irclog2html for #asterisk on 20050622

00:01.42Ariel_sivana, it's been up and down all day
00:02.00SarahEmmfor days...
00:04.03dougheckadrink water
00:04.07dougheckaits better than coffee
00:04.17fugitivotry decafeinated
00:04.31dougheckatry unleaded
00:04.47fugitivodrink tea
00:04.57dougheckadrink ocean water
00:04.59Robot_i agree with tea
00:05.00fugitivoor mate
00:05.02Ariel_not the same coffee is well coffee.. yummmm
00:05.16fugitivodid you try mate?
00:05.20hardwirehttp://www.redcoat.net/pics/ponchomo.jpg
00:05.42dougheckahahah
00:05.52dougheckawas that taken at the astricon?
00:05.55Ariel_ponch....
00:08.08*** join/#asterisk Nukemizer (~Nuke@67.137.28.165)
00:15.15fugitivohttp://www.thinkgeek.com/caffeine/drinks/6147/
00:15.20*** join/#asterisk MagicFab (~chatzilla@modemcable094.36-203-24.mc.videotron.ca)
00:15.32MagicFabHello
00:15.42fugitivomuch better than coffee
00:15.55MagicFabWondering if there a french / quebec asterisk  user group(s) ?
00:17.09NewSoledamd french... they always want to be seperest....
00:17.37NewSolethey want to seperate from canada and now they want thier own groups
00:17.57fugitivolol
00:18.21MagicFabAny other smart people around ? :P
00:18.38NewSoleasterisk is a US Product from califonia.... you want to seperate from that... microsoft offers Live server
00:18.46fugitivonot me, sorry :)
00:20.05NewSolesorry... I FELL NO PITTY for Quebeckers.... The Demand you speek french and english in Ontario.... but they make it illegal to put an english sign up in quebec
00:20.34MagicFabOf course everyone should speak only english and be as ignorant as NS - get a life
00:20.40*** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca)
00:20.47MagicFabthe law changed long ago BTW
00:20.52MagicFab:D :D
00:20.53NewSoleno I belive fair is fair...
00:21.19NewSoleno... it changed that you have to have french first then english...
00:21.19sivanaNS = nova scotia? :)
00:21.29fugitivoi agree with quebec goverment, some languages are dying (ex: french)
00:21.38sivanaand english can only be 1px font size
00:22.07fugitivoenglish is going to die soon
00:22.28NewSoleenglish will die and everyone will speek geek
00:22.32Ariel_english is not going to die.
00:22.45colinm_pig latin for the masses!
00:22.48Ariel_english is going to morph
00:23.42fugitivowe'll all speak speranto
00:23.51MagicFabTrolling is so '90s
00:24.04MagicFaba well.. back to the n00b forums
00:24.18MagicFabtx for nothing
00:26.29NewSolesorry folks... I live near Ottawa and I think its bull crap... people from hull look at people like they are skum cause they dont talk to them in french
00:26.49sivanaquoi?
00:27.00NewSoleand when ever there is something good out they demand a french version of it.
00:28.12sivanaAsteriske
00:29.09NewSolemy friend worked in public works in hull...
00:29.29NewSoleand they fired him because all his C code was not French
00:30.25sivanayou mean the comments?
00:30.34NewSoleno the code
00:30.54NewSolehis comments were in english and french
00:31.04sivanahow can C code be french?
00:31.38NewSolethats why it was stupid...
00:31.46NewSoleC code is english based
00:31.55SpaceBassalors qu'est-ce-cest le problem?
00:32.09Ariel_SpaceBass, says what's the problem
00:32.20SpaceBassvery poorly at that
00:32.22NewSoleI know
00:32.24JerJerthe functions he created could be named using the frechy lingo
00:32.36sivanaah ya
00:32.39NewSoleya but not print
00:32.46NewSoleor while
00:32.49*** join/#asterisk popooya (~popooya@07738e59f333d31b.session.tor)
00:32.49NewSoleor for
00:32.58NewSoleetc...
00:34.01b0ltHey
00:34.06JerJerhoe
00:34.11b0ltI just installed asterisk, and want to use it with broadvoice
00:34.18JerJergood luck
00:34.40b0ltis there any way I could plug a phone into a modem on the asterisk server to get a dial tone, etc?
00:34.44SpaceBassb0lt did you check out the article from geek gazette on it
00:34.51b0ltSpaceBass, yeah
00:35.04b0ltexcept it uses official digitel hardware
00:35.05b0lt00:03.0 Communication controller: Rockwell International HCF 56k Data/Fax Modem (rev 01)
00:35.11b0lti have a conexant softmodem =)
00:35.22b0lt(using linuxant drivers)
00:35.34b0ltthe modem operates through /dev/modem just like any other
00:35.39SpaceBassb0lt typically regular modems lack the digital audio converters for voice
00:36.06b0ltthis one said it was a voice/fax modem on the box
00:36.25sivanaSpaceBass: where's the article?
00:36.41SpaceBassb0lt my understanding is that it could work in theroy if you had drivers... but I've never heard of winmodem drivers existing
00:36.52SpaceBassyou can get an x100p card on ebay for $10.00
00:36.56SpaceBassthats your best bet
00:36.57b0lthttp://www.linuxant.com/company/
00:37.10*** join/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net)
00:37.45SpaceBassbut those aren't * drivers
00:38.09b0ltyeah, they are
00:38.25b0lthttp://www.linuxant.com/drivers/
00:38.50SpaceBasshummm not seeing any
00:39.31SpaceBasssivana I'm looking for the article
00:39.42sivanaSpaceBass: about Broadvoice?
00:39.44niZonyay
00:39.46b0ltSpaceBass, http://www.linuxant.com/drivers/hcf/downloads-license.php
00:39.49SpaceBasssivana yep
00:39.50niZonlink2voip gave me a refund
00:39.58b0lthttp://www.google.com/url?sa=U&start=3&q=http%3A//geekgazette.com/index.php%3Foption%3Dcom_content%26task%3Dview%26id%3D20%26Itemid%3D26&ei=W7O4QsGuAqCUsAHyuayWCw&sig2=p3gbWcoHyQQRo4F7EOBq4w
00:41.14SpaceBasssivana thats the one   http://geekgazette.com/index.php?option=com_content&task=view&id=2&Itemid=26
00:41.46SpaceBassb0lt that linuxant.com link was for a license
00:41.46mepplgute nacht  -  good night
00:41.59harryvvsummer has not reached vanouver bc yet
00:42.00harryvv:)
00:43.00*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) [NETSPLIT VICTIM]
00:44.41*** join/#asterisk drbrown (~chatzilla@user-0cdv208.cable.mindspring.com)
00:44.56Pete_Largoanyone using voip 'did' numbers for incoming faxes?
00:45.38*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
00:45.42SpaceBassPete_Largo not yet... plan on trying
00:45.57Pete_Largohow are you thinking of doing it spacebass?
00:46.30SpaceBassPete_Largo in what sense?
00:46.38SpaceBassthinking of using a BV account
00:46.52Pete_Largopoint the did to a fax machine/ata or some sort of fax to mail?
00:47.02SpaceBassusing amp?
00:47.10Pete_Largoamp?
00:47.13Pete_Largo~amp
00:47.14jbotit has been said that amp is an Audio MPEG Player.  [non-free], or http://amp.coalescentsystems.ca/
00:47.21Pete_Largothat amp?
00:47.30SpaceBass* management portal - web front end
00:47.43SpaceBasscomes with *@home
00:47.51Pete_Largowell, ok, but what about the back end?
00:48.08SpaceBassleme look at my fax setup
00:49.10*** join/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com)
00:49.18SpaceBassPete_Largo well, I have a ext-fax context, so I have the did just do a s,1,goto(ext-fax,in_fax,1)
00:49.53SpaceBassbut that ext-fax is from amp/asterisk@home I think
00:50.10hasslerdid I just jump in the middle of a fax discussion?
00:50.13Pete_Largowhat is in_fax ?
00:50.21Pete_Largoyeah, trying to figure out if * can do it
00:50.40SpaceBassit works on my pstn line
00:50.49harryvvnow is asterisk at home a iso and what OS does it use?
00:50.49hasslermine are always garbled -- someone said timing problems in zap are causing it, but haven't been able to verify
00:50.58SpaceBassI'll patebin my ext-fax
00:51.25Pete_Largothanks
00:53.55*** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
00:54.20SpaceBasshttp://pastebin.ca/15449
00:55.37khemirharryvv: Centos 4
00:55.41colinm_harryvv: it won't run from the cd, but it is distributed as an iso. based on centos, which is based on rhel.
00:56.02bkw_oh guys
00:56.16bkw_go check pbxfreeware.org out.. I just posted a more extensive res_js example
00:56.23bkw_that interfaces with an external website
00:57.42bkw_lalalla
00:57.44bkw_NEXT!!!!!
00:57.49WilliamKhey brian
00:57.51dougheckahail bkw_
00:57.57bkw_WilliamK, yes?
00:58.28bkw_wasabi
00:58.51WilliamKlots
00:58.52WilliamK=)
00:59.03WilliamKlibpri was broken a bit ago after that one major update of ANI
00:59.05harryvvkhemir: is centos like fedora?
00:59.13bkw_you coming to cluecon?
00:59.14dougheckawasabi
00:59.16dougheckahuh
00:59.19WilliamK-- Extension 'AD D' in context 'incoming-trunk01' from '9038834303' does not exist.  Rejecting call on channel 0/1, span 1
00:59.32WilliamKno idea how it came up with AD D
00:59.57dougheckabkw_ make it closer and I will go
00:59.58SpaceBassPete_Largo - does that help? I use it for distenctive ring right now
00:59.59clueconharryvv: centos is based on rhel.
01:00.12Pete_Largocan you pastebin the macro?
01:00.19SpaceBasswhich'en?
01:00.27SpaceBassfaxrecieve?
01:01.02*** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca)
01:01.10doughecka:)
01:01.18SarahEmmheya ;)
01:01.24SpaceBass[macro-faxreceive]
01:01.24SpaceBassexten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
01:01.24SpaceBassexten => s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
01:01.24SpaceBassexten => s,3,rxfax(${FAXFILE})
01:01.24SpaceBassexten => s,103,SetVar(EMAILADDR=${FAX_RX_EMAIL})
01:01.24SpaceBassexten => s,104,Goto(3)
01:01.28SpaceBassoops... sorry
01:01.28harryvvi see
01:01.33SpaceBassitchy trigger finger
01:01.36bkw_SpaceBass, DUDE
01:01.39bkw_he said pastebin
01:01.41bkw_NOT HERE boi
01:01.43bkw_*SMACK*
01:01.54hasslerare you folks receiving faxes correctly?
01:02.07bkw_yes
01:02.09bkw_why?
01:02.13bkw_you have squished ones?
01:02.16hasslermine are always garbled
01:02.21bkw_on a PRI?
01:02.22dougheckabkw_ how reliable are your faxes?
01:02.24hasslersquished may be a good description
01:02.24bkw_check your clocking
01:02.31bkw_your clock src is wrong
01:02.34bkw_and you have frame slips
01:02.39bkw_you won't hear it usually on voice calls
01:02.42bkw_but its a clicking noice
01:02.44SarahEmmi'm not.. but i don't have it set up
01:02.45bkw_er noise
01:03.06hasslerhow? using digium wildcard tdm400p card
01:03.14dougheckahahaha
01:03.43bkw_hassler, no go
01:03.49bkw_you can't use a tdm400p for faxing
01:03.51bkw_it just won't work
01:03.53bkw_give up now
01:03.58bkw_you'll never have it work right
01:04.14hasslerthat's what I've been hearing, someone else said it was a timing issue broken in zap code
01:04.26dougheckabkw_: would you rely on ALL faxes coming in through asterisk?
01:04.29bkw_no its not the code
01:04.30harryvvbkw,what do you recomend?
01:04.33*** join/#asterisk remmo (~rem@smack.isp.net.au)
01:04.35bkw_dougheckawe do right now
01:04.35bkw_why?
01:04.45bkw_well 1. make digium fix it
01:04.51bkw_2 bitch and moan till they do.
01:04.54bkw_or
01:04.55dougheckaI want to know if I can rely on it for ALL my faxes...
01:04.56bkw_3
01:04.59bkw_get some X101p's
01:05.04dougheckaeveryonce in awhile I will get a messed up page
01:05.10harryvvthe 101 is a replacment for the 100?
01:05.13bkw_doughecka yes you will get some messed up pages from time to time
01:05.19dougheckabkw_: :I could never get x101 to recieve faxes ok
01:05.22bkw_but usually its the sending fax machine's fault
01:05.28bkw_I got mien to do it 100%
01:05.33SpaceBassmy x100p recieves faxes
01:05.35SarahEmmwhat's the difference between x100p and x101p?
01:05.35bkw_I setup our system to fax hammer it one night
01:05.40bkw_for 8 hours straight
01:05.41bkw_it worked
01:05.44bkw_100%
01:05.47harryvvbkw, in what way those faxes dont follow fax protocol standards?
01:05.53dougheckabkw_: cool
01:05.57bkw_have you read the fax standards?
01:06.01harryvvnope
01:06.02bkw_its worse than SIP
01:06.09hasslerx100p?
01:06.13dougheckahaha
01:06.22bkw_hassler, yes same thing
01:06.26*** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net)
01:06.35bkw_some fax machiens just boldly break the standard
01:06.38hasslerahhh, the one port FXO card.
01:06.40bkw_some just don't follow it much if any
01:06.43harryvvbkw, so what card did you use for the fax machine or was it machines ?
01:06.46dougheckago where no fax machine has gone before!
01:07.00bkw_harryvv, our hylafax system sending faxes to my x101p in my box here
01:07.03bkw_for 8 hours straight
01:07.08bkw_one fax every 3 min
01:07.12bkw_2 pages long
01:07.14hasslerI'd love to have the fax line detect fax or voice
01:07.15harryvvyes, i have read about hylafax
01:07.56hasslerdoes the x100p work the same otherwise as the tdm400p?
01:08.05harryvvno
01:08.36harryvvi need to get my ip500 completed. fun fun.
01:09.18bkw_its just an FXO
01:09.18*** join/#asterisk [illuminatus] (~illuminat@cpe-65-185-103-95.woh.res.rr.com)
01:09.47[illuminatus]any idea on why i can make outgoing calls but I can't receive incoming calls?
01:09.52hasslerbut would work the same as a tdm400 with a single fxo daughter?
01:09.55SpaceBassha!
01:10.06SpaceBasssshed into my * box... saw my wife call my cell before it rang
01:10.30Pete_LargoI would like to receive incoming faxes over a voip 'did' number...
01:11.19harryvv[illuminatus]: check your context
01:11.27dougheckaSpaceBass: didja kill the connection before it went through?
01:11.43SpaceBasslol
01:11.44SpaceBassshould have
01:11.46Chuji[illuminatus] : We're going to need a little more info
01:11.49*** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net)
01:12.01ChujiSpaceBass : One day I printed out a usage report for my wife out of the CDR tables and she wasn't amused
01:12.11SpaceBassshe doesnt like those eithers
01:12.13ChujiI told her, you talk on the phone too much, and here is proof
01:12.19SpaceBassand she claims not all sip channels ring
01:12.22*** join/#asterisk jerlique (jerlique@lnk2.adl.adsl.esc.net.au)
01:12.29SpaceBasswhen i call in
01:13.01[illuminatus]Chuji: such as?
01:13.01wunderkinSpaceBass, grandstream phones?
01:13.23SpaceBassone hitachi, one cisco, one siemens, one pos ata
01:13.26wunderkinoh
01:13.27Chuji[illuminatus] : Did it ever work? SIP? IAX? tdm?
01:14.12*** join/#asterisk fugitivo (~ajf@168.226.244.221)
01:14.39[illuminatus]incoming calls never worked. I just got outgoing calls to properly work. It's SIP
01:14.43[illuminatus]there's no NAT in betwee
01:14.47[illuminatus]between*
01:15.18Chuji[illuminatus] : Does the CLI give an error?
01:15.47[illuminatus]let me SSH in and look
01:15.56[illuminatus]http://pastebin.ca/15454 for config
01:17.30Chuji[illuminatus] : I'll need to see the Dial line
01:19.01[illuminatus]ok console is spitting out rap whe ppl call
01:19.21*** join/#asterisk dalaila (Santon@adsl-200-136.tricom.net)
01:19.42dalailaI have problem with Asteriskhome
01:19.45*** join/#asterisk LapTop006 (~laptop006@sparc006.chriskaine.com.au)
01:19.56[illuminatus]http://pastebin.ca/15455
01:20.52Chuji~asterisk@home
01:20.53jboti guess asterisk@home is http://asteriskathome.sourceforge.net/, or http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home
01:21.03*** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
01:21.34[illuminatus]Chuji: what do you mean by the dial line?
01:21.46*** join/#asterisk dos000 (~dos000@66.11.173.123)
01:23.12*** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com)
01:23.59Chuji[illuminatus] : That all looks good. Doesn't look like your provider is accepting the call
01:24.39[illuminatus]but it seems to be getting to our * box...
01:25.03[illuminatus]why do you think the provider isn't accepting the calls? (j/w)
01:25.32Chuji[illuminatus] : Take a look at this
01:25.33ChujiFound peer 'pstn'\
01:25.33Chujiit's using that peer for incoming
01:25.38Chujibut in that peer you have not defined a context
01:26.23[illuminatus]ok. so what should I do?
01:26.31Chujiput a context = incoming (or wherever your incoming is)
01:26.39Chujiin that sip peer entry
01:27.46ChujiHave you defined an incoming context in your dialplan?
01:27.46[illuminatus]let me look at it
01:27.46dalailaMy problem is with the user that asterisk create, this user doesn't autenticated by web
01:27.46SpaceBassdalaila web what? voicemail?
01:28.43[illuminatus]sorry i'm a little bit of a nub at this
01:28.51[illuminatus]i can't even find the dial plan now :(
01:29.10SpaceBass[illuminatus] is this a sip or pstn ?
01:29.13Chuji[illuminatus] : extensions.conf
01:29.19Chuji[illuminatus] : or
01:29.22[illuminatus]it's a SIP trunk to our provider
01:29.28Chuji'show dialplan' from the CLI
01:29.32SpaceBass[illuminatus] bv?
01:29.43[illuminatus]CentricVoice
01:29.47dalailaSpaceBass The user that Asterisk gave to me, Does authenticated
01:30.26dalailaSpaceBass the user that I put is admin
01:30.35SpaceBassdalaila don't follow... what are you trying to authenticate to?
01:31.12dalailaAsterisk Management Portal
01:31.13[illuminatus]pstn is not defined in extentions
01:31.18[illuminatus]only from-pstn is
01:31.31SpaceBass[illuminatus] using amp?
01:31.35[illuminatus]unless i just said something stupid
01:31.37[illuminatus]AMP is installed yes
01:31.45SpaceBassdalaila and what user do you want to authenticate?
01:31.51dalailaIt give me for default Admin
01:32.08SpaceBassdalaila and you want to create another user?
01:32.09dalailathe password for default is password
01:32.15Chuji[illuminatus] : as a termporary fix, you can put 'context = from-pstn' in sip.conf under [pstn]
01:32.31Chujisee if that gets you going
01:32.53SpaceBass[illuminatus] you need to look in /etc/asterisk/extensions_additional.conf or in amp for the context
01:33.10SpaceBass[illuminatus] ignore me and follow chuji thats a good idea
01:33.37SpaceBassChuji its easiest with *@h to treat sip trunks as pstn for inbound
01:33.50SpaceBassdalaila you can change that password if thats what you want
01:33.59dalailaSpaceBass I tried with another user but it came up with the same error
01:34.03Chujimore and more people are using it around here. I guess I should install it once to see what it looks like
01:34.10dalailaSpaceBass I change the password
01:34.11ChujiSeems unnecessary to me
01:34.24SpaceBassdalaila a OS user or an extension you created in asterisk?
01:34.35jontowugh.. installed a SCO OpenServer machine tonight.. *shudder*
01:34.38jontowwhat a nasty thing that is :)
01:34.45Chuji~sco
01:34.46jbotit has been said that sco is the new name for Caldera, and they're making right fools of themselves, or the antichrist, or supposed to sue linux users today, or a bunch of jerks
01:34.55SpaceBassdalaila you can create another unix user like maint, but you cannot use an extension you have created to log into AMP
01:35.02jontowyou got it, jbot.
01:35.25dalailaSpaceBass `How can I do this?
01:35.27SpaceBassChuji its slick for what it is, but it creates newbiews like me and [illuminatus]... but thats probabbly a good thing in the end
01:35.39SpaceBassdalaila create a linux user?
01:35.44SpaceBass#linuxhelp :)
01:36.02[illuminatus]<PROTECTED>
01:36.04dalailaSpaceBass I create one name dalila
01:36.08Chuji~man adduser
01:36.15dalailaSpaceBass longtime ago
01:36.43SpaceBassPete_Largo I counldnt get anything until 1.0 to work correctly
01:36.53SpaceBass*@H 1.0 that is
01:37.08Chuji[illuminatus] : 'show dialplan from-pstn' | pastebin.ca
01:37.10Pete_LargoI'd rather use the CLI anyway
01:37.27SpaceBasscan use cli with *@h
01:37.41SpaceBass*@H enabled me to learn enough to be able to manually create dial plans, etc
01:37.44Pete_Largoshould have read 'CLI version'
01:38.40Qwellomg...my friend just explained the phone/network situation at his work...
01:38.50SpaceBassi read the handbook several times and was getting the hang of it.. I think I *could* have done it... what thew me was all the agi stuff
01:38.52QwellThey're sitting on about 20 phone lines (no, they don't have a T1), and using 56k dialup
01:39.11DaminMorning everyone..
01:39.17SpaceBasscluecon if that existed I'd be a happy man!
01:39.22dalailaSpaceBass my problem is not to create a user, my problem is thAt i can not enter with the user and the password that  I asigned
01:39.40clueconQwell: is this a company located in the US?
01:39.48Qwellcluecon: yes!
01:39.54Qwellmakes me sad
01:40.09QwellI'm gonna go there and regulate...smack the IT guy around a bit
01:40.32harryvvQwell, no t-1? thats dumb with that many lines.
01:40.33SpaceBassdalaila I'm not sure how amp is set up... may need to create the user in /var/www/html/pannel
01:40.38Sedoroxwhat does the place do?
01:40.40Qwellharryvv: exactly my point
01:40.40SpaceBassi haven't looked closely
01:40.46QwellSedorox: motorcycle dealership
01:40.52SpaceBassdalaila check #amportal those folks could tell you quickly
01:40.59harryvvI wonder how much thay are being charged with those 20 lines
01:41.00Qwelltwo locations, actually
01:41.06SedoroxI could see the lines... but I dun think they would really need very high speed internet...
01:41.07[illuminatus]http://pastebin.ca/15461
01:41.09Qwellharryvv: alot more then a T1 would cost, thats for sure
01:41.16harryvvwell, at least thay could get a fractional t-1
01:41.18[illuminatus]the other two includes don't eit
01:41.20QwellSedorox: they have a second location...
01:41.27Sedoroxah
01:41.27Qwelldata transferred back and forth all day
01:41.32Sedoroxdammn
01:41.35clueconQwell: If i'm doing the math right...a standard phone line is at least $20 bucks.  That is at least $400 a month.  Your average T1 is gonna run about that.
01:41.37dalaila\j #amportal
01:41.39Qwellover a single 56k...heh
01:41.49Qwellcluecon: business lines are alot more, I'm sure
01:41.50harryvvQwell, whats the location there and how much for a t-1 can that be combined voice/data?
01:41.53SpaceBass[illuminatus] there is no s in from-pstn
01:41.55Sedoroxupgrade time!
01:41.56Sedorox:p
01:42.02SpaceBassi think thats your problem
01:42.06Qwellharryvv: not sure about pricing out there, Fort Worth, IN
01:42.18*** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net)
01:42.19Pete_LargoIN = Indiana?
01:42.22Chuji[illuminatus] : Well, that is your problem. You need to define what happens when an incoming call hits your dialplan
01:42.23Qwellyeah
01:42.25milkyflavahello
01:42.28Chuji[illuminatus] : What should it do?
01:42.38harryvvcluecon: our t-1 here in bc cost anywhere from $750-1,100 CDN here
01:42.42[illuminatus]it is defined though. it goes to extention 502...
01:42.44SpaceBassChuji there is a context he can goto with amp/aah
01:42.46SpaceBassleme look
01:42.47[illuminatus]it works when i dial 7777
01:43.04Pete_LargoOff the top of my head I can't think of any CLECs that service IN
01:43.10*** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca)
01:43.17milkyflavastupid question - can I set up asterisk on my local network and make calls to computers in my local network without having to sign up for a service?
01:43.20harryvvQwll, thats a pretty good price :)
01:43.29Qwellharryvv: for what?
01:43.36harryvvt1/pri
01:43.39SedoroxI'm helping the one IT guy... the new school they just built here is having all cisco equipment put in, including VoIP phones in every room...
01:43.43Qwellwhats a good price?
01:43.44Sedoroxthe 7960's are pretty nice...
01:43.59harryvvI wonder how vonage does it by terminating there pstn in vaginia.
01:44.14SpaceBassChuji and [illuminatus] i understand the problem and fix, but dont know enough to articulate it... when you hit 777 its going to from-pstn which includes time-check and thus from-pstn-reghours-nofax which then has the dial/sip502,FOO,BAR
01:44.28jontowmilkyflava; yes.
01:44.37milkyflavathanks
01:44.56Pete_Largovonage is probably using carrier ld termination
01:45.01SpaceBass[illuminatus] instead of goto(from-pstn, s,1)...
01:45.19milkyflavabut if I want to make calls going outside to PSTN phones I would need to sign up, correct?
01:45.24SpaceBass[illuminatus] make it goto(from-pstn-reghours-nofax,s,1)
01:45.37SpaceBassmike-ff signup for what?
01:45.50[illuminatus]SpaceBass: i would if I knew how :(
01:46.02clueconharryvv: we have a local T1 that is costing us about $600 USD.  We are about to get some additional T1s put in that are just a per minute charge and will soon also have a DS3 circuit which has a cost of about 135 USD per T1 if we have the whole thing active.
01:46.03*** join/#asterisk dyl0n (~mhappe@p548B28C8.dip0.t-ipconnect.de)
01:46.15SpaceBass[illuminatus] are you comfortable editiing the config files?
01:46.38[illuminatus]sure
01:46.44harryvvcluecon: what services do you sell
01:46.48milkyflavalike with voipjet or broadvoic services
01:46.57SpaceBass[illuminatus] that pastebin you pasted earlier ... which .conf was it from?
01:47.14Chuji135 per T1? in US? you should be able to do better than that w/ a DS3
01:47.27[illuminatus]extentions.conf and then i included what it included from extentions_additional.conf
01:47.29clueconharryvv: we don't sell, we use.  it's a collections company and we run thru quite a few calls.
01:47.37harryvvahhhh
01:47.43SpaceBassmilkyflava you can sign up with one to make calls from 'the net' to pstn phones   or you can use a FXO which allows you to plug a PSTN into * directly (and many other options including t1, etc)
01:47.48harryvvnow thats a company i can hit on for biz :)
01:47.50Pete_LargoChuji, the loop cost is probably raising the price
01:48.10SpaceBass[illuminatus] you have a line that is exten => 12143294838,1,Goto(from-pstn,s,1)
01:48.22harryvvcluecon who implemented the entire project?
01:48.32SpaceBasschange it to goto(from-pstn-reghours-nofax, s,1)
01:48.35clueconharryvv: define implement.
01:48.39milkyflavathats what was confusing me, so all I really need is a tmd400p 11b card from digium then I dont need to sign up with anyone but use my existing PSTN line, correct?
01:48.45SpaceBassand remove the include = > ext-did-custom
01:48.48SpaceBassdont think you need that
01:48.52harryvvwho started and installed and configured all of it.
01:49.08SpaceBassmilkyflava yep, or a x100p from ebay for $10.00usd
01:49.18Chuji~x100p
01:49.19jboti heard x100p is an obsolete card.  you don't want to bother trying to make it (or any of the "digium compatible" clones work.  Get a TDM01P, you will save your sanity.
01:49.25clueconharryvv: i am.  we are about to replace our current dialer with * boxes.
01:49.40SpaceBassmilkyflava if you are just playing around you can get a $5/month account from BV and cancel anytime... some people have reported issues with them though
01:50.07harryvvcluecon how many channels is it going to handle? Im asuming the load is distributed among several boxes.
01:50.20milkyflavaSpaceBass thanks, thats exactly what I am doing.
01:50.30SpaceBasswhats the general idea b/h loadbalancing with *
01:50.36SpaceBasshow do you do it... put some channels on each box?
01:50.46*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
01:50.56Chuji~dundi
01:50.57jbotrumour has it, dundi is http://www.dundi.com
01:51.00SpaceBassmilkyflava enjoy... and of note: BV allows more than one incoming call... but they dont tell you that :)
01:51.07clueconharryvv: we are limiting it to 1 quad t card per box (max 96 zap channels and i think 30 zip per box).
01:51.23harryvvbig call center
01:51.24harryvv:)
01:51.44[illuminatus]OMGWTFBBQ?
01:51.52[illuminatus]I can't test it from my PSTN line
01:51.56harryvvbefore your company did this, were thay given quotes by other commerical voip sla providers?
01:51.59SpaceBassthis cheesy hotel room has a f'ing hot tub in the center of it... HA!
01:52.02milkyflavaSpaceBass, excellent, thanks for the tip
01:52.08[illuminatus]because it when I dial the country code it says I don't need and and when I don't dial the country code it says I do need it
01:52.28SpaceBass[illuminatus] sounds like a dial pattern problem
01:52.31clueconinitial plan was to start out with 8 ts active but we just heard that sbc actually got its approval to come onto the property so i am trying to talk the boss into starting it up with 16 active.
01:52.43[illuminatus]SpaceBass: on my regular phone line?
01:52.58harryvvsbc come onto the property and do what?
01:53.00harryvv:)
01:53.02SpaceBassmilkyflava one last one... google geek gazette asterisk@home
01:53.02clueconharryvv: we use pure pri at the moment.  it is better for us (due to the cost of data bandwidth) to use pri.
01:53.07milkyflavaone more question - Should I install using source or *@home
01:53.14clueconharryvv: run fiber to support the ds3.
01:53.19SpaceBassmilkyflava good tutorial for setting up BV with * (if you can find the article, i never can)
01:53.33milkyflavaSpaceBass I think that answered it. :)
01:53.38Pete_Largoit's on the broadvoice site in the support section
01:53.42clueconmilkyflava: use CVS-HEAD.
01:53.54SpaceBassPete_Largo that doesnt work... at least i've never gotten it to successfully
01:53.58cluecon*@H is not what you want.
01:54.02harryvvcluecon yea thats nifty. Any ideas how vonage can sell service here in bc canada when there "I suspect" pstn termination is in vaginia?
01:54.07Pete_Largohttp://www.broadvoice.com/support_install_asterisk.html
01:54.12SpaceBassha ha ha! I'm spreading the *@H plague :)
01:54.13Pete_Largoit's what I used, and it works fine
01:54.17milkyflavaah, ok, thanks cluecon
01:54.20Sedorox*@H sucks.....
01:54.27Sedorox:p
01:54.39Sedoroxlol
01:54.44SpaceBass<-- *@h lepper
01:54.56Sedorox:p
01:55.04Pete_Largolol lepper
01:55.23SedoroxI have it.. but only to use parts of the config from (like setting it up to recieve a fax
01:55.35*** join/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net)
01:55.41clueconharryvv: simple.  pstn termination is just required to get back to the traditional land lines.  if you are big enough (vonage is) you can get great deals with the land line owners (sbc, qwest, sprint, etc.) that allow you to make a profit off of the voip service.
01:56.07SpaceBassSedorox thats my take... use it for some stuff like fax and time check
01:56.14SpaceBassand wake up call and weather
01:56.16Sedoroxyea
01:56.27Sedoroxstuff you don't feel like making thats already done :p
01:56.33SpaceBassexactly
01:57.08SpaceBassbbl
01:57.10Sedoroxactually... I believe I run release
01:57.10Sedoroxlol
01:57.30Sedoroxlol
01:57.44Sedoroxnobody was insisting on it
01:57.47Sedoroxut oh
01:58.43milkyflavaSo is there a one doc that is preferred over others that is specific to on distro?
01:59.06DarthCluevoip-info.org has some pretty straightforward docs.
01:59.08milkyflavaor just follow the docs at the * site
01:59.29*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
02:00.21milkyflavaok, that one I have bookmarked, how about a distro or just whatever I'm used too or does it even matter
02:00.25DarthCluevoip-info.org is slightly better.
02:00.55DarthClueI use FC because it is what I am familiar with.  Some distros have better luck than others.  What is your preferred distro?
02:01.03milkyflavaFC
02:01.11SpaceBass|AWAYfc
02:01.20fugitivoi don't have a preferred distro, i use the best distro, gentoo
02:01.24DarthCluethen use FC.  You will get alot of flac about it, but use it if you like it.
02:01.26SedoroxGentoo
02:01.28DarthClueGENTOO SUCKS!!!!
02:01.35fugitivoDarthClue--
02:01.38milkyflavalol
02:01.42Sedoroxlol
02:01.49Pete_LargoI'm using FC3 and it works fine
02:01.49darwin_35Slack or Debian
02:01.55Pete_Largofor me
02:01.57Sedoroxeach to his own
02:02.08darwin_35Feudoria sucks
02:02.11Pete_Largojust remember to compile zaptel with make linux26
02:02.14harryvvslack has almost been free of crashes...on its own.
02:02.15fugitivofc is for l00sers
02:02.19harryvverr i mean fc3
02:02.21fugitivolol
02:02.35milkyflavawindows 3.1 is rock solid
02:02.38Pete_Largoharryvv, go take a long walk off a short pier
02:02.43[illuminatus]grrr
02:02.47jontowwhy .. i mean, honestly.. why
02:02.48harryvvfug, fc3 runs my phones just phone
02:02.51[illuminatus]still not working
02:02.51[illuminatus]http://pastebin.ca/15467
02:02.52fugitivoharryvv: crashes? only if you use windows
02:02.55darwin_35Slack10 rocks 10.1 you have to replace gcc
02:02.59sivanadoes linux have a .zip unzip command?
02:03.01jontow"WINDOWS XP SUCKS, WINDOWS 2000 ALL THE WAY" <-- seriously, do you guys see what you're saying?
02:03.10jontowshut up and use what works best for you
02:03.14Sedorox[22:02] <Pete_Largo> just remember to compile zaptel with make linux26
02:03.15SedoroxI haven't had to
02:03.19darwin_35FreeBSD
02:03.21fugitivoharryvv: i'm just kidding, i like linux in general
02:03.27DarthClueBut windows XP does suck.
02:03.31milkyflavalove the BSD
02:03.32jontow;)
02:03.48darwin_35Windows is Great as a end user os but not a server
02:03.49fugitivoyeah, openbsd is the winner
02:04.08harryvvlinux is okay..i made the mistake of downloading 64 bit for fc3 and then use it as a work station and server. besides, who needs more memory then what 32 bit can provide.
02:04.09*** join/#asterisk |Ladybug| (~ladybug@219.95.78.33)
02:04.13milkyflavaI read what the openBSD developer had to say about linux
02:04.14fugitivodarwin_35: windows as end user is a pain in the ass
02:04.16|Ladybug|hi all
02:04.40fugitivoharryvv: i use gentoo 64bit in my laptop
02:04.47jontowpersonally, my first 2 choices are always FreeBSD and NetBSD .. but with asterisk; i tend to not think that.. they still have some serious issues to work through (most specifically with random zaptel hardware); so i use gentoo, and i've gladly used debian though i just wasn't comfortable with it.. i now am also using beehive linux on a very weak machine and have a lot more call volume to push through the machine
02:04.50darwin_35weI agree but was trying to be nice
02:05.04harryvvfug, why?
02:05.07fugitivojontow++
02:05.09[illuminatus]anyone know why i can't receive calls?
02:05.10harryvvwhy why
02:05.15*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
02:05.15[illuminatus]everything seems to be setup correctly
02:05.23darwin_35Freebsd and Zaptel are stable
02:05.28[illuminatus]http://pastebin.ca/15467 <-- what happens when i try to make a call
02:05.39|Ladybug|i need help with asterisk... ne1 can help?
02:05.39jontowif i had the time/energy/knowledge, and/or someone was paying me to learn.. i'd gladly fix all of the bugs i could find with freebsd and netbsd using zaptel hardware.
02:05.40fugitivoharryvv: why what?
02:05.53darwin_35I hope to have Festival 1.95 working soon
02:05.54jontowhowever.. none of the above is true, so i'm content with gentoo until most of those conditions can be met
02:06.08jontowdarwin_35; using a T100P under FreeBSD?
02:06.22|Ladybug|i wrote a program using vc++ to interact with asterisk
02:06.29fugitivogentoo is the easiest distribution to mantain
02:06.31|Ladybug|i am using a softphone called sjphone
02:06.35SedoroxI've had ok luck with zaptel and fbsd
02:06.35jontowagreed..
02:06.39milkyflavaThanks all for the answers to my questions, I'm gonna get some CVS-HEAD and start learning
02:06.43Sedoroxbut then it started locking the kernel :(
02:07.08jontow"emerge --sync ; emerge --deep update world" rocks my --deep world..
02:07.08|Ladybug|but everytime i transfer call... my program crashed
02:07.08jontow;)
02:07.09darwin_35not yet . but I have a tdm400
02:07.09|Ladybug|can ne1 help?
02:07.13fugitivojontow: don't forget emerge search, it's great
02:07.18jontow:)
02:07.23darwin_35man I hate dialup
02:07.23jontowi use locate for a lot of my searching
02:07.23*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
02:07.34jontowbut im not too familiar with portage yet.. still learning :)
02:07.37darwin_35I want my highspeed back
02:07.45jontowdarwin_35; me too.  mine burnt.
02:07.54fugitivojontow: just emerge search whateveryouwant
02:08.04darwin_35I only use the ports for base install but build * from cvs head
02:08.06jontownow i have 15 machines on a 26.4kbps dialup
02:08.12|Ladybug|ello???
02:08.25DarthClueLadybug: what is the question?
02:08.47fugitivo|Ladybug|: softphones for linux are not what I call "stable"
02:08.56|Ladybug|oh
02:09.02|Ladybug|i wrote a program using vc++ to interact with asterisk
02:09.04darwin_35there is one from Novel
02:09.06|Ladybug|i am using a softphone called sjphone
02:09.06jontowladybug; part of the issue here is that you asked about VC++ and asterisk + program crashes when calls are transferred; sounds like a debugging issue.. and .... under windows?
02:09.09darwin_35I forget the name
02:09.16|Ladybug|yes...
02:09.18darwin_35SFLPHone
02:09.21|Ladybug|it is running under windows
02:09.29fugitivodarwin_35: novell?
02:09.32jontowok, so you're already outside of the comfort zone for a lot of people here :)
02:09.42darwin_35yes
02:09.46|Ladybug|i know... i never like microsoft
02:09.49darwin_35Novell
02:09.51fugitivodarwin_35: for linux?
02:09.54|Ladybug|but my co's licience is using microsoft
02:09.55|Ladybug|=(
02:10.09|Ladybug|but the asterisk server is running under linux
02:10.20jontowi understand the need for it.. in a lot of ways, as i'm stuck using a lot of it here (and maintaining it.. ugh.  frontpage+coldfusion+win2k+IIS5+ms-sql7 == UGGGGGGHHHH)
02:10.37fugitivojontow: man, i'm sorry
02:10.39jontowtouch any one of the 5 and the rest just come crashing down
02:10.57jontowi've happily deemed it "house of cards syndrome", and laugh whenever possible
02:11.17jontowthe problem is .. i rarely get time to update/maintain my unix machines because.. honestly, they never need attention
02:11.19fugitivolol
02:11.43jontowthe win2k machines always need the attention.. whining crybabies they are.. "someone poked me with a portscan, IM TELLING MOM" etc.
02:11.52jontowstupid crap.
02:11.55darwin_35brb
02:11.57jontow;)
02:11.59|Ladybug|just asking... when i transfer/redirect the calls... what shud be the response of the asterisk?
02:12.08jontowdarthclue; exactly.
02:12.38fugitivo|Ladybug|: what kind of application did you develop with vc++?
02:12.42jontowladybug; depends.. how're you transferring? # or client-initiated or manager-api initiated, etc?
02:12.53|Ladybug|manager api
02:13.03|Ladybug|i am developing a dll... then another MFC program to run it
02:13.05jontowhave you successfully transferred one by hand via the manager api?
02:13.10DarthCluejontow: i put a linux box in as a web / email / database server when i started this job last september.  i haven't had to do anything to it since then.  i have a windows server that i have to 'fix' everyday.
02:13.13jontow(and watched the response?)
02:13.17|Ladybug|yes
02:13.24|Ladybug|the originate button r perfectly ok...
02:13.31|Ladybug|just transfering is a prob
02:13.37|Ladybug|i have 2 ipphone here
02:13.41fugitivo|Ladybug|: did you try with another softphone?
02:13.46|Ladybug|using a softphone called sjphone
02:14.00jontowtry diax2, x-lite, or something else and see if you can replicate the situation
02:14.02fugitivo|Ladybug|: try with another softphone, xlite is free
02:14.06|Ladybug|used anoter phone called sumthing like Bellphone
02:14.08jontowyou may be ahead to find out that its a bug in sjphone..
02:14.11|Ladybug|oh... i m downloading now
02:14.49|Ladybug|try diax2...
02:14.57|Ladybug|it doesnt seem to like my pc at all
02:14.59darwin_35man this exten.conf I am working on is a killer
02:15.06jontowi'd so like to quit this job and get something where i can spend a little more time troubleshooting/developing/maintaining/supporting asterisk and related applications
02:15.18darwin_35has anyone here mapped all th *XX nmbrs
02:15.23fugitivo|Ladybug|: xlite seems stable
02:15.34DarthCluejontow: come to cluecon.
02:15.37jontowas is i get slightly into a project with * and get pulled to fix a windows server thats on fire
02:15.43Sedoroxexten => *XX.,1,YayItWorks :p
02:15.47Sedoroxops
02:15.50jontowdarthclue; and who'll throw my ticket?
02:15.51Sedoroxexten => _*XX.,1,YayItWorks :p
02:15.51|Ladybug|is the x-lite using the SIP too?
02:15.52Sedoroxthere.. done
02:15.53Sedorox:p
02:15.53jontow:/
02:16.05DarthCluejontow: plane ticket? where you located at?
02:16.06fugitivo|Ladybug|: yes
02:16.07jontow(and my gf's .. since we never leave home without eachother.. long story)
02:16.10jontowupstate NY
02:16.11darwin_35no I melike *10 and so on
02:16.24fugitivoi want a ticket too, i live in argentina
02:16.29*** part/#asterisk brian13 (~user@c-24-98-71-208.hsd1.ga.comcast.net)
02:16.32jontow;)
02:16.38darwin_35I have a few of them but there are alot more
02:16.40Sedoroxdarwin_35: ouch.. your seperating them?
02:16.47jontow"yeah.. while you're handing out tickets.. can i have 4?"
02:16.49jontow:D
02:16.54NuggetI miss all the travel I used to do, but I don't miss it too.  it's nice to be able to make plans for more than a week or two in advance.
02:17.22Nuggetbut I miss globetrotting.
02:17.32|Ladybug|ok.. thanks
02:17.39harryvvnugget where and for what reason did you tracel
02:17.43harryvvtravel
02:18.08NuggetI used to fly to our customers and teach them how to use our software.  but I left the company and now I work for one of the customers now
02:18.25harryvvI remember being in a airforce base on leave taking millitary hops for free back home and saw a flight out to hawaii for 10 dollars. that was tempting.
02:18.28fugitivoi don't miss travel at all, i had a bad experience with a plane, and now i have trauma to flying
02:18.36NuggetUK, france, switzerland, australia, and japan.  I flew about 100,000 miles a year for three years.
02:18.46|Ladybug|internet is slow here/... =(
02:18.50Nuggetnow I don't leave my house.  it was quite a culture shock
02:18.51darwin_35http://pastebin.ca/15468 this is what I mean
02:19.09harryvvnugget yea I bet :)
02:19.14jontowsome of our customers pay way better than this company
02:19.18harryvvlady, you f?
02:19.26|Ladybug|yes...
02:19.26jontowkinda makes me sad because many of them are far less skilled and yet still are over double what i get paid
02:19.36|Ladybug|how can a male named themselves as Lady?
02:19.47harryvvanyone can mask anyone on irc :)
02:19.49fugitivojontow: it's always like that
02:20.03NuggetI miss getting to hang out in paris or tokyo on someone else's dime.  but I'm enjoying being able to rediscover what "weekend" means and being more involved locally
02:20.05jontowladybug; san francisco?
02:20.31harryvvnugget yea :) how about doing the occational voip install out of state?
02:20.46darwin_35I found the list on the net of the mapped *XX exten I am just mappping them in *
02:20.49jontowor for that matter any major metropolis? :) i hear there are lots of guys calling themselves lady under the right circumstances..
02:21.02|Ladybug|nope... i m from malaysia
02:21.25harryvvlady was Salam?
02:21.49|Ladybug|Salam?
02:21.54|Ladybug|i am a chinese lady
02:22.05harryvvmandarin or cantonese?
02:22.05Nuggetflying to cleveland just doesn't have the same appeal as flying to sydney.  :)
02:22.14*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
02:22.24harryvvNugget, what did you do for that company?
02:22.31darwin_35my pet project is to map them all in *
02:22.35Nuggetprofessional services.
02:22.39darwin_35for all users
02:22.40|Ladybug|mandarin
02:22.52NuggetI trained people to use the product and I did app development.
02:22.55Nugget(for ud.com)
02:22.55harryvvLady, we have a huge chinese population here in vancouver bc.
02:22.57fugitivoi want to learn mandarin
02:23.13harryvvfug, move to vancouver and learn it :)
02:23.54SpaceBass|AWAYin the year of darkness 2029?
02:24.05fugitivoi know it's impossible to learn to speak it correctly
02:24.16harryvvor 2007
02:24.36jontowdarthclue; dude.. you don't want to know that.. :)
02:24.39DarthClueno.  The
02:24.52harryvvI think both languages have the most difficulty because the way the mouth and tounge muscles work.
02:24.55darwin_35if anyone can help with this please do
02:24.57DarthCluejontow: too late, i already know.
02:25.02|Ladybug|just installing xlite
02:25.03jontowthats too bad :/
02:25.07|Ladybug|how to setup? =(
02:25.10DarthCluedarwin: help with what?
02:25.23NuggetI find japanese easy, but french is totally obtuse to me.  french has sounds that my ear just can't hear.
02:25.24harryvvlady, first of all do you have a firewall?
02:25.30Nuggetgerman is practically english
02:25.34|Ladybug|nope
02:25.35fugitivomandarin is difficult because the same sound, with another tone, means two different things
02:25.37jontowladybug; play with it for a bit.. it happens easily after a few minutes of changing settings :)
02:25.44|Ladybug|our co dun have firewall
02:25.45SpaceBass|AWAYyeah the intonations are crazy
02:25.53|Ladybug|ok... i am going to the website n read a bit
02:26.07|Ladybug|will disturb u guys again if i dun understand
02:26.13jontow:)
02:26.20fugitivojapanese has easy sounds
02:26.24Nuggetyeah
02:26.24|Ladybug|sorry... just started to work.. so a lot of thing i dun understand
02:26.45Nuggetthe hardest thing with japanese is fighting the tendancy to inflect.
02:26.53harryvvI say to my wife "ya lublu teba darinaga"
02:27.23harryvvWhich in russian means "I love you much precios"
02:27.31harryvvprecious
02:27.41SpaceBass|AWAYi want to learn russian and japanese next...
02:27.43SpaceBass|AWAYon my list
02:27.49harryvvrusisan is on my list
02:28.05jontowthat ain't no frodo
02:28.05harryvvAs soon as I get my hamradio station up...then i can start learning other languages.
02:28.10*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
02:28.24darwin_35night
02:28.26*** part/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net)
02:28.29jontowbut anyway --- to do the background research via IRC once again.. anyone played with SMDI and asterisk?
02:28.40harryvvbtw, voip took off on hamradio a few years ago. www.irlp.org
02:28.42jontowspecifically.. the Sun Netra-based Coppercom soft-switches?
02:28.43DarthCluejontow: define SMDI.
02:29.02jontow~SMDI
02:29.03jbotSMDI serial and TCP session protocol as APE library. URL: http://www.voxilla.org/projects/projsmdi.html
02:29.03jontow(?)
02:29.13DarthCluenope.
02:29.16jontownow .. admittedly, APE libraries mean absolutely nothing to me..
02:29.31jontowbut ... i did do a quick rewrite in C vs. C++
02:29.58jontowand it still didn't help, i guess the real issue is.. the serial<-->ethernet converter that the coppercom softswitches use to pass SMDI .. just wanted to know if anyone had seen one and/or made sense of it
02:30.10jontowi suspect not.. pretty new stuff as far as telephony goes [on the market]
02:31.33jontowalternately.. anyone know of a cheap-yet-effective PCI-X board that gives one regular RS232 serial port[s] ?
02:31.41jontow(that works under linux...)
02:32.18*** join/#asterisk Derkommissar (~total@66.64.215.6.nw.nuvox.net)
02:32.34Derkommissaranyone has any cool asterisk wallpapers :-) like a 3d asterisk ?
02:33.17*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
02:33.25fugitivoit's time to do asterisk hats and tshirts
02:33.40*** join/#asterisk tengulre (~tengulre@61.185.238.166)
02:34.08Mavviesomebody here with experience of running asterisk on a sparc with linux on it?
02:34.23shidoit has been done
02:34.26shidoit can be done
02:34.32Mavviethat's a good start!
02:34.34jontowhttp://www.softio.com/ic0607kb.htm
02:34.35Mavviethanks.
02:34.37jontowguess that fits my needs
02:34.37jontow:)
02:35.13jontow(weeee)
02:35.23jontowi do'nt even know if that damned thing boots anymore [once again a fire victim]
02:35.44MikeJ[Laptop]shido, did you get your unicall stuff working?
02:35.53jontowi wonder if linux even supports sparc32 anymore
02:36.11jontowif not; netbsd works great on that box.. haven't tried 2.0.2 though
02:36.20jontowi think i should find a new HDD for it and bring it to work to install upon
02:40.10*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
02:40.57fugitivoanyone is using broadvoice?
02:42.35Sedoroxfugitivo: lots
02:42.43fugitivois it good?
02:42.55SedoroxI don't, I know others that do
02:42.58Derkommissaranyone have a cool asterisk wallpaper ? like a 3d asterisk or something ?
02:47.51SwKdoes david of res_php hang out in here?
02:48.07SwKor has anyone gotten it to compile?
02:48.09jontowswk; AgiNamu, i believe, wrote an alternative, i believe.
02:48.14jontowlemme see if i can dig it up :)
02:48.40SwKI just saw that on sineapps
02:48.43jontowhttp://jontow.hijacked.us/~jontow/asterisk_php.tgz
02:48.57jontowhe pointed the channel at it for feedbakc
02:49.05SwKwerd
02:49.06jontowif you have a good use for it, i urge you to do so; it'll only help
02:49.06jontow:)
02:49.18jontowthats the copy i grabbed from his site; which i fail to remember now
02:49.47jontowDavid Eder .. that his name?
02:50.12SwKyeah
02:50.18SwKI have a copy from his website
02:50.33SwKI'm having compile issues
02:51.04jontowaha
02:54.11fugitivounlimited calls to argentina
02:54.16fugitivoonly for 25 dollars
02:54.43fugitivoi pay more for my pots line, and it's not unlimited
02:55.46wunderkinso what would a  res_php accomplish exactly?
02:55.49*** join/#asterisk _0_0_ ([U2FsdGVkX@96837297623a2575.session.tor)
02:57.57*** join/#asterisk Ahmuck (~chatzilla@24.225.23.102)
02:58.14NewSoleso where is twisted sister
02:58.56SwKwunderkin: res_php is to asterisk as mod_php is to apache
02:58.58*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
02:59.09tengulredoes asterisk support FreeBSD?
02:59.25wunderkinyeah, i dont know how that works either
02:59.29wunderkintengulre, yeah
02:59.49tengulrewunderkin,thanks :), I'm beginner!
02:59.52SwKtengulre: works fine on FBSD but dont bet on hardware support thats worth a damn
03:00.16tengulreSwK,thank u!
03:00.58wunderkini guess somehow it can do the php within asterisk instead of having to spawn a php process for each thing.. or something
03:01.21tengulreI agree u!
03:01.42*** join/#asterisk Junk-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
03:02.43tengulreI like asterisk, but the asterisk can't support more hardware!
03:04.44fugitivoanyone using broadvoice?
03:06.01tengulrefugitivo, yes im
03:06.09fugitivotengulre: what plan?
03:07.30tengulrewhat kind of broad are u use?
03:07.52fugitivoi don't use it, i want to try it
03:08.31*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
03:08.50jontowthere an easy way to hangup a channel? softhangup that is?  ie. not destroying the channel or anything like that?
03:08.58jontowah.. stupid question
03:09.16jontow:)
03:09.23tengulreplaint!!:)
03:09.25*** join/#asterisk Qorky (~Pooa@dip-202-72-131-243.wa.westnet.com.au)
03:09.49Qorkycan anyone here help me with isdn and asterisk. I cant get chan_capi to load
03:09.50Qorkyplease
03:11.15shido4th quarter
03:15.37jontowgreeaaaat; upstream dialplan at the telco switch is failing so i can't call from one of my PBXes to the other
03:15.53*** join/#asterisk freat (~freat@h-69-3-229-184.chcgilgm.covad.net)
03:16.49*** join/#asterisk da_monumental_1 (~da_monume@rrcs-24-172-102-197.midsouth.biz.rr.com)
03:18.26jontowwtf
03:18.33jontowZap/23-1 answered IAX2/intnic@intnic/1
03:18.40jontow(am i missing something here or is this just messed up?)
03:19.01jontowusing a zaptel channel for a 100% voip call?
03:19.45NuxiHmmm... looks like people been talking about me when I was gone...
03:19.47jontowactually .. 2 Zap/N channels for that call
03:19.50jontowthats messssssed up
03:20.36NuxiSwK, jontow, compile problems with res_php?
03:20.45jontowaha !5!#^*  it wasn't AgiNamu at all
03:20.53jontowi knew i'd confused someone :(  [and figured it was myself]
03:21.22jontowSwK was having the issues
03:21.37jontowi was just trying to offer the absolutely vague insight that i had
03:21.38jontow:)
03:22.02Nuxicurrent code is at http://eder.us/projects/asterisk_php/   needs a few tweaks to work with cvs-head.
03:24.17*** join/#asterisk Ahmuck (~chatzilla@24.225.23.102)
03:24.33Nuxijontow, there's also res_bf at  http://pastebin.ca/15414 for your amusement.  (brainfu**)
03:24.42jontowahahahahahah
03:24.53jontowbrilliant; yet so damned twisted :)
03:24.59Hogiemy head hurts, I was at a sausage fest all day
03:25.29*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
03:35.08*** part/#asterisk Cresl1n (~matt@216.207.245.23)
03:35.43SwKok... NOTE TO SELF: **DO NOT** walk of the front door with a blade in your hand when cops are rolling up on you
03:36.01Sedoroxlol
03:36.38jontowgood call
03:36.56NuxiSwK, did you ever get res_php to compile?
03:37.09Corydon76-home~res_brainfuck
03:37.11jbotmethinks res_brainfuck is at http://www.tubgirl.com/
03:37.19SwKNuxi: nopw... but I'm trying against head
03:37.35SwKSedorox: that shit just happened to me at my house
03:37.50Sedorox:(
03:38.20Sedoroxwhy were the cops there.. and why did you happen to have a knife... wait.. I dun wanna know
03:38.28jontowhttp://pastebin.ca/15414   <-- corydon76; go there instead.
03:38.40SwKso a kid about 15yrs old comes beating on my door... doods holding my female cousin nextdoor and is possibly raping her....
03:38.50SwKhilarity ensues
03:38.58Nuxistuff moves around in head.  http://pastebin.ca/15473 has the correct include paths.
03:39.23Sedorox:/
03:39.57Corydon76-homejontow: what, no res_ook?  ;-)
03:39.59*** join/#asterisk ptiggerdine (~ptiggerdi@c210-49-98-194.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM]
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03:39.59*** join/#asterisk newl (~newlook@203-59-112-225.dyn.iinet.net.au) [NETSPLIT VICTIM]
03:39.59*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM]
03:41.34*** join/#asterisk santiago (~santiago@63.245.86.198)
03:41.55*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
03:53.40*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
03:53.40*** topic/#asterisk is Asterisk: The Open Source PBX || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com
03:53.43NewSolesetting up peerin and want to see if rings phone
03:53.48*** join/#asterisk roywish (~roy_wish@202.155.41.20)
03:53.59NewSoleif I try call inside net wit wont go out
03:54.38DarthCluejontow: wikipedia may have more content but it isn't asterisk related.  I am working on a creating a copy of voip-info.org that would be more scalable and less likely to die.
03:54.38roywishhi all
03:55.07postelDarthClue: a mirror or from scratch?
03:55.07roywishanyone have succesfull install oh323
03:55.13roywishplease help me
03:55.33DarthCluepostel: combination.
03:55.35jontowcool.
03:55.57sivanaexit
03:56.00NewSoleDarthClue... can you try it for me
03:56.01sivanaack
03:56.07DarthCluewe need docs that are more specific that we can point newbies to so that we don't keep answering the same questions over and over again.
03:56.34roywishanyone have succesfull install oh323
03:56.35roywishplease help me
03:56.55postelDarthClue: why dont you work a bit on the asterisk handbook? its been left outdated for some time now
03:57.16DarthClueNewSole: not sure that i can.  my box isn't exactly in a position to be tweaked at the moment.
03:57.49NewSoleok...
03:57.58*** join/#asterisk mogorman (~mogorman@207.111.174.1)
03:58.23roywishanyone have succesfull install oh323
03:58.24roywishplease help me
03:58.55DarthCluepostel: asterisk handbook does need work.  I have looked at that and passed it off as being out of date when compared to the wiki.
04:00.50roywishhi all
04:00.53NewSoleAnyone fell helpfull to make a test call for me...
04:01.16postelNewSole: people asked you before, scroll up, WHAT kind of test call?
04:01.44NewSoleonly one did DarthClue
04:01.52*** part/#asterisk roywish (~roy_wish@202.155.41.20)
04:01.55postelhow many you wanted? a test pool?
04:02.21NewSoleI just need someone to call a number and see if it rings phone
04:02.36NewSolesee if guy has peer hooked up right
04:02.38postelgive the damn number then, and someone might call
04:02.55freatmmm... phones
04:02.57NewSole5194884245
04:02.57postelyou want a SLA signed before that?
04:03.35NewSolethnx
04:03.40freatnp
04:05.03Sedoroxnight
04:05.27twistedhtml documentation?  why?  can't read .txt files?
04:06.00DarthCluejust to make you ask why.  No other reason than to make twisted ask why.
04:06.05NuxiThe problem is that documentation has to be written after a release because trying to keep up with head is chaos.
04:06.34twistedDarthClue, nice attitude.  that'll get ya somewhere
04:08.21DarthCluetwisted, i'm joking.  It would be nice to have html docs (even lynx could read them) that work similar to the wiki included with the CVS-HEAD download that would make it easier for newbies to get going.
04:08.29*** join/#asterisk SleepyCow (SleepyCow@wnpgmb09dc1-79-72.dynamic.mts.net)
04:08.39twistedi see
04:08.55DarthClueit's just an idea since the wiki is down so much.
04:09.29fugitivodamn broadvoice
04:09.33fugitivoi can't make any call
04:09.38SleepyCowHello all; I have some questions regarding the business end of providing VOIP services. Does anyone have a few miniutes or some links to resources?
04:11.15SleepyCowanyone?
04:14.05*** join/#asterisk santiago (~santiago@63.245.86.198)
04:14.14SleepyCowhello (echo) ?
04:14.33*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
04:14.51Nuxihmmm. echo cancelation must not be working.  ;)
04:15.01SleepyCowhar.
04:15.18*** part/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com)
04:15.20SleepyCowSeriously, I just need a few questions answered that noone seems to be abel to answer....
04:15.31SleepyCowBasicly , on the provider side: how do you get into the PSTN
04:15.35SleepyCowand how much does it cost?
04:15.48`SauronYou get a smart trunk from the *LEC
04:16.24SleepyCownow those have a fixed number of 'lines' , eg connections  to the pstn simultaneously?
04:16.26SleepyCowright?
04:16.58`SauronYes.
04:17.02`Sauron23, to be exact
04:17.03`Sauronper trunk
04:17.34SleepyCowso, with one trunk , i can have at most 23 incomming + outgoing calls to/from the pstn
04:17.52*** join/#asterisk mxmasster (~maxc@216.152.251.69)
04:17.53mxmassterhi all
04:17.57SleepyCowgreetings
04:17.58`SauronCorrect.
04:18.14*** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM]
04:18.27SleepyCownow that doesnt mean i can have only up to 23 phone numbers: i could have say 35 lines, since most wont always be in use
04:18.31SleepyCowright?
04:18.37fugitivobroadvoice doesnt work
04:19.03mxmassterHA
04:19.14SleepyCowwrong?
04:19.15SleepyCowmabee?
04:19.23`SauronCorrect.
04:19.37SleepyCowokay; now how do i find out how much a trunk is from the ilec
04:19.41`SauronAt work, we have 65 numbers allocated to a single T1/trunk
04:19.43`SauronCall them up.
04:19.59SleepyCowsso at work you hcan have 20odd maximum simultaneous calls
04:20.07`Sauronyeah
04:20.10SleepyCowcan someone give me a general price range in whcih to look
04:20.16`Sauron23
04:20.22`SauronNope.
04:20.25`SauronIt varies from lec to lec
04:20.27Ariel_we have a pri with 100 did's asigned to it.  just love it. we have only maxed it's 23 channels once.
04:20.39SleepyCowOkay, sounds good
04:20.47SleepyCowI have a supplier willing to give me did's at $5 / each
04:20.49`SauronAnd there's bulk discounts, grandfather discounts, competetive discounts, cash cow discounts, etc...
04:20.51SleepyCowwhich seems cheap
04:21.06Ariel_x/o here provides a pri with 23 channels for around $ 549.00 per month plus $ 3.20 for 20 did's.
04:21.06SleepyCowhow much would it costs to do it myself is what im trying to find out
04:21.45SleepyCowso roughly $25 per 'real line'
04:21.50Ariel_100 did's are prices at 3.00 per group of 20.
04:22.02*** join/#asterisk cmk (~cmk_@p54A3E583.dip.t-dialin.net)
04:22.06`Sauronse, now our prices are different. :)
04:22.08SleepyCowhow much does it costs for the actual calls over the line? what is free? what is local? how do you get ld?
04:22.12`Sauronsee above discount list
04:22.21`Sauron$700-some for the circuit
04:22.30`Sauronbut I think we got the numbers for free
04:22.35Ariel_all inbound for us are free and all local calls are free.
04:22.35SleepyCow?
04:22.39`Sauron'course, we already have an entire NXX from them
04:22.52`Sauron324-xxxx is all ours
04:23.00wunderkinwow
04:23.08SleepyCowokay well we aremuch smaller than that
04:23.14wunderkinyou whore :D
04:23.28`Sauronhum, not really.. we have ~10k employees
04:23.32wunderkinah
04:23.36`Sauron7-8k, actually
04:23.50SleepyCowso how do i get ld?
04:23.56`Sauroncall them up
04:23.58SleepyCowhow does that work?
04:23.59`Saurononce again
04:24.05Ariel_7 to 8 thousand employee's large co for me.
04:24.07SleepyCowyou get ld from the ilec?
04:24.14`SauronDepends
04:24.18`Sauronwho do you want LD service from
04:24.24`SauronI think we get it from MCI here
04:24.25SleepyCowwhomever is cheap ;)
04:24.27`Sauronor at least used to
04:24.32`Sauronwhatever MCI is called now
04:24.43Ariel_mci
04:24.52`SauronUnless SBC managed to give us a better deal
04:25.01`SauronHell if I know, that's all telecom stuff, not my worry. :)
04:25.12SleepyCowthats what i need to leanr about :(
04:25.23fugitivodamn, this really pissed me off
04:25.24`Sauronwell
04:25.32`Sauronstart taking LEC sales drones out to lunch
04:25.33Ariel_fugitivo, what does?
04:25.38fugitivoAriel_: broadvoice
04:25.39`Sauronor, rather.. have them take you out to lunch
04:25.46fugitivoAriel_: i can't make any call
04:25.48`SauronBV having problems tonight?
04:25.51Ariel_ahh the up and down service
04:25.51`SauronHum di dum
04:25.59fugitivoi don't know, i just signed in
04:26.07fugitivoand i only can call the technical service
04:26.12SleepyCowhow do i get the list of xLEC's in an area?
04:26.13Ariel_sip behind a nat?
04:26.17fugitivono
04:26.34wunderkinyeah xo wanted to take me to lunch.. im like.. dude its just a quote hehe
04:26.35Ariel_did you put insecure=very
04:26.44fugitivoi call a number, and the voice says "we're sorry, we can't establish the call right now" or something like that
04:27.10Ariel_all circuits are busy please try your call later.
04:27.21`Sauronhum
04:27.24`Saurondi dum
04:27.25fugitivo"we're sorry, you're call cannot be completed at this time, please hung up and bla bla bal"
04:27.32`SauronSorry
04:27.33`Sauronworks fine here
04:27.34*** part/#asterisk tengulre (~tengulre@61.185.238.166)
04:27.42`Sauronjust called my cell from BV
04:27.42wunderkinthink im going with sprint for ld pri t1 here
04:27.54fugitivothen should i wait or should i complain?
04:28.06`Sauroncomplain
04:28.17fugitivogreat, the dtmf doesnt work
04:28.21fugitivo:/
04:28.24*** join/#asterisk sbingner (~thanotos@adsl-699.flex.com)
04:28.55sbingnerhey I got more freezing up with my zaptel card, had been very stable until recently
04:29.15sbingnernot worth opening a bug report, I dont have enough info :b
04:29.41`Sauronhum
04:29.41Ariel_wunderkin, mistake
04:29.46fugitivooh, only inband
04:29.50`SauronI get to multihome-nat my machines at home
04:29.55`Sauronhow fun does that sound
04:30.18wunderkinAriel_, what kind of problems have you had with them
04:30.21SleepyCowso how do i find a list of all the lec for an area?
04:30.33`Sauronhum
04:30.38`Sauronwonder if nether.net has a lec list
04:30.41`Sauronpuck.nether.net
04:30.43Ariel_wunderkin, from billing problems, poor data lines and just plain stupid support people nothing.
04:30.44`Saurongo look
04:30.57wunderkinAriel_, was that you that replied on the list a day or two ago about sprint?
04:31.11Ariel_I did reply once.
04:31.39wunderkinok
04:31.41fugitivogrrrrrrrrr
04:32.10Ariel_fugitivo, in the morning I just got an email that a customer wants me to setup bv for him. Argh.
04:32.40fugitivoAriel_: lol, i'm calling support number and dtmf doesnt work
04:32.42wunderkinAriel_, did you just have data with them?
04:33.02SleepyCowsauron, the NOC list?
04:33.27`SauronUmm
04:33.27`Sauronno
04:34.10Ariel_wunderkin, no just t1 phone service.
04:34.20Ariel_they way to expensive for data
04:34.34SleepyCowwhere then?
04:34.35wunderkinoh
04:35.08wunderkinAriel_, what kind of billing problems? dont have the email anymore
04:35.10Ariel_we have a support department that needed to call via modem to take over systems. They were not able to via sprint.
04:35.26*** join/#asterisk techie (gus@antibala.com)
04:35.33Ariel_we finally switch ld to WorldCom for the support department.
04:36.20QwellOfftopic...would anyone happen to know how the intel 915g chipset and the ADD2 cards works?  I'm trying to figure out how it works, but there is very little info available on google...
04:36.45*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:38.52jdv79how can make samples fail?  haha - this is starting to get hilarious.
04:39.42drumkillabecause you're not teh root!
04:40.14jdv79i am
04:40.17jdv79make: *** No rule to make target `/usr/include/asterisk/version.h', needed by `cli.o'.  Stop.make: *** No rule to make target `/usr/include/asterisk/version.h', needed by `cli.o'.  Stop.
04:40.53Qwelljdv79: it isn't a cvs upgrade, is it?
04:41.12drumkillamake clean!
04:41.15jdv79no
04:41.19jdv79its a clean checkout from head
04:41.23Qwelldrumkilla: I was getting to that. :p
04:41.26`SauronI love how there was make clean
04:41.31`Sauronthen there came make distclean
04:41.37`Sauronthen there's make spotless
04:41.39Qwellmrproper?
04:41.43jdv79i'm just thinking today is cursed
04:41.45`SauronI'm waiting for make spitshine
04:41.54Qwell`Sauron: give me 5 minutes
04:41.56jdv79maybe if i go to sleep everything will work tomorrow
04:42.00jdv79:)
04:42.20Qwell`Sauron: rm -rf . && cvs co asterisk
04:42.26`Sauronhehe
04:42.38`Sauronmake antibacterial?
04:42.48`Sauron<PROTECTED>
04:42.52`Sauronhehn
04:43.31fugitivooh well
04:43.38fugitivowho is using broadvoice with asterisk?
04:43.48fugitivodmtf is not working for me
04:44.24jdv79the CTO of broadvoice used to be my boss
04:44.30jdv79for a very short period, thank god
04:44.57fugitivoi signed up today, i'm going to cancel the service tomorrow
04:45.04jdv79i don't doubt it
04:45.18fugitivomaybe i'll not wait until tomorrow
04:45.50jdv79what's the deal?
04:46.32fugitivoi signed up, i can't make any call (when i make a call, it says "we're sorry, we can't complete your call at this moment, please hangup and ...")
04:47.02fugitivoand dtmf doesnt work, but maybe that's my fault
04:51.11jdv79msg me your #
04:51.22jdv79maybe i can get something done about it
04:53.00`SauronI'm using it
04:53.08`Sauronfugitivo: you need ot use inline
04:53.11jdv79my contact has no attention span by the way
04:53.34fugitivoi'm using inline
04:54.14jdv79is that your BV #?
04:54.22fugitivoyes
04:57.29jdv79best way to deal with it is through BV support
04:57.46fugitivoyes, buy i can't access support because dtmf isnt working for me, lol
04:58.02jdv79can't you get to another phone?
04:58.18fugitivoit seems it works only to another BV #
04:58.18jdv79good god man you think VOIP is reliable?!
04:58.51fugitivo`Sauron: could i try to call your BV #?
04:59.46`SauronUmm, sure
04:59.47`Sauron'sec
05:01.19*** join/#asterisk kisu (~daniel@3ffe:831f:cbe9:9a0a:0:fbcc:25d9:5f26)
05:01.34`Sauroncheck /msg
05:03.21`Saurontry forcing it to alaw/ulaw
05:03.43fugitivowhat about no?
05:03.45fugitivonow
05:03.49fugitivono error, right?
05:03.49`Sauronno errors
05:03.52fugitivook
05:03.57`Sauronyup
05:04.02fugitivoi'll try the support line now :)
05:05.11`Saurontry other numbers
05:05.15`Sauronmaybe that's what was causing problems
05:05.24fugitivono, still not working
05:05.32fugitivoand dtmf isnt working neither
05:05.38`Sauronhum
05:05.49Qorkycan anyone here help me with isdn and asterisk. I cant get chan_capi to load
05:05.51Qorkyplease
05:06.30X-Robqorky, you're usually better asking in the morning
05:06.32X-Robthe yanks are awake then
05:06.36X-Robbut what's your problem?
05:06.44jdv79the yanks not suffering from insomnia that is
05:07.00*** join/#asterisk tengulre (~tengulre@61.185.238.166)
05:07.15*** part/#asterisk tengulre (~tengulre@61.185.238.166)
05:07.17X-RobWell yeah. __Except__ the yanks suffering from insomnia
05:09.40Qorkywell X-Rob. im just a bit lost. cant seem to load everything right.
05:09.50fugitivoits working
05:09.56Qorkyi've tried different drivers and still cant seem to get anywhere.
05:10.13Qorkyim using a fritz pci card.
05:10.32*** join/#asterisk Ayano (~erik_leee@dsl093-034-050.snd1.dsl.speakeasy.net)
05:10.54Ayanohas anyone authenticated to primus network before?
05:11.08fugitivowell, i can call to any number now, but dtmf is not working yet
05:11.22X-RobOK. So Quorky, you can't load the module, asterisk crashes when you try?
05:11.30X-Robor what exactly is happening
05:11.48Qorkywell i've tried lots of ways, so its hard to explain.
05:11.54wunderkinfugitivo, are you using ulaw and inband dtmf?
05:11.56Qorkyim currently trying on a 2.6.12 kernel
05:12.18Qorkyi have the following loaded
05:12.18Qorkyroot@asterisk:/tars# lsmod
05:12.18QorkyModule                  Size  Used by
05:12.18Qorkyhisax_fcpcipnp         12416  0
05:12.18Qorkyhisax_isac             10388  1 hisax_fcpcipnp
05:12.19Qorkyhisax                 115104  2 hisax_fcpcipnp,hisax_isac
05:12.21Qorkyisdn                   89120  1 hisax
05:12.33X-Roblooks good
05:12.38Qorkyso the kernel pcpcipnp driver is loaded. as far as i know thats what i want.
05:12.58Qorkyso yeah.. i have all the /dev/capi/blah stuff. that all looks good.
05:13.16Qwellwhat actually happens?
05:13.26Qorkywhen i start asterisk it has a cry.
05:13.39Qorkylet me see
05:13.52Qorky== Parsing '/etc/asterisk/capi.conf': Found
05:13.52QorkyJun 22 13:04:47 NOTICE[2557]: chan_capi.c:2636 load_module: CAPI not installed!
05:14.17QwellDid you install capi?
05:14.19Qorkyand i've made and installed chan_capi-0.3.5
05:14.48Qorkyand added the lines in the modules.conf file
05:14.49X-Roblsmod | grep -i capi
05:15.26Qorkylsmod | grep -i capi = blank
05:15.33Qorkybut.
05:15.34QorkyCAPI Subsystem Rev 1.1.2.8
05:15.34Qorkycapi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
05:15.34Qorkycapifs: Rev 1.1.2.3
05:15.34X-Robyou haven't loaded the capi module
05:15.37X-RobAh
05:15.38Qorkytis from dmesg
05:15.42X-Robyou compiled it in, rather than as a module?
05:16.00Qorkyyar. if i dont. and i load the modules i never get the /dev/capi stuff.
05:16.07Qorkybut the modules load fine.
05:16.32Qorkyyou reccommend using the module eh ?
05:16.45X-RobDoesn't make a difference
05:17.09X-RobUm
05:17.10Qorkyhmm ok.
05:17.11Qorky<PROTECTED>
05:17.11QorkyJun 22 13:07:59 NOTICE[2825]: chan_capi.c:2636 load_module: CAPI not installed!
05:17.11QorkyJun 22 13:07:59 WARNING[2825]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returni     ng -1
05:17.21Qorkythe load_module failed sounds like my prob.
05:17.26Qorkybut dunno what to do.
05:17.36X-RobHrm.
05:17.53Qorkyalso. is there another way i can see if the drivers are ok? like capiinfo etc ?
05:18.02X-Robyeah
05:18.06Qorkybecause i cant get capiinfo and capiinit to work. i do it manually
05:18.41Qorkyalso, i dont need a special version of asterisk or anything do i? because im just running the 1.0.7 version.
05:18.50X-RobI don't have a fritz card, so I've never tried it
05:18.57X-RobIt's better to run HEAD
05:19.09Qorkybut it should work no ?
05:19.09X-RobStable == not moving, it doesn't mean 'won't crash'
05:19.30X-RobI'd start with head
05:19.33X-Robwhat distro are you using?
05:19.39Qorkyslackware 10.0
05:19.40jdv79it implies that its been looked at though and it has less of a chance to crash
05:20.05Qorkyi can change distro if necisary. but I'm very used to slack.
05:20.20X-RobThat should be fine
05:20.43jdv79isn't slack the fBSD lovers linux?
05:20.50Qorkyit ships with 2.4.26 but i've upgraded and got further with 2.4.12.
05:20.50X-Robnah
05:21.14X-Robhttp://www.aussievoip.com.au/wiki-Asterisk-HEAD
05:21.15drrayHEAD gets changed and broken from time to time, STABLE should work
05:21.18X-RobQorky - look at that.
05:21.22drrayI run HEAD
05:21.25X-Robhead works at the moment
05:21.40X-Robsip was borked over the weekend, but that's the first time it's been broken in ages
05:22.38*** join/#asterisk DrRighteous (~DrRighteo@68.199.175.49)
05:22.39X-Robjdv - anyone who runs slackware has a clue
05:22.48X-Robas it's not a distribution for the weenies.
05:24.41Qorkyi have a little clue. but not that much
05:24.48Qorkyhecne why im stuck
05:24.52DaminWow..
05:25.03DaminPulling up carpeting sucks ass...
05:25.48drrayI use fedora core 3, so I can save my clue for * and not setting up X windows
05:31.21Qorkymeh. i'll keep trying.
05:31.28Qorkytah for trying to help
05:35.06X-Robqorky - got a free world dialup accoutn set up?
05:37.12der[mat]g00d morning *
05:40.46twistedwheee
05:41.45*** join/#asterisk jerlique (jerlique@lnk2.adl.adsl.esc.net.au)
05:43.04Ayanohas anyone used primus before?
05:44.46twistedAyano, yeah, they were my lienholder for my car
05:45.06DrRighteousany user-mode linux guru's around?
05:45.26X-Rob...you're trying to run asterisk under uml?
05:45.45DrRighteousX-Rob: yeah, a number of virtual * boxes
05:46.11DrRighteousX-Rob: all dual-xeon systems
05:47.02X-Robwhat's not working?
05:48.22DrRighteousX-Rob: Debian box, UML is loading, using TUN/TAP, but UML networking can only ping/reach host machine, not public internet.
05:48.28DrRighteousX-Rob: driving me crazy
05:49.31X-RobHow many machines do you have on your LAN, and, is the UML machine the internet gateway?
05:50.45DrRighteouslarge number of machines, have a full class C, .1 is a Cisco 12000GSR and is gateway/
05:51.41X-Robhttp://user-mode-linux.sourceforge.net/networking.html
05:51.43X-RobHave you read that?
05:51.48DrRighteousyes...
05:51.55X-Roband you're using tun/tap?
05:52.57DrRighteousX-Rob: yes I am... this is my first attempt at tun/tap.. but once again the IP I assign the UML eth0 as a public IP can ping the hosts public IP, but can't ping any other box on the LAN
05:53.11DrRighteousno NAT/private IPs on LAN either
05:53.12*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
05:53.13RoyKanyone that knows how i can use soxmix to mix two audio files into one stereo file with each input file as left and right channel?
05:53.58X-Robso what's your command line parameter for the uml kernel?
05:55.15RoyKasterisk under uml???
05:55.23DrRighteousRoyK: yes
05:55.25RoyKunder vmware
05:55.36DrRighteousRoyK: no vmware
05:55.48RoyKstill
05:56.12RoyKyou'll lose the realtime performance in linux by using a virtual machine
05:56.17RoyKwhich is bad for telephony
05:56.44cochisounds like he's doing virtual server * hosting. then he has not many options
05:56.47RoyKasterisk on colinux on win2k3 on vmware on linux 2.2!
05:57.00cochivmware, uml, virtuozzo, ..
05:57.09DrRighteousRoyK: ACK WINDOWS...!!!!
05:57.47RoyKperhaps just chrooting stuff could help
05:58.36DrRighteousRoyK: I need a degree of security between virtual machines. And I don't think something like VMWare with Windows would be better, the GUI overhead alone!
05:58.57RoyKi was joking about the windows stuff
05:59.17RoyKDrRighteous: but how many asterisk installations do you want to do per box?
05:59.43RoyKand what sort of hardware is this?
05:59.49RoyKcan you isolate one vm per cpu?
06:00.41RoyKor could you do better with at stack of mini-itx via c3 cards?
06:01.09*** join/#asterisk shido6 (~shido6@d57-87-253.home.cgocable.net)
06:03.31X-Robthe via boards don't have a very fast FPU
06:03.39X-Roband they suck at transcoding
06:03.45X-Robas long as you don't plan on doing any, they rock.
06:04.29DrRighteousRoyK: Dual-Xeon's, 4-8G of ram
06:04.55DrRighteousRoyK: have a ton of these and don't want a small * install to take over each box.
06:08.08*** join/#asterisk kaldemar (~kaldemar@xdsl-204-1.nblnetworks.fi)
06:10.35RoyKDrRighteous: i wonder what sort of asterisk installation that will require > 256MB RAM .........
06:11.00RoyKmy largest one has allocated 5MB
06:15.19|Ladybug|i m back
06:17.01*** join/#asterisk Kieran2 (~kieran@2.197.221.203.velocitynet.com.au)
06:21.00Kieran2Are there any documents on machine hardware requirements for Asterisk servers?
06:21.07Kieran2I havnt been able to find anything specific.
06:23.14shido6what do you want to do Kieran2 ?
06:24.20Kieran2Just a basic system to route maybe 10 voip lines, and 15 - 20 isdn lines. The machine will deal with 3 different incoming numbers, and will need about 10 different voicemail boxes.
06:24.27Kieran2It'll need to handle stuff like call queues too..
06:24.28|Ladybug|hi all
06:24.44Kieran2So i'm thinking a 386 won't quite do it :)
06:25.06Robot_has to support MMX
06:25.47|Ladybug|ne1 know wat kind of processing i need for the origintae n transfer command?
06:25.49*** join/#asterisk gres (~gres@81.222.48.242)
06:26.06|Ladybug|i developed a vc++ program to interact with asterisk
06:26.20|Ladybug|for the oroginate command... it crash my prog sumtimes..
06:26.33|Ladybug|n i cant even trasnfer
06:26.39|Ladybug|ne1 can give me some advice?
06:26.44*** join/#asterisk Inv_arp (junya@adsl-3-251-225.mia.bellsouth.net)
06:27.24SwK|Ladybug|: sounds like an issue w/ the vc++ program
06:27.49*** part/#asterisk shido6 (~shido6@d57-87-253.home.cgocable.net)
06:28.24RoyK|Ladybug|: doesn't really sound like an asterisk problem
06:28.30SwKanyone got that res_php module to compile right?
06:28.53jdv79use perl instead, you'll thank me later
06:28.54jdv79;)
06:29.09Qwellreal men use res_assembly
06:29.22SwKhah
06:29.23jdv79i didn't see that one
06:29.35RoyK&drink(0xc0ffee) while ($tired--);
06:29.36SwKjdv79: i loath perl
06:29.37Qwell~lart jdv79
06:29.40Qwellthere
06:29.50jdv79haha
06:30.08RoyK~lart SwK for saying bad things about perl
06:30.22QwellRoyK: nice
06:30.51SwKyeah roy
06:30.54SwKit doesnt work
06:31.01RoyKit does indeed :)
06:31.05jdv79that's an interesting point of view SwK cause PHP basically wants to be perl when and if it ever grows up.  plus the irony thickens since perl gave birth to PHP inn a manner of speaking:)
06:31.23SwKmaybe so
06:31.28jdv79that's my rant for the night
06:31.30SwKbut i still dont care for perl
06:31.33RoyKphp5 is pretty fscking far from perl
06:31.34jdv79ok
06:32.06jdv79so is perl6 but lets not go there 'til it exists at least:)
06:33.48RoyKi remember talking about the soon-to-come perl6 in 1998
06:34.07jdv79yeah, but now we have a robot on the task and his name is autrijus
06:34.21jdv79;)
06:34.33jdv79who knew AI existing already
06:35.01JerJerjust hope Autrijus doesn't go crazy and bust into your operating room
06:35.03|Ladybug|royk... do u know what kind of response would asterisk give back when i click transfer buttib?
06:35.16|Ladybug|royk.. u have ne idea how to improve the processing so that i dont clash?
06:37.59SwK|Ladybug|: use non-blocking IO to the manager interface
06:41.11|Ladybug|oh.. ok
06:41.41|Ladybug|i m using asterisk manager api n a sjphone
06:42.03*** join/#asterisk whmok (whmok@218.208.97.102)
06:43.38*** join/#asterisk jjg (jjg@adsl-69-226-248-4.dsl.pltn13.pacbell.net)
06:47.07*** join/#asterisk amir (~amir@195.226.9.186)
06:54.42jjganyone doin any production VoIP over VSAT?
06:55.45SwKhah
06:55.52SwKit works
07:02.58*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
07:04.11*** join/#asterisk lehel (~lehel@82.79.20.17)
07:04.28lehelhello
07:07.12*** join/#asterisk denon (denon@synapse.subneural.net)
07:07.12*** mode/#asterisk [+o denon] by ChanServ
07:17.57*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
07:17.58*** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com) [NETSPLIT VICTIM]
07:17.58*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM]
07:21.03*** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
07:21.10*** join/#asterisk Broesmeli (~broesme@195.65.2.68)
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07:24.42*** join/#asterisk stkn (~stkn@stkn-active-pdpc.developer.gentoo)
07:24.54*** join/#asterisk kks (~kks@203.115.208.140)
07:28.11*** join/#asterisk kks (~kks@203.115.208.140)
07:30.32*** join/#asterisk rLg (~umairbari@202.142.189.86)
07:30.32|Ladybug|what is the mwaning of non-blocking IO?
07:31.36*** join/#asterisk flyingmayo (~flyingmay@ip-66-218-239-201.cableaz.com)
07:34.44jansaellwoip+info seems to have serious problems at the moment
07:34.59jansaellvoip-info.com i meant
07:35.14*** join/#asterisk rLg (~umairbari@202.142.189.86)
07:35.17DA-MANhehe voip-info has been damn shitty the past few months
07:35.21DA-MANwonder what the problem is
07:35.28JerJercpu
07:35.30drraypopularity
07:35.52DA-MANJerJer, you run that box?
07:35.57JerJerfuck no
07:36.01QwellDA-MAN: I have a feeling it'd be up if he did
07:36.23DA-MANQwell, wouldn't know . . .
07:36.24DA-MANhehe
07:36.33DA-MANJerJer, meant no offense by it
07:36.45JerJerthis is irc
07:36.59JerJerif someone takes offense to anything said in irc they need their head examined
07:37.09drrayI'm offended by that
07:37.24JerJergood
07:37.27DA-MANJerJer, lol
07:38.33DA-MANman, I wish I had picked a more reputable iax provider, I went with sixTel and those bastards still haven't activated my toll free #. It's been a month
07:38.47JerJerfun
07:39.25DA-MANpretty good for non-toll free though, i use it for home service
07:40.08jansaellnow vopi-info seems to work again
07:40.20JerJersomeone must have kicked it
07:40.30jansaellyes i think so to
07:41.00JerJeri offered to provide collocation space and bandwidth
07:41.21JerJerbut they never took me up on the offer
07:41.36drraywhat about just a mirror?
07:41.39DA-MANJerJer, you work for a colo
07:41.46drraylike linux from scratch does
07:41.51DA-MANJerJer, who is "they"
07:42.00JerJerwiki's are not friendly to mirror
07:42.33JerJersomeone else in here said they need to run media wiki
07:43.22JerJerwhich could get installed on a dual xeon box i just have running distributed.net stuff since it has nothing real to do
07:43.53JerJerbut then data has to be either migrated or created from scratch
07:43.59*** join/#asterisk rLg (~umairbari@202.142.189.86)
07:44.03JerJerwhich i ain't doin'
07:44.05JerJer:)
07:44.19DA-MANJerJer, sounds like a bitchin machine, what kinda pipe would you put somethin like that on?
07:44.36JerJer66.225.202.65 would be the router
07:44.55JerJerphat pipe - last i checked we had 21 gigabits per second aggregate
07:44.58*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
07:45.15JerJer100BaseT switched to gig-e core
07:45.18DA-MANwow, that's awesome
07:45.38JerJerdowntown chicago
07:46.41DA-MAN11  unknown.ord.scnet.net (66.225.202.65)  64.765 ms  64.963 ms  64.983 ms
07:47.01JerJeryeah reverse dns is broken, but decent latency
07:47.18JerJerand 11 hops is about average
07:47.33JerJerunfortunately
07:47.51DA-MANnot too bad
07:47.53DA-MANseen worse
07:47.57JerJeroh yeah
07:48.07JerJerit gets better to though
07:48.17JerJeri'm like 8ms from there and i'm in michigan :P
07:48.36DA-MANwhats scnet do
07:49.04JerJerthat 8ms is basically the pushing limits of the speed of light
07:49.13JerJerserver central
07:49.22Qwell38  unknown.ord.scnet.net (66.225.202.65)  662.173 ms  610.391 ms  598.640 ms
07:49.27Qwellnice
07:49.30JerJerthey run the kick ass ip network
07:49.34*** join/#asterisk fenlander (~neils@82.152.81.57)
07:49.45DA-MANsweet
07:50.03JerJerJordan and company rock
07:50.19DA-MANi do very little network stuff, mostly a system guy myself
07:50.27minidoes asterisk in version 1-0-7 understands ptime in SDP for rtp?
07:50.29JerJerif you notice sixtel is also very close to that ip address  (there is a reason for it  :)
07:50.55Qwelloff to bed
07:51.09QwellJerJer: the above was fake, btw, in case you cared. :p
07:52.13JerJeri figured vsat or something
07:52.19DA-MANholy crap, you're right. I never did traceroute sixtel
07:52.31JerJeror some crap in .pk or .iq
07:53.02JerJerdamn!!!
07:53.02JerJer<PROTECTED>
07:53.06JerJerit was 8ms
07:53.10JerJerthis is great
07:53.46Qwell...my god
07:53.47DA-MAN<PROTECTED>
07:53.51DA-MANthis always trips me out!!!
07:53.52QwellI get 10ms to my first hop
07:53.58QwellSIX FREAKING INCHES AWAY
07:54.12JerJeratm?
07:54.17Qwellsadly, no
07:54.23JerJerodd
07:54.26JerJerbad cable?
07:54.27Qwellnot sure whats going on with that...
07:54.34Qwellslow ass router
07:54.39Qwelldownloading at 500k/s currently
07:54.40JerJerpolish the fiber again  :P
07:54.50DA-MANqwell, wireless?
07:55.03Qwell<PROTECTED>
07:55.10QwellDA-MAN: nope, ethernet
07:55.16Qwell5' cable maybe
07:55.25JerJercat-3  :)
07:55.32Qwellnah, its cat5
07:55.36JerJerlol
07:55.37DA-MANwow, to quote a common slashdot phrase, "that is teh sucks"
07:56.53DA-MANso anyone else have the horror of trying to configure the 4port welltech fxo?
07:56.54*** part/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
07:56.59*** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
07:57.01DA-MANoops
07:58.32Qwellanyhow, bed
07:58.53DA-MANnight
08:00.20*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
08:06.09*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
08:10.51GuERoI what to send a variable to emialbody of voicemail, but, this variable need coming from perl script (agi). ¿ This is posible ?
08:11.50*** join/#asterisk shido (~greg@d57-87-253.home.cgocable.net)
08:13.52*** join/#asterisk pooh_ (~pooh_@a213-84-220-3.adsl.xs4all.nl)
08:17.40*** join/#asterisk olivier__ (~olivier@sud35-3-82-240-204-182.fbx.proxad.net)
08:24.38*** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il)
08:38.28key2yop
08:38.33*** join/#asterisk RoyK (~roy@80.239.107.80)
08:39.17key2anyone knows how one asterisk could tell an otherone to transfer a call to a specific SIP phone ?
08:40.07infisemaphore?
08:40.32key2infi: ?
08:40.37infino no, rfc1149 IP over Avian Carrier
08:40.56key2what u mean ?
08:41.16infipassenger pigeons, man
08:41.51ZeeekWarning: Heathcliff is UNREACHABLE, lag=1241342314231233324132413322 milliseconds
08:46.08*** join/#asterisk jackthe (~jesse@d594f03e.ftth.concepts.nl)
08:47.49*** part/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
08:53.11Romik_somebody uses voipjet? it down now?
08:54.15*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
08:54.38Zeeekwhat is down?
08:54.43Zeeekthey appear to be up
08:56.31Romik_http://pastebin.ca/15516
08:56.51*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
08:57.53Romik_zeeek: to any number i dialed i getting : == No one is available to answer at this time
08:59.28Zeeekwhat is that 800 numer in pastebin? Something I can try to call?
09:00.11Zeeekno that's yours I guess... let me look around
09:01.20ZeeekRomik_ which server do you use? I've heard there were problems with the west coast one
09:01.37Romik_zeeek: which one you use?
09:01.45Zeeekeast, I'm in Eu it's closer
09:02.11Romik_i use new york one
09:02.28Romik_i make test on this one : 18882175648
09:03.28Zeeekworks for me
09:03.38Zeeekyou owe me $0.014
09:03.48Romik_do you accept paypalk?
09:04.00Zeeeksorry, $0.013
09:04.07Zeeekno karma willbe fine
09:04.17Zeeekso that number works for me
09:04.24Romik_1.4 per min...? increment 6 sec?
09:04.36Romik_which ip do you use?
09:04.37Romik_<PROTECTED>
09:04.53Zeeeksorry: total cost of call: 0.00130
09:05.03Romik_ok
09:05.07Romik_tell me which server?
09:05.31Romik_which protocl? ulaw or ilbc?
09:05.36Zeeekthat's the one, New York primary - so the problem is with you
09:05.41Zeeekulaw
09:06.22Romik_<PROTECTED>
09:06.23Romik_<PROTECTED>
09:06.23Romik_<PROTECTED>
09:06.24Romik_strange...
09:07.12Zeeekcheck your callerid - mine is 8663241324 - no '1'
09:07.43Zeeekthat's the format I mean not the exact callerid I use
09:07.48Romik_zeeek: callerid i can anything...
09:08.04Zeeeksome 800 nums will not answer calls with certain prefixes, especially 800 ones
09:09.11Romik_like problem with their server
09:11.46Romik_since 20-jun - there no calls via voipjet
09:12.08Zeeekfor me it works
09:16.28*** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com)
09:21.24*** join/#asterisk [Ladybug] (~ladybug@219.95.78.33)
09:21.34[Ladybug]hi all
09:24.30drraylo
09:24.38Zeeekyo
09:25.24[Ladybug]need help with the asterisk n little with my programing
09:25.33Ahrimanesmornin'
09:25.34[Ladybug]just need to know more behavior abt asterisk
09:25.38[Ladybug]morning/evening
09:27.02[Ladybug]i wrote a vc++ to interact with asterisk
09:27.25[Ladybug]but once i idle more than 5 mintues... n i call out using the softphone.. it crashes
09:27.42[Ladybug]can i know if there is any internal activities inside asterisk itself...
09:27.51[Ladybug]what;s it format pattern?
09:28.09ZeeekNAT probs?
09:28.36[Ladybug]what does NAT means?
09:28.42[Ladybug]sorry.. i am very new to *
09:29.09Zeeekwhat is your c++ app? A softphone?
09:30.16jansaellNAT - Netrowk Address Translation
09:30.18[Ladybug]my c++ application is a program which got the manager api button
09:30.22[Ladybug]thanks jans
09:30.30jansaellnp
09:30.50[Ladybug]for ex.. when i click Login... the asterisk will response back the response to me.. telling me login successful n etc
09:30.59[Ladybug]the softphone i am using is sjphone
09:31.12ZeeekI never got thatone to work
09:31.17[Ladybug]which 1?
09:31.21[Ladybug]u means sjphone?
09:31.50Zeeekyes, sjp^hone - so I can't help you with that
09:32.11ZeeekI thought you were writing the phone... which is why I asked about NAT
09:32.25[Ladybug]oh...
09:32.30[Ladybug]i m not writing the phone
09:32.33[Ladybug]i am using that
09:32.39[Ladybug]i have 2 usbphone
09:32.52[Ladybug]n i use my sjphone to dial to the usbphone
09:33.11[Ladybug]my c++ program will detect the response n send the response back to my vc++
09:33.36*** join/#asterisk Qorky (~Pooa@dip-202-72-131-243.wa.westnet.com.au)
09:34.09der[mat]anyone knows a d-link cpe device with mgcp support?
09:40.17Qorkyne1 know why im getting this: No ISA tormenta card found at d0000 ?
09:45.59*** part/#asterisk lehel (~lehel@82.79.20.17)
09:48.13*** join/#asterisk smeevil (~smeevil@gremesh1.demon.nl)
09:48.34smeevilgood localtime :)
09:50.06smeevili was wondering if any of you could tell me if its possible to generate some static on calls that are in progress.. a lot of people think the line is dead when one keeps silent since they are used to hear some static... i tried to search on google for it and only thing i ran into is that this phenomenon is called comfort noise ?
09:51.06drraya solution in search of a problem
09:51.11drraycomfort noise
09:51.50drraygive everyone bothered by it a wrapper from a pack of smokes and let them crinkle that over the phone
09:52.03smeevilLOL true enough :)
09:53.36smeevili guess a _real_ solution would be to create a small gsm file that contains low volume static and play that looping in the background ?
09:55.27jackthemost SIP-phones have some advanced settings for CN
09:56.08jacktheless load for your *-box
09:56.29jacktheadding CN to a call is eating CPU...
09:57.47Ahrimanesapp_meetme2.c:646: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) <- hmmm
09:57.51smeevilusing cisco 7905....go figure
09:58.34drrayreal solution is to tell them that like cave painting going away, static free phone calls are a good thing
09:59.04jacktheit has CNG
09:59.07jacktheVoice-activity-detection (VAD) and comfort-noise-generation (CNG) support
09:59.20jackthehere is the source http://www.cisilion.com/7905.htm
09:59.42jackthedon't know how to set the CNG on, but Cisco does ;)
10:00.05AndyCapdrray: static free is good, dead silence is not. :-)
10:00.26drrayI fail it, I guess
10:00.47drrayideas like CNG boggle my brain
10:00.54smeevilty jackthe
10:01.00smeevilwill take a look in that
10:01.00drray:)
10:01.45jackthegood luck with Cisco getting the manual to change your CNG-settings
10:01.57drray:)
10:11.13*** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt)
10:24.58*** join/#asterisk Mavvie (edwin@dsl-35.56.240.220.dsl.comindico.com.au)
10:27.57*** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de)
10:29.35*** join/#asterisk lehel (~lehel@82.79.20.17)
10:29.44lehelhello
10:31.19*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
10:32.18*** join/#asterisk meppl (mephisto@p54AAD58D.dip.t-dialin.net)
10:33.57*** join/#asterisk zotz (~zotz@208.196.247.140)
10:34.35*** join/#asterisk speakman (~speak@t2o30p7.telia.com)
10:34.45speakmanhi folks
10:35.51speakmannone awake?
10:36.00Ahrimaneskinda
10:36.07jackthesleepwalking
10:36.08soulshehe
10:36.10speakmanso many people... so many idlers... :P
10:36.11soulslurking mode :P
10:36.22speakmanit might be night in the US? :)
10:36.35soulsi am not in the US ...
10:36.37Ahrimanespeople should be waking up in an hour or so
10:36.38soulsso no worries
10:36.41jacktheme nether
10:36.46zotz@ 6:30 am on the east coast
10:36.46speakmanhehe ok
10:36.48speakmanok
10:37.04speakmanI do agree it's too early yet then.. ;)
10:37.33speakmanAnyone of you using ZapRAS in any application?
10:37.40jacktheit's lunchtime here ;)
10:37.51speakmanjackthe: CET?
10:38.05speakmanor GMT+1
10:38.15jacktheGMT+1
10:38.29*** join/#asterisk FreezeS (~gido_b@83.103.170.130)
10:38.30speakmanor +2 depending on light saving times
10:38.37speakmanor something.. hehe
10:38.40jackthenetherlands
10:38.48soulsjackthe: hehe nice
10:38.49jacktheeasier
10:38.50FreezeShello
10:38.51souls.nl r0x
10:38.52speakmanoh, netherlands.. sweden here..
10:38.56jacktheah
10:38.58soulsgermany here :P
10:39.00FreezeSI have a problem with an ISDN card
10:39.01soulshehe
10:39.04Ahrimanes.dk 0wnz
10:39.12FreezeScan anybody help me ?
10:39.23soulsFreezeS: depends on some more details
10:39.45FreezeSI'm using zaphfc
10:39.52FreezeSand aparently it's all ok
10:40.01FreezeSexcept that I can't make any calls
10:40.06speakmanI have problems with ZapRAS but no one seems to be using that..
10:40.12speakmantried the mailing list for months.. :P
10:40.26soulsspeakman: sorry, never used that one
10:40.39jacktheuse nether of them
10:40.43soulshehehe
10:40.54speakmanwho? what?
10:40.59soulsFreezeS: how do you take it it is all ok?
10:41.00jacktheI'm more the research type
10:41.14speakman?!
10:41.30FreezeScan I paste 4 lines here ?
10:41.53jackthetry it, if it doesn't work try 2 lines twice
10:42.01FreezeSI mean, is it agains policy ?
10:42.02FreezeSok
10:42.04*** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net)
10:42.04FreezeSChan Extension  Context         Language   MusicOnHold
10:42.04FreezeS<PROTECTED>
10:42.04FreezeS<PROTECTED>
10:42.04FreezeS<PROTECTED>
10:42.18FreezeSthat shows up when I type
10:42.22FreezeSzap show channels
10:42.33key2FreezeS: pastbin
10:42.56FreezeSkey2: what's that ?
10:43.04key2for not pasting like that in the chan
10:43.07key2before getting KB
10:43.13FreezeSok, that's why I asked
10:43.32FreezeSI knew that on some channels they use things like that
10:43.52key2FreezeS: http://pastebin.ca
10:44.05FreezeSthanks
10:45.07*** join/#asterisk psywar (psywar@rasterburn.org)
10:46.18psywarHey can anyone tell me what digits I hit on a Sipura SPA-2000 to get the features listed at the end of the manual?  For some reason it has blanks where the two digits should be.
10:46.59FreezeShttp://pastebin.ca/15534
10:47.02speakmanno one using pppd at all within asterisk? :P
10:49.18FreezeSsouls: did you look at pastebin ?
10:49.36soulsabout to
10:50.02FreezeSthere is another problem... I've seen it just now...
10:50.11FreezeSthe signalling is PRI
10:50.21soulshmmmm
10:50.26soulsgood question, never had that error
10:50.31FreezeSbut in zapata.conf I have signalling = bri_cpe
10:50.45*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
10:50.53FreezeSand zaphfc is loaded normally (not NT)
10:50.59soulshmmm
10:51.10soulsas far as i know NT mode is what you would want
10:51.22FreezeShmm
10:51.28FreezeSmy setup is as follows:
10:51.34soulsat least from my experience with ISDN
10:51.47soulseven though i havent used that in years literally
10:51.56FreezeSone computer, with debian, with one isdn card, and a lot of SJPhones connected via network
10:52.17FreezeSok, so I'll try NT
10:52.23FreezeSI'm a noob with telephony
10:53.16soulseveryone here has been at some point in his life
10:53.23soulsand being "pro" is very relative
10:55.01FreezeSok, tried with NT
10:55.03FreezeSsame result
10:55.41smeevilno luck yet with cng on the cisco's :/
10:55.50FreezeSthe weird thing is that although I have  signalling = bri_net, it shows Signalling Type: PRI Signalling
10:56.57jackthesmeevil: try the Cisco helpdesk, maybe they can help.
10:56.58*** part/#asterisk popooya (~popooya@07738e59f333d31b.session.tor)
10:57.48smeeviljackthe: yeah, will do that
10:58.30jacktheotherwise always have some music on in your office, then there is enough Comfort Noise (try to find a nice jazzy radiostation to fall asleep during worktime ;))
10:58.51smeevilfriggin cisco phones really, every logical thing is in the wrong place, to transfer a call you need to go to a sub menu before you can actually transfer, default only has one ringtone, etc etc etc
11:00.00FreezeSso, nobody else has any experience with zaphfc ?
11:03.07postelsmeevil: it doesnt haev a single one, it has Chirp 1 and 2, resample new sounds , edit RINGLIST.DAT and reboot to get more sounds, to tranfer its in the same bottom menu with everything else (hit more once) on CCM you can even remove items or change the order, do you have a question or just randomly bitching?
11:06.02smeevilpostel: well i know all that, using atftpd to see what happends, the phone only asks for the lddefault files, even though i have all the files in the directory of tftpd, default it only has chirp1 (cisco 7905) , but i still need to try the latest firmware. that might solve a lot prolly
11:09.08postelyou got right permissions for /tftpboot? got the files from CCO or emule?
11:10.31*** join/#asterisk emboss (emboss@caffeine.blacknight.ie)
11:10.52embossHi, I have 3 numbers from my telco, i'm looking to setup a pbx for about 20 users
11:11.08embosscan anyone recomend a card to suit?
11:17.50FreezeSI've made some advancements.... now, if I use signalling = bri_net_ptmp, it doesn't show the error
11:18.13FreezeSbut the call isn't coming through eighter....
11:19.11*** join/#asterisk gres (~gres@81.222.48.242)
11:19.44greshi all. Can anybody help me?
11:27.40FreezeSdoes anybody know how can I dial from asterisk command line ?
11:29.34*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
11:31.16Zeeekshow applications
11:34.28leheli had an extension "211" .. i deleted it, but still finds somewhere
11:34.38leheli can't find anywhere..
11:37.41*** join/#asterisk pooh_ (~pooh_@a213-84-220-3.adsl.xs4all.nl)
11:38.19Zeeeklehel how can it be?
11:39.04Ahrimanesdeleted it in the conf file but didnt reload?
11:39.05drrayhave you reloaded the server?
11:39.06Ahrimaneshey Zeeek :D
11:39.13Zeeekbeer?
11:39.17Ahrimaneswhere?
11:39.40jansaellFreezeS: you have to have the console chanel loaded to be able to dial from the command line i think
11:46.15Qorky.
11:47.09*** join/#asterisk Astermigraine (~psolomon@69-165-217-96.atlsfl.adelphia.net)
11:47.52*** join/#asterisk pawalls (~pawalls@pawalls.teamgleim.com)
11:48.24pawallsCan someone PLEASE help me diagnose a problem with a partial T1 PRI I'm having?
11:48.59pawallsI'm using a TE110P, our incoming line from the phone company is 18 channels (1-18), plus the D-channel on channel 24
11:49.35*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
11:50.32pawallsI've configured zaptel using every single combination of fx(s|o)(k|l|g)s and none of them work. All I get when I dial in to the system is a busy signal. There are no alarm states on the card, all of the LEDs on the DS1 look exactly like they do when it's in the current phone server. /proc/zaptel/1 shows all of the channels correctly and no alarm states.
11:50.42pawallszttool shows "OK" as the status of the span.
11:50.47FreezeSaparently bristuff hates 2.4.x kernel
11:51.02FreezeSI'm trying now with the latest 2.6.
11:51.26lehelZeeek: it's ok
11:51.39pawallsBut for some forsaken reason, it is giving me nothing but a busy signal when I call. I've tried every possible combination of options I can think of. I know the connection is esf/b8zs so it can't be that...
11:51.40lehelthe extension number had autoforward to itself..
11:51.44pawallsCan ANYONE think of what would cause this?
11:52.56pawallsI've followed the instructions step by step from the Digium website for configuring a system with a PRI trunk.
11:53.22pawallsBut I'm running into this same wall, and there is not a single bit of useful debugging information to tell me what is wrong.
11:53.52pawallsdmesg says nothing, all proc files show that it is working properly. I'm definitely at a loss at this point, any remote amount of help would be greatly appreciated at this point.
11:54.07Ahrimaneshttp://pastebin.ca/15543 anyone know what could cause this kind of error?
11:56.22Delvarferits
11:56.32Delvarferits with evil mind rays
11:58.38drrayis the light green on the PRI?
12:00.59ZeeekAhrimanes a stab in the dark, you sure all the version of various stuff are up to date?
12:01.41AhrimanesZeeek: yeah, same source code that i am running from now, just added this module
12:01.53Ahrimanesah but could be that the module requires cvs and not 1.0.7
12:02.01Ahrimaneswhere's areski when i need him, hehe
12:02.14Zeeekthat's the kind of thing I was talking about
12:05.58MikeJ[Laptop]everyone checked out the cool stuff at pbxfreeware.org?
12:08.23InfraRedMikeJ[Laptop]: any good porn ?
12:09.04MikeJ[Laptop]sure
12:09.15MikeJ[Laptop]<sigh>
12:10.31*** join/#asterisk mrproper_ (~psynode@CPE-60-225-51-188.nsw.bigpond.net.au)
12:10.46mrproper_what do i need to connect normal analogue lines to asterisk?
12:11.21Zeeekin PC FXO card or modules or FXO devices
12:11.26psywarif you don't need support, an IA92 winmodem
12:11.30Zeeeklike SIpura, Grandstream ATA etc
12:11.51psywaryeah the winmodem talks FXO (to the telco)
12:12.16psywarfor FXS (handset), either a softphone or an ATA like the SPA-2000, about $75
12:12.26psywarthe winmodem is like $7
12:14.39mrproper_basically i have 4 normal analogue lines going out to the telco, i want to feed them into asterisk and have the sip phones use them for in and out dial
12:15.45*** part/#asterisk lehel (~lehel@82.79.20.17)
12:16.51psywarah you may want something higher-density than the IA92
12:16.58MikeJ[Laptop]mrproper_, I don't recomend the $7 modems..
12:17.00psywarcheck out digium's FXO cards
12:17.03jansaellmrproper: You can use the digium tdm400p for that setup
12:17.04MikeJ[Laptop]especially for that many
12:17.11MikeJ[Laptop]what they said ^^^
12:18.21mrproper_thanks guys
12:18.26jansaellnp
12:24.56*** join/#asterisk cjk (~cjk@80.92.64.103)
12:25.08cjkhi, anyone here who uses loadbalancers and asterisk
12:26.09*** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au)
12:26.53Qorkyanyone here setup a fritz card and capi ? im stuck! please
12:27.57*** join/#asterisk heka (~heka@82.114.68.126)
12:35.12*** join/#asterisk Romik_ (~romik@212.143.5.146)
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12:38.51*** join/#asterisk Tili (~Tili@202-133-67-212-dialup.sat.net.pk)
12:45.16*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
12:45.23*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
12:45.43Ariel_Hello everyone
12:47.00*** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo)
12:48.23[TK]D-FenderGood morning
12:50.18jansaellwell aftrenoon
12:53.31*** join/#asterisk [flop] (~flop@socks.epitech.net)
12:53.48Ariel_jansaell, if you say so then it's afternoon at your location.
12:55.28*** join/#asterisk Morex (~blah@host81-157-226-241.range81-157.btcentralplus.com)
12:55.32MorexHello all
12:55.36[flop]I'm using asterisk 1.07 on 5.4 FreeBSD station and  i have a problem : it seems that mpg123 is going into an infinite loop
12:55.51*** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com)
12:55.54Tilihello
12:56.01gambolputtywhat do you mean by infinite loop?
12:56.08gambolputtyconsuming memory like crazy?
12:56.23Tiliis there someone here from Pakistan and has to throw some dialogic hardware. cuz I want some stuff
12:56.32[TK]D-FenderDear God... Anthm has been busy with res_EVERYTHING.......
12:56.35[flop]i'm lauching asterisk with -vvvgc then when trying to stop now, it loops
12:56.52gambolputtykill the process then
12:57.14[flop]with a ps aux, it seems that mpg123 is the only process related to aterisk wich is not stoped!
12:57.30[flop]gambolputty indeed but it should work by itself
12:57.44Ariel_Tili, your not wanting to use dialogic boards with asterisk are you?
12:57.51gambolputtymaybe don't use mpg123 then
12:57.58gambolputtyconvert mp3 files to gsm
12:58.04gambolputtysomething like that
12:58.11[flop]my main problem is that i don't have any sound on my sip phones and i tought it was related to mpg123
12:58.29[flop]gambolputty is the demo using an mp3?
12:58.38gambolputtywhat demo?
12:58.46[flop]the demo context
12:58.53Ariel_[flop], no sound on sip phones is a problem with nat settings most of the time.
12:58.54gambolputtyI think so
12:59.00[flop]you know, when it says welcome to asterisk... :)
12:59.32[flop]Ariel_ we are on the same subnet
12:59.45[flop](the phone and the server) nothing more classical :)
13:00.38drraynat/udp
13:00.59[flop]drray not sure to understand.. :\
13:01.24Ariel_[flop], do you have a firewall on the fbsd where asterisk is running?
13:01.37[flop]no
13:01.42*** join/#asterisk dsfr (~dsfr@207.111.174.1)
13:02.03[flop]let me explain :
13:02.06drrayis the IP phone and server on the same subnet?
13:02.13[flop]yes
13:02.27[flop]no connexion problem between both
13:02.45drraywhen did you build asterisk? and was it head or stable?
13:02.52[flop]the ip phone can even identify itself to the asterisk
13:03.12[flop]drray from ports (supposed stable)
13:03.16drraysip show peers
13:03.20drray?
13:03.45[flop]when typing 2 on the ip phone, the asterisk got it, then launch the demo, but no sounds on the phone
13:04.02[flop]and i tought that mpg123 going on infinite loop
13:04.18[flop]drray yes it shows
13:04.27*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
13:04.50*** join/#asterisk inspired (mikael@host-81-191-123-197.bluecom.no)
13:05.33[flop]let me know, a sound card isn't necessary to make asterisk works (i'm in doubt)
13:05.45Ahrimanesit isnt
13:06.07drrayyou do need a timing device
13:06.15drraybut sip should work for a while
13:06.37[flop]timing device?
13:06.46[flop]wich means?
13:06.52drrayI'm actually outside my area of expertise
13:07.23drraybut the mailing list was talking about how some sip calls went out of sync without a zaptel timing device
13:08.15*** join/#asterisk jr99 (~jr99@73.230.165.24.cfl.res.rr.com)
13:09.01jr99anyone have experience with the sipura device which provides an FXO interface? Can you use the FXO interface with asterisk or is it just for failover/routing within the sipura?
13:09.21drrayflop - have you specified in the sip.conf what protocol you are using?
13:09.57[TK]D-FenderJR99 : you can use it as a full incoming line to * and it can act as a failover all by istelf as well
13:10.02*** part/#asterisk smeevil (~smeevil@gremesh1.demon.nl)
13:10.37*** join/#asterisk gres (~gres@81.222.48.242)
13:11.38jr99[TK]D-Fender: ok. So in theory these could replace my TDM400P with one FXO and one FXS port?
13:11.55MikeJ_[TK]D-Fender, you like the toys out on pbxfreeware.org?
13:12.00*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
13:14.21gresI have question. I have active channel IAX2/incoming-from-x. In op_buttons.cfg, i wrote [IAX2/incoming-from-x] ... - but on FOP. I don't see anything. Can anybody help me?
13:15.10gresMaybe i mast write in op_buttons.cfg [_IAX2/incoming.*]?
13:15.38gresBut it dosn't work too.
13:16.36[TK]D-Fenderjr99 : exactly, and at a cheaper price-point.  However you would lose your Zaptel timing device and have to rely on ZTDUMMY
13:17.58*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
13:18.04jr99[TK]D-Fender: ahh.. thats right.. forgot about the timing.. My problem is that my TDM400P has always been flakey..
13:18.32jr99[TK]D-Fender: does ztdummy work well?
13:18.59[TK]D-Fenderjr99 : mANY HAVE NO COMPLAINS.  mY tmd400 IS FLAKEY AS WELL... WHAT QUIRKS DOES YOURS HAVE?
13:19.07[TK]D-Fenderoops, caps...
13:19.37jr99heh.. well sometimes it will fail to initalize.. and you must reload the zaptel drivers a dozen times until it works..
13:20.50*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
13:20.54[TK]D-FenderHad that happen to me for a while (I modprobe it in rc.local now) and mine "spontaneously" drops calls.
13:21.19MikeJ[Laptop]call digium support
13:21.25[TK]D-FenderMikeJ[Laptop] : Yeah the Res_*Stuff* looks neat, but complicated to set up.  I look forward to ResPHP running here soon
13:21.32MikeJ[Laptop]that's what you pay for when you buy there cards
13:21.34jr99My thoughts were that having the sipura would keep eveything SIP based...And would be $32/FXS..
13:21.47jr99Yea.. I've called them.. never got anywhere.. maybe I should try again.
13:21.47MikeJ[Laptop][TK]D-Fender, should not be that complicated to set up...
13:22.50MikeJ[Laptop]Will touch base with bkw and anthm today to see if we can get a better out of tree module installer andded to contrib.  there is already one that works for single app modules
13:27.01*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
13:27.01*** mode/#asterisk [+o twisted] by ChanServ
13:27.19Qorkyanyone here setup a fritz card and capi ? im stuck! please
13:30.31miniQoeky: ?
13:32.30*** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com)
13:35.11*** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com)
13:38.43*** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com)
13:39.29gambolputtyhow can gsm be used for music on hold?
13:40.18*** join/#asterisk Stephnie (dfsdf@203.215.180.254)
13:40.19Delvarhmm there was a mod to do that... basicly a perl sccript that did a cat of the file and piped to asterisk
13:40.37Delvarno idea what its called tho :)
13:40.47StephnieI just compiled and installed asterisk ..... first time it run fine.
13:40.52StephnieI rebooted my system
13:40.53StephnieUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
13:40.57Stephnienow getting this msg
13:42.09MikeJ[Laptop]Delvar, rawplayer
13:42.33MikeJ[Laptop]you can also use native MOH in head
13:42.33Delvarah thats it!
13:42.45Stephniehow to run asterisk as Daemon?
13:42.46MikeJ[Laptop]it's in contrib\utils
13:42.56Delvarcool
13:42.56MikeJ[Laptop]Stephnie, type asterisk
13:43.14Stephnieok
13:43.24Stephnieyes, now asterisk is fine
13:47.13[TK]D-FenderMikeJ[Laptop] : Just biding my time until these new tools are more easily merged and for 1.0.8 to be released.  I am quite close to * acceptance here at work now.
13:47.31der[mat]is it possible to run asterisk as server and simultaneous as a sip-client?
13:47.45cochio.O
13:47.51cochiyou mean to register at sip providers?
13:48.43*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
13:52.47*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:52.47*** mode/#asterisk [+o anthm] by ChanServ
13:53.59der[mat]cochi: jebb
13:54.24der[mat]afk
13:54.24der[mat]pardon me
13:54.25*** part/#asterisk der[mat] (~mat@gate-nue0.bintec.de)
13:55.09*** join/#asterisk Katty (~angela@68.112.15.110)
13:58.29*** join/#asterisk orospakr (~orospakr@ip-168.82.126.206.dsl-cust.ca.inter.net)
13:58.36*** join/#asterisk VoIpMaster (VoIp@194.105.96.243.static.cablesurf.de)
13:58.47VoIpMasterhi  to all
13:59.24VoIpMasterthe first question is does anybody know a german asterisk channel? I'm here in Germany and English is only a native language for me ...
13:59.28*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
14:00.30InfraRedyou can always start #asterisk-de
14:00.52*** join/#asterisk olivier_ (~olivier@sud35-3-82-240-204-182.fbx.proxad.net)
14:01.33orospakrhi! I am trying to build asterisk on an ubuntu x86_64 box, and I'm seeing: cc1: error: bad value (k8) for -march= switch.
14:01.49VoIpMasterthx InfraRed
14:02.44VoIpMasterbut there is only a bot into the channel :) so i think stay here :)
14:03.32Stephniewhen I reboot my machine....asterisk doesnt run automatically...
14:03.48Stephniehow to make it run automatically when computer resets...
14:04.05*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
14:04.23InfraRedStephnie: what distro
14:04.28StephnieRH9
14:04.50InfraRedput A STARTUP SCRIPT IN /ETC/INIT.D
14:04.57InfraReddamn capa lock
14:05.26Stephniehow ?
14:05.45InfraRedcaps
14:05.52*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
14:06.20[flop]i'm using te405p on a freebsd 5.4 with zaptel drivers and i have no sound on my sip phones, but when i unload the kernel module of the card i do have sound and everything's working. do someone experienced such problem?
14:07.31VoIpMasterhi flop, is it possible that your soundcard and your te card uses the same irq's?
14:07.34StephnieI dont know how to do that
14:07.41*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
14:07.46Stephnieto put a startup script in /etc/init.d
14:08.13Corydon76-homeThen hire someone who knows what that means and can do it
14:08.16StephnieI think I can do it thru command...to make asterisk run automatically with the OS
14:08.25StephnieI hired myself :p
14:09.04Stephnieok ok . gonna do that .
14:09.06[flop]VoIpMaster don't thnik so, but what's the relationship beetwen the sound card and the te
14:10.22VoIpMasteri don't know flop
14:11.10VoIpMasteractual i'm fighting on a clean install of RH9 and the i get no network ... that's very cracy...
14:11.13orospakrah, my problem was ubuntu'
14:11.20orospakrs choice of gcc 3.3, not 3.4
14:11.42Hmmhesays~seen bkw_
14:11.42jbotbkw_ is currently on #asterisk
14:12.13*** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com)
14:12.43VoIpMasterI'm away for the next 60 minutes, if important leave a PM
14:13.28Hmmhesaysanyone in here using dell poweredge's?
14:15.16[TK]D-FenderI keep getting a "file not found" type error looking for "/etc/init.d" ;)
14:16.09Hmmhesayshaha RoyK
14:18.23*** join/#asterisk brookshire (~matt@207.111.174.1)
14:18.37*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
14:19.18Hmmhesaysthat girl on the digium site is kind of cute
14:20.11Wonkahmm, he says...
14:20.34Wonkayeah, definitely considerable
14:20.35Hmmhesaysindeed
14:20.41KattyHmmhesays: mew
14:20.53HmmhesaysKatttaaaaay
14:21.28brimstonepamples?
14:21.33Kattyyes
14:21.36Katty:<
14:22.00Hmmhesayshow goes it this morning?
14:22.24Kattyam sleepy
14:23.11*** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net)
14:23.58stormfrhello, did anybody already see a iax trunk not anymore working ? (cvs head 2005-06-09)
14:24.31orospakrhi! I'm getting chan_zap not found whenver I start asterisk.  I didn't compile in zapata, and I don't need to use the cards, so what do I need to do to tell asterisk to not load the plugin?
14:25.27InfraRedlook in the conf files asterisk.conf or something like modules.conf
14:25.28InfraRedcant remmber the name
14:25.34InfraRednoload => module
14:25.35*** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com)
14:26.11brimstone/etc/asterisk/modules.conf
14:26.36brimstonenload => chan_zap.so
14:26.43`SauronSNORK
14:26.56*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
14:28.06*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM]
14:28.06*** join/#asterisk newl (~newlook@203-59-112-225.dyn.iinet.net.au) [NETSPLIT VICTIM]
14:28.06*** join/#asterisk ptiggerdine (~ptiggerdi@c210-49-98-194.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM]
14:28.31brimstonei prefer meaty smoothies
14:29.04*** join/#asterisk Hmmhesays (negative3k@66.173.103.108) [NETSPLIT VICTIM]
14:29.11Hmmhesaysand suddenly I felt very alone on asimov
14:29.14Hmmhesayslol
14:30.27orospakrah, I found my problem. it was actually a larger config problem.
14:36.38*** join/#asterisk mjman (~mike@205.158.42.66.ptr.us.xo.net)
14:37.46mjmanHi, I am having some issues with a SNOM 190 SIP phone, and the page on the wiki does not have what I need. Call waiting simply does not work on these phones. When I test it out, I get a fast busy on the second call. Does anyone have experience with this or a similar phone??
14:38.58*** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net)
14:40.49*** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net)
14:41.01file[laptop]#42 7*6
14:41.26file[laptop]I'm still lacking in sleep
14:41.43mjmanfile[laptop]: what?
14:42.04file[laptop]my prom was lastnight, and safegrad
14:42.04*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
14:42.11drumkillafile[laptop]: !!!!!!!!!!!!!!!
14:42.15drumkillahow was it?!?!?!?!?!
14:42.22file[laptop]hi drumkilla
14:42.27file[laptop]it was good
14:42.31drumkillayay!!!!!
14:42.37drumkillathat's just so cute ...
14:42.40file[laptop]in the end I said screw it to everything safegrad and went to the loft and slept
14:42.47*** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net)
14:42.55file[laptop]during which I won a cake in my sleep
14:42.58bkw_OMG you use punctuation like a gay man!!!!!!!!!!!!
14:43.00drumkillapics?
14:43.18file[laptop]pics? meh
14:43.39brookshireme too
14:43.43foodnow eat me
14:43.46*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
14:43.47brookshirei'm hungry and it's like 9:43
14:43.56DrRighteoushi bkw & file! morning
14:43.59file[laptop]it's 11:43AM, and I haven't eaten since... well, I don't remember
14:44.03file[laptop]hi
14:44.04brookshirehey file
14:44.08brookshire:P
14:44.18file[laptop]brookshire: Matttttt
14:44.31mjmanHi, I am having some issues with a SNOM 190 SIP phone, and the page on the wiki does not have what I need. Call waiting simply does not work on these phones. When I test it out, I get a fast busy on the second call. Does anyone have experience with this or a similar phone??
14:44.47drumkillaI'm drinking red bull
14:44.47brookshireOH GNO!
14:44.47file[laptop]drumkilla: when aren't you drinking red bull?
14:44.47brookshireNOT THE REDBULL
14:44.54drumkillayes.  the RED.  BULL.
14:44.59brookshireRed.bull.
14:45.17file[laptop]yay - stumbles!
14:45.46brookshireREDBULL.ORG.UK is free
14:45.47brookshirehehe
14:46.04orospakrCan't locate Asterisk/AGI.pm in @INC. wjat
14:46.12orospakrs/wjat/what's up with this?/
14:46.16DrRighteoustrademark warming!!! whoo! whoo!
14:46.30*** join/#asterisk Cramnoselo (fwuser@host-22.216-16-72.iw.net)
14:46.33brookshireRedBull :)
14:46.42brookshireRedBull!!
14:47.35file[laptop]orospakr: you don't have the Asterisk AGI perl stuff installed
14:47.35*** join/#asterisk tagore (~ircap8b@r200-125-24-169-dialup.adsl.anteldata.net.uy)
14:47.41CramnoseloAnyone using TDM04B and Asterisk @ home succesfully???
14:48.02CramnoseloI am having issues with the card not recognizing after running yum to update
14:48.03*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
14:48.19orospakrfile[desk], is that included in one of the official asterisk packages or is that completely 3rd party?
14:48.28orospakrer
14:48.32orospakrfile[laptop]. whoops.
14:48.34doolphCramnoselo follow the instructions
14:48.50Cramnoselowhich instructions???
14:48.55doolphat their page
14:49.02doolphthere's a how to to install TDM
14:49.07doolphcards
14:49.07file[laptop]orospakr: Google it
14:49.15shmaltzwhat does this mean:
14:49.17CramnoseloI run the genzaptelconf -s -d and it does not work...
14:49.17shmaltz<PROTECTED>
14:49.20Hmmhesaysbkw_: were you the poweredge fan?
14:49.27file[laptop]orospakr: type in Asterisk::AGI - first result that comes up
14:49.54*** join/#asterisk Betu| (~betul@62.244.193.101)
14:50.44shmaltzwhy am I getting this:
14:50.46shmaltz<PROTECTED>
14:51.10brookshiretoo much lag, or the server isn't responding
14:51.39brimstonedoes Asterisk @ Home have a support channel?
14:52.02drumkillabrimstone: #amportal
14:52.19brookshirelol
14:53.26orospakrugh. now this:
14:53.28Hmmhesayslemme guess, amp screwed up your config
14:53.32orospakrUse of uninitialized value in pattern match (m//) at /var/lib/asterisk/agi-bin/dialparties.agi line 123, <STDIN> line 39.
14:53.33Hmmhesaysand you can't fix it
14:53.36CramnoseloI run the genzaptelconf -s -d and it does not work...
14:53.46file[laptop]what am I, an AGI debugger?
14:54.37orospakrwell, it *would* be rather odd for a regular user like me to run into a whole new compile-time bug, yes?
14:54.39Ahrimanesyeah
14:54.51*** part/#asterisk kaldemar (~kaldemar@xdsl-204-1.nblnetworks.fi)
14:54.53Betu|hi, i have DIGIUM TDM11B and want to all module for pstn,Do i need to by fxo modules for TDM11B,dont i
14:55.29brookshireyes
14:55.39brookshirex100m
14:55.56file[laptop]orospakr: compile-time bug?
14:55.58file[laptop]orospakr: dialparties.agi isn't part of asterisk that I know of
14:56.12orospakrhmmm
14:56.18Hmmhesaysi believe that is part of amp
14:56.19orospakrperhaps amportal is munging something
14:56.20file[laptop]it's from AMP
14:56.55Betu|Is x100m another type of card or the name of fxo module ?
14:57.06Hmmhesaysyou don't have to use amp with * @ home... in fact the use of custom-context is a good way to go
14:57.13dsfrBeirdo: it's the FXO module.
14:57.20Beirdo?
14:57.21bkw_it tells you the problem
14:57.34dsfroops, went that for Betu|
14:57.37Hmmhesaysthat way you can leave the amp configs alone... and just use your own.... it provides a good interface for editing the *.conf files
14:58.10orospakryeah, rerunning AMP solved it.
14:58.13orospakrd'oh!
14:58.15Hmmhesaysbkw_: were you the dell poweredge fan?
14:58.22*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
14:58.27bkw_Hmmhesays, yes
14:58.30BoRiSbkw!!!!!!!!
14:58.44orospakranyway, it seems that I am back at the same problem as I had before: asterisk segfaults on SIP phone hangup.
14:58.57CramnoseloAnyone using TDM04B and Asterisk @ home succesfully???
14:59.20Betu|thanks
14:59.23*** part/#asterisk Betu| (~betul@62.244.193.101)
14:59.28Hmmhesayscurious I'm poking through the dell website... what OS do you like to run on those... dell gives you windows or redhat from the factory
14:59.33CramnoseloI am having issues with the card not recognizing after running yum to update
15:00.28JerJeruse the source
15:01.16file[laptop]yay people's court
15:01.36*** join/#asterisk ilan (ilan@69.60.110.251)
15:01.39*** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
15:01.49tagoreCramnoselo, i'm using TDM22B and is running fine
15:02.04IPmongerhey folks
15:02.08Hmmhesaysit's not people's court without judge wapner
15:02.20IPmongerquick question: is there a open-source SIP call generator?
15:02.36Cramnoselotagore, when I run the genzaptelconf -s -d, it tells me there is no zaptel...
15:02.39Hmmhesays~google
15:02.39jbotit has been said that google is a search engine found at http://www.google.com/
15:02.39CramnoseloAny ideas?
15:02.41JerJerIPmonger: asterisk
15:03.07tagorechange the pci slot and try again
15:03.10tagore:O
15:03.35CramnoseloDo you think I should reinstall the whole system or just move the card?
15:04.12tagoreyou can try move the card, if this not work , reinstall, and not ryn yum again :P
15:04.15tagorerun*
15:04.16*** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
15:05.04tagorethe card is working before you run yum?
15:06.05Cramnoseloit seems to be the LEDs light before yum, but not after...
15:06.24tagorecrap !!
15:06.49tagoretry reinstall, and not run yum :S
15:07.21Cramnoselowill it run stable without running the updates?
15:08.21bkw_crontab -e
15:08.24bkw_doh
15:08.41tagorei'm running the stable without updates
15:08.50tagoreand is runnign fine
15:08.58tagoreand is in lan ;)
15:09.13tagorelow risk
15:09.45Nuxires_php works with cvs-head (it even compiles)  http://eder.us/projects/asterisk_php/asterisk_php.0.1a.tgz
15:10.02DarkSpectrehey, where are the voicemail recordings actually stored at
15:10.35Nuxi<PROTECTED>
15:10.54DarkSpectrecool.
15:11.12Cramnoselowill do tagore, thanks for the help!!!
15:11.23tagoreu'welcome
15:12.32dalaberaIPMONGER take a look onto this? http://www.inaccessnetworks.com/projects/asterisk-oh323/utils
15:12.42*** join/#asterisk jansaell (~jan@c80-216-185-161.cm-upc.chello.se)
15:16.45*** join/#asterisk ArkyLady (ArkyLady@adsl-208-191-253-160.dsl.ltrkar.swbell.net)
15:17.36*** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net)
15:18.43Hmmhesaysanyone using a dell poweredge 1800?
15:19.15*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
15:20.08file[laptop]hi Mike
15:21.16*** join/#asterisk Corydon-w (grey@vcchgate.vcch01.springfield.tn.us.vcch.net)
15:23.09*** join/#asterisk grolloj (~grolloj@slim-eth0.horizonlive.net)
15:34.18rephormhas anyone ever had zap channels get stuck in ast_waitfor_nandfds? (like, for ~120 hours after the call was hung up). this is on a zap to zap call using a channel bank and a wc t100p
15:36.19*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
15:36.27shido6ouch'
15:36.47[TK]D-FenderI'm trying to pick a server to use for * for my company and am not liking the $ figure Dell is giving me.  Any suggestions?
15:37.42*** join/#asterisk Cresl1n (~matt@216.207.245.23)
15:37.49doolphyou can build your own [flop]
15:37.52doolphi mean
15:37.52clueconTKD: how soon do you need an answer and what price range are you targetting?
15:37.56Ariel_[TK]D-Fender, look into a supermicro
15:37.58doolph[TK]D-Fender
15:38.47[TK]D-FenderWell I quoted 2000$CAD (rough) for a quicky server that is a local store white box with known * compatible parts (just the server w/o cards)
15:39.12[TK]D-FenderAnd told them I may be willing to look a little higher, but a raided Dell 2850 is running over 5K
15:39.45[TK]D-Fenderbut is much better supported.  I like the low-end idea for the ability to use ATA drives which can be swapped at the drop of a hat though and cloned as easily.
15:40.06*** join/#asterisk mjman (~mikem@205.158.42.66.ptr.us.xo.net)
15:40.07*** part/#asterisk mjman (~mikem@205.158.42.66.ptr.us.xo.net)
15:40.14[TK]D-Fenderand of course the cost factor.  ATA raid has me a little nervous because of how often the OS sees both drives independantly (Slack 10)
15:40.20[TK]D-FenderSo I'm a little conflicted
15:40.29clueconTKD: I am building an AMD64 3000+ with 4 Gigs of RAM and 1 TB of storage for around $1500 USD
15:41.28[TK]D-Fendercluecon : I know I can do that white-boxing, but its a question of corporate approval as well...
15:42.13[TK]D-FenderAnd I've heard here that there may be interrupts issues on AMD based solutions (although I swear by them for PC's)
15:42.15jontowhey wtf is this "X-Lite v2.0 Build 1105d for Linux" ;)
15:47.34orospakrhttp://pastebin.ca/15558 <-- log of segmentation fault. on hangup.
15:47.38orospakrit's consistent.
15:48.28clueconorospakr: what version of * are you using?
15:48.37orospakr1.0.7 on x86_64.
15:48.42orospakrcompiled with gcc 3.4.
15:48.57orospakrthe original ubuntu package for 1.0.6 on x86_64 did the same.
15:51.40*** join/#asterisk dennis (dennis@dennis.mod.unixboard)
15:51.52clueconorospakr: have you tried CVS-HEAD?
15:52.05Silik0norospakr or did you get backtrace?
15:52.25orospakrSilik0n, I can't seem to run gdb on asterisk (either my build or ubuntu's)
15:52.31orospakrit just crashes right away in some nptl stuff
15:52.49orospakrhmm, perhaps I could try attaching...
15:53.13*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
15:54.25orospakrah HAH
15:55.28orospakrhttp://pastebin.ca/15559 <-- backtrace.
15:56.46*** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc)
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15:57.35brettnemgood morning
15:57.45[TK]D-FenderWhat is the easiest way to follow the changes being made in CVS-Head?  Scanning the mailing list archives looks a little messy right now.  Is there a better way (with an eye on following mod's and documentation on their use)
15:58.07brettnemeyow!
15:59.54*** join/#asterisk kFuQ (~somedude@c-67-185-114-199.hsd1.wa.comcast.net)
16:01.15*** join/#asterisk vooduhal (~christoph@67.19.25.178)
16:01.41*** join/#asterisk Stephnie (dfsdf@203.215.180.254)
16:02.09jontowx-lite on linux runs .. extremely well
16:02.17harryvvit does?
16:02.24harryvvI have some errors from it.
16:02.29harryvvmissing some kind of library
16:02.47vooduhalQuick question guys.  We're using one of the 4 port modular cards with 4 FXO modules and when dialing on a Zap channel, it takes ~4-5 seconds before the call starts ringing.  Does anyone know why this is and a way to get shorten that period?
16:03.10*** join/#asterisk bonez41 (~aint@drjones.dsl.xmission.com)
16:03.10harryvvI can see xlite as a temperary solution to ip phones but thats it.
16:03.10Silik0nvooduhal: out pulse time to send digits to the CO
16:03.39orospakrkphone works well now.
16:03.44vooduhalSilik0n: I'm not sure I understand.
16:03.45bkw_cvs-stable needs to die
16:03.46Ariel_vooduhal, that is due to it's trying to wait for caller ID info
16:03.47rephormvooduhal: dialing out?
16:03.52harryvvjontow what distro are you using?
16:03.53vooduhalYes, going out.
16:03.54bkw_I think some people would agree with me
16:04.02bonez41is it possibl with asterisk, to have a specific incoming call routed to another number? i.e., if party X calls and I want them to just call party Y...can asterisk do that?
16:04.02vooduhalWe've resolved the inbound time.
16:04.07orospakranyway, no one has any ideas about that mysql_real_escape_string problem?
16:04.08bonez41possibl=possible
16:04.16Ariel_bkw_, no I like stable
16:04.18*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
16:04.31jlewisbkw_: whats wrong with cvs-stable?
16:04.31file[laptop]vooduhal: it takes time to dial the digits...
16:04.39Ariel_vooduhal, are you dialing from a sip phones?
16:05.07vooduhalAriel_: Yes, but the call from the phone to asterisk is pretty much instantious (sp).
16:05.09jlewisI've got several servers running various -stable snapshots
16:05.17rephormvooduhal: what device are you calling (i.e. what is ringing 4-5 seconds after you dial?)
16:05.39Ariel_jlewis, I have lots of asterisk boxes running stable.
16:06.03vooduhalrephorm: Nothing.  That's the problem.  It takes 4-5 seconds before the ringing on the line starts.
16:06.05*** join/#asterisk unabonger (BhongwanSh@stilyagihall.themoon.org)
16:06.25jontowharryvv; Fedora Core 3 on the machine i tested with
16:06.42*** join/#asterisk cmk (~cmk_@p54A3EBD2.dip.t-dialin.net)
16:07.22*** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
16:07.36harryvvis that the 32 bit version?
16:07.40jontowyes
16:07.53harryvvokay had problems with it in 64 bit version.
16:08.17jontowno idea about 64bit :/
16:08.17jontownever owned one of them :)
16:08.46harryvvI might just download burn and install 32 bit version of fc3. just to much is missing to make this a effective combined ws/ser
16:08.58jontow:/
16:09.31harryvvor fc4 seems that it has alot of the fixes thats a part of fc3
16:09.58unabongerany FreeBSD Asterisk gurus?  Got a st()()pid question about a performance problem after changing from Shrike (RH9.0) to FreeBSD for my Asterisk server.
16:10.08*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
16:10.18rephormvooduhal: seems like that's just the way it is. takes a few seconds for the card to answer the call, then dial the numbers, and then for the fxs at the other end to send back a ring tone
16:12.40vooduhalrephorm: No, this is from an FXO going out.
16:13.17vooduhalIe, SIP phone calls asterisk, asterisk routes the call out via the FXO on a CO line to our DMS.
16:13.38vooduhalOne thing that we've noticed that is really odd, is that it when it dials the card pauses for 1 second before the last digit.
16:13.50vooduhalSo its like 555121---2
16:15.27*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
16:15.41unabongercan anybody tell me how to check disk cache performance and read/write speeds on FreeBSD?  I think my disk might be munging my performance during certain calls.   Either that or Asterisk Server to NuFone latency probs.
16:16.12tagorehdparm -Tt ?
16:16.32Stephnieexten => _7.,1,Dial(SIP/${EXTEN:1}@outgoing) <--- I dont want the prefix 7 ....I want to number to be dialed as it is...what should I use?
16:17.02unabongertagore ,thanks I couldn't remember.
16:17.06tagore:D
16:17.07*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
16:17.07*** mode/#asterisk [+o bkw_] by ChanServ
16:17.31unabongerdangit
16:17.42tagoreexten => _.,1,Dial(SIP/${EXTEN:1}@outgoing)
16:17.45unabongerthat's how I would do it on RedHat, but that doesn't seem to work on FreeBSD
16:18.13Stephnietagore: asterisk told me what to use:)  it says ..use   _X
16:18.27tagoreok
16:18.39Stephnienot working :S
16:20.07tagoreand _. ?
16:20.32Silik0n_. catches everything include s, i, and h
16:20.44tagoreis true
16:20.47KattyHmmhesays: mrow?
16:20.51Silik0n_X. is what you really want
16:21.13Silik0nexten => _X.,1,Dial(SIP/${EXTEN:1}@outgoing) or something
16:21.26HmmhesaysHey Katty, I got split before
16:21.36tagoreunabonger http://lists.freebsd.org/pipermail/freebsd-questions/2004-October/061533.html
16:21.40Hmmhesayssuddenly I was on asimov with like 3 people
16:22.52Kattyasterisk does faxes  :>
16:22.53Kattyi'm all excited and bouncy now :>
16:22.53HmmhesaysI was rambling on about my weekend too.. to myself
16:23.01Kattyheh
16:23.22Hmmhesaysit sure does
16:24.00StephnieSilik0n:  _X.  is not working
16:24.08Hmmhesaysin fact if you have to asterisk endpoints  you can store and forward
16:25.17Beirdobouncy?
16:25.21Kattyyes
16:25.27Stephnie_7.  works   but   _X.   doesnt work.
16:25.36Beirdohehe, you really shouldn't put such thoughts in the minds of geeks :)
16:25.38HmmhesaysKatty: i should send you an episode of bonkers
16:25.43Silik0nStephnie what kind of phone are you using?
16:25.49Hmmhesaysi'm reminded of that every time you say "bouncy"
16:26.35Silik0nkatty: A/S/L/
16:26.46Silik0nKatty: oh sorry... forgot this was a tech chan heh
16:27.02StephnieSilik0n : X-Pro
16:27.56*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
16:28.00Silik0nStephnie can you paste that portion of your dial plan to pastebin.ca with a verbose capture from the console you you try to dial out?
16:28.37Stephnieok
16:29.44StephnieSilik0n:   _X.   means    any prefix ...but there should be a prefix...for example: to dial 5551212 ...it could be X5551212
16:29.47StephnieI just checked it out.
16:30.09Silik0n_X. mean anything that starts with a digit 0 - 9
16:30.16Stephnieyes....
16:30.19Silik0nso 0123123123 or 0 would match
16:30.27Hmmhesayssingle digits don't match that
16:30.31Stephnieyes...correct but I need to dial number direct..
16:30.31Silik0nture
16:30.35Stephnie5551212   ....
16:30.39Silik0ni meant 00
16:30.53Silik0n_X would match single digits
16:31.07Hmmhesaysja
16:31.23Hmmhesaysdoncha know
16:31.23Stephnieyes. but I need to dial 5551212  without  0, 7 , 8 ...
16:31.31Stephniewithout any prefix..
16:31.36Hmmhesayspastebin.ca
16:32.00Silik0nonce its on pastebin.ca let us see it
16:32.29jansaelldo you men that you whant to dial wothout a prefix?
16:32.36Silik0nyou might be better doing something like _9. for the pattern then you would dial 95551212 and ${EXTEN:1} would give you 5551212
16:32.38jansaellwithout...
16:32.48Stephnie_X. means  X5551212 ..... I need to dial 5551212
16:33.48jansaellthe ${EXTEN:1} drops the first character in the current extension
16:33.54StephnieSilik0n:  I dont want to send any extra digits ( _X. ) with number
16:34.08jansaellso if you dial 95551234 ${EXTEN:1} return 5551234
16:34.38lesouvageI run festival --language cepstral_diane --server after running the cepsral script and copying to the festival dir. This is the errormessage: "Unsupported language, using English" SIOD ERROR: unbound variable : voice_rab_diphone. Any clue?
16:34.52StephnieI think u guys dont understand what I mean..
16:34.53Stephnie:)
16:35.14StephnieI dont want the number (5551212) to be followed by any DIGIT.....
16:35.42StephnieIf I need to dial 5551212 then I should dial it 5551212 from my x-pro...not 75551212 or 95551212
16:35.45Stephniegot it ?
16:35.55Nuggetyes, we get it.
16:36.00Nuggetwe got it before
16:36.01Nuggetwe get it now
16:36.16Stephniethen what were you doing?? sleeping ?
16:36.17Stephnie:p
16:36.19*** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net)
16:36.23DarkSpectrein asterisk, can you have the option of when a user calls in, it can login to the system via a "username" and "password"?
16:36.46Nuggetgetting angry with the people who are trying to help you is not a path for success.
16:37.05Stephnieif is it for me ? then I am not angry :)
16:37.25Stephniethere is difference between anger and joke ;)
16:37.32Stephnie_X.,1,Dial(SIP/${EXTEN:1}@outgoing)  <----- so ?
16:37.41jansaellok Stepanie:  exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@outgoing)
16:37.53Nuggetbetter to use NXXXXXX
16:37.53*** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
16:37.55jansaellthat match 7 chars
16:37.56Seyrheyas people
16:39.10lesouvageStephnie: make an extension with the external numer you want to call and pass it to the zap channel exten => 1234567,1,Dial(Zap/1/91234567).
16:39.43lesouvagesorry i tried to cancel my suggestion but I pressed Enter by accident.
16:39.45NuggetWhat Stephnie is missing is the function of the ":1" in the variable.
16:40.34Nugget:1 makes it strip off the first digit, which I guess has led Stephnie to assume that a prefix is mandatory, since asterisk was always dropping that first digit.
16:40.35*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.168.115.68.195.rev.coltfrance.com)
16:41.02Nuggetbut asterisk was just doing what the dialplan said.  Drop the :1 and the while exten will be dialed
16:41.06Nuggets/while/whole/
16:41.52*** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net)
16:41.59DarkSpectredargh
16:42.00Nuggetthis whole ordeal is rapidly turning into "just tell me what to type" and it sounds like you could be well-served by spending some more time with the documentation.
16:42.14orospakrhow would I go about disabling the mysql CDR module? just another noload?
16:42.32Nuggethow variables are expressed in the dialplan, including using :1 to strip off a prefix, is fundamental asterisk knowledge that you should have before you dive too deep
16:43.53Stephnieexten => _XXXXXXX,1,Dial(SIP/${EXTEN:1}@outgoing)
16:43.57Stephnienot working
16:44.04Stephnieoops
16:44.05*** join/#asterisk Kernel_core (Raph@9.229.dial-up.xter.net)
16:44.07Nuggetof course not.
16:44.10*** join/#asterisk Romik_ (~romik@212.143.5.146)
16:44.14StephnieI got it
16:44.14jansaellno drop the :1
16:44.15StephnieI got it
16:44.16Stephnie:)
16:44.18jontowwell.. the good news is.. x-lite works on netbsd to a point
16:44.19Stephnie:D yes
16:44.25jontowi still haven't been able to get it to register to my * though
16:46.00Kernel_corehi all , I am looking for equvalinet command ( output attenuation , input gain) in cisco for asterisk! For SIP and IAX channel !does asterisk have such command ?
16:46.01*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
16:47.06*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
16:47.22MikeJ[Laptop]anybody have a good source for nortel referbished gear?
16:48.04StephnieThank you guys.....:) working now....
16:48.05Stephnie:)
16:48.33StephnieNugget: thanks.......
16:48.36Stephniejansaell: thanks
16:48.37Stephniebye
16:49.56*** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34)
16:53.22*** join/#asterisk cluecon (cluecon@wsip-68-99-73-32.tu.ok.cox.net)
16:53.28DarkSpectrequick question
16:53.47DarkSpectrei'm gonna have a box running to accept at least 6 lines. which card would be best suited for this type of project?
16:54.18MikeJ[Laptop]you want 6 lines on 1 card, then you are using a t1 with a channel bank
16:54.30MikeJ[Laptop]or you could use a couple tdm cards
16:55.07DarkSpectreso, a Wildcard TE405P would work then
16:55.42Ariel_DarkSpectre, yes if your going to pluging 4 channel banks or  4 t1/pri lines
16:56.07DarkSpectreworking on a project for a radio station. I'm not sure exactly what their incoming lines would be
16:56.30*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
16:56.57PBXtechits not normal when a phone re-registers to pick a new port each time is it?
16:57.10SeyrIf I have SER in front of Asterisk and I am just passing outbound calls to *, what config changes do I need to make to the * box?
16:58.04Seyrso far all I have found is to set "autocreatepeer=yes" in sip.conf
16:58.41dalaberacorrect me if I wrong please: if I do a cvs checkout zaptel,libpri I'm getting latest dev code. Can I mix it with asterisk stable code ??
16:59.37jontowwell i had x-lite under netbsd registered and making a call; now im stuck at "Discovering Firewall..." when i start it
16:59.40jontowthat sucks :)
17:00.14cluecondalabera: it is not recommended to mix dev and stable.  the results are unreliable at best.
17:01.08harryvvjontow, is that x-lite on the same network as the * box and you dialing out to another x-lite or how is your x-lite setup?
17:05.34jansaellme
17:05.55*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
17:06.07SeyrAnyone familiar with using SER in front of Asterisk?
17:06.46*** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net)
17:06.46harryvvseyr, have you bothered to read the documentation?
17:07.30Seyrharryvv: whats a documentation?
17:07.43unabongerWiki-Tiki-Tavi, baby!
17:08.12*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
17:08.28MikeJ[Laptop]Seyr, you have this really bad echo.. the same thing keeps coming out of you every hour or so for days now...
17:08.37MikeJ[Laptop]do you need help with SER and asterisk?
17:08.55jontowsame network as * (192.168.2.0/24)
17:09.11jontowdialing to an internal extension on that particular *, not outbound at all
17:09.14SeyrMikeJ[Laptop]: You have a bad memory :-) yesterday I asked about difference between SER and OpenSER, today I am asking about config :-)
17:09.37MikeJ[Laptop]if you need help, I know some people who can help...
17:09.54MikeJ[Laptop]but not free
17:10.00SeyrMikeJ[Laptop]: I have SER up and running, just having problems with outbound calls going through *
17:10.21MikeJ[Laptop]well, if you are willing to pay, let me know and I will have them contact you
17:10.27dalaberacluecon thanks for advice
17:11.12cluecondalabera: no problem.
17:13.32*** join/#asterisk allanon (allanon@c-24-18-190-208.hsd1.wa.comcast.net)
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17:14.53*** join/#asterisk Nukemizer (~Nuke@67.137.28.165)
17:16.19PBXtechits not normal when a phone re-registers to pick a new port each time is it?
17:17.50*** part/#asterisk Cramnoselo (fwuser@host-22.216-16-72.iw.net)
17:22.58*** join/#asterisk fugitivo (~ajf@168.226.244.27)
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17:25.53yaaarword
17:25.54fugitivohello
17:25.59clueconPBXTech: what kind of phone and define picka a new port.
17:26.03clueconsentence.
17:26.16PBXtech7960 out a nat
17:26.27PBXtechdefine what?
17:26.45clueconwhat do you mean by it picks a new port?  Is it using sip protocol?
17:27.02PBXtechevery time it re-registers is has a new nat port
17:27.04PBXtechsip yup
17:27.45PBXtechit seems to loose the ability to dial to the phone
17:27.54clueconsip uses 5060 to register and RTP 10000:20000 to carry voice packets.
17:28.05clueconis it using a port outside that group?
17:28.27PBXtech3819/3819        66.219.xx.xx   D   N      255.255.255.255  1125     Unmonitored
17:28.57PBXtechso * is calling the phone on 1125
17:29.18*** join/#asterisk ArkyLady (ArkyLady@adsl-66-142-125-19.dsl.ltrkar.swbell.net)
17:29.40*** join/#asterisk hholzer (~hholzer@ip130.may.co.at)
17:30.13PBXtechand it will register after 3600 and pick a new port
17:30.13PBXtechdoesnt seem normal
17:30.46clueconi'm not familiar with the 7960 but does the phone have a setting that allows it to pick a random port?  i know the grandstream i'm presently testing with does.  if so, that needs to be disabled so it will use the standard port.
17:31.10PBXtechnot aware of that setting
17:31.46orospakr\
17:31.51orospakroops!
17:32.34*** join/#asterisk afrosheen (~afro@c-24-0-141-232.hsd1.tx.comcast.net)
17:32.52file[laptop]PBXtech: it's the NAT that is doing it...
17:33.01file[laptop]PBXtech: my friend's D-Link does it...
17:33.08afrosheenfile[laptop]: what, 1 way audio?
17:33.08file[laptop]PBXtech: just enable qualify to keep the UDP mapping open
17:33.16file[laptop]no, port madness
17:35.18*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
17:36.29hholzerhi, can someone help me at a problem with a Digium TDM20B Card ?
17:36.39Cresl1nhholzer: what's up man?
17:36.40hholzerthe wcfxs module didnt want to load and i got "ProSLIC 3210 version 2 is too old" in the message log
17:36.47*** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com)
17:36.53Cresl1nhholzer: update to latest CVS
17:36.59Cresl1n(your zaptel)
17:37.03afrosheenproslic?
17:37.07Cresl1neither CVS-HEAD or CVS-STABLE
17:37.12Cresl1nI think both should work
17:37.23hholzeri am using zaptel-1.0.7
17:37.24Cresl1nor check to see if there's a newer version of cvs-stable
17:37.29hholzeris this too old ?
17:37.35afrosheenwhat is proslic anyway
17:37.38Cresl1nhholzer: apparently :-)
17:37.47Cresl1nafrosheen: it's the line interface part on the fxs module
17:38.42CoffeeIVI am recording my calls through *.  I am using the example from voip-info which does stereo mp3 files.  I would also like to email the recording to the email address for that extension; is there an easy way to do that ?
17:39.42CoffeeIVI am about to write a perl script to parse voicemail.conf and do it "by hand", but I don't want to if there is another way
17:41.23clueconCoffeeIV: use res_js
17:41.57clueconhttp://www.pbxfreeware.com
17:42.58Nuxior res_php with php's builtin ini parser.
17:43.18*** join/#asterisk truz24 (~raydogg@12-220-103-82.client.insightBB.com)
17:44.22Nuxihttp://us2.php.net/manual/en/function.parse-ini-file.php
17:45.42*** join/#asterisk Maxxed (~max@cpe-70-114-238-9.houston.res.rr.com)
17:45.51Maxxedbeh
17:46.18Maxxedhey, i have a issue with my cisco ip phones, the time in the upper part of the lcd disapears after a short while
17:46.23Maxxedi reboot the phones
17:46.25Maxxedand its back again
17:46.34*** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:46.52Maxxediv spokem with another asterisk user the otherday in regards to the issue, and they have the exact same problem
17:47.05truz24I just installed an analog card and asterisk, what is a quick way to test to make sure it works?
17:47.18truz24asterisk is running.
17:47.19CoffeeIVcluecon: thanks, i'm glad to know about those links -- I think I am going to write my own perl script though, it seems simpler than learning a whole new interface
17:47.20Maxxedthe solution to bring the clock back that they found was to set the time to 24hr format and switch it back
17:47.29Maxxedand it will come back with out rebooting the phones
17:47.35Maxxedbut it will even disapear after a while
17:47.43wunderkinmaybe its a feature
17:48.12Maxxedanybody know about this issue? its gota be somthing cisco related
17:48.18Maxxedfirmware issue or something?
17:48.29clueconCoffeeIV: res_js is pretty simple to use and will make it really easy to do the record and email that you want to do.
17:49.17Maxxedim running sip 7-4-00 and bootloader pc03a300
17:50.36loudMaxxed, you use any ntp server ?
17:50.51cluecon~cluecon
17:50.51jbotit has been said that cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
17:51.52*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
17:52.15Maxxedloud: nope, at least i dont think i do
17:52.28*** join/#asterisk sangee (~rkuru@207.188.77.86)
17:53.50Maxxedah, well, i do have the phones set to get there time from another ntp server
17:53.56Maxxednot directly from the asterisk
17:54.16Maxxedi have a ntp server on my gateway that the ipphones use
17:54.26sangeeI want to pass through fax, how do i setup Asterisk for that
17:54.43loudreboot the phone, * 6 + settings without the ntp
17:54.56loudyoull see the time.
17:57.00*** join/#asterisk riksta (~rick@212.85.228.176)
17:57.50Maxxedloud: the ntp is what freaks the time out?
17:58.13Maxxedloud: there could be an issue with my firewall that isnt permiting the phones to "talk" to the ntp server
17:58.50lesouvageCan somebody please tells me what file hides the standard voice for festival? I run festival --server
17:59.08lesouvageI mean the setting of the standard voice.
17:59.12clueconlesouvage: are you wanting to change the voice?
17:59.23loudwaht i did works, to me. don't know about the firewall issue, my 7960 has a public address.
17:59.46*** join/#asterisk ctooley (~ctooley@rrcs-24-227-212-181.sw.biz.rr.com)
17:59.57*** join/#asterisk Trakk (~Trakk@adsl-8-244-88.mia.bellsouth.net)
18:00.01lesouvagecluecon: yes everything is in place except the change of the default voice.
18:00.07ctooleyDoes asterisk's agi "STREAM FILE" have the ability to stream from a file socket?
18:00.36Nuxinope
18:00.41ctooleycat file.gsm > /tmp/SOCKET
18:00.46loudyou unpolite people never say hi when joining this room.
18:00.48loud:)
18:00.50Maxxedloud: il check it out, thanks :)
18:00.51ctooleyand have asterisk read from that?
18:00.58Trakkloud: sorry
18:00.59Trakkhi
18:01.06Trakkmy client is about to quit
18:01.07Maxxedhi Trakk
18:01.09Maxxed:)
18:01.16TrakkIRC client that is
18:01.39Trakkbe back on later then.
18:02.24lesouvagecluecon: I expected a .conf file where I had to change a line. The point is that I can't find out what file to change.
18:02.41*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
18:03.10pooh_Is there anybody who has a few minutes to check something about AGI with me pls ?
18:03.27clueconlesouvage: one sec, I'm looking for mine.  it is part of the festival conf not *
18:03.35Nuxipooh_, whatcha got?
18:03.59*** join/#asterisk P-Chan (~jpfingstm@68.142.66.200)
18:04.56clueconlesouvage: you need to edit the file siteinit.scm.  Mine is located under festival/lib.  My voice line is as follows:
18:04.56P-ChanWhat would cause Asterisk to give a "Zap/2-1 is busy" when trying to call out?  I keep getting "All circuits busy"?  We have more than enough circuits available.
18:04.56cluecon(set! voice_default 'voice_cstr_us_jmk_arctic_multisyn)
18:05.14ctooleyNuxi, was your "nope" directed at me?
18:05.24pooh_Nuxi: need to change the context for a call with 'set context'
18:05.54Nuxiyes, but I was thinking network sockets.  don't know about unix sockets.
18:06.14pooh_Nuxi: based upon a dbget value the context has to be set to continue dialing
18:06.37*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
18:06.56pooh_Hi ManxPower, Eindhoven was fun! :-)
18:06.58Nuxipooh_, so what is the problem SET CONTEXT <context>
18:07.04truz24How do i install the zaptel module in ubuntu ?
18:07.10truz24I got the x100p digium card
18:07.20lesouvagecluecon: thanks
18:07.23pooh_Nuxi: The problem is that I do not have anything, just this wish
18:07.29ManxPowertruz24, Um, Digium no longer sells that card
18:08.19clueconlesouvage: your welcome.
18:08.27*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-5-125.d4.club-internet.fr)
18:08.40truz24ManxPower, so?
18:08.54*** join/#asterisk frogy (~edmund@cm218-255-137-80.hkcable.com.hk)
18:09.37pooh_ManxPower: got my idea *almost* working (HotDesk, but need to be able to set the correct context before commencing the dialplan withing the correct context
18:09.40frogyanyone has problem use the Meetme conference here?
18:10.22pooh_NuxI: I guess it would be a 1 or 2 line agi before continue a call in the given context
18:10.57frogyMy conference is having huge delay, and makes it less than usable.
18:11.04[TK]D-FenderExtensions.conf should have an "include" facility to link in other .conf files so as not to create 1 monstrous file.....
18:11.05*** join/#asterisk clive- (~pirch@rrba-146-88-153.telkomadsl.co.za)
18:11.32*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
18:12.14sivana[TK]D-Fender: you can
18:12.25[TK]D-Fenderhow?
18:12.28*** join/#asterisk bofh42 (~bofh42@p5482B957.dip0.t-ipconnect.de)
18:12.36[TK]D-FenderI must have missed that somewhere...
18:12.44sivana; Macros File
18:12.44sivana#include macros.inc
18:12.56sivanathe file.ext can be anything
18:13.11clueconTKD: #include "my-ext.cluecon"
18:13.26sivanaI dont' have quotes
18:13.43frogycan anyone give me some advise on how to solve the Meetme delay problem?
18:14.05JerJermeetme delay problem?
18:14.15clueconquotes are optional i think.
18:14.25pooh_Nuxi: ping ?
18:14.27clueconthey are shown in the docs, but i don't use them either.
18:14.44Nuxipooh_: pong
18:15.04Nuxiwhat language are you friendly with?
18:15.06clueconI have heard that the meetme delay problem may be addressed at cluecon.
18:15.07bkw_frogy, hop to cluecon
18:15.48pooh_Nuxi: for now the few agi scripts I have (stolen/use) are in perls and I have the perl stuff installed
18:15.56heisondoes anyone know if Asterisk can act as a SIP register proxy? i.e. when User-Agent A register with Asterisk via sip.conf, can Asterisk take the registration and register with SER for User-Agent A, such that incoming calls received by SER will be directed to Asterisk
18:15.57[TK]D-FenderOk, I see it now
18:16.01[TK]D-Fender<- blind
18:16.06HmmhesaysI hear I still need to find money to go to cluecon
18:16.08DrRighteouscluecon?
18:16.12bkw_asterisk is a B2B UA
18:16.24heisonthat is incoming calls to user agent A received by SER
18:16.30*** join/#asterisk Lethargicclown (~chatzilla@ool-44c28c91.dyn.optonline.net)
18:16.34cluecon~cluecon
18:16.34jboti heard cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
18:17.03Nuxipooh_ I don't know anything about the perl stuff.
18:17.04JerJerheison: why not have ppl register to SER?
18:17.13bkw_ya what JerJer said
18:17.14bkw_hehe
18:17.19pooh_Nuxi: a little background, I need to be able to direct a call to the correct context based upon a dbget related to a user, NOT a device (e.g. sip.conf)
18:17.21*** join/#asterisk pjz (~pj@place.org)
18:17.36*** part/#asterisk DrRighteous (~DrRighteo@68.199.175.49)
18:17.37heisonbecause SER is a proxy but I need to control features provided by Asterisk
18:17.45frogyyeah, I use SIP phones to connect to the Meetme. At first, it works ok fine. But after 30 seconds, it starts come the delay. And it can be as much as 4 seconds for someone to hear each others.
18:17.55heisonSER is currently running on hardware I have no control over
18:18.04bkw_frogy, we told everyone how to fix it
18:18.08bkw_and even tried to give back changes
18:18.08pooh_Nuxi: so when a user places a call, a dbget value is checked, and accordignly the correct context is parsed instead of the context given in sip.conf
18:18.18bkw_but they were rejected!
18:18.21heisonI could do static routing on the SER, but like to avoid if possible
18:18.27pooh_Nuxi: fine by me
18:18.44anthmpooh, are you using *choke* stable
18:18.46pooh_Nuxi: php >= 4.3.x is ok
18:18.49frogyI read the post in the Mantis too.
18:19.20heisonJerJer: what we currently have is SUA -> SER -> Cisco, and we want to implement SUA -> Asterisk -> SER -> Cisco
18:19.26frogyAnd applied the patch, but it doesn't help. I'm running 1.0.7.
18:20.18pooh_OR.... I need to be able to change the cached dialplan and cached sip.conf values....
18:20.42JerJerheison:  why?
18:20.45frogybkw_ where can I find the solution?
18:20.58Nuxipooh_ you should be able to do that without agi.  I think anthm can help.
18:21.01clueconfrogy: come to cluecon.
18:21.21heisonJerJer: i want to provide call features with Asterisk
18:21.22bkw_frogy, well the problem comes up because mark is chincy with threads in meetme
18:21.22Nuxi[[anthm is much smarter than Nuxi]]
18:21.47bkw_if you dedicate a thread to playing sounds and other conf tasks it works great
18:22.02anthmaww, go on, Nuxi made res_php!
18:22.11pooh_anthm: Greetings stranger, the force led me to you with my search for a solution
18:22.13bkw_we have fought with mark over this.. he thinks it would be too resource intensive to fire up a thread
18:22.53anthmwell using stable is your first obsticle, I'm pretty sure there is a secret bristuff for head
18:23.04bkw_considering you can fire off 1000's of threads per second on a nice box ..
18:23.05bkw_and do millions of malloc'
18:23.05JerJeranother new thread for each new call?
18:23.05bkw_no
18:23.05bkw_one total thread for the entire conf
18:23.06bkw_its job is to do things like play files
18:23.06Hmmhesaysis there a patch floating around that might do that?
18:23.06truz24Does anyone run asterisk in ubuntu?
18:23.20anthmif you do want stable and you want to use agi just rember the golde rule
18:23.27bkw_JerJer, maybe mark was thinking two threads per caller.
18:23.40anthmin and out, asap
18:23.56pooh_anthm: I guess my wish has nothing to do with whatever patch/version, but rather to change the given context entry within sip/iax.conf
18:24.08anthmagi should do minimal task and exit back to the dialplan never prolong an agi session
18:24.42wunderkinis that to get bfs if you use asterisk? what about gfs?
18:24.43bkw_Nuxi, what was that link again
18:24.45frogybkw_ so is there any way to get it to work in 1.0.7 ?
18:25.09anthmwell, um it has to do with it a little since you could do it in a 1 line callflow in head =p
18:25.17Nuxires_bf.c => http://pastebin.ca/15414
18:25.51bkw_where was the page about bf?
18:26.15Nuxi~google bf
18:26.23pooh_anthm: so where do I go from here and with 1.0.7, to change the context given in sip/iax.conf preceding processing a call by a user
18:26.37Nuxihttp://www.muppetlabs.com/~breadbox/bf/
18:27.35*** join/#asterisk sudhir492 (~sudhir@pool-71-114-77-25.washdc.dsl-w.verizon.net)
18:27.51heisonJerJer: I have a work around... although not what I wanted... I can have register lines in sip.conf for all the numbers I want to take from the SER over to Asterisk, if an incoming call from PSTN terminates on Asterisk and the SUA is not available, I can play Congestion
18:27.59anthmyou want to alter the default context of the peer triggered by a call?
18:28.24pooh_anthm: yes, exactly
18:28.27Hmmhesayswhy not just send them to that context
18:28.30Hmmhesays?
18:28.39Hmmhesaysbased upon their callerid
18:29.12pooh_Hmmhesays: not possible, give it a try and you will see that the cached dialplan will play you tricks based upong sip/iax.conf
18:29.15frogybkw_ can u please tell me more about how to fix the meetme?
18:29.39anthmsend them all to the same context and use dynamc switching logic to direct them to the desierd context.
18:29.49Hmmhesaysheh, i hope that wasn't for me anthm
18:30.30anthmboing!! boing!! no its for  pooh woo hoo hoo
18:30.44pooh_so simply put, I need to set the context=.... value within sip/iax.conf from the dialplan to overrule the given values within sip/iax.conf, at the time of calling
18:31.04anthmno you are over-complicating yourself into a corner
18:31.08Hmmhesayspooh_  cmd gotoif would be one way to do it
18:31.20anthmtake a step back and let the abstraction aroma sink in
18:31.25*** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il)
18:31.26JerJerheison:  i don't follow the problem
18:31.51pooh_Hmmhesays: yeah, RoyK advised me that, using accountcode and gotoif from a default context, but that does not work for now
18:31.51Hmmhesaysanthm: i'm stealing that
18:31.57frogycan anyone tell me more about how to fix the meetme delay problem?
18:32.05JerJerwhat is this delay problem?
18:32.20JerJeri don't experience any delay on meetme
18:32.25kajtzuhmm. what delay problem?
18:32.30Hmmhesayspooh_ : then you are doing something wrong
18:33.04bkw_its a problem when playing the enter exit tones in meetme
18:33.05bkw_and or using the menu
18:33.06pooh_yes
18:33.06pooh_I know
18:33.06pooh_:-)
18:33.06bkw_I think its more voip related than TDM JerJer
18:33.14frogyJerJer: When using SIP phones to connect to the Meetme. it can has as much as 4 seconds delay for everyone to hear each others.
18:33.30pooh_ok... let me explain in a bit more detail
18:33.38pooh_* is very device oriented
18:33.39*** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com)
18:33.47pooh_I want * to be user oriented
18:33.53pooh_so......
18:34.00*** join/#asterisk AgiNamu (~Michael@200.6.215.184)
18:34.04AgiNamuhello
18:34.39frogyI have used the q option, but delay is still there.
18:34.40pooh_walk to a device, login and from that moment that device is your phone and ALL calls in/out will be controlled by your account, not the device
18:35.03pooh_e.g. not controlled by sip/iax.conf
18:35.07anthmhow many ways you want to solve that?
18:35.12anthm2, 4?
18:35.14AgiNamuHey, any downsides to storing a north american 10-digit number as a 32-bit int?
18:35.15pooh_and I mean everything related to your account, vm, calls in/out
18:35.16sudhir492I see the following error on the server when trying to send a fax using PAP2-NA: ast_rtp_read: Unknown RTP codec 100 received
18:35.20file[laptop]anthm: 42
18:35.53pooh_I am very close, just need to be able to set the correct context, and I am done
18:36.02sudhir492AgiNamu: why would you want to do that?
18:36.07DannyFres_js is neat *grin*
18:36.12Hmmhesayssounds like a pain in the @ss pooh_
18:36.28file[laptop]anthm: such insanity!
18:36.35AgiNamusudhir, well, it is WAY less to store an int32 than a string.
18:36.46AgiNamuI'm guessing 28 bytes less
18:36.51pooh_Hmmhesays: I know, I have been trying for 2 days now.... no solution yet
18:36.57pooh_Hmmhesays: but I will not give up :-)
18:37.05AgiNamuand indexing, etc. will be smaller, faster, etc.
18:37.13DannyFAgiNamu, unless the number never starts with a 0 all is good ,)
18:37.20kajtzuAgiNamu: you're assuming north american numbers? there's the rest of the world too ;-)
18:37.28AgiNamuDannyF, yea, a north american 10 digit
18:37.36Hmmhesayspooh_ are you using realtime?
18:37.36bkw_DannyF, did you see the orderstatus example I posted
18:37.40DannyFshould be fine
18:37.41AgiNamukajtzu, that is correct. only 10 digit numbers. this thing can't touch other numbers anyways
18:37.50DannyFbkw_, yeah, playing with it ;)
18:37.52pooh_Hmmhesays: nope, all from the dialplan or a AGI script
18:37.53AgiNamui thought it'd be fine, but i just wanted to see if i was missing anything hue.
18:38.00bkw_did you see the format of the return data from the remote side?
18:38.08anthmhere's 1 way. store all that info in a mysql or any db, and make frontends to all of the functions you need to offer that are agi that look up the info in the db based on the callerid when you log in and out you install your caller id into the db
18:38.09Lethargicclowndoes the wildcard tdm400p make the asterisk box work like an ATA?
18:38.11file[laptop]bkw_: OMG BECKY!!!
18:38.13DannyFmyhash?
18:38.27sudhir492AgiNamu: faster indexing etc. seems a valid reason. Not the storage. If you are sticking to US and Canada numbers only, then I guess no problem in storing as int
18:38.28bkw_its asterisk config file like... [blah]\nname=blah\n
18:38.36bkw_but the remote CGI can output that on request
18:38.42bkw_and you can suck it up into myhash
18:38.42DannyFahvalue pairs eh
18:39.07DannyFor whatchacallem ini blocks?
18:39.15Hmmhesaysanother way you could do it is use an agi to modify your sip.conf and reload sip
18:39.18fugitivoLethargicclown: not exactly, ATA means analogic telephone adapter, asterisk with an fxo card is more complex than that
18:39.24bkw_DannyF, yep
18:39.30AgiNamusudhir492, storage in memory == faster
18:39.35DannyFvery usefull
18:39.39file[laptop]memory is always faster...
18:39.44AgiNamuno,
18:39.54AgiNamui mean, less memory , i.e., less storage, results in faster code
18:39.57Hmmhesaysunless you've been drinking ;) <chuckle>
18:40.08AgiNamukeep as much stuff in cache as possible
18:40.16LethargicclownFugitivo: More complex how?
18:40.21pooh_the problem is that * always starts with the context given in sip.conf/iax.conf thus device oriented, while I look at communiction as users using a stupid device, whatever device
18:40.31AgiNamuthanks everyone
18:40.33DannyFbkw_, sure makes some nice things a helluwa lot easier to make in a few minutes
18:40.33anthmyou make them all use the same context
18:40.55pooh_anthm: Tried that, not working because of the cahed dialplan
18:41.11pooh_or I need to be able to change the cached dialplan
18:41.12anthmtried what
18:41.33pooh_but I do not know the cached variable names if at all accessable
18:41.33file[laptop]Incense and Peppermints!
18:41.43anthmyou have 1 context the sip device tries to call somewhere and hit's _X.1 to an agi that sets the proper goto context exten pri
18:42.03anthm_X.,2 goto's the contents of ${GOTO} that the agi set
18:42.43pooh_anthm: exactly what RoyK adviced, and I treid, but within the 'end' context there is no real 's' exten because the context consists of includes only
18:43.20pooh_AND I rather change the context from the 'start'...
18:43.37anthmone more time
18:43.48*** join/#asterisk dwiltjer (~dwiltjer@63.227.22.62)
18:44.09file[laptop]it's great, I got it from bkw
18:44.13anthmyou install the context in the db associated with the user where you surely are storing it anyway if you want to bother having users
18:44.31anthmyou dial 1234
18:44.34DannyFbkw_, did u try run festival stuff (or similar) though it?
18:44.51dwiltjeri have an error 'Ring/Off-hook in strange state 6 on channel X' and i've researched it a bit, it told me to turn off a few settings in my zap.conf file, but that dind't fix it
18:44.54anthmthe catchall looks up your callerid in the db and sets the context variable
18:44.57anthmthen
18:45.18anthmGoto(${MYCONTEXT},${EXTEN},1) in the next line
18:45.24pooh_anthm: that is the idea, but how to make it work. I tried a 1Mio ways
18:45.24anthmtada dynamic contexts
18:45.39dwiltjerdoes anyone have any suggestions? its on a T1 with a TDMX100P card
18:45.41anthmi just practicly showed you working code
18:46.05file[laptop]anthm: you should know better then to help in here, it just makes you angry and raises your blood pressure
18:46.13pooh_(20:41:50) pooh_: anthm: exactly what RoyK adviced, and I treid, but within the 'end' context there is no real 's' exten because the context consists of includes only
18:46.37anthmwhen you dial the login ext you go to an agi that does auth if it's a success you put the remote device in the db as a relation for the dynamic outbound
18:46.57anthmi read that
18:47.08anthmand said to myself, wtf does that mean and moved on
18:47.23pooh_anthm: It sounds like the way to do it, but since the real trageted (outgoing) exten is included, it can not be found within the 'Goto' context
18:47.37*** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
18:47.38dwiltjerdoes anyone know anything about the 'ring/off hook in strange state' error?
18:48.16yaaarcan anybody take a quick look at these 4 lines? http://pastebin.com/302599 ....they're error messages I get when trying to make an outbound call across my x100p....
18:48.44pooh_anthm: please explain a bit more about : when you dial the login ext you go to an agi that does auth if it's a success you put the remote device in the db as a relation for the dynamic outbound
18:48.50anthmif you say Goto(mycon,1000,1) and 1000 is in mycon by way of an include it will still find it
18:49.34pooh_anthm: I have all the includes in mycon, but it will not find the exten, instead it will loop forever
18:49.59MikeJ_file[laptop]!!!!
18:50.03pooh_but then again, I am a real nOOb
18:50.06anthmyou dial login ext, agi answers, it knows your remote sip id so it inserts it into the db in the same table as your profile so you know ext 1234 is SIP/3.4.5.6
18:50.06clueconfile: report to room 996 for your meeting with the principal.
18:50.33pooh_anthm: I have all of that working
18:50.56pooh_anthm: even technology is set
18:51.02pooh_sip/iax2
18:51.23pooh_cause the dial command NEEDS the technology of a device
18:52.08anthma sip call reaction to the context= is not at all different from how the goto would have done it
18:52.09pooh_anth: I am trying to seperate sip/iax.conf from what a user can do on the system
18:52.33pooh_Goto requires a real exten or priority
18:52.36sivanawhen a file has * next to it, what does that mean?
18:52.50pooh_while my contexts has includes and no real extens
18:53.06anthmdoes goto yell at you saying it doesnt exist ?
18:53.39pooh_Nope it will loop. It will go to the right context, can not find it and loops, cause the fallback context says try again
18:53.48*** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
18:53.56file[laptop]nice dialplan logic.
18:53.59*** join/#asterisk santiago (~santiago@63.245.86.198)
18:54.09pooh_the dial command says to look for context,exten,prio
18:54.16pooh_while context only has includes
18:54.26pooh_no go
18:54.26bkw_oh file dear
18:54.35file[laptop]bkw_: oh go jump off a bridge
18:54.36bkw_can I see the pictures..  please please please
18:54.41*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
18:54.52bkw_I take it you had a bad night?
18:54.52bkw_NOT DIGITAL?
18:54.56file[laptop]no, not digital
18:55.03fugitivowhy i get "sorry, your call can't be completed at this time.... " using broadvoice? any idea?
18:55.05file[laptop]I haven't slept in over a day
18:55.09MikeJ_what kind of geek are you?
18:55.15MikeJ_sad... so sad.
18:55.17anthmthis is 1.07 ?
18:55.26pooh_ye
18:55.28pooh_syes
18:55.32file[laptop]okay people, answer me this... I'm in the prom stuff, how was I supposed to take pics? :P
18:55.32pooh_damn, yes
18:55.50file[laptop]clone myself perhaps...
18:56.15MikeJ[Laptop]yes
18:56.17MikeJ[Laptop]ummm
18:56.23file[laptop]they did take pictures
18:56.26MikeJ[Laptop]yes
18:56.33MikeJ[Laptop]well, we want to see them
18:56.36file[laptop]my mother doesn't like the digital camera because she likes real prints, and they're supposedly expensive
18:56.41Ariel_file[laptop], did you get lucky?
18:56.42file[laptop]so shutup before I hunt you down and kill you
18:56.47file[laptop]I'm not in the mood to bitch about pictures
18:56.49MikeJ[Laptop]:P
18:56.56MikeJ[Laptop]ok
18:57.00MikeJ[Laptop]your still loved
18:57.03yaaardoes anybody have any idea what would cause * to say "could not create channel of type Zap'" when I try to execute something like 'exten => s,1,Dial(Zap/1/${EXTEN})'               ?
18:57.03*** join/#asterisk Derkommissar (Derkommiss@241.sub-166-139-71.myvzw.com)
18:57.08DerkommissarHello
18:57.20*** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be)
18:57.20clueconyaaar: do you have a Zap channel 1 defined?
18:57.29MikeJ[Laptop]yaaar, it's inability to create a zap Channel
18:57.53yaaarMikeJ[Laptop]: gee, that's brilliant
18:57.58MikeJ[Laptop]~lart cluecon
18:58.02MikeJ[Laptop]thanks
18:58.57MikeJ[Laptop]~lart cluecon
18:59.09pooh_anthm: so, goto a context that has all kinds of includes will not recognise the overall available extensions from the includes
19:00.46*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
19:00.51*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
19:03.05Derkommissarim having trouble compiling oh323
19:03.12Derkommissaris there anyone here that can help me ?
19:04.20*** join/#asterisk Darien (~darien@office.mvcard.com)
19:04.25Darienhi all, some questions on IAX trunking
19:04.37DarienI'm reading the IAX2 bandwidth page on voip-info.org and reading about codecs
19:04.50Hmmhesaysyeah Derkommissar: use debian install the packages
19:05.10Dariennow, the setup we're going to have is phones -> local PBX -> remote call router
19:05.41Darienit looks like ILBC is the most efficient in terms of bandwidth, but I'm wondering what drawbacks it has
19:05.43anthmare you sure, there is no code in goto to enforse the existance of the destination
19:06.22Darienwait, I misread that
19:06.31DarienHmmhesays: was that to me?
19:06.38HmmhesaysDarien: yeah
19:06.40Darienyeah
19:06.51Darienwe have one location in India
19:06.54Darienbut most of our clients will be here in Montreal
19:07.06DarienI know
19:07.12*** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr)
19:07.22Hmmhesayshow many calls are you talking about?
19:07.22jhiverhi all
19:07.23Hmmhesayssimultaneous
19:07.34Darientwenty at the most
19:07.40Darienat least, for the time being
19:07.51Hmmhesaysand you are conerned enough about bandwidth to use iax trunking?
19:08.24Dariennot really bandwidth
19:08.26Darienbut I don't know what the state of broadband is in India
19:08.26harryvvthere should be a channel on realestate thats where alot of the money is being made at :)
19:08.28Hmmhesaysbuy 20 g.729 licenses and call it good
19:08.40harryvvdarien, bandwith is not a issue in some parts of india.
19:08.45Darienok
19:09.01Darienit's probably safe to assume that his call centre is located in a not-an-issue part
19:09.27Hmmhesaysyou putting asterisk boxes at each location?
19:09.28DarienHmmhesays: is g.729 worth paying for?
19:09.31DarienHmmhesays: yes
19:09.36Darienand each will trunk to us
19:09.39harryvvask him if there is alot of high tech buildings where he is at. if not and its poor area then no chance for good bandwith.
19:09.45HmmhesaysDarien: it's not worth it if you don't need it
19:09.58Darienwell, I can just ask him what he has for an internet connection
19:09.58DarienHmmhesays: that's what I thought, and it doesn't look like I need it
19:10.02jhiverg.729 give you good quality for little bandwith
19:10.10Hmmhesaysif you can get away with ilbc then use it
19:10.10DarienGSM should be fine though for what we're doing
19:10.17Hmmhesaysgsm does't sound quite as good
19:10.19Darienmm, it says it's robotic-sounding
19:10.56HmmhesaysIMHO ilbc sounds better
19:10.57Darienbut I'll use whatever gives toll-quality sound for less than a megabit per 20 callers
19:10.58harryvvwhat do cell phones use mostly?
19:10.58jhiverif you can't use g.711 then really g.729 is second best...
19:11.10Hmmhesaysi dunno, there are some intermediate codecs that sound pretty damn good
19:11.12jhiverplus it only uses 24 kbits/s (using SIP)
19:11.25*** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl)
19:11.32Darienthe thing is, the sites are going to be using vicidial/astguiclient
19:11.34pooh_back, connection problem
19:11.41Darienwhich I guess can do 30 callers per 3 ghz machine
19:11.43*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
19:11.54pooh_anthm: send me a message in the last 5 minutes?
19:12.11*** join/#asterisk loick (~loick@APuteaux-151-1-27-74.w82-124.abo.wanadoo.fr)
19:12.40Hmmhesaysever listen to g.726?
19:12.49jhiverI didn't no
19:13.03DarienI've never heard any of it (or at least, I don't know what I've heard)
19:13.04jhiverI tried g.711, g.729, GSM, iLBC
19:13.11Darienthe way we're doing billing is to basically modify astcc to do it, and use the trunk username as the account number
19:13.21Darienso we're using trunking regardless
19:13.22Hmmhesaystry it out sometime
19:13.25anthmare you sure, there is no code in goto to enforse the existance of the destination
19:13.28HmmhesaysDarien: why are you using trunking?
19:13.40jhiverg.711 is excellent but you need so much bandwith!
19:13.53harryvvany of you using the ip500
19:13.59DarienHmmhesays: bandwidth efficiency and to have a single username/connection to watch for
19:14.03Darienwhich I guess would be the case anyway
19:14.07jhiverg.729 sounds almost as good (except for some 'f' that sound like 's' with certain voices)
19:14.12HmmhesaysDarien: indeed
19:14.22Darienanyway, bandwidth is enough reason for me
19:14.27Darienwe do have to pay for it after all
19:14.32HmmhesaysI thought you said bandwidth wasn't an issue
19:14.36Darienwell
19:14.42Darienit's not a technical issue
19:14.49Darienit's a financial issue
19:14.55HmmhesaysI didn't ask if it was a technical issue
19:14.59Darienand we're going to have a machine at their location anyway
19:15.13*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
19:15.14Darienso unless there are downsides to trunking....?
19:17.13ManxPowerin 1.0.x trunking breaks the iax2 jitterbuffer
19:17.23Hmmhesayslast I read about it was quite some time ago, and then it had jitter problems
19:17.47clive-manx, i belive cvs head that is fixed
19:18.03ManxPowerclive-, *nod*  that's why I said "in 1.0.x" 8-)
19:18.56clive-manx, how stable is CVS head at the moment >?
19:19.38*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
19:20.20Hmmhesayswhat is head at now?
19:20.56clive-I am waiting for 1.2  but its taking its time appearing
19:21.01Darienhmm
19:21.11Dariendo you guys think GSM would be sufficient for a call centre environment?
19:21.21Darien100% clarity is not a requirement, nor likely an expectation
19:21.36clive-Iuse GSM, and its fine
19:21.43Dariengood enough for me
19:21.43*** join/#asterisk torment0r (~torment0r@md-frdrck-cmts3a-b-222.shphwv.adelphia.net)
19:21.48clive-I use,.,,,
19:21.56DarienI have another question about PRIs, which maybe someone can answer
19:22.09Darienor rather, long-distance in general
19:22.17DarienI'm looking for the best deal I can get on North American long-distance
19:22.22DarienAmerican mostly
19:22.36Darienright now we have 0.9 cents/min, but does anyone know any cheaper/better rates than that?
19:22.53torment0rI have an X100P card, and for some reason at bootup the status is in RED Alarm
19:23.07torment0rwhat could cause that?
19:23.12rephormtorment0r: have you configured it with ztcfg?
19:23.19torment0ryep
19:23.33rephormand you have the t1 plugged into it?
19:23.34enderDarien: CHeck out timewarner.
19:23.41Darienender: ?
19:23.44torment0rit's just a single card, i have fxsks=1
19:23.53torment0rin the /etc/zaptel.conf
19:23.56torment0rthen i run ztcfg -vv
19:23.57Darienender: for commercial dial-out ?
19:24.11enderDarien: we got a really good deal on a T1+PRI line (bonded, PRI with higher priority) and really good long distance.
19:24.15rephormoh x100p. sorry
19:24.26Darienender: what kind of price, if you don't mind my asking?
19:24.49DarienI'm just looking for a ballpark
19:24.50enderDarien: TW has a nifty little box that as you add T1 lines, you immediately get more bandwidth and you can choose how much of that bandwidth to allow for callout.  It's all balanced and shared on the fly
19:24.51rephormtorment0r: do you have a line plugged into it?
19:25.00enderDarien: for the line cost or just the long distance?
19:25.09torment0rno, does there need to be for it not to be in red alarm?
19:25.17Darienender: total
19:25.18rephormtorment0r: i believe so.
19:25.22enderDarien: and local long distance or long long distance?
19:25.24torment0rlemme give this a try
19:25.34pooh_back, damn ISP
19:25.35Darienender: lower 48 states
19:25.37enderDarien: actually nwo that I think about it, we used AT&T for our long long distance calling, TW for local long distance.
19:25.42Darienender: very rarely alaska/hawaii
19:25.52Darienender: it would just be for long-distance though, we don't have any physical presence in the US
19:26.06Darienso we'd be colocating in probably New York or Los Angeles
19:26.20enderDarien: for a 2T1 set and the PRI stuff, it was 1000 a month I do believe.  Tack on the AT&T on top of that.
19:26.43enderDarien: and that was with 200 Direct Dial numbers.
19:26.52torment0rinteresting.. now it's working.. awesome :)
19:27.08torment0ri have a vonage ATA box connected to an x100p box
19:27.17torment0rand it's working.. awesome
19:27.18Darienender: so that's 47 lines and local long distance for $1k?
19:27.36torment0rnow to test it with my cell phone
19:27.37Darienhow local is the local long distance?
19:27.53enderDarien: there wasnt' really 'lines' to speak of, just bandwidth.
19:27.57Darienhrm
19:28.20enderDarien: time warner took the phone stuff and turned it into voip to go from our office to their office, and then it went back to voice.
19:28.23Darienender: was this a digital telephony/VoIP solution?
19:28.24Darienaaah
19:28.26DarienI see
19:28.30*** join/#asterisk brettnem (~brettnem@207.90.232.34)
19:28.38enderDarien: but I think it was only 23 lines because we used the second T1 for internet bandwidth only.
19:28.40Darienwell how local was the local LD then?
19:28.46enderDarien: we could do any split we wanted with teh channels.
19:28.49Darienlower 48, or just in-state?
19:28.53enderDarien: I thin kit was just instate.
19:28.56Dariendamn
19:29.02Pete_LargoDarien, how many channels do you need and how many minutes of use per month?
19:29.11enderDarien: TW said they didnt' have good long long distance, the big boys like AT&T were better with that.
19:29.21Darienender: noted
19:29.46DarienPete_Largo: 20 channels (at the moment; needs to be quickly expandable) and our first client is looking about about 480,000 minutes/month
19:29.49*** join/#asterisk wozto1s (~wozto1s@host81-155-20-99.range81-155.btcentralplus.com)
19:30.16Darienwell, he'll be starting at 144,000 but he's going to move up to 480,000
19:30.31Pete_Largo~ (24*60)*30
19:30.31jbot43200
19:30.31*** join/#asterisk Meaty (~cp_simbul@office.abi.ca)
19:30.46Darien?
19:30.47Pete_Largo~43200*20
19:30.47jbot864000
19:31.05Pete_Largojust using jbot for some math :)
19:31.07Dariennod
19:31.31Pete_LargoDarien, are you going to colocate or will it be your location?
19:31.48DarienPete_Largo: we have a colocation for the central call routing server
19:31.53Darienall the calls will be going through that
19:32.09Pete_Largowhere is your colocation located (what state/city)?
19:32.13*** join/#asterisk flotox (~jovan@host52-75.pool80183.interbusiness.it)
19:32.31Robot_~2^2
19:32.31jbot2^2 is, like, a bad question.  You want 2**2.
19:32.31DarienMontreal,QC
19:32.43Robot_~2**2
19:32.43jbot4
19:32.48Robot_~2**10
19:32.49jbot1024
19:32.51Darienhowever, if we can save money, we will open up other call centres and put colocations there (e.g. New York, Los Angeles)
19:32.53Robot_~2**32
19:32.54jbot4294967296
19:32.58Robot_~2**1000
19:32.58jbota number with quite a few digits...
19:33.05Robot_~2**100
19:33.05jbota number with quite a few digits...
19:33.12Robot_~2**64
19:33.12jbot18446744073709551616
19:33.19Darien:/
19:33.34DarienPete_Largo: why do you ask?
19:33.40h3x0ryou cant fit 480k minutes in 20 channels
19:33.41Pete_LargoI found a LA colocation for approx 550 month, in which we can get a single full PRI for approx 250 month
19:33.48Pete_Largoor a DS3 of PRI for 1500
19:33.51Pete_Largomonth
19:34.04h3x0rhttp://www.carrierone.net
19:34.08brettnem480k is a few t1s if I recall. I think 140k is about 1
19:34.12Darienh3x0r: 60 minutes per hour times 20 hours per day times 20 days per month times 20 channels
19:34.13Pete_Largowhat kind of per minute rate do you want?
19:34.17brettnemdepends on your application of course.
19:34.24wozto1sevening....don't suppose anybody has experience of using a TDM card with BT in the UK do they?  I'm NOT a newbie btw and my setup IS working, but the TDM card isn't quite doing what it should!
19:34.27DarienPete_Largo: less than $0.009/min USD
19:34.40h3x0rDarien: well, that is including non billable time i guess
19:34.59Darienh3x0r: yeah, it's 20 stations, 20 days, 20 hrs/day
19:35.03Darienwhich is a lot
19:35.04harryvvDarian, from where to where for that price?
19:35.24h3x0rwhat, are you having people call in or something?
19:35.48Darienharryvv: from Montreal, QC to the US (lower 48 is fine), or if the savings are significant enough, we can set up a colocation in the US (NY/LA) to handle it as well
19:35.50Pete_LargoDarien, is it all outbound or inbound or mixed?
19:35.51brettnem20 hrs/day.. you got migrant workers or something?
19:36.08Darienbrettnem: 20 hrs/day per station, probably 6 hour shifts or something
19:36.14DarienPete_Largo: my impression is outgoing only
19:36.14brettnemah
19:36.29Pete_Largolet me make a quick phone call...
19:36.30Darienor 10 hour shifts, I dunno what normal working days are in India
19:36.32Darienok
19:36.53h3x0ranyway like brettnem said, you cant really bill more than 150k MMOU per T1
19:36.56h3x0rif you arent using it 24/7
19:37.02h3x0rif you were using it 24/7 then like 250K MMOU
19:37.06h3x0rer MOU
19:37.10DarienMOU?
19:37.16h3x0rminutes of usage
19:37.28Darienwhat do you mean, 'can't really bill more than' ?
19:37.31brookshirehttp://www.digium.com/index.php?menu=press/pr_2gen_firm
19:37.33brettnemyeah.. consider average call time and peak calling times
19:37.34Derkommissarcan anyone here help me with chan_oh323
19:37.48brettnemyou simply can't put more than 250k minutes onto the T1..
19:37.51h3x0rbecause, calls spend a lot of time ringing, busy, etc
19:37.54brettnemor somewhere around then
19:38.12brettnemor channels are just idle.. like a 4am
19:38.13Pete_Largoyou could put approx 480k minutes per month on a T1
19:38.16Pete_Largo24/7
19:38.24Darienwell, this is an approximate
19:38.25wozto1shope you havn't better luck than me Derkommissar...I didn't even get a hello :(
19:38.37h3x0ryeah if you left every call off hook
19:38.39Pete_Largo~ ((24*60)*30)*23
19:38.39jbot993600
19:38.39brettnemassuming a perfectly uniform calling pattern 24/7
19:38.41h3x0rbut how many uses are there for that
19:38.51Ariel_wozto1s, hello
19:39.02Darienwe won't know what the REAL numbers are until we start doing them
19:39.05brettnemyou got to engineer it for your peak calling volume
19:39.07Ariel_Derkommissar, hello how was your trip? And what's the problem?
19:39.16h3x0ri know what the real numbers are
19:39.16h3x0rive sold like hundreds of t1s
19:39.20DerkommissarIt was good :)
19:39.23h3x0rive never seen anybody do more than 250k a t1 before ever
19:39.26Derkommissarh323 hates me:(
19:39.29Pete_Largolike hundreds?
19:39.35brettnemyeah, there is pretty good historical data for T1 usage
19:39.39Derkommissarbut the trip was good
19:39.39*** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com)
19:39.45_-Jon-_Hey
19:39.47brettnemDerkommissar: don't feel bad, h323 hates everyone.
19:39.48Ariel_Derkommissar, did you not allready have this working before?
19:39.51wozto1so right...don't ask about my problem then! :)
19:39.55_-Jon-_Anyone else here having problems registering with BroadVoice?
19:39.57brettnemDerkommissar: you know, just yesterday, I found out that chan_sccp hates me
19:40.48Darienso what you guys are saying is that thanks to busy signals, etc. we probably won't hit 480k minutes of usage, right?
19:40.53brettnemwozto1s: you'll get a better response with a hello, then a question, and some active participation
19:40.56Derkommissarsccp ?
19:41.22brettnemDarien: it's more like typical usage.. you'll peak out durning the day, taper off at night..
19:41.23Derkommissarhere is the errors
19:41.24Derkommissarhttp://pastebin.com/302627
19:41.30Pete_LargoBroadvoice is working ok for me right now...
19:41.32Darienbrettnem: probably, yeah
19:41.33wozto1si did say evening...which is short for 'Good Evening' to my knowledge...and I don't know anything about what your talking about so how take I take part?????
19:41.36brettnemDarien: lots of ringing and call failures.. no anwers..
19:41.47Dariennod
19:42.18Darienlike I said, we won't know real numbers until we start routing his calls
19:42.20brettnemit's unlikely that you'll use more than 250k per T1 unless you are doing something weird.. like not talking to PEOPLE but machines.
19:42.26h3x0ronly in vegas...
19:42.53brettnemh3x0r: how about being 100% humidity and not raining.. only in texas...
19:42.59h3x0rhahah
19:43.15brettnemwhat is 100% humiditiy anyway.. seems like you'd be underwater
19:43.29h3x0rit means the air has saturated as much water as it possibly can
19:43.43h3x0rand i guess thats relative to berometric pressure
19:43.50brettnemand tempreture
19:43.55h3x0rwell yeah
19:43.59_-Jon-_Hmm, I wonder why my registration is failing then
19:44.11h3x0rso its 100 degrees, 17% and raining go figure
19:44.21brettnemDerkommissar: sccp is Cisco Call Manager emulation
19:44.30Ariel_Derkommissar, your using asterisk head?
19:44.32*** join/#asterisk pifiu (~myassisbi@208.205.181.170)
19:44.41Derkommissaryes
19:44.57brettnemok.. back to work
19:45.20*** join/#asterisk wasim (~wasim@203.81.220.100)
19:46.48Ariel_Derkommissar, you downloaded all the PWlib's
19:48.17Ariel_also your not on FreeBSD are you?
19:48.30Darienhmm
19:50.05Derkommissaryes
19:51.05Ariel_Derkommissar, yes to which one?
19:52.12Derkommissari downloaded all the pwlibs
19:52.19Derkommissarits all compiled as the readme says
19:52.53torment0rw00t
19:53.00torment0ri got it working like a ninja
19:54.06*** join/#asterisk L|NUX (~linux@202.5.145.58)
19:54.44*** join/#asterisk cluecon (cluecon@wsip-68-99-73-32.tu.ok.cox.net)
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19:55.47Dariendoes anyone know what kind of a talk time ratio vicidial can get?
19:55.55Darienlike 40 minutes out of the hour, 50, etc?
19:56.43*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
19:57.03MrbBelvedri have asterisk set to start at bootup, and it works great
19:57.14*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
19:57.17shidoZzZZ
19:57.20shidoI wanna be
19:57.20MrbBelvedrnow, I want to make sure it never starts at bootup
19:57.21shidoI wanna
19:57.25shidoi wanna be , I wanna
19:57.30shidoI wanna , I wanna
19:57.34shidoI wanna be a hilton
19:57.49clueconwhat is the best way to resolve the issue of "this driver does not support 1st gen modules" with a te405p card?
19:58.23MrbBelvedrhow do i set it up to never start at boot?
19:58.47Darienremove the entry in rc.d or init.d
19:59.08*** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com)
20:00.03*** join/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net)
20:01.25*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)
20:02.26*** part/#asterisk wozto1s (~wozto1s@host81-155-20-99.range81-155.btcentralplus.com)
20:04.45clueconhas bug 4155 been resolved?
20:05.44jdv79is there a test suite for asterisk?
20:06.47*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
20:07.39*** join/#asterisk jmacz (~jmacz@63.245.86.234)
20:08.49*** join/#asterisk proti (~seb@calypso.frankengul.org)
20:09.34harryvvhas there been any bad issue between the ip500 and tftp or ftp server?
20:11.02Ariel_harryvv, I use the IP-500 via ftp. Works great
20:11.17lesouvagecluecon: I finmally found out what it was. With a debian sustem the settings are in /etc/festival and not in siteinit.scm I have it up and running now although it sounds a little bit loud.
20:11.38lesouvageand distorted
20:11.50*** join/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET)
20:15.49*** join/#asterisk Umaro (~umaro@209.140.74.64)
20:15.56*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
20:16.00UmaroHey guys, I've got a pri.. how do I set the callerid to private?
20:16.20eKo1callerid=private?
20:17.15MrbBelvedrso if i remove the file called "asterisk" from the init.d directory, it will not start at boot?
20:19.08*** join/#asterisk Tili (~Tili@202-133-67-239-dialup.sat.net.pk)
20:19.39*** join/#asterisk Netview (~Netview@p54AF8279.dip0.t-ipconnect.de)
20:20.06rephormMrbBelvedr: what os are you on?
20:20.24MrbBelvedrrechat - centos
20:20.34MrbBelvedrredhay
20:20.37MrbBelvedrfucj
20:20.44rephormMrbBelvedr: man chkconfig
20:20.49*** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net)
20:21.03UmaroeKo1: so like SetCallerID(private) would work?
20:21.25rephormMrbBelvedr: 'chkconfig asterisk off' will stop it from loading on boot
20:21.49NethabSet(CALLERID(name)=private)
20:22.15jhiver~seen p0lar
20:22.15jbotp0lar <~p0lar@64.254.225.62> was last seen on IRC in channel #asterisk, 95d 23h 45m 21s ago, saying: 'time to order..hehe'.
20:22.34jhiver~seen Zeeek
20:22.34jbotzeeek <~Zeeek@Zeeek.sustaining.supporter.pdpc> was last seen on IRC in channel #asterisk, 15d 3h 23m 20s ago, saying: 'good luck'.
20:23.04MrbBelvedri did chkconfig --del asterisk
20:23.28MrbBelvedrso that will completely remove the startup script right?
20:25.10wizhippojust the links in the runlevels not the init script
20:26.48UmaroNethab: huh?
20:27.03Nethabthe new func_callerid.so stuff
20:34.20*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
20:35.52Silik0nanyone remember about how long ago ast_request_and_dial() changed?
20:39.21*** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net)
20:39.37torment0rin the extension.conf, how do you get the caller id information
20:41.08rephorm${CALLERID}
20:41.20torment0rthat's it?
20:41.34torment0rexten => s,1,${CALLERID}
20:41.40torment0ror whatever
20:41.44MikeJ[Laptop]torment0r, no, the rest are in readme.variables
20:42.04rephormwell, if you just want to see it in the cli, do a Noop: exten => s,1,Noop(${CALLERID})
20:42.32torment0ri'm actually trying to use it from an AGI script, but i think that will work
20:42.38torment0ryou r0x0r, thx :)
20:44.04Cresl1nr0x0r is pretty cool
20:44.10rephormtorment0r: from agi, you can do a GET VARIABLE to get CALLERIDNUM and CALLERIDNAME
20:44.16Cresl1nd0 u sp3ak in 1337?
20:44.16*** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net)
20:44.19Cresl1n:-p
20:44.31Silik0nnac i fi tahw
20:44.51*** join/#asterisk greg_work (~greg@d221-73-237.commercial.cgocable.net)
20:44.52Silik0ntsetseb eht si kaeps 73313
20:44.53*** part/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET)
20:45.00Silik0n31337 sP[-@k !S +h[- |3E$T[-S~|~
20:45.05torment0rlol
20:45.44Silik0neVeN If !+ DOEs dr1ve 5()|\/|3 PeOPLe tO 1N$TA|\|!tY'S |)0OR S~|~Ep
20:45.52Cresl1nyeah ;-)
20:45.55Cresl1nthat's the problems
20:46.02NivexSSdsbCBzZWUgeW91ciBsMzN0IGFuZCByYWlzZSB5b3Ugc29tZSBiYXNlNjQu
20:46.03Silik0nheh
20:46.20jontow$foo
20:46.23brimstonehey, that's an idea
20:46.29brimstonelet's md5 encode everything we say
20:46.46`SauronThat looked like base64
20:46.48Silik0nok so does anyone know what bug/patch went in for changing the ast_request_and_dial() call?
20:47.12brimstoneSilik0n: a little cvs on diff action will tell ya
20:47.12pifiuhey guys wasup
20:47.23pifiucan anyone recommend a good stable motherboard for an asterisk pbx?
20:47.34Nuggeta powermac.  :)
20:47.53brimstoneasus?
20:48.18pifiuthose are stable for a pbx?
20:48.24pifiujust a standard one?
20:48.38pifiuor would you guys recommend like an entry level intel server board?
20:48.52develpifiu, we're running an intel board, regular desktop.  rock solid.
20:49.00pifiuwhich model?
20:49.25Nuggetif you plan to do any zaptel hardware at all, you'll want to make sure the bios lets you diddle with interrupts.
20:49.41`SauronI finally beat my close card into submission
20:49.43Nuggetif no zaptel is planned it doesn't matter at all.  just buy something stable
20:49.50`Sauronyeah, it's lame to use a clone card, but it works
20:49.52develpifiu, i'd have to open the box back up to tell you.
20:49.54pifiuyeah for this new setup I wouldnt need zaptel anymore
20:50.04Nuggetthen get a powermac or an xserve  :)
20:50.06pifiulol
20:50.09`SauronI didn't diddle interrupts, though. Linux did it for me.
20:50.32pifiuand what is asterisk heavy on when designing a new system?
20:50.33NuggetI had to turn off some of the onboard crap on my home machine because the bios kept wanting to share the interrupts
20:50.40pifiuwould need to be handling like 5-10 calls at a time
20:50.50pifiuactually correction. 5 calls each server
20:50.56pifiumemory? cpu? cache?
20:51.28*** join/#asterisk dos000 (~dos000@66.11.173.123)
20:51.54`SauronNugget, what's dB's preferred addressed name online.. dB or his irc nic?
20:51.56`Sauronnick
20:55.01*** join/#asterisk `Sauron (sauron@h-69-3-12-50.hstqtx02.covad.net)
20:55.06`Sauronhum, that didn't work
20:55.23*** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
20:55.26Nuggetdecibel or thecougar.
20:55.31NuggetI know he's bitched that decibel is taken here.
20:55.43NuggetI can't keep track of all his fscking nicks.  it's confusin
20:55.48Nugget:)
20:56.05NuggetI wrote an irssi script that sprays privmsgs to all of them, that way I don't have to keep track
20:56.19Nuggethttp://slacker.com/~nugget/stuff/irssi/scripts/decimessage.pl
20:58.12pifiuok update now
20:58.25pifiui will need ONE zaptel card to be able to dial 911 in case of emergencys
20:58.36`Sauronrofl
21:00.27`Sauronso irssi is a commandline client, huh?
21:00.30`SauronIIiiiinteresting.
21:01.42*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
21:01.53cochiirssi is like ircii or bitchx but cooler ;)
21:02.09*** join/#asterisk bitslave (~psolomon@69-165-217-96.atlsfl.adelphia.net)
21:02.13`SauronI hate bX
21:02.19`SauronI <3 ircII
21:02.26`SauronI might have to try out irssi
21:02.27cochiirssi is cool.just give it a tr
21:02.27*** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu)
21:02.30cochiy
21:02.42cochiused to chat with the developer for like 1.5 years back then. nice guy
21:02.51Romik_anybody can advice about IAX2 server <-> server encryption?
21:02.51`SauronIs there a way to convince it to run ircII scripts? :)
21:02.59`SauronI might have to check out new versions of epic too
21:03.26Nuggetirssi is the best.
21:03.43Nuggetit does ssl natively, scripting in perl or tcl, solid multi-server support
21:03.44Nuggetit rocks
21:04.10`Sauronwe'll see
21:04.34cochii wrote a patch for converting the timestamp to swatch beats. dunno if timo included it somewhen
21:04.35file[laptop]Romik_: it doesn't work, it'll crash your box
21:04.39`SauronI'll have to figure out how to hack in all my mods to epic before I'll be able to use it fulltime
21:05.35Nuxiirssi aslo has php support.
21:05.41Romik_file[laptop]: any advice what work?
21:05.51cochidoes it have ruby, too?
21:05.56*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
21:06.41Nuggetsure, but using php for that is beyond stupid.
21:06.55cochiusing php is generally quite.. well.. anyway ;)
21:07.25Nuxias opposed to perl?
21:07.35Nuggetpeople who insist on using php for anything other than page-generation are falling victim to the "one tool in my toolbox" problem.
21:08.06cochiperl.. cough.. erm unqualified to give out any statements there ;)
21:08.27cochiphp is just ... ugly. a badly designed language
21:08.40cochiwith frequent unannounced api changes. thanks guys, had a lot of fun with these
21:08.43Hmmhesaysphp isn't that bad
21:08.52`Sauronit's not bad
21:08.52PatrickDKcochi, it may be, but it's still quick to code in
21:08.58DannyFNugget, each to his own...
21:08.59`Sauron<PROTECTED>
21:08.59Nuggetsure, but it was never intended to be a general purpose language and it shows.
21:09.01jontowyes; but the reason for that as i see is the quickly ballooning ability of the language
21:09.03PatrickDKbut that does make it pretty ugly
21:09.06cochiand very error-prone. and not really straight forward
21:09.12`SauronI like the combination of php for web, perl for non-web - in applications.
21:09.13cochithat's the reason why i love Ruby
21:09.31PatrickDKI don't like ruby
21:09.39PatrickDKheh, why yet another language
21:09.55NuxiWhy does everyone always insist that every application has a single language and every language has a single application?
21:09.56cochiit's not easy to get used to it. but the oop stuff in there is so clean <3
21:10.00Nuggetnobody's said anything religious.
21:10.06cochiyet
21:10.07cochi;)
21:10.09DannyFyet :)
21:10.12DannyFlol
21:10.13cochi*g*
21:10.19cochithat just -had- to happen ;)
21:10.27DannyFyupp
21:10.27PatrickDKI just find it annoying to have 25 different compilers on the system is all :)
21:10.28X-Robuh. Praise Larry!
21:10.38Hmmhesayscoming from the author of res_php, lol
21:10.58Nuxijavascript is only for web browsers.
21:10.58DannyFPHP grew on me ;) I like bkw's res_js it's cute as hell
21:11.02jontowas i went about smackin' people down yesterday about the distribution wars.. each language is useful for something or other.. use what you're comfortable with and don't argue :)  [im being gentle today 'cause i got too much to do] :D
21:11.21Nuggetit's not a matter of comfort, it's a matter of appropriate use.
21:11.35jontowwhen i program something; if it takes 3 languages to get a task done efficiently and effectively; then i will use 3 languages
21:11.37cochiwell don't get me wrong. i used to be a pro php developer since 1997 now. working with it every day, writing huge amounts of code. but that just showed my its deficies ;)
21:11.38Nuxiperl is the only language!
21:11.42Nuggetletting comfort dictate language choice is what leads to people writing scripts in php.  :)
21:11.57tclarkNuxi: http://pbxfreeware.com/res_js.tgz
21:11.59Nuxi* should have been written in perl!
21:12.01Nugget(or trying to write full apps in perl)
21:12.02X-Robperl is the one true language
21:12.03DannyFNugget, although it's quite convenient if you have stuff that dont require 1000 calls / sec to re use PHP code u have laying about
21:12.05jontowmaybe so, but if that is what they're good with.. that is what they're good with
21:12.08`SauronJust listen to nugget, people.
21:12.13X-Robanyone who says difference is a heretic.
21:12.15`Sauronhe knows that he talks about ;)
21:12.17cochiwasnt freepbx written in perl, nuxi?;)
21:12.30dos000is there an /etc/init.d/asterisk for autostarting asterisk .. i am googling and i see somewhere it mentions is part of the source but i could not find it.
21:12.39DannyFdos000, safe_asterisk
21:12.44Nuxilinux should have been written in perl to run on windows!
21:12.56dos000DannyF, was that ? is it in the source ?
21:13.01X-Robnuxi, linux should have been written in vbScript.
21:13.04X-RobDoh.
21:13.08DannyFcreated when u do make install
21:13.09jontowyeah, with activeX
21:13.15Nuxidos000 look in the redhat folder for an init script.
21:13.17cochilinux shouldnt have been written.. but been -designed- ;)
21:13.43Nuggetheh
21:13.56DannyFnot much on the net today technology wise that didnt suck initially ;)
21:14.03Nuxiwe should write laws against writting in anything but perl.
21:14.06`Sauronyou're seeing the movie, right?
21:14.08Nuxiperl should be written in perl.
21:14.08dos000Nuxi, i am on debian ... DannyF where does it put it.
21:14.11cochidanny really not? read the Geforce7800 reviews already?
21:14.18`SauronAck, ignore that here. :)
21:14.19cochiwasn't pascal 7 written in pascal 7 ? *mh*
21:14.29cochidos on debian? /etc/init.d
21:14.34Nuxidos000 dl the source and it is in <*src>/redhat
21:14.41cochigotta enable startup in the /etc/defaults anyway
21:14.42DannyFdos /usr/sbin/safe_asterisk usually
21:15.10*** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
21:17.32`Sauronalright
21:17.48dos000Nuxi, hopefully this is ok in debian based systems ?
21:18.54DannyFNuxi, same thing diff name
21:21.11NuxiI know, I know I should be running it on windows with http://www.asteriskwin32.com/ lol
21:27.30pifiuanyone running an intel entry level server board for asterisk?
21:27.34pifiua 775 basd one?
21:30.52harryvvmmm my ip500 is failing to register against the asterisk box.
21:31.08*** join/#asterisk Topslack (~topslakr@24-53-13-169.lndnnh.adelphia.net)
21:31.37fugitivodamn
21:31.58fugitivoall day, trying to figure out why broadvoice wasn't working, and it's because videosupport=yes
21:32.02harryvvwhat fug
21:32.11harryvvreally
21:32.21harryvvthat should not matter
21:32.28fugitivoit does!
21:32.40rephormharryvv: how are you configuring it? (xml files or web interface?)
21:33.50fugitivoi'm going to put that on the wiki
21:33.55FuRR_anyone know where i can find an emulator for cisco ip phones
21:34.08FuRR_or does their SoftPhone support their XML applications
21:34.34harryvvwell, the initial ip and what was on the phone has been done. astrisk sees the ip address trying to register against it but its failing. I have gone into the web interface in register and only provided the basics on the first block of information.
21:34.45Nuxiwoohoo documentation going on to the wiki!
21:36.11harryvvFurr, I have theidentification block filled out but is the Server 1 really needed?
21:36.50fugitivois this a bug? broadvoice not working when videosupport=yes ?
21:36.52harryvvalso I am assuming the address in the identification is asking for the asterisk server ip?
21:37.17rephormharryvv: set 'address' to the name in sip.conf
21:37.36rephorm(identification address)
21:37.41harryvvohh
21:38.15Nethabbroadvoice works fine and i have videosupport=yes
21:38.22Nethabbut i don't have a video phone
21:38.32harryvvname as in regexten or username ?
21:38.39fugitivoNethab: me neither, but i comment the line videosupport=yes, and it works
21:38.39harryvvor username?
21:38.44fugitivoNethab: what asterisk version?
21:38.54*** part/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
21:39.03Nethabi use todays HEAD
21:39.10*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
21:39.11fugitivomaybe it's fixeds
21:39.15fugitivoit doesn't work on 1.0.7
21:39.21Nethabbut it's always worked with videosupport since january
21:39.38Nethabi just leave it on cause i thought it was cool
21:40.19fugitivoi don't really need videosupport, so i'll leave it commented until i upgrade to head
21:40.51rephormharryvv: username
21:40.57harryvvk
21:41.20harryvvseems a little confusing why it would ask for a address in the registration
21:41.23harryvvof the ip500
21:41.44rephormharryvv: yeah. the polycom wording is different from asterisk wording
21:42.13rephormits a sip thing. the full address of the phone is SIP:username@serverip
21:42.23harryvvi see
21:42.29rephormor, in polycom terms, SIP:address@serveraddress
21:42.37harryvvand the point of having server 1 ?
21:42.50rephormserver1 is the asterisk box
21:43.42harryvvfantastic its working
21:44.05harryvvThanks rephorm :)
21:44.56rephormnp
21:45.02*** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net)
21:45.15jdv79call failed to go through, reason 8 - is that common?
21:46.01jdv79no!
21:47.19pjzhow do I check to see if my zaptel cards are working correctly?
21:47.43pjzah, hrm, nm
21:47.47pjzit looks like they're not configred right
21:47.53harryvvwhat do you have
21:47.53pjzI added a tdm400 to my existing te110p
21:48.02pjzand now neither is setting up correctly
21:48.08pjzit looks like maybe the tdm is being found first?
21:48.22pjzwhat should my zaptel.conf look like?
21:48.32pjzI've got teh stuff in it for the te110p, what do I need for the tdm400 ?
21:49.34pjzand how do I tell which one is providing which channels?
21:50.03*** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu)
21:51.47pigpenanyone have anything good or bad to say about voipjet.com?
21:52.33file[laptop]yes.
21:52.37VoIpMasterdoes anybody knows a good instruction (step by step) for isntalling Asterisk REALTIME ??
21:52.41pigpenok..which one.?
21:52.51file[laptop]yes.
21:52.56pjzanyone?
21:52.59pjzanyoen able to help me?
21:53.28rikstai wish people would get the fuck off the -users list writing useless shit
21:53.28pigpenfile[laptop], hmm...good or bad?
21:53.36pigpenI am hoping good.
21:53.37file[laptop]pigpen: see mailing list.
21:53.44pigpenpjz, not me.
21:53.53pigpenhmm..ok..
21:53.57*** join/#asterisk jaike (~a@203.131.137.76)
21:54.11pigpena hint?
21:54.26file[laptop]nope, no hint
21:54.34file[laptop]let all the posts be your help!
21:54.36VoIpMasterdoes nobody habe experience in working with asterisk REALTIME?
21:54.39file[laptop]thus you can make your own decision
21:54.51pjzanyone know how to set up both a te110p and a tdm400p in the same box?
21:55.28jaikehi. newbie here. if i wanted to enable blindxfer and atndxfer, i enable it on features.conf and add Tt to Dial?
21:55.53harryvvwhat codec is ulaw anyway
21:55.55jaikeexten => 312,1,Dial(SIP/312,20)
21:56.05jaikeexten => 312,1,Dial(SIP/312,20,Tt)   ?
21:56.29file[laptop]jaike: using # yes
21:56.35pigpenfile[laptop], ok..I see some concerns...but nufone had issues too...are they doing better?
21:56.41file[laptop]er well using the internal asterisk mumbo jumbo transfer thing
21:56.47file[laptop]pigpen: not as many
21:56.47VoIpMasterokay, another way, does anybody here have experience with asterisk and MySQL .. ? or know a easy to use how to ?
21:57.01file[laptop]VoIpMaster: there's a guide on voip-info.org I do believe
21:57.23jdv79hmm, 404 not found for a sip call
21:57.44pigpenfile[laptop], well, I guess I will give nufone another chance.  I think they have a better reputation...
21:57.55file[laptop]pigpen: or try http://www.asterlink.com/
21:57.57VoIpMasterreally ? file[laptop] ... i can't find it today when i'm searching for it on the website ...
21:57.58jdv79Channel:  SIP/bla@10.10.10.10 - does that looks right?
21:58.13file[laptop]jdv79: if you're not using a peer entry for authentication, yes
21:58.26file[laptop]VoIpMaster: are you gonna make me look?
21:58.46jdv79why would i get a 404 then?
21:58.53jdv79bla exists in sip.conf on 10.10.10.10
21:59.10file[laptop]VoIpMaster: type realtime config in for a search string :p
21:59.22jdv79even so, shouldn't it not go 404 and instead go right to the context as defined in the general section of sip.conf?
21:59.28file[laptop]jdv79: you don't dial a peer entry, you dial an extension in extensions.conf
21:59.32twisted[work]WORD
21:59.32twisted[work]i have the most awesomest warning message evar
21:59.40twisted[work]it reminds me of BKW at astricon atlanta last year
21:59.42file[laptop]twisted[work]: ...
21:59.42twisted[work]Jun 22 16:43:05 WARNING[13326]: chan_sip.c:1026 __sip_pretend_ack: Have a packet that doesn't want to give up!
21:59.47file[laptop]lol
21:59.51file[laptop]that's a new one
21:59.51pigpenfile[laptop], hmm..well, I already have a pri, just looking for good reliable cheap long distance
21:59.57twisted[work]file[laptop], that's what I said
22:00.04VoIpMasterokay, i will try that tomorrow ... thx the first .. i will came back to you when i can't find a good howto :)
22:00.21jdv79oops
22:01.54*** join/#asterisk jief- (~jief@digitalized.ca)
22:01.56file[laptop]darn I helped, what was I thinking
22:02.02jdv79can i not do a SIP call without finding a valid extention?
22:02.02harryvvpigpen what did your pri cost you
22:02.10*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
22:02.14file[laptop]jdv79: correct
22:02.17VoIpMasterokay guys i wish all here a nice night .. open your brains to bring the nicest PBX up !!!
22:02.27VoIpMastercu ... we see us tomorrow!
22:02.28jdv79then what is the context entry in sip.conf for?
22:02.33jief-hello, im going to deploy a call center soon with *. I was wondering, is it possible for agents to have not-available codes? like 611 bathroom, 612 on break, etc ?
22:02.39file[laptop]jdv79: actually your question confused me, rephrase it
22:02.43file[laptop]I'm not awake too much
22:02.52pigpenharryvv, $175/mo
22:02.55harryvvji, as in a meeting or on break ?
22:03.03jdv79i want all sip session to terminate in a context - that's all
22:03.05jdv79just for testing
22:03.13file[laptop]jdv79: you can send them to a default context, sure
22:03.16harryvvt-1 pri 24 channels for 175 a month?
22:03.22jdv79i tried but i get a 404
22:03.26jief-harryvv: yes like what you can get with Symposium for example
22:03.30file[laptop]404 means the extension doesn't exist
22:03.33pigpenyep...and for $20 I got 100 did's
22:03.44shidook
22:03.45pigpenand yes..it is inbound/outbound.
22:03.46harryvvji, polycom ip500 has those features in the phone.
22:03.48shidoits FOOOOOOOOOOd time
22:03.48shidobrb
22:03.49file[laptop]if you do a sip debug on the other box, it'll say where it's looking for the extension (which context) if you wanna make sure
22:04.02jdv79ok
22:04.05harryvvpigpen who gave you that deal?
22:04.13jief-harryvv: ok, so its a phone related feature and not relative to the pbx?
22:04.15file[laptop]but anyway, your issue is completely configuration issue
22:04.28harryvvji, dont know about that.
22:04.33pigpenMy box is located in a telco hotel.   I have over 600MB of bandwith with a telco...they like me.
22:04.34jief-harryvv: ok
22:04.53jdv79got it
22:04.56harryvvpigpen, obviosly!
22:04.58jdv79thanks file[laptop]
22:05.07file[laptop]jdv79: you're welcome
22:05.09harryvvpigpen, in a hotel? doing what
22:06.02wunderkinpigpen, what city?
22:06.18file[laptop]no one loves oily Homer!
22:06.26harryvvI have never heard of those rates for a t-1
22:07.03wunderkinmy ld t1 pri quotes were much less
22:07.08pjzif I have multiple zap devices, how do I tell which are which channels in zaptel.conf?
22:08.17harryvvwunderking, Only T1 rates I have seen here is 650-1,100 per month in BC canada.
22:08.51wunderkinwell this is also in a telco hotel
22:11.02InfraRedhttp://www.ananova.com/news/story/sm_1436341.html
22:12.27*** join/#asterisk meppl (mephisto@p54AAD58D.dip.t-dialin.net)
22:12.33*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-5-125.d4.club-internet.fr)
22:13.13pjzargh
22:13.30pjzdoes no one know how to deal with both a pri and a coupel fxo ports in teh same box?
22:13.42pjzthey're both zaptel, but I don't know how to write my zaptel.conf
22:14.53pjzsince it needs both span= and fxoks=
22:19.40harryvvI am impressed with the sound quality and lack of any echo on the polycom soundpoint ip500.
22:19.48jief-harryvv: Sprint has the best T1 prices in Canada
22:19.49harryvvintercom worked great.
22:19.59harryvvjief-:  ohh really
22:20.07jief-harryvv: as far as i know yes
22:20.16jief-harryvv: at least around montreal
22:21.41wunderkinsprint has had the best pricing for me here too, im getting quotes through another reseller, he usually gets a little better deals
22:21.46*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
22:22.04jief-yeah we are a Sprint partner, might be why we get a better deal though
22:24.15jief-i was attending a Cisco conference on VoIP today. they have neat products
22:25.07pawallsA "custom Lucent 5E" switch would be the "5ess" switchtype, right?
22:25.16pawallsjief-, Cisco has insane stuff.
22:25.22jief-which reminds me, does * have support for encrypted communications?
22:25.33jief-pawalls: CallManager + IPCC is kinda cool
22:25.41pawallsWhen I toured a Cisco facility in 1999 (9th grade), we got to use several of their different VoIP phones.
22:25.54pawallsThey have IP based cameras, TVs, a 10Gbit network, etc...
22:25.55TokyoJimuWhat's a good way to debug sip registrations?  I have several "register" lines in sip.conf but "sip show registry" shows nothing.
22:26.03pawallsEverything was being controlled from a wireless PDA.
22:26.06TokyoJimuI have sip debug on.
22:26.20pawallsHe was changing channels, repositioning the camera, starting a video conference.. wicked stuff.
22:26.21jief-pawalls: Cisco has mobile labs you can borough and use for tests
22:26.25Romik_anybody can advice about IAX2 server <-> server encryption?
22:26.50harryvvpawalls, i think that was the year cisco started to put out voip products.
22:27.08jief-harryvv: they started in '96
22:27.13pawallsharryvv, Yeah. I was taking CCNA courses. So we got to see the bleeding edge.
22:27.27harryvvohh as far back as then. I took the CCNA course in 1998
22:27.44*** part/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net)
22:28.15pawallsCisco is a pretty cool company.. somewhat evil at times, but they've got wicked technologies.
22:28.22pawallsAnd great product line.
22:28.33jief-yes and their conferences are cool
22:28.39jief-in their state of the art labs ;)
22:29.08pawallsAnyway, so my phone company says the switchtype on our incoming circuit is a "custom Lucent 5E", anyone know if that is the same as "5ess" switchtype ?
22:29.23TokyoJimupawalls: yes it is.
22:29.33pawallsTokyoJimu, Great, thanks :)
22:29.43pawallsFinally replacing a 15 year old Comdial POS.
22:29.51TokyoJimuComdial.  Yuck.
22:29.53pawallsThat machine is a nightmare...
22:30.02pawallsSadly all of our phones are proprietary Comdial Impacts.
22:30.08TokyoJimuAmerican technology at its worst.
22:30.12pawallsThey were 300$ a piece...
22:30.19pawallsAnd I think all ~200 of them are going to go to waste.
22:30.35TokyoJimuWhat are you replacing them with?
22:30.36jief-at the office, it came up cheaper to buy all new SIP phones, and new Catalysts than upgrading our Lucent system
22:30.58pawallsWe're using softphones for most of our personnel now.
22:31.08pawallsHowever, I can't for the life of me find a softphone with configurable buttons!
22:31.18pawallsIt seems like this would be an obvious feature.
22:31.19TokyoJimuAre they happy with that?  I don't think I'd want to use a softphone all day.
22:31.30pawallsOur sales people use headsets anyway.
22:31.42pawallsSo it's no difference to them really.
22:31.50jief-ok, Cisco CallManager let you encrypt communication between phones using IPsec, is it possible with *?
22:32.33pawallsIf only I could find a free softphone with programmable buttons... like if you click Button FOO it dials "*510" or something.
22:33.47harryvvwhat bit length encryption do credit cards use
22:33.50pawallsAnyone know of one by chance?
22:33.54pawallsjief-, Sorry, no clue :-/
22:33.55*** part/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl)
22:34.46jief-that's one of the features that Cisco came up with that's really nice. and some other stuff they coded for the US DOD
22:34.49pawallsjief-, http://www.voip-info.org/wiki-Asterisk+Bounty+SIP+encryption
22:34.58jief-pawalls: yeah just saw that :(
22:35.43harryvvso what kind of encryption does it have
22:35.51jief-none
22:35.56harryvvohh nice
22:36.03pawallsNone.
22:36.08pawallsBut you could use a VPN tunnel I guess..
22:36.16harryvvso much for selling it to the federal goverment.
22:36.29jief-MD5 only for IAX
22:36.55*** join/#asterisk Chrif (~fritsch@p54BF376B.dip.t-dialin.net)
22:36.57jief-pawalls: yes, but with Cisco, the communication between any Cisco phones can be encrypted, no need for VPN. it uses 3DES and AES128
22:36.59ChrifHi
22:38.13pawallsHmm..
22:38.27pawallsI wonder what kind of need there is for an xml configurable softphone.
22:38.33Chrifsome time ago, someone told me a bug-number for asterisk with two IPs (Lan and WAN) - i cant find this entry in the bugtracingsystem anymore - anybody a hint?
22:38.34pawallsWhere you can add custom buttons and such.
22:38.55pawallsBasically interface would be defined as an XML document or something..
22:38.56jief-pawalls: call center app on your phone maybe
22:39.04*** join/#asterisk khemir (user119@201.133.129.105)
22:39.06harryvvso the phones them self can do 3des and aes128 and asterisk will pass it.
22:39.36jief-harryvv: i doubt it, its configurable in CallManager, not on the phone as far as i remember
22:39.43harryvvi see
22:40.19terrapenhttp://www.bordergatewayprotocol.net/jon/humor/images/lic_plate.jpg
22:40.56enderterrapen: heh.
22:41.12enderterrapen: I"d be surprised if that is a real plate and the state let them get away w/ it.
22:41.26terrapenim not sure
22:42.32khemirhi
22:42.52khemirsome have expirience in Ovislink whit h323?
22:43.48*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
22:44.27jdv79are there best practices for tuning an install or is the answer immersion again:)
22:44.39jdv79i just hit my proc limit
22:44.48jdv79without even doing anything cool
22:48.08jdv79is call setup and teardown intensive?
22:48.26jdv79i saw an initial surge of load and then it tapered rapidly
22:52.45*** join/#asterisk iq (~iq@204-26-74-11.omah.qwest.net)
22:53.01iqhi all
22:55.40pawallsHmmm...
22:55.55pawallsIn our current phone system, we have call parking (which I set up in Asterisk also)
22:56.31pawallsHowever, with our current system you can see which one currently have people waiting on them by looking at the phone. There is an indicator LED for parked call slot.
22:56.46pawallsIs there any way to do with with Asterisk and the hardware IP phones?
22:56.59enderpawalls: how would you define the slots?
22:57.09endercalls can be parked anywhere from 701 to 799
22:57.20znoGis there an easy to setup stun server for linux?
22:57.31TokyoJimuHow can I see current IAX registry?
22:57.37znoGim not sure i even need it. I have a SIP client behind NAT, however, the SIP server is NOT behind NAT.
22:57.47TokyoJimuThere is a "sip show registry" but not an "iax show registry"
22:58.01robl^pawalls, nt easily.  unless you use a phone like Cisco 79xx using SCCP
22:58.32MiccTokyoJimu, iax2 show registry
22:58.39TokyoJimuAhh.  Thanks!
23:03.07*** join/#asterisk Defraz (~t0tal@tim.ibccom.net)
23:03.08*** join/#asterisk Nemesis760 (~nemesis@71.36.28.33)
23:03.09TokyoJimuOur voicepulse stopped working.  We are registered but: chan_iax2.c: Host 66.234.228.170 failed to authent
23:03.09TokyoJimuicate as voicepulse-in-01
23:03.15znoGso anyone know much about SIP, NAT, STUN?
23:04.15fugitivoit sucks
23:04.58Nemesis760I'm placing an order for a channelized DS3 (PRI) to be terminated via M13 MUX -> Sangoma A104s/Asterisk and need some assistance with various line options... can someone here help. Will pay (reasonable) consulting fee if nec.
23:05.19*** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net)
23:07.24*** part/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
23:07.25Nemesis760Over 200 peeps here, and no takers?!
23:07.36*** join/#asterisk Exstatica (exstatica@65.119.22.200)
23:08.03*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
23:08.19jdv79national bz8 - oh i forget that shit
23:08.25ilanNemesis760: have you tried the asterisk-biz list?
23:08.59Nemesis760Not yet... was hoping for instant gratification. ;)
23:09.27*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
23:10.19brookshireyeah.. good luck with that
23:10.19pawallsrobl^, That phone does support it?
23:10.26pawallsis SCCP the name of the protocol that does that sort of thing?
23:11.02pawallsAnd are there any other phones that have similar functionality..
23:11.05Nemesis760I got the basics... NI1 B8ZS ESF, etc. But no clue on "D-Channel Backup" or Wink, or # of outpulsed digits.
23:13.13pawallsHmm... also..
23:13.20pawallsWhat phone support automatic answer? :-/
23:13.21altpawalls: SCCP is Cisco's proprietary IP Phone protocol.
23:13.52pawallsWith the current phones, if you dial an internal extension, it automatically picks up the phone.
23:13.56pawalls(on speaker)
23:14.14harryvvanyone have experaince with the cdr_addon_mysql ?
23:14.56*** join/#asterisk imcdona (~imcdonald@imail.speakeasy.net)
23:15.08harryvvpaw mm you mean pickup good question.
23:15.23pawallsharryvv, Yeah
23:15.31robl^pawalls, sccp is tghe protocol.  With the right tricks you can make it display any text message on the phone display or you can control lights / status icons next to LCDs.  you can can create the function you want
23:15.52harryvvpawalls, I know there are some heard of them but dont recall.
23:16.01pawallsrobl^, Interesting.
23:16.09pawallsrobl^, Does asterisk speak sccp ?
23:16.17Sedoroxyay for skinny
23:16.37Sedorox(aka sccp)
23:16.45pawallsSedorox, Ah.
23:16.49robl^pawalls, alost.  its not as mauture as SIP or IAX2 support.. but there is chan_sccp and chan_skinny that both talk sccp to some degree
23:16.50Nemesis760How 'bout another topic... Anyone here familiar with the recent CVS Commit dealing with X86_64 processors? It seems to me that the patch assumes that a X86_64 processor is going to be -march=k8. Doesn't this discount the existance of EM64T procs?
23:17.27Sedoroxbah
23:17.34Sedoroxthere is no such thing as a EM64T
23:17.34Sedorox:p
23:17.48Sedoroxwhistles*
23:17.54*** join/#asterisk ozus (~level3@h-67-101-176-189.lsanca54.covad.net)
23:17.54robl^what about XY64Q? :)
23:18.32enderNemesis760: the difference between k8 opts and em64t ops is virtually nonexistant.
23:18.33Nemesis760Funny... I just bought 6 Dell Machines w/ 'em. And RHES for EM64T.
23:19.00enderNemesis760: for most/all compiler cares, EM64T is just x86_64
23:19.12Nemesis760Ok... so I guess it'll be up to me to prove that.? Hopefully I prove you right.
23:19.27key2does Freebsd support SMP ? (for dual xeon) or is it more recommended to set up a linux ?
23:19.29jdv79is a two way g711 call about 70K or 150K of traff?
23:19.49enderNemesis760: feel free.  There was a large discussion about this on the fedora-devel-list.
23:19.57brookshireyes.. freebsd supports smp, but linux is better for asterisk anyways
23:20.18xkevjdv79, about 80k each way
23:20.21xkev64k payload + headers
23:20.55robl^but if you use "trunking" you and reduce the headers  :)
23:20.56Nemesis760Considering I'm supposed to go live in 2 weeks with a pretty large project, I'll be hammering the hell outta these boxes... If I still have a job after drop dead date, then I guess it works. ;)
23:21.12xkevand about 30pps, depending on how many voice frames per packet you configure (not sure if * even has an option for frames per packet)
23:21.34jdv79so for 2way audio we're talkin' ~ 160K roughtly?
23:21.45key2brookshire " can u explain why ? cause i think freebsd has a better kernel and better stability ?
23:22.00Sedoroxkey2: zaptel isn't as developed on freebsd
23:22.01brookshirecuz asterisk works better with linux..
23:22.10Sedoroxand asterisk has some bugs (granted I haven't ran into any)
23:22.15brookshireand it was designed for linux
23:22.22key2oki
23:22.27Nemesis760I remember reading that there are a couple 64bit instructions that are not implemented in k8 that are in EM64T... So is the consensus that if it's built for k8 it'll work on both, but not nec. the other way around?
23:22.29Sedoroxthe way it was written was around linux.. which has a totally different kernel design then freebsad
23:22.30brookshireand is supported only on linux
23:22.56Nuxilet me take this opportunity to remind everyone that asterisk runs on windows, but better on freebsd.
23:23.13wunderkini hate linux, i did an emerge --unmerge ssmtp and now libstdc++.so.5 is missing
23:23.15key2:0
23:23.27brookshireoh.. you hate gentoo
23:23.30brookshireme too
23:23.31brookshire:D
23:23.37wunderkinno i hate all distros
23:23.37Nemesis760ouch.
23:23.56harryvvrun away from 64 bit if you are to use it as a work station
23:24.28ilanhmm, anyone using asterisk on the IBM openpower servers?
23:24.36enderNemesis760: I think thats about right.  adn those few differences really make no difference to applicaiton performance.
23:24.40Nemesis760It is to be used solely as a server... in a pretty heavy environment. 8-T1's / box.
23:25.22Nemesis760Need every performance advantage I can get. =)   Can't wait for Sangoma's DSP cards to be released. 70 days and counting.
23:25.34brookshiredigium cards are out now
23:25.40brookshirewith echo can
23:25.42enderif I recall, the differences were for perhaps some desktop or graphical apps.
23:25.48brookshireand better firmware
23:26.01enderbrookshire: including the E1/T1 cards?
23:26.04brookshireyes
23:26.07enderawesome
23:26.08Nemesis760Yeah? Echo can + hardware channelization?
23:26.18brookshireyup
23:26.29enderare they just newer revisions of existing cards, or will we have to order something specific?
23:26.29brookshireand out perform the current sangoma cards
23:26.30pawallswunderkin, I hate windows. I installed new soundcard drivers, and then Windows XP refused to boot (even into safe mode) after that :-P
23:26.30Nemesis760Good to know.
23:26.39pawallsRequired a complete reinstall.
23:26.51brookshirehttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE411P&tab=details
23:27.04brookshireand they are certified
23:27.14pawallsWe all have our preferences I suppose. Your problem is an odd one.
23:27.25bkw_brookshire, where is your data to back that up... I just wanna see numbers
23:27.44brookshirebkw_: ask malcolm
23:27.59bkw_and you should trust that opinion.. I want hard facts and figures here
23:28.19brookshirebkw_: why are you hating?
23:28.25brookshire:)
23:28.32bkw_i'm not promoting either card.. but if you're going to say its better you better have numbers to back it up.
23:28.35bkw_don't be like signate
23:28.36bkw_:P
23:28.41brookshirethey are there.. we did tests
23:29.10brookshiresangoma couldn't do a full e1 + g729
23:29.13brookshirebut digium could
23:29.40brookshireon a A dual-processor, 3-GHz 800FSB Intel XEON server with 1MB L2 cache
23:29.52brookshirethis is without echocan
23:29.57brookshirecard
23:30.27bkw_and I still don't see hard performance numbers
23:30.36brookshireehh.. oh well..
23:30.55brookshirebecause i don't know them off the top of my head
23:30.56bkw_you buy a porsche because you know its gonna go fast
23:31.00bkw_it has that history of going fast
23:31.14brookshireif you want them.. ask malcolm
23:31.19bkw_and I know mark did alot of work on those cards to improve performace
23:31.29brookshirehttp://www.digium.com/index.php?menu=press/pr_2gen_firm
23:32.15Nemesis760Maybe a stupid question... but what the heck is "J1"
23:32.16Nemesis760?
23:32.19bkw_japan
23:32.23Nemesis760ah.
23:32.24brookshireJapanese T1 basically
23:32.28brookshiredifferent protocol
23:32.28bkw_and I like how they say they support J1 when infact know it doesn't work
23:32.39brookshirenope.. j1 support is in
23:32.45bkw_no it doesn't work right
23:32.50bkw_just ask benjk
23:32.50brookshireand tested on digium hardware
23:32.57brookshirewon't work with sangoma
23:33.01bkw_what won't?
23:33.02bkw_J1?
23:33.05brookshireyup.. won't work
23:33.07bkw_riight
23:33.09bkw_ok
23:33.12brookshireuntil they update their driver
23:33.25brookshireusing digium's code of course
23:33.33bkw_and you work for digium?
23:33.43brookshireyeah
23:34.03bkw_J1 does work on sangoma
23:34.10brookshirecrestl1n finished the testing on J1 two weeks ago
23:34.11bkw_benjk and I had a talk about it last week infact
23:34.18bkw_it might work on this new board
23:34.31MikeJ[Laptop]ahhhhhhhhhh
23:34.47MikeJ[Laptop]PURPLE MONKEYS!!!
23:34.51brookshireall digium cards support j1 now
23:34.57brookshireeven the old ones
23:35.11*** part/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com)
23:35.19bkw_well if you're going to be saying these thigns you better post hard facts and figures to back it up
23:35.26bkw_the exact configs
23:35.32*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
23:35.34bkw_if you wish to compare them
23:36.37bkw_and for the record I don't hate
23:36.50bkw_I have never personally bought any digium or sangoma hardware.
23:36.58bkw_if you wish to hate go check out the home page of voipsupply
23:37.07bkw_and look at the sangoma board with digiums name below it on the front page :P
23:38.00bkw_I think they fixed it now
23:38.16shido.
23:38.47bkw_the company I work for on the other hand has a boat load of 410's
23:38.48bkw_haha
23:39.03*** join/#asterisk torment0r (~torment0r@md-frdrck-cmts3a-b-222.shphwv.adelphia.net)
23:39.07torment0ri'm back
23:39.11bkw_OH NO
23:39.15bkw_:P
23:39.18ender*chuckle*
23:39.19lesouvageI have festival running with Diane from Cepstral. The last problem s that she is kind of shouting. Does anybody know how I can pump down the volume too a sophisticated level?
23:39.54torment0rthis agi script i'm working is pretty 0wn4ge.. i'm using asterisk to deturn an ex-girlfriend :)
23:40.05torment0rgood stuff
23:40.59Nemesis760Does anyone here know about "D Channel Backup"?
23:41.12MikeJ[Laptop]torment0r, learn to type english.. cuz all that cr4p is l4me
23:41.40MikeJ[Laptop]Nemesis760, yes
23:41.45torment0rpfft.. don't be scared :)
23:42.04MikeJ[Laptop]torment0r, you are such a l337 hax0r...
23:42.05Nemesis760I think I get what it is, just don't know how / if it's implemented, and weather or not I want it.
23:42.18torment0ri never said all that now
23:42.39enderMikeJ[Laptop]: it's a maturity measurement device.
23:42.40Nemesis760MikeJ: Care to share some knowledge?
23:42.47MikeJ[Laptop]Nemesis760, do you have a question, or are you just speaking (err, typing) out loud?
23:42.54MikeJ[Laptop]nfas
23:43.30MikeJ[Laptop]lets you have 1 or more d channels running a bunch of pri's (usually want to keep it around 20 at most by convention)
23:43.34Nemesis760I'm ordering a channelized DS3 (28 PRIs) and want to know if this is something I can / should take advantage of.
23:43.48MikeJ[Laptop]Nemesis760, that is where it is useful, yes
23:43.57torment0rlets you have?? what kind of english is that?
23:44.00torment0rlol
23:44.07*** join/#asterisk Weezey (WeezeyD@206.210.109.233)
23:44.08MikeJ[Laptop]my english
23:44.10MikeJ[Laptop]:D
23:44.25torment0rexactly, 0wn3d
23:44.46MikeJ[Laptop]Nemesis760, the big advantage in that situation is you can use fewer chan's for D chan.
23:44.54Nemesis760So another part to this question... does this imply that not every PRI has to utilize a D-Channel and that I can gain extra voice channels?
23:45.05Nemesis760Preempted me.
23:45.10MikeJ[Laptop]so you can get another 20+ voice channels out of your ds3
23:45.12*** part/#asterisk jaike (~a@203.131.137.76)
23:45.28WeezeyWhen I go from SIP into asterisk and out IAX, I get: un 22 19:40:37 WARNING[22181]: channel.c:1607 ast_indicate: Unable to handle indication 3 for 'SIP/2527-0a15'  and the call doesn't ring.  Any clues?
23:45.36enderNemesis760: I think that is exactly what it means.
23:45.41MikeJ[Laptop]well, all of the pri's "use" a D channel, just not one on the same t1
23:45.59Nemesis760And this can be implemented this way in asterisk? Sorry if I'm being redundant... just want to be sure.
23:46.17MikeJ[Laptop]now if you are talking to a quad t1 card, you are going to need to have at least one (perferabally 2, primary and backup) per card
23:46.41MikeJ[Laptop]err.. maybe per server now, I am not sure with the recent 2nd gen enhancements.
23:46.50Nemesis760Ok, and if 2 Quad spans/server... could I still get away with just 2 D Channels?
23:47.07MikeJ[Laptop]I am not sure.. what cards are you planning on using
23:47.08Nemesis760Damn your good... 1 step ahead of my ???s. =)
23:47.13sivanaquit
23:47.16sivanaack
23:47.16Nemesis760Sangoma A104s.
23:47.23MikeJ[Laptop]give them a ring.
23:47.29stormfri have rawplayer using 100% cpu. Load still ok anyway but is there a problem in handle thread with rawplayer/asterisk ?
23:47.40Nemesis760Will do... thanks for the skinny.
23:47.42MikeJ[Laptop]stormfr, stable or head
23:47.43MikeJ[Laptop]np
23:47.56sivanahehe
23:48.15stormfrcvs head 2005-06-09
23:48.26stormfr(currently upgrade to current)
23:48.53bkw_doubt it
23:48.58bkw_you must be doing something very wrong
23:49.03*** join/#asterisk SwK (~SwK@12-219-156-206.client.mchsi.com)
23:49.26Nemesis760MikeJ: Are you available for consulting?
23:49.35Nemesis760Paid of course.
23:49.46shidogo MikeJ[Laptop]
23:49.57shidobefore someone else snatches him up
23:49.59*** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net)
23:50.04shido:)
23:50.27MikeJ[Laptop]what do you need?
23:50.31MikeJ[Laptop]and where are you at?
23:51.31Nemesis760I just want to have someone at my disposal to bounce questions off as I go into final ordering phase for this project... Want to make sure I get no surprises....
23:51.49Nemesis760I'm in Boise, Id. My Company is based out of San Diego, CA.
23:52.15Nemesis760You can be anywhere.... pmt via PayPal, or whatever's clever.
23:52.21*** part/#asterisk bonez41 (~aint@drjones.dsl.xmission.com)
23:52.22brookshirejust call sangoma.. damn
23:52.40Nemesis760brookshire: Have other questions.
23:52.40brookshirei mean.. you did buy the cards from them.. ask them
23:52.59bkw_brookshire, this is #asterisk not #digium
23:53.24brookshiredigium did start #asterisk, remember
23:53.31Nemesis760bkw_: THANK YOU. :P
23:53.35bkw_yes but without the asterisk community it wouldn't be here.
23:53.46bkw_so alot of digiums success is here in the community.
23:53.47brookshirewithout digium, asterisk wouldn't be here
23:53.54brookshirethey go hand in hand
23:53.57bkw_no something else would be in its place
23:54.02Weezeychicken + egg = dinner
23:54.14torment0ryummie
23:54.17Nemesis760Chicken / Egg... Ok Digium was the egg, and it did come first... But the community helped it hatch.
23:54.58bkw_brookshire, without the work of Jim Dixon digium wouldn't exist.
23:54.59brookshiretrue.. but nemesis, really.. maybe 7 employees of digium are here in this chatroom.. and would help you.. but we're not going to support you if you do not support us
23:55.04bkw_so do please give credit where credit is due.
23:55.28bkw_brookshire, digium employees very rarely help anyone
23:55.35brookshirebkw_: without mark + jim, the tormenta would still be isa
23:55.36bkw_I have yet to see much help from the digium staff in here.
23:55.42brookshireand not pci
23:55.50bkw_thats neither here nor there.
23:55.55brookshirethat's the truth
23:56.04bkw_and the tor2 is GPL hardware
23:56.07bkw_moving on
23:56.13bkw_Nemesis760, you had some questions?
23:56.15Nemesis760I'm happy to support Digium... would pay them for consulting... But I need a certified platform, for me that means Dell, and I'd heard of problems with Digium cards on Dell 2850 s.
23:56.15torment0ri talked with mark on the phone once, that was pretty awesome
23:56.16brookshireand was just as much designed by mark as it was jim
23:56.18brookshirejoint project
23:56.22torment0ri was like.. i'm not worthy
23:57.14X-Robwell, I buy TDM400 cards rather than SPA2000 and 3000's, so, pfft.
23:57.42bkw_I just use 10 dollar modems
23:57.46Nemesis760bkw_: Wink / immediate outpulse... what's preferred and why?
23:57.55bkw_you have a CT1
23:58.01bkw_why did you not order PRI?
23:58.04X-Robbkw, you're in the US, you can. Our PSTN impedence is all screwed up compared to yours.
23:58.12bkw_X-Rob, true
23:58.28Nemesis760I am ordering PRIs... Does that mean the Wink question is irrelevent?
23:58.28bkw_Nemesis760, Every CT1 I have ever used  is Wink start
23:58.33bkw_yes
23:58.36bkw_you don't wink a PRI
23:58.40Nemesis760Ok... that was easy.
23:58.49brookshiredigium supports dell 2850
23:59.02brookshireand has been throughly tested with Asterisk Business Edition
23:59.04brookshirefor all cards
23:59.10bkw_brookshire, does digium have a good list of systems that have passed to publish?
23:59.15torment0rasterisk business edition??.. this is news to me
23:59.20brookshirei have three.. i'll get them
23:59.23bkw_its a closed src version of asterisk
23:59.23Nemesis760Number of outpulsed digits, is this a user preference, or dictated by *?
23:59.25SwKNemesis760: just remember on a 2850 TDM400s w/ FXS wont work (no power inside for the SLICs and most of those clones of the x100p kill the PCI buss and the box wont boot
23:59.45bkw_Nemesis760, that shoudln't matter its PRI
23:59.51torment0rthat's pretty interesting, when did this branch come about?
23:59.59bkw_torment0r, go check their website

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