used in teh sip.conf
19:02.35Ariel_~weather ktmb
19:02.46Ariel_~pastebin
19:02.47jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
19:03.21obsidian-studiosMrCh|cken: there you go, just slap code in there, hit submit post, and paste the url here ;)
19:05.00MrCh|ckenobsidian . . . is posible to you to look inside???
19:05.57obsidian-studiosMrCh|cken: look in your box?
19:06.36bumblefsckI have a really basic question about USB phones...  are they basically just a USB speaker and a USB Mic?  Or do they run seperate from the soundcard?  sorry if this is off-topic.
19:07.20Ariel_bumblefsck, depends on the usb device. But most have there own soundboard and mic
19:07.20mrfrenzybumblefsck: it's a soundcard, speaker, mic, numeric keypad
19:07.24Delta34can anybody answer how to display the called party name when dialing an extension on a cisco 7960 phone?
19:07.49Ariel_Delta34, I can't
19:07.50MrCh|ckenobsidian . . .yeeppp
19:07.56Delta34does asterisk need to send something back to the intiated caller
19:08.23obsidian-studiosMrCh|cken: do not really have the time for free :) trying to make some $, but if you pastebin your configs, I will take a quick look
19:08.24*** join/#asterisk epablo (n=epablo@200.75.139.188)
19:08.39epabloHi people.. How's it going?
19:09.19MrCh|ckenobsidian . . okis  . . I'm tring
19:10.14obsidian-studiosMrCh|cken: np, just copy and paste to pastebin, it's a great resource
19:10.28epabloHas anyone used Asterisk::AGI with perl v5.8.6.  My AGI starts, I get no errors, but it doesn't even show me the call environment.
19:10.47shmaltzhow can I capture the CLI to a file?
19:11.29outtoluncepablo: you must have both of these
19:11.32outtolunc$AGI = new Asterisk::AGI;
19:11.32outtoluncmy %input = $AGI->ReadParse();
19:11.40Assidhrmm.. just wondering.. is mysql any faster/slower than pgsql in inserts/reading ?
19:12.00epabloouttolunc: yeap.. I have that.
19:12.10Ariel_shmaltz, use putty
19:12.26hardwireI really want MoH to support an arbitrary channel
19:12.36*** join/#asterisk Primer (n=nnnnNpri@sh.nu)
19:12.44Ariel_Assid, that is a very bad question. Which many have there own views here.
19:12.51Assidhrmm
19:12.53Ariel_Assid, I like and use MySQL
19:12.59Assidokay .. how about any licensing issues ..
19:13.02hardwiredefault => channel:Local/moh@otherserver
19:13.12Ariel_but there are die hard people who just love pgsql
19:13.14Assidi mean mysql has thuis funny licensing.. is any of that affected by the use with asterisk?
19:13.29Ariel_Assid, no
19:13.40*** part/#asterisk santiago (n=santiago@63.245.86.203)
19:14.21Assidwhat if i am a provider.. like a middle man..
19:14.23outtoluncepablo: agi debug on the cli will show you whats happening
19:14.29PrimerCan someone show me an example of an extension that tries a user on a software sip client first, and, if that client isn't connected, goes to the trunk and dials her cell?
19:14.30MrCh|ckenobsidian  . . .how I can capture the pastebin??
19:14.31Assidjust setting up routings and tuff
19:14.39epabloouttolunc:  thanks let me look at it
19:15.00shmaltzAriel_ that only allows for a few lines, I want more than that
19:15.16obsidian-studiosMrCh|cken: what do you mean, just paste in stuff in the box, hit submit, and it will generate a url you paste here
19:15.17Ariel_shmaltz, log file log file settings
19:15.28PrimerI'm mostly wondering about the "detection of the sip client being connected". I'd rather that be immediate than have to wait for a specific number of seconds to time out
19:15.42shmaltzAriel_, you meant to /var/log/asterisk/messages?
19:15.43Ariel_Primer, normal macro's can do that
19:16.02*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
19:16.06Ariel_shmaltz, you can setup your putty to record everything to a log file
19:16.19shmaltzAriel_, thanks
19:16.21Ariel_obsidian-studios, he is new to linux in general
19:16.24*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
19:16.26PrimerMrCh|cken: I have a paste site at http://sh.nu/p that shows a handy curl command line for "pasting" entire files to it
19:16.29Primerit's quite handy
19:16.52MrCh|ckenobsidian  . . . how long could be your review . . and how much it coul be??
19:17.27Ariel_MrCh|cken, you really need to hire a consultant to work with you.
19:17.29obsidian-studiosMrCh|cken: I have no clue, I am not an * expert, but $50 per hour?
19:17.40wasim$250!
19:17.56Ariel_$ 75.00 here pre-paid
19:17.56MrCh|ckenwasims . . .washhhhhhhhhhhhhhhh
19:18.04*** join/#asterisk ianm (n=ianm@63.224.101.51)
19:18.58*** join/#asterisk bassie (n=bas@datarack.xs4all.nl)
19:19.12bassiehello
19:19.20ianmHi
19:19.38bassieI was wondering if there is someone who has successfully setup iaxtel on their asterisk config
19:19.40*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
19:19.46wasimbassie: a billion people have
19:19.48wasimlike RoyK
19:19.58bassiecool :-)
19:20.01wasimeven like RoyK, even
19:20.03bassiemy box registers fine
19:20.15bassiebut I can't make outbound calls to another iaxtel user
19:20.17Ariel_bassie, iaxtel has lots of issues
19:20.22bassieiax2 debugging shows nothing
19:20.29wasimbassie: its prolly on the iaxtel side
19:20.34wasimbassie: is the other side up?
19:20.34ianmiax2 show registry?
19:20.36bassieI added the register line to iax.conf
19:20.36Ariel_seems that there server is down lot
19:20.48*** join/#asterisk Byte (i=byte@2001:4bd0:1000:0:202:44ff:fe47:d3ee)
19:21.30bassie@ianm: 69.73.19.178:4569     bas         80.126.xxxxx
19:21.49bassieso it registers fine
19:22.21*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
19:22.23MrCh|ckenobsidian . . . but with my tracks . . .you think that could you resolve the trouble??
19:22.25funxionhass anyone ever connected an * box to a meridian and tried to use meridian mail through *?
19:22.38bassiebut I sense that iaxtel is not something to use a lot?
19:22.47ianmfunxion: why why why !!!
19:23.08funxionbecause its an intrgration and I would like everyone to use the same voicemail system
19:23.13obsidian-studiosMrCh|cken: I have no clue, seeing the conf files would really help? can you not pastebin them?
19:23.50Primeror http://sh.nu/p them!
19:24.00MrCh|ckenobsidian . . .I'm tried  . . but can't
19:24.22funxionI'm adding 100 * users to a 400 user meridian and everyone knows how to use meridian mail
19:25.25ianmbassie: It should show something like this..... Host                  Username    Perceived             Refresh  State
19:25.25ianm198.22.xxx.xxx:4569     yourusername      65.204.xxx.xxx:60769       60  Registered
19:25.42Ariel_funxion, it should work it just upto you on how you setup the voicemail settings. But your not going to be able to know if you have voicemail on either side of the system.
19:25.56ianmfunxion: I'm sorry - we too have the same problem here at work....
19:26.09ianmfunxion: we ended up going one way or the other.
19:26.33*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
19:26.34bassie@ianm: it shows that
19:26.49ianmbassie: if you're missing the Registered from iax2 show registry?
19:26.52bassiewith my info obviously
19:26.57*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
19:27.02bassieah
19:27.03funxionryte now my problem is that I can get vmail calls to go to the meridian but it sees the call as coming from the extension for which the voicemail is for
19:27.12ianmbassie: so under state is says Registered ?
19:27.16funxionthe getting mwi to werk is the next step
19:27.19bassieHost                  Username    Perceived             Refresh  State
19:27.19bassie69.73.19.178:4569     bas         80.126.152.61:50170        60  Registered
19:27.26ianmbassie: ok
19:27.46bassieso that looks ok, right
19:27.53ianmbassie: so the next thing would be to enable the hell out of iax2 debug
19:28.02ianmbassie: that looks awesome
19:28.27bassieianm, I enabled iax2 debugging, but it keeps its mouth shut
19:28.32ianmbassie: and if no debug shows when someone calls in......
19:28.50ianmbassie: then iaxtel isn't linking the DID with your IAX...
19:29.04ianmbassie: I'm assuming your iaxtel account has a DID right ?
19:29.09bassieuhm
19:29.09epabloUsing a test AGI using perl and Asterisk::AGI.  I turned on the agi debug. I can se the AGI TX, but the script isn't getting the values.  Any ideas?
19:29.22bassiewhat's a DID (sorry for being such a newbie :) )
19:29.27bassie?
19:29.37ianmbassie: direct inward dial number - like a normal PSTN number
19:29.47bassieyou mean my 1700 number
19:29.48r0d3nthi
19:29.51ianmbassie: so the outside world can call yyou...
19:29.53bassieyes, I've got one
19:29.59bassiehold on
19:30.01macTijnepablo: did you do a ReadParse() ?
19:30.10macTijnepablo: you *need* to do that :)
19:30.29epablomacTijn: Yes.. did that and the print also
19:30.39macTijnthe print ?!
19:32.04epabloAny other debug I can do.. to se whats wrong?
19:32.18macTijnser verbose 3 :)
19:32.20macTijnehh
19:32.23macTijnset verbose 3
19:32.28epablocon verbose 33
19:33.07epablo<PROTECTED>
19:33.09epabloAGI Tx >> agi_request: auth-local-Tellfree.agi
19:33.09epabloAGI Tx >> agi_channel: SIP/5519913209-be07
19:33.09epabloAGI Tx >> agi_language: en
19:33.09epabloAGI Tx >> agi_type: SIP
19:33.09epabloAGI Tx >> agi_uniqueid: 1126714961.25
19:33.10epabloAGI Tx >> agi_callerid: Pablo <9913209>
19:33.12epabloAGI Tx >> agi_dnid: 0013057220500
19:33.14epabloAGI Tx >> agi_rdnis: unknown
19:33.16epabloAGI Tx >> agi_context: Tellfree
19:33.18epabloAGI Tx >> agi_extension: 0013057220500
19:33.20epabloAGI Tx >> agi_priority: 1
19:33.22epabloAGI Tx >> agi_enhanced: 0.0
19:33.24epabloAGI Tx >> agi_accountcode: 551
19:33.26Ariel_pastebin
19:33.26epabloAGI Tx >>
19:33.30epablogaim played me one.  Sorry
19:34.03epabloThats what I see, but the script doesn't receive the info :s
19:34.24macTijn%bla = $agi->ReadParse();
19:34.34macTijnthen %bla contains that stuff
19:35.27epabloThats what I did:  $AGI = new Asterisk::AGI;
19:35.28epablomy %input = $AGI->ReadParse();
19:35.28epabloforeach $i (sort keys %input) {
19:35.29epablo<PROTECTED>
19:35.29epablo}
19:36.01epablodoesn't print anything
19:36.15*** join/#asterisk zzzirk (n=lzirkel@c-67-164-205-60.hsd1.ut.comcast.net)
19:36.38*** part/#asterisk zoo (i=nobody@ip-117-16.travedsl.de)
19:38.45*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
19:40.04Darwin35ls
19:40.48*** join/#asterisk Spy_Pc (i=t7DS@201.19.44.227)
19:41.21*** part/#asterisk Spy_Pc (i=t7DS@201.19.44.227)
19:42.15macTijnepablo: what is "OUT" ?
19:42.36epablofile open so I could see the output..
19:42.45blitzrageepablo: are you connecting to the Asterisk console via a -r ?
19:42.48blitzrageor in screen ?
19:42.59epablomacTijn:  Thanks for your help got it working
19:43.15*** join/#asterisk vietyen (n=blurk@oxygen.adsl.utwente.nl)
19:43.15spackleIf you could have a stable (no crashes)  asterisk with only the features currently avialable in head, or continue the way it is now, which would you prefer?
19:43.17vietyenhi
19:44.04blitzragespackle: thats a loaded question :)
19:44.15macTijn<PROTECTED>
19:44.46vietyeni've got a 537EP clone that gives a ZT_CHANCONFIG error, could it be the case that the Serial driver is locking the modem in such a way that wcfxo cannot get a grip on it?
19:45.47*** join/#asterisk smcmahon (n=admin@digitaldatabits.net)
19:47.35ianmmacTijn: did you just remove the OUT from the print line ?
19:48.31*** join/#asterisk cursor (n=kevin@andromeda.office.cursor.biz)
19:50.09macTijnianm: that wouldn't be wise :)
19:50.34spackleblitrage - why is it?  I would rather have 20% of the features if it ran solidly, as expected 100% of the time.
19:50.55epablomacTijn: Reintalled the perl package.. and it worked.. the script was OK
19:50.59ianmmacTijn: sorry bud - I realised I sent the message to the wrong person ;)
19:51.06ianmOhhhhhhhh
19:51.12ianmThat's interesting....
19:51.30ianmreinstalled  Asterisk::AGI ?
19:51.36epabloyes!
19:51.43ianmWow - from CPAN ?
19:52.10epabloNo.. I had it from somewhere in my machine..
19:52.20ianmcool - that's good to know - thanks
19:52.29epablowhat version is in CPAN?  I'm using 0.8
19:52.38*** join/#asterisk jpablo (n=jpablo@dsl-200-78-66-23.prod-infinitum.com.mx)
19:52.48ianmHmm - don't know - let me take a look
19:52.55jpabloanyone can recommend a multi fxo (around 6) ATA ?
19:53.15jpabloto hook into asterisk
19:53.20epabloI saw one on voip-supply.com
19:53.34jpabloepablo: there are several, i was looking for a recomendation ..
19:54.14epablo<PROTECTED>
19:54.27ianmCPAN doesn't even have Asterisk::AGI !!!
19:55.12jpabloepablo: i have tried the digium cards, but they are kind of unstable.
19:55.15SwK[Work]wheres the audiocodes users at?
19:56.04epablojpablo: I just orderd my first digium cards today.. I'll give you my impresion in a couple of days
19:56.23jpabloSwK[Work]: are audiocodes any good ?
19:56.33jpabloepablo: what kind of card?
19:56.43epabloTDMxxxx
19:56.56jpablothey suck, at least in my experience.
19:57.18jpablothe fxs modules work fine, the fxo modules have problems.
19:57.32epabloIt's nice to know.  If the don't work for me I'll sell them ;)
19:58.07ianmjpablo: what card(s) would you recommend ?
19:58.30jpabloianm: im investigating ATAs solutions.
19:58.47jpablothe problem with cards is that you can't put that many in a single machine.
19:58.51ianmjpablo: make sure they pass callerid !!!
19:58.53epabloianm, macTijn.  Thanks for your help.
19:58.56synthetiqif u have exten => 1, blablabla   on top of exten s,1,balabla  will s or 1 be answered first?
19:59.01epablogot to go.  C'ya around
19:59.12ianmCya epablo
19:59.15synthetiqi will ahve to assume 1 but s also indicated the start of a context?
19:59.16*** part/#asterisk epablo (n=epablo@200.75.139.188)
19:59.18SwK[Work]jpablo: for somethings... I'm haveing a fax prbloem with them right now tho... its not kicking into fax passthru properly
20:00.05jpablosynthetiq: s would answer if s is called, 1 will answer if 1 is called ...
20:00.29ianmsynthetiq: isn't s always assumed?
20:00.32SwK[Work]jpablo: the 4 and 8 port ones are nice fpr putting in front of KeySystems to convert small offices to VoIP
20:00.38synthetiqi thought s stood for all extensions in general
20:00.57ianmHmmm - I'm a neebie !! :(
20:01.01ianmsorry
20:01.06SwK[Work]they also make a kool little 8 port FXO box for bringing 8 lines into a asterisk box
20:01.14synthetiqbecause the extensions above s need to be allowed any time but below s have time limits
20:01.22SwK[Work]s is start
20:01.32ianmthat makes sense :)
20:01.36SwK[Work]its a special extension
20:01.36blitzrageStar Wars Kid!
20:01.37jpablosynthetiq: s is the start extension, would be called from zaptel channels.
20:01.49synthetiqok
20:01.57synthetiqi have to use a gotoif time
20:02.06SwK[Work]BLITZ KRIEGE!^H^H^H^H^HRAGE!!
20:02.11ianms could be the start extension from sip and iax too
20:02.16jpabloSwK[Work]: i have a small (6 trunks) office, need to get thoses trunks into asterisk somehow.
20:02.36synthetiqif i want to allow exten 1 to eb called anytime and exten 2 to work during the dya only...how would i go about that, not usign different contexts
20:02.40SwK[Work]jpablo: TDM400s or the Audiocodes MP-108/FXO box
20:02.40jpabloianm: most of the time a start extension doesn't make sense for sip and iax.
20:03.03jpabloSwK[Work]: having worked with tdms before i think i will try the mp-108
20:03.13synthetiqexten => 1,dial  exten=>s,gotif  exten=> 2,dial
20:03.17*** join/#asterisk hassler (n=hassler@cpe-65-31-36-179.woh.res.rr.com)
20:03.18ianmjpablo: really? I find if I miss the context off the register string then it looks for s start extension...
20:03.34synthetiqor should i use ${EXTEN} in place of s
20:04.06*** join/#asterisk nicox (n=nicox@h082218027030.host.wavenet.at)
20:04.49BrianR___crap. my asterisk box just caught an nmi :(
20:04.52*** join/#asterisk sudhir492 (n=sudhir@pool-71-114-87-156.washdc.dsl-w.verizon.net)
20:04.56sudhir492Hi all
20:05.00ianmnmi ?
20:05.01nicoxhello, is anybody there who know something about the zaptel and/or libpri code?
20:05.12BrianR___ianm: non maskable interrupt - usually means hardware trouble on x86
20:05.23synthetiqwhat im doign is sorta subcontexts
20:05.26BrianR___also has a flashing amber led on front panel. need to check the embedded management :(
20:05.37ianmBrianR___: wow - would it help to change the PCI slot ?
20:05.51*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com)
20:06.02synthetiqnicox ask in #asterisk-dev
20:06.09nicoxthanks
20:06.16Ariel_synthetiq, what are you trying to do?
20:06.24BrianR___ianm: no idea.. didn't crash though, which is refreshing..
20:06.36ianmBrianR___: sweet ;)
20:06.37BrianR___first day with live traffic going through our asterisk box... A crash would be very embarassing...
20:07.11ianmHow many users ?
20:07.21BrianR___ianm: over a hundred ;(
20:07.41ianmwow - I see what you mean ;)
20:07.42synthetiqim ttying to prevent the group of extensions below the gotoiftime from makign calls at a certain time while on the other hand anythign about should have eprsmission to calla nytime
20:08.35sudhir492Anyone using Queues here?
20:08.44Ariel_synthetiq, make 2 different contexts for them with the correct includes
20:08.47EssobiAny heavy SER users here?
20:08.54*** join/#asterisk jesster (i=jesster@207.71.207.39)
20:09.00fileEssobi: yes no maybe so
20:09.10Ariel_Sip Express Router == no not me.
20:09.13sudhir492How to block incoming calls in a queue?
20:09.17jessteranyone bychance using Polycom with 1.5.3 SIP firmware?
20:09.35*** part/#asterisk tubes41 (n=tubes41@203-59-179-19.dyn.iinet.net.au)
20:09.41nicoxdoes anybody know why asterisk is resetting every channel of an E1 at least once in an hour?
20:09.41Ariel_jesster, I have not seen the 1.5.3 I have the .2 only.
20:09.55jessterAriel_: 1.5.3 is a few weeks new
20:09.58Ariel_nicox, standard settings
20:10.07Ariel_and if one is in use it skips it
20:10.42jessterAriel_: are you using multiple registrations (backup proxy) in your Polycom config?
20:10.45synthetiqwhat if i dont want to use different contexts?
20:10.51Ariel_jesster, no
20:10.52jessterAriel_: 1.6.2 is out for beta users too
20:10.56synthetiqariel_
20:11.07Ariel_synthetiq, hummm going to be hard
20:11.15synthetiqi mean
20:11.18synthetiqthe question is
20:11.30synthetiqdoes asterisk go thru the dialplan in order
20:11.31nicoxAriel how can i change this?
20:11.32Ariel_asterisk is context drivin
20:11.51synthetiqor are the extensions spread otu randomly in memory
20:11.54Ariel_synthetiq, it goes via it's context first then sorts the other contexts
20:12.16synthetiqyea, but with in each context, does it go in order
20:12.25Ariel_synthetiq, sorts
20:12.28synthetiqie , will it look at extension 1 before extension 2
20:13.02synthetiqif u have it in order or exten => 1,1....     exten => 2,1
20:13.03Ariel_synthetiq,???? sorts 1 does come before 2
20:13.22synthetiqhmmm
20:13.25synthetiqok but
20:13.41synthetiqif i have in the file exten 2 before 1,,,, will 2 be answered first or 1
20:13.43Ariel_synthetiq, do show dial plan
20:13.52synthetiqtoo big lol
20:14.01*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
20:14.02Ariel_putty log
20:14.38EssobiFile You use SER?  I'm trying to figure how how to packet mangle headers in a "200 OK" before it gets pushed on to another host.
20:16.24*** join/#asterisk hellagony (n=egutierr@200.121.129.180)
20:16.58*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
20:18.08*** part/#asterisk pa (n=Paolo@unaffiliated/pa)
20:18.27*** part/#asterisk jbroome (n=jbroome@63-168-10-93.celito.net)
20:19.15Drizzt321If I do 'show channels' on the asterisk console, and I get nothing in return, that means I won't be able to have incoming, or outgoing calls even if my softphone and SIP-PSTN connection is registered and connected, right?
20:19.42dexteruk<PROTECTED>
20:20.58sudhir492jesster: I am using SIP ver 1.5.2.0054
20:21.43Drizzt321anyone?
20:21.55Assidwhats better.. tos=lowdelay / tos=throughput ?
20:23.09spackle~seen maryjane69
20:23.10jbotmaryjane69 <n=eetfunk@MTL-HSE-ppp196800.qc.sympatico.ca> was last seen on IRC in channel #asterisk, 5d 27m 44s ago, saying: 'I am having problems receiving incoming calls from the PSTN or by doing "7777" on my softphone.  From the PSTN, I get a fast "busy" signal, and from the softphone, asterisk simply hangs up on me.  My PSTN line is connected to my ...
20:26.35Ariel_Drizzt321, you don't show any channels until there in use.
20:26.41*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
20:27.02Drizzt321Ariel_: ahh, ok. now I just need to get them in use... :(
20:27.16*** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
20:27.29Drizzt321Ariel_: cause for some reason I can't seem to get anything incoming, or dial out
20:28.28Drizzt321Ariel_: I'm trying to use broadvoice, I got the SIP connection from asterisk to broadvoice registered(seems correct), and from my softphone on another machine to asterisk connect, with what I think is an approrpiate dialplan, but nothing...nothing incoming, and very nothing outgoing
20:28.41FuriousGeorgehi all
20:29.01*** join/#asterisk ixx (i=foobar@cpe-24-27-44-163.austin.res.rr.com)
20:29.06Drizzt321Ariel_: any chance you could help me out?
20:30.27Hmmhesaysbad to the bone
20:30.28*** join/#asterisk Byte_ (i=byte@proxima.arlott.org.uk)
20:30.29Hmmhesaysbbbbbbbad
20:30.37NORANDOMSsay i have an asterisk server behind NAT and a client behind different nat
20:30.45NORANDOMSdo i need to forward anything other than 5060?
20:31.04Drizzt321NORANDOMS: from what I've read(read I have no experience), you need a man in the middle to get the connected
20:31.11Ariel_Drizzt321, yes now when you dial to your box do you see anything in cli ? (broadvoice incoming)
20:31.12Drizzt321NORANDOMS: at lesat with sip...
20:31.23Drizzt321Ariel_: nope, and I set the verbosity to 5
20:31.31Ariel_Hmmhesays, really badddddd
20:31.41Ariel_Drizzt321, sip debug
20:32.06Ariel_I only have a few minutes I need to go to an appointment in about 10 minutes.
20:32.43Drizzt321Ariel_: ah, ok. well, I have a connection between the softphone(xten-lite), and I see packets go by
20:32.45NORANDOMSdriz: so my server needs to not be behind nat?
20:33.20Drizzt321NORANDOMS: erm...I think. have you looked on voip-info.org? they have some good info
20:33.33Drizzt321Ariel_: it ends with a 404, oddly enough
20:33.38Ariel_your servers can be behind nat firewalls there settings need to be set correctly.
20:33.55NORANDOMSah
20:33.58Ariel_Drizzt321, I will have to say that you have some setting incorrect.
20:34.04HmmhesaysLOL
20:34.06NORANDOMSariel: do you know of which offhand?
20:34.09Ariel_use pastebin.ca for your settings remove your password.
20:34.12Drizzt321Ariel_: yea, but I'm at a loss to figure out what it is
20:34.17Ariel_NORANDOMS, yes
20:34.32Ariel_so do allot of others here and it's listed on the wiki
20:34.35Ariel_~docs
20:34.37jbotsomebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
20:34.37NORANDOMSok thanks
20:35.12Drizzt321Ariel_: any thoughts on which settings might be off? my dialplan?
20:35.42Ariel_Drizzt321, at this stage without any real error or looking at your settings no it's just a guess
20:36.23Drizzt321Ariel_: hrm...any chance you might be around later this evening to help me?
20:36.50Ariel_Drizzt321, yes there is a good chance of that.
20:37.03Drizzt321Ariel_: about what time?
20:37.12Drizzt321Ariel_: I'll be eternally grateful
20:37.59Ariel_time well that is a question. After here I will be going home. At home I have to at least play with my baby girl for about 1 hour. So around 7 or 8
20:38.08Ariel_it's 4:38 pm now
20:38.26Drizzt321Ariel_: cool, I'll make sure to be around then
20:38.27Drizzt321:)
20:38.31*** join/#asterisk fugitivo (n=ajf@201.255.101.33)
20:38.50synthetiqHow can you send street name and house # to e911?
20:38.57enderwhat is the default timeout time for exten => t  ?
20:38.57synthetiqbecause they require that
20:39.30Ariel_argh e911 ani settings that is a big problem your going to need to do some reading and programming or hire a co to do it for you.
20:39.48synthetiqrut roh
20:39.50Drizzt321Ariel_: sweet, I look forward to that eventually...lol
20:39.59fugitivohello
20:40.08synthetiqill ask the developers
20:40.14*** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com)
20:40.18synthetiqif there is soemthign in the plans
20:40.33Craziman2can someone tell me how to stop this message:
20:40.34Ariel_fugitivo, como esta's
20:40.35Craziman2Sep 14 15:42:19 WARNING[905]: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?)
20:40.44fugitivoAriel_: bien y vos? :)
20:40.55Craziman2I get it on trunk connection between latest cvs and 1.0 stable
20:41.25Ariel_fugitivo, estoy bien. (That is about all my spanish)
20:42.04fugitivoAriel_: at least you could start a conversation ;)
20:42.05Ariel_ok meeting is going to start see you all later.
20:42.05NORANDOMSok i got the phones able to dial each other but i'm not geting audio
20:42.07*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
20:42.34Craziman2Sorry for the spread message... trying to stop this message between 1.0stable and latest cvs on an IAX trunk...  Sep 14 15:42:19 WARNING[905]: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?)
20:42.55Ariel_NORANDOMS, look at externip and localnet settings in sip.conf also make sure you have ports 10,000 to 20,000 open for sound.
20:43.26NORANDOMSah
20:43.44NORANDOMSi thought ony 5060 was used
20:43.45NORANDOMSthanks
20:44.11doughecka_Anyone have a solution for a way to manually set incoming calls to goto an answering service? I need the ability to dial a code, and have it send all calls to an external number.
20:45.43fugitivodoughecka_: i don't understand what you mean
20:45.43anthmat least 10 ways off hand
20:47.02*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
20:47.41anthmthe fastest one would be little ol' astdb
20:47.45hardwirehttp://blog.tmcnet.com/blog/tom-keating/images/cisco-ip-phone-7985G-videophone.png
20:47.46hardwireweird.
20:47.52anthmthen there is agi gateway to dbi
20:47.57anthmflatfile
20:48.02anthmexten reweite api
20:48.24anthmglobal variable
20:50.32doughecka_well
20:50.46doughecka_what I want is whenever the last person leaves
20:50.56doughecka_they dial *79 (or whatever)
20:51.27NORANDOMShell yea ariel
20:51.29NORANDOMSi got it workin
20:51.33*** join/#asterisk denon (i=denon@synapse.subneural.net)
20:51.33*** mode/#asterisk [+o denon] by ChanServ
20:51.45doughecka_and that will switch something around so when a call comes in, it knows to just pass it through to an outside number
20:52.09Craziman2Does anyone know the option to turn off iax timestamps?
20:52.25doughecka_why
20:52.27*** join/#asterisk fulgas (n=fulgas@a81-84-117-187.cpe.netcabo.pt)
20:52.39sudhir492doughecka_: There are many ways you can do that
20:53.01anthmmake ecten *79 do a dbput with the new dest and make the new call do a get on the same key would be the most painless
20:53.02sudhir492Are your people in a queue?
20:53.31sudhir492How are your calls handled in usual case?
20:54.05doughecka_no
20:54.10doughecka_it rings into a ring group
20:54.22doughecka_after 30 seconds, it goes to an IVR
20:54.30IOscannerIs there a kernel patch for zaptel drivers?  I would like to build into the kernel source instead of patching each system.
20:54.36doughecka_but they want to take out the ring group and have it go directly to the asnwering service
20:54.38sudhir492all ring at once I presume?
20:54.41doughecka_yes
20:55.08*** join/#asterisk rene- (n=rene@dsl-201-128-87-236.prod-infinitum.com.mx)
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20:55.38*** part/#asterisk Craziman2 (n=Craziman@boromir.apid.com)
20:56.28sudhir492One of the easiest way is to edit the queues.conf file and reload when someone dial *79 (forward to answering service), or *80 (dont forward to answering service)
20:56.48doughecka_well, I am not using queues
20:56.56doughecka_its just ringing all the phones ,then times out to the IVR
20:57.04sudhir492np.
20:57.14clive-can anyone comment on if its worthwhile using ztdummy for iax2 trunking timming purposes ?
20:57.31sudhir492edit your extensions.conf file and extensions reload on dialing *79 or *80
20:57.50doughecka_s
20:58.03sudhir492you can have alternative extensions.conf files, and load the appropriate one
20:58.10anthmif it's head they are functions
20:58.12rene-does nextel works on celullar protocols like gsm and such?
20:58.17anthmprehead it's apps
20:58.17rene-s/on/over
20:58.25doughecka_oh, well, I have 2 phone systems running through 1 asterisk box, very complicated setup
20:58.28file[laptop]nextel uses iDEN, a proprietary thing
20:58.32anthmshow function DB
20:58.46file[laptop]it's based on GSM
20:58.46doughecka_its stable
20:58.49doughecka_1.0.9
20:59.14anthmthen you need show application dbput and a lot of luck
20:59.35rene-file: i was looking at the dock-n-talk cradles, do you think they interfase with the phone or do they actually speak iDEN?
20:59.39doughecka_blah :)
20:59.45file[laptop]rene-: interface with the phone
20:59.56enderwith a static config file, no agi or anything, is there a way to do time based call handling?  IE Between the times of foo and bar, dial baz, otherwise dial baal ?
21:00.14doughecka_ender: YES! dont ask me how, but its very possible
21:00.18anthmapp_rtfmiftime
21:00.21sudhir492doesnot matter how many phone systems, all you have to do is to keep multiple extensions.conf files, and in your agi, just do: mv rightexten.conf extensions.conf, and then reload
21:00.44doughecka_sudhir492: ooh, ick, I dont like that :)
21:00.47spackleender: yes, want an example?
21:01.45enderspackle: yesh, that would be very nice.
21:02.46doughecka_do I have to initialize anything with dbput?
21:02.56sudhir492thats why asked if you use queues.
21:03.00doughecka_or can I just do dbput(ans/blah)
21:03.18*** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net)
21:03.56spackleender: http://pastebin.ca/22937
21:04.10anthmyou can dbput all yo uwant
21:04.20anthmits a dbm file
21:04.26tzangerwhee
21:04.39tzangerjust spent all day making a CAN controller board
21:04.45doughecka_CAN controller board?
21:04.47doughecka_NO WAY
21:04.50doughecka_OMGWTFBBQ
21:05.07spackleBBQ?
21:05.08doughecka_anthm: cool, let me see if I can hack soemthing up :)
21:05.21doughecka_spackle: yea, like the cooked meat and sauce
21:05.24BrianR___Anyone know if the early dial stuff on the grandstream gxp-2000 acutally works? It seems that for variable length entries in the dial plan (ie, _9011.), it does the wrong thing.
21:05.26tzangerdoughecka_: heh
21:05.47spackleDougchecka, I see, like quel frommage?
21:05.51doughecka_yea
21:06.48enderspackle: interesting.  I can't quite make out how the time checking does anything different.  Goes to next priority, but if the time isn't in that period, it goes there anyway... or am I wrong?
21:07.11*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
21:07.45spackleender, oops, forgot I changed it to keep them from getting calls on cell phone.  You can see what to change to remedy that though.
21:08.28tzangerthere's no fucking way this is gonna fit on that baord
21:08.29tzangerholy shit
21:09.01jpablopolycom phones works under asterisk, right ?
21:09.19denonyep
21:09.25fugitivoyes
21:09.35enderdoughecka_: hrm, I played a Counter-Strike team named 'OMG WTF BBQ?' a couple weekends ago.
21:10.46Hmmhesaysi'm kind of partial to "SlamDanglers" myself
21:11.19doughecka_lol
21:11.45doughecka_tzanger: whatcha makin
21:12.04enderTelco Softball team: DSlammers.
21:12.46spackleShaike 'n' baike.  And I helped.
21:15.16spackleSlamDanglers sounds painful.  Like America's (not) funniest home videos and such.
21:15.58spackleI'm on that like Oprah on a canned ham.wq
21:16.55NuggetMy 859 day uptime box died.  :(
21:17.25spackleNugget, what services did it run?
21:17.38Nuggetweb, dns, backup mx, ldap.
21:17.53Nuggetdunno what else. probably a few other things
21:18.03spackle<darth vader> Impressive </darth vader>
21:18.26*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
21:19.36fugitivoNugget: what died? the hd?
21:20.03Nuggetdunno.  it had page fault errors on the screen when they hooked a monitor to it, but after a power cycle it won't even post.
21:20.07Nuggetthe thing's just dead.
21:20.20NuggetI'm having someone swing by the colo facility to grab it and ship it to me
21:20.54spackleprobably a bug in the Power Supply.
21:21.07spackleIt's getting to be that time of year
21:21.57*** join/#asterisk santiago (n=santiago@63.245.87.180)
21:22.41NuggetThe machine was garage built crap, and it's been going strong since 1998.
21:22.49NuggetIt's well past time to die.  Hard to get too upset.
21:22.52*** part/#asterisk santiago (n=santiago@63.245.87.180)
21:23.33Assid<PROTECTED>
21:23.33AssidSep 15 02:53:08 WARNING[23173]: chan_iax2.c:9390 load_module: Unable to open IAX timing interface: No such device or address
21:23.33Assid<PROTECTED>
21:23.39fugitivodon't say that, i'm sure it can be saved
21:23.42*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
21:23.45Assidis that common?
21:23.52opus_Hey guys
21:24.04syzygyBSDhey opus
21:24.07opus_is this syntax correct: Dial(SIP/username:password@IPaddress/Extension) ???
21:24.30*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
21:24.41spacklenugget, do they know if it let any magic smoke out?
21:25.00Nuggetthe guy peeked inside and wiggled everything, said nothing was obviously wrong.
21:26.51*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:27.15enderIf I have both exten => 5729,1,Dial(something) and exten => _5XXX,1,Dial(somethingdifferent)   will 5279 if dialed do the something as opposed to the somethingdifferent eventhough _5XXX would catch 5729 ?
21:27.18opus_hmmm ?
21:27.29enderbasically, can I have exclusions to catchalls like _5XXX ?
21:28.16*** join/#asterisk bjohnson (n=bjohnson@i216-58-59-77.cybersurf.com)
21:28.40rene-How much does an asterisk tech makes per hour in the US? not at guru level but dCap worth
21:30.42gambolputtywhat is dCap?
21:30.44syzygyBSDopus_: I don't know for sure, that is the syntax for IAX it looks like but looking at the conf sip looks more like Dial(SIP/Extention@host)  but this could be way off, I dont know how the auth is done on this system
21:31.12syzygyBSDender: list the exclusions first in the dialplan
21:31.22file[laptop]Digium Certified Asterisk Professional
21:31.30bkw_Ya know what
21:31.34Assidhrmm.. i am just getting : Sep 15 02:59:16 ERROR[23214]: cdr_addon_mysql.c:201 mysql_log: Failed to insert into database.
21:31.39Assidshouldnt it mention whats the error?
21:31.47bkw_Cresl1n, that was such a slap in the face to Royk
21:31.49Assidi mean why cant it connect
21:32.02*** join/#asterisk NewSole (n=dave@d226-110-153.home.cgocable.net)
21:32.06endersyzygyBSD: thats what I figured, thanks.
21:32.11fugitivoAssid: check mysql logs
21:32.28syzygyBSDAssid: Isnt' there more then one line for that error message?
21:32.46syzygyBSDwhen i have a problem with pg it kicks out 5 or 6
21:33.16NewSoleanybody got a few min... need to pick someone brain about asterisk and math
21:33.18Assidnope
21:33.24Assidcdr mysql status
21:33.25AssidConnected to asterisk@localhost, port 3306 using table cdr for 7 minutes, 51 seconds.
21:33.25Assid<PROTECTED>
21:33.26fugitivoAssid: check mysql logs
21:34.25*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
21:34.37spacklebkw_ what channel are you thinking you are on?
21:35.46Assidwhere the hell is a readable log of mysql???
21:35.58WhiteWolfy/var/log/mysql.log
21:36.01WhiteWolfyprobably
21:37.46rene-me thinks newer mysql do binary logs per default
21:37.58rene-theris of course an utility to dump into text file
21:38.08pauldymaybe ya gota edit your /etc/my.conf file
21:38.13Assidyeah
21:38.18Assidthey do
21:40.06fugitivologs always give you the answer
21:40.37opus_arge.
21:40.48Assidhrmm..
21:40.53Assid.err file has some stuff
21:40.54fugitivoexcept microsoft logs, they doesn
21:40.58Assidbut.. nothing related to this
21:41.04fugitivothey doesn't tell anything
21:41.36spacklenow that's what I'm talkin' about: http://edition.cnn.com/2005/WORLD/europe/09/14/germany.catfuel.reut/index.html  Finally a good use for cats.
21:41.45FuriousGeorgewhy would my remote asterisk box say that chan_iax2.c:3910 register_verify: No registration for peer 'claudia'
21:41.46*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-209.cybersurf.com)
21:41.50*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
21:41.54FuriousGeorgewhen i try to log my other * box in as a peer
21:42.09FuriousGeorge[username] and secret are right
21:42.54*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
21:43.36fugitivoFuriousGeorge: do you have claudia in your iax.conf ?
21:43.56Cresl1nbkw_: yeah, well, it was a pretty annoying argument.  It kind of pisses me off when people act like that
21:44.15FuriousGeorgeyeah, i defined it as a peer at the bottom of the server *'s iax.conf, and as register => in the client's side
21:44.26bkw_Cresl1n, well you can't go around saying that.. you can't request money to have bugs fixed .... you can just say we aren't interested in fixing it
21:44.41bkw_asking for money for bug fixes outright is just not right in an Open Source project... if someone offers it.. take it
21:44.48bkw_but requesting it?
21:44.54Cresl1nfor a backport?
21:45.14bkw_while 1.0.x is billed as production ready you should be willing to fix bugs
21:45.18bkw_not charge to fix them
21:45.26Cresl1nthat probably would have taken a while and obviously nobody was interested in doing it for free
21:45.33spacklebkw_ who should fix them?
21:45.46Cresl1nhe just offered to do it if he was payed to do it
21:45.59Cresl1nmaybe the programmers out there who don't have to eat to live
21:46.23fugitivothere're no programmers who don't have to eat
21:46.25Cresl1nor somebody that gets excited about a big nasty backport
21:46.30bkw_haha
21:46.39Cresl1n(not me :-) )
21:47.05NuggetI dunno.  I'm still convinced that with enough coffee food can become optional.
21:47.07Cresl1nobviously there weren't a lot of others jumping up and down raising their hands to do it either
21:47.08spackleThere aren't enough "make it rock-solid" asterisk developers, everybody wants to add glamorous new features.
21:47.21mykewhere's my comfort noise generator?
21:47.29Cresl1nspackle: it was fixed already though
21:47.34Nugget#asterisk is a discomfort noise generator.
21:47.38spacklein general.
21:47.45Cresl1nspackle: just nobody wanted to backport it
21:47.48bkw_Cresl1n, well digium should be more than willing to fix something that is in the "stable" branch
21:47.54Cresl1nwhy?
21:48.03spackleexactly, why?
21:48.12bkw_because its software that people use.. and the software that sells cards
21:48.20MikeJ[Laptop]AST_FRAME_DNG
21:48.33spacklebkw_ but isn't that supposed to be Asterisk Business edition?
21:48.35MikeJ[Laptop]it's just a faint moan....
21:48.45Cresl1nspackle: exactly
21:49.03bkw_spackle yes and that causes me to question Digium's stand on the Open Source version
21:49.05fugitivowhat is asterisk business edition anyway? it has bug fixes or more functions?
21:49.15bkw_we were told ABE wouldn't differ
21:49.18Cresl1nbkw_: why, the fix was already in HEAD
21:49.30bkw_well lets get something to replace stable with :P
21:49.34spacklebkw_ you are right to question it, I don't think it is truly "open" although others will argue differently
21:49.49spackleasterisk 'snapshot' ;-)
21:50.06NuggetThere are plenty of large, successful open source projects which are able to manage the conflicting demands of a stable, production-ready branch and a developent, new feature sandbox branch.  It's unfortunate, but asterisk has thus far been unable to manage this sort of issue.
21:50.29spacklenugget, Asterisk doesn't seem to have that much traction yet.
21:50.36fugitivoasterisk business edition is like redhat, and asterisk like debian?
21:50.38bkw_Asterisk is one of the most mismanaged Open Source projects I have seen
21:50.40NuggetThat doesn't mean the situation isn't bad.
21:50.43Cresl1nnugget: You could be the one that starts it though :-)
21:50.53bkw_direction, focus and a roadmap would be most helpful
21:51.07Cresl1nnugget: that's the beauty of open source
21:51.16Cresl1nnugget: you can personally make a difference
21:51.16NuggetI'd argue that it's one of the big reasons why asterisk doesn't have that much traction yet.
21:51.17bkw_No its not.  Its the UGLY
21:51.20MikeJ[Laptop]asterisk does not have enough voluteers to handle the load, and not enough paid personell to handle the load required to even handle the working, submitted patches most of the time.. it's a problem of resources in every direction.
21:51.28FuriousGeorgefugitivo: i was spelling the user name wrong.  now its telling me the peer is not dynamic...  externally it does have a dynamic ip, but i didnt specify it as dynamic anyway
21:51.37spacklebkw_ agreed.  Even at cluecon when Mark had the opportunity to lay it out, at least in theory, he just tolaked about his challenges.
21:51.49bkw_MikeJ[Laptop], if they keep getting run off it will never have the number of people to handle the load.
21:51.59bkw_I have seen good developers get run off
21:52.01Juggieasterisk needs to be handed over to a self governing board, and not in 100% digium control....
21:52.10bkw_Juggie, haha FAT CHANCE
21:52.13Juggiejust like freebsd for example, has an elected team of leaders
21:52.21bkw_Juggie, that is a nice idea
21:52.31Juggiei know its a nice idea... but like you said, fat chance.
21:52.40Juggieits a monopoly
21:52.52Juggiei'd compare digium & asterisk to phone companies & dsl
21:52.57fugitivowhy fat chance? isn't it opensource?
21:53.13spackledoes anyone not employed by digium have commit rights?
21:53.21denonyes
21:53.22Juggiedigium is afraid to let it out of their total control, because they dont want "other" cards supported in the main tree
21:53.27fugitivoget the code, start another project with new leaders! muhahaaha
21:53.28MikeJ[Laptop]I have seen many good developers who are not willing to go through the hoops any more.  That being said, I have been assured that there is at least an effort to resolve the resource bottleneck at the commiting end at least
21:53.46Juggiesaganoma cards could be supported much easier with a chan_sanganoma, but it will never make it into cvs
21:54.00MikeJ[Laptop]hopefully that component of developers stopping contributions will help a but
21:54.02Cresl1nyeah, maybe we should make a tree that has NO quality control, that's a great idea <sarcasm>
21:54.16JuggieCresl1n, there needs to be a more expiremental thing
21:54.23Juggiewhere its ok to totally break things to try new ideas
21:54.25file[laptop]there's quality control and then there's an insane level
21:54.25Juggie*tree
21:54.32spackleCreslin - almost feel like we are already there.
21:54.40MikeJ[Laptop]spackle, yes, people not employed have commit rights
21:54.51MikeJ[Laptop]but are limited
21:54.55Juggiei'd like to see * run by a board of directors
21:55.00FuriousGeorgecan someone explain what this means:  http://pastebin.ca/22944
21:55.03Juggiesuch that, mark, and other digium people sat on the board
21:55.08Cresl1npeople whine about stable being stable, and now you guys are arguing about why we should make a SUPER unstable tree
21:55.09Juggiebut there were outside digium members as well
21:55.24file[laptop]FuriousGeorge: put host=dynamic for the peer entry
21:55.27MikeJ[Laptop]I am not saying we should make another tree at all.
21:55.27*** join/#asterisk dextro (n=Miranda@209.163.224.58)
21:55.38Juggiei would definitally like to see an expiremental tree
21:55.48Juggiewhere patches can go in without 2 months of hoops
21:55.51MikeJ[Laptop]that's easy enough to do juggie..
21:55.52Juggieand if things break, oh well they break
21:56.06FuriousGeorgefile[laptop]: i was thinking of doing it that but i thought it was saying that b/c it didnt like dynamic peers :)
21:56.22file[laptop]FuriousGeorge: it was saying that because you didn't have host=dynamic...
21:56.30FuriousGeorgei gotcha now
21:56.37Juggiemike, i'm going to take a look at that SQL patch tomorow btw.
21:56.43Juggienever had time today
21:57.43sivanawhat is GPL 2?
21:58.10obsidian-studiossivana: Generic Punk License 2 :)
21:58.19bkw_Juggie, dear check your private messages
21:58.22*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
21:58.22*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
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21:58.32FuriousGeorgeanyone know if nufone did's work with fwd?  the free partr
21:59.01obsidian-studiossivana: really is GPL version 2, http://www.gnu.org/copyleft/gpl.html
22:00.09obsidian-studios* has support for mysql and postgresql. Is anyone working on the ability to use Firebird?
22:03.27*** join/#asterisk Moc (n=mochouin@modemcable139.70-131-66.mc.videotron.ca)
22:04.31NuggetIsn't firebird a web browser?  ;)
22:06.06obsidian-studiosNugget: mozilla assholes, stole another open source project's name years later, started a huge legal battle, but no it's a RDBMS forked from open source Interbase v 6.0 http://www.firebirdsql.org/
22:06.26Nuggetof course.
22:06.47Nuggetbut thanks at least for linking to the firebird site and not the intolerable slashdot flamefests about it.
22:07.28obsidian-studiosNugget: as an open source DB, firebird is commonly over looked, it's only contender is postgresql, but fb 2.x should blow all other open source ones away. Much less Firebird's base has been is use for a very long time
22:07.41NuggetOf course.  I know all this.
22:07.47NuggetThat's why I made the *joke* up there.
22:08.00obsidian-studiosNugget: ok, sore subject 4 me ;)
22:08.15Nuggetno worries
22:08.48obsidian-studiosNugget: I am trying in small ways to get some of the core FB people to start making more of a presence, in mags, conventions and etc
22:09.30NuggetIt's doomed.  People are blissfully happy using mysql and getting what they deserve.
22:09.35obsidian-studiosI want to do cc processing with *, but I need db support for that, and do not want to port apps to 2 dbs
22:10.01obsidian-studiosNugget: sure, till they try to do more with mysql, & either can't or have to purchase a license, or licenses
22:10.29Nuggetopen source databases have as much a chance displacing mysql as linux has in displacing windows on the desktop.  It's just not happening.
22:10.35jarrodhey
22:10.54Nuggetit's crap, but it's familiar crap -- so people use it.
22:11.00jarrodis there a way to have FOP distinguish which Zap calls belong to a specific panel context?
22:11.03obsidian-studiosNugget: postgresql has always faired well against mysql
22:12.00obsidian-studiosNugget: are you a gold one or toilet one :)
22:12.00*** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net)
22:12.10FarrisGis IPSwitchBoard worth using?
22:12.16Nuggethrm?
22:12.21obsidian-studiosNugget: or a green nugget :)
22:12.27Nuggetchicken.
22:12.42obsidian-studiosNugget: real chicken like at arby's ;)
22:13.17fugitivoNugget: postgresql is a great database, needs people to stop using mysql and start using a real database
22:13.50*** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim)
22:13.58mswobsidian-studios: real chicken like at bojangles
22:14.00NuggetPeople are, for the most part, content eating crap.  Nothing to be done about it.
22:14.41obsidian-studiosmsw:  ah, good old jangles, can't go wrong, makes the winn dixie chicken I am eating taste like, well chicken :)
22:14.53mswsoooo tasty
22:14.57obsidian-studiosNugget: in both food and open source dbs
22:15.01Nuggetyup
22:15.43obsidian-studioswhich reminds me of one of the favorite passwords a person I knew used, OICU812
22:15.56obsidian-studiosoh, no it was OU812 :)
22:16.13NuggetOICU812 is an album by Van Halen.
22:16.43obsidian-studiosreally, and I thought they were being original, guess they were lamer than I thought
22:18.16NewSoleanybody got a few min... need to pick someone brain about asterisk and math
22:19.05filebleh
22:20.35NewSolein globals I am trying to inc and dec a count to active calls
22:20.39*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
22:21.06NewSoleit will add call count to global but will not execute command after dial even with 'g'
22:23.02Darwin35hey kram how much to have a proof read of a extensions.conf
22:24.45Darwin35now I need to add calling cards
22:24.50Darwin35this will be fun
22:25.28jarrodf5 seems to be the only load balancer that can properly keep persistence of sip/mgcp calls over udp and manage across multiple softswitches
22:27.53*** join/#asterisk asteriskmonkey (n=phil@67.71.178.116)
22:29.05*** join/#asterisk shermster (n=chatzill@rndf-146-51-71.telkomadsl.co.za)
22:29.15asteriskmonkeydoes anyone hear have an iaxy?
22:33.06enderon a Polycom 501 with 2 lines configured for one ext and one line configured for another, get a call on the other ext and transfer it to somebody using one of the other extension lines?
22:33.52asteriskmonkeyi alwasys seem to get echo when using a DIGIUM IAXY anyone else have one of those things?
22:34.17hardwireanything that converts analog to digital for telephones is bound to have to tackle echo
22:34.27hardwireI am finding
22:34.48hardwireI have to cancel echo for the endpoint themselves !
22:35.08asteriskmonkeyhardwire : how you do that with an ata like the iaxy?
22:35.17hardwirethe tdm cards
22:35.27asteriskmonkeyah your using like an x100p
22:35.37hardwiret1 cards
22:35.41asteriskmonkeywoo
22:35.45asteriskmonkeyi have a te110p
22:35.46asteriskmonkey:)
22:35.49hardwirewith your iaxy what are the outbound calls going over?
22:35.49asteriskmonkeyshow me how :)
22:36.02asteriskmonkeynet>asterisk>pstn(pri)
22:36.19*** join/#asterisk Wonka (i=produzie@madwifi/support/wonka)
22:36.30hardwireasteriskmonkey: I thought you grokked all this!?
22:36.40hardwireits engrained in your nic !
22:36.41hardwire:)
22:37.02hardwireso
22:37.06asteriskmonkeyhardwire: whats even wierder is i use my iaxy to call someone and the damn client comming from the pstn gets echo of there own voice on there side
22:37.07hardwireI ordered the 3com NJ220
22:37.13hardwireI am really really excited about getting it
22:37.13asteriskmonkey?...
22:37.18hardwiretalk about cool voipage stuff
22:37.21hardwireanyhoot
22:37.32hardwireasteriskmonkey: I get that
22:37.40hardwirehad to enable echo cancel :)
22:37.40fugitivoasteriskmonkey: using rxgain and txgain?
22:37.50asteriskmonkeyin the zapata.conf
22:37.55hardwirewhats weird is.. you should never have to enable rx/txgain for PRI
22:38.01asteriskmonkeyyes i know
22:38.02hardwirethat should be padded at the ilec imho
22:38.07asteriskmonkeyi tried no luck with that though
22:38.08fugitivoasteriskmonkey: set gain to 0
22:38.13hardwireon both
22:38.16asteriskmonkeyit is 0.0 on both
22:38.22hardwireand echocancel=128
22:38.27hardwireechocancelwhenbridged=yes
22:38.31hardwireechotraining=no
22:38.35hardwireI know..
22:38.37asteriskmonkeyno echotraining?
22:38.39fugitivo128 is too much
22:38.44fugitivotry echocancel=64
22:38.44hardwirenot in my case
22:38.46hardwirethat cleared it up
22:38.54asteriskmonkeyoh really?
22:38.56asteriskmonkeyok sec.
22:38.59fugitivoechotraining=800
22:38.59dougheckahardwire: true, but still need echo cancel on pri
22:39.03hardwireasteriskmonkey: one thing you should do
22:39.05hardwireis use record
22:39.14hardwireand load in the in-out files for the monitor
22:39.18hardwireuse monitor.. not record
22:39.25hardwireand start talking once the remote party answers
22:39.27hardwireits realyl freaky
22:39.33hardwirebecause you can see how terrible the echo is
22:40.22*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
22:40.32hardwirehttp://tdxnet.com/~hardwire/ACS_Echo/
22:40.39hardwirethese are the results I had
22:40.45hardwirehttp://tdxnet.com/~hardwire/ACS_Echo/Test-Analog-1.png
22:40.49hardwirethats an answering machine
22:41.18hardwirehttp://tdxnet.com/~hardwire/ACS_Echo/Test-Analog-2.png everything in the dark selection is me talking.. and the remote party doing nothing
22:41.21*** part/#asterisk zed_zed (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
22:41.42hardwirehttp://tdxnet.com/~hardwire/ACS_Echo/Test-Digital-2.png
22:41.47hardwirethats the exact same test
22:41.49hardwirebut to a cell phone
22:41.53hardwirenotice.. no echo
22:42.04hardwirenot even a drop
22:42.10hardwirethats with echocancel off completely
22:42.15*** join/#asterisk nick125 (n=nick@unaffiliated/nick125)
22:42.16hardwireso you have to do echocan for the remote party too
22:42.16asteriskmonkeyhey you have issue with cellphones not goign to voicemail ?
22:42.28hardwireasteriskmonkey: thats so not the issue at hand :)
22:43.12asteriskmonkeyhardwire: i know thats another issue i got to :P seems to hangup instead of goign to vm when calling a cell
22:44.38asteriskmonkeyecho cancel is a set to ye... i should change it to a value of 128?
22:44.40Rowterhow could I dial sending a digit, am trying dial(ZAP/1/23233232,D,1#) no luck
22:44.49hardwire128 is default
22:45.11asteriskmonkeyga.. but it says yes...
22:45.13asteriskmonkeymmm one sec
22:45.35fugitivoasteriskmonkey: i'd recommend a smaller number, but try with 128 first
22:45.53hardwireI went up from 16
22:45.58*** join/#asterisk Inv_arp (i=junya@adsl-3-255-139.mia.bellsouth.net)
22:46.12asteriskmonkeyhardwire : http://pastebin.ca/22946
22:46.15hardwireand I made it a point to test it against the NOC jerk who didn't believe me
22:46.20asteriskmonkeyfugitivo: http://pastebin.ca/22946
22:46.21*** part/#asterisk Cresl1n (n=matt@gateway.digium.com)
22:46.23hardwirethey should have never given me their number.
22:46.47asteriskmonkeyso it can take numberic and yes/no ?
22:46.56hardwireits so weird to see pri_cpe have to juggle echo issues
22:47.08hardwireasteriskmonkey: as per the docs :)
22:47.20asteriskmonkeycan you pastebin yours :)
22:47.24fugitivoasteriskmonkey: if that doesn't help, try echocancel=64 echotraining=800
22:47.30hardwireif I were you.. I would kill threeway and transfer.
22:47.34hardwireand call waiting
22:47.37asteriskmonkeywhy?
22:47.45hardwireyou probably don't need them
22:48.00hardwireand you definatly don't have a need to transfer outside of asterisk itself.
22:48.01obsidian-studiosI have an extensions range like _[100000-999999], but when I use the ${EXTEN} var all I get is 1? when I dialed 10000? any ideas?
22:48.25hardwireobsidian-studios: thats a bad extension mapping
22:48.35hardwirethat means it accepts 1 or 0.. and 9
22:48.42hardwirewell
22:48.49hardwire1.. 0 through 9
22:48.52*** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
22:48.52hardwireand thats it.. all single digit.
22:49.10hardwireinstead do _nxxxxx,
22:49.16FuriousGeorgethis may be a stupid question but does festival, or any sort of sound playback require a soundcard?
22:49.19obsidian-studioshmm, it allowed me to enter 100000 as the extension and did what I wanted from there
22:49.29fugitivoFuriousGeorge: no
22:49.30obsidian-studioshardwire: ok is that the proper way to map a range?
22:49.37hardwireFuriousGeorge: no.. it just samples the sounds to 8000hz mono and it gets played back via the channel driver
22:49.48hardwireobsidian-studios: yurop
22:49.49hardwireerr
22:49.50hardwireyup
22:49.54FuriousGeorgegreat thanks guys
22:50.03hardwireFuriousGeorge: just don't get angry.
22:50.07hardwirethats all I ask
22:50.08obsidian-studiosok, I was thinking wrong then
22:50.21fugitivoFuriousGeorge: also, try cepstral, it's cheap and much better than festival
22:50.25hardwireobsidian-studios: I was pissed when that didn't work for me either.
22:50.28hardwirethen I read the docs :)
22:50.39hardwirefugitivo: used sphynx?
22:50.58fugitivohardwire: no, it's installed but never tried to set it up
22:51.03hardwireok
22:51.05asteriskmonkeyman had anyone got sphynx to work yet?
22:51.05FuriousGeorgefugitivo: cepstral=commercial festival?
22:51.11mykethat's some old shit right there
22:51.13hardwirehah
22:51.17hardwireits not terrible
22:51.20fugitivoFuriousGeorge: cepstral is commercial
22:51.21*** join/#asterisk earthsound (n=webmonke@138.26.35.115)
22:51.30FuriousGeorgefugitivo: will look into it
22:51.33mykei did some volunteer testing on sphynx...in like 1988
22:51.43hardwiremyke: so you had to say a lot?
22:51.45FuriousGeorgefugitivo: was thinking of just using someone as a voice model
22:51.45fugitivohardwire: are you using sphinx?
22:51.54mykeyeah just talk to it a bunch
22:51.56hardwirefugitivo: there is an ADA requirement on one job.
22:52.02mykeback when it was still a navy project at CMU
22:52.04hardwireso I am tempted to start looking into it
22:52.19mykeif nothing else it's a mature codebase
22:52.38fugitivoFuriousGeorge: if you don't need dynamic audios, it's better a real voice
22:52.39hardwiremyke: hah
22:52.44hardwirewhat OS was it under at the time?
22:52.47*** join/#asterisk jimlinux (n=jim@69.49.168.58.swcp.com)
22:52.56hardwirefugitivo: what if you just don't like humans?
22:52.58hardwire:)
22:53.04fugitivohardwire: hire a dog
22:53.04hardwireprobably shouldn't be in the phone business.
22:53.05mykeprobably something horrible like AOS running on an IBM RT RISC workstation
22:53.12hardwiremy puppy can't speak
22:53.14asteriskmonkeyok
22:53.14hardwireits terrible
22:53.20asteriskmonkeyHardwire: are you in toronto
22:53.28fugitivohardwire: yes he can, you just can't understand what he says
22:53.31hardwireasteriskmonkey: was that a "buy you a beer" kinda question?
22:53.40asteriskmonkeylol for sure
22:53.43hardwirefugitivo: he moans like a wookie.. but never barks
22:53.44jimlinuxWhy does Cisco have to be so annoying about getting a firmware image for my phone?!?  I DON'T WANT an $80 service agreement just so I can test a phone.
22:53.52fugitivohardwire: how old is he?
22:53.59hardwireasteriskmonkey: no.. Alaska
22:54.02asteriskmonkeyanyone in toronto i can call so i can get an idea of echo ?
22:54.04*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
22:54.09fugitivojimlinux: when you get cisco, you get that
22:54.25jimlinuxfugitivo: It doesn't mean I have to like it  :D
22:54.33hardwirefugitivo: 2 years
22:54.42hardwirehe barks at big scary things.. but never us..
22:54.43fugitivojimlinux: sure :)
22:54.43mykejimlinux, i had that problem, xyplex wanted $400 for the firmware for the term concentrator i got for $80 on ebay, the next day some guy on a mailing list boosted it to me for free
22:54.52hardwirewe tried to teach him to sing.. cause thats what he does
22:54.53jskcrhy all
22:54.59hardwirehe doesn't understand we want him to make funny noises.
22:55.22jimlinuxmyke: lucky bastard
22:55.25hardwireso we have no idea how to train him to speak/sing/woof/meow
22:55.25jimlinux;)
22:55.32asteriskmonkeydamn
22:55.39fugitivohardwire: mine sings when you play air instruments
22:55.45hardwireasteriskmonkey: call your house :)
22:55.55hardwirefugitivo: he rocks down to air guitar?
22:55.59hardwireor do you mean wind instruments :()
22:56.05fugitivolol
22:56.08fugitivowind instruments :)
22:56.13hardwirecool
22:56.22hardwirewould have been neat if a dog knew what air guitar was
22:56.35mykejim, no, sorry
22:57.01Kattyhi lads.
22:57.04jimlinuxmyke: thx anyway
22:57.12asteriskmonkeytiny minor echo
22:57.13fugitivohardwire: heh
22:57.35fugitivoasteriskmonkey: now, try with lower echocancel
22:57.41asteriskmonkeylike 32?
22:57.44fugitivoasteriskmonkey: 64
22:57.54asteriskmonkeyis was at 64
22:58.02fugitivoechotraining=800 ?
22:58.02hardwireasteriskmonkey: get recording working
22:58.05Kattysad. no one said hi.
22:58.09fugitivohi katty
22:58.10jarrodanyone deployed redundant SER boxes?
22:58.12hardwireit will allow you to see the delay and adjust
22:58.15asteriskmonkeyechotrainig=no
22:58.20hardwireits very insightfull when training echo manually
22:58.25Kattyfugitivo: ((=
22:58.40asteriskmonkeyhardwire: what do i have to mash in for the monitor?
22:58.40hardwireok
22:58.45hardwireg/f told me she needs booze
22:58.47hardwirethis can't be good
22:58.53filea properly configured SER with a stable build should never go down
22:59.02hardwireasteriskmonkey: before a dialout add a monitor command
22:59.03filethe only time I've ever crashed it was out of my own stupidity
22:59.16hardwirepost your dialout stuff to pastebin
22:59.21hardwireextensions
22:59.35hardwireso
22:59.44hardwireanybody here ever heard of mummified evergreens
22:59.49asteriskmonkeyhardwire: me post my extension?
23:00.01hardwireasteriskmonkey: your extension for dialing out locally
23:00.08hardwirefrom extensions.conf
23:00.18fugitivohardwire: www.airguitar.com
23:00.58asteriskmonkeyhttp://pastebin.ca/22947
23:01.13*** join/#asterisk Romik (n=romik_@84.77.130.188)
23:01.19*** join/#asterisk SwK (i=momfwt@12-219-144-126.client.mchsi.com)
23:01.44*** join/#asterisk znoG (n=gs@200.115.218.81)
23:02.27hardwireasteriskmonkey: gimme a sec
23:02.34asteriskmonkeyhardware: k thanks
23:02.51*** join/#asterisk kpettit (n=keith@69.15.174.114)
23:03.30kpettitI'm using asterisk 1.1 and I want to get spandsk with rxfax and txfax installed
23:03.30hardwireasteriskmonkey: http://pastebin.ca/22948
23:03.32hardwireok
23:03.33hardwiretada
23:03.50hardwirenow in /var/spool/asterisk/monitor there will be two files made when you dial out
23:03.58hardwirerecording-in.wav and recording-out.wav
23:04.11obsidian-studiosjimlinux: you are not the only one suffering the Cisco firmware issues, just wait till you get an EOL product, and can't get a smartnet contract for it. Getting firmware is even harder
23:04.17hardwireso..
23:04.20kpettitspandsp-0.0.3 dosent even include the rx and tx files and the older 0.0.1 version dosen't compile with the newerasterisk.  Any ideas?
23:04.37hardwireasteriskmonkey: now if you look closely at the two files closeley in say.. audacity
23:04.43hardwireloading up both files into a stereo wav.
23:04.51hardwireyou will see the offset of the echo
23:04.58asteriskmonkeyhardwire sec.
23:05.01jimlinuxobsidian-studios: I'm surprised there's not a larger effort to reverse engineer Cisco firmware
23:05.02hardwireaccording to the zaptel driver atleast
23:05.06hardwireit takes a while to get used to doing this
23:05.14hardwirebut thusly classifies you as an echo master.
23:05.28jimlinuxobsidian-studios: It's a great business model for forcing new hardware purchases though
23:05.49obsidian-studiosjimlinux: that's not the type of talk that's going to open up Cisco, I bet because of the reverse engineering aspect they are the way they are about firmware
23:05.52*** join/#asterisk paulc (n=Paul@216.187.75.190.novuscom.net)
23:06.04*** join/#asterisk mmlj4 (n=jkelly@redfishnetworks.com)
23:06.13FuriousGeorgemy asterisk console is no longer verbose when a call comes in on an fxo.  did i change something?
23:06.13asteriskmonkeyso i just call someone and it recods it?
23:06.38obsidian-studiosjimlinux: yes when they have new products to purchase, but they have EOL all cable routers? They keep the 827-4v around because of it's fxs ports when hardly any of their other soho devices have fxs ports
23:06.40jimlinuxobsidian-studios: but it's exactly that type of attitude from Cisco that forces those who are unsupported to find a way to make their hardware work
23:07.31paulcSupposing you were really stupid.. and copied something over top of extensions.conf.. you've got it in memory, but can't "save dialplan" cos "writeprotect=yes" was in the original file. You can do "show dialplan" but you don't see the global variable definitions.. any way to write the file? or is it a case of manually adding an extension and NoOp()'ing the variables you need?
23:07.32jimlinuxobsidian-studios: There's nothing illegal or wrong with reverse engineering.  As long as the engineering efforts are 100% home-grown
23:07.36hardwireasteriskmonkey: only one way to find out :)
23:07.40hardwirebrb
23:07.41mykeugh
23:07.48mykewho said AMP made * config easier?
23:07.51obsidian-studios<PROTECTED>
23:07.51asteriskmonkeyhardwire: ok called someoen :P
23:08.16obsidian-studiosjimlinux: yes, but it's the reverse aspect I bet they want to prevent or limit
23:08.22fugitivopaulc: a "friend" copy over his extensions.conf?
23:08.30mykereverse engineering is mostly illegal under the DMCA
23:08.30jimlinuxobsidian-studios: I'd agree that finding a good Open Source business model isn't the easiest thing to do
23:08.49paulcfugitivo: no - it was me - I'll fess up :-)
23:09.00paulccopied sip.conf from one location over top of extensions.conf in another :-s
23:09.00*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-18-197.cybersurf.com)
23:09.14jimlinuxmyke: not if it's 100% home-grown, and without the intent of disrupting their business model
23:09.28jimlinuxmyke: so far, rev-eng on hardware has stood up
23:09.31FuriousGeorgefugitivo: did you ever try the mbrola voice with festival?  the default one is pretty terrible
23:09.34jimlinuxmyke: look at the X-Box hacks
23:09.39obsidian-studiospaulc: oh yeah, well I rm -fR * in my /etc/asterisk dir of a new install I was half way into configuring, I thought I was in the log dir :)
23:09.58jimlinuxmyke: they are legal except where MS code is copied over...
23:10.04jimlinuxmyke: it's called fair use
23:10.06fugitivoFuriousGeorge: no, i use cepstral in spanish :)
23:10.13paulcobsidian: ok now I feel better - that's a bigger hassle than what I did!
23:10.20FuriousGeorgegotcha
23:10.25paulcI'm guessing there's no way to override the "writeprotect=yes" right?
23:10.33paulcor "save dialplan" to a different name?
23:11.12asteriskmonkeyis there linuix proggi for merging the to monitor files?
23:11.12fugitivoonce i did crontab -r, damn flag, R is at the side of E
23:11.21fugitivoit was a crontab of 100 lines
23:11.44*** join/#asterisk BoyGenius (n=ldvoipen@ip68-108-88-208.lv.lv.cox.net)
23:11.45paulcasteriskmonkey: soxmix
23:11.50fugitivoand it wasn't mine, so i couldn't write it again
23:11.53paulcfugitivo: DOH! what a mistaka to maka
23:11.58BoyGeniushello all
23:12.12fugitivopaulc: hell yes
23:12.16paulcBoyGenius! Here to save the day?!
23:12.17obsidian-studiosfugitivo: well my record move was in my first month on a Linux box, a colo cobalt server, I chowned /, can't remember what users, but it hosed the system :)
23:12.18jimlinuxSo for now, I'll try and use my 7940 with SCCP until my SLA comes in
23:12.29fugitivoobsidian-studios: LOL
23:12.33asteriskmonkeyso ... soxmix recording-in.wav recording-out.wav out.wav?
23:13.10obsidian-studiosfugitivo: suffice to say in the first few months that cobalt had to be reimaged a few times :)
23:13.13paulcyeah - somethign like that.. the wiki has an example of how to do it from the dialplan if you don't want to do it manually.. including panning the input/output to left/right channels so you can still hear them separately if need be
23:13.25*** join/#asterisk santiago (n=santiago@63.245.86.203)
23:13.28BoyGeniusI dunno about that, but I'll certainly try
23:13.59BoyGeniushow are you paulc?
23:14.03bkw_I AM I AM
23:14.09bkw_how ya doing paulseeeeeeeeee
23:14.15paulcBoyGenius: It's doubtful.. I was clutching at straws.. (I overwrote my extensions.conf and am trying to reconstruct from a "show dialplan" together with an old version of the file)
23:14.19paulcbkw_! I AM I AM!
23:14.23paulcI'm good.. and you?
23:14.24bkw_yes you are you are
23:14.30bkw_great grea grea
23:14.37BoyGeniusouch, sounds like my lost backups
23:14.41hardwireugh
23:14.46hardwireI hate community radio sometimes
23:14.54hardwireesp when they play the happy b-day song in all meows.
23:14.54BoyGeniusBeen a very long day chasing troubles on an old NEC switch
23:15.15obsidian-studiosBoyGenius: did you catch it?
23:15.24BoyGeniusLOL, and killed it dead on the spot
23:15.36hardwireheh
23:16.00obsidian-studiosBoyGenius: if you were in la you could have made the news, they love chases there :)
23:16.20BoyGeniusNo, it is an old NEAX switch, with DRU remote units, one or two of them are very flaky, and couplke that with a campus network > three miles
23:16.28BoyGeniushahaha
23:16.35BoyGeniusnope been there done that, heted it
23:16.45BoyGeniusno cali for me, I hate vegas as it is
23:16.58BoyGeniusgive me a house in the foothills, away from lots of people
23:17.19BoyGeniusonly good thing about vegas, it alows me to beat the house at the nfl
23:17.22Rowterhow could I dial sending a digit, am trying dial(ZAP/1/23233232,D,1#) no luck mmh any clues?
23:17.38BoyGeniuslike this week, san fran 49ers +13 pts!!!  that's unheard of
23:17.53BoyGeniusthey are going to kill the eagles
23:17.54jpabloroamer323: did you RTFM?
23:18.05obsidian-studiosBoyGenius: vegas is cool, but the traffic and growth, OMG, but I always drop tax brackets when I go to vegas
23:18.14asteriskmonkeynow i have this was file what do i use to determine what to tweak?
23:18.44BoyGeniusyes vegas has no state tax, but it is expensive, and no one wants to pay you anything, especially as a technical person
23:18.47jpabloRowter: did you RTFM ? http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
23:19.33obsidian-studiosBoyGenius: I meant federal tax bracket, as in loosing allot of $ :)
23:19.42BoyGeniusagain, the only way I survive is by going to school full time and drawing va bennifits
23:19.55BoyGeniusLOL, ok, again, been a very long day
23:20.04Rowterjpablo, had you ever send digits after dialing? I had read that, been trying diferent forms.. you know?
23:20.45*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
23:20.45*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
23:20.47BoyGeniushere is a quick question for all, I just upgraded my cpu, but it is the same arch, should I recompile?
23:21.04drumkilla_laptopno need
23:21.06jimlinuxBoyGenius: I wouldn't
23:21.12BoyGeniusthank you, I didn;t think so
23:21.24BoyGeniusbut I am not the most linux savy(yet)
23:21.27xhelioxHow can I turn on console debugging for just chan_zap?
23:21.52Drukenzap debug on ?
23:22.05Drukener... no just zap debug
23:22.32xhelioxzap debug doesn't work
23:23.35hardwireblah
23:24.34*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
23:24.38malverian[work]Hmmm..
23:24.56*** part/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net)
23:24.57malverian[work]What's the best way to play a tone to the CALLED party when they answer the phone?
23:29.09*** join/#asterisk Rez (i=lorez@freenode/staff/lorez)
23:30.12fugitivowhat kind of tone?
23:30.21malverian[work]It can be a gsm file.. doesn't matter.
23:30.22hardwiremalverian[work]: using dial
23:30.29hardwireI play a beep at people I intercom
23:30.32malverian[work]I'm currently using the Dial(||D(5))
23:30.42malverian[work]But it's not long or distinct enough..
23:30.48fugitivosendDTMF
23:30.55malverian[work]hardwire, That's what I'm using it for.. but a dtmf tone isn't really good enough :-/
23:30.56*** join/#asterisk pablix (n=root@r200-125-3-189-dialup.adsl.anteldata.net.uy)
23:30.58hardwire'A(x)' -- play an announcement to the called party, using x as file
23:31.06malverian[work]hardwire, Ooooh.. awesome thanks :0
23:31.08hardwireso I use a(beep)
23:31.10hardwireerr
23:31.12hardwireA(beep)
23:31.19asteriskmonkeyhardwire what do i use to analys the wav?
23:31.21hardwireI made beepbeepbeep.gsm
23:31.26hardwireasteriskmonkey: anything
23:31.31hardwireI use audacity under debian
23:31.44malverian[work]hardwire, That waits until the party answers?
23:31.54hardwirefugitivo: you can send DTMF to the called party using that?
23:31.54pablixanyone can helpme. i have a box connect to PBX and the lines analog in pbx get by number 9.   can anobody pastebin the sintax in extensions.conf to do that with my x-ten soft?
23:32.03hardwireI have got to see your dialplan entries.
23:32.14Drizzt321Ariel_: you around?
23:32.29Ariel_yes but eating dinner
23:32.40*** join/#asterisk iswm (i=iswm@unaffiliated/iswm)
23:32.47Drizzt321Ariel_: ok, cool. I'm around, whenever you have a chance to help me out, let me know :)
23:33.16Ariel_ok
23:33.50enderis there any way to restrict the protocols allowed when using zap channels?
23:33.53pablixanyone can helpme. i have a box connect to PBX and the lines analog in pbx get by number 9.   can anobody pastebin the sintax in extensions.conf to do that with my x-ten soft?
23:35.32pablix??????
23:35.39pablixnobody?
23:36.08enderpablix: exten => _NXXNXXXXXX,1,Dial(Zap/whatever/9w${EXTEN})
23:36.32enderpablix: puts a 9 and then a wait before sending the 10 digits to the zap line.  I assume you're connected via zap to your PBX?
23:36.40hardwirew is so cool
23:36.46hardwireI had no idea w existed until the ther day
23:36.52pablixyes i will try that thanks ender
23:36.53hardwireit makes me happy
23:36.54hardwirew
23:36.59enderlol
23:37.32pablixin what context i need to insert this line?
23:37.40hardwireyour outgoing one :)
23:38.18asteriskmonkeyso if its a minor echo that comes and goes now hardwire which setting do i tweak? if i have echotraining=no and echocancel-64?
23:38.43hardwireechocancel.. make sure echocancelwhenbridges=yes
23:39.40asteriskmonkeyhardwire its set to yes already
23:39.47hardwireok
23:39.49jimlinuxSo, what are the chances of someone sending me the Cisco 7940 6.3 and 7.x firmware images?
23:39.56hardwirethen you adjust echocancel=
23:40.09asteriskmonkeyits at 64 should i make it go up or down
23:40.29hardwireasteriskmonkey: this is where I have to say.. record them at different levels
23:40.34hardwireand visually inspect the difference
23:40.36hardwireits hard work
23:40.37asteriskmonkeyi ask becuase im running out of random ppl to call hehehe
23:40.39hardwireyou ready for that :)
23:40.55obsidian-studiosjimlinux: if you got a cisco rep, beg :)
23:41.30*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
23:41.31asteriskmonkeyso i can leave echotraining=no right ?
23:41.38hardwireyeeeeees
23:41.43jimlinuxobsidian-studios: I'd love to.  I just hate waiting.
23:42.16obsidian-studiosjimlinux: I know, I had the same issues with a 7960, I am going to keep my own repository of firmware, but I do not think it wise to share or pass it along, or I would if I had it
23:42.24jimlinuxobsidian-studios: I broke down and ordered the 1-year support for the 7940 from CDW, but now it looks like it'll take another week
23:42.43jimlinuxobsidian-studios: I hear you
23:42.44*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
23:42.54obsidian-studiosjimlinux: yes I am advising clients, however I am saying get 1 smart net per phone model, instead of one per phone
23:43.28FuriousGeorgedoes festival not like playing on zap channels?  check out the one on top, it doesnt work where the one on bottom does
23:43.39FuriousGeorgehttp://pastebin.ca/22950
23:43.47*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
23:43.50obsidian-studiosjimlinux: that or I will buy one for each phone model, and keep the firmware on hand for when I need it
23:46.10BoyGeniussomeone needs cisco firmware???
23:46.20asteriskmonkeywhat you guys find better iax of sip
23:46.38obsidian-studiosBoyGenius: yes jimlinux: needs 7940 fm
23:46.40BoyGeniusor ios software?
23:47.04obsidian-studiosBoyGenius: the phones do not seem to run IOS?
23:47.12BoyGeniuswell no they dont
23:47.16SwK*yawn*
23:48.19BoyGeniussip image or h323?
23:50.45BoyGeniusI have 7940 rel 7.2.3
23:50.53BoyGeniuslatest I know of
23:51.09BoyGeniussip
23:52.53hardwireman
23:52.55hardwirespeakign of image
23:53.00hardwireanybody used SendImage ?
23:53.06hardwirewhats that all about?
23:53.13hardwirecause I would totally have fun using that
23:53.30BoyGeniuswhat is it?
23:53.41opus_Arge! make zaptel goes into an infinite loop
23:54.12hardwirehaha
23:54.45harryvviax.cc was doing some wierd things today.
23:54.46harryvv:)
23:54.57harryvvCannot wait to drain that account and find another.
23:56.18*** join/#asterisk eris (n=jburnes@orthanc.estreet.com)
23:56.43hardwireharryvv: just use it as backup
23:57.07erishelp: can anyone help me with asterisk firewall routing?
23:58.51erisi'm trying to efficiently use my public IP space and don't mind assigning the asterisk box it's own IP
23:59.04znoGguys, is the quality of a g726-16 any good?
23:59.27erisproblem is I'd rather not use a separate subnet since that burns 4 IPs at once
23:59.39erisis 1::1 natting just  as good?
23:59.40BoyGeniusg729 is better
23:59.58harryvvhardwire voipjet is the fail over service.

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