irclog2html for #asterisk on 20050726

00:00.09Kattytwisted: heh
00:00.37Kattyhttp://www.penny-arcade.com/docs/nes_4.jpg <- file's collection
00:00.45twistedROFLMAO
00:01.04twistedI used to have a normal NES back in the day
00:01.06twistedthen i modified it
00:01.19twistedmade it top loading and tried to build my own rf controller
00:01.19Qwelltwisted: mine runs asterisk
00:01.20syle2my NES runs on my xbox now lol
00:01.25Delta34gambolputty: can u post sip.conf
00:01.30gambolputtyhold on
00:01.43twistedof course, trying to build RF devices when you're 12 doesn't work out too well
00:01.45Kattyi have an original nes still
00:01.55Kattywith zelda and rampage and tetris
00:01.58twistedzelda
00:02.01twistedi LOVE the original zelda
00:02.02Kattyoh and the star wars game
00:02.10Kattyi still play it every year :>
00:02.19twistedoooh
00:02.21syle2super mario bros 3, mike tyson;s punchout
00:02.33twistedif I ever come visit your area, I'm going to make you let me play it :P
00:02.41Kattyk
00:03.19twistedNES was the first and only console gaming system I ever owned
00:03.29twistedbesides the xbox, but that never played games, that ran linux from the first day :P
00:03.34syle2my xbox is suped up with NES, SNES, playstation, sega etc games and emulators, and tons of xbox games, what i find is when people come over , all the guys want to play latest xbox games and the girls want to play mario brothers or tetris on NES hehe
00:03.46opus_whoah he even has the control pad with the turbo button
00:04.05twistedopus_, not only that, but the quad extender
00:04.13gambolputtyhttp://pastebin.com/320659
00:04.19opus_whaoh
00:04.23twistedopus_, AND the droid!
00:04.27opus_yeah the droid
00:04.30twistedI SOOO wanted that droid
00:04.42opus_that was like only the first 3 months when nintendo came out
00:04.49opus_or somthing
00:04.54twistedit was a particular game
00:04.57twistedbut yeah, they were hard to come by
00:05.00twistedso they were expensive
00:05.04opus_whoah, two droids
00:05.14twistedomfg, you're right
00:05.14opus_i think they were free when it came out
00:05.30*** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca)
00:05.35twistedi need to find one
00:05.38twistedi'll make it work under linux
00:05.42opus_hahaha
00:05.50opus_make it a DVD repliactor
00:05.54twistedhahaha
00:05.55twistednah
00:05.57twistedtape autoloader
00:06.15*** join/#asterisk trig_hm (~jb@home.monkeypr0n.org)
00:06.17DeeJayTwowe have an IAX connection between 2 asterisk system... when people are talking thru it, it happens that a person doesn't hear the other while 30 seconds while the other can hear..
00:06.17*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
00:06.20Qwellno...make it an ATA
00:06.22DeeJayTwois it a known issue?
00:06.40fileI fear I don't play games
00:06.53jsmithDeeJayTwo: Could be any number of issues... but my best guess is a codec problem.
00:07.02DeeJayTwo...hmm.. it is possible..
00:07.06DeeJayTwothere's a codec conversion..
00:07.07jsmithDeeJayTwo: Make sure you're using the same codec on both sides.
00:07.22DeeJayTwonice... that's what I wanted to hear... that's the only thing we've not tried yet ;)
00:07.47jsmithDeeJayTwo: It's a total guess, but it sounds like the right problem :-)
00:08.10DeeJayTwothanks a lot ;)
00:08.34twistedhahaha
00:08.36twistedOH YES
00:08.40twistedthis SHALL be my new ringtone: http://www.scott-o-rama.com/Audio/Mahna%20Mahna.mp3
00:10.15jsmithtwisted: Yeah, figure out how to convert that into a Nokia ring-tone, will you please?
00:10.25twistedjsmith, no way man
00:10.32twistedmy phone will play mp3's stock
00:10.48jsmithtwisted: Then you're just that much cooler than I am :-)
00:10.52twistedjsmith, hehe
00:11.11Katty:<
00:11.20twisteder
00:11.27Katty:>
00:11.27twisteddamn tab completion
00:12.09niZonno asterisk PHP users around?
00:12.23Kattyk, all better
00:12.29syle2is it illegal to have music on hold at your own house with copyrighted material?
00:12.31jsmithniZon: I use PHP and Asterisk
00:12.39jsmithsyle2: Yes.
00:12.44fileKatty: you're silly
00:12.45opus_syle2 not if your server is offshore
00:12.49syle2even if you bought the cd
00:12.50Kattyfile: you started it
00:12.54filetrue
00:12.57jsmithsyle2: Even if you bought the CD.
00:13.06syle2blah
00:13.11DeeJayTwoyou must buy the band
00:13.15DeeJayTwoand the disc company
00:13.17DeeJayTwo=)
00:13.32opus_you need a licence for each speaker:)
00:13.34Sedoroxsyle2: so yes.. it is illegal
00:13.58Kattyfile: do you tone your forearms?
00:14.01niZonjsmith: does it work nicely?
00:14.01twistedsyle2, you can pay a small fee to the recording labels to do it, IIRC
00:14.03opus_how come I hear people with radio all the time
00:14.31fileKatty: nope
00:14.33filewell
00:14.34Kattyk
00:14.35syle2yeah , yet radio is fine?
00:14.38jsmithniZon: Yes.
00:14.39filethey're, toned
00:14.41Kattyhot
00:14.42twistedfile, wrong kinda tone
00:14.43jsmithsyle2: http://www.copyright.gov/carp/
00:14.45filedarn
00:14.47jsmithsyle2: If you're in the US.
00:14.55jsmithsyle2: Radio is technically illegal too.
00:14.55Kattyi once got toner smudged on my cheek
00:14.56gordonjcpsyle2: depends on the copyright agreement
00:14.59Kattyit took forever to get it off
00:15.01opus_if your SERVER is not in the US you're fine:)
00:15.03Darwin35any word on if e911 is going to be fought ?
00:15.03jsmithsyle2: Unless you pay the royalites
00:15.04twistedKatty, toner is EVIL
00:15.07AyanoThe radio has permision to play the songs.
00:15.08gordonjcpKatty: don't use hot water...
00:15.09Kattytwisted: tis :<
00:15.20syle2well its in canada so i highly doubt US laws don;t apply
00:15.24twistedbut it's lotsa fun to use for office pranks ;)
00:15.44opus_us owns the planet and you must ahve a license
00:15.46gordonjcpsyle2: I have copyrighted music, but it's perfectly OK to use for music on hold
00:16.00twistedKatty, haha, i wouldn't do that to you
00:16.00Kattyeventually, i used tape to get it off me
00:16.12*** join/#asterisk fluidicsl (~asdf@adsl-63-200-54-51.dsl.snfc21.pacbell.net)
00:16.12Kattythe clear scotch tape stuff
00:16.13gordonjcpheh
00:16.22gordonjcpever seen a cat get sticky tap on it's paw?
00:16.27fluidicslI am haveing a probelm getting asterisk to start all of a sudden
00:16.28twisted3M makes the best adhesives
00:16.33Kattygordonjcp: indeed
00:16.36gordonjcps/tap/tape/
00:16.42twistedgordonjcp, dude, that's some funny stuff
00:16.46gordonjcpman
00:16.51opus_what was it, egg yoke and viniger
00:16.51Kattyhis typing?
00:16.55Kattyor the cat and the tape? :P
00:17.07twistedKatty, cat + tape
00:17.09gordonjcpwatching my cat stand on a bit of discarded gaffa tape, and bounce around trying to get it off his foot...
00:17.11Katty:P
00:17.22gordonjcp... until he discovered he could stick things with it and be really annoying
00:17.38twistedgaffers tape is good stuff
00:17.54gordonjcplike, stick it to a sheet of newspaper, pull the sheet across the room, stand on it with his other front paw to pull it off the tape, repeat
00:18.24twistedgordonjcp, we went through an entire box of gaffers tape at the rolling stones show in nashville
00:18.38gordonjcptwisted: it's good stuff
00:18.44twistedyes indeed
00:19.18*** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET)
00:19.24*** join/#asterisk generalhan (general_ha@63.133.146.82)
00:19.31generalhanwhats going on everyone
00:20.17generalhananyone have a second for a question ?
00:20.54Nuggetyes.
00:20.56Nuggettime's up.  :)
00:21.03generalhanlol
00:21.25Sedoroxlol
00:21.36*** join/#asterisk cpatry (~junky@Toronto-HSE-ppp3780869.sympatico.ca)
00:21.37*** join/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
00:21.57generalhanmy office is currently using a cisco based VoIP system that is hosted by our ISP, who i REALLY hate. some one told me about Asterisk and it seams like an awsome solution for me but im not that familiar with Linux, am i going to be able to set this up for our office? or is it extremely difficult ?
00:22.27generalhanor are you guys just so great that you could talk me through it ! LOL
00:22.45opus_generalhan - its not that difficult if you know linux. If you don't, try Asterisk@home iso.  other people write commerical based web interfaces
00:22.52Sedoroxsorry.. that goes $1/character
00:22.52Sedorox:p
00:23.06jake1932i would recommend http://www.xorcom.com/
00:23.13generalhanwow ! i could have paid for my entire cisco system by now !
00:23.18opus_generalhan - the hard part will be redoing the cisco licensing
00:23.46generalhanbah licensing?? LOL !!!
00:23.52Sedoroxlol
00:24.02opus_cisco is terrible
00:24.04Sedoroxyup.. if you use cisco phones... technically each phone has to be licensed
00:24.12Nuggetasterisk is not difficult, but it can be opaque.  all it takes is patience to plod through some times when it doesn't work well and tolerance for googling and reading.
00:24.40Nuggetit won't work right away, it won't work well right away, and the docs are not centralized or even particularly easy to find
00:24.48opus_generalhan - you can hire a guy, post on asterisk-biz asking for a consultant or check voip-info.  craiglist is also a good source to find a consultant
00:24.49generalhanwell i would have to do all that reading before hand ... i work for a law firm and down time is unexceptable
00:25.10Nuggetand there are parts of asterisk that will always be sort of flaky -- at least for the forseeable future
00:25.10opus_where are you located?
00:25.16jsmithgeneralhan: www.oreilly.com/catalog/asterisk will help, in a month or so
00:25.19generalhanScottsdale, AZ
00:25.22jake1932you canmsg shido6 - he helped me get started and was reasonable on ratesw
00:25.37opus_jsmith whats that about
00:26.15jsmithopus_: An asterisk book, written by the Asterisk Documentation Project.
00:26.32opus_nice. what version of asterisk will it cover? just 1.0?
00:26.40QbYhttp://pastebin.ca/18498 -- could someone take a look and tell me what i'm doing wrong..
00:26.56jsmithopus_: It covers a few 1.2 features, but not everything, as we had to write it before all the 1.2 features were in there :-)
00:27.04opus_hehe
00:27.07jsmithopus_: I'm sure the second edition will cover 1.2
00:27.20opus_haha its not even out yet and you're saying second edition, cool.
00:27.44jsmithWell, that's because they're asking us to start working on the second edition.
00:28.04*** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
00:28.17file[laptop]KITRICH!
00:28.22syle2anyone running voip company in here?
00:28.27SarahEmmFILE!
00:28.46generalhanwell heres the deal ... there is another company that is currently using this software, and they have agreed to give us what they have already for me to mess with for our own use. but the problem with that is giving them access to our systems. given my lack of knowledge with linux, my boss is VERY concerned with data that we currently have on the system
00:29.01*** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-144-219.buff.east.verizon.net)
00:29.04file[laptop]syle2: tons of people probably
00:29.09SuPrSluGhi all
00:29.17file[laptop]SarahEmm: how are you?
00:29.40SuPrSluGis there a default number of messages you can leave with asterisk?
00:29.51jsmithSuPrSluG: I think it's 100.
00:30.04SuPrSluGhow can you reset that
00:30.09jsmithSuPrSluG: You can change the code and recompile.
00:30.10file[laptop]it's hardcoded in 'da source
00:30.24SuPrSluGoh ok
00:30.30file[laptop]jsmith: 'tsk 'tsk
00:30.31SarahEmmfile: sucky! but i just geeked out for a bit and moved myself to 5GHz (802.11a) so that's good
00:30.46file[laptop]I like my 802.11g thank you
00:31.17SarahEmmheh
00:31.26SarahEmm26 other access points in 2.4GHz visible from here
00:31.34SarahEmmit was getting unreliable and slow with so much crowding
00:31.37SarahEmmi'm the only 802.11a visible :)
00:31.54file[laptop]cool
00:32.27SedoroxHmmm
00:33.32pfnhmm, wasn't mpg123 available from asterisk cvs?
00:33.39pfnI coulda sworn it was there
00:33.43SuPrSluGthanx found it. it
00:34.03file[laptop]make mpg123 may magically download and compile mpg123 (the correct version of course)
00:34.05Nuggetit's a makefile tag
00:34.06SuPrSluGit's in the vmai.c correct?
00:34.56pfnah, it's a make target
00:34.58pfnI guess that'll do
00:38.34harryvvnugget, disabeling the call forwarding still has not resolved my issue.
00:38.39*** join/#asterisk wulfy814 (~lorentz@c-67-165-37-20.hsd1.pa.comcast.net)
00:39.04pfnmeh, I didn't save my unixodbc config when upgrading my machine
00:39.14wulfy814ok - have to put up a production system tomorrow for a small company whose Lucent Partner System died today
00:39.27wulfy814I'm thinking : Ubuntu & CVS head?
00:39.33wulfy814or should I stick with stable
00:39.54Delta34whats ubuntu?
00:39.55harryvvI seriosly think there is some kind of bug in this distro of @home
00:40.05pfnsome sorta linux dist
00:40.28pfnI dunno, I go for fedoracore
00:40.30pfnor centos
00:40.32harryvvanyone happen to have the asterisk@home authors email.
00:40.46Sedoroxisn't it centos?
00:40.52Sedoroxwhich is RHEL based?
00:40.52pfnis it?
00:41.03pfnyes, centos is rhel "based"
00:41.05Delta34sedorox: yes
00:41.18pfnwhere "based" = s/RedHat/CentOS/g
00:41.20*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
00:41.24wulfy814distro based on Debian
00:41.42Ariel_Hello everyone
00:41.46harryvvhi ariel
00:41.48wulfy814Delta34: seriously, you haven't heard of Ubuntu or are you pulling my chain
00:42.07opus_i think centos is like 99% RHEL
00:42.14harryvvdelt, do a /list on here
00:42.15Ayanohi ariel
00:42.15harryvv:)
00:42.20Ariel_ubuntu is in my view more of a desktop debian build
00:42.22Delta34nah, neverheard
00:42.31wulfy814I like the debian based distro - I guess my question was more Head or Stable
00:42.35Delta34whats a /list?
00:42.35Ariel_harryvv, how are you doing. Get your wifes car fixed?
00:42.44generalhancan anyone recommend a good VoIP provider for the West Coast ??? some one other than freaking Vonage ... please !
00:42.48Delta34not following your lingo
00:42.56wulfy814I far prefer apt to rpm hell anyday
00:43.02Ariel_generalhan, race.com
00:43.26opus_i perfer gcc over rpm
00:43.29Qwellwow, spammy
00:43.35Ariel_I like yum and rpm's But that is why there are many distro's.
00:43.52SarahEmmsivana: you around? :)
00:44.00Sedoroxrpm
00:44.03Sedorox's SUCK
00:44.11Sedoroxyay for dependancy hell!
00:45.03Delta34wow hella peeps on ubuntu channel, but gentoo most popular
00:45.22*** join/#asterisk iswm (iswm@iswm.user)
00:45.26*** join/#asterisk |nix (~inix@cm11.gamma116.maxonline.com.sg)
00:45.28wulfy814ok so, back to my original question CVS Head or stable?
00:45.41Ariel_wulfy814, what are you setting up.
00:45.45Sedoroxmost people would say HEAD...
00:45.55Ariel_I use stable for production systems
00:45.57Sedoroxyay for gentoo's
00:46.11SarahEmmSedorox: you're a gentoo grrl too? ;)
00:46.30Sedoroxwell.. gentoo boi
00:46.30Sedorox:p
00:46.33SarahEmm:)
00:46.50Sedoroxanything redhat based sucks.. but thats my personal opinion :p
00:46.58Delta34anybody use an ata188 with *
00:47.07wulfy814Ariel_ : system for small company using sipura spa-3k's for 5 pots lines and 6 Polycom IP501's
00:47.14Delta34thinking about buying one for my analog fax machine
00:47.20Delta34and was wondering if it would work
00:47.25Ariel_wulfy814, how much do you know about asterisk?
00:47.43wulfy814so so, compiled and installed it on my Clark Connect box at home
00:47.50Ariel_Delta34, the sipura work just fine. I am using them all over the place.
00:47.56wulfy814modified configs on a gentoo based install that I hired out
00:48.32wulfy814I guess the real question is are they any great gains in Head over Stable
00:48.32Ariel_wulfy814, If you want a normal setup I would use stable. If your good with asterisk then go and use head.
00:51.14Ariel_wulfy814, for all my customers I use stable I know it works and I don't change much. Now if there are features you want from head then take and build it test it and then deploy it but don't update it unless you do testing first.
00:51.43fluidicslasterisk is failing to start can any one give me a hand ?
00:52.09Ahewesfluidcsl: maybe
00:52.11Delta34how u starting it?
00:52.27fluidicsljust " asterisk" its not hte first time I started it
00:52.42Aheweswhats the error
00:52.44*** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net)
00:52.48fluidicslUnable to specify channel 1: No such device or address
00:52.48fluidicslUnable to open channel 1: No such device or address
00:52.48fluidicslhere = 0, tmp->channel = 1, channel = 1
00:52.48fluidicslUnable to register channel '1'
00:52.48fluidicslchan_zap.so: load_module failed, returning -1
00:52.48fluidicsl<PROTECTED>
00:52.50fluidicsl<PROTECTED>
00:52.52fluidicslLoading module chan_zap.so failed!
00:53.28Aheweswhere is chan_zap.so located. is it in /usr/lib/asterisk/modules/ ?
00:53.35JunK-Yrun ztcfg -vvvv 1st
00:54.12*** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)
00:54.40*** join/#asterisk |Vulture| (~V@76.233.204.68.cfl.res.rr.com)
00:54.41|Vulture|Anyone know what port DUNDi uses for lookups?
00:55.08fluidicslas
00:55.17fluidicslyeah
00:55.23fluidicslthats where it is
00:55.28*** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net)
00:55.58Ahewesis that where /etc/asterisk/asterisk.conf says it should be?
00:56.19AhewesI guess either your zaptel modules are broken or misconfigured.
00:56.28wulfy814ok somewhat unrelated
00:56.38wulfy814on the asterisk I installed on my Clark Connect
00:56.47jdv79anyone know any good machine detect links?  the subject isn't the best documented as far as i've found.
00:56.49wulfy814Call parking is not working properly
00:56.49fluidicslyeah thats where it should be
00:56.52niZonthis is odd
00:56.58daniel101Do I absolutly need a wildcard like the digium x100p for asterisk to work
00:56.59niZonmy festival is being mean
00:57.02wulfy814file.c:568 ast_readaudio_callback: Failed to write frame
00:57.03niZonit's all garbled and fast
00:57.09niZonanyone have any ideas?
00:57.14wulfy814it's not reading back the extension call is parked
00:57.25Ahewesfluidcsl: well, I'm struggling with a similar problem on my laptop.  I'll take a look at that and let you know.
00:57.27Q-At-Homelong time no see all :)
00:57.48*** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com)
00:58.07nDuffI have 'exten => _9NXXXXXX,1,GotoIf(["${CALLERIDNUM:-4:1}" = "5"]?2:3)' in my dialplan (as part of a script to set externally-viewable CID for a specific group of extensions). However, it's producing unexpected behaviour:
00:58.10nDuff<PROTECTED>
00:58.11nDuffand, when handling a number which *does* come from  a 5xxx extension,
00:58.11nDuff<PROTECTED>
00:58.11nDuffI would expect at least *one* of these to goto 2 rather than 3.
00:59.17Q-At-Homespeaking of caller id, has anyone cobbled up something to match on cid name rather than number?
00:59.45Ahewesfluidcsl: how did you install asterisk?
01:00.11Ariel_argh did I say I hate Mandrake.  I have a system that well just plain does not work right.....argh
01:00.15fluidicslasteirsk was working
01:00.22fluidicsllike a couple of days ago
01:00.24fluidicsljust stoped
01:00.27*** join/#asterisk joshpbx (~joshpbx@ayc122.neoplus.adsl.tpnet.pl)
01:00.40nDuffdanalien, no.
01:00.45nDufferm
01:00.48nDuffdaniel101, no.
01:00.49niZonstopped like my festival... :(
01:02.28QbYwhat does _ mean? in extensions.conf?
01:02.37Q-At-Home"match"
01:03.01Q-At-Home_9NXXXXXX = match any number after a 9
01:03.07*** join/#asterisk Brixius (Brixius@c-24-118-215-163.hsd1.mn.comcast.net)
01:03.43QbYso _8X?  what would i dial for my phone to get that?
01:03.53QbY81?
01:03.55Q-At-Home8 followed by any number
01:04.00daniel101ok
01:04.16Ahewesfluidcsl: can some of your stuff have been damaged,overwritten, or permissions changed?  Can you back up your config and re-install asterisk?
01:04.23wzlwzlanyone ever experience double dtmf with certain calls via broadvoice lines?.... when i call from my cell->bv->*, asterisk is registering double dtmf... same if i call from vonage->bv->*.. .. obviously the problem is broadvoice, no?
01:04.27fluidicslno
01:04.35QbYQ..  How would I make a conference call?  if _8X is MeetMe.. Why does it hang up after it connects?
01:04.38fluidicslbut I might have messed up a config file before I left
01:04.54fluidicslI am trying to figure out what I should be looking at
01:05.28hypa7iaoy, done compiling... now to actually get the damn thing working :-)
01:05.34BrixiusHello, stupid question but I can't seem to find it on the wiki,  In Extensions.conf, if I want to have extensions 68(0-4 and 8)X, would it be 68[0-4,8]X or 68[0-48]X or am I wrong on both counts?
01:05.52Q-At-Homeexten => 81,1,Meetme,1234
01:06.09Q-At-Homemake sure you have a matching 1234 conference in meetme.conf
01:06.16QbYah
01:06.21nDuffneeded to be $["${CALLERIDNUM:-4:1}" = "5"] (was missing that initial $)
01:06.33QbYQ.  Lets say someone calls me, and I need to just bring in another individual (on our system) on the line with us..  How?
01:06.45Q-At-Homehave that person call the meetme
01:06.50Ahewesfluidcsl: you should look at /etc/asterisk/zapata.conf.
01:06.57QbYok
01:06.57nDuffQbY, your SIP phone will frequently have a "conference" button.
01:06.58fluidicslok
01:06.59Q-At-HomeI believe the new meetme supports outcall, but I've not used it
01:07.00Ahewesbut at times like this, I always look at my navel.
01:07.01Sedoroxor if they are on a seperate line on your phone.. transfer them to it
01:07.14QbYnDuff.  Unfortunately we are testing with X-Lite, and the Conf doesn't work
01:07.15nDuffQbY, you can use that in cases where you're in a 2-person discussion and need a to bring a 3rd in.
01:07.23nDuffoh.
01:07.39Q-At-HomeQbY: are we talking bringing someone into a meetme?
01:07.40nDuff(right, that's one of the pro-only features there IIRC)
01:07.55fluidicslI didint c any thing wrong with this cancallforward=yes
01:07.55fluidicslcallreturn=yes
01:07.55fluidicslechocancel=yes
01:07.55fluidicslechocancelwhenbridged=yes
01:07.55fluidicslrxgain=0.0
01:07.55fluidicsltxgain=0.0
01:07.57fluidicslgroup=1
01:07.59fluidicslpickupgroup=1
01:08.01fluidicslimmediate=yes
01:08.03fluidicslmusiconhold=default
01:08.05fluidicslchannel => 1
01:08.08JunK-Yfluidicsl: stop flooding, use pastebin
01:08.11JunK-Y~pastebin
01:08.11jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
01:08.11QbYQ.  I have a customer on the phone, and I need to get the next department on the line with us at the same time to authorize something..
01:08.41Q-At-Homewithout conference, you cant... you can however xfer using # if you have # xfers setup
01:09.07QbY# is set up.. but sometimes we just need them on the line for a brief moment and then they will drop off
01:09.11Q-At-HomeI've been off IRC for like 6 months... wow
01:09.15*** join/#asterisk matt_cn (~matthew@210.22.166.62)
01:09.24Q-At-HomeQbY: get yourself some hardware phones
01:09.36QbYyeah i'm planning on it..
01:09.41Q-At-Home(thats my answer for everything)
01:09.48Q-At-HomeI love hardware
01:09.55Q-At-Homeyou could try iaxcomm
01:10.03Q-At-HomeI think it supports conf, and its free
01:10.36Q-At-Homeyou could open the manager up, using Gastman (if it still exists) and drag the new ext onto your exiting call to "test"
01:10.36AhewesQuestion: How can I force asterisk to attempt to register a SIP trunk in the CLI?
01:10.40Q-At-HomeI think that works
01:10.52Q-At-Homesip reload should reparse the config
01:11.19Ahewesthanks
01:11.51Q-At-Homestill trying to match the caller id "name" and or number.... if anyone has a hint
01:12.11Q-At-HomeI have a massive faxer here that always uses the same cid name, but different numbers
01:12.17Q-At-Homesick of the faxes
01:13.08ManxPowerQ-At-Home: Ah.  GotoIf($["${CALLERIDNAME}" = "Evil Telemarketer"?45:50) or something like that
01:13.28ManxPowerpbx-1*CLI> show applications like black
01:13.29ManxPower<PROTECTED>
01:13.29ManxPower<PROTECTED>
01:14.03*** join/#asterisk rene- (~rene-@dup-148-221-117-113.prodigy.net.mx)
01:14.06Q-At-Homehrm.. that would work
01:14.19Q-At-HomeI've been neglecting my pbx for some time :)
01:14.20Q-At-Homethanks
01:14.55*** join/#asterisk hardwire (~hardwire@209.112.194.45)
01:15.03Q-At-HomeI think I played with the blacklist, it didnt work right for my needs. but that gotoif looks promising
01:15.06Q-At-Hometime to play
01:15.10Q-At-Homethanks :)
01:15.11fluidicslso asterisk suddenly wont start http://pastebin.ca/18503 ( pertinant info tell me if you need more)
01:15.27rene-Hello, i wonder if anyone knew whether using Steve's Underwood Unicall R2 libraries notice any important penalty hit versus using plain PRI lines
01:15.30Q-At-Homeoh, does the if match "substrings" ?
01:15.45Q-At-Homeor is it an exact string for the =
01:16.19niZondoes anyone know why festival would suddenly sound like crap?
01:16.29*** join/#asterisk outtolunc (~me@adsl-69-110-52-25.dsl.pltn13.pacbell.net)
01:16.33hardwirefestival always sounds like crap
01:16.44SarahEmmlol
01:16.45SarahEmmagreed
01:16.46Ariel_niZon, maybe it's just festival that is like that.
01:16.50SarahEmmfluidicsl: what does zttool show?
01:16.51niZonwell like mushy just had the flue crap
01:16.56Q-At-Homefluidicsl: missing a zap module/hardware?
01:17.02wzlwzlanyone ever experience double dtmf with certain calls via broadvoice lines?.... when i call from my cell->bv->*, asterisk is registering double dtmf... same if i call from vonage->bv->*.. .. obviously the problem is broadvoice, no?
01:17.07niZonas in a 10 word sentence in less then a second
01:17.19QbYhttp://pastebin.ca/18498 -- could someone take a look and tell me what i'm doing wrong..
01:17.22Ariel_fluidicsl, have you started that zaptel service?
01:17.28QwellniZon: use punctuation
01:17.47QwellI've had mine sound really out of breath after a few sentences without periods
01:18.16Ariel_wzlwzl, bv uses ulaw so make sure your using dtmf=inband.
01:18.34Ariel_also canreinvite=no
01:18.36Delta34fluidicsl, u can do lsmod to see if its loaded
01:19.24fluidicslzaptel and wcfxo is loaded
01:19.47Ariel_fluidicsl, what does the ztcfg -vvv say?
01:20.04Q-At-HomeManxPower: think I got it
01:20.12fluidicslChannel map:
01:20.12fluidicslChannel 01: FXS Kewlstart (Default) (Slaves: 01)
01:20.13fluidicsl1 channels configured.
01:20.53fluidicslI fixed it
01:20.55fluidicsl:)
01:20.59fluidicslhave no idea how though
01:21.16wzlwzlAriel_: http://pastebin.com/320687
01:21.40wzlwzli've also tried commented out the allow=alaw line, but that didnt affect anything
01:22.08joshpbxsomone maybe know why that person who i call hear my voice slowly..this same hapend local when i record it.
01:23.10*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
01:23.28Ariel_joshpbx, slow hard drive or your running out of memory.
01:23.50niZonaha!
01:23.52niZonfixed!
01:23.53wzlwzlAriel_: updated with log: http://pastebin.com/320690
01:23.57joshpbx512mb, P4 3.0Ghz, i dont think so that hardware problems..
01:25.19Q-At-Home512 might be light... depending on what you're doing
01:25.33Kizmetjoshpbx, make sure your running the latest kernel (linux kernel) 2.6.11 works nice for me.
01:25.41*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
01:26.13Ariel_wzlwzl, your using amp.  The problem with cell phones and bv and others is in my view a problem.  I have had it before.  Don't remember what I did to fix it. let me look at a customer that is using bv to see there setup.
01:26.21joshpbxKizmet: it`s not linux. it`s bsd ;)
01:26.39wzlwzlAriel_: thanks
01:26.41Kizmetwell install Debian and all your issues will be resolved
01:26.42Kizmetlol
01:27.02Kizmetjoshpbx, i only just got Asterisk realtime stuffs working :D
01:27.52joshpbxi cant, it`s not home or office pc, it`s big serrwer and it will be not nice to move all thinks to debian.. better idea it`s just add some other box..
01:28.01joshpbxthen nobody know?
01:28.17wzlwzlAriel_: yea, i was using AMP.. i've since just switched to manually editing config files..
01:28.20joshpbxKizmet: u run cvs head or stable?
01:28.38joshpbxwzlwzl: manually it`s better.. ;P
01:28.45wzlwzlyea, no kidding
01:29.05Kizmetjoshpbx, sounds like you need more memory... i also run "Asterisk CVS-NHEAD-07/24/05-00:09:34"
01:29.14*** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com)
01:29.22wzlwzlive gone from asterisk newb -> asterisk-dangerous (that level where knowledge makes you more dangerous than useful) in just a cpl days
01:29.33Kizmetwzlwzl, lol.
01:29.40joshpbx;) more memory? but 60% it`s not used..
01:29.54Kizmetjoshpbx, then its probs a hdd problem.
01:30.16Kizmeti done have any issues on my 4 calling card servers i just installed in a datacentre.
01:30.25joshpbxudma133? :P i dont think so..
01:30.30*** join/#asterisk bjohnson (~bjohnson@i216-58-13-224.igs.net)
01:30.34joshpbxKizmet: u have any other clue? :P
01:30.40Kizmetthen again they all have 4+ GB of memord and Dual Opteron Dual Core chips
01:30.46Kizmetjoshpbx, nope.
01:31.05joshpbxand how many person reach u pbx?
01:31.13Kizmetenough.
01:31.39Kizmetim talking 800+ lines going into Digium PRI cards.
01:31.53ManxPowerThat's a lot of servers
01:31.59Kizmetpeople call in and then use the calling card to dial internationaly for cheap.
01:32.10joshpbx;)
01:32.32KizmetManxPower, well they arnt exactly mine. they are the company that im working for's
01:32.48joshpbxgr, nobody hells now why this sound it`s so fucked,
01:32.57Kizmetthere is another 12 sheduled to be installed around the country. (Australia)
01:33.11Kizmetjoshpbx, have you tried different codecs
01:34.24daniel101What do I need to run asterisk ? I mean do i need special hardware or something ?
01:34.30ManxPowerjoshpbx: what is your specific problem?
01:34.35Kizmetjoshpbx, i know we are using G729 for the actual phones in the office.
01:34.37joshpbxyes, gsm,ulaw,g729, but what it can be, if i hear voice ok and 2nd part hear my voice so slowly..
01:34.56Kizmetjoshpbx, have you got a licence for g729 ?
01:35.15joshpbxnope, i find some modules precompiled in some website.
01:35.17joshpbx;)
01:35.30Kizmetjoshpbx, they are the modules. yes.
01:35.38Kizmetbut you need a licence to activate them.,
01:35.48ManxPowerjoshpbx: I can't help you with problems with pirated codecs.
01:36.00Kizmet_DAW, its ok. he doesnt actually have g729 he just has the module..
01:36.02joshpbxManxPower: but it`s not codecs..
01:36.09Kizmetbut no licence for it..
01:36.14_DAWright no license
01:36.18_DAWboooo
01:36.30ManxPowerjoshpbx: disallow=all allow=ulaw should be the only allow/disallow lines in sip.conf
01:36.58ManxPowerjoshpbx: do that.  You will remove the red herring of the illegal codec.
01:37.08joshpbxManxPower: i use ulaw, and when i make echo test or just call voicemail my sound looks to slowly..
01:37.10*** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net)
01:37.28KizmetManxPower, i would suggest ' disallow=all ' and ' allow=alaw allow=ulaw allow=ilbc allow=gsm'
01:37.35joshpbxManxPower: i`m not using it i just try, because i thought that it`s something with codecs..
01:37.39ManxPowerjoshpbx: And you confirmed the codec by using "sip show channels" during a problem call?
01:37.51SwKyou can use g729 in passthru
01:37.52ManxPowerjoshpbx: there's nothing wrong with the codecs.
01:38.01joshpbxi`m not using sip ;P iax2
01:38.02ManxPowerif you are using ulaw
01:38.03SwKonly need the codec if its going into voicemail or an IVR
01:38.13Q-At-Homek, I'm lost... ManxPower: can you explain "Evil Telemarketer"?45:50 the ?45:50 part is what I dont get
01:38.13ManxPowerjoshpbx: Diagram your setup
01:38.31joshpbxdiagram?
01:38.39ManxPowerQ-At-Home: if true it will jump to priority 45 of the current extension, if false jump to priority 50
01:39.02joshpbxManxPower: can i pm?
01:39.13Q-At-Homeah ok, tips on "pattern matching" i.e I want to match "Evil" in any calleridname
01:39.15ManxPowerMy Home Asterisk:  POTS/PSTN<->X100P/Asterisk/CiscoSwitch/SIPura
01:39.34ManxPowerQ-At-Home: THAT I don't know off the top of my head how to do.
01:40.00Q-At-Homewoohoo, I stumped ya :)
01:40.09Q-At-HomeI'm gunna go reading for abit
01:40.25ManxPowerQ-At-Home: Expecially README.variables  Read it twice for good measure
01:40.35SuPrSluGhaving problems compiling cvs. a problem w/ say.c . is there a fix
01:42.43QwellSuPrSluG: perhaps if you paste errors to a pastebin
01:42.49Qwell~pastebin
01:42.49jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
01:42.56ManxPowerYa know what might be cool?  Asterisk putting the UNIQUEID in the logging messages
01:43.29stormfrHello, i have many of these error : "chan_zap.c:4394 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 63 - audio may have been lost". I have read this due of packet lost, but i have this error sometimes even on the same lan without any packetlost on the network. Any idea of the reason, and a way to remove these warning how fill log ?
01:43.37ManxPowerWhen you have 4 people checking voicemail at the same time, for exmaple, you would be able to see what files were playing at which time to which call
01:43.56Q-At-Homeamen
01:44.15ManxPowerstormfr: check duplex of the pc and switch
01:44.54stormfrmanxpower : already very... i manage an european isp backbone, first thing i have check ;)
01:45.35ManxPowerstorm is the ethernet port on that machine very busy?
01:47.08wzlwzlAriel_: any luck?
01:47.31stormfrmanxpower : no
01:47.49Q-At-HomeI believe I'm looking for a : instead of an =
01:47.50Q-At-Home:)
01:47.55_DAWhey, everyone.. On a previously working system (that has been working great on stable 1.0.7 ) I am now having problems with transfers dropping due to Zombie Sip channels out of the blue.  The net admin at the client site has just seperated the * box from all the sip phones via vlans and internal routing on his switch.  Could that account for these occasional problems.  The problem sure seems netowrk related.
01:48.09*** part/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net)
01:48.25ManxPowerstormfr: I have no more questions/suggestions.  Why couldn't you have an easy problem, like not looking at the CLI? 8-)
01:48.39*** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net)
01:49.24*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
01:49.25ManxPower_DAW: canreinvite=no in sip.conf.  It may have to go in each sip.conf stanza.  If the problem goes away, then you have a vlan/routing problem.
01:49.25Kizmethas anyone got an idea what this is :
01:49.26Kizmet08715048066
01:49.33KizmetJul 26 11:49:25 WARNING[5236]: res_config_mysql.c:323 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info.
01:49.53Kizmetthats just after a :     -- Saved useragent "Wu Chuan" for peer 2001 line
01:50.02stormfrKizmet : is realtime working before ?
01:50.08Kattywebcam is on (=
01:50.14Kizmetstormfr, its working fine.
01:50.29Kizmetjust i get a screen flood of that message and its getting annoying.
01:50.39Kizmetwhen all the phones re-register
01:50.44Kizmetall 200 of them.
01:50.44_DAWManxPower - Much thanks, will try it out.
01:50.49stormfrKizmet : your database is not corrupt ? i never see this except bad configuration or mysql down maybe
01:51.04Kizmetstormfr, nothing is down.
01:51.27Kizmetfrom what i can gather it is trying to save the useragent into the database.
01:51.51Kizmetbut there is no-where for it to save it too.
01:53.11stormfrkizmet : normally it don't store useragent
01:53.22stormfrkizmet : add log to mysql and see what sent asterisk to mysql
01:58.55twistedyay!
02:02.09puzzlednite all
02:03.19Kizmetstormfr, what would the log table be ?
02:03.38*** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net)
02:04.01stormfrKizmet : add log to your my.cnf file
02:04.21SuPrSluGhere's the say.c problem when compiling. http://pastebin.ca/18504
02:04.32SuPrSluGany ideas?
02:04.46stormfrKizmet : if you dont have create it with [mysqld] \n log
02:06.05*** join/#asterisk xtrvd (~x@d209-121-36-44.bchsia.telus.net)
02:07.01niZonis anyone here a master with regular expressions in php?
02:07.41xtrvdCould somebody point me in the direction of a method to call up *, hang up, and have it call you back presenting you with an IVR so that you can dial out?
02:08.48xtrvdSo far I've got my IVR setup, but I can't figure out how to have * call back after 'x' seconds, nor have it present an IVR upon calling.
02:10.55Kizmetxtrvd, read the documentation :)
02:10.58bkw_drumkilla,
02:12.53twistedi'm sorry, drumkilla is not currently reachable.  please leave a message after the <tone>
02:12.55twisted<tone>
02:15.21xtrvdKizmet: I've read the documentation!!! *sigh*   a little help wouldn't hurt... what specific documentation you're talking about, etc.
02:16.08twistedeh
02:16.10twisteder heh
02:16.27Kattyfile makes nice chair
02:17.01file[laptop]that's what they tell me
02:17.23*** join/#asterisk SwK (svtild@12-219-156-206.client.mchsi.com)
02:17.39*** join/#asterisk MikeJ__ (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
02:18.33twisteddon't fall over
02:19.00twistedfalling over is bad
02:19.12file[laptop]quite true
02:19.26*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
02:19.34*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
02:20.21twistedholy crap
02:20.22twistedlike
02:20.34twistedeveryone on my buddy lists is online
02:20.54twistedit must be a sign of the apocolypse
02:21.00file[laptop]even me?!?
02:21.05WilliamKwow
02:21.11Qwellfile[laptop]: You aren't on his list. :(
02:21.13twistedfile, oh.
02:21.14Kattyfalling off teh interweb is bad
02:21.19file[laptop]actually I am! OMG!
02:21.25twistedyou are?
02:21.25hypa7iabuddypocalypse
02:21.29file[laptop]twisted is toasted
02:21.36twistedi am not toasted
02:21.40twistedi am extremely tired
02:21.44file[laptop]same thing
02:21.49twistedno
02:21.51twistedbig difference
02:21.52Kattyno one from asterisk is on mybuddy list
02:21.54brookshirenubb
02:21.55file[laptop]nooooo yessssss
02:21.59file[laptop]brookshire: Matttt
02:22.05twistedKatty, aw
02:22.25brookshirehey
02:22.27Kattyeveryone from slashnet is though
02:22.38twistedheh
02:22.47twistedbrookshire
02:22.52twistedyour dog wanted to eat my car
02:22.56brookshiretwisted, so who's the one person on your list?
02:23.08twistedbrookshire, the one that does what?
02:23.10Darwin35twisted you work is silkion right
02:23.20twistedDarwin35, you english is borked right
02:23.40Darwin35brain is borked over worked underpaid
02:23.40brookshireengrish
02:24.09file[laptop]cry me a river.
02:24.11Q-At-Homesitting here maintaining the "evil" list makes me think I should use an AGI...
02:24.12twistedlol
02:24.43file[laptop]you play violin? COOL!
02:24.48craziman2Can some one point me to a reference for the 'hello world' AGI?
02:24.59twistedfile[laptop], on certain occasions ;)
02:25.10file[laptop]scary
02:25.13*** join/#asterisk SwK (sdxfmw@12-219-156-206.client.mchsi.com)
02:25.40Darwin35swk whats up homer
02:25.57SwKlooking for donuts
02:26.07*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
02:26.07Q-At-Homebbl
02:26.37twistednah, what's scary is watching a hay bailer make a wide right turn into oncoming traffic
02:26.40Darwin35to support the projects I want to do
02:26.53Darwin35thats scarry
02:27.07Darwin35and twisteds driving is 2nd in that group
02:27.18twistedyou have never ridden with me
02:27.21twistedso stfu n00b
02:27.34file[laptop]twisted: let's go on a killing spree!
02:27.45twistedfile[laptop], not tonight, i'm too tired.
02:27.51file[laptop]that's sad :(
02:28.11*** join/#asterisk Rakko (~Eric@68-115-21-42.dhcp.mdsn.wi.charter.com)
02:28.34*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
02:28.38twistedoh no.. MikeJ ran into services!
02:28.44file[laptop]or they ran into him.'
02:28.48file[laptop]one or the other
02:28.49twistedyeah
02:28.59twistedbut they both need to watch which way they're going
02:29.06KizmetDoes anyone know of a MySQL logging thingo for Asterisk.
02:29.23Darwin35look in asterisk-addons
02:29.28Darwin35and the wiki
02:29.31MikeJ[Laptop]sigh...
02:29.39hypa7iaor use the one in HEAD, via odbc
02:29.46XamoDoug2a
02:29.49*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
02:29.54MikeJ[Laptop]my laptop had a leftover route from the office, so i had 2 0.0.0.0 routes, one to nowhere land...
02:30.03file[laptop]yay nowhere land
02:30.10MikeJ[Laptop]no wonder I was having such a hard time staying connected
02:30.13twistedoooh
02:30.16twistedshiny object
02:30.17Darwin35never never land
02:31.25Darwin35SWK  your in Kentucky right
02:31.33Darwin35miss spelled I bet
02:31.42SwKHSV
02:31.46Darwin35thats it time for caffiene
02:31.52Darwin35ok
02:32.00Darwin35Alabama
02:32.13Darwin35Ala Bam a
02:32.15kramalabama? :)
02:32.22SwKhah
02:32.30twistedkram!
02:32.31SwKKram in the hizzy
02:32.32twistedwtf
02:32.33file[laptop]double away!
02:32.34twisted<PROTECTED>
02:32.42KizmetDarwin35, as far as i have searched there is NO mysql Asterisk logging on the wiki
02:32.47MikeJ[Laptop]twisted... we don't care ;)
02:32.57twistedMikeJ[Laptop], I do
02:33.04MikeJ[Laptop]:P
02:33.10file[laptop]I just looked at Entourage and tried to navigate it like an IRC client, I kept looking for the asterisk channel
02:33.20twistedfile[laptop], nice.
02:33.43Darwin35ding
02:33.49Darwin35round 1
02:33.49twistedMikeJ[Laptop], i think he wants us to fight
02:34.44Darwin35fine I can take them
02:34.48file[laptop]"me father was a tree!"
02:34.50Darwin35I am use to the abuse
02:35.07Rakkoand your mother smelled of elderberries
02:35.42twistedwhoa
02:35.44twistedthat was random
02:35.51MikeJ[Laptop]Darwin35, square circle.. impressive...
02:36.13twistedMikeJ[Laptop], have you never seen a square circle?
02:36.15MikeJ[Laptop]twisted, nah.. I have no beef with you...
02:36.26MikeJ[Laptop]oh.. and you are bigger than me.
02:36.28MikeJ[Laptop]hehe
02:36.35filebigger where it counts?
02:36.40brookshirecircle circle dot dot?
02:36.40twistedhttp://www.cornermark.com/kinetic/images2/squarecirclerelationship.jpg
02:36.46twisted^^ square circle
02:36.58Qwellheads up: If you buy a product from somebody, and they ship it to you (even though you very likely paid for the shipping), you aren't UPSs customer.
02:37.00MikeJ[Laptop]hmmmm
02:37.10twistedMikeJ[Laptop], i have no beef with you either
02:37.11Qwell</pissed>
02:37.15Darwin35swk whats life like out there
02:37.21hypa7iaQwell: common carrier
02:37.21MikeJ[Laptop]I have beef in the freezer
02:37.26twistedhaha
02:37.31twistedi don't even have beef in the freezer
02:37.34MikeJ[Laptop]and deer too
02:37.43Qwellhypa7ia: sorry?
02:37.44Darwin35everyone has a beef with me it seems just because I am a  non linux user
02:37.58SwKits like hell
02:38.00twistedDarwin35:
02:38.00twistedDarwin Acantha.local 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc
02:38.01SwKonly more humid
02:38.02Qwellhypa7ia: I can't tell if you're saying I have a right, or if you're saying I'm screwed. :p
02:38.10brookshireDarwin dhcp-157.digium.com 7.9.0 Darwin Kernel Version 7.9.0: Wed Mar 30 20:11:17 PST 2005; root:xnu/xnu-517.12.7.obj~1/RELEASE_PPC  Power Macintosh powerpc
02:38.10brookshire;D
02:38.12hypa7iaQwell: they are a common carrier.  they don't really have all that much responsability to anyone :-/
02:38.19Darwin35coool
02:38.22Qwellahh, the latter then
02:38.27brookshirebsd still sucks for asterisk tho
02:38.30hypa7iayup
02:38.31Darwin35no
02:38.31twistedbrookshire, heh
02:38.41Qwellsadly, I'll I'm trying to do is refuse a package before it gets here, heh
02:38.42SwKDarwin swk.homeip.net 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc
02:38.49Qwellthey won't let me refuse it, until it gets to my doorstep...
02:38.54twistedmy last commit was porting muted to darwin
02:38.56Darwin35I have festival and res_sqlite and res_perl and sphinx and loads more working
02:38.59Qwellexcept that I won't be here, and by default makes me accept it. :D
02:39.04twistedwell
02:39.05twistedactually
02:39.06twistedto osx
02:39.10Darwin35just sucks if you must have the latest hardware
02:39.25MikeJ[Laptop]Qwell, just say you never got it
02:39.35MikeJ[Laptop]let them prove you did without a signature
02:39.50QwellMikeJ[Laptop]: the main office will accept it, since I won't be here
02:39.52MikeJ[Laptop]leave a note on the door that you do not accept any packages w/o a sig
02:40.01MikeJ[Laptop]tell them not to
02:40.04QwellMikeJ[Laptop]: I haven't seen the UPS driver in weeks
02:40.15Qwellmy apt people are idiots, unfortunately
02:40.25QwellI often have to prompt 2-3 times if I even have a package sitting there
02:40.29Darwin35Swk what was your former nick
02:40.38*** join/#asterisk HellAgony (~HellAgony@200.121.216.229)
02:40.39QwellI doubt they'd remember, even if I told them 5 minutes beforehand
02:41.05twistedQwell, so write it with a magic marker on a big cardboard sign
02:41.07Qwellso now I'm on hold, because the UPS supervisor was too busy to talk to me...thats so awesome
02:41.10twistedand stand that up in the lobby
02:41.32brookshireQwell: they just don't want to talk with you
02:41.41Qwellbrookshire: yeah, I'm kinda an asshole
02:41.48QwellI demand lube
02:41.51brookshirewe should conf ups
02:41.57Qwellbrookshire: we should
02:42.02Qwellactually...
02:42.07Qwellyou guys wanna help me wardial them? :P
02:42.09twistedi'm awesome at bitching out shipping companies
02:42.10Darwin35tomarrow I am embedding fbsd 5.4 with asterisk/festival/sphinx/spandsp/res_sqlite/res_perl and more
02:42.14Qwell</childish>
02:42.37Darwin35on the new pc/104 1.5 system I got
02:42.42brookshireDarwin35: good for you!
02:42.49brookshirebsd still sucks for asterisk
02:42.50brookshire:D
02:42.59Darwin35why have you tested it latly
02:43.11brookshireno tdm interfaces
02:43.13Darwin35or is that just a anti bsd remark
02:43.14Qwellbrookshire: You should modify iaxtel to force all outgoing calls to hit 800-pick-ups :p
02:43.14brookshireso yes.. it sucks
02:43.16brookshire:D
02:43.21Darwin35excuse me
02:43.27brookshireQwell: that is if it worked
02:43.28Darwin35I have a tdm40b
02:43.32Darwin35and it works
02:43.33Qwellbrookshire: there is that, yes
02:43.37Darwin35we have drivers
02:43.45Darwin35for ver1 cards not ver 2
02:43.52*** join/#asterisk SwK (lffagp@12-219-156-206.client.mchsi.com)
02:43.56Darwin35we are working on ver 2 card drivers now
02:43.58*** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net)
02:44.22*** join/#asterisk netnameus (~netnameus@pcp05000344pcs.shrpsr01.tn.comcast.net)
02:44.23brookshiresee!
02:44.28brookshirelinux better for asterisk
02:44.35brookshire:P
02:44.42Darwin35what ever
02:44.52Darwin35use the os you like and know
02:44.58MikeJ[Laptop]no, digium just does not want to have customers who have anything but linux :P
02:45.00netnameushow's it going everyone?  Just found out about this chan...
02:45.17brookshiremike: linux and macosx
02:45.20*** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net)
02:45.51daniel101Amp is it free ?
02:46.05Darwin35no its just marked that way to mess with your mind
02:46.12Darwin35its really .99c
02:46.37harryvvfound out my issue why getting all circuits are bussy.
02:46.43Rakko99/100 of a cent?
02:46.56daniel101it says on their web site that it is open source portal
02:47.15Qwelldaniel101: open source almost always means free.  especially when there is a "download now" link
02:47.20Qwell(almost...not always)
02:48.00Qwellhmm
02:48.03QwellI have 6 lines on my 7960
02:48.10Darwin35really
02:48.12QwellI bet I could call and get 6 supervisors to put me on hold
02:48.20Darwin35wow and I thought the only came with 5
02:48.22*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
02:48.23daniel101ok lol, ty but their support are very expensive !!
02:48.47QwellI am so going to waste more money from UPS then they're getting from me on this shipment
02:48.49Darwin35my x401's will be here tomarrow
02:49.18Qwellsorry about my ranting btw...other channels don't pretend they care nearly as well as you guys. :p
02:49.30Kizmetlol
02:49.45Darwin35we only act like we care then mock everyone once they leave
02:49.55QwellDarwin35: I never leave though, unfortunately
02:49.55Kizmethahaha.
02:50.37Darwin35ok where did I put the network drive
02:50.45Darwin35my mail and config drive
02:51.02Darwin35and fax storage
02:51.56Darwin35there it is
02:52.38*** part/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com)
02:53.44Darwin35now to disable all the extra logging
02:53.48Darwin35grr
02:53.56hypa7iatake THAT x100p
02:54.00hypa7iadammit :-)
02:54.03netnameushow come pressing # on my phone (connected to * by an SPA-2002) doesn't always result in putting the call on hold? (or allow me to transfer/park)?
02:54.59netnameusI haven't figured out when it does and does not work... haven't dug into it that deeply yet... but was just wondering if anyone here would know of something i'm missing
02:55.36Darwin35have you updated the firmware and checked the company website for know issues
02:55.44harryvvare there one button voice mail phones on the market?
02:55.53Darwin35did you check to make sure the dtmf and dtmfmode are set
02:55.59drumkillawho said my name?!?!
02:56.04drumkillahypa7ia: !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
02:56.06Qwelldrumkilla: sorry :(
02:56.11hypa7iaOMGHI2U
02:56.18hypa7iahehe
02:56.23harryvvnetnameus, i have the spa 1000
02:56.56netnameusharryw, do you ever experience what i've described?
02:57.17netnameusDarwin35, thanks... i'll look into the firmware update
02:57.45Darwin35any time
02:58.39harryvvnetnameus where did you find out the key combinations for the use on a standard phone and a ata/
02:59.34brookshirerulless!
02:59.39brookshirei mean russell!
02:59.49harryvvnetnameus my music on hold for # works
03:00.00brookshire# ?
03:00.12drumkillabrookshire: 012938123
03:01.25Kizmetdoes anyone have a good way to get the billing rate (per min) from the number dialed.
03:01.27Darwin35grrrr
03:01.51Darwin35maybe I can get kram to hire mem to do bsd/asterisk
03:02.00brookshireyeah right
03:02.02brookshiregood luck with that
03:02.15brookshirelol... linux only :D
03:02.18harryvvmem?
03:03.24Darwin35it cant be linux only it already compiles and works on net/open/free bsd and osx and from what I am told slowlaris
03:04.29brookshirejust because it compiles doesn't mean it's made for it
03:04.41harryvvsolaris is still a trusted os.
03:04.53brookshireand lot more of asterisk is slowly being pushed into the kernel
03:04.57brookshireso it runs in kernel space
03:05.23brookshireoops.. maybe i said something i shouldn't
03:05.24brookshirelol
03:05.25opus_so whats the best moh implementation
03:05.38kimo_sabebrookshire: eww, why?
03:05.50Darwin35madplay
03:05.54brookshirekimo_sabe: because asterisk can die, yet keep going
03:05.59brookshirehigher reliablity
03:06.05kimo_sabe
03:06.08brookshiregreater performance
03:06.16kimo_sabebrookshire: uh, NO, putting stuff in the kernel does exactly the oposite
03:06.16brookshireaccess to all the resources
03:06.34Darwin35mp3play
03:06.36brookshirekimo_sabe: i've seen a demo of marks
03:06.44Darwin35mpg123 must die
03:06.49kimo_sabeit doesn't need access to all resources, and direct bit fondling tends to be more fragile
03:06.50brookshirehe killed asterisk completely, and the calls that were connected were still going
03:06.53*** join/#asterisk Moc_ (~mochouina@modemcable203.101-70-69.mc.videotron.ca)
03:07.25brookshireyeah.. you don't have to put it in the kernel if you don't want
03:07.28kimo_sabeso he's separated call control from operations
03:07.32harryvvfacinating
03:07.39brookshirebut some people might want it
03:07.44opus_can madplay play from the cd-rom?
03:07.48*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
03:07.49Moc_freaking cool... they will draw prizes at cluecon !!! T1 Card !!! I better beable to participate too !!!
03:08.06harryvvin california?
03:08.22kimo_sabebrookshire: you know the web server-in-the-kernel is out performed by one or two user-space servers, right?
03:08.37Darwin35well night kids
03:08.47Rakkowhat about kernel-mode linux?
03:08.49Darwin35lots to do tomarrow
03:09.22brookshireasterisk is not a webserver
03:09.23kimo_sabeRakko: a bad idea generalized
03:09.32*** join/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg)
03:11.53Rakkokimo_sabe: hehe
03:16.16*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
03:16.20*** join/#asterisk hmodes (hmodes@pcp03772956pcs.potshe01.pa.comcast.net)
03:18.32*** join/#asterisk Xeaded (~Xeaded@69-88-216-229.thewavz.com)
03:18.52opus_hmmm
03:18.58opus_No streaing music on hold
03:19.18opus_where is the debug
03:19.32opus_<PROTECTED>
03:19.38opus_thats it
03:19.59XeadedHas anyone here successfully configured a Mediatrix 1204 to work with asterisk?
03:20.14tzafrir_laptopkimo_sabe, no, the embedded httpd in the linux kernel doesn't perform *that* great compared to others.
03:20.33opus_embedded httpd?
03:20.46tzafrir_laptoptux, mentioned above
03:21.34tzafrir_laptopother user-mode servers often use sendfile and gain comparable performance on linux
03:22.27tzafrir_laptopYou shouldn't put things in the kernel without a *very* good reason
03:22.56kimo_sabesuch a good reason I've never seen
03:24.00brookshirewell.. it probably won't happen, lol
03:24.10brookshirei'm just full of crap :D
03:24.38opus_whats wrong with my music on hold string mp3:/var/lib/asterisk/mohmp3-empty,http://195.137.248.36:8020/
03:25.25opus_ls: /var/lib/asterisk/mohmp3-empty: No such file or directory what is this file? is it suppose to be null?
03:26.17*** join/#asterisk Strom_C (strom@66.159.243.60)
03:26.54Strom_Cwhy would I get the error message "failed to authenticate as dundi" on a direct IAX2 call?
03:28.04opus_do you use dundi? disable it
03:28.28xtrvdIs it possible to have * dial a number (outbound call) and present an IVR menu instead of just responding to an incoming call?
03:28.34Strom_CI use dundi on the PBX I'm trying to call, but I don't have dundi set up on the PBX that's calling
03:28.53kimo_sabextrvd: sure, why?
03:29.26kimo_sabeopus_: why do you have the mohmp3-empty in there?
03:29.40xtrvdkimo_sabe: I'm trying to setup a little system where I call Asterisk with my mobile phone, press an extension, hang up, have it call me (so I am using my unlimited incoming minutes) and use the IVR to place outbound calls.
03:29.54kimo_sabextrvd: ah, tricky
03:29.57xtrvd=)
03:30.04xtrvdI can't figure out how to get the IVR to dial me though,
03:30.19kimo_sabextrvd: I think the wiki has something about generating calls
03:30.21hmodesxtrvd: drop a file in /var/spool/asterisk/outgoing
03:30.32hmodesthere's a sample.call file included with the distribution
03:30.35hmodesand the wiki also has details
03:31.03xtrvdhmodes+kimo_sabe: Thanks both of you, I couldn't find anything in the wiki, but I'll try under 'generating calls'
03:32.15hmodesa search for call file gets it on the first hit
03:32.24hmodesasterisk auto-dial out
03:32.28kimo_sabextrvd: "call files" could be a good keyword
03:32.30*** join/#asterisk stuntshell (stuntshell@200.180.185.92)
03:32.34xtrvdthanks. =)
03:32.56xtrvdI had no idea where to look, =) Thanks
03:33.25hmodesand you can System() in the dialplan to either copy a static call file or generate one with variables
03:33.33kimo_sabextrvd: I generate ransom calls from procmail :)
03:33.35stuntshellQuick question: How can I configure Asterisk so it only dials out and not receive calls thru the PSTN?
03:33.54kimo_sabestuntshell: don't answer incomming calls
03:34.27stuntshellwise answer Kimo, and how about * itself?
03:34.45kimo_sabestuntshell: let me rephrase. Don't Answer() incomming calls
03:35.15xtrvdhehe
03:36.32stuntshelland where would I configure it so it does not answer incoming calls?
03:36.51kimo_sabestuntshell: the dialplan
03:37.07xtrvdunder [incoming] in the dialplan.
03:37.45kimo_sabestuntshell: it's a simple matter of not doing it. Hell, set your incoming dialplan to something invalid
03:38.55*** part/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
03:39.12*** join/#asterisk dijungal (~ovr@206.113.106.114)
03:39.18dijungalhello call
03:39.59dijungalis there a VOIP to PSTN service i can signup to to deliver my Asterisk PBX calls unto the PSTN network..?
03:40.23kimo_sabedijungal: lots, in the wiki
03:40.35stuntshellOhh I see
03:40.48dijungalin other words i want users to not call one another ... but be able to make regualar calls to phone lines
03:40.49stuntshellThank you Kimo, for a minute I took as a joke ;)
03:41.16filedijungal: there's TONS
03:41.29dijungalanyone u might want to suggest i look at..?
03:42.02dijungalthe site does not have a WIKI...
03:42.10dijungalhttp://www.asterisk.org/
03:42.32colinm_~wiki
03:42.33MikeJ[Laptop]Corydon, you around?
03:42.51btmdijungal: http://www.voip-info.org/wiki-Asterisk
03:43.07dijungalthanx.. found it.. :)
03:43.24dijungalsooo... any suggestions for VOIP to PSTN gatways...?
03:47.38*** part/#asterisk stuntshell (stuntshell@200.180.185.92)
03:47.54*** join/#asterisk doughecka (~Miranda@doughecka.user)
03:49.19*** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
03:49.26xtrvdJust a quick question regarding the .call files, how does one trigger them through the dialplan?
03:49.28AyanoHey hey hey
03:49.49kimo_sabextrvd: just drop them into the outgoing directory
03:50.02Ayanotrue true
03:50.07QwellPWNED...UPS supervisor = fired
03:50.11xtrvdHow will I control when it calls?
03:50.24Ayanowhen it is dropped in, it calls
03:50.55hypa7ianice Qwell, how did you manage that?
03:51.06Qwellhypa7ia: by not hanging up for an hour when he put me on hold
03:51.11hypa7iahehehe
03:51.13hypa7ianice!
03:51.14kimo_sabextrvd: you won't, it'll call immeditately
03:51.14Qwellhis manager was like "OMG, there's been a call on hold for an hours?!"
03:51.25hypa7iahahahaha pwnt
03:51.29QwellI explained the situation, and he's gone :D
03:51.34xtrvdOkay, how do I make it only call when I want it to? ....
03:51.41QwellAND I'm sending an email to the top 5 executives of UPS.
03:51.59QwellI did that with Capital One, and they called me the next morning to KISS MY ASS.
03:52.15*** join/#asterisk _deg_ (~deg@200.139.119.148.adsl.gvt.net.br)
03:52.25hypa7ianice :-)
03:52.30hypa7iasqueaky wheel
03:52.34Qwellindeed
03:52.48QwellI would have done the same to Adelphias CEO, but...he was kinda in jail at the time I needed it. :D
03:52.54hypa7iaoops :-p
03:53.14_deg_anybody using unicall + mfcr2 ?
03:53.14QwellCEOs get MAD when they get emails from customers...heh
03:53.38xtrvdKimo: is there some sort of way to create the call file from the dialplan?
03:54.06xtrvdAnd where should I be looking for these answers; the results in the wiki search were very limited.
03:54.06kimo_sabextrvd: make a script to generate it and run your script from System() in your dialplan
03:54.10dijungalok guys i'm a little lost again
03:54.19dijungaldo u guys understand what i am trying todo
03:54.23dijungalor should i go over it again
03:54.26xtrvdAlright, thanks Kimo, I need to figure out how to run scripts now. =)
03:54.31dijungalthe wiki is not helping.. :(
03:54.41xtrvdYou've been a big enough help so far though, thanks.
03:54.59dijungalthe wiki is nice through... it's LOADED with information that  i will need but not now...
03:55.14kimo_sabextrvd: exten => s,2,System(/path/to/script)
03:57.02xtrvdThanks Kimo_sabe, =)
03:57.34dijungalok so lets go again.... i want to deliver calls from my * server unto the pstn phone network
03:57.58dijungalis there a service i can sign p to..? or do i have to get some special hardware ?
03:58.07kimo_sabedijungal: either
03:58.43dijungalok i want to know about the cheaper solution which i think would be the service...
03:58.57dijungalcan u PLEASE direct me to what i should be looking at..?
03:58.57_DAWdijungal - depend on whether on not you want to terminate the call via voip or pstn..  cheapest would be voip.
03:59.23dijungalwhat do u mean terminate the call via VOIP or pstn..?
03:59.33dijungaloooh.. i want to ternimate via voip
03:59.35*** part/#asterisk Rakko (~Eric@68-115-21-42.dhcp.mdsn.wi.charter.com)
03:59.44_DAWdijungal - start looking here http://www.voip-info.org/wiki-VOIP+Service+Providers
03:59.52_DAWlots of voip providers listed there
04:00.01dijungalafter that...  it's non of my business how the pstn service routs the calls... as long as it gets there
04:00.03dijungalok
04:00.05_DAWfor both sip and iax term..
04:00.06dijungalthanx.. :)
04:09.19opus_arge windows xp is buggy
04:09.21MikeJ[Laptop]outtolunc, you here?
04:09.31harryvvhi daw whats up
04:14.40*** part/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
04:28.54*** join/#asterisk Craziman2 (~dcm@208.3.11.172)
04:30.35*** join/#asterisk brettnem (~chatzilla@user-0ccsrag.cable.mindspring.com)
04:31.02opus_damn i hate windows..
04:31.16opus_found multiple bugs just in explore
04:31.27brettnemsuprise suprise
04:33.57Strom_Chttp://www.stromcarlson.com/misc/phonesex.jpg
04:34.08Strom_Cme being bored while setting up a PBX :)
04:34.35brettnemlovely color
04:34.36QwellStrom_C: I'm scared
04:34.49Strom_Cdon't be ;)
04:34.50Qwellhaha
04:34.54Qwelland reasonably so
04:35.47Corydon76-homeStrom_C: I tried phone sex once... but my dick got caught in the number 1 on the rotary dial...
04:37.03Strom_Cit
04:37.13Strom_Cit's small enough that you even attempted that? :)
04:37.51fileStrom_C: geek.
04:38.23Beavethats pretty bored :)
04:43.58Corydon76-homeStrom_C: well, I used to think it was small... until I started sticking it in and guy bellowed like no tomorrow... :-P
04:44.09Corydon76-homes/guy/guys/
04:46.02Corydon76-homeOh, wait, was that TMI?  ;-)
04:47.50xtrvdkimo_sabe: I have been looking around for how to generate call files and haven't found anything that fits, could you or somebody else point me in the direction of dynamic callfile generation?
04:49.03kimo_sabextrvd: http://pastebin.com/320798
04:49.06kimo_sabextrvd: my random script
04:49.12xtrvd=)    Thank you ever so much.
04:51.38xtrvdYou call this like so: exten => s,1,System(/script/is/in/here.sh) ?
04:52.09kimo_sabextrvd: should be, mine is trigger off an email
04:52.38kimo_sabethat way I can make threatening calls by emailing from my cell phone
04:52.59xtrvdlol, =)  that's hilarious,
04:53.11*** join/#asterisk gaffer (~cliff@216.216.112.2)
04:53.50xtrvdOne other quick one that you should be able to answer; instead of calling Zap/1, calling a number in my outbound context should be as easy as 'outbound,9871234321'  correct?
04:53.55kimo_sabextrvd: what's more hilarious is when one of my friends with a sprint phone uses it. Sprint adds a "This message was sent from a Sprint PCS phone", which is of course read aloud with the rest of the message
04:54.16xtrvdlol!
04:54.36xtrvdOne day I'll figure out how to do that. =)
04:54.40kimo_sabextrvd:I donno about that
04:54.59xtrvdIt would be easiest to just set the IAX2 information in the script then?
04:55.20kimo_sabextrvd: I made it for a friend's wedding last year so we could kidnap the bride then shake down the crowd for ransom money
04:55.27kimo_sabeprobably
04:55.44xtrvdlol! That's classy right there.   A good coverup for your normal usage, but classy none the less.
04:55.52xtrvdWhat's your CallerID for it anyway?
04:56.00kimo_sabextrvd: it wasn't my idea, I just went along :)
04:56.41kimo_sabextrvd: at the time I think the caller ID was the groom's.  Now we've got a TelIAX account where we can set any callerid we want. Not that it does
04:57.13Craziman2what did this thing do?
04:57.14xtrvdPersonally, I enjoy setting the callerID as 'god' where I work. =)
04:58.39xtrvdCraziman: He setup a script that would read emails and open a voice conversation with another caller, then * would proceed to read the email (commonly a ransom note) to the receiving caller.
04:58.59xtrvdQuite classy if you ask me. =)
04:59.02Craziman2neat
04:59.04Craziman2I like
05:08.27opus_whats another alternative to MoH?
05:08.50*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:09.49kimo_sabeopus_: just hanging up on people instead of putting them on hold?
05:15.15opus_no
05:15.18opus_like a real solution
05:16.16kimo_sabeopus_: leave them in silence? What are you asking?
05:16.44opus_file, what do you use for MoH
05:16.50hmodesreverse lookup their callerid and dispatch a squad of ninjas!
05:16.57hmodesthat'll keep them entertained while they're on hold
05:17.02Delta34i'm using rawplayer
05:17.16opus_do you like it?
05:17.31kimo_sabehmodes: ooh, I like that idea. s,1,Answer(); .... s,2,DispatchNinjas()
05:17.45hmodeshehe
05:18.02Delta34so far, only been using for a couple of days, i dont like mpg123 cause it sometimes keep having run away processes
05:18.05file[laptop]I use mpg123
05:18.07file[laptop]and it works fine and dandy
05:18.49opus_i'm having problems streaming from shoutcast :(
05:18.52Delta34someone mentioned sox the other day
05:19.07opus_yes, sox sounds good
05:19.21opus_i want the ability to use the cd-rom tray, mic input, streaming, etc
05:19.42Delta34wow u need all that =)
05:19.48DarthClueformat_mp3
05:21.50opus_hmm. i think i'll use sox raw format and a fifo
05:22.12DarthClueplay_fifo exists as well i believe.
05:22.13opus_and pray to the unix god it doesn't fail
05:22.19opus_really
05:22.57DarthCluehttp://www.pbxfreeware.org/archives/2005/06/new_download_--_5.html
05:25.17xtrvdAnybody able to aid with this one: WARNING[24886]: pbx.c:1291 pbx_extension_helper: No application 'System(/etc/asterisk/callj' for extension (dialmenu, 128, 1)
05:25.44xtrvdI can't seem to get asterisk to run the 'callj' script
05:26.07DarthCluextrvd: check permissions?  does the file exist?
05:26.30xtrvdPermissions 777, file exists,
05:26.52xtrvdBoth are a roger,
05:27.40DarthCluepastebin the relevant lines from your conf
05:27.55*** join/#asterisk Twister (Twister@216.30.232.108)
05:28.32opus_thanks Darth
05:30.15xtrvdDarthClue: http://pastebin.com/320823
05:31.07DarthCluextrvd: you need to fix the line in your extensions file...you are missing a closing )
05:31.58xtrvd*sigh*
05:32.00xtrvdThanks,
05:32.35xtrvdIt's the time of night that's getting to me,
05:32.42xtrvdone more of these slip ups and it's time to sleep
05:32.47*** join/#asterisk pressure_man (~pressure_@ip-202-37-228-1.internet.co.nz)
05:32.48xtrvdOr get more Mountain Dew,
05:33.27*** join/#asterisk oej (~oej@apollo.webway.se)
05:33.35pressure_mani'm seeing some weird behaviour with the wildcard pattern match in extensions.conf. it only appears to match pass one extra digit to the macro
05:33.36xtrvdSee, you had the upper hand!
05:33.39pressure_manis that normal?
05:34.10DarthCluextrvd: yeah well, it just means that i've got a few hours of work ahead of me.
05:34.28*** join/#asterisk tengulre (~tengulre@61.185.238.166)
05:37.50file[laptop]goodnight all
05:38.14pressure_mananyone?
05:38.33pressure_manit appears that the pattern is being matched, but the tail of the pattern is being discarded and not passed to the macro
05:38.49opus__XNNNNN.
05:39.18pressure_manthat only passes 7 digits to the macro
05:39.29pressure_manit matches any number of digits, but only passes 7 digits
05:39.37pressure_manwell, matches 6+ digits
05:39.57pressure_mantake international numbers for instance - they can be variable length
05:40.07Qwellpressure_man: how are you calling the macro exactly?
05:40.24Qwellsay "Hi" when you come in, btw.  The powers that be get mad
05:40.39pressure_manok.
05:40.41pressure_manexten => _100.,1,Dial(SIP/${EXTEN}@cisco-out)
05:40.50pressure_man(my int'l prefix is 00)
05:41.13pressure_man(and '1' for an external line)
05:41.15Qwelland where does it call the macro?
05:41.29pressure_manumm actually that one isn't calling a macro
05:41.43pressure_manbut ${EXTEN} doesn't contain the full string that matched the pattern
05:41.59QwellCheck your digittimeout.  Its probably timing out before the user is done typing
05:42.05Qwells/typing/entering digits/
05:42.20pressure_manthis is even from a softphone though, which presumably sends all digits at once
05:43.09QwellI think that would be a true statement
05:43.10*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
05:43.17Kizmetpressure_man, exten => _100.,1,Dial(SIP/${EXTEN:1}@cisco-out)
05:43.20QwellIs there some other match it might be hitting?
05:43.33Kizmetif im right. as you dont want the 1 to be sent along too ?
05:43.48pressure_manKizmet: the cisco needs the '1' - i know, it's weird - integrating with an existing system
05:43.58Kizmetpressure_man, :/
05:44.14pressure_manhmm, sorry guys, i just tried from a softphone again and it worked. maybe it is the digit timeout
05:44.22Kizmetpressure_man, exten => _100.,1,Dial(SIP/cisco-out/${EXTEN},25,r)
05:44.41Kizmethrmm.
05:44.59pressure_manwhat's the default digit timeout, and where is it set?
05:45.26Kizmetpressure_man, you need to set it.
05:45.30Kizmetas far as i know
05:46.01Kizmetexten => _100.,1,Dial(SIP/${EXTEN}@cisco-out)
05:46.07Kizmetwhoops
05:46.11Kizmetexten => s,4,ResponseTimeout(10)
05:46.24Kizmetis what i have in my configs for a IVR menu
05:46.30pressure_manexten => _100.,1,Dial(SIP/cisco-out/${EXTEN},25,r) will just let it ring for 25 seconds
05:46.36pressure_manhas nothing to do with digit timeout
05:46.57pressure_manthe phone has a 'no key' timeout set at 4s, but i'm dialling faster than that
05:46.59Qwellpressure_man: after you said its a softphone, I don't think thats the problem at all
05:47.28pressure_manit works from a softphone, but not from a hardphone
05:47.43pressure_manlet me try and dial real quick on the hardphone
05:48.00*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
05:48.34*** join/#asterisk |nix (~inix@218.208.24.248)
05:48.47pressure_manok found the problem
05:48.54Kizmetpressure_man, ?
05:48.57pressure_manit's working from some hardphones, not others.
05:49.07pressure_manthe one it isn't working from are trying to do early dial
05:49.40pressure_manand of course, when 100. is matched, asterisk responds that the pattern matched (then the phone thinks the rest of the digits are menu keys)
05:49.44Qwellmust be something in the phones dialplan
05:49.58pressure_manthe phones don't have dialplans - they grandstream
05:50.53xtrvdI have a quick question... can I have an extension continue a priority list even after I have hung up?
05:50.54pressure_manjust going to try disabling early dial
05:51.08QwellI know just about nothing about SIP hardphones.  Don't all of them have some sort of dialplan?
05:51.37Qwellxtrvd: show application dial
05:51.45Qwelltells you if you can or not
05:51.57xtrvdThanks, =)
05:52.56pressure_manyeah, worked fine without early dial
05:55.35pressure_mancisco phones support a dialplan, which is basically early dial on steroids
05:55.54pressure_manit's all about getting rid of the painful delay at the end of the number before the call connects
05:56.11opus_i think what you are talking about is a dialplan mask, which is how many digits should the phone wait before automatically contacting asterisk, or the cisco voip system, to do with the call
05:56.19xtrvdQwell: it says that 'g' will extend the call to the next context even if a hangup has occured, where do I put this 'g'?
05:56.29pressure_manyep
05:56.30Qwellxtrvd: in the dial command
05:56.38QwellDial(something/blah@blah,g)
05:56.38opus_polycom has this marked as 'dialplan' as well, its confusing.
05:57.01xtrvdI'm trying to figure out where in the dial command to make the syntax correct, but I'll find it.
05:57.05pressure_mani'll refer to it as a dialplan mask in future then. i'll sound smarter than i really am.
05:57.55pressure_manthanks guys... more code to write, so i'm bailing outta here. c u.
05:58.00*** part/#asterisk pressure_man (~pressure_@ip-202-37-228-1.internet.co.nz)
05:59.50harryvvqwell
05:59.56harryvvi have the same phone
06:00.09Qwellharryvv: which?
06:00.12harryvv500
06:00.28QwellI don't have a polycom..
06:00.33Qwellor, whatever that is :p
06:01.31harryvvwas talking to opus
06:01.31Qwellyeah, Polycom, right?
06:01.38harryvvyup
06:02.27*** join/#asterisk znoG (~gs@200.115.216.109)
06:04.39Strom_Care any of you having jitter problems with asterlink?
06:10.49*** join/#asterisk kimera (~kimera@host162-176.pool8256.interbusiness.it)
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06:13.15*** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc)
06:13.45DarthClueStrom_C: what?  um, no, what's your number?
06:15.03|nixanyone using polycom knows if it has any settings for disconnections?
06:15.19|nixi'm facing some disconnection problems and i've narrowed it down to the phone
06:15.22|nixsipura works perfect..
06:19.43*** join/#asterisk daniel101 (~daniel101@dsl15-088.express.oricom.ca)
06:20.11daniel101Does anyone have heard of cisco call manager ? .. Just wondering what it does
06:21.18JerJerlol
06:23.32DarthClueMorning JerJer.  How are you today?
06:23.33Qwellwow
06:24.01JerJerpissed off
06:27.36kimeraHi to all...does anyone has experience with siemens IP phones with * ?
06:32.16hardwireyou rang?
06:33.23*** part/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)
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06:51.55fitzelMoika
06:52.11fitzelAnyone here with some practical experience for a softphone on a windows mobile PDA with wifi/wlan?
06:53.29*** join/#asterisk jtodd (~jtodd@ti.fox-den.com)
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06:59.35*** part/#asterisk umesh (~u@203.200.50.230)
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07:07.13Delta34anybody using rawplayer?
07:09.58*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
07:12.44_gigi_im looking for some software for analize RTP streams (delay, jitter, delta, and loss), someone know something ? :)
07:17.11xtrvdWhat's the best way to cause a delay in the creation of call files?  Can I tell bash to wait before it processes the script?
07:18.38Juggiextrvd, set the date on the file to the future
07:18.44Juggieit wont process until that time
07:19.18xtrvdHmm, interesting method; would it be easier to impliment a 'sleep' command for a few seconds?
07:19.28xtrvdOr rather, how does one set the date ahead?
07:19.45xtrvdI'm not familiar with how to do either... At least I'm not familiar yet. I'm learning at a fast pace.
07:23.50xtrvdWell, that solves it. I put  a 'sleep 10' in my bash script,
07:25.56*** join/#asterisk newmember (user@S010600036d1139fb.cg.shawcable.net)
07:27.56*** part/#asterisk newmember (user@S010600036d1139fb.cg.shawcable.net)
07:31.35*** join/#asterisk CyberSword (~c@cablea0mle.cybercable.net.mx)
07:34.31*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
07:35.28*** join/#asterisk _omer (dfsdf@203.215.180.254)
07:35.35_omerhi
07:36.10_omer"Sip show inuse" ...doesnt work....how do I know about the busy channels/peers?
07:37.14kaldemarshow channels
07:37.39_omersip show channels <--- shows the incoming calls as well....
07:41.27kaldemarone channel being open doesn't necessarily mean the phone is busy though.
07:43.39_omerI think in latest version of asterisk ...SIP show inuse doesnt work
07:47.26DarthClue_omer: what do you get with sip show inuse?
07:47.52_omer* User name               In use          Limit
07:47.52_omer* Peer name               In use          Limit
07:47.54kaldemar_omer: if you only give 'sip show inuse', it shows the peers/users with group limits.
07:47.55_omerthat's it ...
07:48.04kaldemartry 'sip show inuse all'.
07:48.08_omerwhen I do sip show inuse all
07:48.18DarthClue_omer: what do you get with sip show peers?
07:48.28kaldemari have cvs head from last week, and it works.
07:48.37_omerit shows   0  ZERO with all channels.....even with the channels who are taking calls..
07:49.00_omerlist of peers
07:49.23*** join/#asterisk pa (~Paolo@pa.user)
07:49.41DarthClue_omer: hold new / old is your cvs-head?
07:49.48DarthClues/hold/how/g
07:50.16_omerwhat's that?
07:50.35DarthClue_omer: how new / old is your cvs-head?
07:51.01_omerMYLINUX*CLI> show version
07:51.01_omerAsterisk CVS-HEAD-06/02/05-08:36:40 built by root@MYLINUX on a i686 running Linux
07:51.49_omerI have a asterisk box that I installed in december 2004.....which works at SIP SHOW INUSE
07:51.50DarthClueI'm using it from 7/21 and it works.
07:52.01DarthCluesounds like you have a bad version.
07:52.12_omerhow do I update the version?
07:52.14*** join/#asterisk Mw3 (mw3@daisy.chains.ch)
07:53.50_omer~update
07:53.50jbotmethinks update is dselect update, grabs the Packages.gz files from their sources and refreshes the available packages. Use before an apt-get upgrade, or you can use apt-get update instead of dselect update
07:54.03_omerwoof
07:54.13DarthClue_omer: take a look at the wiki ...
07:54.24*** join/#asterisk Aze` (~aze@85.18.136.114)
07:54.35_omerok ...
07:54.41_omerthanks..
07:54.42Aze`Anyone use cisco 7960 sip ?
07:54.59DarthClueshould just be a cvs update, but i'm up way past my bedtime so i recommend going to the wiki
07:55.14DarthClueAze`: somebody does, but not me.
07:56.31_omer:)
07:57.37Aze`tnx DarthClue
08:10.30*** join/#asterisk montag___ (~montag@f.desys.it)
08:11.40montag___hi, it's possible to use 2 tdm 400 with 4 FXO port for each one on a single asterisk
08:11.41montag___?
08:16.06Math`montag___: hmm you could use 1 asterisk for your 4 Zap channels and connect to your other asterisk using IAX
08:16.09xtrvdI am aware of the methods for using Dial() with Zap and SIP phones, but how does one use Dial with outbound calls to a VOIP provider?
08:16.27Aze`Anyone know how my cisco 7960 accept only 2 line registration ? i need a license ?
08:16.59Math`xtrvd: is your provider SIP ?
08:17.05xtrvdIAX2
08:17.18xtrvdI'm setting a callfile right now and I can only figure out how to use SIP and Zap,
08:17.36Math`theres 2 ways u can make calls using IAX2
08:17.48xtrvdI want my call file to send outbound to a PSTN number...
08:17.55Math`1st: Dial(IAX2/user:passwd@provider/extension)
08:18.03xtrvdThat's not the one I want to use... =)
08:18.07Math`2nd: you define the peer in iax2.conf and use IAX2/provider_entry/extension
08:18.13Math`er in iax.conf
08:18.25xtrvdAhh, thank you. =)
08:18.27xtrvdI'll look in there
08:18.40Math`ex... I got: exten => _011.,1,Dial(IAX2/voipjet/${EXTEN})
08:20.11xtrvdAnd your 'voipjet' is listed in your iax.conf as a seperate entry?
08:20.25Math`yeah I just pasted it in privmsg not to flood the channel
08:20.30xtrvd=)
08:20.38xtrvdThanks a bunch,
08:23.08*** join/#asterisk smeevil (~smeevil@gremesh1.demon.nl)
08:32.22*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
08:32.22*** topic/#asterisk is Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted
08:32.59smeevilflorz: hmm can i paste 4 line to show what i use to test ?
08:33.48florzI guess so, there isn't much going on ATM anyway.
08:34.32smeevil[from-zap]
08:34.32smeevilexten => s,1,Answer;
08:34.32smeevilexten => s,2,SayDigits(${EXTEN});
08:34.32smeevilexten => s,3,Hangup;
08:35.10smeevileffect is, pickup , say nothing , hangup
08:35.39Jas_Williamssmeevil: what is the line type ?
08:36.49florzsmeevil: Well, of course, you can't do that in the s extension
08:37.03smeevilJas_Williams: is this what you mean ?  switchtype = euroisdn, signalling = bri_cpe_ptmp
08:37.21smeevilzaphfc based isdn cards
08:37.25Jas_Williamssmeevil: Correct
08:38.09smeevilflorz: should i use _. then ?
08:38.29florzsmeevil: smeevil Well, anything that matches the actual numbers
08:39.05florzsmeevil: Dunno, though, whether a . works in that case without much delay
08:39.10Jas_Williamssmeevil: it sounds like you are not being passed a number try posting the output of a pri debug span 1
08:39.12smeevilflorz: hmm but would i not be matching on the caller's number then in stead of the number the caller dialed
08:39.43florzJas_Williams: Nope, he just has immediate=yes in the channel config
08:39.59*** join/#asterisk bikokola (~bikola@211.27.37.248)
08:40.05florzsmeevil: Nope, the dialplan (of course) always matches on the destination number
08:40.10smeevilflorz: thats correct, should that not be there ?
08:40.33smeevilokies ! just a moment then
08:40.39florzsmeevil: Well, would the option exist if it was not meant to be used?
08:42.45bikokolahey guys, i just intalled linux and downloaded asterisk (not including zaptel and the other one),i done this to learn how to use asterisk, but i dont know exactly what do to with the tarball file on my desktop
08:43.29florzbikokola: unpacking would be a good first step, I guess
08:44.26bikokoladone that so i have a folder of the unpacked contents on the desktop
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08:44.35*** part/#asterisk Wardead (~ShellPad@g220110.upc-g.chello.nl)
08:45.36*** part/#asterisk Robot_ (~Robot_@84.47.4.242)
08:46.23bikokolaso now what, do i have to put the unpacked file in the etc folder
08:47.08Jas_Williamsbikokola: follow the steps here http://www.asterisk.org/index.php?menu=download
08:47.23bikokolathanks mate
08:47.33xtrvdMath`: for some reason, my iax.conf method wouldn't work, so I just entered the loginname:pass@provider the long way and it works fine.
08:47.36xtrvdThanks for your help
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08:56.52smeevilflorz: thank you for your time, the change of immediate=yes to no did the trick.
08:56.59smeevilJas_Williams: thanks as well.
09:01.23*** join/#asterisk Assid (~assid@203.115.64.60)
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09:08.52*** join/#asterisk clive- (~pirch@rndf-146-56-76.telkomadsl.co.za)
09:09.24bikokolahey guys, im really bad in linux console, is there another way to install
09:10.25bikokolaasterisk im refering to
09:10.58clive-pay someone to do it for you...lol
09:11.14florzbikokola: Probably none that makes sense if cou really wanna use it.
09:11.56florzbikokola: If it's just for some simple standard scenario, maybe
09:12.29Assidbikokola: just use AAH
09:12.29bikokolanah, i can do the scriptinhg, but not the first time install
09:12.45bikokolawhat's aah
09:14.20toothey - can anyone confirm to me the correct zapata.conf entries for uk caller id ? i've spent weeks and am getting nowhere. My current zapata.conf is at http://pastebin.ca/18521
09:14.27*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
09:14.53tootbikokola - make clean && make install in the various directories - its not tooo bad :)
09:16.07Zeeekbikokola here's a full install guide: http://automated.it/guidetoasterisk.htm
09:16.14Jas_Williamstoot: what card are you using ?
09:16.15Zeeekfollow it and all will be fine
09:16.19tootTDM400P
09:16.33tootand caller id is working fine on the line
09:17.30tooti'm just unsure of that the offical config should be - as i see people with usecallerid=uk/yes and cidstart=history/hist/ring/bell, etc
09:18.12*** join/#asterisk tuxinator_linuxM (~tuxinator@ip68-109-146-168.ph.ph.cox.net)
09:18.35Jas_Williamscidstart should = polarity
09:19.00Jas_Williamstoot: history is when using a patch x100p
09:19.22Jas_Williamswhich does not detect polarity reversal
09:19.43Jas_Williamsuse callerid = yes
09:21.19tootwhen i usecallerid=yes i get - WARNING[12328]: chan_zap.c:3707 zt_handle_event: Didn't finish Caller-ID spill.  Cancelling.
09:21.45tootwhen i use usecallerid=uk i don't - but i thought that might be cause it equated to no :)
09:26.46tootreset to polarity but no luck. hmm
09:27.23Zeeektoot do you have a wait() before answering?
09:27.46*** join/#asterisk Piranha- (piranha@209.89.80.129)
09:28.49tootnope? should i?
09:29.08Zeeeksometimes it's needed: wait(1) or (2)
09:29.23Piranha-quick question before I get started here... I have voip service setup (unlimited calling), and I have 2 different houses that are going to be using this line... Would asterisk be able to ask the caller which voip phone connection to ring?
09:29.30Zeeeksomething to try: start with 3 to be sure, but three is too long. Also, much said on the mailing list
09:29.55ZeeekPiranha- sure
09:30.15Piranha-like pressing 1, to a certain box, pressing 2 to the other box, and so forth
09:30.22ZeeekYES
09:30.26Piranha-;P
09:30.36ZeeekInteractive Voice Response (IVR) menus
09:30.36Zeeekhttp://users.pandora.be/Asterisk-PBX/IVR.htm
09:30.36Zeeekhttp://www.voip-info.org/wiki-Asterisk+tips+ivr+menu
09:30.59Piranha-thanks.. didnt know what to search for
09:31.08Zeeekthat can be a problem
09:31.20Piranha-yeah :) normally I don't ask for help
09:31.29Zeeeknow, what do you know about smarty templates?
09:31.44Piranha-I know a whole lot of 0
09:31.57Zeeeksame here. I may have to actually read a little :)
09:32.03Piranha-hehe
09:32.56tootadded in the wait 3 - still getting the Didn't finish caller id spill issue
09:32.56tootif relevant i have also upgraded as recommended to the latest zaptel src
09:33.10Piranha-yeh im movin away here and I finally got my parents to subscribe to a voip connection.. the provider locks their voip box (normal) and charges you to use THEIR software to use it on the computer... so, I unlocked it, looked at the config file and got the login/pass :) unlimited clients hehe
09:33.22Zeeektoot ok well then remove the wait, it was just a thought
09:33.41Piranha-just ordered up the SPA-2002 and awaiting arrival
09:34.06tootna - worth a shot - will leave it in and remove once i figure out what the hells going on :)
09:34.16tootwas all working lovely on my x100p :)
09:34.31Zeeekah this is TDM400P issue?
09:34.38tootyep
09:34.46ZeeekI hate international non-standards
09:34.51toot:)
09:35.13Zeeekbut this HAS to have been discussed a lot and dealt with. YOu searched the list?
09:35.15tooti mean it may well be my config - but i had it working on x100p so i know the line,etc,is all fine
09:35.31ZeeekI often see UK CallerID
09:35.31tooti get all manner of answers when searching on the list :D
09:35.39Zeeekyeah that's the problem there
09:35.51Zeeekwhy not call or email digium?
09:35.51*** join/#asterisk Romik_ (~romik_@212.143.5.146)
09:35.55tootthe definitive guide will be a winner :)
09:36.15tootyeah think i will - tried avoiding it as i guess it was me but think i have covered every permutation at this stage :)
09:36.15Zeeekdigium should do a paper on callerid in various countries
09:36.59RaYmAn-BxIt is rather stupid how many callerid "standards" there are...Some with just minor differences that make them slightly incompatible and stuff
09:37.01Piranha-caller id is soo worthless
09:37.01Piranha-;)
09:37.20Piranha-seems rather stupid
09:37.24ZeeekPiranha- it's important in business where people don't fuck with it
09:37.37Piranha-:P just joking.. I dont know if I could live without it
09:37.38InfraRedcallerID was developed on local level
09:37.48InfraRedbefore peopoe thought about making calls over the internet
09:37.55Zeeekso was cocaine, now it's universal
09:37.55InfraRedlocal telecos with local standards
09:37.59InfraRednothing new
09:38.06Piranha-well all drugs pretty well were
09:38.07Piranha-hehe
09:38.09InfraRedyes Zeeek
09:38.17Zeeekrap music
09:38.26Piranha-true
09:38.30Zeeekwho knows about smarty templates or serendipity ?
09:38.50tooti know plenty about smarty
09:39.08toottis the best/most flexible template engine i have used
09:39.21Zeeekso in a .tpl, how do you know what variables are available?
09:39.31tootyou don't :)
09:39.54toot{if !empty($VARNAME)}{$VARNAME}{/if} or just {$VARNAME}
09:39.55Zeeekfor example  {foreach from=$dategroup.entries item="entry"}
09:40.11Zeeekwhere would I find thelist used?
09:40.19Zeeekwhere is dategroup defined?
09:40.27tootthelist used?
09:41.10tootnot 100% with ya .. :)
09:41.15tootbut its early for me
09:41.49ZeeekI'm trying to figure out how to know what variables (like dategroup - maybe it's objects?) are available
09:41.49*** join/#asterisk christo (~chris@office.enovi.com)
09:41.50Piranha-same.. 3:40 am
09:41.56christomorning all
09:41.57Zeeek11:41 AM
09:42.16Zeeekobviously these are in the calling code
09:42.23tootto be annoying - thats the part of your php code - not your template to work out normally
09:42.25ZeeekI though maybe there was a naming convention
09:42.36Zeeekyeah
09:42.49Zeeekok, I'll have to figure out how it's called
09:42.54tootsmarty.foreach.foreachname.....
09:43.19tootor you can 'hack' your way down the defined smarty vars and call it like php using an if defined, etc, but thats a bit naughty
09:43.22*** join/#asterisk Specky[W] (~sspecken-@pD95B0C27.dip0.t-ipconnect.de)
09:43.27*** part/#asterisk Specky[W] (~sspecken-@pD95B0C27.dip0.t-ipconnect.de)
09:43.51tootor use smarty debug console to see where it is defined if that would help (ie if it does not change each time called)
09:44.41tootright bugger this - i'm going flying (where uk caller id can't bug me)
09:44.52christoI'm sending out faxes over E1. I find that if I use many channels (say 20 or more), I suffer frame slip and the faxes come out with sections missing/squashed.  Is there any way to tell asterisk not to place a call if there are more than 15 channels in use?
09:45.42*** join/#asterisk Tili (Tili@202-133-67-21-dialup.sat.net.pk)
09:50.27pais it possible to use asterisk to do talk with 2 people? that is 3 ppl connected togheter
09:51.29Jas_WilliamsPa yes you can conference calls together or use a meetme for more people
09:51.38paoh nice :)
09:51.39Tilipa: do you mean 3 way calling
09:51.45payes, prolly
09:51.52pai dont know english term for it
09:52.38Zeeekthat may depend on the phone
09:52.48Zeeekasterisk can do conferencing though
09:52.52_gigi_im looking for some software for analize RTP streams (delay, jitter, delta, and loss), someone know something ? :)
09:52.55*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
09:53.28Tili_gigi_: ethereal can help a bit
09:53.42Tili_gigi_: there are other commercial apps otherwise I think
10:01.05ManxPower~docs
10:01.05jbothmm... 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
10:01.15Zeeekhi Manx
10:03.51ManxPower'morning zeedo
10:03.54ManxPower..er...
10:03.56ManxPower'morning Zeeek
10:04.58Zeeekit just became afternoon, 12:04PM :)
10:05.19ManxPowerit's 5am here
10:06.54dwmw2daystar appears to be up
10:10.30*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
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10:31.52kkshi all, i wan to install Areskicc2 which requiring phpagi. phpagi_v1 or phpagi v2 that i needed?
10:34.04*** join/#asterisk kks (~kks@202.73.8.130)
10:34.51kkshi all, i wan to install Areskicc2 which requiring phpagi. Which version of phpagi that i needed?
10:35.58*** join/#asterisk kks (~kks@202.73.8.130)
10:52.41*** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-7-21.w81-248.abo.wanadoo.fr)
10:54.41jimmybob46do u have to use hisax to use a fritz isdn card? or do u just use the fcpci module
11:06.08Assidumm.. if i have to forward all calls from one box to another..
11:06.15Assiddo i just dial that context?
11:06.39Assidits like this.. i have an IVR on the main box.. depending on what they choose.. i will forward the call to the other * box.
11:07.38Assidso do i just dial the context and not provide the extension ?
11:12.48kaldemaryou could for example save the called number to a variable, and then:
11:13.16*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
11:13.21kaldemarexten => 1,1,Dial(IAX2/user:secret@otherbox/${CALLEDNUMBERVARIABLE})
11:14.03Assidokay and what if i just want it as a routing?
11:14.15Assidlike all calls i want to send to the other box
11:14.16pawith iaxcomm i cant use the # button for services who requires it.. why?
11:14.22kaldemarexten => _X.,1,Dial(IAX2/user:secret@otherbox/${EXTEN})
11:14.24pawhen i press it i get a strange sound
11:15.49Assidhmm.. i gotta play with it
11:15.57Assidoh yeah.. is there a way to record a call
11:16.07Assidlike you know how some places.. there is a sales team
11:16.10Assidand they record the call
11:16.21kaldemarAssid: there is.
11:16.28Assidi know i can record an answer
11:16.41Assidbut.. if the line is picked up.. how do i have a conversation take place and have it recorded
11:17.05kaldemarAssid: give 'show application monitor' in CLI.
11:17.10*** join/#asterisk |nix (~inix@cm11.gamma116.maxonline.com.sg)
11:17.29DarthClueAssid: you can also use muxmon from http://www.pbxfreeware.com
11:18.52*** join/#asterisk manaz (~manaz@acis.garda.sk)
11:19.04Assidi want it to record all conversation of that particular extensions
11:19.23Assidautomatically
11:20.14kaldemaryou can add the monitor in your dialplan before the dial command.
11:20.35Assidokay.. before the plan
11:20.45manazhi. it is possible to use dialogics D/600JCT-2E1 hardware with asterisk ? i found it under supported hardware, but chan_dialogic isn't by default in contrib tree ..
11:21.03Assidi was wondering.. coz once the dial plan is done.. i cant really do much more
11:21.11DarthCluemanaz: under supported hardware? where at?
11:23.01manazDarthClue: http://www.asterisk.org/index.php?menu=hardware
11:24.56*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
11:25.39DarthCluemanaz: call digium and ask them.  i believe that you have to pay for the dialogic driver and i'm not sure it is even actively developed anymore.
11:28.12manazDarthClue: yes. i found some posts in mail archives, but they was so old ... so i need to know actual status of that drivers .
11:28.17pado you know voipbuster.com?
11:28.21paanyone knows it?
11:28.38pahttp://www.voipbuster.com/en/index.html
11:28.54manazDarthClue: they want to pay $15 per channel i think . but it was post from 2003 .
11:29.30Assid??? how the hell?
11:29.58Assidpa: it doesnt really work
11:30.04Assidtill now not 1 successful call
11:30.25paAssid: a friend of mine called me with it last day (from windows)
11:30.27Assideither the other party canr hear me.. or something
11:30.43Assidit has issues..
11:30.44pabut for only 1 minutes since he didnt have credit
11:30.53Assidhell i'll call you..
11:30.58Assidyou tell me if you can hear me
11:31.01paok
11:31.02Assidand i have credit
11:31.06paok
11:32.17kaldemarAssid: what's the problem?
11:32.38pado you know if can voipbuster be integrated into asterisk box?
11:32.40Assidwith? voipbuster? they cant hear me.. or i cant hear them.. or sometimes voicew crakcs..
11:32.47Assidor some crap or another..
11:32.50Assidi barely use it
11:32.52padoes it use SIP or other standrard protocols?
11:33.05Assidi think they use iax2.. but you cant connect *
11:33.13Assidor atleast i dont think u can
11:33.20paoh bad :(
11:33.26pait would be great..
11:34.26Assidnot really
11:34.31Assidi cant make a call yet ritght now
11:34.34Assidi cant call myself
11:34.42pathat would be the cheapest way to call not mobile phones for free...
11:34.53pait is free :)
11:35.07Assidwho cares.. doesnt work 99% of the time
11:35.35pathe friend of mine told me he seldom uses it and it works almost everytimes
11:35.44Assidweird
11:36.08Assidstill cant makke a cal today
11:36.16Assid"The other party disconnected"
11:36.17patry to call me..
11:36.18Assidalways the same error
11:36.26gambolputtytrying to use realtime with cvs * and sip show peers doesn't show my extensions.
11:37.03kaldemargambolputty: sip show peers shouldn't show your extensions.
11:37.24gambolputtyit has before
11:37.35gambolputtywith a prior cvs version of *
11:37.51gambolputtyend of june
11:38.16gambolputtymy phones don't register
11:38.23gambolputtycvs was just downloaded and compiled
11:39.49kaldemaris your sip.conf ok? bind address and user definitions?
11:40.36gambolputtyworked before
11:40.58gambolputtydo you use realtime and cvs?
11:41.25Assidhrmm how much load do you think it will take to monitor a call and encode at mp3?
11:42.49*** join/#asterisk andrebarbosa (~andrebarb@gate.criticalsoftware.com)
11:42.49Assidsay 5 calls being transcoded/monitored?
11:43.18andrebarbosaei guys
11:43.42andrebarbosacan you tell me if it's possible to connect asterisk to voipbuster?
11:43.53Assidcant
11:44.02Assidi think
11:44.17andrebarbosa:(
11:44.23ManxPowerandrebarbosa: What protocol does voipbuster use?
11:44.40andrebarbosai search the list and voip.info, and i didnt found anything
11:44.58ManxPowerwell, unless you know the protocol the service uses, we can't help you.
11:45.18andrebarbosalet me search
11:46.06andrebarbosasip
11:46.21andrebarbosasip.voipbuster.com
11:46.24ManxPowerthen you should be able to use it with Asterisk (unless they try to stop you)
11:46.30AssidWRONG
11:46.33andrebarbosahum
11:46.53Zeeekit works with asterisk
11:46.59*** join/#asterisk njan (~james@james.user)
11:47.04andrebarbosaso i connect like i used to connect to gossiptel?
11:47.10Assidsays iax2
11:47.11njanHas anyone here tried runnign asterisk on adamantix, either from source or from the binaries?
11:47.25*** join/#asterisk matr24ct (~lkj@p54996E97.dip.t-dialin.net)
11:47.35matr24cthi
11:47.46njanhi, matr24ct
11:48.32matr24cti have a question about compiling asterisk on a Via Platform
11:48.40matr24ctVersion 1.0.8
11:49.06matr24ctchanged PROC=i586 in Makefile
11:49.10andrebarbosaya
11:49.13andrebarbosait supports iax too
11:49.43matr24ctwell, the compiler bombs out with n internal compiler error
11:49.52matr24ctwhen compiling chan_sip
11:50.51matr24ctsome other guy experienced the same problem, he posted to a mailing list.  but nobody replied to him
11:51.06matr24cthe also had a Via Epia main board
11:51.16matr24ctalso running suse 9.1
11:51.30matr24ctanybody got an idea?
11:52.21joshpbxyeah paste in
11:52.23joshpbx~pastebin
11:52.23jbotsomebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca
11:52.27matr24ct*-1.0.6 works perfectly
11:52.46*** join/#asterisk darkskiez (~darkskiez@194.247.78.146)
11:52.48joshpbxand cvs?
11:53.13andrebarbosajust: register = username@sip.voipbuster.com
11:53.30andrebarbosa?
11:53.31andrebarbosa:D
11:53.37matr24ctnot tried - whats the case with CVS? what version is it and is it stable
11:53.49ManxPowerYou usually need a password when you register
11:53.50*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
11:53.58andrebarbosaya
11:54.06andrebarbosaok
11:54.08andrebarbosalet me try it
11:54.11andrebarbosai tell you news
11:54.24*** join/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
11:55.55*** join/#asterisk syslod (~yurplsl@65.114.15.70)
11:57.29*** join/#asterisk Romik_ (~romik_@212.143.5.146)
11:58.15QbYi need to adjust the volume (increase) for outgoing SIP calls..  i have found info about txgain for zaptel but is there anything for SIP?
11:59.18Assidyou really dont need to..
11:59.19*** join/#asterisk nounoursfr (~Direct@stardust.noc.frontier.fr)
11:59.26Assid]since sip is generalyl for hardware/ software phones
11:59.29nounoursfrhello
11:59.31Assidip phones have volume control
11:59.38Assidand software.. you can increase /decrease anwyasy
12:00.18QbYAssid..  Well here's the problem, we have Broadvoice..  If you call our regular number volume is GREAT..  if you call the 1-800# (which is pointed to the regular number) the call is hard to hear..
12:00.27QbYand its a customer support number and people are complaining
12:01.23nounoursfrwho use a eyebeam video support ?witch nat ?
12:01.48*** join/#asterisk skiold (~userid@84-121-68-212.onocable.ono.com)
12:01.54AssidQbY: increase it from the hardware device
12:02.01Assidif its a sip
12:02.04*** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com)
12:02.57QbYAssid..  Our customers are overall dumb and would rather bitch than do anything proactive..  So I just need to boost the volume for that sip connection..
12:03.11AssidQbY: increase it from the hardware device
12:03.36matr24ctWhat gcc compiler version does the newest asterisk version require
12:03.40QbYassid.. their phone?  we have no hardware device except for the actual asterisk server itself.
12:03.55matr24ctWhat gcc compiler version does the newest asterisk version require?
12:03.57bikokolaguys, i get a error, when i type make in the asterisk directory in conosle, it says "c compiler cannot create executible"
12:03.57Assidyes the phome
12:04.17Assidbrb
12:04.23lters_any idea where we could get a ccm-to-7960 tcpdump to show how to turn 7914 lights on/off?
12:06.17bikokolaany1 else get a error, when they type "make" in the asterisk directory in conosle, saying "c compiler cannot create executible"
12:07.55joshpbxpaste u errors in pastebin
12:08.21bikokolaim installing on diff pc
12:08.32bikokolano net atm on that pc
12:09.28bikokolado i have to install devellpment tools with my copy of fedora
12:09.45bikokolai unticked the development toools during install
12:09.45*** join/#asterisk RandomAndy (~randomand@adsl-63-207-12-192.dsl.snfc21.pacbell.net)
12:09.49DarthCluebikokola: yes, you need dev tools
12:10.02bikokolaoh ok, thanks
12:12.37*** join/#asterisk MikeJ__ (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
12:21.26*** join/#asterisk bjohnson_ (~bjohnson@i216-58-62-102.igs.net)
12:22.57InfraRedis there asterisk-addon package in debian ?
12:31.14*** join/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it)
12:33.12lters_InfraRed, don't know, but you can do cvs co asterisk-addon :)
12:33.30lters_and get the current/latest addons
12:33.43lters_InfraRed, did you update the wiki?
12:33.53InfraRedno
12:33.55InfraReddidnt get the chance
12:33.58InfraRedbeen ill
12:34.05lters_sorry to hear.
12:34.09InfraRedand i have to rescue this server now
12:34.18InfraRedit's been down2 days
12:34.22lters_did u try chan_sccp ?
12:35.00lters_apt-cache search asterisk
12:36.10*** join/#asterisk Vandien (~stephan@p50904201.dip.t-dialin.net)
12:37.56Vandienhi, does anyone know where or how i can find out what ports freenode irc servers use? i got no voice in #freenode... dunno where to ask -.-
12:38.18Vandienand i know this channel cause it already helped me several times ;)
12:38.38InfraRedlters_: it's not there, looks like i have to compile it aginast the source
12:38.41*** join/#asterisk zotz (~zotz@24.231.36.100)
12:40.42*** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net)
12:41.50lters_InfraRed, yeah looks like u are right.
12:44.16InfraRedheh
12:44.33InfraRedi just had to copy asterisk.h from the src to the adodn dir to get it ti compioe
12:44.36InfraRedcompile
12:46.37MikeJ__InfraRed, if you make install in asterisk, it should move that file into usr\include, so you don;t need to do that
12:46.39*** join/#asterisk RandomAndy (~randomand@adsl-63-207-12-192.dsl.snfc21.pacbell.net)
12:48.47Kattymew
12:49.22oejCat's around. Watch out all mice in the channel!
12:49.39Darwin35des
12:49.43*** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3986041.sympatico.ca)
12:49.44Darwin35ok this is killing me
12:49.55Kattyoej: ...
12:50.01Kattyoej: you have insaned.
12:51.35QbYcan someone make a test call for me?
12:52.52Kattyso much sleepy this morning.
12:53.19Katty:<
12:53.39Kattyi cut my paw last night
12:53.41Kattywith a gensu
12:53.51DarthCluei'll sleep next week.  14 road trip will give me plenty of time to rest.
12:53.52*** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net)
12:53.53lathos42woo, almost have my boss talked into Cluecon
12:53.59Hmmhesayswell i'm running back and forth from here to south carolina
12:54.06KattyDarthClue: k
12:54.25newlIt slices, it dices, it chops Kattys in a flash!  If you're one of the first 100 callers, you'll receive...
12:54.33DarthClueKatty: that won't get you out of cluecon.  we are still coming to get you.
12:54.34Katty:<
12:54.45KattyDarthClue: i figured brian would be driving (=
12:55.01lathos42DarthClue: Is there internet access in every room at the Best Western?
12:55.01Dovidhey
12:55.11Dovidanyone know if CID spoofing is legal ?
12:55.30DarthCluelathos42: i doubt internet access will be an issue, but bkw would be the best person to answer that.
12:55.39DarthClueDovid: in most cases, yes.
12:55.47Dovidok
12:55.53Dovidin what cases would it not be ?
12:56.08Hmmhesaysahh cluecon
12:56.14DarthClueDovid: I am not a lawyer, but the FCC would be best able to answer that question.
12:56.16Dovidlooks like it can be a big biz. i can make 8 cents a min
12:56.20Dovidkk
12:56.21Dovidthanks
12:56.41Hmmhesays:) morning
12:56.45christoWhere can I find the different reason codes for a call failing? I am seeing 'reason 1', 'reason 5' and 'reason 8' and would like to know their meaning
12:56.52Kattyit is /not/ morning! i protest!
12:57.00lathos42bkw_ : You around this morning? :)
12:57.03Hmmhesayswell maybe not in germany
12:57.12KattyHmmhesays: then it is germany.
12:57.14KattyHmmhesays: kthxnaptime
12:57.15DarthCluelathos42: it's not even 9am yet, let the man sleep.
12:57.23Hmmhesays:)
12:57.30Kattyif only :<
12:57.40Hmmhesaysas long as I don't get stuck in a room with some really strange person at cluecon
12:57.45DarthClueKatty: unfortunately, it is morning.
12:57.45lathos42DarthClue: Oh c'mon, I had to be here at work at 8, everyone should be up :)
12:58.10KattyHmmhesays: i'm a really strange person :P
12:58.18Hmmhesaysruh roh
12:58.22DarthCluelathos42: i've been up since midnight, i technically don't have to be working till 9, but that doesn't stop me.  of course, bkw was up while i was passed out.
12:58.28Kattyexactly.
12:58.37Hmmhesays<chuckle>
12:58.38KattyDarthClue: are they going to double us up on rooms?
12:58.42DarthClueHmmhesays: what's your name again?  i'm pretty sure you are in a room by yourself.
12:58.52Hmmhesaysmatt williams
12:59.02*** join/#asterisk greg_work (~greg@d221-73-237.commercial.cgocable.net)
12:59.09DarthClueKatty: for those persons who paid full price, it is most likely the room all to yourself, but bkw has those details.
12:59.20KattyDarthClue: did i pay full price?
12:59.22Hmmhesaysthat reminds me I owe you guys some money
12:59.41Kattyactually, i guess that counts as full price.
12:59.55KattyDarthClue: i'll pester brian later (=
13:00.02*** join/#asterisk indra (~indra_wat@microinfo.rain.fr)
13:00.25indra'gday folks
13:00.53*** join/#asterisk seong (~seong@218.111.18.207)
13:01.08indrahave a question about SIP register/subscribe/notify
13:01.10Hmmhesaysok off to work
13:01.15indraZeeek : hi :)
13:01.52indrai managed to send a REGISTER (replied by Trying and Ok by Asterisk)
13:01.56Zeeekhello indra
13:01.57DarthClueKatty: yeah, you should be in your own room.  I'm pretty sure that they are doing single rooms for full price people...and Hmmhesays, you really need to login and make that payment or we just might put you in a room with really strange people.
13:02.00*** join/#asterisk matr24ct (~lkj@p549959D1.dip.t-dialin.net)
13:02.03*** join/#asterisk grimse (~grimse@p5481D528.dip.t-dialin.net)
13:02.10indrabut next, when i send a SUBSCRIBE, asterisk answers Not found ?!?
13:02.12*** join/#asterisk jwitte (~jwitte@port-212-202-101-206.static.qsc.de)
13:02.27KattyHmmhesays: if you get stuck with a Really Strange People, you can knock on my door :P
13:02.32matr24ctAnyone know how to compile the capi channels for the current Asterisk version from CVS
13:02.44matr24ctCapi channel version 0.3.5
13:02.56indraZeeek : no summer holidays? Almost every French has left their office by now :)
13:03.01*** part/#asterisk jwitte (~jwitte@port-212-202-101-206.static.qsc.de)
13:04.44DarthCluelathos42: $650 gets you the conference and hotel for the 2,3, and 4th.  If you hurry (as in before noon CST), I'm pretty sure we could still get you a spot without problems.
13:04.52Zeeeka) I'm only 1/2 FRench sort of, and b) most leave in August
13:05.15lathos42DarthClue:  Ok, i'll work harder on my boss then
13:06.34KattyHmm, a local reporter wants to do a story on a teen vegan.
13:06.39KattyToo bad I'm too old :<
13:06.53*** part/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it)
13:06.58Zawi'm sponsoring a vegan
13:07.00DarthClueit's supposed to rain and get cold tonight!  instead of being near 100, it's only gonna get up to about 84 tomorrow!
13:07.07KattyZaw: so you said. many times.
13:07.35matr24cthas no-one got experience with the capi channels from junghanns?
13:07.36Zawi would sponsor you, but it's not as much fun if you never actually eat lunch with the sponsored vegan
13:07.58Katty...
13:08.05KattyUhm, no (=
13:08.07Kattykthxbi
13:08.10jake1932matr24ct: I'm workingo n it now
13:08.23indraZeeek: half of the people here left already since last week
13:08.34jake1932matr24ct: I got it compiled
13:08.37*** join/#asterisk tobiasWolf (~konversat@195.162.255.10)
13:08.56tobiasWolfhi all
13:09.42DarthClueI'm sorry, all is not available to take your call now.  If you would like to leave a message, please do so after the beep...BEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEP
13:10.21matr24ctok..
13:10.25tobiasWolfdoes anybody experiences problems while unloading the wcte11xp kernel module ?? i just did this and the machine froze totally.
13:10.32jake1932matr24ct: what distro?
13:10.48matr24ctsuse 9.1
13:11.11jake1932matr24ct: ok - i used debian - should be much easier with SUSE
13:11.57jake1932matr24ct: are you getting errors on the compiling?
13:12.20*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
13:14.10*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:14.31*** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au)
13:15.12*** join/#asterisk salvini_fs (~felipesal@200165218124.user.veloxzone.com.br)
13:15.29*** join/#asterisk hotgrits (~hotgrits@192.160.238.156)
13:15.55Hmmhesayseverything has caffiene
13:15.56webmancan anyone explain why a roundrobin queue with agents added via agentcallbacklogin works the same as rrmemory, instead of roundrobin? (or, I suppose it could be random, but it seems pretty consistent so far)
13:15.58*** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net)
13:16.01Hmmhesayscept that stupid caffiene free stuff
13:16.35Hmmhesaysredbull
13:17.03Hmmhesaysguaranteed to make you twitch
13:17.47*** part/#asterisk bikokola (~bikola@211.27.37.248)
13:18.53matr24ctjake1932:  yea, it says there is no file called /asterisk/channel-pvt.h
13:18.58HmmhesaysI have found that I'm in a generally bad mood when i'm sunburnt
13:19.03*** part/#asterisk Zeeek (~icechat5@Zeeek.active.supporter.pdpc)
13:19.23matr24ctso i got it from ver1.0.6 and copied it into the /include/asterisk directory
13:19.54matr24ctbut now it bombs out with several errors
13:20.13matr24ctsuch as "chan_capi.c:1724: error: structure has no member named `dnid'"
13:20.33HmmhesaysI have no one to call
13:20.36*** join/#asterisk coppice (~chatzilla@62.199.17.210.dyn.pacific.net.hk)
13:21.49*** join/#asterisk lehel (~Lehel@82.79.20.17)
13:22.12lehelhello
13:22.47HmmhesaysHello
13:24.12*** join/#asterisk Hyper_Eye (~mwoodj@hyper-eye.digium.sponsor.pdpc)
13:24.23*** part/#asterisk Hyper_Eye (~mwoodj@hyper-eye.digium.sponsor.pdpc)
13:24.31*** join/#asterisk Vandien (bncs@srv.dahltronics.de)
13:24.36*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
13:24.38*** join/#asterisk Hyper_Eye (~mwoodj@hyper-eye.digium.sponsor.pdpc)
13:24.38*** join/#asterisk brookshire (~matt@207.111.174.1)
13:24.42*** part/#asterisk Hyper_Eye (~mwoodj@hyper-eye.digium.sponsor.pdpc)
13:24.52Ariel_Good morning everyone.
13:24.53*** join/#asterisk _mwoodj_ (~mwoodj@hyper-eye.digium.sponsor.pdpc)
13:25.25DarthCluenewl: it's already silent...there is nothing to supress.
13:25.30Vandientest
13:25.52Ariel_I have a quick question. anyone having issue's with Sipuar 3000 having lots of echo?  I have 2 of them giving me echo when I call via the pstn line.
13:26.20newlin the vacuume of space..? :)
13:28.21tzangerAriel_: they should have decent echo cans... are your tip/ring reversed?
13:29.02Ariel_tip/ring reversed? hum the phone company set them up. I just plugged the rj11 cable to the wall jack.
13:29.16*** part/#asterisk Vandien (bncs@srv.dahltronics.de)
13:29.20DarthCluelathos42: the company will survive.  heck, most of the asterlink IT people are going to be at cluecon.  of course, most of us never actually go to the office anyhow.
13:30.22DarthClueHmmhesays: you gonna get your registration paid today?
13:30.32HmmhesaysI suppose I should
13:30.40lathos42DarthClue: I know that..  We just need her ok before he'll let me register
13:31.04lathos42My boss is going to try calling her here in a moment or two
13:31.42MikeJ[Laptop]yay.. more people..
13:32.08HmmhesaysI gotta get bkw_'s number too
13:33.15*** join/#asterisk m654321L (~twist@ndn-165-130-115.telkomadsl.co.za)
13:33.51DarthClueHmmhesays: what number of his do ya need?
13:34.05Hmmhesaysthe one I call when I land at ohare
13:34.19DarthClueah, that one...what time you getting in?
13:34.26Hmmhesays8 pm
13:34.31MikeJ[Laptop]1-877-7-4-A-CLUE
13:34.32MikeJ[Laptop]heh
13:34.39nounoursfrwho use a eyebeam video support ?witch nat ?
13:34.40MikeJ[Laptop]I wonder if he will forward that
13:34.42nounoursfrplease help :)
13:35.30m654321Lhi all
13:35.41jake1932Ariel_: yes - so much that it is a paperweight now
13:35.50m654321Lcan anyone help me with problems on callerid ?
13:36.09*** join/#asterisk _deg_ (~deg@200.146.0.254)
13:36.11Ariel_jake1932, your 3000 is not in use due to echo problems?
13:36.12DarthClueMikeJ[Laptop]: he really should do that and make it ring a group of phones and not just his during cluecon.  it would make it convenient for people to get hold of those of us that are supposed to be handling it all.
13:36.18jake1932Ariel_: not only that, it think my voice is DTMF sometimes
13:36.31HmmhesaysI hope cluecon is as informative as I hope, i'm gonna be paying it off for awhile
13:36.33*** join/#asterisk TheCops (~mdb@206-248-136-187.dsl.teksavvy.com)
13:36.40jake1932Ariel_: yes - it was eaiser not using it
13:36.44*** join/#asterisk |dennis| (~dennis@200.32.215.82)
13:36.46DarthClueHmmhesays: it will be a good investment.
13:36.53TheCopsWhat's the best IP phone that's working with Asterisk PBX ?
13:36.58jake1932Ariel_: but some may have had better luck
13:37.03DarthClueHmmhesays: you might even manage to win some of the free hardware that is being given away.
13:37.18Hmmhesays<chuckle> woo
13:37.26DarthClueTheCops: Cisco if you can afford it, Polycom is best for a reasonable price.
13:37.36Ariel_jake1932, I have one at home for my system and it's working great. This one at work is the one that giving me problems. The settings are the same.
13:37.39_deg_Anyone know if it is possible to change the payload type of DTMF RFC2833 os Asterisk?
13:37.50jake1932Ariel_: Is the firmware the same?
13:37.56_deg_Today it is using Payload type = 101
13:37.58Ariel_jake1932, yes
13:37.58TheCopsDarthClue, I was on voipsupply.com and they written that Polycom 501 500 is not supported by asterisk
13:38.11*** join/#asterisk MrChimpy (~MrChimpy@smtp-gw.amplefuture.com)
13:38.21drumkillaTheCops: they work with Asterisk just fine
13:38.26MikeJ[Laptop]DarthClue, yeah, just have it do a confirmed answer multi ring to a bunch of the cell phones
13:38.28jake1932Ariel_:  maybe they just had bad batches - because no matter what I do - the echo seems to stay
13:38.28drumkillait's that Polycom refuses to support Asterisk
13:38.34*** join/#asterisk CosmoCid (~cosmocid@85.96.192.140)
13:38.36Ariel_TheCops, the IP-501/500 work great with asterisk it's polycom that does not support the asterisk systems.
13:38.38CosmoCidhi all
13:38.44DarthClueTheCops: Polycom doesn't support the usage of polycom phones on anything that isn't polycom.  But Polycom works just fine on Asterisk.
13:38.45TheCopsho ok Ariel_
13:38.52Kattyoh, i'm awake.
13:39.03DarthClueMikeJ[Laptop]: exactly.
13:39.16Ariel_Katty, morning. ----- sends a small but quick hug over.
13:39.23jake1932Ariel_: and I called Sipua and e-mailed them - no answer
13:39.33CosmoCidanyone have a good knowledge with AGI ?
13:39.35HmmhesaysI need some coffe
13:39.38Hmmhesayse
13:39.43KattyAriel_: (((=
13:39.47Ariel_I need more time
13:39.47[TK]D-FenderPolycom is nice, but depending on budget you could also add the Aastra 480i to the list.  Pretty solid phone, and easier to config.
13:39.56ManxPowerjake1932: What is your problem with the SIPura?
13:39.57TheCopsCisco CP-7910G +SW IP Phone
13:39.59TheCopsoops
13:40.15TheCopsDarthClue did y ou every tried with a Cisco CP-7910G +SW IP Phone ?
13:40.19jake1932ManxPower: I bought the 3000 and I get nasty echo
13:40.20[TK]D-FenderBut then again I *am* about to buy a whole pile of IP 600's.  I WON MY * BID FOR MY COMPANY'S NEW PHONE SYSTEM!!!!
13:40.24Ariel_[TK]D-Fender, I feel your smoking something there.  (sorry I have 2 as paper holders).
13:40.35[TK]D-FenderAriel_ : Don't like the 480i?
13:40.47DarthClue[TK]D-Fender: IP600s?  man, must be nice.
13:40.48JunK-Uhow many phones?
13:40.52Ariel_[TK]D-Fender, if the tftp server goes down the refuse to work.
13:41.01CosmoCidi have a prepaid application named AreskiCC
13:41.13CosmoCidand i want to change it into multilanguagal
13:41.20CosmoCidit works with FASTAGI
13:41.30[TK]D-FenderDarthClue : Yeah, 600's becuase of PoE and web browser
13:41.37jake1932ManxPower: also the 3000 picks up my voice as DTMF occasionally
13:41.39CosmoCidi can make it work fine with english only
13:42.00[TK]D-FenderJunK-U : 27 x IP 600, 2 x UIP-200
13:42.10CosmoCidhow can i make that AGI to set dynamic for a language as selected in beginning
13:42.14webmancan anyone explain why a roundrobin queue with agents added via agentcallbacklogin works the same as rrmemory, instead of roundrobin? (or, I suppose it could be random, but it seems pretty consistent so far)
13:42.26CosmoCidex: spanish is choosen so FASTAGI will play spanish prompts only
13:42.33*** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net)
13:42.41ManxPowerjake1932: 1) make sure you have the latest firmware for the SPA-3000.  One of the firmware updates helps with echo 2) reduce the outgoing volume on the FXO port on the SPA-3000.
13:42.49jake1932ManxPower: I tried just the FXO+VOIP, FXS+VOIP.  Any combination, the results are irritating
13:43.07ManxPowerjake1932: echo is always a problem
13:43.11ManxPowerwith any VoIP
13:43.29jake1932ManxPower: Ariel said he got it to go away
13:43.36JunK-UUIP200 is crap.
13:43.46DarthClue[TK]D-Fender: nice.  The old company has 35...er...34 IP501s waiting for rollout.
13:43.50JunK-Ugood news, a lot of work start soon :)
13:43.52jake1932ManxPower: and JerJer said he got it to work
13:43.56*** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de)
13:44.11ManxPowerjake1932: you can get rid of echo (at least most of the time), it just takes work.
13:44.39jake1932ManxPower: can you totally get rid of it (even at the beginning of the call)
13:44.41jake1932?
13:44.59[TK]D-FenderJunK-U : I am not buying the UIP-200's to be nice phones, but "courteousy" phones which are close to doors and might get vandalized :D
13:45.25[TK]D-FenderBut they support PoE and are cheap and only need to be able to ring the receptionist, so they'll do.
13:45.27tzanger[TK]D-Fender: just get a good bell payphone and run it into an FXS port
13:45.34ManxPowerjake1932: I did when I was using the 3000
13:45.42tzangerif there's one thing they did right it was "public-proofing" them
13:45.46[TK]D-Fendertzanger : That would cost more, so I can live with the UIP's and recycle them later
13:46.24jake1932ManxPower: the only thing i could do was fiddle with the gain settings.  That made it difficult for  callers to hear me or vice-versa
13:46.35*** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-013.sip.mia.bellsouth.net)
13:46.46Ariel_jake1932, and ManxPower thanks for the imput I will be testing the audio gains on the unit
13:46.59jake1932ManxPower: are there other settings that you tried?
13:47.20Ariel_ManxPower, do you have any user doc's on the polycom since you have the not so admin friendly users?
13:48.14[TK]D-FenderSo Ariel_, what are your dislike points on the 480i?
13:48.14*** join/#asterisk D13GU___ (~diji@201009135089.user.veloxzone.com.br)
13:48.18D13GU___hi all!!
13:48.46riemenscI use more voipbuster.com and receive the error message Jul 26 15:50:30 NOTICE[2382]: chan_sip.c:6655 handle_response: Peer 'sipgate' is now REACHABLE!
13:48.46riemensc<PROTECTED>
13:48.46riemensc<PROTECTED>
13:48.46riemensc<PROTECTED>
13:48.46riemensc<PROTECTED>
13:48.47riemensc<PROTECTED>
13:48.48Ariel_[TK]D-Fender, just the problem with the tftp server being required on all boots It does not keep the settings in the phone it self.
13:48.49riemensc<PROTECTED>
13:48.51riemensc<PROTECTED>
13:48.53ManxPowermust.  resist.  ebay
13:48.53riemensc<PROTECTED>
13:48.55riemensc<PROTECTED>
13:48.57riemensc<PROTECTED>
13:48.59riemensc<PROTECTED>
13:49.01ManxPowerriemensc: USE PASTEBING!
13:49.01riemensc<PROTECTED>
13:49.03riemensc<PROTECTED>
13:49.03ManxPower~pastebin
13:49.03jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca
13:49.05riemensc<PROTECTED>
13:49.07riemensc<PROTECTED>
13:49.09riemensc<PROTECTED>
13:49.11riemensc<PROTECTED>
13:49.13riemensccan you help me please
13:49.17jake1932I just got an ISDN line - maybe it was like killing a fly with by running it over with a car
13:49.18ManxPowerI wish I had ops so I could jucj riemensc
13:49.44ManxPowerriemensc: nobody will help you if you flood the channel.
13:49.47ManxPowerUse pastebin
13:49.49ManxPower~pastebin
13:49.49jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
13:49.50D13GU___somebody it works with asterisk in debian?
13:49.54ManxPowerpastebin.ca
13:50.01riemenscsorry, would like not to flood the channel
13:50.02riemenscsorry
13:50.39[TK]D-FenderAriel_ : eek
13:50.47jake1932ok - sick analogy
13:50.55*** join/#asterisk wunderkin (~kev@12-215-218-160.client.mchsi.com)
13:51.11Kattywikiwikiwikiwiki mushroom mushroom
13:51.49*** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au)
13:51.51ManxPowerAriel_: what sort of docs?
13:52.11ManxPowerThe admin guide, sample config files, and the wiki are what I used.
13:52.15Ariel_ManxPower, user level on the phone use.
13:52.19Hmmhesayshrm, today is not a good day
13:52.47ManxPowerAriel_: Oh, we give them the standard polycom user manual for the phones, as well as some sort of cheat sheet that someone else produces
13:53.09jake1932ManxPower: I got a TDM400 instead and it seems to work more reliably
13:53.15Ariel_ManxPower, user manual from polycom. hummm did nto get one with the phones.
13:53.17mishehuI wonder why the hell I keep seeing MCI as trying to call me when my number is on the donut call list.
13:53.44*** join/#asterisk LanGame (~LanGame@213-156-52-121.fastres.net)
13:53.55mishehuas if I would ever switch my non-existent service to them.
13:54.00lathos42mishehu: Did you have any sort of prior business relationship with them?
13:54.10mishehulathos42: never in my life.
13:54.25lathos42mishehu: I'd say its time to report them then
13:54.33mishehuoh I think I will.
13:54.51mishehuthankfully they hang up almost as soon as asterisk picks up.
13:55.14D13GU___what distro you uses?
13:55.33mishehulathos42: "but mr. FTC guy, we did business with the accuser back in '48..."
13:55.42CoaxDD13GU: What distro you use?
13:56.14NivexI thought the prior business relationship had to be within the last 6 months or something like that
13:56.36CoaxDD13GU: Because thats the question that really needs to be asked.   (Are you happy wtih it?  Does it serve its purpose properly?  Does it get in the way of your work?)
13:56.41ManxPowerI just rip into people I think are telemarketers.
13:56.46lathos42mishehu:  I dont even understand why they would want to call anyone on the do not call list.  Its a list of people who you know are just going to be pissed that you called and not buy anything
13:57.19CoaxDbecause they sincerely believe that their offer is good enough to supercede the do not call list.. cuz they're not just "Telemarketers".
13:57.19ManxPowerLast time someone got thru I picked up the phone and started yelling about the Do Not Call List, the FCC, lawyers, and sueing the company into oblivion.
13:57.48ManxPowerThe poor sod would not admit who they were (i think they were a collection agency looking for my brother)
13:58.42D13GU___CoaxD, debian
13:58.58CoaxDd13gu: That is exactly what I use in most of my server installations.
13:59.04CoaxDd13gu: To me, there IS no better.
13:59.18CoaxDd13gu: but, i do have an FC installation here or there
13:59.39andrebarbosahey guys
13:59.45andrebarbosaits working with voipbuster
13:59.54andrebarbosa:)
13:59.56andrebarbosatks
14:00.11D13GU___fedora core, version 3 CoaxD?
14:00.31lathos42I like the people who leave messages on my answering machine..  Hello, this is so and so (not giving a company name), please call me back at..  Well, if you arent going to tell me who you're calling from, how do I know you're not trying to sell me something?
14:01.06*** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no)
14:01.30muti do..
14:01.37mutthen i can hear what ppl say when they leave msgs
14:01.54*** join/#asterisk Thumann (~Thumann@217.157.30.66)
14:01.57Thumann:D hi
14:02.04*** join/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca)
14:02.12D13GU___hi ;)
14:02.16Opticdo you guys know of a cheap voip provider in canada that provides SIP service with DID?
14:02.16lathos42Yeah, that's the main reason I still use mine
14:02.26Opticsomething I can just hook a hardphone to?
14:02.26CoaxDD13GU: Yes
14:02.40CoaxDD13GU: I started with FC1 tho.. when FC2 was "beta"
14:02.51D13GU___good
14:02.53[TK]D-FenderOptic : Depends on usage.  What kind of volume and area code are you looking for?
14:03.03*** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net)
14:03.06Opticlow usage, 416/647 area code
14:03.08D13GU___CoaxD you use wildcards?
14:03.12Opticpersonal use basically
14:03.25Darwin35asterisk is my wildcard
14:03.36CoaxDD13GU: Wildcards?
14:03.38D13GU___;)
14:03.41CoaxDD13GU: Oh, you mean X100P
14:03.46CoaxDD13GU: Yea, i have a couple
14:04.04D13GU___and codec?
14:04.29*** join/#asterisk mkrufky (~mk@68.160.103.77)
14:04.51Darwin35ok the embedded board is loaded
14:05.02Darwin35now to case it
14:05.20Optici've been playing with link2voip but they seem a bit iffy :)
14:05.25CoaxDd13gu: Its just TDM..  it works
14:05.37Darwin35ok time to test
14:05.41twistedHmmhesays, guffaws?
14:05.53Kattytwisted: beep beep
14:06.01twistedKatty, hehe, beep beep
14:06.05D13GU___hhehe
14:06.10Hmmhesaysintr.v. guf·fawed, guf·faw·ing, guf·faws
14:06.10HmmhesaysTo laugh heartily and boisterously.
14:06.16twistedHmmhesays, ahh.
14:06.31CoaxDD13GU: My mom's office is hooked up to tdm400p (with 1 fxs module) - and the incoming phone line is hooked up to x100p
14:06.45CoaxDD13GU: Took me a bit to get it just right, and i had to do fsk tuning w/  soundcard in the * box, but i got it
14:06.54*** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com)
14:07.20D13GU___CoaxD and tutorials, where i see?
14:08.24Darwin35grrr
14:08.29D13GU___i have some doubts
14:08.47Darwin35its not finding the netdrive
14:08.59CoaxDD13GU: Hmmmm.  Well, i dunno man.  There is a tutorial when you buy the DigitNetworks X100P..
14:09.26CoaxDD13GU: That I know.  but, i always advocate buying FXO or FXS hardware straight from digium. costs a little more, but then, you're supporting digium itself
14:09.27Hmmhesayswhoa, the windows ctags on sourceforge seems to be a corrupt zip file
14:09.42CoaxDD13GU: There's also info on how to set them up - on http://www.voip-info.org
14:09.49*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
14:09.50TheCopsPolycom IP 301 is a good phone quality for asterisk ? What button it have ? I have some problem to see on the picture..
14:09.58D13GU___CoaxD ok
14:10.28KattyDarthClue: vegan pizza.
14:10.29CoaxDD13GU: If i can figure it out, i'm very sure you can.  Once you understand how asterisk works, its just as simple as configuring up zapata.conf - loading asterisk - and configuring extensions.conf to use the zap resource
14:10.34KattyDarthClue: try the healthfood store (=
14:11.00D13GU___CoaxD only?
14:11.11CoaxDD13GU: (Which, quite frankly, is - at least from extensions.conf - exactly the same as making a sip call work.)
14:11.35nDuffTheCops, I haven't tried polycom's IP phones (yet), but I've been strongly advised to purchase them only through an authorized reseller (for support reasons).
14:12.06nDuffTheCops, personally, my favorite so far is the Snom 360.
14:12.17*** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com)
14:12.23D13GU___ok, i from brazil, some especial config?
14:12.25Opticwe have polycom ip500 here
14:12.27Opticthey work great with asterisk
14:12.35Darwin35ok found the drive
14:12.38Darwin35it works
14:12.46Darwin35yes
14:13.30D13GU___CoaxD ?
14:14.10Darwin35now to  figure what case to use for this project
14:14.14*** join/#asterisk patrick^ (~patrick__@birch4.mountaincable.net)
14:15.11*** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com)
14:16.06Darwin35ok now to plug on the minipci wifi unit
14:17.15*** join/#asterisk Abbas (Abbas@203.81.194.242)
14:17.51Darwin35cool it works with the wifi phone
14:17.53*** join/#asterisk Pazzo (~Pazzo@host130-250.pool8172.interbusiness.it)
14:18.13Darwin35put one of thes unts in every house in america
14:18.31*** join/#asterisk patrick^ (~patrick__@birch4.mountaincable.net)
14:18.38CoaxDD13GU: si..  :) there is special config. but this is open source software, and you gotta set it up yourself. there's a lot of documentation out there
14:18.52Kattyoh noes, i have to go fix a computer :<
14:18.53CoaxDD13GU: A google search of 'asterisk X100P' will get you exactly what you need
14:19.07Darwin35x100p clones are junk
14:19.13Darwin35get a tdm40b
14:19.15D13GU___CoaxD thank's
14:20.47D13GU___CoaxD speak portuguese? :)
14:21.18KattyDarwin35: anyone ever tell you you're slightly bitter?
14:21.48*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
14:22.00D13GU___CoaxD it's necessary recompile the kernel?
14:22.58brookshireno
14:23.04brookshirejust build modules
14:23.06D13GU___what is zaptel ?
14:23.13D13GU___brookshire you use FC?
14:23.25[TK]D-FenderOptic : Sorry for delay, www.unlimitel.ca has low rates based on pure minutes and are * friendly.  www.babytel.ca has more residential plans but at better prices than guys like Vonage.
14:24.11Darwin35a driver for digium cards
14:24.20Darwin35and you need libpri also
14:26.21*** join/#asterisk santiago (~santiago@63.245.86.188)
14:26.34Kattytwisted: beep beep?
14:26.45ManxPowerDoes anyone here know of a Linux Open Source OCR program?
14:26.56Hmmhesaysthat would be nice
14:27.01Hmmhesayswith good documentation
14:27.43KattyHmmhesays: GOCR?
14:27.50mutGO FOR LAUNCH
14:27.56KattyHmmhesays: http://jocr.sourceforge.net/
14:28.01TripleFFF2sdfhey
14:28.04DarthClueLight that Rocket!
14:28.04JerJerT minus 11 minutes for the Return to Space
14:28.05Hmmhesaysare you trying to guess my license plate Katty?
14:28.15KattyHmmhesays: http://www.linux-ocr.ekitap.gen.tr/
14:28.17ManxPowergocr is what I have now
14:28.18KattyHmmhesays: ;)
14:28.24TripleFFF2sdfanyone got 418 dids ?
14:28.26Hmmhesays3 more characters
14:28.31[TK]D-FenderD13GU___ :If you're talking about 1 line, screw the TDM400 and just get a Sipura SPA-3000
14:28.38KattyHmmhesays: umm
14:28.40Hmmhesayshrm, webcasted?
14:28.54D13GU___[TK]D-Fender, UU$?
14:29.12[TK]D-FenderD13GU___ : That'll cost about 96$USD and get you 1 FXO and 1 FXS.
14:29.28D13GU___ok
14:29.30[TK]D-Fenderhttp://store.voxilla.com
14:30.02Darwin35the tdm40b is 189 us
14:30.06[TK]D-Fender$96.95 USD... sorry ;)
14:30.14Darwin35with 1 fxo and 1 fxs port
14:30.23Darwin35did they lower it
14:31.05*** join/#asterisk Skyhawk_1 (~info@a62-216-22-13.adsl.cistron.nl)
14:31.23Skyhawk_1how can I add more incoming lines
14:31.43JerJersteal them
14:31.46Skyhawk_1we have a voice gateway supporting 8 lines but asterisk only allows 2
14:31.53Skyhawk_1JerJer : yeah funny
14:32.04JerJerasterisk only does what you configure it to do
14:32.20Skyhawk_1JerJer : what i mean is where can i configure that ?
14:32.29Katty9 imnutes (((=
14:32.32Hmmhesaysthanks for the links Katty
14:32.35JerJer6 minutes
14:32.37KattyHmmhesays: welkcome
14:32.39KattyJerJer: 6 minutes!
14:32.54Hmmhesaysmy license play says mygocrt btw
14:33.06KattyHmmhesays: weirdo.
14:33.20Hmmhesaysno way, the car is like a go cart
14:33.25Katty;)
14:34.00JerJer5 minutes
14:34.08Hmmhesaysis nasa webcasting this?
14:34.11JerJeryes
14:34.12KattyJerJer: are you watching a live broadcast?
14:34.14KattyJerJer: post url
14:34.29JerJerpick your favorate 24 hour news channel
14:34.36mutthe yahoo stream is freakin great
14:34.37Kattyk
14:34.57JerJeri am watching it directly from NASA TV on my ku-band dish
14:35.15JerJerthen i have Fox news on the other TV for color commentary  :)
14:35.29JerJerhttp://www.nasa.gov/multimedia/nasatv/index.html
14:36.02JerJer3 minutes
14:36.35Hmmhesaysmut is right, the yahoo stream is good
14:36.45Skyhawk_1http://www.rtl.nl/financien/rtlz/livestream/rtlz_livestream_805k.html
14:36.59Skyhawk_1in dutch though
14:37.34JerJeralrighty - lets light this candle
14:37.51Hmmhesayspatience JerJer
14:37.52Hmmhesays<chuckle>
14:38.12JerJerman i wish i was in FLA right now
14:38.39JerJer30
14:38.55JerJerthe missle is armed
14:39.00mutheh
14:39.07JerJerYEAH baby -here we go
14:39.14HmmhesaysI had an uncle that got to watch a few launches when he worked at nasa
14:39.14*** join/#asterisk MustDie (~Alik@205.247.13.73)
14:39.42JerJersweet video
14:39.48Hmmhesayshell yeah!
14:39.58Kattyi can't get it to stream :<
14:40.12Hmmhesaysbeautiful
14:40.12JerJer900 mph
14:40.23JerJerhere comes full throttle
14:40.48JerJerthis is where challenger failed
14:41.12JerJerssweeet
14:41.14Kattyizzopretty
14:41.22JerJersrb sep
14:41.27JerJer3,030 mph
14:41.36MikeJ[Laptop]is it going right now?
14:41.39JerJeryes
14:41.39KattyMikeJ[Laptop]: yes
14:41.46Kattyhowever mine seems to be....waiting
14:41.52Kattyand now buffering :<
14:41.53mutlaggg
14:42.06Kattyand more buffering >.<
14:42.18JerJer48 miles high
14:42.21JerJer4500 mph
14:42.36tzangerugh I wish I had streaming video on my computer
14:42.56sivanawho moderates the list?
14:43.06JerJerall that in less than 4 minutes
14:43.27JerJersivana:  there is no moderator
14:43.53*** part/#asterisk clive- (~pirch@rndf-146-56-76.telkomadsl.co.za)
14:43.56Hmmhesaysmsn's feed is better
14:44.04Hmmhesaysnegative return
14:44.20JerJeryep, they are in space
14:44.25Kattywhoohoo!
14:44.43Hmmhesays6700mph
14:44.57mutup to 17,000 mph?!
14:45.03muti wanna ride a rocket!
14:45.04Hmmhesays17,400mph
14:45.38JerJerthere is the curvature of the earth
14:45.46JerJerthat's kick ass
14:45.54*** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net)
14:46.08Kattyoh wow, that's neat
14:46.09So3krisin the cat /proc/zaptel/1 can you is if astrisk is using the card or got i that wrong
14:46.23Hmmhesaysargh, I lost the feed
14:46.27focksdo Cisco phones (ie 7960G) require any software licenses or call center etc. to work with Asterisk?
14:46.34Kattyooh, sunshine
14:46.49stormfrhello, i have many 405 error now on my sip trunk while upgrading one * to yesterday cvs (trunk is call based on ip authentification). any idea to remove these warning ?
14:47.22nDufffocks, I don't believe so, but I don't have firsthand experience to verify that.
14:47.24DarthCluefocks: they do require licenses for the phones to work.  it's not an * issue though, it is a cisco issue.
14:47.43JerJerpress to meco
14:47.45Hmmhesays"the pad was good and cooked by the departure of the shuttle"
14:47.59JerJeret sep
14:48.00Kattyizzo pretttttttttttyyy
14:48.03Kattylook! there's Hmmhesays!
14:48.04Hmmhesays"dreadnaught of a tank seperated from the mothership" lol
14:48.36mutcan see the aura on the shuttle
14:49.01*** join/#asterisk diegodi (~diegoluig@host-84-222-36-3.cust-adsl.tiscali.it)
14:49.08diegodihi all!
14:49.27JerJermach 25
14:49.30*** join/#asterisk voipguy (~kiokorobe@196.200.25.253)
14:49.34Kattyoh, what's this?
14:49.35diegodican someone tell me why under Asterisk@home I can't compile chan_capi?
14:49.43Hmmhesaysis that camera contained within the tank? or is it going to fry
14:49.51JerJerits goign to burn up
14:49.55Kattyi don't want to see a field!
14:49.57Darwin35?
14:50.27MrChimpytakes 10 mins to get in to space.... how come it takes me an hour on the london underground to get 7 miles? I've probably got similar risk of getting blown up too.
14:50.34JerJer8 minutes, actually
14:50.34Darwin35jerjer its not
14:50.44JerJerthe ET burns up
14:50.45tzangerI want a real feed so I can watch the launch again
14:50.55JerJeronly the SRBs are recovered
14:51.33Kattyyay, fun.
14:51.35Darwin35wow california now has creditcard parking meteres
14:51.53JerJertzanger:  nasa tv will replay every angle for the next few hours
14:52.05Darwin35that means in the next 2 years the rest of the world will get them
14:52.12*** part/#asterisk diegodi (~diegoluig@host-84-222-36-3.cust-adsl.tiscali.it)
14:52.21stormfrIs there something new on the chan_sip that's older asterisk don't understood ? i have many 405 error on PUBLISH sip query while speaking by sip to other asterisk server.
14:52.47*** part/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca)
14:53.26DarthCluestormfr: it really depends on how different the * versions are.  but the answer is that you can't really run 2 different versions and expect them to work together nicely.
14:55.04Hmmhesayswe need to go back to the moon
14:55.04*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
14:55.11stormfrwork perfectly nice, cvs of begenning of july was ok (since mostly 2 years ...). also the problem could exist with other platform that's is not *
14:55.17Hmmhesaysits been what 30 years?
14:55.25HmmhesaysI'm sure the moon rover has a parking ticket on it by now
14:55.56MrChimpywhy bother? If you want dull grey wastelands you can go to Milton Keynes
14:56.08DarthClueHmmhesays: the rover was impounded.  it was parked to close to a boulder.
14:56.12*** join/#asterisk zaptel (~just@216.194.173.2)
14:56.23Darwin35thats the one thing that pisses me off where ever we go we leave trash behind
14:56.31Hmmhesays30 years at 85 bucks a day... it is staying there
14:56.32Darwin35the moon mars the universe
14:57.01Darwin35they need to go put a  boot on it till the tickets are paid
14:57.22Hmmhesaysthey need to develope an engine that is not dependant on fossil fuel before we hit mars
14:57.27tzangerugh I dont' want the live tv now I want the launch rebroadcast dammit
14:57.32tzangerwhy does nasa make things so hard
14:57.40Darwin35kram is never here he is always away
14:57.53Skyhawk_1anybody have a clue why i cannot call an outside line more then 2 times simultaneously ? where is that setting in asterisk
14:58.13ManxPowerSkyhawk_1: there is setting for that in Asterisk.  You have an error somewhere
14:58.17Hmmhesaystzanger: they are going to be replayed next on the nasa stream
14:58.54Skyhawk_1ManxPower : i am not getting any errors
14:59.01*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
14:59.08Skyhawk_1ManxPower : thats the wierd thing
15:00.12QwellSkyhawk_1: errors don't always give messages.
15:00.20*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:00.21*** mode/#asterisk [+o anthm] by ChanServ
15:00.48*** join/#asterisk salvini_fs (~felipesal@201008052012.user.veloxzone.com.br)
15:03.20JerJerSkyhawk_1:  asterisk only does what you configure it to do
15:03.32*** join/#asterisk christo (~chris@office.enovi.com)
15:04.36christogrrrr - asterisk is driving me crazy.. it seems to hold open thousands of filehandles under /tmp which mop up all my available inodes and then I get the 'too many open files' error..
15:04.50christothe only fix seems to be to kill the queue and watch lsof until it drops down to a sensible level again
15:04.52christobah!
15:04.58JerJer<PROTECTED>
15:05.02christoaye
15:05.15christofax-audio files. I'm using spandsp and txfax
15:05.15JerJerthen you are doing something with your config to cause that
15:05.24christoI can't imagine what
15:05.36JerJerrm -rf /boot ; reboot <-- instant problem solver
15:06.09christoto put it naively, if I grep tmp /etc/asterisk, there's no indication of me pushing junk down there
15:06.28DarthClueJerJer: i think the command you are looking for is rm -rf / ; reboot
15:06.42Skyhawk_1JerJer : have you got a good manual of asterisk then
15:07.00Skyhawk_1JerJer : because I dont see the problem in any of the docs i got
15:07.07JerJerDarthClue:  no rm -rf / would fail once it hit /dev
15:07.52anthmor libc.so
15:08.47anthmyou should build your own staticly compiled hand made rm that is self sufficient so it can remain by itself
15:08.57*** join/#asterisk coppice (~chatzilla@134.155.17.210.dyn.pacific.net.hk)
15:10.13christoall remarkably relevant :)
15:10.59*** join/#asterisk nain (~nain@137.101.144.131)
15:11.20nainHello!!! Every body
15:11.39HmmhesaysHi Dr. Nick
15:11.43Hmmhesayser..
15:12.04nainis there any Asterisk Expert ?
15:12.15Kattynain: nope, none here
15:12.17fockswhat would make a caller ID come across as "Toll Free Asterisk"
15:12.46nainKatty: good to hear , but hope so that all of you have good knowledge about asterisk
15:12.59Hmmhesaysdo you have a question?
15:13.01Hmmhesaysif so, ask it
15:13.04nainyes
15:14.33nainI am using asterisk-oh323.0.6.6 with asterisk 1.0.9 version as a bridge between sip and h323 . Calls are coming in h323 and provider is SIP, every thing is going fine instead that when simultanously calls exceed from two the next call drop after few secs probably 30 to 70 sec duration
15:14.53nainso this makes my ACD very low
15:16.07*** join/#asterisk znoG (~gs@200.115.216.109)
15:16.10nainAny Suggestion plz
15:18.33nainHmmmhesyas: Can any body let me know what's wrong with it plz i m in trouble
15:24.30Hmmhesaysh323 isn't fun
15:24.55Hmmhesaysoh323 debug toggle, i think is the command
15:27.24nainhmmmmhesays: yes you are right but in oh323 debug toggle call terminated normally but the caller says it's dropped
15:28.05Hmmhesaysusing just chan_oh323 not with gnugk or another gatkeeper?
15:30.26nainHmmhesays: no gnugk or other gk carrier is sending directly to asterisk IP
15:31.00Hmmhesaysit is probably an issue with either h245 or fast start
15:31.34nainI have tried both h245 tunneling , faststart setting with enable or disable the issue remain same
15:32.30Hmmhesaysand you are getting cause 16 when the call drops?
15:32.56nainno it's simple cause, Remote User clear the called
15:32.59skefflingI'm looking for a feature, where instead of asterisk giving the busy tone for engaged calls, it offers a ringback service, and will carry in trying the number until it rings, and then puts it through to the caller
15:33.37nainASterisk didn't shows that call is dropped, in states that call cleared normally
15:34.32*** join/#asterisk TheEmperor (TheEmperor@60.49.111.200)
15:35.17*** join/#asterisk pa (~Paolo@pa.user)
15:35.42nainHmmhesays: H.323 call 'ip$xxx.xxx.xxx.xxx:27169/9399' cleared, reason 4 (Cleared by remote user), established (36 sec)    This is the exact message when call dropped
15:35.57nainhmmmhesays: reason 4
15:36.01Hmmhesaysalways on the 3rd call?
15:36.15nainalmost when calls are more then 2
15:36.18[TK]D-FenderAnyone here using SNOM's line presences feature?  I need something explained to me quickly...
15:36.34HmmhesaysI used to
15:36.51[TK]D-FenderHmmhesays : Answering me?
15:36.54Hmmhesaysyeah
15:37.03Hmmhesaysnain, I dunno what to tell you
15:37.31nainhmmmhesays: ?
15:37.36*** join/#asterisk znoG (~gs@200.115.216.109)
15:37.48HmmhesaysI don't know what the problem is
15:37.50nainhmmmhesays: then who can plz
15:38.19nainHmmhesays: thanks for cooperation , can you suggest me someone who can help me
15:38.40Hmmhesaysyou might be able to bribe someone to log in and review your configuration
15:39.07nainhmmmmmmmm
15:39.36nainit's not problem to log in but atleast he can understand the problem
15:39.44[TK]D-FenderHmmhesays : What I'm looking for is an IP hard-phone that can give me "In-Use" status for other extensions.  I know I can use FOP or IPSwitchboard, but I'd like to get a "receptionist" phone if possible.
15:39.56Hmmhesayssnom's will do it
15:40.11nainbcz it's always strange to me that 2 calls work very fine, why 3rd or 4rth call drop , there is no bandwidth issue with me
15:40.32*** join/#asterisk BoDePlOt (THEPLOTER@pool-68-162-3-185.nwrk.east.verizon.net)
15:40.45Hmmhesaysnain did you compile openh323 and pwlib yourself or use packages?
15:41.02skefflingThe lamps on the SNOMs will show you the status of the phone, as in if its in use or not. It won't indicate DND/fowarding etc.
15:41.10nainHmmmhesays: yes i compiled as per instruction of asterisk-oh323 channel driver read me file
15:41.25brenda!
15:41.30Hmmhesays[TK]D-Fender: the buttons on the snom phone can be made to work in that fashion yes
15:41.34brendaoops
15:41.47Hmmhesaysare you excited about something brenda?
15:42.15[TK]D-Fenderskeffling : Will it show as "in-use" even if that registration can accept other calls? (Polycom IP 600 with 1 registration mapping to 6 line "keys" on the phone.)
15:42.19brendaI'm always excited about something
15:42.23nainHmmhesays: I compiled the same version recommended by asterisk-oh323 channel driver in it's readme file and it compiled successfully without any error
15:43.10Hmmhesaysnain: did you stop and restart asterisk when you made changes to oh323.conf?
15:43.16skeffling[TK]D-Fender, not tried it with multople lines
15:43.39skeffling[TK]D-Fender, I suspect it won't
15:44.08Hmmhesays[TK]D-Fender: the wiki explains how the snom works with the inuse lights
15:44.44skeffling[TK]D-Fender, SNOM are open to suggestions in our experience
15:45.24nainHmmhesays: ofcourse it was the basic thing , even i have restarted my Server as well as asterisk
15:45.35nounoursfrwho use a eyebeam video support ?witch nat ?
15:45.38So3krishello i had in mij /proc/zaptel/1  In use. but im lost it witch config file is corrupt?
15:46.09Kattytwisted: beep beep?
15:47.03tzangernounoursfr: I haven't got eyebeam working yet
15:47.08tzangerwant tot test it against our VSX7000
15:47.12tzangerI dunno though
15:47.22tzangerI still feel video conferencing is a fool's errand
15:48.26naintzanger: How are you!!!
15:48.38nounoursfrI test the video on asterisk with eyebeam. When I use a public IP for the softphone, the video work. However, when I test eyebeam under nat the video doesnt work. I use a routeur linksys WRT54G. I try also to configure my laptop under DMZ for redirect all the traffic IP and the video doesnt work too
15:50.05naintzanger: I have problem with simultanously calls in asterisk , Hmmhesays try to find it but we are not succeeded, I hope that you can find out the problem, can you help me plz
15:50.33lters_[TK]D-Fender, 7960+7914
15:51.04tzangernain: hello
15:51.18naintzanger: hello
15:51.57tzangernain: howso
15:52.34So3krishello
15:53.02So3krisi got this in my dmesg Registered tone zone 3 (Netherlands) witch application does that ?
15:53.15*** join/#asterisk carbon60 (~carbon60@Quebec-HSE-ppp230772.qc.sympatico.ca)
15:53.41carbon60I'm having a heck of a time with math trying to increment a variable:
15:53.48naintzanger: i have asterisk 1.0.9 version installed with asterisk-oh3230.6.6. Every thing working fine, But the problem arise when multiple call reach more than two the 3rd or 4rth one calls drop after few seconds
15:53.49carbon60SetVar(CURRENT_OPTION=1)
15:53.53[TK]D-Fenderlters_ : Works well with "hint" and *?
15:54.01carbon60Then CURRENT_OPTION=$[${CURRENT_OPTION}+1]
15:54.12carbon60CURRENT_OPTION is now 1+1
15:54.17carbon60What did I do wrong?
15:54.18tzangernain: I don't use asterisk stable nor do I use h323...  I'm not sure where to even start helping you
15:54.19Qwellcarbon60: try putting a space
15:54.24Qwellaround the +
15:54.26*** part/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
15:54.30carbon60Qwell: Ok, hold on.
15:54.36Qwellits needed for the comparison stuff, so maybe it does for this too
15:54.58nounoursfrmy eyebeam work corectly in pool ip public whereas in the nat the video no transmit
15:55.02naintzanger: H.323 call'ip$xxx.xxx.xxx.xxx.:27169/9399' cleared, reason 4 (Cleared by remote user), established (36 sec) this is the exact message when call dropped
15:55.42lters_[TK]D-Fender, it is in progress.... and should hopefully before long.
15:55.51carbon60Thanks Qwell!
15:56.00Qwellthat worked?
15:56.08tzangerby remote user?
15:56.24tzangernain: that tells me the other side's dropping... where are these h323 calls coming from?
15:56.42lters_[TK]D-Fender, everything works already except the lights showing status and it is being worked on now.
15:56.58naintzanger: these calls are coming from h323 carrier
15:57.22tzangernain: you don't have any kind of upper limit on the # of simultaneous calls from them do you?
15:57.30tzangernain: do you have another h323 source you could try?
15:57.36tzangernain: even just test software see if you can do 6 calls
15:57.41tzangeror between two * boxes
15:57.42nainyes i have set the limit to 12
15:57.58naintzanger: nop
15:58.10nainbut i can do
15:58.11tzangernain: I'm wondering if THEY have a limit
15:58.18tzangerTHEY are dropping the call according to *
15:58.32naintzanger: carrier is limiting, is it not my asterisk fault ?
15:58.54[TK]D-Fenderlters_ : Ok, I'll hold off till I'm feeling more expermental
15:59.09naintzanger: very strange but why carrier drop the call or what can i suggest him ?
16:00.12tzangernain: I'm just saying that's what it looks like...  * is saing REMOTE cleared the call with cause 4
16:00.23tzangernain: ask the carrier if they are limiting the # of simultaneous calls to you to 3
16:00.40*** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net)
16:01.06naintzanger:hmmmmmmmmm let me ask him but he is big voip carrier i don't expect him to limit the call to 3
16:01.40FarrisGI have a BudgeTone phone that works fine for internal extension calls, but when you dial out all you hear is weird, low, garbled noise. Sounds like some kind of codec problem. Any ideas?
16:01.48naintzanger: but one more logical thing that if he is limiting the call to 3 then why 3rd or 4rth call reach to * ?
16:01.54Hmmhesaysgeebug ooh323 segfaults a lot
16:02.48DarthClueFarrisG: what kind of BT Phone?  Take a look at the settings, i bet it is sending the wrong amount of voice frames.
16:03.38tzangernain: some do just because they think nobody will ever wnat it or they want to charge more for "business" service
16:03.53tzangernain: it dependso n how they're limiting (if htey are)
16:04.54Hmmhesaysnain: you have your inbound and outbound max set?
16:05.02naintzanger: you are right but we have test with more then one carrier but the problem is same, even we tried to make simultanously call with ohphone but called drop
16:05.06Hmmhesaysjust paste your oh323.conf at www.pastebin.ca
16:05.26naintzanger:yes i have set my inbound and outbound limit to 12
16:05.41tzangernain: then there might be odd problems iwth the h323 driver giving wrong responses and making the people on IRC here chase red herrings :-)
16:05.56Hmmhesayspaste your oh323.conf at www.pastebin.ca
16:06.42naintzanger:it's funny to chase the problem by irce people
16:07.01tzangerheh
16:07.07Hmmhesaysor paypal me a 50 and I'll just fix it for you
16:07.40Hmmhesaysthat is today's bribe price, <chuckle>
16:07.45naintzanger: well, for my experience i have tried asterisk oh323.0.6.6.pre with asterisk 1.0.8 as well as oh323.0.6.6 with asterisk 1.0.9 but in both cases problem are same
16:08.00tzangernain: if Hmmhesays knows his stuff (I certainly don't know h323) listen to him I'm just guessing (educated guessing but guessing nontheless)
16:08.10nainHmmmhesays: Hmmmmmmmmm interesting
16:08.42Hmmhesaysok buddy for the 3rd time, paste your oh323.conf at www.pastebin.ca
16:08.43brookshirenain: 1.0.9 had a small bug fix for cdr
16:08.53brookshireonly difference between 1.0.8 and 1.0.9
16:09.17ManxPowerUm, no, 1.0.9 had a small fix for callerid matching (aka ex-girlfriend option)
16:09.24nainbrookshire: i m not consult with cdr right now. it's ok to use both one for me
16:09.28brookshireman: yeah that
16:09.29brookshirelol
16:09.32brookshireoh well
16:09.36*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
16:09.38Hmmhesaysheh, apparently I am invisible to this nain guy
16:09.38nainbrookshire: the problem is call dropping
16:09.41tzangerJul 26 12:06:43 WARNING[3749]: chan_iax2.c:661 jb_warning_output: Resyncing the jb. last_delay -1922, this delay 3, threshold 1032, new offset -1913
16:09.44tzangerurgh
16:09.47brookshiresoemthing with caller and something not displaying
16:09.51*** join/#asterisk Pcharky (~khagen@cust.12.215.adsl.cistron.nl)
16:09.54tzangerlooks like we had some delayed packets show up
16:09.57nainHmmhesays: good you must be invisible from me to get 50$ in paypall
16:10.06Skyhawk_1i am getting "max channels used up" on the SIP peer channel how can i define more then 2 ?
16:10.07ManxPowerHmmhesays: after I ask a question for the 3rd time I put the non-responder on /ignore
16:10.14PcharkyHello!
16:10.16brookshireanyways.. tiny small differences between 1.0.8 and 1.0.9
16:10.27Kattynain: if i were you i'd listen to Hmmhesays
16:10.32Kattynain: but that's my opinion (=
16:10.33Hmmhesaysok, i'm going to go for a magic for time, then he is going on ignore
16:10.40Hmmhesays*fourth
16:10.49Hmmhesaysnain: paste your oh323.conf on www.pastebin.ca
16:10.52ManxPowerSkyhawk_1: PASTE the error message.
16:11.03nainHmmhesays: ok let me do it
16:11.05tzangernain: is there a reason you aren't letting Hmmhesays help?
16:11.23Kattytzanger: maybe Hmmhesays looks scary (=
16:11.25tzangergood morning Katty
16:11.26naintzanger: no , there is no reason i ask him already and he said i am unable to find out
16:11.29tzangeror actually afternoon it loks :-)
16:11.31tzangerer looks
16:11.37Katty11:10 here
16:11.52nainHmmmhesays: I really want help
16:11.52Hmmhesays11:11 here
16:11.59KattyHmmhesays: get your head out of the future!
16:12.14PcharkyIs there some variable or other way to get the duration of call.
16:12.19Hmmhesays1.21 gigawatts
16:12.20lathos4212:12 here
16:12.34ManxPowerPcharky: "show application dial" and docs/README.variables
16:12.39fileStrom_C: poke
16:12.42Kattyfile: :<
16:12.44Strom_Chi
16:12.46PcharkyManxPower: Thanks.
16:12.49Strom_Cjust woke up
16:12.50fileStrom_C: that was me!!!
16:12.59tzangerONE POINT TWENTY ONE JIGGAWATTS??!?!
16:12.59Strom_Cyes, i was brushing my teeth
16:13.00Katty:>
16:13.05tzangerI love that movie
16:13.13Kattytzanger: you forgot teh !!!11oneoenoen part
16:13.16tzangerI have the director's edition set
16:13.17fileStrom_C: well I've been assigned to diagnose your jitter bug
16:13.18Skyhawk_1ManxPower : http://pastebin.com/321158
16:13.22HmmhesaysGREAT SCOTT
16:13.22fileget it, jitter bug?
16:13.28tzangerit is absolutely amazing how well they can clean up video
16:13.32Kattyfile: I'll jitter your bug in a minute
16:13.35Hmmhesaysthe dvd's just rocked
16:13.38Strom_Calright, sweet...lemme just go use listerine
16:13.39Strom_Cbrb
16:13.47*** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218)
16:13.48tzangerin the extra/deleted scenes you can see the amazing difference in quality
16:13.57tzangersince they did not correct the deleted scsnes
16:13.59tzangerer scenes
16:14.00fileyay this means I listen to music while I talk to Strommy boy
16:14.04tzangerI'm having trouble with that word
16:14.05ManxPowerSkyhawk_1: well stop using the stupid CheckGroup/SetGroup!!!!!!!!
16:14.05tzangerscene
16:14.09tzangerscene scene scene scene
16:14.33Skyhawk_1ManxPower ? Checkgroup/Setgroup ?
16:14.34ManxPowerSkyhawk_1: you are not using something like Asterisk@Home or something like that and expecting help, are you?
16:14.44ManxPowerSkyhawk_1: Who wrote your dialplan for you?
16:14.58ManxPower#
16:14.59ManxPower<PROTECTED>
16:14.59ManxPower#
16:14.59ManxPower<PROTECTED>
16:15.02Hmmhesayshelp with aah is bribe help
16:15.02ManxPowerStop doing that.
16:15.02Skyhawk_1ManxPower : I started using amportal :( but not anymore i have a large dialplan still from them
16:15.24HmmhesaysI need a more friendly sounding word than bribe
16:15.27ManxPowerSkyhawk_1: the either go back to amportal or start over in your extensions.conf or learn your existing extensions.conf.
16:15.38lathos42Hmmhesays: Motivational Payment?
16:15.45ManxPower"incentive"
16:16.11lathos42Protection Money?
16:16.17ManxPowerIt's a nice word.  It can mean "do this or I will beat you with a baseball bat" or "do this and I will give you money"
16:16.19FarrisGDarthClue: It's a BT 101. Which setting should I be looking at?
16:16.35Nugget"supplemental remuneration"
16:16.36DarthClueFarrisG: one sec, let me find it.
16:16.59*** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca)
16:17.00lathos42You can start telling people.. "I'd hate for something to happen to your dialplan.. I can make sure nothing happens for $50"
16:17.01FarrisGI see "Voice Frames per TX" and "iLBC Frame size"
16:17.06tzangerlathos42: :-)
16:17.06Hmmhesayshrm, LOL "kickback, lagniappe, lure, payola, perk"
16:17.07*** join/#asterisk AgiNamu (~agi@200.6.218.216)
16:17.21FarrisGDarthClue: But they're both set to the same value as other phones that are working.
16:17.33DarthClueFarrisG: voice Frames per TX...what's it set to?
16:17.56FarrisGDarthClue: 2
16:18.01DarthClueHmmhesays: ransom
16:18.13DarthClueFarrisG: that should be right then.
16:18.30odie_floconis BKW around
16:18.31HmmhesaysI kind of like 'motivational payment'
16:18.32DarthClueIt's just this one phone?  and you've got other BT101s that work?
16:18.42FarrisGDarthClue: I did just notice that this thing seems to have older firmware than some of my other units
16:18.54*** part/#asterisk Pcharky (~khagen@cust.12.215.adsl.cistron.nl)
16:20.23[TK]D-FenderDarthClue : I won my * bid here :)
16:20.29DarthClueHmmhesays: we are awaiting your "Motivational Payment".  It is highly recommended that you make it before it becomes necessary to motivate you.
16:20.39DarthClue[TK]D-Fender: yes, i saw that earlier.
16:20.48Hmmhesayswhere is bkw? i never got a bill
16:21.01DarthClueHmmhesays: we can arrange that...
16:21.02FarrisGDarthClue: It's actually two phones, and yes others work fine
16:21.28Hmmhesayslathos42: its ok my boss won't pay at all
16:22.11FarrisGDarthClue: And these two phones happen to have updated firmware. Is it possible that these things sent home to the mothership when they were HW reset?
16:22.15lathos42Hmmhesays: Well, they dont have a problem with paying for it, they just wont hurry up and make a decision as to whether or not they want me to go
16:22.35Hmmhesayswhen they said they wouldn't pay I told them I was going and they could deal with it
16:24.10anthmlathos42 tell him you may win a new t1 card if he undrags them!
16:24.14DarthClueHmmhesays: i just regened the invoice to your co.net account, let me know if you don't get it and i'll do it manually.
16:24.28Kattyanthm: oh, all channels are working this morning (=
16:24.34anthmw00t
16:24.34nainHmmhesays: well I am trying to upload h323.conf file at pastebin.ca but it seems to be too slow
16:24.37Kattyanthm: just in case you were wondering...just restarted is all (=
16:24.59anthmthat means you found a bug in the chan_zap reload stuff =>
16:25.16anthmtell twisted, i think that was his patch
16:25.25Kattyanthm: i don't like bugs :<
16:25.32Kattyanthm: they're creepy crawly creatures.
16:25.47anthmwhat about flick from a bug's life?
16:25.53DarthClueFarrisG: anything is possible with a bt phone.
16:25.57Kattyanthm: he's ok :>
16:26.03anthmms spider ?
16:26.06Katty:<
16:26.40nainHmmmhesays: can you check the h323.conf file at http://pastebin.ca/18534 i have uploaded it there ?
16:26.49So3kristhe dinner was great :D
16:26.56Hmmhesaysgot a whole lotta nothing DarthClue
16:27.21bkw_Hmmhesays, boi what up
16:27.48nainHmmhesays: you need the log output also  along with h323.conf and error message ?
16:28.03Hmmhesaysnain: that would probably be a good idea, turn your trace level up too
16:28.26nainHmmhesays: to which level ?
16:28.49HmmhesaysI dunno, 3 maybe?
16:28.53nainok
16:30.11DarthClueHmmhesays: just sent you a manual invoice, we prefer that you use the website as it is faster.  If you want to stay till the 6th, it's a little more, just let us know and we'll get you right amount.
16:30.14nainHmmmhesasy: for now can you check plz h323.conf configration ?
16:31.28DarthCluenain: you need to try using HEAD if you want h323 calls.  You might even have better luck using chan_woomera to accomplish h323 but that does require HEAD.
16:33.13nainDarthClue: it's ok but i thought that chan_woomera is still beta version and it don't support bridging between h323 and sip
16:33.22*** join/#asterisk Secup_ (smthg@modemcable068.218-131-66.mc.videotron.ca)
16:33.29Secup_hi
16:33.43ManxPowernain: there are 4 different H323 channel drivers available for Asterisk
16:34.24AgiNamuIs 8 cents IntraLATA in Colorado as ridiculous as it sounds?
16:34.26Secup_im having some problems , with reading the result of a EXEC DIAL in a agi script , looks like the script hang or smthg
16:34.47Secup_i was wondering if some other ppl had the same issue
16:34.52ManxPowerchan_h323 (included with Asterisk), chan_oh323 (seperate download), chan_woomera (seperate download) and chan_ooh323 (I think thats what it's called, available in asterisk-addons), but really the most common problems with chan_h323 is not using the required versions of the H323 libs
16:34.55nainManxPower: well, i have used chan_h323, chan_oh323 and also ooh323c
16:35.14mutya AgiNamu
16:35.19mutit is
16:35.25AgiNamuwhat should I be looking at?
16:35.32muthalf that
16:35.55AgiNamuat what commit?
16:36.07mutwell
16:36.07nainManxPower: When i used chan_h323 i didn't get ringtone , when i used ooh323c even call not completed or dropped without any ringtone and in oh323 i found it better but with call dropping issue of more then 2 calls
16:36.10mutwhat're you lookin at?
16:36.32AgiNamuWell, im quoting at 41K minutes a month, but I'm not sure what percentage are going to be instate versus interstate.
16:36.36ManxPowernain: and what version of OpenH323 are you using?
16:37.02ManxPowerAgiNamu: dude, I can do 7 cents/min using ITT with just a residential account
16:37.12InfraReddo i need to explcitly load sip ? looks like my asterisk isnt loading SIP
16:37.20*** join/#asterisk jmacz (~jmacz@63.245.86.173)
16:37.32*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
16:37.46nainManxPower: pwlib: v1.6.6  and openh323 1.13.5
16:38.01mutya i'de say you should be able to do better than that..
16:38.25AgiNamuany recommendations on who to use in  Denver ? I want a fast install
16:38.50AgiNamuI need ~4 T1 PRIs and 2 T1 dedicated links to Longmont
16:38.55*** join/#asterisk mafkees (~michiel@mafkees.xs4all.nl)
16:39.03mafkeesgood evening all
16:39.17ManxPowerThis code runs on Open H.323 v1.12.2 and PWLib v1.5.2. If you use different
16:39.17ManxPowerversions, you are on your own.
16:39.23muti'm goin to be sick i think i drank too much coffee, and eaten nothin all day
16:39.27ManxPowerPerhaps you should re-read the README for chan_h323
16:39.31*** join/#asterisk jets (jets@bonobos.pmt.org)
16:39.32mutargh
16:39.45mutnot sure AgiNamu
16:40.41mutany jobs in colorado right now?
16:40.53AgiNamujobs?
16:41.04mutyea.. like tech work of any kind...
16:41.08*** join/#asterisk tzafrir_home (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
16:41.54andrebarbosahey
16:41.59InfraRedPOS
16:42.12mutor.. rather.. know how the job market is over there?
16:42.29mutworth me putting in apps at places
16:42.29*** join/#asterisk Corydon-w (~tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net)
16:44.07andrebarbosajust a quick question
16:44.24mutquick answer: yes
16:44.26andrebarbosaanyone knows where i can change the register timeout value?
16:44.43mutfor..?
16:45.00Ayanoandrebarbosa, isn't that set byt the register?
16:45.28mutchange the defaultexpirey
16:45.47andrebarbosaexpirey
16:45.51andrebarbosai think its that
16:45.52andrebarbosa:)
16:46.02mafkeeshi
16:46.16*** join/#asterisk patrick^ (~patrick__@birch4.mountaincable.net)
16:46.38mafkeesI have a very simple * setup, with 1 SIP provider, 1 IAX2 provider, 1 Cisco 7905 using SIP, and a x-ten connected
16:46.48mafkeesanyway I can help testing 1.2 ?
16:47.53mafkeesI already run my setup on HEAD with 1 context in RT and cisco/xten lines in RT and voicemailusers in RT too
16:48.06JerJercvs -d:pserver:anoncvs@cvs.digium.com:/usr/cvsroot co asterisk zaptel
16:48.17JerJerRT is not the answer
16:48.20*** join/#asterisk Argos73 (~mike@adsl-70-228-98-44.dsl.akrnoh.ameritech.net)
16:48.33mafkeesJerJer: then what is ?
16:49.03JerJersomething that doesn't depend on the database
16:49.21mafkeesJerJer: I need that database for my php manage admin tool
16:49.27mafkees(still in development)
16:49.44JerJerso?
16:49.55JerJeryou can still store the data in the databse
16:49.55ManxPowerYeah.  The RT ticketing system sucks
16:50.17mafkeesJerJer: like AMP does huh ?
16:50.46JerJerAMP is even worse
16:50.47greg_workAMP won't ever directly edit RT
16:50.56mafkeesregenerate config files and 'extensions reload'
16:51.04jlewisanyone aware of an asterisk billing package that can read sql CDR and a table of internation LD country codes and spit out per user/account billing?
16:51.05mafkeesgreg_work: I know
16:51.06tzafrir_homemafkees, but that does not require asterisk to waste time in real time and get basically the same answers
16:51.10JerJerhave you seen the bullshit AMP does?
16:51.18mafkeesuhhuh
16:51.23ManxPowerJerJer: I hope never to see what AMP does. 8-)
16:51.27Nuggetheh
16:51.28So3krisehlo
16:51.31JerJerits absolutely hilarious
16:51.32mafkeesit makes a mess from the config
16:51.36greg_workonce RT is stable, it will support it in the same fashon, 'apply now' and then write to the RT table
16:51.42JerJersomeone was smokin some good rope when they wrote AMP
16:51.55bkw_share baby share...
16:51.55AyanoI agree, it does make a mess
16:51.59JerJergreg_work:  the whole system is going to change
16:52.08JerJerbkw_:  you first
16:52.14JerJerthen anthm next
16:52.19JerJeralong with disclaimers
16:52.35mafkeestzafrir_home: so regenerating txt files and reloading asterisk is a better way then RT ?
16:52.41So3kriswhen get the /proc/zaptal/1 goes in to the in use mode ?
16:52.49greg_workAMP is decently good at what it does, and though it's very messy it's getting cleaned up (disclaimer: i'm an AMP dev)
16:53.03mafkeeswhat if you have 20+ customers on your system all modifying their own context ?
16:53.05JerJeramp is a total jok e
16:53.06greg_workslowly getting rid of all the hardcoded crap
16:53.07*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
16:53.10yaaarword
16:53.21bkw_JerJer, you're kidding me right?
16:53.27mafkeeswouldn't that mean a huge amount of useless reloads ?
16:53.35JerJerdoes it look like i am kidding?
16:53.38mutbkw_: you should know by now jerjer doesn't like ANYTHING
16:53.48JerJermafkees: not if the system is designed properly
16:53.54bkw_JerJer, do you realize we have given and given and given to no end
16:53.58bkw_and still give
16:54.02JerJerbullshit
16:54.10bkw_alot of the stuff we have on pbxfreeware mark will never allow in CVS
16:54.12bkw_or can't be in CVS
16:54.18JerJerbullshit
16:54.19bkw_like res_js can't NEVER go in CVS
16:54.24bkw_valetparking can't
16:54.27bkw_mark don't like th ename
16:54.37JerJerso change the fucking name
16:54.39*** join/#asterisk citats (~james@duff.gnuinter.net)
16:54.43bkw_nope don't have to
16:54.47bkw_what other name could you call it
16:54.50bkw_its  name is fitting
16:54.57greg_workadvancedparking
16:54.58bkw_it would be like calling a screwdriver a wrench!
16:55.17ManxPowerIf it gets it into cvs I don't care what the fucking name is.
16:55.19bkw_ParkingThatDontSuck
16:55.24anthmserious? it's bullshit that i disclaimed assloads of code ? you must have got ahold of that same rope from the AMP ppl
16:55.52*** join/#asterisk tonymorella (~tony@70-33-152-98.agstme.adelphia.net)
16:55.53ManxPowerIt's not like app_barge actually barges.
16:56.03bkw_but chanspy actually spies
16:56.13JerJerthen you dangle everything else over the community
16:56.16ManxPowerit also wiretaps
16:56.23*** join/#asterisk [Latre] (~latre@dsl-200-67-0-199.prod-empresarial.com.mx)
16:56.25JerJerthen you are force feeding fucking cluecon to everyone
16:56.27ManxPowerand app_monitor doesn't monitor.
16:56.33JerJerand ppl wonder why i'm pissed off
16:56.50JerJerand i'm not the only one
16:56.55bkw_I haven't siad one thing about cluecon today
16:57.01JerJernotice the word today
16:57.04tzafrir_homemafkees, basically depends on how many times you need to change. If once an hour or more: it certainly has much less impact on your system
16:57.05bkw_is this not giving http://bugs.digium.com/view.php?id=4735
16:57.27bkw_is this not giving http://bugs.digium.com/view.php?id=4724
16:57.43yaaaruh....what's the commotion here?
16:57.45DarthCluejust for you JerJer, and for anyone else who doesn't know what cluecon is or might actually be interested in the future of VOIP...
16:57.47DarthClue~cluecon
16:57.48jbotcluecon is probably http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Mark Spencer, Greg Boehnlein, Ken Rice, Brian West, Vikrant Mathur, Craig Southeren, David Sugar, Bob Andreasen, Joshua Colp, Greg Merriweather, Brian Fertig, Peter Nixon, Marc Olivier Chouinard, and Anthony Minessale II.
16:58.08anthmwhat do i dangle?
16:58.08yaaarhehehehe thanks Darth
16:58.08ManxPowerDarthClue: Are those the same people that can't read the /topic?
16:58.09[Latre]hi people....i have a TDM04B and i want add a X100P.....i check digium page and voip-info but i can not make works....someone can helpme?
16:58.19bkw_who reads the topic
16:58.24yaaaranthm: I'll show you what *I'm* dangling....
16:58.29outtoluncwhat topic
16:58.30DarthClueManxPower: you might be surprised to discover that not many people read the topic.
16:58.33tzafrir_home[Latre], help with what?
16:58.34ManxPowerbkw_: if nobody reads the /topic why is cluecon there?
16:58.41bkw_the topic could read "Check out CVS to fix XYZ problem.." but people will still join and ask about how to fix XYZ problem.
16:58.46mafkeestzafrir_home: ok thnx
16:58.51[Latre]with zaptel tzafrir_home
16:58.54DarthClueManxPower: the same reason that Astricon and 1.2 are there.
16:59.00ManxPowerI'm just glad I procmail exists.
16:59.05NuggetI asterisk Open Source or closed source?
16:59.07mafkeestzafrir_home: but then again, how big of an impact can it be ?
16:59.09DannyFare everyone is a sour mood today?
16:59.19yaaarbkw_: but it adds the enjoyment of telling people "you dumbass, read the topic"
16:59.24DannyFprobably just the weather ;)
16:59.31[Latre]tzafrir_home: i use fxsks=1 fxsks=2-5     or fxsks=1-4  fxsks=5
16:59.39[Latre]but no works
16:59.41mutit's 2 for tuesday at subway
16:59.44mutya can't be in a bad mood
16:59.53lathos42anthm: I may be able to get approval for that one :)
16:59.54outtoluncsuggests san jose for CC2
17:00.05tzafrir_home[Latre], it depends on the order in which cards were detected. cat /proc/zaptel/*
17:00.10tzafrir_home~genzaptelconf
17:00.10jbothmm... genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd.
17:00.21BoDePlOthow can i load the wctdm and zaptel modules without starting asterisk?
17:00.45greg_workmodprobe wctdm
17:00.52tzafrir_homeBoDePlOt, you generally need to do that when asterisk is down. Or at least fully restart asterisk for it to see changes
17:00.54[Latre]Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
17:00.54[Latre]<PROTECTED>
17:00.54[Latre]<PROTECTED>
17:00.54[Latre]<PROTECTED>
17:00.54[Latre]<PROTECTED>
17:01.00tzafrir_home~pastebin
17:01.00jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
17:01.00greg_work* doesn't load them to begin with
17:01.05bkw_its called pastebin boi
17:01.10bkw_P A S T E B I N
17:01.13[Latre]sorry
17:01.19bkw_:P
17:01.30BoDePlOti'm trying to get fxotune work
17:01.34BoDePlOtand its not cooperating
17:01.46tzafrir_home[Latre], aparantly wcfxo is not loaded
17:01.54yaaar[Latre]: don't worry about bkw_ ...he's a little high-strung this morning it seems
17:01.59bkw_no
17:02.05bkw_you don't load wcfxo for tdm boards with FXO modules
17:02.09bkw_the wctdm driver does it all
17:02.15BoDePlOtthe util keeps "skipping FXO module" for each of them
17:02.22Hmmhesaysanthm: bad news there
17:02.24tzafrir_homebkw_, but you do for X100Ps
17:02.27yaaaranthm: could you shove that down my throat next?
17:02.38Hmmhesaysif you bust out the beer bong around a bunch of IT people, you are going to need a lot of beer
17:02.48zaptelanyone with experience with this problem/message pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel
17:02.57DarthClueHmmhesays: there will be enough beer for everyone.
17:03.02bkw_zaptel, its called frame slipage
17:03.03[Latre]tzafrir_home: modules zaptel and wctdm and for X100 wcfxo
17:03.14anthmkram is buying the whole cluecon beer and pizza speaking of
17:03.22bkw_yes yes
17:03.23Hmmhesaysa friend of mine once bonged a 750ml of hot 100
17:03.24tzafrir_home[Latre], lsmod |grep ^zaptel
17:03.25anthmtaking suggesttions on toppings
17:03.36anthmand brands
17:03.43tzafrir_homefeel free to paste one-liners like the output of that command
17:03.45[Latre]Span 2: WCFXO/0 "Generic Clone Board 1"    6 WCFXO/0/0
17:03.52[Latre]is channel 6 ?
17:04.01[Latre]fxsks=6 ?
17:04.04tzafrir_homeSeems so.
17:04.08tzafrir_homeright
17:04.11[Latre]weird
17:04.24Hmmhesaysbudweiser is my middle class beer of choice
17:04.42yaaarso, at the risk of sounding like a pussy and letting everyone know I'm trying out a web-interface....anybody round here use AMP, and/or know why amportal won't start asterisk? it just says it failed and to check /var/log/asterisk/full, which doesn't have anything in it from today at all.
17:04.45AyanoHmmhesays me too
17:04.59JerJerthen another thing that anonys the piss out of me was I was originally told that ClueCon was an Asterisk expo to acutally do development
17:05.10BoDePlOtheh i used amp until i witnessed firsthand the suckage of the whole thing
17:05.13tzafrir_home[Latre], but I'd try to unload both and then load them. You'll probably get it then in channel 5
17:05.41[Latre]tzafrir_home: works....   put in zaptel.conf fxsks=6 and ztcfg -vv and works
17:05.42anthmand um, yes that is the plan
17:05.47[Latre]5 channels configures
17:05.51[Latre]ed
17:05.51JerJerbullshit
17:05.55tzafrir_homeyaaar, amportal doesn't exactly fail to start asterisk. It is asterisk that fails to load
17:06.09Hmmhesaysamp can be alright if you know what the hell you are doing in the first place
17:06.12JerJerits YAOSC - Yet another open source convention
17:06.27[Latre]tzafrir_home: where is genzaptelconf  ?
17:06.28yaaartzafrir_home: uh, a rose by any other name....     but whatever, asterisk -vc starts it just fine, as does /etc/init.d/asterisk start.
17:07.09yaaarHmmhesays: that's why I played with asterisk and configured it all from the files and made everything work first. now I'm going to see if this will do what i want on a daily basis, and if not I'll go back to the configs
17:07.12tzafrir_homeyaaar, one guess: asterisk fails to run when run as non-root?
17:07.27*** join/#asterisk znoG (~gs@200.115.216.109)
17:07.28anthmFAOSCWDAD Finally An Open Source Convention With Actual Developers
17:07.31Hmmhesaysamp works fine for basic configs and the menu generator is nice
17:07.43yaaartzafrir_home: hmmm....i dunno. i've only tried to run it as root, and i'm running the amportal script as root as well
17:07.51[Latre]tzafrir_home: where is genzaptelconf  ?
17:08.01Hmmhesaysnewcastle is good
17:08.02tzafrir_home~genzaptelconf
17:08.02jbotit has been said that genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd.
17:08.05yaaartzafrir_home: how can i test that? and what would i do about it if that were the case?
17:08.12ManxPowernDuff: You only drink Duff Bear Oh Yeah!?
17:08.16ManxPoweror even
17:08.21ManxPowernDuff: You only drink "Duff Beer Oh Yeah!"?
17:08.28Hmmhesays"Duff man, can't breath OH NO"
17:08.33yaaaranthm: hmm....you can tell it's got actual developers just by the acronym they gave it!
17:08.59JerJerlol - we'll see how much actual development comes out of it
17:09.08JerJermy money is on very very little
17:09.33JerJerif not none
17:09.50ManxPowerJerJer: It's really a secret plot by the illimati to create a secret PBX that is really a combination of Asterisk, YATE, and Bayonne.
17:10.01mafkeeslol
17:10.12JerJeri wouldn't put it past them
17:10.12yaaarit will probably destroy us all
17:10.15ManxPowerI heard it will be called Asterbate!
17:10.26Nuggetheh
17:10.26BoDePlOti asterbated this morning
17:10.36znoGyou asterbator
17:10.37mafkeestmi
17:11.18mut^ never say that agai
17:11.21mutagain
17:11.59mafkeeswhy not ?
17:12.12mutreminds me of a 12yr old
17:12.23mafkeessorry
17:12.27BoDePlOtyou're into 12yr olds?
17:12.30BoDePlOtkinky
17:12.40mut..
17:12.41[TK]D-FenderTMI
17:12.42BoDePlOtand illegal in most states
17:14.13*** join/#asterisk DougRoyer (doug@70-67.69-92-cpe.cableone.net)
17:16.13mafkeesdinner time
17:16.15mafkeeslater yo
17:16.30anthmRight, I do no development for asterisk my 215 karma points I got, without being on the take nonthless, are all bullshit too
17:17.19anthmdamn, i'm busted
17:19.27*** join/#asterisk _deg_ (~deg@200.146.0.254)
17:19.49yaaartzafrir_home: does amportal attempt to run asterisk as user 'asterisk' ? that user doesn't have a shell, so I can't test whether it'll run. When I run it as a normal (non-root) user it tells me permission denied on various files, but they are all owned by asterisk and have owner perms that look ok.
17:20.26nDuffyaaar, you can tell su to use a specific shell, and so you can su to users without shells.
17:21.04BoDePlOthm
17:21.18BoDePlOtcant seem to get WCTDM module to load
17:21.24yaaarnDuff: the man page for su seems to disagree.......
17:21.26BoDePlOtmaybe that is my issue with fxotunne
17:21.50Secup_possible to execute more stuff after a DIAL ?
17:22.01Secup_or not causing hangup ..
17:22.01nDuffyaaar, from "su --help": -s, --shell=<shell>   use shell instead of the default in /etc/passwd
17:23.19BoDePlOtdo i have to manually compile wctdm?
17:23.21yaaarnDuff: hrm...must have a different version of su than you........if i do 'su --shell=/bin/bash' it tells me invalid option and then spits out the help, which does not contain that option
17:23.49tzafrir_homeyaaar, I have no idea about amp.
17:24.00*** join/#asterisk generalhan (general_ha@63.133.146.82)
17:24.34lehelyaaar: join amportal
17:24.45tzafrir_homeyaaar, generally you should make asterisk (the user) of various dirs,
17:24.55NuggetI don't have a "su --shell" or an "su --help" for that matter.
17:24.58Nuggeteven on my linux box.
17:25.08KattyNugget: (=
17:25.13KattyNugget: but linux is poo
17:25.21NuggetYes it is indeed.
17:25.35generalhanhas anyone run Asterisk on Fedora Core 4 ?
17:25.37*** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:25.52NuggetAsterisk doesn't care what flavor of linux you use, generalhan.
17:26.04nDuffKatty, who, me?
17:26.10Nuggetah.
17:26.11znoGnow, what are the chances a PSU fails slightly (enough to make your system unexpectedly crash every so often) after only 8 months of use?
17:26.15KattynDuff: well, there's only one person with your /nick (=
17:26.20Nuggetmy redhat box has whatever weird version of su nDuff has.
17:26.25KattynDuff: no, i'm talking about tzafir (=
17:26.27Seyrim using Cisco 7960 phones and need one that is less than $100.00US for other uses. Anyone know of one thats decent?
17:26.40Nuggetmust be a gnu thing
17:26.46Ayanodecent, no, cheap yes
17:26.52Ayanobudgetone
17:26.57*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
17:27.02nDuffNugget, 3rd-party opinion: Me, bitter?
17:27.09SeyrAyano: the Grandstream one?
17:27.18NuggetI'd rather have a wheel group than a --help option.  :)
17:27.20nDuffIf you don't need speakerphone, the Sipura SPA-841 is better than the Grandstreams.
17:27.56*** join/#asterisk da_monumental_1 (~da_monume@rrcs-24-172-102-198.midsouth.biz.rr.com)
17:27.57Hmmhesays:D
17:28.00tonymorellaJust changed out three X100P cards for a Wildcard TDM400P with three X100M and one S100M mods. Now for some reason 30% of the calls are not sending out the 1st digit IE "1". We are using Grandstream BT-100 VoIP phones and have tried both * stable v1 and the latest CVS. Comments?
17:28.01nDuffNugget, a wheel group is just a groupadd and some /etc/pam.d/* edits away...
17:28.03Seyrim looking more for reliability than features for a <$100.00 phone
17:28.45lters_Seyr, ip301 ? $117 on ebay new
17:28.46nDuffSeyr, then the SPA-841 is your best bet, as long as you don't need speakerphone. The Grandstreams (even their "enterprise" phone) are flaky as hell.
17:29.45Seyrthanks people
17:29.55Darwin35who did it
17:30.02Darwin35I want to know who is guilty
17:30.14Darwin35confess and we will only castrate you
17:30.25AyanoThen go with the sipura
17:30.27KattyHmmhesays: i cut my paw :<
17:30.37Hmmhesaysoh noes
17:30.43Kattyi know :<
17:30.48Darwin35seyr get a x401 from eezeephone.com
17:30.51Katty:>
17:30.55Kattyk, all better.
17:30.56Seyranyone know if those Linksys ATAs (PAP2) are only shipped with Vonage?
17:31.00Hmmhesaysthat stuff works wonders
17:31.00Darwin35diff company but only 69.99
17:31.14Darwin35better then the grandstreams and it does iax
17:31.20KattyHmmhesays: they're also refusing to give me my phone. apparently it was billed to the company instead of me ;)
17:31.21ManxPowerSeyr: Those "linksys ATAs" were designed by SIPura
17:31.41Hmmhesayssounds like a pain, maybe I shouldn't be jealous of your beep beep capabilities
17:32.02Kattyevery mobile company is a pain
17:32.12Kattyi'm just glad /i/ don't have to deal with them ;)
17:32.16Hmmhesayscellone has been nothing but good
17:32.30Kattycellone?
17:32.35Hmmhesaysnear 2 years I've had them
17:32.38Kattyis that...cingular or something?
17:32.53Hmmhesaysno, they are their own company
17:32.57Kattyneat.
17:33.03lters_Seyr, sipura 2001 gets u 2 analog ports... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5790779163
17:33.21Hmmhesaysthey are a smaller company, but service is real good
17:33.28Kattythey're not a reseller then?
17:33.48Kattythey have their own gsm network, etc.
17:34.04Hmmhesaysyeah they have their own network
17:34.04lters_Katty, *every* mobile company...
17:34.09Hmmhesayshttp://www.cellularonewest.com/images/maps/NMAP_07_04.gif
17:34.13Seyrthanks Iters
17:34.21*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
17:34.24KattyHmmhesays: (=
17:34.37lters_Seyr, than u can do cordless walmart phones or whatever :)
17:34.39generalhananyone know a good VoIP provider that sells Big Bulk minute blocks ?? like 20,000 ?
17:34.51Kattyasterlink might
17:35.10Hmmhesaysfor just termination generalhan?
17:35.10Kattybkw_: does asterlink sell minute blocks in bulk?
17:36.20generalhanwell like, right now our ISP provides our call switch and hardware and we get 28,000 minutes a month to use ... but if i move over to asterisk i need a VoIP provider that can do the same amount of a block for me
17:36.33*** join/#asterisk fugitivo (~ajf@201.255.99.157)
17:38.09fugitivohello
17:38.20Hmmhesaysyo
17:38.52*** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:42.26*** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net)
17:43.24HmmhesaysI'm still looking for the winamp classic sdk, I can't find that darn thing anywhere
17:44.40twisted[asteria]Katty, sorry, couldn't answer, on phone with customer
17:44.54Kattytwisted[asteria]: no big (=
17:45.03Kattytwisted[asteria]: i'm not one of those rude people who start talking when you don't answer your beep beep
17:45.33twisted[asteria]Katty, heh.. you're the first person i've encountered that uses the alert rather than just a simple quip though
17:46.07Kattyquip?
17:46.17twisted[asteria]well, keying it up and letting it go
17:46.27Kattywha?
17:46.29twisted[asteria]sends the two simple beeps on the other side rather than the "beep beep beep beep beep" thing
17:46.33twisted[asteria]like this
17:46.35tzangertwisted[asteria]: huh?
17:47.05Kattyoh.
17:47.08twisted[asteria]Katty, ;)
17:47.12Kattyk
17:47.27twisted[asteria]it's not a biggie :)
17:47.32twisted[asteria]but can't answer agian, phone
17:47.35Kattyk
17:47.43So3krisyes my x100p works
17:47.51mafkeesback
17:47.52*** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com)
17:48.04So3krismafkees: are you dutch
17:48.09mafkeesyes
17:48.13So3krisahaa
17:48.14So3krisme2
17:48.17mafkeesah
17:48.21mafkeesnice to meet ya
17:48.22mafkees:)
17:48.27So3krisdat praat een stuk beter :D
17:48.34BoDePlOtshould i be able to insmod wctdm?
17:48.45MiccAnyone know much about iaxclient?
17:48.54mafkeesBoDePlOt: why not use modprobe ?
17:49.06MiccI've got a strange issue with it cutting out when the cpu spikes.
17:49.09BoDePlOtwell let me back up a bit
17:49.11xtrvdWhat method can I use to keep a list of telephone numbers, and when one of them calls in, I have a custom CallerID show up on my side (SIP phones)?
17:49.16BoDePlOti was running old asterisk version'
17:49.19MiccIs that something that a jitterbuffer can fix?
17:49.20ronnhi guys.. I lose the ACCOUNTCODE variable after a call to DISA() .. any idea /
17:49.20*** join/#asterisk mut (~animenodv@65.111.201.79)
17:49.23BoDePlOti updated yesterday
17:49.33BoDePlOti want to load wctdm (which previously didnt exist)
17:49.38ronnis that expected behavoiur?
17:49.41Nuggetxtrvd: LookupCIDName()
17:49.48BoDePlOtso i can use fxotune to turn on echo cans on my boards
17:50.07BoDePlOti rebuilt the source, but wctdm doesnt seem to be compiled
17:50.08*** join/#asterisk pa (~Paolo@pa.user)
17:50.19BoDePlOtthough the source is there
17:50.27BoDePlOtso hence modprobe wctdm fails
17:50.34xtrvdNugget: Where can I tell it what to read from? (The wiki doesn't seem to have much info regarding the database)
17:50.35mafkeesdid you modify the Makefile ?
17:50.38Kattyyeah, my modprove wctdm fails too
17:50.47Nuggetthe wiki has plenty of information on the subject
17:50.47ronnhi guys.. I lose the ACCOUNTCODE variable after a call to DISA() .. any idea ?
17:50.48ronnis that expected behavoiur?
17:50.49Kattyi have to insmod instead
17:50.55ManxPowerronn: that was fixed in the past few days
17:51.12BoDePlOtKatty where do you insmod it from? insmod fails on mine
17:51.15ManxPowerronn: Maybe only in CVS-HEAD, I don't recall for sure.
17:51.22KattyBoDePlOt: uhh?
17:51.26syle2whats that site that offers US dids for 5 dollars a month?
17:51.27KattyBoDePlOt: what do you mean where from?
17:51.27mafkeesBoDePlOt: in the zaptel Makefile is a line that states all the modules to be built
17:51.34BoDePlOt<PROTECTED>
17:51.34BoDePlOtinsmod: wctdm: no module by that name found
17:51.41ronnthanks Manx ... will update CVS
17:51.45KattyBoDePlOt: modprobe zaptel, insmod wctdm, /sbin/ztcfg, asterisk -vvvvvvvvvc
17:51.51mafkeesBoDePlOt: if you do a 'make install' it will place the modules in the correct /lib/modules dir
17:52.33BoDePlOtmake: *** No rule to make target `install'.  Stop.
17:52.53BoDePlOti'm a *nix moron so please forgive me
17:52.54BoDePlOt=P
17:53.20lehelso what does ztcfg -vv tells you? BoDePlOt
17:53.26*** join/#asterisk wunderkin (~kev@12-215-218-160.client.mchsi.com)
17:53.39mafkeesBoDePlOt: where did you get the zaptel sources from ?
17:53.43BoDePlOti did make install in src/zaptel
17:53.50BoDePlOtand it didnt touch wctdm at all
17:54.09BoDePlOtmafkees: i got them from cvs-head
17:54.14lehel;) you should make
17:54.29lehellsmod |grep wctdm
17:55.01mafkeesBoDePlOt: I just did a cvs checkout and it compiles wctdm here
17:55.18BoDePlOti used the shell scripts in /src
17:55.18*** join/#asterisk gaffneyc (~gaffney@70.88.90.25)
17:55.27BoDePlOtto download and build
17:55.33lehelBoDePlOt: modprobe zaptel
17:55.42BoDePlOtyeh zaptel is there
17:55.49tzafrir_laptopwhy a separate modprobe for zaptel?
17:56.04mafkeestzafrir_laptop: I had to do that too with the qozap module
17:56.13*** join/#asterisk gwynpen (~gwynpen@p54AAD640.dip.t-dialin.net)
17:56.20mafkeestzafrir_laptop: modprobe qozap is suppost to autoload zaptel too
17:56.26mafkeesbut it freezes my debian machine
17:56.39NuggetLinux is poo.
17:56.44mafkeesand modprobe zaptel && modprobe qozap fixed the freezes
17:56.47*** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net)
17:56.56mafkeesNugget: I agree, but no zaptel on other platforms
17:56.58tzafrir_laptopI hardly call it a fix
17:57.01tzafrir_laptop:-(
17:57.11mafkeestzafrir_laptop: me neither, but at least it works
17:57.37tzafrir_laptopwhat kind of freeze?
17:57.44mafkeestotal
17:57.44tzafrir_laptopany oops?
17:57.54mafkeesnope, just *dead*
17:58.01tzafrir_laptopanswers pings?
17:58.03mafkeesnothing in logs after reboot
17:58.05mafkeesnope
17:58.08mafkeestotally dead
17:58.14mafkeeslike a kernel panic
17:58.27mafkeesbut nothing is printed on console nor in logfile
17:59.08tzafrir_laptopmafkees, Any idea if this happens on systems without qozap hardware?
17:59.13*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
17:59.47mafkeestzafrir_laptop: my home system vorks fine with an X100P without preloading zaptel
18:00.09mafkeestzafrir_laptop: and that's the only 2 pieces of zaptel hardware I use(d)
18:00.18Darwin35is parking hardset at exten 700 or can it be moved
18:00.20nitrammafkees: just installed qozap on a debian machine today and did not have to preload zaptel
18:00.24mafkeesthe ztdummy needed zaptel already loaded too btw
18:00.42mafkeesuntil I upgraded to Debian Sarge
18:00.46tzafrir_laptopztdummy doesn't pull zaptel?
18:00.51tzafrir_laptopok
18:01.04mafkeesztdummy wasn't pulling zaptel on my woody system
18:01.08Darwin35you have to remove a # sign in the Makefile
18:01.08mafkeesbut it does on sarge
18:01.16Darwin35to get iut to compile ztdummy
18:01.26Darwin35sarge uses rtc
18:01.27mafkeesDarwin35: I know ;)
18:01.47generalhancan some one tell me how to go about picking up on of the T100Ps for my server ?
18:02.00tzafrir_laptopDarwin35, or use my packages, where building it is enabled by default
18:02.19[TK]D-Fendergeneralhan : Picking up on?
18:02.19mafkeestzafrir_laptop: I now see qozap works great without preloading zaptel on Debian Sid too
18:02.20outtolunchttp://www.digium.com/   click on 'order'
18:02.36generalhanhow much are they running right nmow ?
18:02.47Darwin35I am working on a realtimeclock for bsd to use to drop ztdummy
18:03.00mafkeestzafrir_laptop: does your package contain the bristuff patch ?
18:03.00FarrisGOk, so this is very strange. Incoming calls from outside the * server sound fine on this BT101, but if you try to PLACE an outgoing call to a number outside the * server, the audio is garbage, like it's using the wrong codec
18:03.02[TK]D-Fendergeneralhan :About 570$USD for the TE110P
18:03.11*** join/#asterisk craziman2 (~Craziman2@boromir.apid.com)
18:03.12mafkeesDarwin35: that will be super
18:03.29mafkeesDarwin35: that way I can use * at OpenBSD finally
18:03.39Darwin35you can now
18:03.44Darwin35with ztdummy
18:03.49Darwin35its in the ports
18:03.55mafkeesin OpenBSD ????
18:03.57fugitivozaptel doesn't work with openbsd
18:03.59mafkeesor Freebsd ?
18:04.03Darwin35yes
18:04.10mafkeeshuh ???
18:04.12Darwin35zaptel works on bsd
18:04.16generalhank thank you
18:04.18Darwin35we have themm in ports
18:04.25fugitivosince when?
18:04.28mafkeesDarwin35: yeah, FreeBSD
18:04.30mafkeesnot OpenBSD
18:04.31Darwin35long time now
18:04.35brookshireyeah.. surpised me
18:04.36fugitivonot openbsd
18:04.39fugitivojust freebsd
18:04.51Darwin35not open I thought open bsd ported the freebsd ports
18:04.59craziman2I have a Cisco 7960 phone talking to my * box.... the phone has 2 different lines provisioned on it... if one line is in use then I would like the other to not ring... any ideas?
18:05.01mafkeesno way
18:05.20filecraziman2: groups.
18:05.22mafkeescraziman2: use setgroup and checkgroup
18:05.26*** join/#asterisk DougRoyer (doug@70-67.69-92-cpe.cableone.net)
18:05.51craziman2thanks
18:06.19Darwin35well my embedded system works
18:06.32Darwin35not just to deside on a case for it
18:06.33hardwirehola
18:06.55Darwin35and it has wireless in it
18:07.11Darwin35and the zyxel phones work
18:07.34*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
18:07.49hardwireDarwin35: Geode stuffins
18:07.51hardwire?
18:08.40Darwin35this is all fbsd based . and m-systems disk on a chip setup
18:09.27Darwin35it has festival and sphinx and spandsp fax
18:09.50hardwirewhat processor?
18:09.51Hmmhesaysanyone play around with ooh323?
18:09.54hardwirerather.. what sbc?
18:10.06hardwireHmmhesays: this isn't the right channel for that I think.
18:10.21Hmmhesaysthen I suggest you read the wiki again
18:10.22Hmmhesays;)
18:10.34Darwin35duron 1.4
18:10.41hardwireDarthClue: how is that embedded?
18:10.48*** part/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
18:10.51hardwirecause I would like some embedded durons :)
18:11.16Darwin35www.tri-m.com
18:11.26Darwin35look at thier boards
18:11.33hardwireah
18:11.40hardwireI have seen their FFD's but not their SBCs
18:12.10Darwin35get a catalog it has some new boards not listed on the sight
18:12.12harryvvnice lcd vidio
18:12.20*** join/#asterisk Coriantum (~asdfkle@67-41-182-243.slkc.qwest.net)
18:12.21hardwireDarwin35: what is your board?
18:12.42hardwirehttp://www.tri-m.com/products/icp/rocky772ev.html
18:12.53CoriantumCould someone tell me how to cut 4 characters off of a variable without using Cut?
18:13.03hardwireoff the end?
18:13.09Coriantumyeah
18:13.18hardwirecheck out the example dialplan
18:13.21Coriantumthe length changes too
18:13.23hardwirewhere it strips the MSD
18:13.27Darwin35its a tmz104 but a new one not listed on the sight
18:13.42hardwireinteresting
18:13.47*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
18:13.49hardwirewhere you going to throw it Darwin35?
18:13.57harryvvhardwire thats like the old days with the daughterboard
18:14.58Coriantumhardwire: that's for taking it off of the beginning, not the end
18:15.12hardwirehttp://www.tri-m.com/products/engineering/bat104sla45.html
18:15.13hardwirecool
18:15.15Darwin35well I want to make a unit that can be put in mobiehomes
18:15.26hardwireCoriantum: oh yeah :)
18:15.48Darwin35liek then one I am going to look at
18:16.00hardwireover starband eh?
18:16.10Darwin35loads of uses for it
18:16.17muto yea hardwire
18:16.17Darwin35you could
18:16.19muti was going to ask
18:16.20hardwiremight I suggest using speex at a low bitrate VBR then
18:16.21anthmif it were logical you could use neg numbers you try that ?
18:16.25Darwin35no hard drive
18:16.27muthow do you guys power those radios in the middle of bumfuck
18:16.29Darwin35what a waste
18:16.38hardwiremut: solar/gas
18:16.39Darwin35use a network storage drive
18:16.40anthm${VAR:-4:4} if that doesnt work it should
18:16.52mutthat what those barrels were?
18:17.02hardwiremut: those are water weight
18:17.12hardwirekeeping the dish from blowing away
18:17.29mutah
18:17.57Darwin35just hope to find some good open hotspots when I use it
18:18.34kFuQPrivacy Guru Locks Down VoIP
18:18.35kFuQhttp://www.wired.com/news/technology/0,1282,68306,00.html?tw=wn_story_top5
18:18.42kFuQhmm.... pgp with voip
18:19.00kFuQZimmermann has developed a prototype program for encrypting voice-over IP which he will announce tomorrow during a presentation at the BlackHat security conference in Las Vegas.
18:21.20hardwirekFuQ: pgpfone sounds fun all by itself
18:21.29Darwin35ok just orderd a pac-53h case
18:21.33hardwireI just want sips support
18:21.42hardwireDarwin35: I like the can-tainers
18:21.46hardwirethey are sexy.
18:22.36Darwin35yeah I have my eye on one but they are 200 bucks
18:22.43Darwin35they need to coe down
18:22.49Darwin35come
18:23.30harryvvhi darin whats up?
18:23.33*** join/#asterisk fluidicsl (~asdf@adsl-63-200-54-51.dsl.snfc21.pacbell.net)
18:23.37kFuQhardwire: yah
18:23.50generalhanwhat company is everyone using for VoIP minutes ?
18:23.53fluidicslI am trying to make a .call file for asterisk and I am confused on how to specify the target number
18:24.28netnameusi use voipjet
18:24.30harryvvdarwin35 you seem to spend alot of time here what have you done with asterisk
18:24.34DarthCluegeneralhan: asterlink
18:24.43generalhani need a good VoIP comany that can supply my with about 20,000 bulk minutes per month
18:24.54harryvvnice chunck of time
18:25.01JerJerDarthClue:  nothing like more shameless self promotion
18:25.02Darwin35I ported it to bsd  and have spent time keeping it there along with getting other parts to work
18:25.04generalhanwell i have 30 users who are on the phones ALL DAY'
18:25.12DarthCluegeneralhan: inbound / outbound?  source/destination?
18:25.16ManxPowergeneralhan: Get a local PSTN line.
18:25.24harryvvDarwin35 thats cool
18:25.25ManxPowerA T-1 or E-1 PRI would do it.
18:25.33generalhanmore outbound than inbound, but the inbound will be to our Toll-Free
18:25.52harryvvManxPower found my problem yesterday. it was in the ata.
18:26.03DarthClueUS only or do you need international / canada / mexico as well?
18:26.04fluidicslI am trying to make a .call file for asterisk and I am confused on how to specify the target number
18:26.21ManxPowerfluidicsl: Channel: Zap/g1/5551212
18:26.24generalhan99.9% USA, every once and a while we get a Canada
18:26.43[TK]D-Fendergeneralhan : www.unlimitel.ca
18:26.50fluidicslI want to call with a sip channel
18:27.05fluidicslI mean I want it over my voip not pstn
18:27.06JerJerfluidicsl:  then change ManxPower's example accordingly
18:27.14ManxPowerfluidicsl: Channel: SIP/5551212@sipconfentry
18:27.23fluidicslah ok
18:27.38JerJeryou have to be smarter than what you are working on
18:27.54mutNEVAR
18:28.16ManxPowerHmmm...I either ran out of cigs or forgot where I put them.
18:28.21gaffneycWhat driver should be used for the TDM400P board? I have seen both wcfxs and wctdm (which is marked as the new driver). But my Gentoo install doesn't include wctdm. Any ideas?
18:28.35JerJerdepends
18:28.54ManxPowergaffneyc: what version of Asterisk are you using?
18:28.56fugitivogaffneyc: wctdm for 1.0.8
18:29.00yaaarthis really bites. anybody know why i can start asterisk with 'asterisk -U asterisk -G asterisk -vvvvvvvc' but can't with 'amportal start' ? it just says "failed to start, check /var/log/asterisk/full' but that file doesn't have anything from today in it.
18:29.06ManxPowerfugitivo: WRONG!
18:29.16JerJeryaaar: don't run AMP
18:29.20JerJerproblem solved
18:29.21ManxPowergaffneyc: look in the README for the zaptel source you are using.
18:29.24fugitivoManxPower: isn't wctdm the new one?
18:29.27gaffneycI've tried both 1.0.8 and 1.0.7 and it was not found in either
18:29.37JerJerwctdm is -head only
18:29.43yaaargee JerJer thanks. I don't know what I would do whithout you.
18:29.43ManxPowerfugitivo: wctdm is the name in CVS-HEAD.  wcfxs is the name for 1.0.x
18:29.44Darwin35brb
18:30.19fugitivoManxPower: is wctdm included in 1.0.x?
18:30.22gaffneycOk... so wctdm will be the driver in the next major release?
18:31.13JerJerdepends
18:31.19ManxPowerfugitivo: no.
18:31.32ManxPowerI think 1.0.x puts alias wctdm wcfxs or something like that
18:31.37fugitivoi have wctdm in my 1.0.8
18:31.51ManxPowergaffneyc: they are the same driver, just the name changed
18:32.03JerJerbecause the FXO module was released
18:32.15ManxPowerfugitivo: maybe they started symlinking it or you have something left over from CVS=HEAD
18:32.16JerJerand loading wcfxs wasn't intuitive for people
18:32.29gaffneycManxPower: Makes sense, thanks.
18:32.40gaffneycIs there a change log available online from 1.0.7 -> 1.0.8? Is it worth using 1.0.8 over 1.0.7 as 1.0.8 is currently masked in portage.
18:32.56*** join/#asterisk pifiu (~myassisbi@208.205.181.170)
18:34.11*** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl)
18:34.14JerJerhow about ChangeLog file in the source tree?
18:34.19generalhananyone used Voice Pulse for service before
18:34.26yaaargaffneyc: i think you'll find you've got to unmask it.....seems like the versions of some of the deps don't work out right if you try to do it without ~x86
18:34.43fugitivoManxPower: you're right, i'm using wcfxs, maybe i dream about wctdm, lol
18:34.46pooh_Hi all, sorry to be off topic, but does anybody know what is goin on with the bf2 accountservers pls?
18:34.47yaaargaffneyc: i've got 1.0.8 with ~x86 versions of the various masked deps, and it's solid as a rock
18:35.18fugitivopooh_: battlefield 2?
18:35.49gaffneycyaaar: Thanks for the advice, looks like 1.0.8 it is
18:35.57yaaarnp
18:36.02yaaargood luck
18:37.05yaaargaffneyc: this may save you a minute or two...you'll need ~x86 in package.keywords for these too: net-misc/zaptel, net-libs/libpri, net-libs/zapata
18:38.30fugitivogaffneyc: add "net-misc/asterisk ~86" to /etc/portage/package.keywords
18:38.38fugitivosorry, ~x86
18:38.51fugitivoif not, you'll break asterisk if you do emerge world
18:39.01gaffneycThanks, it's already being re-emerged
18:39.09JerJerwhy let someone break things for you?
18:39.19JerJercheck out out of cvs and compile it yourself
18:39.27gaffneyclol
18:39.30JerJerits not that hard
18:39.36*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
18:39.50gaffneycIs there an change log online though? I'm interested in seeing what has changed from version to version
18:39.51fugitivoJerJer: why? emerge is easier
18:40.17JerJerthen do you really know what you are getting?
18:40.47yaaardoesn't the ebuild just pull from cvs anyway?
18:40.48fugitivoJerJer: yes and no, it's the same with cvs
18:41.06yaaarJerJer: by that logic I should be downloading the source for everything on my system and compiling it manually
18:41.14JerJeryes you should
18:41.15JerJeri do
18:41.25yaaarthat's great JerJer.
18:41.35fugitivoJerJer: that's why gentoo was released
18:41.39yaaarthe rest of us have better things to do
18:41.39fugitivoJerJer: :)
18:41.53JerJeris your OS 15 megabytes total ?
18:42.03fugitivoJerJer: my firewall is
18:42.10JerJeri can squeeze it down to 9 meg without too much trouble
18:42.20solargaffneyc: emerge -pvl packagename ; # will show you what has change between the installed revision and the update
18:42.26*** join/#asterisk Pooter (Fleb@24.181.176.181)
18:42.46fugitivoJerJer: if i want to use a compactflash for my os, i can do it
18:42.53*** join/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com)
18:43.09fugitivoJerJer: but for normal tasks, i use emerge
18:43.14JerJeri use CF all day long
18:43.18yaaarJerJer: I don't really care to make it that small. The smallest hard drive you can buy is huge these days. I'd rather have a system that is managable and which can be upgraded without too much work
18:43.32JerJerlol - too much work ?
18:43.37Pooteranyone noticing a problem with audio degradation with nufone?
18:43.40*** join/#asterisk hardwire (~hardwire@209-112-147-72-cdsl-rb1.nwc.acsalaska.net)
18:43.40yaaarI mean, what would I gain that's more valuable than bragging rights?
18:44.00*** part/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
18:44.04hardwirehi
18:44.26fugitivoJerJer: asterisk boxes with CF?
18:44.29JerJerwhy waste valueable space with code that will never get executed ?
18:44.34JerJerfugitivo:  most certainly
18:44.48JerJerleave the HD space for voicemail or config files
18:44.51yaaarJerJer: yeah. Too much work. As in, my time is worth money, and I'm not wasting it compiling my whole system from scratch and keeping track of security issues and everything else when I can just run emerge -uDav world instead
18:44.55JerJerlol
18:44.57fugitivoJerJer: yes, that's the way
18:45.14JerJerit took me a whole day to roll my own distro
18:45.43JerJerwhat security issues?  there isn't much on there to be exploited
18:45.44yaaaryeah, ok, and it'll take you significant time throughout the life of the system to keep it secure and up to date
18:45.56yaaarlike the next time there's a remote root in sshd
18:46.07JerJerwho says i use sshd ?
18:46.19JerJerand who says its hard to update ?
18:46.20*** join/#asterisk jarrod (jarrod@juniperyour.net)
18:46.33mutyou emerge daily then yaaar?
18:46.43*** join/#asterisk meppl (~mephisto@84-245-169-14.ipool.celox.de)
18:46.58yaaarwhatever. this is stupid. i'm not going to sit around here wasting the aforementioned valuable time trying to convince someone of the value of having a real system instead of some homebrew bolt-together.
18:47.04yaaarmut: most days
18:47.04PooterJerJer: do you have any updates with the nufone issues, i would call the nufone number but no one seems to pick up
18:47.06jarrodis realtime extensions the best way of syncing extensions configuration across a multiple node asterisk network?
18:47.15FarrisGcan someone help me troubleshoot this production problem that just occurred? The * server went down, and I restarted it. Now everything works internally, but inbound calls are not hearing any audio or getting properly routed, though the console log says they are
18:47.25mutheh
18:47.51JerJerPooter:  updates?
18:48.13JerJerGreg and hayzell answer our number as much as they can
18:48.21JerJerwe have more than just one customer, ya know
18:48.23FarrisGActually, it appears that NO audio will work. If I dial out, I hear a ring (though that's probably fake), but then when it's picked up I get nothing
18:48.33PooterJerJer: I keep getting choppy audio, even when I call the nufone number its choppy from a lanline phone - I did get somoene a while ago who said you were having issues with asterisk
18:48.51JerJerso you take their word?
18:48.53harryvvFerris, can you call internally from phone to phone?
18:48.56JerJerhave you bothered to do a traceroute ?
18:50.26*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
18:50.26*** topic/#asterisk is Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted
18:50.28FarrisGharryvv: Not likely. Firewall hasn't changed, and this was all working 5 minutes ago
18:50.32PooterJerJer: I JUST said I spoke to someone a while back, I'm looking for " updates ", which is why im asking you since I cant get a hold of anyone
18:50.36*** part/#asterisk tonymorella (~tony@70-33-152-98.agstme.adelphia.net)
18:51.01FarrisGharryvv: And the outgoing calls go out fine. If I call my cell phone, the cell phone rings, but when I pick up there's no audio
18:51.04pifiuhold on brb jarrod, i would appreciate it if you help me
18:51.04PooterJerJer: Also the 248-724-VoIP # is choppy from a landline
18:51.13FarrisGlikewise, if I call the * server from my cell phone, I get no audio
18:51.38jarrodpooter: probably poor latency/bandwidth
18:51.41jarrodto the softswitch
18:52.00pifiujarrod did you get my IM?
18:52.07jarrodno?
18:52.28pifiui sent you a private message
18:52.28jarrodpifiu: do you have the example configs for the polycoms?
18:52.41pifiuyes and no but hold on let me finish this phone call and we can keep talking
18:52.46pifiuhow much longer are you going to be on?
18:52.46jarrodyou'll need to modify them for the mac addresses and sip information for each phone
18:52.49pifiui need like 30 minutes
18:53.02oejAnyone here with PUBLISH problems in chan_sip HEAD?
18:53.21zoanot me oej
18:53.44hardwirepolypaudio just made my damn week
18:54.00pifiujarrod send me a private message
18:54.03[TK]D-Fenderharryvv : I won my * bid with ScopServ's solution :D
18:54.44zoasomebody here with a lot of signalling etc on E1/T1s ?
18:54.59loudThat scopserv panel is GOOD, but really expensive man.
18:55.45loudi talked yesterday with them, 4500 dlls, (not the itsp, the previous).
18:56.07harryvvTK, fantastic and the phones comming from CCP?
18:56.29[TK]D-FenderYup
18:56.32harryvvgoood
18:56.33jarrodthat panel is per server, yes?
18:56.33[TK]D-FenderCCP wins it.
18:56.43loudper server. yes.
18:56.45[TK]D-Fenderjarrod : AFAIK
18:56.55loudno matter if you have a centralized management.
18:57.16harryvvThats very good news
18:57.17harryvv:)
18:57.18zoai will release an open source queue log analyzer this week
18:57.20jarrodi need a better way of syncing extensions with multiple servers..
18:57.26jarrodi guess im going to go realtime with the sql backend
18:57.27[TK]D-FenderI got to demo a full server here live.  SWEET.  These guys are aiming for turn-key mass-production, and they should annihilate the competition.
18:57.34JerJerrealtime is not the answer
18:57.36harryvvohh
18:57.38zoaand maybe also a switchboard
18:57.47loudyes, that demo will work for 15 days
18:58.01pifiuok jarrod here is what i need to do
18:58.03harryvvTK, is there server anything special?
18:58.10loudthey have some sort of digium licence server or those, itneresting stuff
18:58.11jarrodjerjer: how come?  they would all read from the backend
18:58.16pifiui know you set it in the phone to look for the config files in an ftp server and such
18:58.38pifiubut i need to write in that config file for in the future to look somewhere else, such as http
18:58.51pifiuso that i can just edit one image and it applies to all of them
18:58.53pifiueasier
18:58.53pooh_fugitivo: yes, bf2, I have no other way to check what is going on
18:58.58jarrodpifiu: oh i did not know it supported http
18:59.00JerJerjarrod:  realtime is just a hacked implementation
18:59.06pifiuthe new boot loader does
18:59.10oejzoa: Cool, queue analyzer
18:59.16bkw_two words.. Sandy vagina!
18:59.19zoaand very nice looking :)
18:59.34bkw_JerJer, realtime just returns an astconfig object just like config files do
18:59.57bkw_it is really no different than reading from flat files vs database
19:00.00JerJerand did I say the astconfig object was any good?
19:00.01blitzragezoa: yo!
19:00.10zoahey blitz
19:00.12bkw_well you seem to think that its fine for flat files
19:00.14jarrodwell its what we have to work with
19:00.15ManxPowerI'll use RealTime at some point.
19:00.22JerJerbkw_:  you are putting words in my mouth
19:00.42jarrodits easier than me syncing my ael's across multiple servers and reloading extensions
19:00.45bkw_well you say you like flatfiles right?
19:00.52JerJerhave i said that?
19:00.57bkw_in the past you have
19:01.01JerJerno
19:01.09JerJeri've said realtime is not the answer
19:01.19bkw_yes you have you have sung the high praise of of using a perl script or something to write out .conf files on disk
19:01.24JerJerum no
19:01.25jetsflat files are okay and have there place, much like simple batch files or simple shell scripts have a place.
19:01.27bkw_um yes
19:01.31JerJerprove it
19:01.35bkw_I totally recall you doing that
19:01.38JerJeri have my own config handler
19:01.46JerJerthat panny and I wrote
19:02.02jarrodjerjer: what does yours do
19:02.11bkw_the end result is still the same
19:02.14JerJerconfigures asterisk
19:02.16bkw_no matter if  you use your own config handler
19:02.23bkw_its still ends in the same form in the core
19:02.27JerJerits not realtime nor is it flat files
19:02.56fugitivoJerJer: database that generates flat files?
19:03.09*** part/#asterisk lehel (~Lehel@82.79.20.17)
19:03.10bkw_well I do recall jerjer saying he used a script in the past to do that
19:03.24bkw_JerJer, why are you so bitter and hateful today?
19:03.31file[laptop]lol
19:03.34*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-80-43-50.red.bezeqint.net)
19:03.40JerJerfugitivo:  no
19:03.42JerJerbkw_:  no
19:03.58JerJerbkw_:  you tell me
19:04.05bkw_tell you what?
19:04.37JerJerwouldn't you be pissed off when someone used your company name without permission?
19:04.41jarrodwhy do you feel realtime is not the answer?
19:04.48JerJerand the other party didn't bother to verify anything
19:04.49jarrodi would assume the handler you use is also a 'hack'
19:04.52jarrodas you would call it
19:05.00JerJerits not realtime
19:05.16JerJermy solution doesn't depend on the database to operate
19:05.24yaaarJerJer: could you maybe focus on the trademark infringers then? and, like, quit being a dick to the rest of us?
19:05.35JerJerleave if you don't like it
19:05.42JerJerthis is irc yo - deal with it
19:05.42bkw_yaaar, it was his employee that is speaking at Cluecon
19:05.43jarrodi see nothing wrong with querying a database.. many applications require a database and allows for centralized configuration.
19:05.57jarrodthe only thing necessary is to maintain direct network connectivity to the machine/cluster
19:06.00JerJerbkw_:  without any permission - who simply asked for a time slot
19:06.02bkw_yaaar, so I put exactly what greg gave me to pu tup
19:06.04yaaarhey, look, it's not bad enough to make me want to leave or anything....I was just asking
19:06.18JerJerbkw_:  after you and I talked about the specific subject
19:06.35JerJerwhich tells me how desperate you are for speakers
19:06.41bkw_Nope
19:06.47bkw_If he don't show up .. big deal.
19:06.52bkw_I'll take care of it
19:07.12tessierbkw_: JerJer? Bitter and hateful? No way. JerJer is all sunshine and flowers all the time.
19:07.22bkw_in what alternate universe?
19:07.33tzangerjarrod: I haven't seen a good burning NEED for realtime
19:07.36FarrisGThis is very aggravating. Something is stopping audio for inbound or outbound calls, but the log just shows timeouts. Where should I start?
19:07.49JerJerFarrisG:  firewall
19:07.57anthmactually, we had to add time and shorten some of the slots to accomidate all the speakers
19:08.09jarrodfarrisg: put a phone on the outside and try it again
19:08.15JerJeryeah only after the whole concept of the conf changed
19:08.17FarrisGJerJer: I have already checked the firewall, and even brought the server outside the firewal
19:08.20tzangertessier: *snorts*
19:08.21PoWeRKiLLHi all :)
19:08.32FarrisGJerJer: With or without firewall, same issue
19:08.45JerJerFarrisG:  tcpdump - see if frames are acutally showing up
19:09.16FarrisGJerJer: yes they are
19:09.17dantApr 06 04:10:58 <JerJer>        what's wrong with flat files?
19:09.33yaaarFarrisG: what kind of phone? is silence-suppression on?
19:09.35JerJernow those few early members that have commited to speak now feel obligated
19:09.55JerJerdant: show the rest of the context, smart guy
19:10.13dantJerJer, sure, I will as soon as I find that grep'd line
19:10.26yaaardant: try with grep -A 10 -B 10
19:10.26jetshrm
19:10.42FarrisGyaaar: Three different phones. X-Ten, BudgeTone 101, and a Polycom 501
19:10.43jarrodman from where i'm sitting realtime is one of the best options available
19:10.50yaaarFarrisG: ouch
19:11.02FarrisGSilence suppression is not on, but turning it on affects nothign
19:11.05bkw_jarrod, just use realtime.. we do with the perl config handler.. works great
19:11.05JerJerjarrod:  and it is my opinion that realtime is not the answer
19:11.09dantApr 06 04:10:28 <eric>  I'm using the flat files and I read this could all be stored in a db and I'm not sure how to make that happen
19:11.15harryvv<PROTECTED>
19:11.16harryvv?
19:11.25JerJerbkw_:  and i'm not talking extconfg handler either
19:11.26FarrisGharryvv: Yes
19:11.39JerJerwe implemented our own config back-end
19:11.45jetsQuack!
19:11.47jarrodjerjer: how does your config handler operate safer/more stable ?
19:11.57JerJerhave i ever claimed that?
19:12.03JerJeri simply said its not realtime or flat files
19:12.04bkw_JerJer, well you recall that day you said you wish you could reload just X or Y context?
19:12.07jarrodyou are obviously using it?
19:12.16jarrodso you feel it is a better solution
19:12.26JerJerbkw_:  so i cannot provide suggestions that I receive from my customers?
19:12.31tzangerjarrod: I don't like realtime because now you've got that ass-nugget MySQL as an SPOF
19:12.33hardwirestill battling trying to make the perfect MoH server
19:12.38hardwirewhat a pain in the patootie
19:12.48Lee__hardwire: define "perfect"
19:12.55bkw_tzanger, ODBC can speak to anything that has a driver
19:12.56fugitivowhat happens with realtime if the database is down?
19:13.01jarrodtzanger: mysql is very reliable also
19:13.15fugitivoasterisk stops working?
19:13.16tzangerthere's no reason not to generate flat files and reload the specific module you need, it'severy bit as configurable and you don't have the extra code waiting to crash
19:13.23hardwireLee__: I am trying to think of the best way to mix in ads/notifications with a live stream
19:13.41tzangerbkw_: true, but now you're introducing a HUGE chuck of code into your PBXes very core...  I'm a HUGE fan of KISS when it comes to things like this
19:13.46jetsI agree....  I love jesse's XML config engine to throw it out to tftp files and asterisk configs when it's called.
19:13.49hardwireI think I am just going to be screwed
19:13.57Lee__hardwire: use a mixer and send that to a shoutcast server live?
19:13.59jarrodtzanger: i just dont like syncing with 5 other servers via scp or the likes
19:14.02tzangerjarrod: it's gotten better but until it is ACID compliant and doesn't play silly bugger with my data it won't see the light of day in my shop
19:14.03jarrodmodify one backend.. done/done
19:14.03jetsUPdate config/tftp from MySQL, update MySQL from XML and keep it all congruent.
19:14.07hardwireLee__: an automated one.. yes
19:14.13JerJerjarrod:  tell that to BroadSoft then
19:14.15tzangerjarrod: so make it a little smarter
19:14.22Lee__hardwire: have you looked  JACK?
19:14.25hardwireLee__: but for right now I am thinking about how to use polypaudio server
19:14.27hardwireor jack
19:14.28JerJerthey use rsync like its going out of style
19:14.33hardwireI am not a fan of the complexity associated w/ jack
19:14.34Lee__:) jack rulez
19:14.41hardwireLee__: good.. then you wanna help :)
19:14.47Hmmhesaysthis episode where data plays kolrami in stratagema rocks
19:14.52Lee__help with what? your MoH server?
19:14.57hardwirewell learning Jack
19:14.58Lee__or setting up Jack?
19:15.10hardwireso that I can say.. at the end of the day.. I learned jack today.
19:15.19Lee__ha! what do you want to know?
19:15.21yaaarAnybody know why i can start asterisk with 'asterisk -U asterisk -G asterisk -vvvvvvvc' but can't with 'amportal start' ? it just says "failed to start, check /var/log/asterisk/full' but that file doesn't have anything from today in it.
19:15.32JerJerdant:  where is the context of that statement?
19:15.34hardwirehttp://jackit.sourceforge.net/ right?
19:15.40JerJeryaaar:  dont' run AMP
19:15.41bkw_yaaar, the file is too big?
19:15.46dantJerJer, I pasted the question that you responded to
19:15.47bkw_that 2gb crap stuff in linux?
19:15.52yaaarbkw_: which file, the log? it's tiny
19:15.57Lee__yeah. there's a good tutorial for getting realtime performance and capabilities from a 2.6 kernel, but Debian makes it simpler.
19:16.26Lee__hardwire: I'll /msg you, this is totally OT.
19:16.34hardwireI know
19:16.37hardwirebok
19:16.39hardwireerr
19:16.39hardwireok
19:16.48yaaarJerJer: I heard you the first time, but I, unlike you, care some about how much time I spend on tasks I do every day. I setup asterisk and had it working just fine without the interface. now I want to see whether the interface will do the things I want and thus save me time. if it doesn't, i'll go back to editing the configs.
19:17.05FarrisGThe logs even show that my custom greeting is playing for the caller when he calls, but the caller hears nothing, and eventually it times out and hangs up
19:17.22dantJerJer, http://pastebin.ca/18547 <-- a more complete context for you
19:17.22JerJeryaaar:  so you waste time by asking the same question that wasn't answered the first time?
19:17.48yaaarpiss off
19:17.51JerJerdant:  ok and your point is?
19:17.58*** join/#asterisk shrush (~goldenold@ns2.xoasisnetworks.com)
19:18.04shrushhi
19:18.34*** join/#asterisk joerg (~joerg@p548886CC.dip0.t-ipconnect.de)
19:18.35joerghi
19:18.41jarrodive been using CDR to mysql with no issues at all
19:18.46shrushanyone looking for a used T410P?
19:18.56yaaarshrush: pm you?
19:19.25shrushyah
19:19.25joergI have two different sip providers...they provide connectivity to pstn. one works fine with asterisk. with the other, I can't hear the person who answers the phone but he hears me...
19:19.29joergany idea?
19:19.41dantJerJer, just giving you a hand filling in the blanks
19:19.53JerJerwhere does it state that i use flat files?
19:19.55generalhanwho are the 2 providers
19:20.03fugitivojoerg: is the configuration the same for the 2 providers?
19:20.06FarrisGhere is my log from a call: http://pastebin.ca/18548
19:20.06FarrisGLooks normal
19:20.48lters_shrush, why not get it upgraded?
19:22.50dantJerJer, well, text based config files tend to be flat?
19:23.33JerJerok where does it say i use text based config files?
19:23.59JerJerhe was stating that he had to stop and start asterisk to make changes - i told him differently
19:25.16joergfugitivo: yes, just checked that :)
19:25.24Hmmhesaysseriously folks this episode just rocks
19:25.40dantJerJer, http://pastebin.ca/18549
19:25.57tzangerHmmhesays: eipsode of?
19:26.11*** join/#asterisk drbrown (~chatzilla@63.238.117.40)
19:26.13FarrisGI would really appreciate it if someone could give me a hand with this
19:26.30joerggeneralhan: sipgate and 1und1
19:26.45generalhanand which one seams to be working for you ?
19:26.59drbrownhave any of you guys dealt w/ pfsense
19:27.00JerJerdant:  yeah notice the fucking date
19:27.18*** join/#asterisk MicC_ (~sum1@CPE000c419ce901-CM000a7363f92c.cpe.net.cable.rogers.com)
19:27.23generalhanim sorry to ask you these questions so that you think i might be able to help. but im looking for a reliable SIP provider and this info will help me out a lot ! LOL !
19:27.38JerJerand that was a dicussion about wanting to use RADIUS
19:27.40Hmmhesaystzanger: this episode of tng where the hathaway has to challenge the enterprise
19:27.43JerJerwhich you didn't bothere to send
19:27.45tzangerah
19:28.19MicC_what is the best softphone out there
19:28.24MicC_commercial and OSS
19:28.29MicC_?
19:28.30tzangerI've been here over a year and I still don't know why RADIUS is such a poor choice, I dont' need it but I just never got a real good explanation about how its accounting was wrong compared to the ohters
19:28.31Nuggetthey all suck.
19:28.33Hmmhesaysi like firefly
19:28.33Nuggetwhat os?
19:28.37*** join/#asterisk ckruetze (~ckruetze@i3ED6833B.versanet.de)
19:28.39tzangerI'm guessing something to do with being able to miss STOP packets
19:28.39dantJerJer, nope, wasn't about RADIUS
19:28.43joerggeneralhan: sipgate
19:28.52MicC_Hmmhesays: firefly? URL?
19:28.57generalhanthey dont ALL suck ... im using IPBlue's soft phone that has the Cisco skin and its pretty good
19:29.25MicC_generalhan: my prob right now is Skype works better than my VOIP service
19:29.35MicC_and my VOIP service is local lan only
19:29.53fugitivogeneralhan: url of ipblue?
19:29.54JerJerdant:  i have logs here too
19:29.56dantJerJer, http://pastebin.ca/18550
19:30.00generalhanwww.ipblue.com
19:30.06JerJerchayewla was all over me about radius
19:30.09generalhanits a little expensive but worth it
19:30.22dantJerJer, decode was talking about getting his * config in a database
19:30.53JerJerwe don't force asterisk to depend on the database
19:30.54generalhani have 30 users that all use cisco 7960s and when our voice stream goes out with our crappy VoIP provider, the soft phones still work, so its worth the investment for our company because we cant have ANY downtime
19:30.58JerJerjust like real switches
19:31.44JerJerand again notice the date
19:31.53tzangerJerJer: hmm I was doing something like that
19:32.10tzangeroffice* - iax2 - colo* - iax2 - switch1 - {whatever you do to TDM}
19:32.16dantJerJer, it's in the past, which is in the timeframe bkw had said you'd mentioned it
19:32.25JerJerbullshit
19:32.47JerJertzanger:  which woudl be  switch-1 -> gw-chi-X -> PSTN
19:33.02tzangerso that's 4 IAX2 hops and a Zap term.
19:33.31tzangeroffice* - iax2 - colo* - iax2 - switch1 - iax2 - gw-chi-x - zap - pstn
19:33.41JerJerwhat's at the office?  SIP?
19:33.57tzangerJerJer: nope.  Norstar - PRI - office*
19:34.02MicC_hey...I have a trunk setup from 1 PBX(no pots) to 1 PBX with pots. I have trunk set to dial 9 on behalf of users to get out the PBX with pots.
19:34.02*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
19:34.05JerJerok so Zap
19:34.06MicC_is there a better way?
19:34.15tzangerso zap - 4*iax2 - zap
19:36.44HmmhesaysMicC_ google it
19:38.25*** part/#asterisk gwynpen (~gwynpen@p54AAD640.dip.t-dialin.net)
19:38.30lters_generalhan, which soft phone do you run from ipblue?
19:39.12*** join/#asterisk tim27 (~tim27@97-70.dr.cgocable.ca)
19:39.19generalhanthe VCGO Adv.
19:39.28*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
19:39.30generalhanVTGO-PC Adv. rather
19:39.38tim27any here have sip image for 7905 phone
19:40.22JerJertim27:  www.cisco.com
19:40.37*** join/#asterisk matr24ct (~lkj@p54994B8F.dip.t-dialin.net)
19:41.02matr24ctHi, does anyone know the newest chan_capi-0.5.4
19:42.51focksanyone use Polycoms?
19:43.24focksi'm trying to figure out a way to prevent them from showing a missed call if the call was rolled to you because noone answered the main number
19:43.44JerJergood luck
19:45.06focksok, how about a way to use a distinctive ring if its a broadcast ring
19:45.09*** join/#asterisk Error_X (~Error_X@217-131-211.5001.adsl.tele2.no)
19:45.23Error_XIs it possible to set up a chat room, like fwd coffee house with asterisk?
19:45.25focksguess that would be similar to how i did the intercom
19:45.33JerJerif that call isn't answered the phone's counter will increase
19:45.56focksJerJer, which counter
19:46.02JerJermissed calls
19:46.09focksyeah, it's annoying
19:47.08tim27any here tested the GPX-2000 and SPA-841
19:47.22generalhanwhat format are the auto-attendants greetings stored as in asterisk ?
19:47.25*** join/#asterisk znoG (~gs@200.115.216.109)
19:48.49*** join/#asterisk PG3 (~c@cablea0mle.cybercable.net.mx)
19:50.27matr24ctnobody ever heard about chan_capi-0.5.4?
19:51.06focksanyone noticed in VoiceMailMain it says 4 is temporary greeting, but 4 is change password and it says 5 is change password but 5 doesn't work?
19:51.24zoageneralhan: gsm normally
19:51.47matr24cti'd just like to establish what ist the generally known version of chan_capi in the asterisk world
19:52.23zoacheck www.junghanns.net
19:52.29zoathats the most common version
19:53.52_deg_Anyone know if it is difficult to backport SIPGetHeader to stables version?
19:54.23pifiujarrod are you still here?
19:57.41*** join/#asterisk Umaro (~umaro@67.189.110.20)
19:57.54anthmwho with stable wants to test a backport of chanspy?
19:58.25FarrisGI could very well get fired today
19:58.40anthmFarrisG's day off?
19:58.49FarrisGanthm: cute
19:58.50Lee__anthm: I can try but I don't know what chanspy does so I might not be of much help.
19:58.53PoWeRKiLLwho use firmware 3.2 of sipura please ?
19:58.55Nuggetmmmmmmmmia sara.
19:59.00Lee__I have a sarge dev box.
19:59.23FarrisGI'm serious. It looks to me as if someone has screwed with the configuration on this server, and now everything APPEARS to work correctly, but doesn't actually
19:59.26Lee__or do you mean stable Asterisk  :)
19:59.27Umaroanthm: hey there
19:59.43anthmchanspy is like zapscan without the ZAP limitation
19:59.50anthmhi umaro
20:00.18PoWeRKiLLwho manage to compile res_php I'm trying to but I always get error
20:00.21Umaroanthm: could you answer a question for me? in include/asterisk/manager.h, there's #define MAX_LEN 256
20:00.33*** join/#asterisk NoOS (~askme@cust.8.241.adsl.cistron.nl)
20:00.35Umaroanthm: If I raise that to say, 512, where else do I have to change it?
20:00.36NoOShi
20:01.05matr24ctwww.junghanns.net hasn't been updated in ages
20:01.08anthmprobably nowhere
20:01.16anthmit's a constant you are redefining
20:01.19matr24ctthere is a newer version at https://sourceforge.net/projects/chan-capi/
20:01.57anthmyou may need to make clean to make sure
20:02.49NoOSIs 2x isdn bri + 2x HFC card + bristuff  a good start ?
20:03.03*** join/#asterisk SiD3WiNDR (luser@bastard-operator.from-hell.be)
20:03.34generalhanwhat/and how many of the interfaces do i need to support 30 VoIP users ?
20:03.48Umaroanthm: should that cause any problems?
20:04.38anthmoh from looking at the code
20:04.42anthmnot at all
20:05.08zoaanthm, i have problems with the damn zaptel.conf :(
20:05.14anthmyou'll just use more ram
20:05.29zoait doesnt seem to be compatible with whatever i read about framing / signalling in the us
20:05.58anthmfor t1 ?
20:06.50NoOSWhat is better chan_capi or bristuff by junghanns?
20:08.05*** join/#asterisk Juxt (~Juxt@64.135.20.202)
20:08.07Juxtgood day
20:08.24Juxti just upgraded to the latest head and my setup has gone wild
20:08.28zoaim off
20:08.30zoacheers
20:08.36Juxti am getting Jul 26 16:07:09 WARNING[8738]: chan_sip.c:8666 handle_response: Host '192.168.16.22' does not implement 'PUBLISH'
20:08.44SiD3WiNDRcan the alsa module be used with a modem which is recognized by alsa (snd-intel-8x0m, AC97 AMR card) or doesn't that work?
20:08.46Juxtand my phones keep ringing after the calling party has hung up
20:09.22Juxthow can i turn off this PUBLISH thing?
20:09.35JerJerJuxt:  cd channels
20:09.50JerJercvs up -r 1.791 chan_sip.c
20:10.02ManxPowerSiD3WiNDR: that doesn't work
20:10.05Umaroanthm: it'll take more ram all the time, or just when the manager command is being initiated?
20:10.24SiD3WiNDRManxPower: what is the alsa module for then?
20:10.49anthmevery manager session and every manager command
20:10.55JuxtJerJer: that will downgrade chan_sip right?
20:10.58anthmit's like not that much tho
20:11.09anthmyou said you want 512 instead of 256?
20:11.14Umarofor example, yeah
20:11.25*** join/#asterisk znoG (~gs@200.115.216.109)
20:11.46anthm256 bytes more per session in use
20:11.46NoOSWhat is better chan_capi or bristuff by junghanns?
20:11.53ManxPowerSiD3WiNDR: to use the speaker and microphone of the sound card as a "console phone"
20:11.58anthmand 256 more per message you send
20:12.19ManxPowerBut you can't use the modem part, only the soundcard part.
20:12.24drbrownfile, bkw_ wants you to test my asterlink account on your test system
20:12.35*** join/#asterisk gtigene (~chatzilla@70.89.216.41)
20:12.36bkw_drbrown, is having firewall drama
20:12.38bkw_hehe
20:12.44fileok
20:12.48SiD3WiNDRManxPower: ahh, okay, I see. thanks.
20:12.56anthmso if you had 4 sessions at once you would waste an extra 1k of ram
20:13.10Umaroanthm: oic
20:13.18Umaroanthm: that's not bad ;)
20:13.30gtigeneMy Asterisk does not display caller id name for incoming calls from the PSTN. Is this normal? What can I do?
20:13.43*** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net)
20:13.49drbrowntook out the registry info from my box file
20:14.33Darwin3592.99 they say
20:14.45filedrbrown: let's move over to #asterlink :)
20:14.45*** join/#asterisk zoo (nobody@ip-168-16.travedsl.de)
20:14.46Darwin35man they rob you
20:14.55drbrownk
20:15.16*** join/#asterisk newmember (~newmember@dsl-lkbn-66-18-211-34-cgy.nucleus.com)
20:15.17DarthClueDarwin35: be happy.  my bill is running about $200.
20:15.40Darwin35wow
20:15.55Darwin35just cant believe bills run that high
20:16.14zoohello
20:16.32Darwin35just means cutting back on the ac and tv and only use the pc
20:16.41DarthClueDarwin35: that's what you get when the power company is a monopoly.
20:17.08zooi want to have an extention.conf-content for incoming calls that just plays an anouncement. but i don't manage to. any hints?
20:17.36Darwin35the actual use wa 44. dollars but then there is a 30 dollar energy cost recovery fee
20:17.42*** join/#asterisk eagle501 (~icechat5@83-65-72-2.berggasse-II.xdsl-line.inode.at)
20:17.54eagle501hello!
20:18.01ManxPowerMy power bill is usually around $100 year-around
20:18.19Darwin35monthly
20:19.30Darwin35well i think I will just have to go back to the soloar loft  in toronto .  and conver all the pc's to 12v
20:19.34harryvvmanx, sounds like ours
20:19.44*** join/#asterisk pjquinney (~phil@cpc4-walt1-5-0-cust162.popl.cable.ntl.com)
20:20.02harryvvelectricty here is cheap. BC even sells its surplus power to califonia
20:20.02ManxPowerharryvv: I'm lucky.  I live in a newish building with a newish cooling system
20:20.04*** part/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net)
20:20.35gtigeneMy Asterisk does not display caller id name for incoming calls from the PSTN. Is this normal?
20:20.38harryvvThats good.
20:21.07harryvvdo you have noOp in your extentions?
20:21.10ManxPowergtigene: no.
20:21.13harryvvnoop cid
20:21.18ManxPowergtigene: What brand of phone do you have?
20:21.33gtigeneManxPower: Polycom 30x and 50x
20:21.41ManxPowergtigene: and what phone company?
20:21.50gtigeneharryvv: Don't have no op in extensions
20:21.50harryvvManxPower was it you that was making the next asterisk book?
20:21.52pjquinneyHi everyone, I checked out a fresh copy of CVS head so I could help look for bugs but I can't compile asterisk itself - I've done everything as I usually do but the compile seems to be looping with this fragment: http://pastebin.com/321329. Thanks to anyone who can help.
20:21.59ManxPowerharryvv: hell no
20:22.05gtigeneManxPower: Paetec
20:22.06harryvvSomone in here said thay are
20:22.18ManxPowergtigene: what country?
20:22.27gtigeneManxPower: USA
20:22.53eagle501I've currently a problem running asterisk. Asterisk 1.0.8 (gentoo) starts really fine but I can't stay online with my siptronic phone (or any other). I'll get always "Scheduling destruction of call .... in 15000 ms" and after that it destroys "Destroying call". I have already googled around but i stuck on that problem. Anyone experienced the same problem?
20:23.05ManxPowergtigene: does it work when you plug an analog phone into the telco line?
20:23.33ManxPowereagle501: start asterisk using safe_asterisk and connect to it using asterisk -rvvv
20:23.50gtigeneManxPower: I don't know. I have a four port T1 card (TE405P).
20:23.58ManxPowergtigene: Oh!  Using PRI?
20:24.00eagle501ok i'll give it a try
20:24.08gtigeneManxPower: Yes
20:24.17NoOSHi, should I start with chan_capi or zapbri?
20:28.00*** join/#asterisk Saaib (~nabudocon@ns1.ensenada.gob.mx)
20:28.09gtigeneManxPower: Are you there? :)
20:30.53NoOShuh
20:32.00*** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl)
20:32.01NoOSHi, should I start with chan_capi or zapbri?
20:32.12pooh_JerJer, you around ?
20:32.17pjquinneyNoOS: What are you trying to do?
20:32.26NoOSI only use ip phones, should I start with chan_capi or zapbri?
20:32.48pjquinneyNoOS: If you only use IP phones you don't need either of them
20:32.55NoOSI would like to hoop up two bri lines with 4 ip phones using asterisk
20:33.04JerJerpooh_:  no
20:33.09pjquinneyNoOS: unless you want to do IAX trunking or MeetMe: in which case you need ztdummy
20:33.10NoOShook
20:33.27pjquinneyNoOS: hook??
20:33.35NoOSup
20:33.56NoOSI want sip to cisco ip phones
20:34.12pooh_JerJer: no one is available at this time, using the nufone service. Ongoing call was interupted
20:34.19NoOSso is capi best ?
20:34.21*** join/#asterisk ctjctj (ctjctj@192.55.203.130)
20:35.00pjquinneyNoOS: I don't know much about it, but I thought CAPI was for ISDN
20:35.08FarrisGok, something's dying...
20:35.16NoOSWhat hardware do I need for my bri lines to asterisk box ?
20:35.17FarrisGnow the friggin' daemon just keeps dying off
20:35.49FarrisGwhat is one supposed to do in a production environment when nobody is available with enough experience to fix things?
20:35.52ctjctjHello, I've got an asterisk system cobbled up using a pap2-na and viatalk.  It even manages to have working voicemail.  But, how do I get the MWI to just flash the light on the phone and not rering every few minutes?
20:36.10harryvvFarris, have a backup asterisk box made up
20:36.20pooh_JerJer: running again... I am puzzled
20:36.30harryvvFarris, do you have a spare pc to use in the event the main one goes down?
20:36.36NoOSWhat hardware do I need for my bri lines to asterisk box ?
20:36.59pjquinneyNoOS: Sorry - I didn't see you say you had analogue lines as well - my bad
20:37.05FarrisGharryvv: Yes, I do have a spare PC, but in my case that won't really help because I didn't set up the current one, and the person who did is no longer reachable
20:37.18NoOSI have 2 HFC cards so what driver is best capi or bri_stuff
20:37.19*** join/#asterisk ClayReiche123 (fwuser@mail.accxx.com)
20:37.36pjquinneyI'm not sure on that one - the best thing is to try each and see what works for you
20:37.42pjquinneyeveryone's setup is different
20:37.50FarrisGBut seriously, I have to get this machine BACK up and running today. A backup will then be created, but who do I call/email/pay for support to get things back the way they were?
20:37.58JerJerpooh_:  you bother to do a traceroute ?
20:38.01pjquinneyctjctj: Turn off the voicemail splash ring (or set it to 0)
20:38.05ClayReiche123Does anyone know if I can pipe out a cli command to a text file?
20:38.16ctjctjpjquinney: In voicemail.conf?
20:38.22outtoluncfarrisg you might want to hire someone to help you since your need is immediate
20:38.25pjquinneyNo, in the PAP2 config page
20:38.33pjquinneydo you want me to check where it is for you?
20:38.36FarrisGouttolunc: That's what I'm asking... Hire WHOM?
20:38.41harryvvFarris, its not hard to setup a new one. As long as you can copy the config files to the new asterisk box swap out the cards then your in biz. So, how many people depend on this box?
20:38.55outtoluncthere should be a 'consultant list' on digium
20:39.01gtigeneClayReiche123: I redirect cli output to text files. The text file is kind of funky but I can usually find what I am looking for.
20:39.06FarrisGharryvv: About 50
20:39.12ctjctjpjquinney: Nope, that's ok.   Just explains why I couldn't find it in the asterisk documentation.   Two days of websearchs and an "oh, never thought of that."
20:39.15outtoluncthat or post the request to the asterisk-biz list
20:39.40ClayReiche123gtigene: How do you do it.... or are you just using the logger files?
20:40.05gtigeneClayReiche123: asterisk -vvr >temp.txt
20:40.21ManxPowergtigene: look at logger.conf to make asterisk log in the way you want.
20:40.30ctjctjFYI: Viatalk has been very supportive of asterisk.  They run it internally, sent me a locked pap2-na, gave me passwords for the pap2 to unlock it, allowed me to use asterisk to their servervice using sip, is planning on rolling out an iax connection in the near future.
20:40.32pjquinneyIts got to be on the top of the list of annoyances with the Sipura devices
20:40.59FarrisGthis is absolutely terrible
20:41.08harryvvFarrisG msg me
20:41.26gtigeneManxPower: You and I were talking about my problem with caller id name not getting to the phones
20:41.40*** join/#asterisk pifiu (~myassisbi@208.205.181.170)
20:41.53gtigeneMaxPower: You asked me if it was a PRI. Yes, it is.
20:41.57pjquinneyCan anyone help me now?? Please?? CVS-Head won't compile for me - the compilation loops: http://pastebin.com/321329
20:42.16ClayReiche123gtigene: Have you tried a "asterisk -x sip show peers" >temp.txt
20:42.23InfraRedasterisk sucks
20:42.26InfraRedthere
20:42.47gtigeneClayReiche123: Did you see what ManxPower suggested about logger.conf?
20:43.06pjquinneyInfraRed: are you having trouble?
20:43.17ManxPowergtigene: Yeah.  Then I got a phone call.
20:43.42ManxPowergtigene: put a Wait(1) as soon as the call comes in to wait for the callerid name (which is sent after the incomming call is accepted).
20:43.46InfraRedpjquinney: i think it's the provider being cunts
20:43.49ClayReiche123gtigene: yes... I just want to log the output from 1 command though....
20:43.51gtigeneClayReiche123: that should work
20:43.53InfraRedkeep getting error 481
20:44.04ManxPowerif that works then you can reduce the wait to .5 or .25 and find the min required.
20:44.08ClayReiche123thanks
20:44.13pjquinneyInfraRed: sounds bad - anything I can try and help you with?
20:44.13ManxPowergtigene: also, your carrier may not be sending CLID name
20:44.17gtigeneManxPower: I will try it. Thanks.
20:44.32gtigeneManxPower: I will ask the carrier
20:44.42ManxPowergtigene: try the waits first
20:44.45*** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca)
20:44.51*** part/#asterisk Juxt (~Juxt@64.135.20.202)
20:44.53gtigeneManxPower:OK
20:45.01InfraRedJul 26 21:37:17 NOTICE[11393]: chan_sip.c:6912 handle_response: Failed to authenticate on INVITE to '"1901001" <sip:1901001@82.x.x.x>;tag=as102d7ffa'
20:45.13SwK[Work]anyone using AudioCodes Gateways?
20:45.15InfraRed<PROTECTED>
20:45.32InfraRedpjquinney: any ideas?
20:45.42pjquinneyHmmmm - which provider if you don't mind me asking?
20:45.52InfraRedthe lost planet
20:45.59InfraRedmagrathea  telecom
20:46.00InfraRed:)
20:46.48hmodesw00
20:46.50hmodescolor me impressed
20:46.56pjquinneyInfraRed: my best advice would be to try using x-lite. If it works there then get an ethereal dump of the login process. Then tweak your asterisk setup to match. Bring to the boil and simmer for 10 minutes...
20:47.18hmodes500 ivr calls on a dual opteron and it can still keep up with 20calls/sec being setup
20:47.58eagle501I have still a problem using asterisk. The sip connection gets always destroyed: http://xinfo.net/asterisk/sip.conf http://xinfo.net/asterisk/index.txt If anyone could help me, it would be great! :) google didn't help me :(
20:48.43*** join/#asterisk hhh_ (Gone@i-195-137-50-162.freedom2surf.net)
20:48.47hhh_hi guys
20:50.40ctjctjpjquinney: It was under user1 and user2 and is VMWIduration:   I set it from 0.5 to 0 and it is now pleasently quiet.   Thank you again.
20:51.17pjquinneyctjctj: Hmmm, that sounds different to my sipura. Anyhow - at least you can enjoy the peace and quiet!
20:51.30pjquinneyeagle501: You aren't authenticating properly: "SIP/2.0 401 Unauthorized"
20:51.49InfraRedanyone here using astcc with iax?
20:52.01*** part/#asterisk ctjctj (ctjctj@192.55.203.130)
20:52.04FarrisGhttp://pastebin.ca/18558
20:52.13pjquinneyeagle501: what is the output of "sip show peers"
20:52.48FarrisGI'm getting a seg fault, and I can't tell why
20:53.13joergI'm at the end with my latin :)
20:53.23eagle501/msg pjquinney 33/33            83.65.72.9       D          255.255.255.255  1720     Unmonitored
20:53.25*** join/#asterisk hellop (~hellop@cpe-70-95-18-61.hawaii.res.rr.com)
20:53.36hhh_doesn anyone here use gnugk in proxy mode?
20:54.02eagle501that would be the output :/
20:54.28pjquinneyeagle501: try turning on debugging for that peer "sip debug <peername>" where peername is the name defined in sip.conf
20:55.52harryvvcan asterisk go into seg fault if one of the vm passwords was put in wrong ?
20:55.58*** join/#asterisk Error_X (~Error_X@217-131-211.5001.adsl.tele2.no)
20:56.39*** join/#asterisk sawyernet (~lsawyer@sawyernet.com)
20:56.45jetsharryvv: very unlikely
20:57.04Error_XWhat is this error? NOTICE[2147]: chan_iax2.c:5773 socket_read: Rejected connect attempt from 65.39.205.12, requested/capability 0x4/0x4 incompatible with our capability 0xff03?
20:57.07*** join/#asterisk doolph (doolph@201.226.146.178)
20:57.14doolphhow can I debug an agi script?
20:57.24bkw_Error_X, going from CVS-HEAD to stable?
20:57.26bkw_cd /usr/include
20:57.28bkw_rm -rf asterisk
20:57.30bkw_cd /usr/src/asterisk
20:57.32bkw_make clean
20:57.32bkw_make
20:57.33bkw_make install
20:57.36bkw_NEXT!!!
20:57.36*** join/#asterisk craziman2 (~Craziman2@boromir.apid.com)
20:57.40harryvvjets, and if the partion thats used to fill with vm is full, would that cause a feg fault?
20:57.42Error_Xbkw_: apt-get install asterisk :p
20:57.55Umarohmodes: you're running that 500 ivr calls on a dual opteron?
20:58.31SwK[Work]bkw_:  are you a little cranky today?
20:58.50bkw_na
20:58.51bkw_i'm ok
20:59.00SwK[Work]Error_X: you can deal with those old crusty packages
20:59.11sawyernetjust upgraded to latest CVS-HEAD today.  Having one issue.  When a call comes in it is ringing the ext the correct time and then going to voicemail, but the ext keeps ringing?  I am seeing a SIP 405 method not allowed error???
20:59.32FarrisGhttp://pastebin.ca/18560
20:59.32FarrisGHere's the error I'm getting...
20:59.49FarrisGOr rather, NOT an error. Just a call (that goes through, but with no audio), and then a segfault
20:59.50SwK[Work]so ok I know someone here uses AudioCodes gateways... where are you?
20:59.50sawyernetit also does it on ext that i have ringing multiple phones...
20:59.53eagle501http://xinfo.net/asterisk/debug.txt is the output. it's probably the same (after a short look above the lines)
21:00.18*** join/#asterisk R3DB0x (nobody@66.142.28.36)
21:00.25joerghmm, when I call via call file, the connection is established...but there are no udp packets sends containing the voice...
21:00.28pjquinneybkw: Could I ask you for some help please? - I'm compiling CVS-Head for the first time in about a month and my make is looping: http://pastebin.com/321358
21:00.35joergideas? :)
21:00.51joergonly packets from and to udp 5060
21:01.10pjquinneyeagle501: are you sure you've got the right user / password?
21:01.36pjquinneyeagle501: that looks like you aren't authenticating properly and so when you go to make a call it is rejected
21:03.00eagle501/msg pjquinney 100% it's really annoying, and those user/passwords are really short it's just user=33 and pass=33
21:03.24eagle501/msg pjquinney i re-checked it several times
21:03.40*** join/#asterisk jamestt (~chatzilla@22.DHCP46.enoreo.on.ca)
21:06.42Error_Xis meetme included in asterisk?
21:07.08pjquinneyError_X: yes
21:07.16Error_Xk
21:07.58Error_Xis meetme like fwd coffee house?
21:08.27jamesttis sip registration working for 'realtime' in HEAD?
21:09.23hellopIs there any resaon why we can't introduce ogg to Asterisk?
21:09.33*** part/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl)
21:10.03hellopLike as a default recording format.
21:10.13pjquinneyhellop: licensing probably
21:10.49sawyernetI d/l CVS HEAD today and am getting the same error with a strange behavior.  If I ring a number that calls two sip extensions and pick up the call on one of the extensions, then the other extension continues to ring indefinitely.
21:11.15*** join/#asterisk clive- (~pirch@rrba-146-74-170.telkomadsl.co.za)
21:11.31SiD3WiNDRpjquinney: licensing of ogg? I thought it was supposed to be free? :)
21:11.59pjquinneyi thought it was licensed under some kind of license
21:12.15pjquinneytherefore digium wouldn't be able to have the code disclaimed in the same was as normal asterisk code
21:13.28harryvva pri cable going into the back of a server looks more ribbon like right?
21:15.14*** join/#asterisk Tili (~Tili@202-133-67-126-dialup.sat.net.pk)
21:16.08*** join/#asterisk loick (~loick@APuteaux-151-1-21-108.w82-124.abo.wanadoo.fr)
21:18.44*** join/#asterisk zoo (nobody@ip-168-16.travedsl.de)
21:19.12zoowhat did i do wrong, if * says on incoming calls:  Unable to create channel of type 'SIP' ?
21:19.26*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
21:19.49pjquinneyzoo: It couldn't connect usually, or the peer is offline
21:23.22*** join/#asterisk jake1932 (~jake1932@pool-68-236-16-157.phil.east.verizon.net)
21:23.46jake1932is BRI and PRI S/T compatible?
21:24.36jake1932IOW - can I use a T100P with a BRI 4 wire connection?
21:24.45clive-jake NO
21:24.55jake1932tnx
21:25.02clive-your welcome:)
21:25.59*** join/#asterisk brettnem (~Brett@207.90.232.34)
21:26.18brettnemhey all
21:26.59brettnemanyone want to point me to a good source for bulk cisco phones (sip 7940).. need about 20-30 or so..
21:27.05*** join/#asterisk file[laptop] (~file[lapt@mctnnbsah25-142166093154.nb.aliant.net)
21:27.22xtrvdWhat kind of method can I use to establish a call que where if I am on the phone, incoming calls are told that the number is busy and that they are on hold, and will be answered when I become avalible? Could anybody point me in the direction to look in the wiki?
21:27.50brettnemxtrvd: Look for 'camp on' in the wiki for some ideas
21:28.02xtrvdThanks Brett, =)
21:28.28brettnemsure
21:29.47zoocan i call a SIP-phone from the cli for testing purposes?
21:30.12brettnemzoo: I think you need chan_oss for that
21:30.26brettnemthen you can do 'dial myphone@mycontext'
21:30.39brettnemessentially dials from your soundcard
21:31.01zoobut i have no sound in my machine
21:31.06sawyernetI d/l CVS HEAD today and am getting the same error with a strange behavior.  If I ring a number that calls two sip extensions and pick up the call on one of the extensions, then the other extension continues to ring indefinitely.
21:31.18brettnemso. bulk cisco phones.. are you telling me that in a conf room of 300 people no one wants to sell me cisco phones?? ;)
21:32.01brettnemzoo: try loading chan_oss and see if you get a dial command.. I don't know if you actually need the hardware. I know I've used it before on computers I didn't think had soundcards..
21:32.34brettnemsawyernet: we'd need a sip trace to debug that.
21:32.50*** join/#asterisk jamestt (~chatzilla@22.DHCP46.enoreo.on.ca)
21:32.56brettnema CANCEL should go out the the phone that didn't get answered.. and it should reply with a 487
21:33.17sawyernetbrettnem... ok... let me check that...
21:33.27brettnempastebin please
21:33.44matr24ctWhere can one find a list of all the modules in asterisk and what each module does?
21:33.51sawyernetpastebin??
21:34.10fugitivo"By now you are aware that StanaPhone does not currently provide 911 service"
21:34.14fugitivohehe
21:34.17brettnem~pastebin
21:34.17jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
21:34.28sawyernetthanks
21:34.32brettnemsure
21:34.46zoobrettnem: i don't have chan_oss, as i am using an embedded system (openwrt)
21:34.48brettnemmatr24ct: try 'show modules'
21:34.54*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca)
21:34.57brettnemhmmpf
21:35.14sawyernetbrettnem:  what is the best way to capture the trace?
21:35.21brettnemwell don't know what to tell you then.. you gotta connect the near end of the call to something
21:35.35harryvvso how many major voip providers have sucessfully implemented E911 services?
21:35.36brettnemsawyernet: tethereal port 5060
21:35.37brettnemyou'll need ethereal
21:35.45brettnemharryvv: I have
21:35.47sawyernetgot it...
21:36.23harryvvbret, is your equipment in the same geographical locate as your customers?
21:36.34*** join/#asterisk gst (~gst@85.124.175.134)
21:37.07brettnemharryvv: the equipment locale is irrelevant except for geographical diversity
21:37.12gstis it possible to use dundi together with realtime extensions?
21:37.34brettnemshould be able to.. I avoid realtime.. so I'm the wrong one to ask
21:38.25harryvvbrettnem so did you get the 911 info from the local clecs?
21:39.00brettnemharryvv: I am the local clec. :)
21:39.08harryvvahhh
21:39.16harryvva traditional phone company
21:39.17harryvv:)
21:39.19gstbrettnem: ok.. the only worries that i have is that there will be _many_ db queries until the peer got enough HINTs of our dialplan. but i think i'll just try it out.
21:39.34brettnemharryvv: no. not a traditional phone company. :)
21:40.25brettnemgst: I'm not sure how realtime all of that stuff is.. some of it might not really be realtime.. like loaded into memory
21:41.42*** join/#asterisk jhava (~icechat5@200.58.26.21)
21:42.13gstbrettnem: our extensions here are really realtime. for each call we currently get a db query. while it is possible to cache the sip peers it isn't possible to cache the extensions :/
21:42.41brettnemah
21:42.52gst(which would be useless anyway for dundi as most of the queries will be for numbers which we don't host and so they won't be cached)
21:43.06sawyernetbrettnem:  working on it.. trying to capture just the relevent traffic and not all of the other phones that are on the system
21:43.47eagle501exit
21:44.02*** join/#asterisk pjquinney (~phil@cpc4-walt1-5-0-cust162.popl.cable.ntl.com)
21:44.23brettnemsawyernet: you can also try ngrep like:' ngrep -qW byline <text in call setup> port 5060' where that text could be the 10 digit number you are dialing.
21:44.38sawyernetgood idea... I'll do that
21:45.11jhavahello all: question on queues: if I have two agents, one is on a long call, using roundrobin, the available user only gets one every two calls the other call gets dropped, why ? Is there another ring strategy more suitable for this ?
21:45.34brettnemgst: why not just have all your servers point to the same DB for realtime instead of dundi?
21:45.56SplasPoodhrm, is the asterisk in debian testing, 1.0.7.dfsg.1-2, worth using or should I compile from CVS?
21:46.20gstbrettnem: we are currently setting up a peering in the public dundi system
21:46.36brettnemah
21:46.44brettnema "free" provider eh
21:46.50gstbrettnem: therefore i'm not sure if the db will scale - i have no experience how many queries we will get
21:47.02brettnemgst: asterisk doesn't scale.. heh ;)
21:47.39*** part/#asterisk mkrufky (~mk@68.160.103.77)
21:48.23gstbrettnem: our users are registered to openser and we just use asterisk for the call routing. so we can use the dispatcher module of openser for the load balancing. although i'm not sure yet how to load balance dundi.
21:48.37*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
21:48.48brettnemgst: I thought of doing just that... not sure how I feel about it..
21:48.53*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
21:49.18brettnemgst: what exactally do you mean by "the call routing" ?
21:49.24brettnempstn gateway?
21:49.29sawyernetbrettnem: got it http://pastebin.ca/18561
21:49.46*** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-144-219.buff.east.verizon.net)
21:49.52SuPrSluGhi all
21:50.24pjquinneyHi SuPrSluG
21:50.35gstbrettnem: at first we did everything with openser and used asterisk just a pstn gateway. but the problem is that asterisk doesn't support sip transfers when not all of the call legs are on asterisk. so we now route all calls (even sip2sip) through asterisk. (otherwise e.g. a sip user who talks with a pstn user wouldn't be able to transfer the pstn user to another sip user).
21:50.36SuPrSluGgetting a strange message when reloading my extensions
21:50.41SuPrSluGWARNING[16578]: pbx.c:710 pbx_find_extension: Maximum PBX stack exceeded
21:50.53SuPrSluGyo pj
21:51.23brettnemsawyernet: that shows me just one call setup.. I don't see the other leg..
21:51.36Error_XWhat is LSSL?
21:51.43SuPrSluGwhen i show dial plan some extensions are missing. what up wit dat
21:51.58sawyernetbrettnem:  Ok...let me check that
21:52.06pjquinneySuPrSluG: have you tried stopping asterisk and starting it again? Reboot the machine to make sure
21:52.35brettnemgst: hmm.. I haven't tried that.. really.. that seems odd
21:52.59brettnemgst: hey, have you tried the promisredirect directive? I can't remember how it is spelled.. but sounds like it's right up your alley
21:53.07SuPrSluGi'm doing it remotely so if something goes wrong..... i'd prefer not to reboot the machine
21:54.41brettnemgst: +;promiscredir = no      ; If yes, allows 302 or REDIR to non-local SIP address
21:55.05gstbrettnem: isn't promiscredir just for redirects?
21:55.17gstbrettnem: the problem is with REFERs/INVITEs:                         ast_log(LOG_NOTICE, "Supervised transfer requested, but unable to find callid '%s'.  Both legs must reside on Asterisk box to transfer at this time.\n", replace_callid);
21:55.49brettnemoh crap.. heh
21:55.51*** join/#asterisk pifiu (~myassisbi@208.205.181.170)
21:55.54brettnemyour right
21:56.01gstbrettnem: i think that there may be already a patch for the INVITE stuff, but there's still a 50% chance that asterisk does get the REFER instead of the invite (with the replaces tag).
21:56.05*** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
21:56.23sawyernetbrettnem:  not sure... that is all i get on the capture
21:56.24pjquinneySuPrSluG: are you starting asterisk using safe_asterisk?
21:56.26_deg_I have a PABX connected to my Asterisk with E1.
21:56.45brettnemgst: I'd like to get rid of asterisk all together really.. <gasp!>
21:56.49_deg_I need to make calls from this pabx through a VoIP trunk using g729, is it possible?
21:56.58brettnemI've had so much more success and stability with SER
21:57.10bdunnOKay... maybe stupid question.  How do I call a voice mailbox and enter the code?  When I dial the extension that goes to Voicemail(u3500) it only allows me to record a message.  Shouldn't I be able to push * or something to enter my access code?
21:57.14_deg_It olnly works when I have g711 on the trunk.
21:57.22brettnemsawyernet: well in your capture you are only sending calls to one phone..
21:57.32_DAWbdunn - VoicemailMain
21:57.37brettnemsawyernet: a snom?
21:57.59sawyernetbrettnem:  yea... but it is doing the same thing... When ast voice mail picks up the phone contiues to ring.  Polycom 600
21:58.54brettnemsawyernet: how many phones are we talking about (in that last scenario?) You mean you call from one phone to VM and you hear ringing AND VM in the same phone?
21:59.01gstbrettnem: wrong channel to say this :) but a voip setup would be much easier if asterisk supported call transfers without all legs being routed through it (because you wouldn't need to route all calls through it).
21:59.35_deg_Anybody here have expeirence with E1 and Asterisk?
21:59.37brettnemgst: nah,I think this is an ok channel.. maybe a developer will hear your cries for features.. :)
22:00.06brettnemgst: I'm trying to figure out how to make all the features work in SER.. avpops gets you quite a bit..
22:00.39brettnemgst: and Dustin over at vecsector.com mentioned to me that he wrote a voicemail server w/IMAP storage for vovida.. sooooo...
22:00.44sawyernetbrettnem:  in the trace I sent you I called my desk phone from my cell phone.  My deskphone rang for 20 sec, then I hear the voice mail message on my cellphone, but my desk phone keeps ringing.  If at that time I pick up my deskphone nothing is on the line.
22:00.53bdunn_DAW - Thanks... I get it.
22:01.06gstbrettnem: i suggest the openser fork. there are many cool features in their cvs-head just waiting to be released :)
22:01.31_deg_Someone could help me?
22:01.32pifiuanyone have experience with polycom 501 phones?
22:01.38brettnemsawyernet: ok, let me think about that..
22:01.55_deg_Peanuts problem....
22:01.56Umaroanthm: you around?
22:02.05sawyernetok... it just started today when I upgraded the CVS-HEAD to todays version... was running head before that
22:02.06brettnemgst: I'm not sure what you are refering to.. you just mean using openser??
22:02.19gstbrettnem: openser instead of ser
22:02.33DarthClueUmaro: anthm is busy at the moment, what ya need?
22:02.39brettnemgst: yeah.. I'm using a module that I couldn't get to compile in openser..
22:03.13UmaroDarthClue: I need to find someone who can help me/do it for me for $$ with app_talkdetect.c
22:03.28brettnemUmaro: do what?
22:03.53DarthClueUmaro: anthm is quite busy at the moment, but i'm sure if you /msg him he will get back to you as soon as he can.
22:04.20Umarowell, it just doesn't scale well.. even when I only have 3 calls going on, and the quality of the other 2 calls when one call is in backgroundetect is affected
22:04.42gstbrettnem: the problem is that when doing everything on asterisk you can't use features like CPL. currently we have written our call logic in Scheme (which is called by the Python asterisk module :P). with CPL this would be a little bit easier.
22:05.12brettnemgst: so why not switch it all to SER??
22:05.39brettnemsawyernet: I'm not sure what's going on here.. it looks like you are completing a call to that phone..
22:05.53gstbrettnem: because we need to support call transfers from pstn to sip and this isn't possible with asterisk as pstn gateway (because of the problem with the call legs i mentioned above).
22:05.59sawyernetyea but what is that 405 msg about?  Never got those before
22:06.00gstbrettnem: maybe this will be possible in 1.2
22:06.08brettnemgst: what is your pstn interface?
22:06.15brettnempri?t1?
22:06.21gstbrettnem: euroisdn pri
22:06.29brettnemgst: ah perfect.. use sems. :)
22:06.31*** join/#asterisk buddah (~hnic@208.179.86.5)
22:07.20buddahanyone know how to unlock the configuration menus on a cisco ip phone 7940?
22:07.42brettnembuddah: <settings>9 "cisco"
22:07.48buddahahh, thanks
22:07.54brettnemif it's still default..
22:08.12buddahany idea if they, or the ata 186s, support digest authentication?
22:08.23brettnemok.. it's been fun.. I gotta get home.. if anyone wants to sell me cisco phones.. find me..
22:09.27bdunnHow can I set things up so that when an extension is ringing, they hear musiconhold instead of ringing?
22:09.39brettnembdunn: show appliactions dial
22:10.33pifiuanyone have experience with polycom 501 phones?
22:10.56sawyernetpifiu: a bit... what you need
22:11.05*** part/#asterisk brettnem (~Brett@207.90.232.34)
22:11.07pifiulol i somewhat need a lot
22:11.17sawyernetok fire
22:11.35gstbrettnem: cool - i didn't know that sems does include an isdn plugin :)
22:11.35bdunnbrettnem - Thanks!
22:11.38pifiui have the phone setup already so it looks in tftp for the sip images
22:11.40pifiuthats all fine
22:11.55pifiubut i want the actual config file to know that NEXT time to look in a certain http address
22:12.02pifiuI was wondering how that could be done in the config file
22:12.31sawyernetcan't as far as I know.  you can't tell it where to look from the config file.  That has to be setup on the phone from my exp.
22:13.25pifiuthat makes no sense though
22:13.39pifiuif i want to change a stupid little thing in a setup of like 50 phones i have to go to that specific phone and change it?
22:14.01Error_XWhat is this error? NOTICE[2147]: chan_iax2.c:5773 socket_read: Rejected connect attempt from 65.39.205.12, requested/capability 0x4/0x4 incompatible with our capability 0xff03?
22:14.25sawyernetpifiu... well the way we did it here was using DHCP and FTP (NOT TFTP).  Do you have to use HTTP
22:14.30PyroSteveis my keyboard working ?
22:14.37_DAWpifiu - once you point to phone to a tftp server than it will look to that server for all changes.  You only need to goto the phone to tell it which tftp/ftp or http server to use.
22:14.47pifiuwell http seems cleaner than ftp
22:14.47Error_XPyroSteve: no :p
22:14.54*** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
22:15.20SuPrSluGforgot a priority in the dial plan. that's a bad thing.
22:17.12yaaarcatch you guys tomorrow....
22:17.19sawyernetnot really... but maybe you can do the same using DHCP for HTTP... the DHCP tells the phone where to go
22:17.58gst_deg_: do you have an g729 license?
22:18.12*** join/#asterisk hypa7ia (~leigh@f6c2b9a02c5834fd.session.tor)
22:18.15gst_deg_: otherwise you won't be able to convert it to g711
22:20.44*** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985890.sympatico.ca)
22:20.48*** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET)
22:20.58jeffikanybody familiar with aah/amp?
22:23.58Error_XI've changed my meetme config file.. What do I call to enter it? My asterisk number?
22:24.30hardwiretaking an IQ test
22:24.32hardwirethis is dissapointing
22:24.46hardwirethe main reason.. not enough IQ to know not to do this during work.
22:25.34*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
22:26.16*** join/#asterisk boch (~boch@201.255.169.146)
22:31.09*** join/#asterisk MikeJ__ (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
22:31.42*** join/#asterisk craziman2 (~Craziman2@boromir.apid.com)
22:33.29*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
22:33.54*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
22:40.16*** join/#asterisk L|NUX (~linux@202.5.146.154)
22:41.55niZonanyone know how i can play music on hold for a few seconds then do something?
22:42.23*** join/#asterisk Lathos42 (~Lathos42@68.77.108.51)
22:42.33Lathos42Evening #asterisk
22:42.58craziman2niZon  there is a music on hold command that lets you specify a time
22:43.36craziman2niZon WaitMusicOnHold(time)
22:43.50Lathos42Any Cluecon folks around?
22:44.00DarthClueLathos42: yes, why?
22:44.06*** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net)
22:44.16Lathos42DarthClue: Is there still room to register?
22:44.26DarthClueyes, if you do it right this second.
22:44.49FarrisGFor what are we registering?
22:44.56DarthClue~cluecon
22:44.56jbotcluecon is, like, http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Mark Spencer, Greg Boehnlein, Ken Rice, Brian West, Vikrant Mathur, Craig Southeren, David Sugar, Bob Andreasen, Joshua Colp, Greg Merriweather, Brian Fertig, Peter Nixon, Marc Olivier Chouinard, and Anthony Minessale II.
22:45.51Lathos42DarthClue: Ok
22:46.23Error_XHow does the meetme thing works? I have set my asterisk up for fwd, and added a room in meetme.conf.. How do I enter the room?
22:46.37*** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985455.sympatico.ca)
22:46.53file[laptop]Error_X: write your dialplan to do it...
22:47.04shmaltzError_X, you take out a cig and start smoking
22:47.04JerJermore shamless self promotion in #asterisk
22:47.05shmaltzRTFM
22:47.06hardwirebut how file!
22:47.17DarthClueLathos42: you registering now?  did we manage to get the boss convinced?
22:47.22file[laptop]hardwire: I believe it involves using a computer.
22:47.29hardwirehmm
22:47.32hardwirethis iaxy is being a mofo
22:47.49hardwirejust cause its a mofo doesn't mean I need a poking.
22:48.02Lathos42DarthClue: I think so.. when I called him earlier he mentioned that I should be able to go.. but I wasnt exactly in a state of mind to talk details with him
22:48.32JerJerdinner!
22:48.53Error_XLathos42: How old was he? heh
22:49.31Lathos42Error_X:  He was 7 I think..  we had just taken him to the vet yesterday and they said he had a bladder infection
22:49.41Error_Xk :s
22:50.32Lathos42DarthClue: I left my boss a voicemail on his cell phone.. should I get on and get myself registered before I get the final word to guarantee a spot?
22:50.44*** join/#asterisk mstocco (~mario@207.212.29.195)
22:51.31*** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com)
22:51.48DarthClueLathos42: just let us know as soon as you can.  we'll note you a spot until we hear otherwise.
22:52.25JerJerLathos42: save your boss some loot and have him hire an asterisk consultant (not me) for a few hours
22:53.20DarthClueJerJer: it is so nice to see that you have no clue about reality.  a consultant will cost him more than going to the conference and learning it for himself.
22:53.41JerJerif you say so
22:53.54JerJera consultant has already gotten over the learning curve and can implement exactly what they want
22:53.56drumkillawellll ... depends how much you value your own time :D
22:54.10JerJernot what those few speakers want to talk about
22:54.27DarthClueand then they are dependent on the consultant until they learn it themselves.
22:54.54JerJernot if the consultant comments his configs well
22:54.58JerJerand teaches along the way
22:55.19JerJerand subscribes to the KISS philosophy
22:56.06DarthClueCluecon will provide more than enough information to get Lathos42 what he wants.  We look forward to having you there if you can make it.
22:56.06Lathos42I'm pretty comfortable with the configuration of it right now
22:57.33Lathos42I'm thinking it'll be alot more useful than the MFG/PRO conferences my boss is always going to
22:57.56JerJerif you just want to drink some beers and talk shit with other asterisk phreaks, go
22:58.04JerJerbut don't plan on learning all that much
22:58.44Lathos42DarthClue: Did JerJer not get an invite or something?  He's awful bitter
22:58.50bkw_JerJer, please don't even go there
22:59.07*** part/#asterisk clive- (~pirch@rrba-146-74-170.telkomadsl.co.za)
22:59.21Lathos42Ooh, I think my boss is calling
23:00.40drumkillammmmmmmmmm ... beeeerrrr
23:01.02Corydon-wMmmmmm... warm beer...
23:01.19Error_X*Ush*
23:02.04dalaberaQuick Question: I have a T400P card, and the interrupts that it's using keep growing. Is there a limit for that or should I reset the server in order to reset the interrupts??
23:02.24drumkillathe interrupts should keep growing
23:02.27drumkillathat is correct operation
23:02.34drumkillaif it *wasn't* growing, then you should be concerned :)
23:02.41drumkillathat would mean that the card was not taking interrupts
23:03.00Corydon-wMore to the point, what is it doing wrong that you think the interrupts are the cause?
23:03.24anthmpfft, one thing i cant stand is warm fuckin' beer makes me want to fucking puke!
23:03.32Corydon-wOr if you have bipolar violations, those are also of concern
23:03.38harryvvor just dont drink it
23:03.50Corydon-wanthm: warm beer is its natural habitat...
23:04.04drumkillawarm beer is gross.
23:04.18Corydon-wCold beer loses most of its complex taste
23:04.26anthmaww nobody got it. that was a frank booth quote from blue velvet.
23:04.28*** join/#asterisk xheliox (~jeff@user-0c6se1v.cable.mindspring.com)
23:04.29Corydon-wdeadens the taste buds
23:04.30anthm=p
23:04.34hardwireok
23:04.34hardwireso
23:04.35hardwirehi
23:04.35hardwirehow
23:04.36hardwireare
23:04.36hardwireyou
23:04.47hardwiredamn.. soda is kicking in.
23:04.49drumkilla~thwack hardwire
23:04.49jbotACTION beats hardwire on the eblow with a UNIX Manual
23:04.51hardwireand the iaxy works.
23:04.51drumkillaquit that
23:04.56hardwiredrumkilla: i know..
23:05.14drumkillahardwire: you should check out gtkiaxyprov :)
23:05.15Exstaticawtf mate
23:05.32Exstaticai don't need a mofo request
23:05.35hardwiredrumkilla: I was thinking I should check out the iaxprov.conf and templates
23:05.43hardwireExstatica: everybody got one..
23:05.58hypa7iapercussive maintenance!
23:06.06hypa7iaworks every time.
23:06.11hypa7iaokay, almost :-)
23:06.27drumkillahypa7ia: !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
23:06.40hypa7iaHIHI
23:06.53dalaberathanks for your response about the interrupts, it just that I was concern about a limit for the interrupts keep growing ...
23:06.59drumkillahypa7ia: you're my hero
23:07.54drumkillaha, not!
23:07.57drumkilla~frag hypa7ia
23:07.57jbotACTION readies the nuke launcher and fires some rounds at hypa7ia
23:08.14jeffikanyone familiar with *@home?
23:08.19hardwiredrumkilla: where do you even find gtkiaxyprov?
23:08.25xhelioxThis isn't really an Asterisk question, but you all would probably know best. :) I'm trying to dial from Asterisk to a Partner II system, does anyone happen to know what prefix I need to dial on a Partner II to get an internal extension (or intercom as they seem to call it)?
23:08.39JerJerbkw_:  i did go there and will continue to do so
23:09.14JerJerLathos42: you will see after the con is over
23:09.18drumkillathere's no hiding from the nuke!
23:09.26JerJerbe sure to report back how much you actually learned
23:09.45JerJerand how drunk you got
23:10.05*** join/#asterisk hhh_ (Gone@i-195-137-50-162.freedom2surf.net)
23:10.13outtoluncand if there were 'stickers' <G>
23:12.05hardwiremmm
23:12.06hardwirestickers
23:12.08*** join/#asterisk Legend (~legend@24.244.142.133)
23:12.08hardwireerr
23:12.09hardwireI want snickers
23:12.13hardwiremmmm
23:13.46outtolunci'm hungover, i want taco bell <G>
23:13.55blitzragetaco bell is gross
23:14.03outtoluncgreat for hangovers tho
23:14.04hardwiremeet of dooom
23:14.13hardwireno.. meet the deedles is great for hangovers
23:14.17hardwirethats such a dumb movie
23:14.25outtoluncnever heard of it
23:14.37JerJernothing like freeze dried meet that is reconsituted 20 minutes before the customer consumes it
23:14.45outtoluncnods
23:14.52Legendits teh yum!
23:15.04outtoluncbbl
23:15.13hardwirereconstituted is a fun word
23:16.11hardwirefor instance.. I had some reconstituted rice the other day.
23:16.13JerJerLathos42:  and i resent your comment about being bitter - If they have the right to push their con, i have the right to push my reasons for not attending
23:16.26hardwireI have yet to see reconstitutable beef or beans
23:16.28hardwirebut I am waiting
23:16.29hardwireslowly
23:16.31hardwirefor it to happen
23:16.33MikeJ[Laptop]JerJer, hey!
23:16.38JerJerhardwire: taco bell
23:16.45hardwireJerJer: I wanna see it before hand
23:16.50JerJerits nasty
23:16.53JerJereven the cheese is powder
23:16.58hardwirewell if I saw it before hand
23:17.05hardwireit would stop me from my usual orderings at taco bell
23:17.10hardwireeven though the lady there is super friendly
23:17.11*** join/#asterisk meppl (~mephisto@87.193.4.17)
23:17.15hardwirein my case
23:17.22JerJerMikeJ[Laptop]:  no
23:17.24bkw_wooo Someone has sand in their vagina
23:17.30MikeJ[Laptop]:P
23:17.35hardwirebkw_: ?
23:17.36MikeJ[Laptop]uhhhh
23:17.39JerJeryep my pussy is broke
23:17.42*** join/#asterisk darkskiez (~darkskiez@host-84-9-85-42.bulldogdsl.com)
23:17.44JerJerso deal with it
23:17.47MikeJ[Laptop]you can say that again
23:17.54hardwireI should work
23:18.08JerJercuz its gonna bleed until August 6th
23:18.22bkw_JerJer I don't know why you're being such a prick to everyone today
23:18.31anthmooh the day we start the campaign for ClueCon II
23:18.40JerJerthen its gonna bleed longer
23:18.45dalaberabtw has anyone have any experience with Dialogic and Asterisk, specific board: D480JCT2T1
23:18.55JerJerbkw_:  you tell me
23:19.25MikeJ[Laptop]I know how it is.. mine goes all month long
23:19.27MikeJ[Laptop]hehe
23:20.01blitzrageis there a jbot command to print instructions for identifying yourself to join Asterisk?
23:20.10bkw_JerJer do you want me to read you?  You're still all pissed off that your employee snatched up the speaking spot at Cluecon behind your back and you're acting like a two year old.
23:20.19blitzrageor do I need to make one? what keyword do you like? :)
23:20.35JerJerok you want truth
23:20.42bkw_sure shoot
23:21.21JerJerbkw_:  i'm tired of your "Holier-than-thou" fucking attitude
23:21.36bkw_what?
23:21.36JerJeranthm's too
23:21.38MikeJ[Laptop]blitzrage, seems kind of silly.. how would someone use it cuz they can't get inthe chan
23:21.45blitzrageMikeJ[Laptop]: look in #asterisk-doc
23:21.51bkw_I have never had a "Holier-than-thou" attitude
23:21.53JerJeryou constantly bitch about getting shit into cvs
23:21.54blitzrageMikeJ[Laptop]: now it makes sense :)
23:22.02MikeJ[Laptop]heh.. he's in there too
23:22.03JerJerthen when it doesn't go your way you whine
23:22.11JerJerthen you pull disclaimers
23:22.15hardwireand to think I have been bitchier than that for less all day
23:22.16JerJerand pull code out of cvs
23:22.17hardwireI feel bad now
23:22.20JerJerthen bitch some more
23:22.25bkw_res_perl was NEVER disclaimed to digium
23:22.26JerJerthen start a conf just to make money
23:22.28bkw_NEVER
23:22.38bkw_I see what this is all aobut now
23:22.43bkw_I totally see
23:22.49JerJerwhich you lied to many about what the conf was about
23:22.58bkw_what?
23:22.59bkw_how?
23:23.10anthmwe whine to get a bug fix in CVS?
23:23.16anthmdamn us
23:23.19bkw_we have dev talks weekly on the conf
23:23.23bkw_we drive development
23:23.23JerJer(13:19:46) bkw_: kpfleming, http://bugs.digium.com/view.php?id=4760
23:23.30JerJer(13:21:15) kpfleming: and you don't think i'd look at it unless you told me about it here?
23:23.33bkw_yes and we DO NOT use that feature
23:23.45bkw_but we did fix it
23:23.50JerJerwho gives a fuck
23:23.53bkw_it might as well go in because it fixes a very real issue
23:24.09JerJeryou still were a squeeky wheel and kpfleming greesed it
23:24.12JerJerto shut you up
23:24.17bkw_no he didn't
23:24.26bkw_its still open
23:24.37bkw_its not commited
23:24.44JerJer(13:30:21) kpfleming: and certainly pestering people about your new patches makes them get in a lot faster too, since i always drop everything i am doing and go look when you tell me a patch has been updated
23:24.50bkw_I posted to the lists to get more people interested in it
23:25.02JerJerbig picture
23:25.09JerJerif anything doesn't go your way you bitch about it
23:25.24bkw_what?
23:25.33bkw_trying to fix a bug and help out is bitching
23:25.40bkw_well color me purple
23:25.49anthmyou're trying to generalize today into and excuse for bitching at us.
23:26.02bkw_he doesn't realize I bitch about everything.. every day.. no matter what
23:26.06bkw_it doesn't matter
23:26.11JerJerthen all this fucking pushing of cluecon every 9 minutes
23:26.19JerJeri am not the only one that is anonyed here
23:26.26JerJeri am just the only one making an issue of it
23:26.27anthmhas it been that long already HEY COME TO CLUECON
23:27.13JerJeri'm still waitng for someone to tell me when was the last time oej or sokol mentioned astricon in here
23:27.20JerJermuch less pushed it
23:27.29bkw_JerJer, the first astricon they were all over the place with it...
23:27.35JerJerum no
23:27.43JerJerno where near
23:27.53anthmthey are too busy counting the $$ from all the useless certs they sell for several grand each.
23:28.01JerJeryou guys have an alter ego that does nothing that pushes your con
23:28.19harryvvI dont
23:28.21JerJerand don't lie i've watched the idents
23:28.28bkw_what?
23:28.33harryvvohh brother
23:28.41bkw_JerJer, no you're just making shit up
23:28.48JerJerif you say so
23:29.01*** join/#asterisk laserfox (~jimbob@81-179-127-14.dsl.pipex.com)
23:29.05bkw_we have not had anyone come in here nor have we created fake nicks to advertise cluecon
23:29.12JerJerlol
23:29.20JerJerso DarthClue is someones real nick
23:29.24JerJerfunny
23:29.25bkw_yes
23:29.29file[laptop]yes it is
23:29.30JerJerif you say so
23:29.32bkw_it actualy is you wanna talk to him on the phone
23:29.54twisted[asteria]c'mon now guys, play nice
23:29.58file[laptop]yes, I have an alternate personality that lives in Tulsa
23:29.59harryvvbkw msg
23:30.28Darwin35File{flattop} lives in tulsa
23:30.49*** join/#asterisk zapa (~zapa@200.66.20.72)
23:30.56bkw_flattop?
23:30.57JerJerbkw_:  then what about all the bitching "Mark won't let that in to cvs because he doesn't like the name"
23:30.59JerJerand so on
23:31.04JerJerMark won't let this on
23:31.05JerJerin
23:31.11JerJeror Mark this or Mark that
23:31.27mstoccofor those of you new to the asterisk channel, it really is never this hot
23:31.28bkw_thats exactly why valetparking isn't in CVS
23:31.31bkw_mark didn't like the name
23:31.38bkw_and we refused to change it
23:31.40twisted[asteria]mstocco, haha... it's like 85F
23:31.41JerJeri'm sure there is much more to it
23:31.42bkw_thus it didn't go in
23:31.48bkw_JerJer, actually no thats it
23:31.55bkw_mark didn't want people to confuse parking with valetparking
23:31.58bkw_that is the bottom line
23:32.17JerJerso why not yank parking if yours is that much better?
23:32.21twisted[asteria]I'm not taking sides here, but I can attest that the real reason is the name.
23:32.23JerJerever thought about that
23:32.26bkw_I mean is it not enough that we have given the modules away on pbxfreeware.org?
23:32.34bkw_you can't yank parking its built into res_features
23:32.39bkw_which is where the stupid bridge function lives
23:32.43JerJerits open source yo
23:32.48JerJeranything can be changed
23:32.48bkw_why bother?
23:33.00Darwin35has to pass threw mark first
23:33.09JerJerno Kevin
23:33.14Darwin35and mark does not want to change it so it wont change
23:33.15JerJerthen possibly Mark if its major
23:33.21JerJerbut its mark's baby
23:33.23MikeJ[Laptop]threw?  hmmmm
23:33.28twisted[asteria]valetparking/parking issue was pre-Kevin
23:33.30JerJernone of us would be here bitching today if Mark had not released it
23:33.38JerJerin the first place
23:33.45JerJerdon't forget about that
23:34.10bkw_well don't forget the MANY hours anthm and I have put into this
23:34.17bkw_we have thousands of lines of code we have give back
23:34.23MikeJ[Laptop]who cares if people bitch.. why is this worth caring about
23:34.26Darwin35my question is this why are we diging up the past and redredging it. its over
23:34.29Darwin35move on
23:34.32*** join/#asterisk rvhi (~rv@66.175.65.89)
23:34.34bkw_no clue
23:34.42JerJerDarwin35: he wanted the truth
23:34.42Darwin35you 2 dont see eye to eye and should just walk away
23:34.51MikeJ[Laptop]if somone is upset, they can bitch... if somone does not want to hear them bitch, put them on ignore...
23:34.54MikeJ[Laptop]who cares.
23:34.56anthmyah darwin i am enjoying this
23:35.22twisted[asteria]can't we all just, hit a gong?
23:35.24Darwin35I stand by Anthm and BKW they have done alot for asterisk
23:35.28MikeJ[Laptop]twisted, nice
23:35.29filetwisted[asteria]: bong!
23:35.30Darwin35they put in lots of time
23:35.44twisted[asteria]file: i'll leave that open to creative intrepretation
23:35.47rvhiin ACD, is it possible to make agent 1 always gets the call if he is not busy?
23:35.50Darwin35and I think they have gotten the shaft on some things
23:35.51filetwisted[asteria]: I thought so.
23:35.54JerJerDarwin35:  so you are saying I haven't?
23:36.09MikeJ[Laptop]rvhi, sure, don't add any other agents
23:36.10JerJerDarwin35:  only due to their own stubbornness
23:36.48Darwin35JerJer I dont know what you have done. but no I am niot saying that but insted of fighting why not communicate and work as a team to better things for the future insted of fighiing
23:37.06MikeJ[Laptop]awww.. can't we all get along
23:37.12mstoccorvhi: if I am not mistaken, round-robin starts at the lowest agent number first
23:37.32Darwin35no get along persay but learn to put things in the past and learn to work with eachother
23:37.34JerJerDarwin35: not if they won't disclaim their code
23:37.43bkw_what do you mean JerJer
23:37.49JerJereverything that i have released has been disclaimed
23:38.10MikeJ[Laptop]duuuuudddee
23:38.29MikeJ[Laptop]I'm feeling gooooooood
23:38.39filemellow.
23:38.47blitzragecalm blue sky
23:38.52*** join/#asterisk eville83 (~sdfas@CPE001195351498-CM014370000248.cpe.net.cable.rogers.com)
23:39.10*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
23:39.29*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
23:39.45xtrvdAnybody know where I can find information on Intelligent Call Distribution?
23:39.54bkw_JerJer, really I wanna know where in the book it says we must disclaim code to digium?
23:39.59blitzrage~google intelligent call distribution
23:40.06*** join/#asterisk derek_1234 (~derek@203.167.203.10)
23:40.07JerJerbkw_:  its called doing the right thing
23:40.11harryvvwhat book?
23:40.14bkw_what?
23:40.17xtrvd:P
23:40.17bkw_the right thing?
23:40.19anthmso you profess that I am required to give away every line of code I ever write?
23:40.27JerJerand that attitude right there is what pisses me off
23:40.28anthm80% of it is not enough
23:40.38bkw_what attitude? we give it away.. how much more free can it be?
23:40.41xtrvdThere isn't any valid information in the wiki, and I haven't been able to find anything on google. That's why I'm asking here.
23:40.43JerJeranthm:  you are putting words into my mouth
23:40.49derek_1234Freedom man.
23:40.54derek_1234Freedom is what you talk about
23:41.05bkw_the src is out there.. I don't see where we are required to disclaim it
23:41.07derek_1234Anthm has the freedom to do what he chooses with his code.
23:41.13bkw_if thats the case jump on coppice too for spands and rx/txfax
23:41.20derek_1234Do Americans believe in freedom ?
23:41.24derek_1234surely they do.
23:41.26anthmcan you please elaborate on what you mean exactly in terms of disclaming code then so I do not misunderstang?
23:41.26bkw_I do
23:41.29derek_1234then leave anthm alone.
23:41.32JerJercoppice isn't forcing a conf down our throats every 9 minutes
23:41.53JerJeror bitching about not getting a bug dealt with
23:42.01JerJeror complaining that Mark won't let something else in
23:42.17*** join/#asterisk allanon (allanon@c-24-18-189-146.hsd1.wa.comcast.net)
23:42.20JerJerneither is capigod
23:42.34bkw_JerJer, someone has to be a forward motion in the project... or bugs sit and rot.. patches go stale... people loose interest
23:42.37eville83does anyone here know why I'm getting     "modprobe: can't locate module wxfxo"   when i input the line "modprobe wxfxo".   im trying to install the drivers for a Wildcard X100P card
23:42.47bkw_eville83, the module isn't installed
23:42.50bkw_depmod -a
23:42.52anthmoh yeah that reminds me
23:42.53bkw_and tray gain
23:42.56bkw_er try
23:43.01eville83k, thanx i'll try that right now
23:43.14Darwin35Kram has alot on his plate and BKW has been doing alot to keep asterisk in a forward motion on his free time
23:43.15NewSoleanyone up for a challenge.... when someone calls using iax2 to a sip client or peer asterisk uses a g729 codec license even thogh both are using g729 can we not make a proper passthough for both iax2 and sip so it does not use license when passing it though
23:43.20anthmooh ooh you just reminded me
23:43.31anthmlets quote mark from the asterisk CREDITS file
23:43.34anthmAnthony Minessale - Countless big and small fixes, and relentless forward push
23:43.40JerJerDarwin35: don't forget about Kevin
23:43.48Darwin35I dont know Kevin
23:43.48*** join/#asterisk remmo (~rem@smack.isp.net.au)
23:43.52JerJeranthm:  good for you - i am not even in the credits file
23:44.06JerJeryet how much shit have i put up with?
23:44.12anthmthat would suggest pushing forward was desired
23:44.19anthmmy only point
23:44.20Darwin35BUt I have been dealing with BKW and ANTHM since I first came to the project . and they gave me help where others did not
23:44.26bkw_yes you are
23:44.43bkw_eremy McNamara - SpeeX support
23:44.45bkw_er Jeremy McNamara - SpeeX support
23:44.46Darwin35even though then did point out to me alot that fbsd was not a supported os they still helped
23:44.53anthmlogicaly if it thanks me for relentless forward push then it should be my duty to proceed with it
23:45.01JerJerWishlist  yay
23:45.06JerJerso i paid mark to develop something
23:45.11derek_1234Jerjer, how about putting your time into helping people, rather than whinging ?
23:45.13eville83bkw_: i tried that and it gave me the same response again.     i thought i installed the module.  im doing everything line for line according to the "getting started with asterisk" guide
23:45.18JerJerderek_1234:  nope
23:45.38eville83do you know of any good documentation for setting up the X100P?
23:45.39JerJereville83:  lspci - is it listed?
23:46.14JerJerderek_1234:  i am in bitch mode tonight and there isn't much that's gonna stop me
23:46.20eville83let me check
23:46.36NewSoleanyone up for a challenge.... when someone calls using iax2 to a sip client or peer asterisk uses a g729 codec license even thogh both are using g729 can we not make a proper passthough for both iax2 and sip so it does not use license when passing it though
23:46.53*** join/#asterisk CyberSword (~Cyberswor@cablea0mle.cybercable.net.mx)
23:47.01JerJerNewSole:  g.729 is allowed thru the whole path?
23:47.14JerJerand you are not using |r or t or T dial modifers?
23:47.24*** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com)
23:47.38NewSoleit does not everytime I call a sip line from iax it uses a license
23:47.46eville83JerJer: what do you mean by lspci - is it listed?
23:47.52JerJertype lspci
23:47.58fileLathos42: mmm iBook
23:48.30JerJerdo you see the X100P device listed
23:48.35JerJerin the output
23:48.47Lathos42file: I have the G3 500 model, but it does pretty good for its age
23:49.03blitzragepowerbooks suck
23:49.05JerJeranthm:  Mark doesn't like rocking the boat
23:49.09fileblitzrage: YOU SUCK!
23:49.11JerJerso perhaps he was just being nice
23:49.15*** join/#asterisk stustu (~stustu@fluffy.fatburen.org)
23:49.21CyberSwordhi , how can i do for sip externals users can log into my server?
23:49.24JerJerand/or politically correct
23:49.33JerJerwhich i am most certainly not
23:49.47NewSolethats lame it should pass though even if you are using dial modifiyers
23:49.56JerJerit can't
23:50.56stustuIs anyone here getting the warning "Have a packet that doesn't want to give up!" from chan_zip?
23:51.13stustu(Except me...?)
23:51.18*** join/#asterisk cgcorea (~cgcorea@63.245.14.194)
23:51.27JerJerchan_zip
23:51.37anthmactually he has told me more than once to never stop pushing so how bout we ask him and if he says so, I'll never push another thing into asterisk
23:51.38JerJeris that like Z compressed IP ?
23:51.39stustuSorry! chan_sip!
23:51.58remmocan anyone tell me the status of freebsd asterisk and zaptel?
23:52.21eville83jerjer: when i type ispci i get bad command, should i type that when im connected with asterisk?
23:52.22stustuI'm running Current on FreeBSD.
23:52.29JerJereville83:  no
23:52.35JerJerthe shell smart guy
23:52.38derek_1234lspci is a shell command
23:52.50derek_1234lspci is a command you type in when logged in as root to a shell.
23:53.11stustuAs for zaptel, I can only comment on the wcfxs driver, and it seems to be working ok.  Only one analog phone, though.
23:53.15derek_1234lspci is a bit like the ls command.
23:53.25derek_1234except that lspci list the compoents on the PCI bus
23:53.50JerJeranthm:  because mark is not like that
23:55.06stustuRegarding FreeBSD: I do have a problem with SIP registration currently, but I do not know if it is FreeBSD specific.
23:58.09JerJeranthm:  but after all Asterisk is Mark's baby - he deserves the right to a have a decision on how it is going to grow up
23:59.07derek_1234yep, mrk has the right to make decisions on his baby.
23:59.16JerJerthese same issues come up with Linus - hence why there are so many kernel variants out there
23:59.20derek_1234However, Anthm has the right to not follow marks lead.
23:59.26*** join/#asterisk jsaunders (jsaunders@70.70.74.153)
23:59.32anthmwhy sure, that's why I give him a heads up on everything I make
23:59.37anthmasl him
23:59.39anthmask
23:59.47JerJerbut do we really want to fragment the effort?

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