irclog2html for #asterisk on 20060316

00:00.03hfb"ManxPower vile, You have to buy new handsets.  PBX companies make their handsets incompatable with ANYTHING else so their customers are locked into buying their overproced phones."
00:00.06hfbArgh.
00:00.31ManxPowerHmm?
00:00.37ManxPowerAh.
00:00.48*** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com)
00:01.00hfbWow something you posted 2 years ago.  :)
00:01.09ManxPowersofh, Um, all bandwidth measurements are ONE WAY
00:01.16justinuthere's a company that sells very expensive SIp gateway for definitey, avaya, nec d-term, nortel phones
00:02.08justinusettle down, beavis
00:02.49FuriousGeorgeyour nothing without me
00:02.59FuriousGeorge*you're
00:03.04ManWithYellowBatyou = pwnd.
00:03.17ManWithYellowBatMONKEY >:O
00:03.28FuriousGeorgeSapien
00:03.37ManWithYellowBatboo
00:03.52FuriousGeorgeyou know i'm 5 to 7 times stronger than the average man dont you
00:04.00ManWithYellowBatyou're not a gorilla
00:04.06ManWithYellowBatyou're a very small monkey
00:04.19FuriousGeorgedamn straight im a carnivour
00:04.19ManWithYellowBatand i have opposable thumbs :)
00:04.23FuriousGeorgeand a bad mofo
00:04.38FuriousGeorgedude, i got opposable thumbs on me feet
00:04.56FuriousGeorgethat means i can punch you and kick you with all four appendages
00:05.00ManWithYellowBathaha
00:05.36*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
00:06.18gigglesanyone know how to program a mediatrix 2102?
00:06.30FuriousGeorgeManWithYellowBat:  that was me:  http://www.cbsnews.com/stories/2005/03/07/national/main678634.shtml
00:06.34justinumonkey feet
00:06.36fugitivoanyone with experience with R2 + unicall ?
00:09.05justinudo any IP phones support ipv6?
00:09.20buuArgh. Fufkcing hold music
00:10.28buuThere has to be some way to easily change which music it's playing.
00:10.31ManWithYellowBatGeorge :http://www.cfhf.net/lyrics/images/super-mario.jpg
00:10.33ManWithYellowBatthat's me :)
00:10.46ManWithYellowBatoop... http://www.cfhf.net/lyrics/images/super-mario.jpg
00:10.58buuAnyone?
00:11.29justinubuu: put different mp3's into /var/lib/asterisk/moh
00:11.32justinuor whatever the directory is
00:11.38buujustinu: And how do I stop it playing the current one?
00:11.42buuWhat if I remove one?
00:11.55justinumoh reload
00:11.58justinudelete it
00:11.59xevoPoE doesn't require different patch cables, does it?
00:11.59buumoh ?
00:12.03justinumod reload from the CLI
00:12.05justinuxevo: no
00:12.08r_evolutionhey buu... you should also use lame and sox to convert it to a raw file : )
00:12.13justinuhowever, if we're talking polycom
00:12.15buur_evolution: Um. Why?
00:12.21xevothey're just trying to upsell me then
00:12.23justinuonly the 601 supports native 802.11af
00:12.39r_evolutionbecause it will free up some of your processor load
00:12.39justinuthe 301/501 need a special cable/adapter
00:12.51xevooh really?
00:12.53buur_evolution: Oh, well, eh
00:12.54xevoodd
00:13.08buur_evolution: More concerned with making it work than making it work fast.
00:13.23r_evolutionwell... part of making it work is making it work with more users, right? :)
00:13.33justinuxevo: i know... i just ordered 20 IP301's w/ the special connector
00:13.37*** part/#asterisk Peaceful (n=Peaceful@70.98.162.62)
00:13.51xevohow much were each of the connectors?
00:13.59justinulike 30 bucks
00:14.03xevoouch
00:14.05justinuyeah
00:14.10justinucustomer's money tho...
00:14.12justinutheir problem
00:14.16xevohehe
00:14.34*** part/#asterisk dec (n=tom@ppp151-30.lns3.adl2.internode.on.net)
00:14.46xevoDo 501s come with power adapters?
00:14.58r_evolutionwha justin? the power injectors?
00:17.32justinuyeah, the injectors
00:17.41justinu501s can be ordered with the injectors
00:17.53dja_Hi.  Any hints on how to minimize echo?  I'm going from AnalogPhone->ATA->Asterisk->VOIPProvider->PSTN->AnalogPhone.  The echo is driving me crazy.  :)
00:18.08justinudja: on your ATA, turn down your gains
00:18.53dja_justinu: thanks -- I'll give that a try.
00:18.58justinugood luck
00:23.32xevoSo it's just a special cable if they have 802.3af switches?
00:24.43asterboyAnyone with some insight into why I can call out no problem, but when I try to call in, no voice!
00:25.21asterboyztmonitor registers the * side, but nothing coming from the pots side.
00:25.56asterboyWorks when I use a Digium FXS card, but not for my Polycom SIP setup.
00:26.13asterboySo I'm sure the FXO ZAP Channel is fine.
00:26.20justinuxevo: a cable with a small box in the middle of it
00:26.42justinubox has an LED that lights up when it gets power from the switch, but isn't plugged into the phone
00:27.03*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
00:27.15asterboymoved the FXO cards to a new server...same.
00:27.19*** join/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net)
00:27.22xevoahh ok
00:27.35*** part/#asterisk outtolunc (n=me@c-24-23-162-126.hsd1.ca.comcast.net)
00:30.21*** join/#asterisk brockj49464 (n=brockj49@63.87.56.236)
00:36.47Zodiacalanyone know what could cause constant static on an fxo module? a reboot fixes it, but only for about 12 hours, the it reapears on a random fxo module in my system
00:37.00Zodiacalthe = then
00:37.18justinuirq conflict
00:37.20Zodiacalnope
00:37.40*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
00:37.43justinusince the module is random, it's gotta be a software problem
00:38.23Zodiacaljustinu any ideas where to begin figuring out where?
00:39.03justinufigure out what happens when it goes bad
00:39.08justinulook at your messages.log
00:39.10justinufull log
00:39.26justinucheck /proc/interrupts when it's working, recheck when it's borked
00:39.31*** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com)
00:40.01X-RobZodiacal, does loading and unloading the module fix it?
00:40.28Zodiacali'll have to try again when it happens
00:40.30Zodiacalbut i don't think so
00:40.41Zodiacalerr, can't remember ;P
00:40.53Zodiacalits happend 3 times so far..
00:41.04Zodiacalit will probably happen again at noon
00:41.24Zodiacali'll try those suggestions, its hard to figure out exactly when it happens tho
00:41.35Zodiacali will have to constantly pickup the line every 5 mins or so
00:48.29*** join/#asterisk NeonLevel (n=NeonLeve@200.52.142.186)
00:49.12*** join/#asterisk nahirean (n=nahirean@67.132.43.2)
00:49.44NeonLevelgood day everyone, anyone has setup a sip provider on a diferent sip port than 5060? i cannot get this working. i've read it and i think it may be the port= parameter but it wont accept it. has someone has done this? thanks in advance
00:50.03justinubindport=
00:52.14ManxPowerNeonLevel, I believe port= is the DESTINATION port for that device.
00:52.38NeonLevelManxPower: that is exactly what im trying to do
00:53.00NeonLevelthe destination according to my provider it would be 5070
00:53.03NeonLevelno 5060
00:53.09ManxPowerdo you have a remote SIP device that's listening on port 5070?
00:53.21NeonLevelis a SIP terminator to the PSTN
00:53.31NeonLevelit's a big provider here in my country
00:54.00nahireanhello, I am using a .call file to trigger a call in the spooler.. however, once it references the context - it keeps looping to the dial string, and doesnt pass beyond that even when I pick up the call.. may I post a 6 line context, perhaps someone can take a look?
00:54.01ManxPowerso you have a [provider] section in sip.conf with a port=5070 in it?
00:54.11justinucall a SIP uri: like sip:user@voipprovider.com:5070
00:54.15NeonLevelthat is corret
00:54.35NeonLeveli haven't thought of that....
00:54.39NeonLevelit might work
00:55.00NeonLevelbut since i find the parameter port, i'd be struggling with it
00:55.07NeonLevelsorry to bother you guyes with it
00:55.16nahireanthere are no goto commands in the context, i dont know why it's looping
00:56.14nahireanit follows the context correctly until the call is answered, then it repeats the context
00:57.29*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:58.08synthetiqanyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why?
00:58.31asterboyHow do the channels #s get assigned to wcfxo cards?  Are they whatever you set in zaptel.conf?
00:59.37asterboyI'm getting channels 1 and 2, but my 3rd card won't load even though it shows in interrupts
01:00.26NeonLeveljustinu: how would be the syntax for a sip uri, with a number to call, username, password, host, port ?
01:02.24NeonLevelanyone???
01:02.34r_evolutionNeon try registering the provider in the sip.conf
01:03.15r_evolutionregister => :@66.185.167.228
01:03.42NeonLeveli did register it, and it registers ok
01:03.47r_evolutionregister => SIP:PW@PROVIDER/EXTEN
01:03.57r_evolutionthen use that for outbound calling in the extensions
01:04.03NeonLeveland the incoming calls are OK
01:04.06*** join/#asterisk asteriskmonkey (n=phil@69.158.146.217)
01:04.20NeonLevelbut when i try to place a call using this SIP channel, it simply wont work
01:04.35asteriskmonkeyanyone know any possible reasons why a tmd2400 card cant be modprobed?
01:04.36r_evolutionhow are you formatting it?
01:04.46asteriskmonkeyalsthough i do see an extra ethernet card in lspci
01:05.20r_evolutionexten = _1ZXXXX.,1, Dial(SIP/1${EXTEN:1}@provideraddress)
01:05.23r_evolutionis what i use for long-dis
01:05.35r_evolutionyou can actually not use the EXTEN:1
01:05.38r_evolutionand just use EXTEN
01:05.43r_evolutionand drop the 1 after the SIP/
01:05.46NeonLevelthis is the line i have in the general sip.conf <<<< register => username:password@200.66.96.57:5070/335004301 and it registers ok
01:05.56r_evolutionok what about your extensions.conf?
01:06.06*** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net)
01:06.37NeonLevelextensions.conf <<<< exten => _8.,1,Dial(SIP/${EXTEN:1}@username:password/200.66.96.57:5070||Tr)
01:07.11*** join/#asterisk exten123 (n=exten@60.49.6.190)
01:07.14r_evolutiondid you register a peer in the sip.conf?
01:08.12NeonLevelwell i did commented the peer, in sip.conf because i thought it won't be necesary dialing with a sip uri
01:08.37NeonLevelyou want me to uncommenting it?
01:09.10r_evolutionyou can... it's usually easier that way (for me)
01:09.24r_evolutionb/c then you just put SIP/ etc @ [whateverisbetweenbrackets]
01:09.26NeonLevelok hold on
01:09.47asterboyHow are wcfxo cards assigned channels?
01:10.27*** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
01:11.10r_evolutiondunno aster... i don't use a tdm card
01:11.24asterboyThey are the fxo.
01:11.33r_evolutioni know... i dont use one
01:11.48r_evolutioni was under the impression that they were activated in the zaptel.conf file
01:11.56Zodiacalnow that would be neat
01:12.00asterboythats what I thought.
01:12.01r_evolutionno kidding
01:12.03r_evolutionthat would be the hotness
01:12.36asterboyaudio wireless sip fones don't work, so why may video?
01:12.57Zodiacalwhich have you tried?
01:13.16Zodiacalasterboy
01:13.17r_evolutioni dunno... the one our CIO uses seems to work pretty well
01:13.35r_evolutionthe ones pulver was selling.. the bcm taiwanese phones
01:13.43asterboyThe Zyxel
01:13.55Zodiacalisn't the zyxel the cheapest of the bunch
01:14.05r_evolutionZyxel and UT Starcom
01:14.15asterboyStarcom has bad delay
01:14.18r_evolutionthe grandstream of WiSIP phones ;x
01:14.43Zodiacali just want ViWiSiP too much to ask?
01:14.45asterboyI haven't a hope of landing this contract using *
01:14.51r_evolutionway too much
01:15.02Zodiacalonce my fios gets lit up... that would rock
01:15.07*** join/#asterisk Kumbang (n=kumbang@167.205.24.5)
01:15.26asterboyEcho on the line, so I reduce TX gain, then it's too low a volume...catch 22
01:16.08asterboyCan call into ZAP FXO from SIP phone but can't hear calling party.
01:16.32asterboyCan't figure how the fucking thing assigns channel numbers to try other FXO cards.
01:16.51asterboyStill need to figure out call transfer and call forward.
01:18.07Zodiacalanyone know if its posible some how to use a web cam and a sip hardphone ?
01:18.18asterboyTime to look at other solutions...OpenPBX any good?
01:18.52r_evolutionopen PBS?
01:18.54ZodiacalopenPBS isn't that redudent?
01:18.55Zodiacal:P
01:19.05r_evolutionplanning on running our own Public Broadcasting System?
01:19.33asterboylol
01:19.43asterboyshort for PuBeS
01:19.53asterboyopen pubes
01:20.10Zodiacalr_evolution ever hear of such a setup? pc with webcam and hardphone/sip ?
01:20.17Zodiacalto make a video call. can Sip do things like that?
01:20.20Zodiacaland asterisk?
01:20.46r_evolutiondunno... i know * possibly has some video support
01:22.19r_evolutionshow video codecs
01:22.20r_evolutionDisclaimer: this command is for informational purposes only.
01:22.20r_evolution<PROTECTED>
01:22.20r_evolution<PROTECTED>
01:22.20r_evolution--------------------------------------------------------------------------------
01:22.20r_evolution<PROTECTED>
01:22.22r_evolution<PROTECTED>
01:22.24r_evolution<PROTECTED>
01:22.26r_evolutionsee
01:22.29r_evolutioni havent messed with it at ALL though
01:23.02Zodiacalsomeone on the wiki said that "Cisco VT Advantage" can do it
01:23.04Zodiacalhrmm.
01:23.47Zodiacali guess just setting up some kind of webcam software to turn on when the phone rings and to send the user params is all thats nessisary
01:24.38Zodiacalhehe
01:32.47*** join/#asterisk Eitch (i=[U2FsdGV@unaffiliated/eitch)
01:33.06synthetiqanyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why?
01:33.50r_evolutionim not using spandsp yet synth ;x
01:33.52r_evolutionbut soon!
01:35.11synthetiq=/
01:50.47*** join/#asterisk zotz (n=zotz@24.231.32.85)
01:53.46*** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell)
01:56.44*** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com)
02:02.46*** join/#asterisk file[laptop] (n=jcolp@142.131.190.116)
02:03.11*** part/#asterisk diclophis (n=diclophi@65.203.37.58)
02:03.13file[laptop]hello world
02:03.44justinuit's back
02:03.55justinui rm -rf'ed that file
02:04.06file[laptop]:\
02:04.08EitchLOL
02:04.13Eitchi liked this new verb
02:05.19file[laptop]I was going to get a muffin this morning... but they didn't have any good ones
02:05.51Qwell[laptop]file[laptop], liar!
02:06.09file[laptop]nope, quite true
02:06.32Qwell[laptop]I asked them.  "Has file been there?" "No."
02:06.46file[laptop]I went to the little cafe place in the Marriott...
02:06.48file[laptop]quite good
02:06.56Qwell[laptop]oh?
02:19.24exten123hey guy how to solve this failed to pass IP ACL when trying using Asterisk Manager?
02:21.15*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
02:25.51*** join/#asterisk m_a_g_o (i=maxgluck@201.243.97.246)
02:29.00*** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net)
02:29.50*** join/#asterisk mattwj2005 (n=Matt@user-12l3lm4.cable.mindspring.com)
02:32.12*** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116)
02:33.16*** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net)
02:37.45willtDo I have to use the ztdummy module to use the meetme application? What if im using FreeBSD?
02:38.36*** join/#asterisk asteriskmonkey (n=phil@69.158.146.217)
02:38.49asteriskmonkeyanyone know why a tdm2400 serious card would modprobe?
02:38.53asteriskmonkeyit dosnt show up at all
02:39.00Qwell[laptop]asteriskmonkey, why it *would* modprobe?
02:39.07asteriskmonkeysorry wouldnt
02:39.08asteriskmonkey:P
02:39.15Qwell[laptop]there are various reasons
02:39.20asteriskmonkeyusing centos 4.2
02:39.24Qwell[laptop]first things first - does lspci show it?
02:39.29asteriskmonkeyno
02:39.34Qwell[laptop]call Digium
02:39.39asteriskmonkeyshows unkonw device
02:39.55asteriskmonkeydigiums not open right now
02:40.00Qwell[laptop]email Digium
02:40.07asteriskmonkeyfrom your experience what is this issue?
02:40.11Qwell[laptop]got me
02:40.18asteriskmonkeyive had it in the past but its been due to a bad motherboard
02:40.53Zipper_32hmm, my x100p clone is showing up on lspci as: Communication controller: Motorola: Unknown device 5608 .... I have a feeling that's at least the source of one error of mine.
02:48.23ManxPowercontact the vendor you bought it from
02:51.11*** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116)
02:54.21*** join/#asterisk digg10 (n=john@206-248-135-54.dsl.teksavvy.com)
02:59.34*** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net)
02:59.56synthetiqanyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why?
03:01.44asteriskmonkeyyep its not compiled properly
03:02.56*** join/#asterisk nvicf (n=nvicf@201.250.165.83)
03:03.04nvicfhello
03:03.32synthetiqbut it did compile so...?
03:04.03nvicfI have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?
03:04.14digg10ne1 managed to build a web interface for managing the dialplan?
03:04.29synthetiqweb interface
03:04.32synthetiqhah
03:04.52digg10funny?
03:05.43asteriskmonkeysynthetiq: i had the same isse i had to delete all the old files and recompile it
03:05.54digg10i need to control the dialplan from browser
03:06.05asteriskmonkeyuser a m p then
03:07.01Qwell[laptop]freepbx..
03:07.06Qwell[laptop]~freepbx
03:07.32asteriskmonkeyha freepbx wants money so why call if freepbx lol
03:07.51Qwell[laptop]for a similar reason
03:08.32asteriskmonkeyjust write some php and make your dial plan sql driven :)
03:10.25*** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116)
03:15.02exten123Why I can't connect Asterisk Manager from others computer?
03:20.20digg10why is this irc channel not more active?
03:20.51*** join/#asterisk Psyiode (n=lacigol@205.241.238.186)
03:21.15Psyiodeim having trouble ringing a sip cisco 7940
03:21.40PsyiodeI've setup queues and agents, and it will ring sip softphones, but the ciscos still wont ring... what am i missing..
03:22.50digg10how can i do time of day routing?
03:25.18synthetiqgotoiftime
03:25.20synthetiqexit
03:25.21*** join/#asterisk habakuk (n=chatzill@c-24-6-173-113.hsd1.ca.comcast.net)
03:25.43Psyiodeinclude => yourcontext|time-time|day-day|*|*   that will route the call to yourcontext during the specified time during specified days, do that for all your time and days
03:25.46Psyiodeor here http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours
03:26.00*** join/#asterisk jjg_ (n=doink@dsl081-245-050.sfo1.dsl.speakeasy.net)
03:26.03jjg_hi
03:26.43*** join/#asterisk tengulre (n=tengulre@61.185.224.66)
03:26.58jjg_i'm having trouble getting my granstream budgetone 100 to register using NAT through a linksys .... the phone is in the DMZ and i am getting 401's over and over, any recommendations?
03:27.17jjg_i followed teh instructions on voip-info on setting up the grandstream and sip.conf file
03:27.57*** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116)
03:28.06habakukHey folks. I have a question. I'm using manager app using Originate. The problem is it tends to block until the channel is established. Any ideas on how to configure is so that the channel immediately connects? I was considering using the local channel. Would that solve the problem?
03:29.21habakukjjg_ the phone is behind NAT or the asterisk is behind NAT?
03:29.43jjg_habakuk , the phone is behind NAT
03:29.47digg10the include =>  line goes in a seperate section, or inside a context?
03:29.56*** join/#asterisk bit123 (n=bit123@203.115.15.252)
03:30.16bit123how to do ad-hoc conferencing with asterisk ?
03:30.27Psyiodedigg10: i have it in my default context, which is my first context, which only contains openhour included
03:30.31habakukjjg_: so does the phone respond to the 401? is it getting the 401?
03:30.40tengulreanybody know which website provider software exchange services?
03:31.14jjg_habakuk , i don't think so, is there a way to turn on debugging alerts on the phone?
03:31.50habakukjjg_: get a cheap hub and a laptop running ethereal
03:32.13nvicfI have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?thanks
03:33.53orlockman
03:34.00orlocksip/*/grandstreams are loosing me
03:34.04orlockanybody here used Sail?
03:35.45dippono, what is it?
03:36.01dippowhich grandstream phones are you using?
03:36.05orlockits an interface for managing asterisk via the smeserver interface
03:36.13orlockgxp2k
03:37.04dippoi bought a slew of grandstream budgetones for our office
03:37.08dippoperhaps not a wise decision in retrospect
03:37.45*** join/#asterisk litage (n=nick@203.220.55.70)
03:37.53FuriousGeorgeso i think qualify is causing some of my iax peers to drop out and not come back.  specifically my iax provider
03:37.53bit123hi, anybodyknows how to do ad-hoc conferencing with asterisk ?
03:38.05FuriousGeorgeyou guys think switching to sip will remedy that?
03:38.12dippowhat's qualify?
03:38.51FuriousGeorgea setting that measures the latency for sip protocol communication and desides whether that peer qualifies to be used
03:40.12nvicfI have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?thanks
03:40.24FuriousGeorgebetween this, and *'s apparent inability to handle .dynu addresses with any measure of success, im thinking about just restarting all the servers at 4AM every day
03:41.37willtcan't you set qualify=no?
03:41.51nvicfme?
03:41.55*** join/#asterisk chendy (n=hello_vi@218.80.62.113)
03:42.05orlockdippo: yeah
03:42.13FuriousGeorgeyeah, but i like that safety net.  my boss wont care why the call quality stinks
03:42.23orlockdippo: when i punch in the auth details, update, reboot... the old auth username is still there!
03:43.01willtFuriousGeorge: o i c.. I haven't got a chance to play with IAX yet all sip here..
03:43.15orlockrt?
03:43.16orlockwhats that?
03:43.43FuriousGeorgei guess ill talk to my procider about if changing to sip will help
03:45.04*** join/#asterisk Flauto (n=zhao@71.194.38.112)
03:46.21*** join/#asterisk jjg__ (n=doink@dsl081-245-050.sfo1.dsl.speakeasy.net)
03:46.33jjg__habakuk i have the hub now, gonna see if i can see the messages
03:49.38*** join/#asterisk escribzz (n=manzzzee@VDSL-130-13-28-196.PHNX.QWEST.NET)
03:51.57*** join/#asterisk bmg505 (n=leon@dsl-146-16-10.telkomadsl.co.za)
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03:52.37De_Monhrm, I recompiled app_meetme with some new code and did a 'reload app_meetme.so', but nothing's changed. Do I have to restart asterisk?
03:52.41*** part/#asterisk Psyiode (n=lacigol@205.241.238.186)
03:53.50*** join/#asterisk MarioGamboa (n=yo@201.133.229.135)
03:54.00MarioGamboahi everyone
03:57.36nvicfque hace marito
03:58.15*** join/#asterisk punkgode (n=punkgode@r200-125-63-195-dialup.adsl.anteldata.net.uy)
03:59.36*** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com)
04:00.19punkgodehello, anyone knows what this error message is about? "chan_sip.c:12628 reload_config: Can't add wildcard IP address to domain list, please add IP address to domain manually."
04:00.54punkgodei've triple-checked the configuration files, I can't seem to find the problem
04:01.10rpmdo you have any wildcard ip's?
04:01.16rpmx.x.x.*
04:01.20punkgodeno
04:01.32rpmpaste your sip.conf
04:02.15punkgodek
04:04.55justnulling2<PROTECTED>
04:05.14gigglesanyone know how to program a mediatrix 2102?
04:05.42habakukgiggles: yeah throw it in the trash
04:06.20punkgoderpm, http://debian-uy.pastebin.com/604789
04:06.27*** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
04:06.32habakukgiggles: actually you can configure it via SNMP
04:06.48punkgoderpm, i've removed the lines that started with ";"
04:06.52gigglessnmp has been disabled
04:10.18gigglesthere appears to be a faciltity to download a config file in xml, anyone know the format?
04:13.52punkgoderpm,  well, it was the bind address, sorry to bother
04:14.04FuriousGeorgei thought iax was supposed to be good with nat?
04:14.23habakukFuriousGeorge: what's the issue?
04:14.30FuriousGeorgei got a bunch of boxes behind nat and dynamic ips, some cant register with eachother.  others have their peers drop out
04:14.37sevardDoes anyone have a HOP8T?
04:14.42*** part/#asterisk punkgode (n=punkgode@r200-125-63-195-dialup.adsl.anteldata.net.uy)
04:14.55FuriousGeorgepart of the problem is that asterisk looks up the servers by ip not by url
04:15.01FuriousGeorgethe ip is dynamic the url isnt
04:15.16FuriousGeorgeok, fine, that sucks.  but then the ip changes and they cant seem to find eachother
04:15.37FuriousGeorgeadd that to the fact that im noticing my iax provider peer drops out and the only solution is to restart the server
04:15.44FuriousGeorgei guess restarting networking would probably help too
04:15.49FuriousGeorgelet me try that now
04:16.22*** join/#asterisk litage (n=nick@202.168.41.172)
04:17.45habakukFuriousGeorge: so you are saying your server is on a dyn ip? or your provider is?
04:18.50FuriousGeorgemy server is.  i run a service that monitors the ip and links it to a url even when it changes
04:18.52*** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net)
04:19.10FuriousGeorgebut nhow for instance, a reboot didnt get my other dynamic ip peers back.  it will fix my provider dropping out
04:19.20FuriousGeorgethis is all adding up to very frustrating
04:19.23*** part/#asterisk escribzz (n=manzzzee@VDSL-130-13-28-196.PHNX.QWEST.NET)
04:19.38FuriousGeorgecould you imagine if ssh just kinda sorta stopped working from time to time
04:20.28FuriousGeorgewhen 1.2 was comming out i thought this junk would be addressed, but it seems to be getting worse not better.  just my opinion
04:24.01FuriousGeorgeBOX1:  4 iax2 peers [2 online, 2 offline, 0 unmonitored     BOX2:  4 iax2 peers [3 online, 1 offline, 0 unmonitored]    BOX3:  4 iax2 peers [4 online, 0 offline, 0 unmonitored]    BOX4:  4 iax2 peers [1 online, 0 offline, 3 unmonitored]
04:24.20FuriousGeorgethese computers are all attempting to log into eachother, wtf gives with that
04:24.39FuriousGeorgehow is one supposed to debug that
04:25.42habakukFuriousGeorge: so, you are using something like dyndns.org right? Why not add an  iax2/extension reload to your ifup script?
04:26.10FuriousGeorgeifup=cron?
04:26.28habakukand then have a sed script that updates extensions.conf , iax.conf as well
04:26.37FuriousGeorgein any case it wont help
04:26.47FuriousGeorgeupdate how?
04:26.54FuriousGeorgesearch by ip?
04:27.05habakukwell with dyndns you install a client on each server right?
04:27.08*** join/#asterisk Eroick (n=chatzill@Ottawa-HSE-ppp269684.sympatico.ca)
04:27.11*** join/#asterisk x86 (n=x86@p3m/member/x86)
04:27.13habakukto update dyndns servers right?
04:27.22FuriousGeorgehabakuk: no, on the lan's gateway,firewall box
04:27.56FuriousGeorgeand its not like im using a microsoft wireless router or anything, this is a wired linux box running iptables at every location.  makes no difference
04:28.01willtim seeing this when leaving voicemail: WARNING[1348]: file.c:981 ast_writefile: No such format 'gsm,wav'
04:28.16FuriousGeorgewhat surprises me is that no one else seems to have this problem
04:28.16willtand voice mail doesn't work
04:28.18EroickOk, so is this the idea behind Asterisk? I can have someone dial into my computer over Skype/Gizmo/Other VoIP and then program it to do whatever I want? Is there a tutorial to jump in?
04:28.26FuriousGeorgenot skype
04:28.33FuriousGeorge~docs
04:28.36jbotdocs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
04:29.08habakukFuriousGeorge: so you don't have a ddns client?? you lost me
04:29.31FuriousGeorgethe gateway on the 4 respective networks runs the clients
04:29.42FuriousGeorgefor the dyndns service
04:29.45habakukmove it to the servers.. problem solved
04:29.51FuriousGeorgewhy?
04:29.59FuriousGeorgepinging the url gives the right ip
04:30.05FuriousGeorgeits asterisk that cant seem to handle the nat
04:30.08*** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe)
04:30.52FuriousGeorgeim only using iax b/c i heard it was better with nat
04:31.11FuriousGeorgeconsidering my clients all use sip its starting not to make sense
04:31.54habakuksip behind NAT is a pain. I actually prefer SIP to iax, but it is a pain with NAT
04:32.43FuriousGeorgeas opposed to what im experiencing which is what?  a minor discomfort.  the only thing im sure of is that my sip clients, be it eyebeam, snom phones, sipura phones, or ANYTHING NOT USING IAX doesnt have this problem
04:32.47*** join/#asterisk viLeR (i=1000@66.128.47.232)
04:34.56orlockHmm..
04:35.00orlocknon * sip question -
04:35.09orlockI've got a cisco 7940 at home, running sip firmware
04:35.21orlockevery so often it stops making/receiving calls
04:35.23orlockreboot doesnt fix it
04:35.25habakukFuriousGeorge: yeah go with sip then. sorry without understanding what you are trying to do, its difficult to figure out what's causing this
04:35.27orlockbut then it will work again
04:36.13FuriousGeorgehabakuk: there is nothing really complicated going on.  i have asterisk boxes and i want them to be able to communicate with eachother, so they have corresponding firends in iax.conf
04:36.14habakukFuriousGeorge: btw did IAX reload work? I'm assuming you are using IAX registration
04:36.31FuriousGeorgei set up everything right, and it will work at random 75% of the time with no ryme or reason
04:36.39*** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net)
04:36.45FuriousGeorgeiaxreload helps a bit, but some peers never come back
04:37.05orlockand my * setup, i cant seem to register with upstream
04:37.15habakukFuriousGeorge: heh.. not good.
04:37.35FuriousGeorgei guess ill turn off the web proxy on the gatways before i give up
04:37.59FuriousGeorge(or the peers come back a day later, but by then another 2 have dropped out for good)
04:38.00habakukFuriousGeorge: any indication if it is the client side or the server side that causing the issue?
04:38.42FuriousGeorgeboth, because asterisk is both the client and the server.  box a is both a client and a server to box b.
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04:39.39*** part/#asterisk MarioGamboa (n=yo@201.133.229.135)
04:39.47habakukof course... but the client side seems to have the issue right?.. i.e. not able to find other server
04:40.33*** join/#asterisk litage (n=nick@203.220.55.70)
04:41.11FuriousGeorgewell, no because since they are dynamic they have corresponding registers=>  to eachother, but even though those dont come back after ip change, as long as the peer side doesnt screw up, they can still call between eachother
04:41.16habakukFuriousGeorge: the reason why is I'm building an IAX client, and want to make sure I don't have these problems on the server side... i.e. if I design my client right, hopefully I won't have this problem
04:41.43FuriousGeorgei dont use any iax clients but i have a feeling they wouldnt have the same issue
04:42.39habakukyeah they would if not designed right
04:46.49FuriousGeorgesure, and sip clients would probably do the same thing if not designed right, but they dont, which tells me there is something wrong with *
04:47.32De_MonI'm dialing the exten => 50,1,Playback(agent-loginok)
04:47.39habakukFuriousGeorge: yeah sounds like it
04:47.45De_Monand all I hear on the phone is 'ent logged in'
04:48.47De_MonI was just padding the playback with a wait(1), but it doesn't work in all cases... There's gotta be a more global solution
04:49.36Z-Knightanyone have a VIA based motherboard?   If so, do you still have to do the PROC=i586 when compiling Asterisk 1.2.5?
04:53.00willtany reason why I should save voicemails in more then one format?
04:53.19De_Monwillt to avoid transcoding (not really)
04:53.28*** join/#asterisk Eroick (n=chatzill@Ottawa-HSE-ppp269684.sympatico.ca)
04:54.19EroickI dont understandhow other people can dial into my PBX from something like Gizmo if they are on a different computer on a different network.
04:55.08Eroick~docs
04:55.32jbotsomebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
04:55.32willt:s
04:55.49*** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
04:56.37Eroickwhat softphone is recomened?
04:58.18Z-Knightprobbaly the xlite softphone
04:58.29*** join/#asterisk jjg_ (i=jjg@dsl081-245-050.sfo1.dsl.speakeasy.net)
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04:58.41Octothorpe[away]~idefisk
04:58.44jbotfrom memory, idefisk is a great iax2 softphone for asterisk.  See http://www.asteriskguru.com/tools/idefisk_beta.php
04:58.51FuriousGeorgeheres another mystery:  box1.dynu.com = 70.118.26.5  but even when i restart asterisk it looks to the old address.  where the hell is asterisk cacheing this and how can i stop it
05:00.03FuriousGeorge-- Registered IAX2 to '70.118.26.5', who sees us as 72.68.119.22:1026 with no messages waiting
05:00.09FuriousGeorgethat makes 0 sense to me
05:00.21nvicfI have a little problem, I have a little callcenter in which I make calls and send some music, but that music is played only after the receiver end says hello(or something) and if the receiver end stop talking it hangs up, what's the option to avoid this?thanks
05:00.27FuriousGeorgeso now of course the other box has no freakin idea
05:00.29*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:01.10*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:06.16FuriousGeorgecan anyone tell me why, at random, one of my peers choses to use a port other than 4569?
05:06.44SplasPoodhrm.. new 7960/40 firmware
05:08.09*** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe)
05:16.41orlockHHmm
05:16.42orlock<PROTECTED>
05:16.48orlockWhat could cause that? i can ping it
05:17.56*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
05:18.30orlockPeer 'vic.nexvoice.net.au' is now REACHABLE! (6ms / 3000ms)
05:18.31orlockhmm
05:18.50wasimdid your dns flake out?
05:18.58orlocknuh
05:19.09wasimor possible some netfilter inroute?
05:19.12orlockand theres basically just fibre between them and us
05:19.14orlocknope
05:19.20wasimhmm
05:19.46wasimmaybe they got upset at the protean run chase ...
05:19.49jjg_anybody ever get "port unreachable" ICMP errors from a pc running xlite in response to 407s from *?
05:20.04FuriousGeorgei was able to get all my peers working by setting host=boxname.dynuservice.com.  the only problem is that now all the boxes complain all day that Peer "SoandSo" is not dynaimic
05:20.12FuriousGeorgeand the registrations are being refued
05:22.55willtFuriousGeorge: Why can't you get a static?
05:23.03*** join/#asterisk frk2 (n=frk2@202.5.145.13)
05:23.23frk2hey guys!
05:23.27frk2jbalcomb- you home? :)
05:24.17FuriousGeorgewillt why should i have to.  i set it up right to begin weith and i dont feel like spending the extra 800 dollars a year for 4 static ips
05:24.42willtwho is you internet provider?
05:24.52FuriousGeorgeverizon
05:25.09FuriousGeorgeand it would come out to more like 1000 per year extra for 4 static ips
05:25.10frk2why am i the ONLY one having issues with the GXP 2000
05:25.16willtyikes
05:25.25willtwhat kind of service do you get from them?
05:25.47FuriousGeorge1.5/768 mbps
05:25.55frk2my phone hangs NIGHTLY :(
05:25.58FuriousGeorgewhat i might do i switch to cabelvision which gives semi-static ips
05:26.17willthow much is your service right now cost?
05:26.28SwK_anyone noticed a memory leak lately?
05:26.35FuriousGeorge50 per network
05:26.42FuriousGeorgestatic ip would be 70
05:26.51FuriousGeorgedifference of 20 X 4 X 12 mo = $960
05:26.53willtoh you have more then one ip?
05:27.04willti mean more then one connectino
05:27.21FuriousGeorgeyeah, and they wont connect to eachother
05:27.34wasimSwK_: can't remember
05:27.36FuriousGeorgebecause of nat and dynamic ips.  at least not consistantly
05:27.51FuriousGeorgewell, we'll see now that i dropped the registers and changed host=url
05:27.58SwK_trunk is leaking like a bitch on my sparc
05:28.01willta lot of time they just block dsl -> dsl connections
05:28.05FuriousGeorgehopefully i will still need to restart but it will at least be reliable like it is right now
05:28.16FuriousGeorgeunfortunately the CLI wont stop complaining about it
05:28.26willtjust seems like a lot of money. is covad out your way?
05:28.54SwK_4.5minutes of CPU time 85000 calls and its using over 90meg of memory and not giving any of it back when the calls go away
05:29.20FuriousGeorgewilltt yewah i think they are.  dont they sell proprietary voip solutions?
05:29.38willtyeah but they sell dsl and it's pretty good
05:29.47frk2anybody using grandstream's gxp 2000?
05:29.52wasimSwK_: not good
05:29.58willtmaybe I can sell it to you at my cost :)
05:30.28FuriousGeorgewillt:  i aprreciate it, but its not my isp's fault
05:30.42wasimSwK_: transcoding?
05:31.04SwK_wasim: i'm basically sending it 250calls/sec right now with sipp, calls get answered and wait() is called in the dialplan til sipp sends it a bye 1 sec later then the calls go away
05:31.20FuriousGeorgeadn im sure a static ip from covad is gonna be more than a dynamic one from verizon
05:31.22wasimSwK_: we've seen it stablizie about 45M for 4 e1's, about 10k calls per hour
05:31.36wasimofcoruse it crashes, but thats a different story
05:31.38SwK_<PROTECTED>
05:31.59SwK_135K calls since restart
05:32.26SwK_it just topped 100M on size and 99M on res
05:32.59SwK_this box will run for a while like this tho (it has 8G or ram in it)
05:33.15SwK_hmmm
05:33.16*** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net)
05:33.18SwK_soemthing aint right
05:33.25willtFuriousGeorge: example they retail TeleSoho starting at $59.95 comes with 1 or 4 statics can't remember
05:33.28SwK_<PROTECTED>
05:33.34wasimeww
05:33.36SwK_it just jumped to 160+ me
05:37.17willtSwK: What are you using to stress test that?
05:38.12SwK_sipp
05:38.23SwK_sipp.sf.net
05:38.27willtoh sorry im blind
05:38.37Abydos313anyone here ever work with a 3com mbx 100?
05:38.47SwK_is that anything like a nbx?
05:38.52SwK_heh
05:38.53*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
05:39.05Abydos313no idea, haven't actually seen the unit yet
05:39.24SwK_3com's IP-PBX?
05:39.28Abydos313yes
05:40.04*** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net)
05:40.05Abydos313i have a buddy that needs to move it to another office. he was curious if that unit moves easily and doesn't lose config
05:40.50*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:40.59Abydos313the manual doesn't specify any special shutdown or reboot procedures
05:41.33SwK_i should keep its config i've never really used them
05:41.47SwK_i've seen them but thats about it
05:41.50Abydos313that is what i was thinking. don't see why not
05:42.06SwK_how else would it survive a power failure
05:42.19Abydos313that is exactly what i said
05:42.20SwK_i would just mke sure that any configs have been written out on it
05:43.15Abydos313i'll have to go thru it and checkout the dialplan and config. manual talks about backing up database so i take it the config is all held there
05:43.17SwK_several phone systems work much like cisco routers, you can make changes on them, but they dont commit it to flash or non-volitile storage unless you "save translation" or "write mem" or whatever the command for that unit is
05:43.21*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
05:43.34Abydos313ok, kewl so like cisco ios
05:43.35SwK_omg its mikej
05:43.41SwK_I dunno
05:43.45Abydos313thx SwK_
05:43.47SwK_check the manual
05:44.19Abydos313all web based config according to manual. but i'd guess there is shell access also. but i'll see
05:44.36*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
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06:12.08*** join/#asterisk jasonpr (n=jasonpr@c-24-10-236-54.hsd1.ut.comcast.net)
06:13.48jasonprI head about an asterisk interface that an application can connect via a socket.  Then the application can execute commands.  Does anyone know that this app is?
06:13.49*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
06:14.17brookshiremanager
06:15.41jasonprAsterisk Manager.  Does it allow you to do a MeetMe?
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06:16.08*** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116)
06:21.19MikeJ[Laptop]it's brooks!!!!
06:21.49tarheelcoxnI'm having trouble recording my voicemail greeting from the phone I've got. the volume level is loooooowwwwww. call volume levels are fine. Any ideas what might be going on?
06:22.16tarheelcoxnalternatively, where do I stick the audio file if I just want to record one and plop it in?
06:28.48*** join/#asterisk oej (n=oej@apollo.webway.se)
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06:45.07brookshirehey!
06:45.26brookshireit's mike j
06:45.30brookshire:D
06:49.21tarheelcoxnokay so it's in /var/spool/asterisk/voicemail/default/700 ... I guess I'll go record replacement greet.gsm and unavail.gsm files...
06:52.04*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
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06:55.02RoyK_Stockholmhej
06:55.21mogorman?
06:55.34jasonprI'm trying to set up my dial plan but the Background function drops the call imediatly after the file is done playing.  Isn't there suposed to be a 10 timeout before it contiues?
06:58.41tsumemanager sucks pretty bad
06:58.47tsumeso does any implementation out there
06:58.49X-Robjasonpr, set(TIMEOUT(whatever)=foo)
06:59.00tsumebut what can you expect, they are made by peopel who don't now much
06:59.08X-Robjasonpr, -- show function TIMEOUT
06:59.53*** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk)
07:05.26jasonpris there a good c/c++ api or other interface that I could use to connect to asterisk and listen for evens and execute commands?
07:05.37mogorman~agi
07:05.46jboti heard agi is the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages
07:05.46russellbdenied
07:05.52russellbah.  there it is
07:06.18mogormanheh looks like i win again russellb
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07:13.16SuryeHey, do you need a FXO card, or will a voice modem work for some/all functionality for the POTS line?
07:13.27Qwell[laptop]You need an FXO
07:13.35Qwell[laptop]rj11 port does not an FXO make
07:17.39justnulling2how do i disable "SRV mapped to host" msg?
07:20.30*** join/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p143.telkom-ipnet.co.za)
07:20.51Supercrossgood morning
07:20.58Supercrosshow is everybody doing?
07:21.01*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
07:21.18mogormanfantastic
07:21.22tarheelcoxnSupercross: confused
07:21.31Supercrosslol
07:21.33tarheelcoxnbut loving asterisk!
07:21.36Supercrossjoin the club
07:21.41Supercrossya asterisk is great
07:21.51Supercrossi am needing a bit of help with mysql and asterisk
07:21.54tarheelcoxnI wish I could figure out what's going on with the volume
07:21.55Supercrosscan anybody help?
07:22.05Supercrossvolume?
07:22.15tzafrirnot before you give some details
07:22.34Supercrossi am trying to setup my queue.conf into a sql table
07:23.11*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
07:23.54Supercrossthe doc's on the wiki say i must enter the string, realtime_family=family name
07:24.07Supercrossbut it doesnt work
07:24.31Supercrossno queue's are loaded
07:24.51Supercrossthe family name refers to the name in extconfig file
07:25.04*** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it)
07:25.38tarheelcoxnSupercross: when I record my greet and unavail from the phone, the volume is reeeaally low
07:25.46tarheelcoxnbut the volume is fine when making calls
07:26.00Supercrossummm, that is weird
07:26.09*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
07:26.13Supercrosswhat format are you saving them to? GSM?
07:26.19tarheelcoxnI've been playing with sox trying to just record my ow
07:26.23tarheelcoxnown*
07:26.40tarheelcoxnwhatever asterisk is doing byy default... which...
07:26.47tzafrirtarheelcoxn, are you sure they are the right format?
07:26.48tarheelcoxnthe directory has three files for each
07:26.50Supercrossit will be gsm then
07:27.04tarheelcoxngreet.WAV greet.gsm greet.wav
07:27.05tzafrirwhat does 'file' give you for them?
07:27.09Z-KnightCan someone answer a couple of questions about the zaptel/ztdummy modules in linux and how do you add those at boot?
07:27.26tzafrirZ-Knight, what distro?
07:27.32Z-KnightcentOs
07:27.34Z-Knight4.2
07:27.44tarheelcoxntzafrir:  file greet.WAV
07:27.45tarheelcoxngreet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
07:27.47Z-KnightI compiled zaptel ztdummy
07:27.57Supercrosstarheelcoxn, strange it give you three
07:27.58*** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk)
07:28.04Z-Knightbut they don't load on boot
07:28.11tarheelcoxnfile greet.gsm
07:28.11tarheelcoxngreet.gsm: data
07:28.13Z-Knightwhen I do lsmod | grep ztdummy   I get nothing
07:28.27Z-Knightbut when I check /etc/modprobe.conf they are listed there
07:28.30tzafrirZ-Knight, first-off, does 'modinfo zaptel' and 'modinfo ztdummy' show them?
07:28.40tarheelcoxngreet.wav
07:28.40tarheelcoxngreet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
07:28.48tzafrirmodprobe.conf is irrelevant for autoloading
07:29.02tzafrirnither zaptel nor ztdummy need any post-install action
07:29.19tzafrirFeel free to delete the ztcfg lines from there
07:29.24Z-Knightbut, should I see them when I do:   lsmod | grep ztdummy?
07:29.39tzafrirNot to mention that kernel 2.6's tools won't look there, I believe
07:29.51Z-KnightI have Asterisk@ home.....and I have my separate version of asterisk on a different computer
07:29.57tarheelcoxntzafrir: so it's using the .gsm version of each?
07:30.00Z-Knightthe Asterisk@home has those modules listed
07:30.27Z-KnightI'm fairly new to the modules topic so I might be asking stupid questions
07:30.50tzafrirZ-Knight, the script /etc/init.d/zaptel looks for MODULES which can be defined in /etc/sysconfig/modules
07:31.24Z-Knightthere is no /etc/sysconfig/modules
07:31.28tzafrirunrem in the latter file the line referring to ztdummy
07:31.44tzafriror the quick&dirty:
07:32.04tzafrirecho MODULES=ztdummy >>/etc/sysconfig/modules
07:32.39tzafrirI'm not sure about "pure" centos.
07:32.58Z-KnightI need to read more about this
07:33.30Z-KnightI was simply following the  basic installation of zaptel using 'make linux26' which incudes ztdummy
07:33.31tzafrirOn Rapid and on latest Debians the zaptel init script does not load modules explicitly. However it checks for a timing source, and if there is none, it loads ztdummy
07:33.47tzafrirI believe that this approach is saner
07:33.59Z-Knightso it dynamically loads it
07:33.59Z-Knight?
07:34.47tzafrirmodules are always dynamically loaded...
07:35.44Z-KnightI guess I'm confused a bit, I'm reading the Asterisk book (yes for older version of *) but it mentions having to do a 'modprobe zaptel' and 'modprobe ztdummy' to load those modules
07:35.59Z-Knightafterwards you can load asterisk
07:36.15tzafrir'modprobe ztdummy' will load zaptel with it.
07:36.25*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
07:36.44tzafrirbecause the module ztdummy depends on the module zaptel
07:37.00Z-Knightbut you can't simply load:  modprobe ztdummy
07:37.06Z-Knightit complains about not having zaptel
07:37.12tzafrirAnd the kernel modules loader has a bit more brains than Asterisk's modules loader
07:37.13Z-Knightso I have to do 'modprobe zaptel' ifrst
07:37.21tzafrirZ-Knight, you used insmod
07:37.53Z-Knightthat i don't know....i'm unfamiliar
07:37.55tzafrirsurely it did not complain explicitly about zaptel
07:38.01Z-Knightthis is what I got:
07:38.09Z-Knightmodprobe ztdummy
07:38.09Z-KnightNotice: Configuration file is /etc/zaptel.conf
07:38.10Z-Knightline 0: Unable to open master device '/dev/zap/ctl'
07:38.10Z-Knight1 error(s) detected
07:38.12Z-KnightFATAL: Error running install command for ztdummy
07:38.16*** join/#asterisk bails (n=bails@bailsyatton.plus.com)
07:38.22*** join/#asterisk Lino` (n=Lino@i577BD710.versanet.de)
07:38.34Z-Knightonce I did:  modprobe zaptel, then I could do modprobe ztdummy without a problem
07:38.39tzafrirgrep ztcfg /etc/modprobe.conf
07:38.58tzafrirgrep ztcfg /etc/modprobe.conf | egrep 'zaptel|ztdummy'
07:38.59TelamonI've been having a problem with IAX for a while, and can't seem to figure it out.  IAX calls (any IAX phone, I've tried 4 different models) that go to internal Asterisk functions (voicemail, sound recorder) work fine, but IAX calls that go to other phones (SIP or IAX) or out the zaptel interface get voice transmit problems (the sound cuts out for a half second every few seconds.)  Incoming sound is fine.  Any ideas?  Digium has been pretty
07:39.00Telamon<PROTECTED>
07:39.06Z-Knight[root@localhost 2.6.9-22.EL]# grep ztcfg /etc/modprobe.conf
07:39.06Z-Knightinstall tor2 /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg
07:39.06Z-Knightinstall torisa /sbin/modprobe --ignore-install torisa && /sbin/ztcfg
07:39.06Z-Knightinstall wcusb /sbin/modprobe --ignore-install wcusb && /sbin/ztcfg
07:39.06Z-Knightinstall wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
07:39.06Z-Knightinstall wctdm /sbin/modprobe --ignore-install wctdm && /sbin/ztcfg
07:39.08Z-Knightinstall wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp && /sbin/ztcfg
07:39.10Z-Knightinstall ztdynamic /sbin/modprobe --ignore-install ztdynamic && /sbin/ztcfg
07:39.12tzafrir~pb
07:39.18jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
07:39.18Z-Knightinstall ztd-eth /sbin/modprobe --ignore-install ztd-eth && /sbin/ztcfg
07:39.18Z-Knightinstall wct1xxp /sbin/modprobe --ignore-install wct1xxp && /sbin/ztcfg
07:39.19tzafrirsorry, my mistake
07:39.19Z-Knightinstall wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg
07:39.19Z-Knightinstall wcte11xp /sbin/modprobe --ignore-install wcte11xp && /sbin/ztcfg
07:39.20Z-Knightinstall pciradio /sbin/modprobe --ignore-install pciradio && /sbin/ztcfg
07:39.22Z-Knightinstall ztd-loc /sbin/modprobe --ignore-install ztd-loc && /sbin/ztcfg
07:39.24Z-Knightinstall ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg
07:39.25mogormanahhh
07:39.26Z-Knightso they are there
07:39.28Z-Knightok
07:39.32mogormanpastebin my friend
07:39.42Z-Knighthow do you use pastebin?
07:40.12TelamonZ-Knight: Go to pastebin.ca and copy in the text you want to paste, then just type in the URL they give you into the channel.
07:40.14tzafrir'install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg' : totally useless. ztdummy needs no config.
07:40.21tzafrirAnyway, ztdummy should be loaded
07:40.59Z-Knightbut once I reboot, I won't have it show up when I do 'lsmod | grep ztdummy' .... is this a mistake or is this ok?
07:41.19tzafrirAnd anyway, this is not the job of modprobe to run ztdummy. You have a zaptel init.d script. This is where ztcfg should be run
07:41.24tzafrirbah
07:41.42Z-Knightok
07:41.50Z-KnightI think I understand
07:41.56TelamonZ-Knight: Your Linux distro should have a file that lists all the modules to load at boot, you can just add it to that.
07:42.17Z-KnightTelamon:  isn't that /etc/modprobe.conf?
07:42.27tzafrirIn Debian it is /etc/modules . I'm not aware of any such simple thing on RH
07:42.43tzafrirZ-Knight, no. That file is configuration for the module loader
07:42.52TelamonNo, modprobe.conf just helps modprobe figure out dependancies, it doesn't actually do any loading.
07:43.00Z-Knightok
07:43.03Z-Knighthmm
07:43.20Z-Knightcentos is basically redhat,...there is not /etc/modules   or /etc/sysconfig/modules
07:43.24TelamonZ-Knight: IE, telling modprobe that when you load wctdm or ztdummy, it has to load the zaptel module first.
07:44.40tzafrirZ-Knight, so write your own zaptel init script . It should be run before the Asterisk one. See the skeleton init script in /etc/init.d/
07:45.11tzafrirIt should basically run 'modprobe ztdummy' on 'start' and do nothing otherwise
07:45.30tzafrirYou may need to put some content in it later on.
07:45.31bailscentos /etc/modules.conf
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07:46.08Z-Knightshouldn't it add one when I do 'make install' on zaptel source?
07:46.30tzafrir/etc/modules.conf is the old (kernel <=2.4) modutils config file. It can't be CentOS-specific
07:46.50Z-KnightI have kernel 2.6 with centos4.2
07:47.03tzafririnit scripts are quite distro-specific
07:47.34*** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no)
07:47.42Z-KnightI figured though I would not have to go to trouble of making my own init scripts....I'd think that zaptel would have that configured for me
07:48.15*** join/#asterisk htims (n=htims@Vf3fb.v.pppool.de)
07:48.21tzafrirZ-Knight, I think that an init script is the way to go. hmm, time to file a bug report
07:48.22tzafrir?
07:48.49Z-Knightit might be too early...maybe I'm doing something wrong
07:48.55X-RobZ-Knight, the zaptel.init works fine with centos
07:49.01Z-Knightdid you do your own installation of * or did you do *@home?
07:49.14X-Robmake sure you create /etc/sysconfig/zaptel and put 'TELEPHONY=yes' and 'MODULES=wctdm' or ztdummy or whateveryou want to load
07:49.38tsumeyuck, the distro which rips rhel and comes with uber late security patches
07:49.40tsumegreat
07:49.48tsumeone should be using debian for an asterisk box
07:49.57tzafriruber-late?
07:50.09tsumetzafrir: they were 3 weeks late with a sshd exploit update
07:50.16tzafrirnot that you shouldn't use Debian
07:50.25Z-KnightX-Rob...are there any specific instructions out there for your own installation of *?     I was using one of asteriskguru and it made no mention of making additional scripts/etc....neither does the * book
07:50.26tsumedebian is what most use
07:50.32tsumealso the livecds
07:50.37Z-KnightHere is the general installation I followed:  http://www.asteriskguru.com/tutorials/general_asterisk_installation_compilation.html
07:50.42tsumeits jsut much easier to set up :)
07:51.06tsumebut I guess some people are GUI pussies :)
07:51.29*** join/#asterisk oej (n=oej@apollo.webway.se)
07:51.54tzafrirwhat GUI would you recommend that generates debugable dialplan?
07:52.01bailsmaybe its because some people dont have all day to sort out apt-get/dpkg nastiness, and i use debian so i'm not slagging it off
07:52.34tzafrirbails, apt-get install wajig to get a nicer command-line interface
07:52.47bailsan e1 gurus about?
07:52.57bailsany^
07:53.13tzafrirwajig search <package>; wajig get <package>; etc
07:53.37bailsI have problems with a te110p
07:53.52tzafriror use aptitude/synaptic
07:53.52bailslooks like layer 1
07:54.10X-Robtzafrir, if you use a gui that generates buggy dialplan, then tell the gui maintainers
07:54.11bailslots of crc4 errors and sync errors
07:54.24X-RobAMP/FreePBX doesn't have any wierdism at the moment, we think 8)
07:54.49tzafrirX-Rob, but what if you put a wrong username or hostname to the trunk?
07:54.58russellbX-Rob: so what's the story behind renaming AMP?
07:55.16tsumebails: there is no nastiness. there is only incompetance
07:55.17tzafrirX-Rob, have you seen the definitions of SIP/IAX trunks in AMP?
07:55.45X-Robrussellb, fucked if I know, not my call. But I think the idea was to get away from 'Asterisk' management portal, when they want it to be a genereic manager for opbx,freeswitch,whatever you want to plug into it
07:55.47*** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk)
07:55.51tzafrirYou basically need to put there the peer entry, thje user entry and the register line
07:55.56tzafriras plain text
07:56.04russellbX-Rob: i see.  I was just curious if you knew.
07:56.05tsumeweb based configs are a fkin joke
07:56.15russellbI personally hate the new name, because obviously, AMP itself is *not* a PBX at all.
07:56.17X-Robtzafrir, I'm a FreePBX developer. I'm _aware_ that it sucks, and 'real soon now' there'll be a provider wizard.
07:56.20brookshiretsume: they have their place :)
07:56.22tsumeIE, Moz keep changing how JS, DOM, other stuff works.
07:56.43tsumemakes it complete hell to actually make a maintainable project, especially with DnD
07:56.48X-Robrussellb, well, yes, I kinda agree, but it had nothing to do with me. I'm just a grunt developer in .au, not living the high life in canada 8)
07:57.09russellbX-Rob: ha, it's all good.  I still think it's a great effort.  I just don't like the name.  :)
07:57.16tzafrirX-Rob, so you have to be able to debug errors yourself. And AMP generates such a verbose output that it is practically impossible to figure out
07:57.25russellbanyway, back to work ...
07:57.31tzafrirIt is also quite good at masking out the real reason for the problem
07:57.43brookshireyou guys should have just kept the aberviation 'AMP'
07:57.48tsumeheh, bails is very incompetant :)
07:57.50brookshireand dropped the name ;)
07:57.57X-Rob'A management portal'? 8)
07:58.05russellbsure, why not
07:58.06tsume(CR)AMP just isn't very good
07:58.16tsumeit really gives me the (CR)AMPs
07:58.17brookshireSAMP!
07:58.31*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
07:58.54tsumeneed a decent client GUI based application
07:58.56tsumenot java
07:59.08tsumenot proprietary platform code
07:59.09tsumemaybe something mono based ;)
07:59.12tsumeand GTK#
07:59.18X-RobSo your issue with it, tzafrir, is that it's hard to debug username/password issues in AMP, and I agree, that can be a bastard.
07:59.20brookshiremono is just as bad as java
07:59.30tzafrirX-Rob, a "wizard" is not what I'm after. AMP clearly lacks decent data structures that could be easily manipulated from scripts
07:59.31tsumebrookshire: in memory, no its not
07:59.34X-Robbut when I tidy up trunks, then you'll put a username, password, and pick a provider from a drop-down
07:59.41brookshirejava + swt = mono + gtk#
07:59.46brookshirelol
07:59.48tsumebrookshire: its not such a fkin hog like java
07:59.49X-Robtzafrir, it uses mysql. Do whatever the smeg you want to it.
07:59.55tsumeand it uses gtk# by default
07:59.58tsumewinforms coming soon
08:00.01tsumenative on each platform
08:00.03X-Robit's all stored in a database, and the database is scraped to generate the dialplan
08:00.08tsumemaking it better than java for one
08:00.10brookshirewinforms have been coming soon since 2000
08:00.14tsumealso C# being more OO based
08:00.23tzafrirIf you have a wizard, then any for any change to it, you'll have to go through the wizard again (think of outlook and of cups's web interface)
08:00.24tsumebrookshire: no, winforms for mono started in 2002
08:00.38tsumebrookshire: peopel have really been working on gtk# more, making it stable
08:00.54X-Robtzafrir, no, you use the wizard to generate the entry. Then if you want to change anything, you go into the 'edit trunk' and all the settings are laid out in front of you.
08:01.02tsumebrookshire: also theres a small fact... Novell/mono project focuses more on the paying customers, mainly SuSE or anyone who pays
08:01.25X-Robincluding a 'I know what the hell I'm doing' box which lets you edit the trunk definition directly (which is what you've got now)
08:01.37brookshireopengl + gtk+ is the only way to code ;)
08:01.48tsumenot really
08:01.52tsumeGTK# is just fine
08:02.02brookshireno... gtk# is lame
08:02.03brookshire:D
08:02.09tsumethat is such a troll :)
08:02.10X-Robno, gtk( is better!
08:02.13brookshireit's a distraction
08:02.23X-Roband gtk% is like l33t.
08:02.26tsumeI should make a dialplan gui and sell it
08:02.29tsumeclosed source
08:02.34brookshirex-rob: it's 1338
08:02.36tsumeor maybe shareware :)
08:02.40X-Rob1338? Fwor.
08:02.43tzafrirX-Rob, again, how do I add a phone from the command-line (or a script)?
08:03.06X-Robtzafrir, you populate the mysql database and then call php's config generate.
08:03.07brookshire1337+1
08:03.14tsumeeww! mysql
08:03.21X-Robif you _Want_ to do it from the command line, feel free to write some code and I'll stick it in AMP
08:03.23tarheelcoxnouch. that joke is painful
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08:03.29X-Robbut I don't think it's what average users want
08:03.49X-RobI think the issue is, AMP is _not_ for people that know what they're doing. It's for beginners, and it's deliberately designed to hide the complexities from the end user.
08:04.07X-Robif they're clueful enough to want to add 100 xtns, then they should be doing it themselves, rather than using AMP
08:04.15iDunnoand annoy the hell out of everyone else ;)
08:04.26tzafrirX-Rob, The problem is that too many clueless people use it for production
08:04.39brookshirex-rob: i know someone who has 1,000 lines with amp
08:04.58X-Robtzafrir, brookshire, what can I do to make it easier?
08:05.02X-RobI really have no idea.
08:05.31X-RobWe're about to release 2.0, which is all new and funky, but the dialplan it generates is basically the same.
08:05.47X-RobI'm ready and willing to take suggestions of ways to make it 'better'
08:05.51brookshirex-rob: i dunno.. i've never used it..
08:06.12brookshirebut, i would definately talk with matt o'gorman
08:06.17mogorman?
08:06.31brookshirehah
08:06.42X-RobWhy am I talking to you, mogorman?
08:06.47mogormani dont know
08:06.56X-Robbrookshire?
08:07.11brookshiretell x-rob how he can make amp better :)
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08:07.45russellbmogorman knows all
08:07.53mogormanthat is true russellb
08:08.01mogormanits just getting it out of my tiny brain
08:08.03mogormanis the trick
08:08.15X-Robcrowbar, stat!
08:09.15tzafrirX-Rob, but what I mentioned is that it doesn't do a good job at hiding the complexity: when it breaks: it breaks into pieces. This means error reporting is bad.
08:09.28tzafrirThe error handling of the dial macro is simply ohrrible
08:09.40tzafrirNot to mention that it is overly complex
08:09.42X-Robtzafrir, yes, but a lot of that has been fixed in 2.0
08:09.52*** join/#asterisk RoyK (n=roy@static-213-115-44-227.sme.bredbandsbolaget.se)
08:10.02X-Roboverly complex? It _does_ do a shitload of stuff
08:10.15tzafrirWhich most users don't need anyway
08:11.12tzafrirMost of the applications in the dialplan can't be changed from the GUI
08:11.32X-Robcall forwarding, call waiting, voicemail no answer, voicemail busy, override caller ID
08:11.34tzafrirCompare that to the nice "applications" menu in destar
08:11.47X-Robtzafrir, yes, and that'll be fixed in 2.1, I totally agree that that is arse.
08:12.03*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-170.claranet.co.uk)
08:12.36X-Roball applications will be pluggable modules, and you'll be able to disable 'em if you don't use 'em to clean up your dialplan
08:14.28*** join/#asterisk tehdely (n=delysiid@home.teambarry.org)
08:14.34X-Robso yes, your two complaints so far I think have been addressed? Next? 8)
08:16.31*** join/#asterisk powerchip (i=powerchi@197.80-202-229.nextgentel.com)
08:16.48X-RobOn the subject of applications, we're also setting up a localisation menu, where you pick 'UK' and it sets up call waiting, forward, etc, numbers as per the standard in that country
08:16.49buuX-Rob: It's not chartreuse
08:17.00X-RobAaah, damn good point, buu
08:17.21buuThanks!
08:17.23X-RobI'm afraid that I'll have to take that as an insult, and now argue with you for three hours
08:17.35X-Robs/I'm/But I'm/
08:17.37buuExcellent.
08:18.18buuhrm
08:19.16buus/hrm/$x/
08:19.16buuNo?
08:19.16X-Robnah
08:19.16X-Robit's quite limited
08:19.16X-Robs/quite/very/
08:19.16X-Robyou can't do backquotes
08:19.16buuAnd it seems to work randomly
08:19.16X-Robyeah
08:19.16X-Robor you just broke it
08:19.21X-Robooh, there we go
08:19.58*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
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08:22.08RoyKahrmf
08:22.09X-Robs/(there) (we)/$2 $1/
08:22.13X-Robooh, there we go
08:22.14X-Robs/(there) (we)/$2 $1/
08:22.23X-Robdon't be gay, jbot.
08:22.27X-Robs/jbot/sparky/
08:22.45X-Roboook
08:22.55X-Robs/o+/o/
08:23.06X-Robvery limited.
08:23.09RoyKa:%s/X-Rob//gi
08:23.18X-RobRoyKa, heh
08:23.23RoyKa:)
08:23.50*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
08:23.59Zeeekhey now
08:24.04RoyKZeeek: wot?
08:24.20Zeeekhej!
08:24.24RoyKhej hejj
08:24.52*** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net)
08:24.52ZeeekI thought you were Danish or something
08:25.08RoyKI'm norwegian
08:25.13RoyKbut I'm in a meeting
08:25.21RoyKand staying over in stockholm for the weekend
08:25.23Zeeekoh yeah that's right. Same thing
08:27.36*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
08:27.48*** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net)
08:27.54X-Rob_Grr.
08:27.58X-Rob_or have you got a couple of seconds to spare for a norwegian who doesn't speak english all that well?
08:28.04X-Rob_(to RoyK)
08:28.17Zeeekoh god, the ghetto begins here
08:29.26RoyKX-Rob_: are there any norwegians that doesn't speak english?
08:29.35X-Rob_Actually, he seems to have got better now
08:29.43X-Rob_I think he may have just been rusty 8)
08:29.53RoyKX-Rob_: who is this?
08:30.02X-Rob_powerchip on #freepbx
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08:44.15Zeeekbleh
08:48.18Supercrosshi everybody
08:48.35Supercrossi am needing some help with queues.conf and mysql
08:48.37shiznatixhi doctor nix
08:48.40shiznatixhi doctor nic
08:48.44*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
08:49.00Zeeekask your question, please
08:49.07Zeeekand drop a quarter in the slot
08:49.14Supercrosslol
08:49.19*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
08:49.34Supercrosswhat line would you put in the queues.conf file to tell it to use the mysql table?
08:49.45Supercrossin the extentions u use the switch =>
08:49.52Supercrosswhat do you use in the queues?
08:52.10Supercrossanybody?
08:52.35X-Rob_I'd check /usr/src/asterisk/configs/queues.conf.sample for documentation
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08:53.29Supercrossok will check
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09:10.23kippiHi
09:11.33kippiTrying to logon a agent, I have added it to queues.conf and agents.conf, when I ring to add the agent it asks for logon, that works fine, then it is asking me for a new extension, what is this?
09:16.41Zeeekdid you reload queues and agents or restart asterisk?
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09:26.01shiznatixHello, I am having difficulty with sending faxes on asterisk 1.2.4. Here http://pastebin.com/605040 is my extensions.conf file and the callfile. If anyone can help that would be fantastic
09:30.54*** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca)
09:31.11konfuzedslePP, eh are you alive at this hour
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09:43.18Winkieasterisk billing support sucks
09:46.06Zeeekso many things in this universe suck worse
09:47.15frk2yup
09:47.20frk2like chinese voip phones
09:47.25frk2they suck big time
09:47.36ZeeekI have three, they're not that bad
09:47.45frk2wait till you put a 100 of them
09:47.48frk2i have 5 too, not bad
09:47.52frk2deployed 75 at a client
09:47.57frk2and random things start happening
09:47.59Zeeekwhy would I want 100 chinese phones?
09:48.09Zeeekyour client is too cheap
09:48.14frk2hahahah
09:48.16frk2indeed
09:48.18frk2have told them so
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09:50.18fourcheezefrk2: which models are they?
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09:50.53frk2atcom at-323
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09:53.36snip3rhi all
09:54.08snip3ranyone having a couple of minutes helping me out with an asterisk/sip/nat issue?
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10:02.36*** join/#asterisk astra^^ (n=muhajir_@59.145.104.74)
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10:11.51kippiZeeek: yea I did a reload on asterisk
10:15.48[ProB]CrazyManhello, anybody had some issue like this with spandsp ? http://www.roterschnee.org/failure.tif
10:16.04[ProB]CrazyManthat faxes doesn't look ok
10:16.22*** join/#asterisk marktt (n=marktt@203.217.18.2)
10:18.19Zeeeksnip3r just ask and ye shall receive
10:20.59snip3rZeeek: thx :)
10:22.01*** join/#asterisk tv_manojkumar (n=Administ@59.93.56.163)
10:22.41snip3rso, I have a simple config: asterisk on a public IP, clients behind NAT, and there's no voice between any client and/or any other endpoint
10:22.52snip3rprotocol is SIP
10:23.12snip3rnat=yes & canreinvite=no is set in sip.conf
10:23.36snip3rfor all clients and my outbound proxy
10:23.56shiznatixdoes anyone have any expierence with sending / recieving faxes with asterisk 1.2.4?
10:24.23snip3rhowever, I'm still getting an "attempting native bridge of SIP/blah" message in the CLI
10:24.58frk2sniper thats fine... it will never actually bridge
10:25.20frk2unless you have no tTowW options in your dial
10:25.29snip3rfrk2: wow. that's good news.
10:25.34snip3rfrk2: nope
10:25.49snip3rI have a 4-line dialplan
10:25.56[ProB]CrazyManshiznatix: my spandsp fax doenst work very well with 1.2.4
10:26.01frk2meaning if your dial is like dial(SIP/100,,) then bridging may happen
10:26.26*** part/#asterisk toot (i=chris@212.20.250.186)
10:26.56snip3rmy dialplan looks like "exten => _06.,Dial,SIP/${EXTEN}@outb"
10:27.02Skidwhat about 1.2.5 ?
10:27.11snip3rwhere "outb" is my outbound link
10:27.14shiznatix[ProB]CrazyMan: have you found a solution to this? Also, where do you get a problem? Maybe its my configuation, can you check out what I have at http://pastebin.com/605040
10:27.20frk2it might bridge... unless you have canreinvite=no
10:27.25snip3rI have
10:27.41frk2if it bridges you'll get another message saying so
10:27.44frk2so no worries
10:27.46snip3rboth in PSTN gateway config and client config
10:28.08snip3rthen, why I'm not getting any voice?
10:28.21kmilitzerHi! Can anybody tell me what I need to test the t.38 capabilities of spandsp and rxfax? If i read it right, the latest version of spandsp and rxfax/txfax can act as t.38 gateway, but I could not find out if I need the svn branch of asterisk or if 1.2.4 works and where exactley to get the latest spandsp 0.3 and the t38-bits.tgz file ....
10:28.29frk2dont know
10:28.31frk2whats the sip client?
10:28.35snip3rthe most interesting thing is that it worked
10:28.42snip3r^^ a moment
10:28.49[ProB]CrazyManshiznatix: The faxes I receive doesn't look very well ..
10:29.00snip3ra Sipura h/w phone and an X-Lite
10:29.08[ProB]CrazyManshiznatix: aswell I get connection interruptions
10:29.14snip3rso, it worked in a previous installation
10:29.26snip3rwith almost no configuration changes at all
10:29.27ZeeekX-Lite - make sure transmit silence is ON
10:29.31shiznatix[ProB]CrazyMan: that is not a problem, i just need it to send *anything* even if its horrible quality.
10:29.34snip3ryep, it's on :)_
10:29.47snip3ri've read a couple of articles on this issue
10:30.02Zeeeksniff it to see what ip address is being sent to asterisk
10:30.11shiznatix[ProB]CrazyMan: I paste the callfile in the asterisk outgoing folder and it starts the fax but when it tries to write to /tmp/ffax/ it juts hangs and times out
10:30.12snip3rbut thought I'm missing something trivial
10:30.30snip3rZeeek: a minute
10:30.35tv_manojkumarHi All, I need some help on asterisk realtime setup
10:32.36[ProB]CrazyManshiznatix: you have to put somehere a number to dial an destination
10:32.49snip3rZeeek: To: <sip:2004@192.168.1.102:5060>
10:33.02snip3rI'll resolve this and come back again
10:33.03snip3r:)
10:33.17[ProB]CrazyManshiznatix: you mixed txfax and rxfax
10:33.18snip3rthanks for pointing out the obvious
10:33.30shiznatix[ProB]CrazyMan: huh ok lemme try it
10:33.36[ProB]CrazyManshiznatix: txfax => sending, rxfax -> receiving
10:33.47[ProB]CrazyManshiznatix: do you use zap ?
10:34.11shiznatixi have to get it to work with sip first them zap, bosses orders
10:34.17Zeeeksnip3r but the client is on the same LAN as * or not?
10:34.25tv_manojkumarHi CrazyMan
10:34.39[ProB]CrazyManshiznatix: do in the callfile sth. like Channel: zap/g1/numbertodial
10:35.55snip3rZeeek: * is on a public IP, clients behind NAT on a different public IP
10:36.05shiznatix[ProB]CrazyMan: Alright, ill try some of this stuf
10:36.51snip3rx.x.x.100 is asterisk and x.x.x.136 is my nat box
10:37.18snip3rthere're 2 clients behind the nat box, the Sipura phone and an X-Lite
10:37.23Zeeekwell sending the local ip address won't get the audio properly
10:37.28snip3rsure
10:37.38Zeeekyou could try STUN
10:38.15snip3rX-Lite uses STUN by default, isn't it
10:38.16snip3r?
10:38.31frk2but setting nat=yes fixes that
10:38.48frk2why dont you just try a echo test first?
10:38.53snip3rnat=yes is set in all clients' config
10:39.31frk2hmmm
10:39.33snip3rI'm getting no audio and I've tried all sample extensions :(
10:39.38frk2maybe nat box is dropping packets?
10:39.46frk2does the call connect?
10:39.51snip3rsure
10:40.09frk2is the nat box also a firewall?
10:40.20snip3rZeeek pointed out that the phone sent its private IP
10:40.31snip3rit isn't
10:40.32shiznatix[ProB]CrazyMan: I get the error 'No channel type registered for zap'
10:40.39*** join/#asterisk backblue (n=igor@82.102.1.42)
10:40.49Zeeekhuh?
10:40.51frk2the phone will send its private ip
10:40.57frk2asterisk will ignore it if you set nat=yes
10:41.01snip3rIC
10:41.01[ProB]CrazyManshiznatix: how do you make externeal calls ?
10:41.06Zeeek192.168..... is public ip? nah!
10:41.10snip3rnope
10:41.24snip3r195.x.x.100 is *
10:41.32snip3r195.x.x.136 is the NAT box
10:41.36Zeeekanyway it's logical tohave nat=yes for a client behind NAT
10:41.47snip3ryep
10:41.53shiznatix[ProB]CrazyMan: I don't call outside of the asterisk network
10:42.18shiznatix[ProB]CrazyMan: And I don't have a fax machine or a phone I am actually trying to dial, I just want it to take the fax in then save it to file
10:42.28snip3rX-Lite also discovers the type of NAT
10:42.40snip3rand identifies it as a port restricted cone
10:42.52Zeeekactually, X-Lite worked for me behind nat right away on FWD
10:42.57snip3rIC
10:43.11snip3ras I said, it worked for me too
10:43.18[ProB]CrazyManshiznatix: ? how do you want to receive an fax? if you try to let asterisk send to itself an fax (same box) that will not work
10:43.27snip3rbut I removed * to install SER
10:43.37Zeeekblasphemy!
10:43.45snip3r'cause I needed DB auth
10:43.50snip3rsry :)
10:44.09snip3rnow I realized that I need 'em both
10:44.19shiznatix[ProB]CrazyMan: Why not? I give asterisk the callfile which says 'start sending the fax' then i want it to just save the fax to a folder on the asterisk server.
10:44.20Zeeektwo wives are better than one
10:44.27snip3rabsolutely true
10:44.55Zeeekshiznatix what is the original source of the fax?
10:45.15shiznatixZeeek: Just a random tiff file
10:45.32frk2dude that is blasphemy. no more asterisk help for you
10:45.33frk2lol
10:45.37[ProB]CrazyManshiznatix: spandsp could not handle that faxes .. (I dont know it, maybe because there is then no timing or so ..)
10:45.51snip3r:)
10:46.43Aze`Anyone use misdn with avm ?
10:47.32shiznatix[ProB]CrazyMan: What do I do then? I just need to be able to send a crappy fax to asterisk with a call file and have asterisk save it to disk. Any way to get that done is a great way
10:49.40Zeeekwhy not just email the TIFF?
10:51.01tv_manojkumarHi People, I need some help on Asterisk RealTime setup
10:55.34FiskfanHow many of you here are working daily with asterisk, as a technichan with installation and maintence?
10:55.46wasimme, me and me
10:55.55astra^^me 2
10:55.56Fiskfanwasim... schizo? :)
10:55.58*** join/#asterisk juanjoc (n=juanjoc@222-32-235-201.fibertel.com.ar)
10:55.59tv_manojkumarme 2
10:56.04wasimFiskfan: you have no idea ...
10:56.13Fiskfan:)
10:56.27kmilitzerWhat do I need to test t.38 gateway capabilities of spandsp?
10:56.51tv_manojkumarHi People, I need some help on Asterisk RealTime setup
10:57.06astra^^wht is it
10:57.22Zeeekyo wasim
10:57.47wasimbonjour monsieur zeek, comment ca va?
10:57.52Fiskfanah lunch... see you later...
10:58.23tv_manojkumarastra: Whenevr I want to add a new context I should add it to extensions.conf, can I avoid this and directly add to Mysql
10:58.49tv_manojkumarIf I do not add in Ext.conf then that context is not recognozed by asterisk
10:59.11Zeeekça va bien, mais j'ai un peu faim là
10:59.22tv_manojkumarany solution for this
11:00.22tv_manojkumarplease help me
11:00.38*** join/#asterisk Whisk (n=a@194.130.117.202)
11:04.48*** join/#asterisk zotz (n=zotz@24.231.32.85)
11:06.05tv_manojkumarHi Whisk,zotz : can u help me on asterisk realtime
11:07.51zotzi would not be the best, have not done much asterisk lately
11:07.58zotzwhat is your question though
11:08.19tv_manojkumarWhenevr I want to add a new context I should add it to extensions.conf, can I avoid this and directly add to Mysql
11:08.34tv_manojkumarIf I do not add in Ext.conf then that context is not recognozed by asterisk
11:08.57tv_manojkumarzotz: this is my question
11:11.35*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
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11:15.51nelahi
11:16.55tv_manojkumarhi nela
11:17.23*** part/#asterisk nela (n=nela@xdsl-213-196-230-59.netcologne.de)
11:17.55*** part/#asterisk tv_manojkumar (n=Administ@59.93.56.163)
11:20.28wasimi guess they hit it off well ...
11:23.21zotzso tv left with no answer
11:23.59snip3rsnip3r is back :)
11:24.19snip3rcould anyone take a look at the log http://cozmo.hu/log.txt ?
11:24.52*** join/#asterisk RoyK_Stockholm (n=roy@static-213-115-44-227.sme.bredbandsbolaget.se)
11:25.16snip3rthat's a full call log with no rtp traffic in neither direction
11:27.09snip3rZeeek, frk2: could you check this out?
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11:29.49Zeeekweird
11:30.35*** join/#asterisk chris_ast (n=Administ@59.93.56.163)
11:30.37fenlanderhmm. so much for astricon earlybird registration starting yesterday...
11:31.15snip3rthe PSTN g/w is working well and as I mentioned, the same setup worked in an earlier installation
11:32.14snip3rI gotta go but if you can comment on my problem, please write and I'll check it out soon
11:33.36chris_astPlease help me configure asterisk dialplan
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11:42.25astra^^if i recieve a call frm a sip server to * . what are the setting to be at the sip server end to route the cal to my * server
11:42.55*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
11:43.22chris_astastra: what is your sip server, ser?
11:43.39astra^^free BBd
11:43.46astra^^bsd
11:44.01astra^^sorry free BSD
11:44.15kippiTrying to logon a agent, I have added it to queues.conf and agents.conf, when I ring to add the agent it asks for logon, that works fine, then it is asking me for a new extension, what is this?
11:45.44chris_astastra: what is the name of your sip server, free BSD is OS of your machine
11:47.06astra^^cris_ast:porta one
11:47.37*** join/#asterisk oracle^ (n=cam@unaffiliated/cameleons)
11:48.36chris_astI can tell how we can do it for ser to asterisk
11:49.24Zeeekkippi I had that happen the other day. I think it means you are logging in it wants to know what your call back is
11:49.24chris_astfor portaone I have no idea
11:50.08chris_astHi People, I need help configuring my DialPlan
11:51.21Zeeekspit it out
11:52.14frk2damn my grandstream locked up again
11:52.24astra^^chris :yes ser to asterisk
11:52.33astra^^i have a dial peer in asterisk
11:54.40astra^^chris_ast: PORTA ONE was my billin .s/w s
11:54.47chris_astYou just need to rewritehost name and port
11:55.18astra^^name to..?
11:55.56astra^^name to peer name of the asterisk..?
11:56.16chris_astcheck http://www.voip-info.org/wiki-Asterisk+at+large
11:56.41chris_astsee inder  Asterisk and SER on same box
11:56.46chris_astunder
11:57.11*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
11:57.21chris_astHi Zeek, here is my problem
11:57.30*** join/#asterisk dyer (i=dyer@unaffiliated/dyer)
11:57.39Zeeekwe're all ears!
11:58.58frk2somebody PLEASE help me with my grandstream :(
11:59.26frk2does anybody have a grandstream with asterisk here?
11:59.34ZeeekI have one in a box, it just stopped working
11:59.48Zeeekyes we have three, what is the problem
11:59.48frk2i wonder if the 100 windows pc on the same network are causing it to screw up
11:59.55frk2random lock ups
12:00.08frk2using 1.0.2.13
12:00.29frk2the phone voice quality degrades.... and then poof.. its unpingable. hard crash
12:00.31chris_astI have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf alll the time
12:00.31chris_astwhenever I want to add new context I will do it in Mysql DB but without mentioning that in ext.conf asterisk is not taking it
12:00.37chris_asthow can we do this
12:00.57frk2this happens with the original firmware also
12:01.03chris_astZeeek: that is my question
12:01.06frk2i am almost clueless
12:01.37frk2i thought grandstreams were good
12:01.45frk2or do i have a unique issue?
12:02.29frk2Zeeek? are you facing something similar?
12:03.01Zeeekmost people will say bad things about GS phones. My older ones work ok but the new GX ones I've always heard BAD about
12:03.29Zeeekchris_ast I don't know about that it sounds like you want to look at realtime on the wiki or something
12:04.00frk2but you said you have 3
12:04.04frk2do they cause similar issues?
12:04.53chris_astzeeek: I looked at realtime but even for that we have to name context in ext.conf and tell it to switch to realtime
12:05.28konfuzedslePP, are you awaje yet ;^)
12:05.55Zeeekchris_ast look at the mailing list and see if you find anything there
12:06.47chris_astThanks for the idea but I already added a post there before two days but sadly no good response
12:06.54Zeeekfrk2 I don't have random lockups but I do have one phone that will not stay registered even on the same side of LAN as asterisk
12:07.19Zeeekchris_ast that means only one thing. You'll have to invent the solution and sell it
12:07.29chris_astI am testing my luck here :)
12:08.06Zeeekyour luck doesn't look too good at the moment :)
12:08.44chris_astBut I was not provided that much time here
12:09.14Zeeekprovided? Oh, you mean no one gives a $hit ?
12:09.18chris_astit's true, anyway is my question clear?
12:09.30ZeeekI think it is
12:09.49frk2zeeek exactly.
12:09.49Zeeekbut youhave to find someone who cares about that kind of thing, maybe they solved it
12:10.27frk2the phone that doesnt stay registered- does it lock up? or do you have to restart it to get it to register?
12:10.40frk2do you have the qualify=yes parameter?
12:11.19Zeeekyeah nothing helped
12:11.23chris_astzeeek where can I get that kind of info, I am a newbie to asterisk and only today I cam to know about this chat
12:11.29Zeeekno lock up though, just unreachable phone
12:11.37Zeeektried all firmwares, no diff
12:11.44frk2yes thats what happening to me
12:11.52Zeeekbut I can call out
12:11.53frk2just that one phone?
12:11.54powerchip<PROTECTED>
12:11.58*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)
12:12.19frk2are you using your phones connected to a pc or standalone?
12:12.26frk2maybe i have TWO phones like the one you have :)
12:12.34powerchipsip
12:12.44kmilitzerI guess I am too dumb, I cannot find out in Bug 5090 if there is a way to use asterisk as a t.38 gateway or not ... last message from steveu from 03-12-06 09:38 seems to imply that, but I cannot figure out how this can be accomplished ...
12:12.52frk2so you DO have qualify=yes?
12:13.41chris_astzeeek: where can I get that kind of info, I am a newbie to asterisk and only today I cam to know about this chat
12:14.50wasim~wiki
12:15.11wasimugh, you silly stupid jbot
12:15.27wasimchris_ast: www.voip-info.orggggggg
12:15.28Zeeekfrk2 I did when it was behind NAT and NOT when it was on the same side as astrisk
12:15.41*** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br)
12:16.29chris_astwasim: I have gone thru that site but my thing is not listed there
12:16.36tzafrir~voip-info
12:16.38jbot[voip-info] the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
12:17.12tzafrirno need to be harsh on the poor bot_
12:17.16frk2hmmmmm
12:17.40frk2mine are all on the same side as asterisk
12:17.45chris_astcan I tell u people what I require exactly?
12:18.01chris_astI have gone thru voip-info
12:18.03frk2can there be a problem with older hardware revisions?
12:19.20chris_astI need a dynamic asterisk setup, I do  not want to change extenions.conf all the time instead I change mysql DB
12:19.36kardecallanIs ther anybody that can help me in configuration of the STUN?
12:21.17*** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br)
12:22.08*** join/#asterisk dragonkh (n=kings@dr4g0nn.gotadsl.co.uk)
12:22.11dragonkhhi
12:22.57dragonkhhow do I load the zaptel drivers?  I installed by TMP400 and installed asterisk
12:23.27dragonkhif I modprobe zaptel - I still see nothing in zttool
12:23.40tzafrirmodprobe wctdm
12:23.59dragonkhI get an error with wctdm
12:24.14dragonkhFATAL: Module wctdm not found.
12:24.17tzafrirrun ztcfg
12:24.18dragonkhhow do I fix that ?
12:24.22chris_asttzafrir: please help me on astersik realtime setup
12:24.25tzafrirDo you get the same error?
12:24.40tzafririf so: module was loaded
12:24.46dragonkhZT_CHANCONFIG failed on channel 4: No such device or address (6)
12:24.49dragonkhI get
12:25.20tzafrirlsmod | grep ^zaptel
12:25.33kippiIf i get a IXAy box, will that let me send faxes?
12:25.52tzafrirchris_ast, sorry, ask others...
12:26.03dragonkhzaptel                186372  0
12:26.06tzafrirAnyway, you better ask your question anyway
12:26.10dragonkhcrc_ccitt               2176  2 zaptel,hisax
12:26.15tzafrirso module was not loaded
12:26.37tzafrirDid you run modprobe wctdm ?
12:27.02kardecallanhello Zeeek!!
12:27.09dragonkhmodrobe wctdm ==> FATAL: Error running install command for wctdm
12:27.23dragonkhmodrobe wctdm ==> FATAL: Module wctdm not found.
12:27.46dragonkhwhere do I get the wctdm ?
12:27.51tzafrirwhat version of zaptel is it?
12:27.59dragonkhhow do I tell ?
12:28.12tzafrirHow exactly did you install it?
12:28.13dragonkhI ran the cvs astinsaller
12:28.29kardecallanasterisk -V
12:28.33dragonkhsudo /.astinstall
12:29.30dragonkhAsterisk CVS-v1-0-01/26/06-14:37:36
12:29.39wasimq
12:29.53tzafrirdragonkh, you're using an ancient version of asterisk and zaptel
12:30.07tzafrirAny special reason to keep using it?
12:30.28dragonkhnope - how do I get a modern version then ?
12:30.51tzafrirhttp://www.asterisk.org/download
12:30.53dragonkhI downloaded the astinstall script and it connects to cvs and installs it
12:31.12tzafrirWhere did you get it from?
12:31.24dragonkhthe asterisk site I think
12:31.26dragonkhin here
12:31.56kardecallanIs there anybody that can help me?
12:32.24dragonkhisnt 01/26/06  - 26th Jan 2006 ?
12:33.08kardecallanI need to install the Stun?
12:33.18tzafrirdragonkh, right. My mistake.
12:33.23dragonkhI guess its USA format  -  2001 - June 6 ?
12:33.30*** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it)
12:33.45dragonkhit does say CVS v1
12:34.04tzafrirdoes 'modinfo zaptel' give something? (don't paste it here) how about modinfo wctdm? modinfo wcfxs?
12:34.33dragonkhtzafrir: so you think the problem is im missing the wctdm driver?
12:34.35tzafrirhmmm... it pulls the branch v1-0 from the CVS. It is ancient code
12:34.45tzafrirmodprobe fcfxs
12:34.50tzafrirmodprobe wcfxs
12:35.11tzafrir(and still, you should work with more recent code)
12:35.32dragonkhall of those commands give not found errors
12:35.52tzafrirfcfxs is a typo . wcfxs should exist
12:36.01dragonkhok let me check
12:36.49kardecallanby the way, I have installed it, but I need now to configure it, could you help me?
12:37.15dragonkhnope all the comands complain - Module wctdm not found
12:37.23dragonkhits clearly missing
12:38.10*** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk)
12:39.41*** join/#asterisk eaglezz (n=eaglez@62.108.213.120)
12:42.37shiznatixhello, does anyone have a tiffg3 file that I can use to test a fax with?
12:43.37*** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu)
12:44.08kardecallandragonhk, you can to go www.asterisk.org and download of the zaptel, to install you run make;make install.
12:47.30Zeeekdragonkh I had that problem - it's a linux config file
12:47.41Zeeekmodules or something
12:49.38Zeeek<PROTECTED>
12:50.13Zeeekor is it modprobe.conf ?
12:50.29Zeeekpost-install wcfxs /sbin/ztcfg
12:52.26fourcheezegot some issues with ringtone not being generated
12:52.56fourcheezeit seems that whenever I'm on my second extension of a call when I dial I don't get ringtone
12:53.17*** join/#asterisk pheo (n=pheo@63.Red-80-36-138.staticIP.rima-tde.net)
12:53.18fourcheezee.g. in an ivr or when handling a * or 0 drop out from voicemail
12:53.20fourcheezeany ideas?
12:53.23pheohi all
12:54.54shiznatixhow can I make a tiffg3 document without ghostscript?
12:55.49fourcheezeinstall ghostscript first?? ;-)
12:58.11shiznatixhar har :)
13:05.12*** join/#asterisk dasenjo (n=dasenjo@208.195.215.153)
13:05.36dasenjoHi, I got just - in the transalation line for a codec (g729) .. why?
13:07.06Zeeekbecause you don't have g729 installed?
13:07.51dasenjoyes I have codec_g729.so in /usr/lib/asterisk/modules
13:08.01Zeeekand the license?
13:10.14fourcheezehow does * work out the times in show translation ?
13:11.05*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:11.09ZeeekI think it's just a table
13:11.15dasenjois the "free" one .. downloaded from http://kvin.lv/pub/Linux/Asterisk/
13:11.27Zeeekno advice for the free, never used it
13:11.33*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:12.26fourcheezeI didn't think there was a free one
13:12.45fourcheezethere's a patent on using g729, so if you live anywhere where patents apply then it's not free
13:13.37dasenjoI know .. there is no free one .. I found the error .. the module was in /usr/lib/asterisk but strangely didnt load automatically ..
13:15.24astra^^hw do we know wher pattent is aplicable
13:17.15fourcheezeastra^^: no doubt you'll be well versed in your country's patent law
13:17.18fourcheeze;-)
13:19.32[ProB]CrazyManshiznatix: you can't
13:31.57*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
13:32.40*** join/#asterisk shiznatix (n=shiznati@213-35-233-152-dsl.end.estpak.ee)
13:34.53*** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com)
13:35.20*** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1)
13:35.56*** join/#asterisk nettie (n=esivieri@85-18-54-38.ip.fastwebnet.it)
13:40.27astra^^:-/
13:40.47*** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.209)
13:40.48Kernel_Corehi all
13:41.13*** join/#asterisk mko-025 (n=korpim@p5498BD34.dip0.t-ipconnect.de)
13:41.31*** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116)
13:41.34astra^^fourcheeze: i have a dbt
13:42.42Kernel_CoreMar 16 07:33:28 WARNING[6451]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x85c83a8', 10 retries! <---- what is wrong ?! ( I am running 1.2.4)
13:43.59*** join/#asterisk gambolputty (n=root@64.74.225.131)
13:44.37gambolputtyAnyone having trouble compiling zaptel 1.2.4 with centos 4.2 kernel 2.6.9-34.ELsmp?
13:45.22astra^^if a ser fwds call at my * wat are the setting to be made ....
13:45.53astra^^i chkd in the sit bt nt so informative..
13:45.58astra^^site i mean
13:46.37astra^^SER to Asterisk ...
13:47.11astra^^ser--->Asterisk--->host server.
13:51.17Aursgambolputty: you have to change a typo in a file in the kernel source
13:51.50Aursgambolputty: I have a URL in an email here somewhere.. let me find it for you.. one sec
13:52.08Aurshttps://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 - there you are
13:52.50astra^^any help on my question
13:52.55astra^^:)
13:54.58AursKernel_Core: http://bugs.digium.com/view.php?id=6445&nbn=15 - something like this?
13:57.21Kernel_CoreAurs: yea
13:57.54Kernel_CoreAurs: when asterisks load increase I get such error and after that asterisk stops serving .....
13:58.03*** join/#asterisk aze (n=lucky@ACayenne-101-1-11-208.w81-248.abo.wanadoo.fr)
13:58.09azehi all
13:58.25Zeeekhello
13:59.39*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
14:00.06AursKernel_Core: not that it helps you, but I had a similar case on a 1.0.9 this morning
14:01.05Kernel_CoreAurs: could you fix this problem ?
14:01.35*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
14:01.38AursKernel_Core: not really. just ignoring it until we make the switch to 1.2.5
14:02.07AursKernel_Core: but this happened on a 1.2.4?
14:02.29Kernel_CoreAurs: yes , exactly
14:02.32AursKernel_Core: are you using IAX to connect to other asterisk servers?
14:03.19Kernel_CoreAurs: no , I am useing chan_h323
14:03.52Kernel_Corebut I think it is generally channel.c problem
14:04.08Kernel_Corenor SIP IAX Problem
14:04.10AursKernel_Core: I was googling this earlier today. check this out: http://bugs.digium.com/bug_view_page.php?bug_id=3848
14:04.40*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
14:04.55astra^^i want to fwd calls frm SER to * wher in asterisk will route to the iconnect server..
14:05.10astra^^what are the setting to b made in tis..
14:05.21astra^^asterisk is working fine as of nw
14:05.24Aurshmm.. but that bug is probably fixed in 1.2.4, Kernel_Core.. i dunno :P
14:06.48Aursastra^^: checked http://www.voip-info.org/wiki-SER+tips+and+tricks ?
14:07.44CurusDo any of you happen to have Elmeg 290 phones?
14:08.16*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
14:09.40chris_asthi all
14:12.16*** join/#asterisk exonic (n=exonic@209.172.11.54)
14:12.22powerchiphow I can make so if Agent take , the call not go to him?
14:13.26*** join/#asterisk KeX-NB (n=KeX-NB@ng1.kurtkrenn.com)
14:13.27powerchiptalk*
14:13.46KeX-NBhi
14:14.09KeX-NBborbably not the correct chan, but is someone using ser/openser with tls?
14:14.26*** join/#asterisk Tili (n=Tili@193.172.20.10)
14:15.15*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
14:17.08tdonahueCurus: if the picture I found of the Elmeg 290 is actually the phone, then it is probably a rebranded Snom 190.  We have had mixed reviews of the 190, people either love or hate the phone.
14:17.15*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
14:18.19*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)
14:18.20SpaceBassmorning
14:18.34SpaceBassanyone know why faxes I receive come in blank?
14:19.11chris_astSpaceBass: did u check codec it is using?
14:19.33bkw_doubt the codec is the issue
14:19.51chris_astreliable faxing only with G711
14:19.53Zeeekgood grief, bkw!
14:20.07Zeeekdoes asterlink still exist?
14:20.24SpaceBasschris_ast, its coming in on my zap trunk
14:21.29SpaceBassfyi as an aside- i fax out over a sip trunk and it works fine
14:22.57Aursbkw_: are you the one who wrote app_dbodbc.c?
14:23.23Zeeekbkw_ are you the one that wrote NEXT! ?
14:23.55chris_astSpaceBass, I have no idea on zap. The best I can tell you is check the codecs
14:24.03Aurshmm.. is he the one who is about to write /ignore aurs, Zeeek ? :P
14:24.26Zeeekwhat did you do for such mean treatment?
14:24.37chris_astCan anyone help me on Asterisk realtime
14:24.57Zeeekactually I'm an asterlink customer and I wondered if there was anyone at the controls
14:25.07Aursdunno
14:25.08Zeeekhaving written a coiple of times
14:25.26Zeeeknot that important, just curious
14:25.29Aurschris_ast, what do you want help with?
14:25.38chris_astcan someone please help me configure dialplan dynamically
14:26.16*** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131)
14:26.39bkw_yes thats me
14:26.53Fedoracore6hai all
14:26.54Zeeekwhere've you been hiding all these months/years?
14:27.00bkw_right here
14:27.05chris_astAurs, I have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf all the time
14:27.42chris_astwhenever I want to add new context I will do it in Mysql DB but without mentioning that in ext.conf asterisk is not taking it
14:27.42Aurschris_ast: use _patternX. things
14:27.43Aursah.. for new contexts
14:28.28chris_astAurs, that is the problem contexts will be changing frequently and we cannot change ext.conf manually all the time
14:28.50Aursbkw_: ok, what version was it written for? I'm using it (modified) on a 1.0.9 box
14:29.02chris_astAurs: please guide what could be done here
14:29.29willtchris_ast: probably not what you want but you could write a quick php scipt do grab info out of DB and right it out
14:29.38willts/right/write/
14:30.01Aurslol. nice bot
14:30.17willtjbot rocks! :)
14:30.22chris_astso with mysql asterisk we cannot do this
14:30.36chris_astI mean the add on we have
14:31.29Aurschris_ast: does this  help? http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
14:32.03Aursguess not, since you still have to write the context in the extensions.conf
14:32.47chris_astthat is the problem, we have to mention each and every context and also give switch to realtime under it
14:33.39chris_astAurs: so practically we add or change did numbers and all the time manually we cannot change ext.conf
14:33.43Aursyou could do a static realtime on the entire extensions.conf file chris_ast
14:33.55Aursbut then you will need a extensions reload after any changes
14:34.41Aursbut that makes it easier to make an automated gui for it
14:35.02chris_astjbot_: From php how I could I write rules with priorities to asterisk?
14:35.21Aursextensions.conf => mysql,database,table_name
14:35.35Aursin extconfig.conf
14:36.09Aurschris_ast: check this out: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static
14:36.34chris_astAurs: I will check that link, one sec
14:36.53dragonkhhi I have asterisk working but when I dial in asterisk says it cant create channel type 'Zap' - and when I pick up the phone I have dialtone and can call out
14:37.11dragonkhanyone know why this could be ?
14:37.38Zeeekdragonkh let's see the Dial command you are using
14:38.16dragonkhZeeek: er I installed that xorcom rapid asterisk - I'll look in the dialplan one sec
14:38.58*** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca)
14:39.04willtasterisk doesn't support saving voicemail in mp3 does it?
14:39.16frk2gsm is way better man
14:39.33willtwhy is it better? doesn't sound better to me
14:39.38[TK]D-Fenderfrk2 : Nah.. WAV.
14:39.39*** join/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br)
14:40.02_Paulo_~seen coppice
14:40.14jbotcoppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 2d 4h 39m 12s ago, saying: '"Nun"'.
14:40.15[TK]D-FenderWAV is inherently better quality and more prolific in its playback capability
14:40.15Aursgsm is not better for email attachments :)
14:40.20willtyes the wav49 sounded pretty good and is about the same size as gsm
14:40.25Zeeekgsm is better if you want faster loading
14:40.38*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
14:40.45Zeeekbut windows has a hard time playing gsm natively
14:40.59willtquicktime plays it
14:40.59asterisk99anyone know why I might get the message "chan_zap.c:6285 handle_init_event: Unable to play dialtone on channel 3
14:41.00asterisk99" when I lift handset of FXS attached phone?
14:41.00Zeeekand now a general OT announcement
14:41.33ZeeekI bought a book about Solaris 10 which sucked badly. I complained to the publisher and they are sending me a free book of my choice from their catalogue - cool, no?
14:41.34dragonkhZeeek: er I dont understand the thing heh - but tell me this - I have zap/1 and zap/4 - the phone is set to zap/1 and so is the trunk  - is this how it should be ?
14:42.01Zeeekask xorcom how it works
14:42.06willtZeeek: what publisher? lol
14:42.35ZeeekMcGraw-Hill - check out "The Complete Guide to Solaris 10" - now that I read the reviews I see why I felt screwed
14:42.46af_a good softphone for linux?
14:42.54ZeeekMy complaint letter was already contained in other's reviews!
14:43.02willtlol
14:43.14Zeeekhey, the squeaky whell gets the grease
14:43.22Zeeeks/whell/wheel/
14:43.28chris_astAurs: I feel that is not the thing we are looking for
14:43.57Zeeek~seen TheLight
14:44.00jboti haven't seen 'thelight', Zeeek
14:44.04Zeeektoo bad
14:44.09asterisk99Zeeek: What was wrong with it? Content missing? The grammer were bad? Full of spelling misteeks?
14:44.16Aurschris_ast: ok. why not? you could make a script that automatically inserts the magic into the table and does a extensions reload for you
14:44.22jsharp~seen Elvis
14:44.23jbotelvis <n=sdad@ipd50a583c.speed.planet.nl> was last seen on IRC in channel #debian, 33d 1h 9m 40s ago, saying: 'is there a combined package on debian to install all perl modules?'.
14:44.38Aursmagic as in "the context"
14:44.43Zeeek~seen Elvisleaveth Building
14:44.44jboti haven't seen 'elvisleaveth building', Zeeek
14:44.55Zeeeknyuk, nyuk
14:45.24Zeeekasterisk99 - it says Solaris 10 but has none of the actual SOlaris 10 differences
14:45.44Zeeekthere is a whole new concept in SOlaris 10 regarding the way the system starts up
14:46.06Zeeekit'd be like "Complete Guide to Asterisk 1.2" with no reference to dialplans
14:46.24Zeeekthere are 100 negative reviews confirming the same thing
14:46.37chris_astAurs: we do not want to reload many times
14:46.41Zeeekfor some reason I didn't see these on my first visit - because I bought from a local amazon
14:47.12chris_astAurs: I strongly feel there must be a way to tell Asterisk to search for contexts directly from mysql DB
14:47.17asterisk99anyone here have a working FXS phone??
14:47.21Zeeek~seen An Elephant Fly
14:47.23jbotZeeek: i haven't seen 'an elephant fly'
14:47.41ZeeekI have phones that work on FXS cards
14:48.00*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:48.12Zeeek~Zeeek
14:48.14jbotmethinks zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."
14:48.31Zeeekjeeze what a memory!
14:48.58chris_astjbot_: Please tell me how I could  write rules with priorities to asterisk from php?
14:49.00asterisk99Zeek: I have one that used to work and no longer does... I'm trying to figure out what I did wrong (probably something dumb) .... do you have your immediate=no or immediate=yes?
14:49.29Zeeekimmediate=no - yes tells it to jump in the context and go
14:49.33Zeeekwhen picked up
14:50.00asterisk99Zeek: You said that? Kewl!! BUt I rather think learning Asterisk isn;t quite as much fun as the other thing ;)
14:50.22Zeeekno, in fact it's actually a lot like masturbation, but I digress :)
14:50.22asterisk99Zeek: A little less messy, tho
14:50.31willtchris_ast: are you trying to get the contexts in extensions.conf to pull dynamicaly from mysql?
14:51.25Zeeekgotta run, asterisk99 I don't know what changed for your FXS
14:51.39chris_astwillt: yes I do want to mention them in extensions.conf but directly tell Asterisk to get them from Mysql DB
14:51.47asterisk99Zeek: K ... bye
14:51.53Zeeeklater
14:51.56*** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
14:52.04*** join/#asterisk viLeR (i=1000@66.128.47.232)
14:52.08kippifrom a * command is there away to set your extension to divert to voicemail or to another number?
14:52.32willtchris_ast: what do you mean mention them?
14:52.57caio1982kippi: diverting like in what situation?
14:53.12chris_astfor realtime we have give context name in ext.conf and under we have to write switch to realtime
14:53.12kippiwhenever a call comes in
14:53.33chris_astwillt: I do want to do this for every context
14:53.56willtchris_ast: this is for your did's?
14:54.06caio1982kippi: i guess i didn't get you meaning of "divert" then
14:54.15chris_astwillt: s for my DID's
14:54.20chris_astyes
14:54.43kippiso that if your away from your desk you can set a divert and it will either to your voicemail or mob
14:56.05*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
14:56.29caio1982a call forwarding you mean, right?
14:56.37kippiyeah
14:57.52caio1982kippi: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding
14:58.02kippithanks
14:58.06kippii'll have a read
14:58.22kippionce the page loads
14:58.52caio1982kippi: yeah, it's also a bit slow today for me
15:00.37*** join/#asterisk Theuni (n=ctheune@alphastar.gocept.com)
15:00.40Theunihowdi
15:01.23TheuniI upgraded asterisk last week (from 1.0 to 1.2) and with that change the music on hold became extremely loud (way overdriven actually). Is there a known itch during upgrade there?
15:04.16*** join/#asterisk michael-i (i=michael@141.41.38.185)
15:07.26*** join/#asterisk Deep6 (n=DEEP6@208.38.35.162)
15:07.58*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
15:08.40ManxPowerTheuni, Was there anything mentioned in UPGRADE.txt?
15:08.52tzafrirdragonkh, any problems?
15:08.57puzzledhi
15:13.09Kernel_CoreAurs: I found something...... when asterisks load increase this messages more repeats .... Mar 16 09:06:27 WARNING[12616]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb5885f08', 10 retries!
15:14.48ManxPowerKernel_Core, that is normal unless it impacts call quality.
15:15.02ManxPowerIt seems more common when doing Monitor, ZapScan, ChanSpy, etc.
15:15.41TheuniManxPower:  only that the format has changed. probably it's a problem of mpg123. I upgraded from 0.59r to 0.59s. That might be the problem.
15:16.07ManxPowerTheuni, only 0.59r is expected to work.
15:16.44Theunidowngraded already
15:16.47Theuniworks again. thanks anyway.
15:17.04*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
15:17.29*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
15:17.29Kernel_CoreManxPower:what about this ? Mar 16 09:10:46 WARNING[18766]: file.c:584 ast_readaudio_callback: Failed to write frame
15:18.22*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfigs.dialup.mindspring.com)
15:18.27tamp4xanyone here use spandsp? when i load asterisk -vvvvvvvvv  it stops loading when app_rxfax.so loads....any ideas why?
15:18.34*** join/#asterisk f7950qs0 (n=redhotte@61.17.213.99)
15:18.41f7950qs0hi all
15:18.45f7950qs0is fourcheeze here?
15:19.00f7950qs0damn I can see in the list if he's there or not
15:19.08*** join/#asterisk cas (n=cas@83.98.233.2)
15:19.41tzafrirhe's here according to the users list
15:19.50chris_astPlease tell me whether I can match a context using regular expression?
15:20.02tzafriralso try /msg dpkg seen username
15:20.48Kernel_CoreManxPower: even when I issue show channels I get this : http://pastebin.com/605508 , active calls should be 1/2 active channels .... but it isnot...
15:22.51f7950qs0tzafrir you use xorcom dont you
15:23.03f7950qs0i get your msgs in mailing list of xorcom
15:23.11f7950qs0rapid
15:23.19*** join/#asterisk MikeJ_ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
15:23.26*** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
15:23.50*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfigs.dialup.mindspring.com)
15:24.21Kernel_CoreMar 16 09:16:50 WARNING[19713]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files
15:24.21Kernel_CoreMar 16 09:16:50 WARNING[19713]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe!
15:24.28Kernel_Corehelp me :|
15:24.33f7950qs0does anybody know of a soft phone application that can configure one extension which can be assigned to a sip device
15:26.19Kernel_Coreres_agi.c:246 launch_script: unable to create fromast pipe: Too many open files :((
15:26.19justinuFor the third straight year, the Department of Homeland Security -- which is charged with charting the federal government's cyber security agenda -- earned a grade of "F" for computer security from a key congressional oversight committee, according to a story at Washingtonpost.com.
15:26.28justinuway to go, guys!
15:26.47*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
15:27.15tzafrirf7950qs0, yes, I am
15:27.22powerchiphow can i dio so if some client take to one , the not sent more phone to the client?
15:27.44powerchipdo*
15:27.55powerchiptalk*
15:29.08justinukernel_core, is that a problem with ulimits perhaps?
15:29.16f7950qs0hi justinu
15:29.21justinuyour kernel is either out of resources, or denying you resources
15:29.26[TK]D-Fenderpowerchip : You can do a "ChanIsAvail" test to see if they are on a call in the dial-plan OR just limit it on the phone (tell it to allow only 1 call)
15:30.41Kernel_Corejustinu: what is this ?
15:30.45Kernel_Corehow do I check it ?
15:30.47tzafrirjustinu, the legacy of the Symantec chief they had?
15:30.47powerchipI have try http://pastebin.com/605457
15:30.52powerchipbut not work:(
15:31.10powerchipwhat is wrong..:/
15:32.34justinuKernel_Core: ulimit -a
15:32.54justinutzafrir: lol, interesting theory
15:33.49Kernel_Corejustinu: http://pastebin.com/605546
15:34.22justinuok, so you have a 1024 open file limit
15:34.24powerchip[TK]D-Fender: u know?
15:34.26*** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com)
15:34.44justinutry changing it with ulimit -n 4096
15:34.46justinuor something
15:34.47f7950qs0show me one way of using asterisk as a call shop billing software
15:35.02justinushow me.... subscriber trunk dialing! <poof!>
15:35.19ManxPowerKernel_Core, yes, I get that too.  It does not cause a problem in my experience.  I assume you are using 1.2.x
15:35.20f7950qs07 concurrent calls and i want to meter them
15:35.24*** part/#asterisk cas (n=cas@83.98.233.2)
15:35.27f7950qs0they come out of my sip devices
15:35.38Kernel_CoreManxPower: running 1.2.5
15:35.57ManxPowerKernel_Core, what PROBLEMS is it causing?
15:36.29Kernel_CoreManxPower: calls being rejected by asterisk ....
15:36.48Lloydie-tI am trying to install res_sqlite3 but get error http://pastebin.ca/45925 Can you help?
15:37.03ManxPowerKernel_Core, then report it as a bug to bugs.digium.com.  If you neglect to include what PROBLEM it is causing the bug will be closed.
15:37.19[TK]D-Fenderpowerchip : You aren't use the "ChanIsAvail" application properly.  Read the instructions for it.
15:37.21ManxPowerKernel_Core, those messages are normal, rejecting calls is not.
15:39.09powerchip[TK]D-Fender: ok
15:39.23ManxPowerKernel_Core, you must have MANY MANY calls to be running out of FDs
15:39.47Kernel_CoreManxPower: I am useing chan_h323 ...
15:39.48ManxPowerKernel_Core, or your AGI is buggy and not exiting correctly
15:40.00vuudFrom a SIP phone attached to *, what is the best way to access the "s" extension?
15:40.04Kernel_CoreManxPower: I am useing astbill for accounting...
15:40.12ManxPowervuud, there's isn't.
15:40.33vuudoh wait - how about I put all that as an ext and goto it from the s exten
15:40.46ManxPowervuud, the "s" extensions is run when asterisk does not receive any information about the destination of the call (usually only when using analog fxo ports)
15:40.49chris_asttzafrir, Please tell me whether I can match a context using regular expression?
15:41.01vuudManxPower: right, I want to test menus and such
15:41.16ManxPowervuud, exten => 666,1,Goto(s,1)
15:41.18tzafrirchris_ast, I can't think of a simple way
15:41.40ManxPoweron MY systems, I do more of exten => s,1,Goto(2101,1)
15:41.43ManxPowerwhere 2101 is my IVR
15:42.05vuudManxPower: yeah, i just started changing to that
15:42.09*** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116)
15:42.28*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
15:42.35*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:42.35*** mode/#asterisk [+o anthm] by ChanServ
15:42.53ManxPowerexten => s really should be changed in the source code to be exten => unknown because "s" is confusing.
15:43.52chris_astManxPower, I could not get it clearly
15:44.10Kernel_CoreManxPower:  is it normal ?! frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end
15:44.13iDunnos for "starting" ;)
15:44.20[TK]D-FenderManxPower : Then we'll need to make a new one for macro's as rell... and so on...
15:44.24ManxPowerKernel_Core, THAT message is normal.
15:44.33ManxPowerKernel_Core, it looks like you have three totally different messages.
15:45.34ManxPower[TK]D-Fender, even in my macros I many times do exten => s,1,Goto(${MACRO_EXTEN},1) so my CDRs look good.
15:45.36Kernel_CoreManxPower: this message is repeating too fast .... Dropping extra frame of ....
15:45.43Lloydie-tAny thoughts on http://pastebin.ca/45925 ?
15:45.52ManxPowerKernel_Core, that is caused by buggy G729 devices.
15:45.54chris_astcan somwone please help me
15:46.16ManxPowerKernel_Core, you can just comment out that message in the source code if you want, or just run at a less verbose debug level
15:46.36ManxPowerchris_ast, I did not see a question from you.
15:47.01ManxPowerLloydie-t, are you compileing asterisk as root?
15:47.01Kernel_CoreManxPower: OKey
15:47.29Lloydie-tManxPower: Yes I am
15:47.32Kernel_CoreManxPower: hehe asterisk crashed at last !!!!!
15:47.35tzafrirchris_ast, perhaps you should state the reason for which you need regex matching of context name. As noone here seems to know how to do that
15:47.44ManxPowerKernel_Core, it will do that if you run out of FDs
15:47.56Kernel_CoreFDs ?
15:48.01justinusigh
15:48.05willt;)
15:48.05Kernel_CoreFile Descriptror ?
15:48.12ManxPowerLloydie-t, perhaps astxs is not set to be executable.
15:48.16ManxPowerKernel_Core, correct.
15:48.20justinuwith a name like kernel_Core, you'd think you'd know about the services a kernel provides
15:48.23justinuat least a little bit
15:48.24Kernel_CoreManxPower: how do I increase it ?
15:48.29Kernel_Core:))
15:48.30justinulol, i told you that already.
15:48.34justinupay attention
15:48.54chris_asthere is my need, I have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf all the time
15:49.00chris_astwhenever I want to add new context I will do it in Mysql DB but without mentioning that in ext.conf asterisk is not taking it
15:49.02ManxPowerKernel_Core, justinu already told you how to do that.
15:49.12ManxPowerEach call should use no more than 4 FDs.
15:49.14Kernel_Coreulimit ... ?
15:49.20ManxPowerif you are using more than that there is a bug somewhere.
15:49.29ManxPowerKernel_Core, correct, before you start asterisk
15:49.30Lloydie-tManxPower: I a linux newbie. How would I set that to be an executable?
15:49.41Hmmhesays1000 did's huh?
15:49.42Hmmhesaysfun
15:49.50justinu1000 dids? weak
15:49.53justinui've got 20k +
15:49.59ManxPowerLloydie-t, I only help with Asterisk questions.  Yours is a basic *nix question.
15:50.14chris_astjustinu, how are you managing
15:50.24justinujust fine
15:50.37justinuextensions reload is starting to take an appreciable amount of time to complete
15:50.38Lloydie-tNo problem, I'll check on google
15:50.42justinubut it seems to run just ifne
15:50.43Kernel_CoreManxPower: so so what should I do now .... :|
15:50.44fourcheezeLloydie-t: chmod +x somefile
15:50.46puzzledManxPower: trade basic *nix knowledge for stroopwaffels and you should soon have that large crate
15:50.59ManxPowerKernel_Core, I recommend buying a book on basic linux
15:51.05Kernel_Core:))
15:51.17ManxPowerpuzzled, THAT is the BEST idea I've heard this week.
15:51.17chris_astjustinu, I do not want to relaod all the time, can't we give everything in mysql DB
15:51.21Kernel_CoreManxPower: it is long time which I don't use linux tooooooo much ...
15:51.27justinuchris_ast: yes
15:51.28willtKernel_Core: or start using the man pages
15:51.50ManxPowerPatience.  You must find your own path to Linux nirvana, Grasshopper.
15:51.54Lloydie-tYeah, I got the redhat bible
15:52.09justinuman pages, there's an idea!
15:52.38chris_astjustinu, even updating contexts is also a problem
15:52.38Hmmhesayslinux for dummies 2nd edition is a good book
15:52.49ManxPowerYou REALLY do need a basic understanding of Linux for you to be able to install, troubleshoot, and manage an Asterisk server.
15:53.09Delvarcan somone help, how can i access ${CDR(billsec)} after a DIAL? i always get 0 back, even though the CDR has a value in the database afterwards.
15:53.25f7950qs0can anyone tell me if I can get this work (www.callerid.com) done by asterisk
15:55.08chris_astplease tell me how can we have contexts dynamically?
15:55.14justinurealtime
15:55.18Hmmhesaysrtfm dude
15:55.21Hmmhesaysseriously
15:55.28dragonkhhow can I improve the sound quality? people sound far away ??
15:55.31tamp4xanyone here use spandsp? when i load asterisk -vvvvvvvvv  it stops loading when app_rxfax.so loads....any ideas why?
15:55.37fourcheezechris_ast: why is updating contexts a problem?
15:55.44jarroddragonkh: thats usually gain levels
15:55.55Hmmhesaystamp4x: set debug in logger.conf
15:55.59*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
15:56.01Dr-Linuxhi
15:56.05Dr-Linuxerror
15:56.06Dr-LinuxMar 16 07:42:26 WARNING[7029]: channel.c:784 channel_find_locked: Avoided initial deadlock for '0x8d63898', 10 retries!
15:56.08Hmmhesaystheres trouble
15:56.11Dr-Linuxwhat does this error mean?
15:56.26jarroddr-linux: does that hose your box?
15:56.27*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com)
15:56.30tahorgast_rtp_read: Unknown RTP codec 115 received
15:56.31dragonkhjarrod: where do I set them ?
15:56.45Dr-Linuxjarrod: what is "hose" ?
15:56.46tahorganyone can tell me what codec 115 is ?
15:56.48Lloydie-tfourcheeze: thanks for that. got a new problem though. I'll try and suss it myself though
15:56.53Kernel_Corenote that if you do not set the ulimit -n 100000 or something similar efore you start asterisk you'll run out of FD's around 151 calls. :))
15:57.01tamp4xdebug => debug    is set
15:57.04jarroddr-linux: the asterisk box becomes unusable
15:57.10*** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com)
15:57.13tamp4xi assume /var/log/asterisk/debug willt ell me
15:57.26Dr-Linuxjarrod: my everything is working fine
15:57.29justinuDr-Linux: unless you're having a problem, it's normal to see that occasionally
15:57.44Dr-Linuxbut when i dial this extension i see this error
15:57.47Dr-Linuxhi justinu
15:57.54SwK[Work]anyone know what protocol Mitel 5020 IP phones use?
15:57.57tamp4xdebug tells me nothing hmmhesays
15:58.04jarrodswk: i believe its proprietary
15:58.05Dr-Linuxjustinu: so it's not a bug ?
15:58.10jarrodso it wont interoperate with *
15:58.12justinuno
15:58.26Dr-Linuxi just wornder if its makes bad my box
15:58.38SwK[Work]jarrod: you know I should know this.. mitel and sprints IP-PBX is the same platform with a different label onit heh
15:58.42Dr-Linuxbut actually what does this warning MEANS?
15:59.27SwK[Work]<PROTECTED>
15:59.28*** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-200-134.miamfl.adelphia.net)
15:59.31chris_astPlease tell me how I configure conexts in ext.conf dynamically
15:59.36SwK[Work]damn its amazing what you can find on google
16:00.07jarrodchris: dynamically?
16:00.38chris_astyes I want dynamically
16:01.23chris_astjarrod, I do not want to give them in ext.conf but directly from mysql DVB
16:01.26chris_astDB
16:01.39jarroduse realtime
16:01.44AlexCTIHi, I need some help, I update my server to the new version and I got this MSG and * doesnt come up Mar 16 11:55:01 WARNING[3185]: loader.c:499 load_modules: Loading module chan
16:01.44AlexCTIdem.so failed!
16:02.06chris_asteven with realtime I have to mention them in ext.conf
16:02.20jarrodand there are two types of realtime I believe... one that loads it into memory upon reload/startup, and one that reads it dynamically
16:02.36*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
16:02.38jarrodchris_ast: incorrect - you can go to a full sql config
16:03.21chris_astjarrod, everthing from sql?how can we do that?
16:04.25jarrodchris: http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database
16:05.53chris_astjarrod: asterisk needs to be reloaded all the time
16:06.16fourcheezechris_ast: extensions reload
16:06.17jarrodExternal configuration is configured in /etc/asterisk/extconfig.conf allowing you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload.
16:06.22Lloydie-tstill got problem with res_sqlite, tried to suss it for nyself, not sure where to start http://pastebin.ca/45928
16:06.47f7950qs0HI FOURCHEEZE
16:06.50f7950qs0oops
16:06.52f7950qs0caps on sorry
16:06.53fourcheezechris_ast: I'm using realtime myself but I don't go as far as putting my extensions in there
16:07.11fourcheezechris_ast: but running extensions reload shoudln't give you too many problems
16:07.37fourcheezef7950qs0: erm hello
16:07.49f7950qs0there's an identifier machine on www.yes-tele.com and i want that work done through some software
16:07.56chris_astjarrod,fourcheeze: even for realtime we need to include context name in ext.conf and under it we have to give switch to realtime, I want to avoid even that
16:07.57f7950qs0how?
16:08.14fourcheezechris_ast: are you putting your extensions.conf into realtime?
16:08.18jarrodyou don thave to specify in two locations
16:08.24f7950qs0my girlfriend has got visas to come to me
16:08.34jarrodif you specify extconfig to look for extensions in sql via realtime
16:08.37chris_astfourcheeze: reload is not a problem but we update or create contexts very frequently
16:08.46f7950qs0i'm happy
16:09.06f7950qs0o.k. back to work anybody had a look at that site?
16:09.08fourcheezef7950qs0: where are you - guantanamo bay ? ;-)
16:09.23SpaceBassanyone using dring?
16:09.32f7950qs0what's that
16:09.38f7950qs0i'm in India
16:09.50chris_astfourcheeze: extensions.conf is realtime, I mean context rules are defined in mysql db
16:10.53chris_astjarrod: even if we specify in extconfig.conf we have include context name and swith to realtime in ext.conf
16:11.06fourcheezechris_ast: what does your extconfig.conf look like?
16:11.42*** join/#asterisk jmacz (n=jmacz@201.244.241.189)
16:11.55Hmmhesayslook at me cause i ain't wearing no frown
16:12.42fourcheezef7950qs0: it looks like the yes telecom thing is just what you were on about yesterday
16:12.48f7950qs0i'm getting tired of my cafe
16:13.02f7950qs0is there anyone here providing asterisk consultancy (configuring) services in India?
16:13.11f7950qs0a day's work for an expert
16:13.12Hmmhesayswhy do you need to be in india?
16:13.26f7950qs0because I haven't yet got a green card to be in the US
16:13.28Hmmhesaysthats the beauty of ssh
16:13.42f7950qs0the call shop is in India
16:13.52chris_astfourcheeze, any suggestion for me
16:13.55Hmmhesayswhy do you need someone physically there?
16:13.55asterboygreat more outsourcing.
16:14.00asterboyno thanks.
16:14.01fourcheezef7950qs0: if you set up a basic debian box and get ssh access people can be anywhere
16:14.10f7950qs0i dont have to have someone physically here.
16:14.15asterboyTired of every call I make giong to India.
16:14.21Hmmhesaysok what problem are you having?
16:14.25Hmmhesaysor what do you need done?
16:14.27fourcheezechris_ast: show us yout extconfig.conf
16:14.33f7950qs0:)) india BPO crazy asses
16:14.40willt~pastebin
16:14.41jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste
16:14.47gniretar_workno way i'm gonne be involved in another buissness going to India
16:15.11SpaceBasscan someone take a look at my zapata.conf http://pastebin.ca/45929  when I use dring EVERYTHING goes to the zap/2's context
16:15.40asterboyThe US has an 8 Trillion dollar deficit...can't see how they can afford to outsource.
16:15.45f7950qs0how much would somebody charge me for this thing? i want metering for my sip devices that's it
16:15.49chris_astfourcheeze: here it is
16:15.56chris_astextensions => mysql,asterisk,extensions_table
16:16.11Hmmhesaysf7950qs0: metering?
16:16.23fourcheezechris_ast: are you using the dynamic or static realtime configuration?
16:16.33chris_astdynamic
16:16.43fourcheezef7950qs0: what you're talking about is a basic asterisk installation
16:16.48*** join/#asterisk file[laptop] (n=jcolp@142.131.190.116)
16:16.54Hmmhesaysi wouldn't say that quite yet
16:17.31chris_astI did all the things shown in voip-info but this thing is not given there
16:17.40f7950qs0that's it? dint quite feel that easy to me fourcheeze
16:17.46*** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net)
16:17.56Hmmhesaysf7950qs0: explain what you want done in more detail
16:17.58fourcheezef7950qs0: get me a debian box and I'll install asterisk for you
16:18.05TheCompWizhas anyone tried to use a nortel pbx as a voip termination?
16:18.08fourcheezeand I'll only charge $100
16:18.15chris_astcan I use reges for context name? Is this possible?
16:18.20Hmmhesaysfourcheeze you're going to end up eating those words
16:18.30fourcheezemaybe :-)
16:18.42fourcheezenever done it before
16:18.43*** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net)
16:18.49willtomg lol
16:19.00Hmmhesaystake my advice, find out what he wants done because "metering for sip devices" means nothing
16:19.11Hmmhesaysand never ever charge a flat fee
16:19.13fourcheezeHmmhesays: well if it's what he was on about yesterday
16:19.25fourcheezeit means storing cdr inmysql
16:19.26f7950qs0Hmmhesays: i have an internet call shop people come in and use my sipdevices (connected to analog phones) when the call ends i need to meter how long did they call that's it. just wanna know the minutes they utilized
16:19.37*** join/#asterisk salviadud (n=ralfalfa@201.137.161.31)
16:19.41Hmmhesaysso you want to set up call detal records
16:19.43fourcheezef7950qs0: if I give you a mysql port somewhere are you able to configure something to talk to it?
16:19.59fourcheezef7950qs0: like windows obdc drivers?
16:20.03f7950qs0i can use my telephony provider's online webpage to track the minutes but they dont provide instant online billing
16:20.10salviadudi got this prank call i recored with mixmonitor
16:20.14*** join/#asterisk Altair256 (n=icechat5@mail.clccorp.com)
16:20.16f7950qs0i dont know what is windows odbc drivers i'm sorry
16:20.18salviadudanybody want to hear it?
16:20.25justinuf7950qs0: install "asterisk-stat"
16:20.29chris_astfourcheeze: can u suggest me something on this?
16:20.37Hmmhesaysf7950qs0: thats a fairly simple setup
16:20.41*** join/#asterisk Aurs (i=aurs@hallo.aurs.info)
16:21.08f7950qs0practically i dont know anything about asterisk and dont think even wanna know just for the purpose of cafe someday will learn just for learning
16:21.08TheCompWizanyone in here work with nortel systems?
16:21.36Hmmhesaysyou know anything about linux ?
16:21.42NuggetLinux is poo.
16:21.48willtf7950qs0: do they pay you cash or charge with credit card?
16:21.50salviadudfreebsd is poo
16:21.51f7950qs0a little bit
16:21.54salviadudwith a devil log
16:21.56salviadudlogo
16:21.56f7950qs0they pay me cash
16:22.35Fedoracore6hai all i was problem i my updatecode  to databases .. any one can help me what my false in this code
16:22.36Fedoracore6http://pastebin.com/605643
16:22.37f7950qs0I charge them two cents or less per minute to use my cafe shop's cabins and the phones and they dont fucking pay me sometimes becuase i can't track if the call went through or not
16:22.56salviadudyou dudes know how many formats does mixmonitor support?
16:23.24f7950qs0call shop sorry not cafe shop
16:23.28AlexCTISomeone can help me to fix this error: Mar 16 12:16:06 WARNING[8712]: loader.c:499 load_modules: Loading module chan_modem.so failed!  happend after update.
16:23.32Fedoracore6cos i press like example 2222 " code for subject computer" but can put in databases
16:24.24*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
16:25.05SpaceBasscan someone take a look at my zapata.conf http://pastebin.ca/45929  when I use dring EVERYTHING goes to the zap/2's context
16:25.38ManxPowerAlexCTI, That is why "make install" warns you about those modules.
16:27.44ManxPowerin fact UPGRADE.txt ALSO talks about those modules.
16:27.56ManxPowerAnd that issue has come up on this channel many times.
16:28.02ManxPowerand it's been talked about on the mailing lists.
16:28.15snip3rHi folks
16:28.33snip3rwonder if you remember my problem with asterisk
16:28.56salviadudwe're not here to remember
16:28.59salviadudwe forgive and forget
16:29.11ManxPowerwell forget at least.
16:29.20salviadudyeah
16:31.48Fedoracore6:)
16:32.05chris_astcan someone please help me solve my problem?
16:32.10AursFedoracore6: what are the datatypes on the fields you are updating in mysql?
16:32.52*** join/#asterisk digg10 (n=john@206-248-135-54.dsl.teksavvy.com)
16:32.58chris_astAurs: still I could not get a clue on my problem :(
16:33.12Aurschris_ast: the extensions thing?
16:33.19chris_asts Aurs
16:34.19AursFedoracore6: the '$clid' in the sql can cause the error if the clid field is a integer (or some other number-type). that was my thought.
16:34.30digg10is it possible to do time_of_day routing with extensions.conf being loaded from the database?
16:35.03Aurs*food*
16:36.14*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
16:40.25*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
16:41.07Zeeekoh my
16:42.25Fedoracore6i wanna update kodsubjek2 in my code
16:42.25*** join/#asterisk tuxinator_linux (n=tuxinato@142.131.190.116)
16:43.23*** join/#asterisk thdei (n=DD@nat1.cri74.org)
16:43.37thdeihello everybody
16:43.45thdeiI have a question about a problem
16:43.54thdeiSometimes, when I call someone out of my company, i have the message below
16:43.59thdei-- Called ISDN1/0450047588
16:44.01TheCompWizdoes anyone have any experience with nortel equipment?
16:44.07thdei<PROTECTED>
16:44.15thdei<PROTECTED>
16:44.22thdei<PROTECTED>
16:44.28thdei<PROTECTED>
16:44.35thdeiDo you have a idea for me ?
16:44.54SpaceBasscan someone take a look at my zapata.conf http://pastebin.ca/45929  when I use dring EVERYTHING goes to the zap/2's context
16:45.00ManxPowerthdei, what is the value of DIALSTATUS?
16:45.21ManxPowerSpaceBass, very few people use dring so not many people will be able to help you.
16:45.22[TK]D-FenderTheCompWiz : Sure, try asking the more direct question......
16:45.32thdeiManxPower, How a know this ?
16:45.58ManxPowerthemacuser, Noop(DIALSTATUS=${DIALSTATUS}) as the priority after the DIAL
16:46.10*** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
16:46.12thdeiok, I try
16:46.18Zeeekhey ManxPower you coming to Astrocn Eu this time?
16:46.30Zeeeks/Astrocn/Astricon/
16:46.30ManxPowerZeeek, no.
16:46.38TheCompWiz[TK]D-Fender ... do you know how to configure the meridian box to accept connections from asterisk & use it to dial out?
16:46.40AlexCTIManxpower, Thanks I'll read the docs about this
16:47.01TheCompWiz(in/out rather)
16:47.03ManxPowerZeeek, too many post-Kartina things to deal with.  I may not even have any holiday this summer.
16:47.14SpaceBassManxPower,  I know...trying to space that apart so as not to be "that guy"...just hoping someone has a clue
16:47.16justinuTheCompWiz: use a T1 PRI to interconnect your PBXs
16:47.18justinuif you can.
16:47.27SpaceBassManxPower, I'm basically with out fax until I can get it fixed....
16:47.28Fedoracore6Aurs: its i wrong using the $clid
16:47.32*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
16:47.34TheCompWizjustinu ... can I use an ethernet interface instead?
16:47.39Fedoracore6or i must change the other feild
16:47.42SpaceBassunless, is there a way for asterisk to just recognize fax as soon as it answers the zap trunk?
16:47.44ManxPowerSpaceBass, the mailing list archives were not helpful?
16:47.52ManxPowerSpaceBass, it can do that.
16:47.55ManxPowersee exten => fax
16:48.07ManxPoweralso see faxdetect in zapata.conf
16:48.16justinuTheCompWiz: doubt it... legacy PBXs use analog or T1/E1 interfaces usually
16:48.33SpaceBassManxPower, didn't find anything there, but just posted to the forums for A@H on SF
16:48.35f7950qs0time for me to leave. wont come to bug you guys again. will come here to hang out though
16:48.43SpaceBassManxPower, thanks!
16:48.54ManxPowerSpaceBass, you are using Asterisk@Home?  Pervert.
16:48.58TheCompWizjustinu... this box of mine isn't really "legacy"
16:49.07justinumeridian is legacy
16:49.28ZeeekManxPower have you (sucessfully) relocated?
16:49.33justinuunless your meridian happens to speak SIP
16:49.38TheCompWizsupposedly... this pbx box of mine is already voip capable...
16:49.43TheCompWiz(I'm guessing SIP)
16:50.00justinusure... anything is VoIP capable if you plug a media gateway into it.
16:50.07TheCompWizheh.
16:51.08Zeeekthere are whole sites about using @hole
16:51.24SpaceBassManxPower, i was orignally drawn to A@H out of lack of expirence and the cool wake up call feature...next box I build will be from scratch...but I still like freepbx for adding and managing extensions and users
16:52.16thdeiManxPower, I have DIALSTATUS=CHANUNAVAIL
16:52.18*** join/#asterisk fugitivo (n=ajf@201.255.177.17)
16:52.31ManxPowerZeeek, I am, however, willing to consider accepting consuling jobs now.
16:52.38[TK]D-FenderTheCompWiz : Careful, it may be a proprietary format though.  Confim your exact scenario first....
16:52.49ManxPowerthdei, that means Asterisk cannot complete the call.
16:52.51thdeiBut as I say that don't happen each time
16:52.57*** join/#asterisk T-Squared (n=T-Square@hidden.serreyn.com)
16:53.04ZeeekManxPower that's good to know!
16:53.11*** part/#asterisk T-Squared (n=T-Square@hidden.serreyn.com)
16:53.12ManxPoweror it means that your PROVIDER can't process the call.
16:53.27*** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc)
16:53.31ManxPowerZeeek, I'm not cheap. 8-)
16:53.45thdeiManxPower, do you know what can do that  Asterisk cannot complete the call.
16:54.01ManxPowerthdei, no.
16:54.02Zeeeksomeone asked me yesterday about recovering from Pickup() when the extension is NOT ringing, e.g., play a message when they arrive at Pickup(). It just hangs up if not ringing.
16:54.14ManxPowerPerhaps your provider only allows you 1 call at a time?
16:54.21ManxPowerperhaps your provider does not have enough lines?
16:54.47fugitivoanyone had this problem before?
16:55.02thdeino, I have 3 rnis lines and before, using a 3com system, there was no probleme .
16:55.03fugitivochan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
16:55.05Zeeekactually the provider usually give a CONGESTION status when it can't happen
16:55.09ManxPowerfugitivo, yes, we have problems with people not telling us their problem.
16:55.20ZeeekHelp me someone please!
16:55.26ZeeekNewbie question!
16:55.29ZeeekCan I ask it?
16:55.31justinulol
16:55.32ZeeekOr not?
16:55.36justinujust ask it
16:55.38justinuwtf?
16:55.46ManxPowerfugitivo, that means data was lost between the card and the zaptel driver.  usually something is locking interrupts for too long.  common causes of this can be IDE, SATA, on board Ethernet, RAID, etc.
16:55.54Zeeekis that your question? You only get one you know!
16:56.02thdeithnks for your help ManxPower, i will search more
16:56.03ManxPoweranything that wants to take control of the PCI bus to improve performance.
16:56.05justinuheh
16:56.13fugitivoManxPower: great
16:56.26fugitivoManxPower: interrupts?
16:57.15ZeeekLove the mailing list thread "must distinguish between voIP and voPI"
16:58.37fugitivoManxPower: and this? sorry, this is my first isdn pri (Don't know what to do with control frame 15)
16:59.07ManxPowerfugitivo, you said you were having HDLC messages, not control frame messages.
16:59.17ManxPowerignore the control frame messages.
16:59.22jbalcomb[TK]D-Fender How long do you estimate needing to revamp our dialplan?
16:59.31*** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it)
16:59.35Micettohi
16:59.38Micetto:)
16:59.44justinufugitivo: you've either got a hardware problem with your zaptel card, or your T1 is not deriving timing from the loop correctly.
16:59.44Zeeek:(
16:59.58Micettocan anyone help me with chan_zap and asterisk to reiceve a fax from isdn interface?
17:00.12Micettohow can I do this ?
17:00.24SpaceBasscan you technically get a fax over isdn?
17:00.33Micettoyes
17:00.33justinuof course
17:00.37justinufaxes go over ISDN all the time
17:00.50Zeeekbut do they ever come back?
17:00.56Micettobut I have set "faxdetect=both"
17:01.06SpaceBassi just didn't know...cool
17:01.08[TK]D-Fenderjbalcomb : within a week.  I'm moving next weekend though.
17:01.41Hmmhesaysi guess i have to move
17:01.42Hmmhesaysthat sucks
17:01.43Micettoand if in CLI type "zap show channel 13"
17:01.43Hmmhesaysi hate moving
17:01.53fugitivojustinu: how can i check if it's a problem with the card?
17:01.55MicettoI see Fax Handled: no
17:02.33justinufugitivo: check to make sure nothing is sharing IRQs w/ your card
17:03.01Micettopuff....
17:03.04Micetto:)
17:03.22Micetto!@#....with asterisk!!!
17:04.03MicettoI'm going crazy to get a signal fax from my ISDN card!!!
17:05.28Micettohelp me! :D
17:05.30Micetto^_^
17:06.12tzafrirMicetto, how about giving some actual details as opposed to ranting
17:06.24justinulol
17:06.30Zeeekmore fun to try to guess the problem
17:06.48tzafrirVersion of Asterisk, They type of ISDN channel, error message you get, etc.
17:07.03Micettook
17:07.19MicettoAsterisk take from bristuff-0.3.0-PRE-1k.tar.gz
17:07.31fugitivojustinu: could this be the same problem? No D-channels available!  Using Primary channel 16 as D-channel anyway!
17:07.31MicettoDriver that I use is qozap
17:07.43Micettowith my quadBRI card
17:07.47justinufugitivo: yeah, d-channel is completely down now.
17:07.48Micetto(Cologne Chipset)
17:08.06MicettoI don't get any error message....
17:08.16MicettoMy problem is that fax is not handle!!!
17:08.18fugitivojustinu: that's not a config problem?
17:08.34ManxPowerfugitivo, HDLC errors are come of the hardest problems to solve with Asterisk
17:08.40justinufugitivo: is it a new setup?
17:08.44ManxPowersome of  that is
17:08.47fugitivojustinu: yes
17:08.53justinufugitivo: so it never worked?
17:09.02fugitivojustinu: i was working with R2 but never received those errors
17:09.09justinufugitivo: look into IRQ conflicts first.
17:09.15justinucat /proc/interrupts
17:09.16Micettotzafrir: in default context exist a fax extensions that execute a NoOp if a fax is received
17:09.43Micettotzafrir: but NoOp it's never been executed
17:09.59SpaceBassManxPower, you didn't tell me that faxdetect just doesnt work
17:10.01SpaceBass:)
17:10.26ManxPowerSpaceBass, it does work
17:10.38ManxPowerit can be confusing, of course.
17:10.59SpaceBassManxPower,  I'm sure thats the case....I found a little about it on the wiki...
17:11.50SpaceBassi have it set to =incoming but its just sending it straight into the dialplan for voice calls...from what I'm reading if the fax tone doesnt happen to beep right as the zap channel picks it up then it doesnt know its a fax
17:11.55ManxPoweryou, of course, have to answer the line for asterisk to hear the fax.
17:12.36ManxPowerthen you need to do something like Ring and then something like background or waitexten for a few seconds
17:12.36SpaceBassthere is an answer in the dial plan
17:12.49SpaceBassyeah, I'd need to build in a wait
17:13.12tzafrirMicetto, do you try to use the fax detection option of zap?
17:13.12Zeeektake a look at this: http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk
17:13.30Micettoyes
17:13.33Zeeekit works
17:13.47Micettotzafrir: faxdetection=both
17:13.51ManxPowerif you want more control then you can build and install NVFaxDetect, but most people don't need that.
17:13.57*** join/#asterisk enzo123 (n=enzo123@200.sub-70-197-78.myvzw.com)
17:14.14ManxPowerMicetto, you don't want that.  Then asterisk will send an OUTGOING fax to the asterisk fax extension instead of the PSTN
17:14.57ManxPowerwell COULD, rather than "will"
17:15.32Zeeekyou know what, we get faxes and there's no answer() to be seen anywhere
17:15.35MicettoManxPower: and how I set up faxdetection ? which value should I use ?
17:15.50ManxPowerMicetto, usually incoming
17:15.56*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
17:15.56*** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.209)
17:15.59Kernel_Corehi alll
17:16.12MicettoManxPower: ok, now I'm testing
17:16.14ManxPowerZeeek, playback, background, and other apps answer the line
17:16.21ZeeekI just have exten => fax,goto(real_fax_area,1) in the beginning of the context
17:16.41Fedoracore6i wanna do update system to databases , its i must using the $query_result1 = @mysql_query($query1);
17:16.50ManxPowerZeeek, that won't work.  you don't have a priority
17:16.58Zeeekthe scottstuff site I gave above has details on how to receive faxes and they work
17:17.02MicettoManxPower: where I can sure that faxdetection is enabled ?
17:17.12ManxPowerMicetto, zapata.conf
17:17.22Zeeekyeah, well I'm bad at typing, but it's there and we get all the junk faxes assholes insist on sending us
17:17.23MicettoManxPower: ok, and in CLI ?
17:17.36ManxPowerMicetto, I don't know.
17:17.48Micetto:(
17:18.08Zeeekshow fax detection :)
17:18.12Micetto#@!#@@!#!!#....
17:18.14ManxPowerif it's set it's enabled
17:18.20Kernel_CoreManxPower: I changed my ulimit and /proc/sys/fs/file-max , after 69 calls , I get this error http://pastebin.com/605728 :|
17:18.28SpaceBassMicetto,  I can emphatize...thats about how I feel
17:18.42Zeeekhas anyone read the article I posted above?
17:18.54SpaceBassfrom Scott? yeah i had it bookmakred even
17:19.02Zeeekwell it works perfectly
17:19.08MicettoZeeek: but it is an old article :P
17:19.09Kernel_CoreManxPower: after that , users will hear busy ...
17:19.21ZeeekMicetto so what? It works
17:19.42*** part/#asterisk chris_ast (n=Administ@59.93.56.163)
17:20.06ManxPowerKernel_Core, and what does ulimit -a say for open files
17:20.19*** part/#asterisk enzo123 (n=enzo123@200.sub-70-197-78.myvzw.com)
17:20.24MicettoManxPower: don't work....:'((
17:20.33justinupoor manxpower
17:20.46Zeeekbeseiged with questions
17:20.47justinumad n00b kung-fu going on
17:20.56Micetto<PROTECTED>
17:21.00ManxPowerjustinu, eventually I get frustrated and stop answering questions
17:21.01Zeeekquestions that have been answered a million times elsewhere
17:21.06justinuManxPower: i know the feeling
17:21.07Micetto18:19:52 16/03/2006 DEBUG[2659]: channel.c:797 channel_find_locked: Avoiding initial deadlock for 'Zap/1-1'
17:21.15SpaceBassn00b would be "can I use my usb modem with asterisk as an answering machine?"
17:21.17ManxPowerMicetto, if you flood the channel with crap I'm going to bitchslap you.
17:21.23justinulol
17:21.39MicettoManxPower: sorry ^_^'
17:21.45ManxPower~pb
17:21.50jbothmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
17:21.51Zeeekhow do I know if bitchslap() is installed on CLI?
17:22.01justinushow application bitchslap
17:22.03justinun00b
17:22.04justinu:P
17:22.09Micettoeheheh
17:22.13[TK]D-FenderZeeek : its all about the arguements ;)
17:22.23ZeeekI can't read the description, it's in dark green!
17:22.30SpaceBassLOL
17:23.21ManxPowerHere's the basic idea.  If Asterisk is waiting for DTMF then it can detect a fax.  If it's not listening for DTMF, then it can't detect a fax.
17:23.55SpaceBassthen wouldn;t it stand to reason that if an IVR answered and faxdetect=incoming it would pick it up?
17:24.10MicettoManxPower: ok
17:24.17bkw_ManxPower, not entirely true
17:24.23bkw_you can play a file and do fax detect
17:24.27ZeeekSpaceBass not only an IVR but a human as well
17:24.30bkw_without the need to wait or even think about DTMF
17:24.33ManxPowerSpaceBass, Yes, that should work fine as long as you have an exten => fax AND you are thinking of "incoming" from the perspective of Asterisk
17:24.47ZeeekI've answered the phone only to hear the fax tone and then have asterisk hang me up to recv the fax
17:25.07Zeeekalso works in vmail
17:25.09SpaceBassI've tried both...answering and sending to IVR...in all instances asterisk never detects it
17:25.18ManxPowerbkw_, I was not aware of that.  I use NVFaxDetect.
17:25.26bkw_faxdetect needs to be on btw
17:25.26SpaceBassand Ricoh is probably tried of me using their faxback to test :)
17:25.30bkw_its not usually on by default
17:25.32Zeeekare you using genuine Inte^h^h^h Digium hardware?
17:25.37ManxPowerSpaceBass, using ZAP interfaces, right?
17:25.37Micettoand NVFaxDetect works ?
17:25.42SpaceBassManxPower,  yep
17:25.48bkw_faxdetect=incoming
17:25.51ManxPowerMicetto, yes.
17:25.55bkw_yes that should do it
17:25.58Idledoes anyone have a recomended place to buy digium hardware? I need 2 more fxo and 1 fxs module.
17:26.08Zeeekfrom digium
17:26.08ManxPowerSpaceBass, BEFORE the channel = lines, right?
17:26.09SpaceBassIdle,  did you check ebay
17:26.16SpaceBassleme look
17:26.17Zeeekor voipsupply
17:26.26IdleSpaceBass: not yet, I would like to buy it new
17:26.27Micettoso....it's possible that the cause of my problem is that I send a fax from an apple iBook ?
17:26.32IdleZeeek: ok
17:26.46ManxPowerMicetto, I don't know.  Does the apple iBook send the fax tones when calling?
17:26.47Zeeekdigium has a stroe online
17:26.52*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
17:27.05Zeeekwarning, many fax machines do NOT work with spandsp at all
17:27.09Idleoh haha,
17:27.10PakiPenguinevening
17:27.14SpaceBassManxPower,  yeah....its before the channel => <line>
17:27.17Zeeekhowever, fax detection works even when the faxes don't
17:27.22MicettoManxPower: I don't know....^_^'
17:27.24IdleZeeek: I was lookingat the find a reseller, didn't realize they had a link at the right
17:27.25ManxPowerZeeek, We have found that recent spandsps work is almost all fax machines.
17:27.27IdleZeeek: thanks man
17:27.36fourcheezecan asterisk hand off SIP calls to other hosts?
17:27.38ZeeekManxPower not here
17:27.48Zeeekalthough recent = ?
17:27.56ManxPowerMicetto, if the caller doesn't send fax tones then there is no way for Asterisk to know if it's a fax.
17:28.00ZeeekI'm using asterisk 1.2 and whatever the spandsp was then
17:28.05SpaceBasswhat I CAN do is pick up the call and transfer it to my fax extension...that works fine
17:28.11ManxPowerZeeek, "recent = in the past 3 months"
17:28.17SpaceBassof course the other problem I have is that faxes are coming in as blank pages
17:28.18Zeeeknope. Not me
17:28.19*** join/#asterisk flynux (i=v8hy3c1@cl-8.bru-01.be.sixxs.net)
17:28.21SpaceBassbut that seems unrelated
17:28.22fourcheezeif a call comes into asterisk box A for box B, how do I hand that off so that box A plays no further part - would that be a 302 ?
17:28.25fourcheeze(sip)
17:28.26ManxPowerSpaceBass, do you year fax tones when you pick up the call?
17:28.34SpaceBassManxPower, oh yeah!
17:28.57MicettoManxPower:  "fax tones" = "beep....beep.....beep....", right?
17:29.35ManxPowerfourcheeze, that will happen by default if all legs of the call are using the same codec, if you don't have oddball options to the Dial command, if no leg of the call is NAT'd, and you don't have canreinvite=no.  Remember SIGNALING will still go thru box A, just AUDIO will go direct.
17:30.29SpaceBassManxPower,  I hear the tones and I've let it beep a few times...Asterisk doesnt even make an attempt (at least according to the CLI)
17:31.06fourcheezeManxPower: but how do I actually do that - is it just a Dial() ?
17:31.09ManxPowerexten => s,6,Answer
17:31.09ManxPowerexten => s,7,Ringing
17:31.09ManxPowerexten => s,8,WaitExten(3)
17:31.14ZeeekSpaceBass you do have app_rx_fax installed?
17:31.17ManxPowerthis is a copy from my extensions.conf
17:31.19justinuanyone running opera and gaim at the same time on windoze?
17:31.26ZeeekFLOOD! FLOOD!
17:31.27asterboyholy crap, fixed that pesky problem with rx transmission failure on ZAP FXO...dam rxgain=1.0 was killing it.
17:31.33SpaceBassactually...my dring solution worked before I upgraded my A@H.... I need to study up on zapata syntax.... see if I can figure it out again
17:31.43MicettoAH!!!!!!
17:31.50asterboydam that sounds geeky when I read what I just wrote.
17:32.00Micettobut exists a simulator (or configuration checker) for Asterisk ?
17:32.07asterboywonder if I could use that as a pickup line?
17:32.09justinuasterboy: welcome to the club
17:32.16[TK]D-FenderMicetto : reality.sh <-
17:32.46asterboyHey baby, maybe I can stick my FXS into your FXO and make a channel.
17:32.46mutilatoranyone here ever worked in tech support?
17:32.57*** part/#asterisk SWAT (n=SWAT@dsl159-68-100.fastxdsl.nl)
17:32.58justinuin a different life, yes
17:32.58Zeeekyes all day
17:33.01*** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
17:33.12mutilatorwhat do you guys do with support tickets that ya don't closed before your workday is over
17:33.19Micetto[TK]D-Fender: ahahaha
17:33.28mutilatorstay til it's done or pass it off?
17:33.39Zeeekask my provider: I'm still waiting for an answer five days later!
17:33.41asterboyor just drop it to the floor.
17:34.05asterboyno...pass it to some guy in India working for 2 cents an hour.
17:34.53*** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com)
17:34.57asterboy8 trillion and climbing!
17:35.02jsharpPick it up tomorrow morning.  It'll still be there.
17:35.12Hmmhesays2 cents an hour, thats a deal
17:35.13Darwin35good morning all my fellow asterisk whores
17:35.16asterboylol
17:35.38ManxPowermutilator, Huh?  We have work tickets that are almost a year old.
17:36.11ManxPower"user wants us to rewrite the Asterisk voicemail system so it's dumbed down enough for the user to understand."
17:36.27mutilatorManxPower: thats sad..
17:36.29Hmmhesaysdumded down?
17:36.39Darwin35whats a flesh light
17:36.42asterboypress 1 for dumb down mode
17:36.50ManxPowerThe users don't like all those menu options when they are checking their voicemail, sending email via Blackberry, and driving down the freeway at 70mpg all at the same time.
17:37.04asterboyI like the voicemail that says, "Dial"
17:37.11[TK]D-Fender70mpg?  WOW, what kind of cars are they driving?!
17:37.12Zeeek"if you want to hear it, do nothing. Else hangup"
17:37.27Hmmhesaysg must be a new unit of measurement for length
17:37.33asterboyanyone actually have a phone you can Dial?
17:37.37mutilatorya
17:37.41tzafrirgallon?
17:37.41ManxPowerHmmhesays, yes, it stands for "irc inches"
17:37.49mutilatorwind used to blow through the dorm hallways at 30+mpg
17:37.51[TK]D-Fender<- putting the "smart" back into smart-ass :)
17:38.09[TK]D-Fendermutilator : Wow, thats a lot of BEER I'm sure ;)
17:38.13Hmmhesays[TK]D-Fender took me two hours to get that ugly kid joe solo down
17:38.20tzafrirBut then again, it could be a plain "gee"
17:38.27Hmmhesays1 hour to fake it half assed
17:38.58Darwin35ok I still need a good realtime persont o help me move my home plan to real time
17:38.59[TK]D-FenderHmmhesays : not bad...
17:39.13Micettoso...another fantastic question for you....!!!! ^.^
17:39.41konfuzedslePP, are you round today
17:39.42Hmmhesaysi definatley need to sell the les paul though
17:39.47MicettoberoNet driver can be used with junghanns cards ?
17:39.50Zeeekyes and no
17:39.52sevardweird, when i dial my own extension and press * and enter my voicemail password i get in, but when I dial my *98 extension and enter my extension and password it says login incorrect.
17:40.48[TK]D-Fendersevard : mayeb you should look and see what the difference is in how VoiceMail is being called....
17:41.16*** join/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr)
17:42.01*** part/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr)
17:42.07*** join/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr)
17:42.43*** part/#asterisk aze (n=lucky@ACayenne-101-1-6-92.w81-248.abo.wanadoo.fr)
17:42.59justinuService Impact Statement
17:42.59justinuDropped calls and possible busies and All Circuits Busy recordings due to facilities being down. Fiber cut between Pittsfield and Toledo, Ohio, Trucks are enroute and no ETR.
17:42.59justinuAffected Locations
17:43.00justinuAffected Rate Centers: BIRCH RUN FRANKENMTH AUBURN FAIRGROVE BAD AXE BAY CITY ROSEBUSH COLEMAN MIDLAND AKRON SAGINAW FREELAND ST HELEN AUBURN LINWOOD WESTBRANCH OSCODA REESE ST CHARLES VASSAR MIDLAND
17:43.20mutilatorwha no?
17:43.49puzzledjustinu: no redundancy?
17:43.58mutilatorwhere ya see that?
17:44.02Zeeekthe trucks are to carry the packets?
17:44.18sevard[TK]D-Fender: I just saw an error message, vm_execmain: Couldn't read username
17:44.36justinuzeek: i think the trucks are enroute to repair the fiber cut :)
17:44.42mutilatori suggest avian carriers
17:44.43justinuthat alert came from Level3
17:44.45Zeeekoh, that won't do at all
17:44.49mutilatorleast that has an rfc
17:45.04ZeeekvoIP with flu
17:45.13puzzledlol
17:45.21[TK]D-Fendersevard : I said how it is being CALLED.  Look at the exec line.....
17:45.21mutilatorand i think there is an rfc for and QoS over avian carrier aswel
17:45.30justinuSummary
17:45.31justinuUPDATE - Regarding Michigan Facilities Down - The affected OFF NET Transport Fiber technicians are on site excavating, the splicer truck has arrived, no ETR.
17:45.51Zeeekwhat will they be having for lunch?
17:46.15mutilatorwonder if i got one from misdig
17:46.45mutilator<PROTECTED>
17:46.47mutilatorso i'm good
17:46.50ZeeekTopic "If you are using AAH or AMP, please say so right away"
17:46.57justinuheh
17:48.07Zeeeksevard it means vmail is waiting for you to identify your number
17:48.09xachenAAH and AMP just suck period
17:48.17SpaceBassi'd disagree
17:48.28Zeeekyour mouth would have to be reallllly small to suck a period
17:48.29justinuban
17:48.40konfuzedgeez some people. I just found out this  guy had his switch unplugged since sturday and was just bitch at me under his breath like its my fault
17:48.43xachentbqh I don't think users should be using a broken down control panel to run software that is way advanced
17:49.10Zeeekthey can do what they want but it's hard to help them sometimes
17:49.10*** join/#asterisk powerchip (i=powerchi@197.80-202-229.nextgentel.com)
17:49.11justinukonfuzed: people are real assholes
17:49.27xachenDo you give a 12 year old who has never shot a gun before a 30/30 for their first shot?
17:49.34konfuzedits nearly enough to make me change carreers
17:49.38*** join/#asterisk heison (n=heison@216.235.9.2)
17:49.53xachenThen you get #asterisk filled with people asking what a bloody extension is
17:49.55mutilatorxachen: i first shot a muzzle loader..
17:49.59[TK]D-FenderHmmhesays : Funny think, I just took up iaido last week :)
17:50.01SpaceBassif you look at something like the 2.x versions of A@H...sure its a dummyed-down version and has its faults...but it allows a lot of people to get into asterisk and VoIP easily and learn...and that will lead to more hardware, more providers, more proliferation so the govt and ISPs can't screw us
17:50.02mutilatorknocked my on my fscking ass
17:50.04mutilatorheh
17:50.08xachenand how to implement one :)
17:50.09konfuzedim gonna go work in a chocolate factory
17:50.15Zeeekxachen be fair - it's usually more like "I read the section about contexts"
17:50.30Zeeekwhen they really didn't
17:51.02tzangerwow Vonage is spamming now
17:51.05SpaceBassxachen,  I think there should be a asterisk for newbies or asterisk@home channel....I even tried to start one....i agree there should be a diffrent place for experts vs hobbiests
17:51.10xachenyeah, you get lazy ppl who can't be bothered to learn a drag and drool control panel either :p
17:51.18heisonhello [TK]D-Fender... problems with a couple new 7940 phones, can't get the stupid thing to talk SIP
17:51.36mutilatoranyone try out them new grandstream gxv-3000's?
17:51.39mutilatorvideo phones
17:51.43[TK]D-FenderSpaceBass : A@H isn't even "hobbyist" its "I don't want to learn any other details and just want to practice my clicking finger"
17:52.01[TK]D-Fenderheison : Sorry, never worked on Cisco's personally....
17:52.08konfuzedya know the cacao bean has therapuetic levels of magnesium and MAO uptake inhibitors
17:52.09xachenyeah and when you get a desync issue
17:52.11xachenyour done :)
17:52.17heison[TK]D-Fender: okay, thx
17:52.32Hmmhesays[TK]D-Fender: I use a@h for small biz pbx installs
17:52.44konfuzedthen I could maintain a smile for the assholes complaining theres not enough sugar in their chocolate
17:52.44xachenI consider * hobbyist if you compile it from scratch, write your own extensions macros agi scripts tec.
17:52.50SpaceBass[TK]D-Fender,  I'd totally disagree with that...I'm  hobbiest and A@H was my launching pad
17:53.04Zeeeklook where it got you :)
17:53.14salviadudwe're mostly hobbyists then
17:53.16SpaceBassI'v learn a fair ammount of dial plan syntax now, but I never would have been able to if I didn't have something that worked out of the box to start with
17:53.20salviadudi write my own extensions...
17:53.26heisondoes anyone here have experience with 7960/7940? I'm trying to convert from MGCP to SIP (following the Cisco docs), and the phone doesn't want to talk SIP, it thinks it's still MGCP and tries to contact the CM for TFTP
17:53.30SpaceBassZeeek,  thanks for the vote of confidence
17:53.31xachenyeah, i started from the console :)
17:53.35xachenstarted with cvs head
17:53.40Micettobye bye
17:53.44MicettoI'm going away
17:53.44Micetto:)=
17:53.48Micetto=:)
17:53.50*** join/#asterisk Nugget (i=nugget@dazed.slacker.com)
17:53.54Darwin35grrr
17:53.56*** join/#asterisk stoffell (n=stoffell@d51A4D12C.access.telenet.be)
17:53.56ZeeekNot only did I learn asterisk but I never had installed linux before
17:54.00SpaceBassby this logic X was the worst thing to happen to unix/linux
17:54.01salviadudmy approach was to SEE a config file, then edit it.  now i make my own
17:54.01Hmmhesaysi started my linux and asterisk ventures at the same time
17:54.15ZeeekNot that I remember 90% of what that took, but I got a lot of help here
17:54.26salviadudwhat's wrong with X?
17:54.27exonicI'm writing a manager API ncurses interface. I am trying to find a way of getting 'detailed info' on channels.
17:54.41SpaceBasssalviadud,  its for lazy people who cannot be bothered to use the shell
17:54.46salviaduddon't you like neat graphics? i can't watch pr0n without x dude!
17:54.49ZeeekI never was able to finish the asterisk make on solaris 10 the other day
17:54.50[TK]D-FenderSpaceBass : Do you actually use A@H with the knowledge you've gained since?  For all the control you give up to it there isn't much "in-between except for special IVR's at best....
17:54.56Hmmhesaysit would be nice to have a web interface for lcdial
17:54.56justinuexonic: i got a little bit of help from AMI yesterday
17:54.59stoffelloej, can i ask you somethin' in private?
17:55.02salviadudi'm a slackware user, the console is my best friend of course
17:55.04ZeeekTTY pr0n
17:55.06SpaceBassand since there is xchat the linux channels get flooded with nubes asking how to run itunes on linux
17:55.11Zeeekmuch more left to the imagination
17:55.11salviadudbut when it gets down and dirty
17:55.11justinuexonic: turns out, if you set async=true in an originate action, it'll return you the channel name, and unique ID
17:55.17Darwin35my xkey keyboard stoped working
17:55.21Darwin35grr
17:55.22exonicjustinu, hey, how's it going
17:55.26[TK]D-Fendersalviadud : Hallelujah!
17:55.29Darwin35I now have to reprogram in
17:55.35exonicjustinu, hmm. good to know
17:55.35SpaceBass[TK]D-Fender, I still use it b/c my box is working fine and all I ever do is add/remove the occasional trunk and thats just easy with freepbx
17:55.46Hmmhesaysi need my Itunes damnit
17:55.52SpaceBass[TK]D-Fender,  my next box in the future will probably be from scratch, but I'll still run freepbx
17:55.57salviadudi don't like it when people diss on X
17:55.59justinuexonic: here's my latest dilemma... i send out the originates, but I don't get the "OriginateSuccessEvent" until the call is actually answered.
17:56.01exonicjustinu, I am I'd like to get detailed channel variables if possible.
17:56.03salviadudi use fluxbox and make it look cool
17:56.04Hmmhesayspffftttt
17:56.05oejstoffell: go ahead
17:56.08justinuso I can't terminate the call until it's answered :(
17:56.09xachenfluxbox is the best :)
17:56.13Zeeekthe new PalyStation comes with asterisk pre-installed
17:56.20salviadudgot my own menu, lots of shortcuts
17:56.21jsharpxachen:  That just made me snort my Pepsi.
17:56.22SpaceBassI'm just saying things like A@H and amp/freepbx have a place and they will ultimatly benifit everyone
17:56.28Hmmhesaysxachen: you see that family guy episode where the crippled prof was talking dirty to his crippled wife?
17:56.32exonicjustinu, hmm.. You can of course track the channel state. 'Up' , 'Ringing' etc
17:56.33xachenyup :)
17:56.34[TK]D-FenderSpaceBass : then again what value does that have to truely understanding * when all you're doing is letting it do the whole job for you?  Not much of a hobby when you're sitting in the stands while AMP is in the driver's seat.
17:56.34HmmhesaysI agree with SpaceBass
17:56.36SpaceBassit will get more people using asterisk and that equals more $
17:56.40salviadudand well, if im running asterisk i don't need it. that's all
17:56.47asterboyif you get it to work.
17:56.53justinuexonic: yeah - except that I have no way of knowing which channel I created... except by the context, which irritates me
17:56.54konfuzedHmmhesays, itunes is just a branded mp3 player but ya got suckered into paying for each tune
17:56.58SpaceBasshow many windows users edit the registery to change their desktop background?
17:57.04Hmmhesayskonfuzed: it was a joke
17:57.08SpaceBasshow man mac users spend all their time in pref files exiting XML?
17:57.08justinueven then, I can't match each originate action with the "new channel event"
17:57.14exonicjustinu, but I thought async returned the unique id?
17:57.21justinuit does... when the channel is answered
17:57.28exonicjustinu, ugh!
17:57.29*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
17:57.39justinui think i should just modify that to give me the originate success when the call is proceeding
17:57.42konfuzedHmmhesays, those itunes people have been triggering me lately ;^)
17:57.45exonicjustinu, have you seen the Dial event, I'm not sure so what to do what that. Perhaps it can help you?
17:57.50SpaceBassI'm building a * box as a project at work....build from source, but I'm still going to put freepbx on it for day to day stuff
17:57.54[TK]D-FenderSpaceBass : I didn't say FreePBX was "bad" per-se, just completely bad if you are intending on using it as a launching pad to actually learning *.  Maybe as a motivator to say "see we can make it work, so can you, only you'll have to start from scratch because understanding our crap will give you a TUMOR"
17:58.01Hmmhesaysi still download my music illegally like everyone else
17:58.09xachenspeech recongintion could be used to detect panting and change the tone of the Festival config parameters :p
17:58.14Hmmhesaysfreepbx pretty much kicks ass
17:58.18Hmmhesaysfor guys like me
17:58.27justinuexonic: i only get a dial event when my originate call hit's a "Dial" command in the dialplan
17:58.30[TK]D-FenderHmmhesays : You on the new beta of it?
17:58.37Hmmhesaysyeah, i love it
17:58.40exonicjustinu, yeah, it doesn't hit Dial all the time?
17:58.46Hmmhesayson the flip side, I know whas going on under the hood too
17:58.52justinuno, because I originate a call, and send it to an IVR extens
17:58.57exonicjustinu, ahh
17:59.00SpaceBass[TK]D-Fender,  look, I agree that trying to learn traditional (and clean) dial plan syntax from reading amp's conf files is like learning french by reading strero instructions in french.... but it does enable people to get setup out of the box
17:59.04[TK]D-FenderHmmhesays : what seperates it from the previous version? (Didn't notice much in the screen-shots)
17:59.18Hmmhesaysoh its way prettier than amp 1.x
17:59.20Hmmhesaysand modular
18:00.01[TK]D-FenderSpaceBass : completely true.  and if that gets them running enough to feel confident in buying the HARDWARE to get started sure, but getting away from AMP is no easy matter, and learning FROM it, less so.
18:00.04asterboyMy Dial plan won't let me simply pickup the phone and start dialing.  I have to put something like, "_9." to dial 9 before I get dial tone.  How do you change that???
18:00.09sevard[TK]D-Fender: I figured out the problem  My context was &#^$ed
18:00.15Hmmhesaysamp will confuse the fark out of a nub
18:00.17sevard[TK]D-Fender: Do I have to create /var/spool/asterisk/voicemail/default/140/INBOX manually?
18:00.24exonicjustinu, yeah, Quite the dilema. I don't think very much thought went into the manager api.
18:00.26[TK]D-FenderHmmhesays : Is there a demo page I could cruise for it?  Didn't see it a Coalescent's site previously.
18:00.39[TK]D-Fendersevard : I was pretty sure thats what it was
18:00.42justinuexonic: heh, nope... i think i'm just gonna have to patch the code to make it work for me.
18:00.46Hmmhesaysyeah you can cruise an install I messed up, it doesn't actually work, but everything is there
18:00.50SpaceBass[TK]D-Fender,  I'd agree with that....when I customize my dialplan its totaly outside of AMP's structure...i just use their 'custom' commands to drop into my own plan
18:00.53[TK]D-Fendersevard : the box should initialize itself...
18:01.15asterboyMy Dial plan won't let me simply pickup the phone and start dialing.  I have to put something like, "_9." to dial 9 before I get dial tone.  How do you change that???
18:01.18Hmmhesayshold on i'll link you
18:01.23SpaceBassbut the bottom line, amp or no amp....I fucking hate faxing right now and I'm going to go Office Space on the next fax machine I see
18:01.25[TK]D-FenderSpaceBass : how much "custom" work have you done with it so far?  Give me a sample.
18:01.31konfuzedAMP is the biggest motivator for all those asterisk manager project a likes to have gotten started
18:01.51[TK]D-FenderSpaceBass : Oh... yeah analog fax + * = pain... just gt POTS for that if you want to remain sane...
18:01.54salviadudoffice space, as in the movie?
18:01.59salviadudthat movie rocks
18:02.08SpaceBass[TK]D-Fender,  nothing extravigant.... i have a 2nd incoming context that routes work calls differently than personal...i have gotos that i use for followme dialing when I'm traveling abroad, stuff like that
18:02.12SpaceBasssalviadud,  yeah it does!
18:02.15[TK]D-FenderI *do* however intend on learning A@H + all its components as it will open up a new customer base for me.
18:02.28exonicI am having incredible trouble w/ faxes too. patched asterisk 1.2.4 w/ T.38 and can receive just fine but sending fails 1/8
18:02.31asterboyAny examples showing how to put get an outside line without pressing a key first?
18:02.39SpaceBass[TK]D-Fender,  I think there is a market for A@H installers
18:02.44konfuzedanalog fax stuck on a pstn with plain fax machine
18:02.50salviadudpattern matching
18:02.55Zeeekasterboy dial ZAP/1
18:03.00asterboyexonic, put the faxes onto HylaFax instead.
18:03.06SpaceBassasterboy, just don't require it in the dial plan :P
18:03.09Hmmhesayshaha I toasted my apache2.conf on this box too it'll be a second
18:03.12Zeeekor whatever inferface the line is on
18:03.12asterboythx Zeeek , trying...
18:03.23[TK]D-FenderSpaceBass : Agreed, and for business reasons I should not ignore it.  I hate it when business ethics clash with my personal ethics :)
18:03.44SpaceBass[TK]D-Fender, exactly...for instance, I'm an MCSE
18:03.48[TK]D-FenderI'm a very "practice what I preach" kind of guy...
18:03.49SpaceBassI'd shoot myself if I could
18:03.52exonicasterboy, Hylafax? it's unknown to me. Does it allow SIP to sent faxes ?
18:04.00[TK]D-FenderSpaceBass : Try target practice first ;)
18:04.06salviadudMCSE?
18:04.06*** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net)
18:04.09salviadudwhat's that?
18:04.15SpaceBasssalviadud, microsoft slut
18:04.24salviadudhahaha!
18:04.27[TK]D-Fendersalviadud : Microsoft Certified System Engineer.
18:04.31salviadudyou poor bastard
18:04.47SpaceBassI don't even work in that copacity anymore...but it comes in handy
18:04.55Zeeekwithout microsoft there would be no linux
18:05.04konfuzedok 4 active lines in the house (not mine but somebody wondering why their phone bill is so high) so keep the analog fax line active and put dsl on it and an x100p for fail over to pstn
18:05.05[TK]D-Fenderor : Multiple-Choice Sysadmin Exam ;)
18:05.10konfuzedyeah that should do it
18:05.23SpaceBassLOL
18:05.36[TK]D-Fender"C" baby!  All the way to 80% !!!
18:05.38*** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net)
18:05.48SpaceBassThey don't tell you % anymore...just pass/fail
18:05.57konfuzedonly faxes call that number so it wont have to be forwarded or answered by asterisk at all then
18:06.00salviadudwindows is terrible with filenames, file permissions
18:06.09salviadudand well... spyware
18:06.09SpaceBasswhich sucks b/c I was proud of my % for a week or so that I cuared
18:06.10[TK]D-FenderGod I'm so good its ~SCARY~
18:06.28asterboypwd
18:06.28asterboyls
18:06.37Zeeekcd
18:06.39asterboyslol
18:06.42asterboyifocnfig
18:06.52Zeeekipconfig /ALL
18:06.58SpaceBassI was pretty anti windows and MS when I switched to OS X recently....but it has its pluses
18:07.12salviadudso, SpaceBass, who do you need to *** to get some source code in microsoft?
18:07.16asterboyipconfig! how dare you!
18:07.27salviadudyeah, wmv videos man
18:07.33salviadudi use windows for pr0n
18:07.36ZeeekI have to see my ip address
18:07.37salviadudand that's it
18:07.44SpaceBasssalviadud,  I recently had to sign a NDA with MS...it involved a LOT of lube
18:08.01justinubut you liked it
18:08.08SpaceBassosx for porn....all the picts, none of the spyware
18:08.24konfuzedthen keep the main house phone number line active with dsl aswell with x100p and asterisk actually answer that line
18:08.38ManxPowerI thought MS didn't believe in lube.
18:08.57SpaceBassManxPower, we had our lawyer negoicate it....my company respects me like that
18:09.06salviadudwho has a xbox 360?
18:09.16SpaceBassseriously...this was a sit down with MS lawyers and ours just over a NDA that no one cares about
18:09.28konfuzedand then move the other two lines to a VOIP provider as 2 DIDs and then drop the kids pstn line and the business pstn line
18:09.38Zeeekthe .NET version of asterisk 2.0 ?
18:10.08salviadudyou think asterisk might get succesfully ported to windows?
18:10.09Zeeekkonfuzed if you have only DSl you don't wanna drop the business line
18:10.16konfuzedi wanted to drop the fax line cause its not really used but
18:10.27Zeeekonly if windows is successfully ported to windows first
18:10.31SpaceBass* will run on windows with cygwin
18:10.42salviadudcygwin is huge
18:10.50SpaceBassi dont care for it myself
18:11.02salviadudi rather use linux than cygwin
18:11.07Zeeekbut it's still in love with you
18:11.11salviadudi can't select text on cygwin
18:11.13xachencygwin is real slow
18:11.17salviadudif asterisk would run on putty
18:11.19salviadudthat be awesome
18:11.37mutilatorhuh?
18:11.40konfuzedwell i figure the the voip provider DID can be set to forward to house pstn number if asterisk doesnt pick up
18:11.41salviadudalthough, that is practically impossible
18:11.44SpaceBassi have one xp box left here at home...use it for outlook and thats it
18:12.00Zeeekhardware is so cheap, just put asterisk on a box and talk to that box from whatever system you like, even windows
18:12.15salviadudyeah, windows is the OS for grandmas, teenage girls and lawyers
18:12.20SpaceBassthats what I did....
18:12.34Darwin35ok now i am pissed  why does realtime vm not work with pgsql
18:12.39SpaceBassgot a 2ghz dell optiplex off ebay for like $75...runs * great!
18:12.58*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
18:13.00salviadudhey
18:13.11salviadudi got a prank call i did to a rehab clinic
18:13.17salviadudmixmonitor rulez!
18:13.22salviadudwho wants it?
18:13.26konfuzedwell crap theres 4 pstn lines in the house one just for fax and one double charged as a business line. i figure leave 2 pstn and move 2 over to VPBX provider and make it a hybrid setup
18:13.26justinubueno?
18:13.51salviadudyes amigo friend platano banana
18:14.17salviadudwell, any takers?
18:14.31salviadudit's about 1.7 megs big
18:14.32justinupost the link
18:14.42salviaduderrr, no link
18:14.49salviadudi got the file
18:14.55justinuuh, upload it to a webserver
18:15.13konfuzedcp file ~/pbulic_html/.
18:15.14salviadudlike, which one?
18:15.20Zeeekhttp://yousendit.com
18:15.33Zeeeksend yourself the link and post it
18:16.00konfuzedeven pastebin lets you attach files
18:16.29salviadudcan i attach a wav in pastebin?
18:16.31*** join/#asterisk NexGen (n=me@adsl-70-135-6-65.dsl.tulsok.sbcglobal.net)
18:16.57konfuzedsalviadud, cp path/to/files /var/www/htdocs/html/prank/.
18:17.04konfuzed;^)
18:17.22konfuzedattach file is generic
18:17.43NexGenhey guys, have a small problem, when connected to my cli I get the following:
18:17.43NexGenParsing '/etc/asterisk/manager.conf': Found
18:17.43konfuzedit rarely bothers to chack file types
18:17.43NexGenConnect attempt from '127.0.0.1' unable to authenticate
18:17.48NexGeni have verbosity set to 10
18:18.04SpaceBassNexGen,  got amp/freepbx ?
18:18.08NexGenbut I have no way what is trying to connect on the server, any ideas?
18:18.08pauldythat just for the remote agi isn't it
18:18.18NexGenamp
18:18.22SpaceBassNexGen,  thats what it is
18:18.28pauldythe pl file
18:18.30NexGenamp trying to connect?
18:18.47SpaceBassits amp connecting to the manager service
18:18.48Darwin35anyone here doing realtime with pgsql ?
18:19.09NexGenok so apparently the password is wrong in the manager.conf?
18:19.10SpaceBassjust mysql here
18:19.31SpaceBassNexGen,  yeah, most likely or in amp.con ... check out #freepbx
18:19.50SpaceBasssee [TK]D-Fender  I'm not a total waste! :)
18:20.27*** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)
18:20.28pauldyop_server.pl
18:20.45bancusIf my system clock is running slightly fast, would that cause the audio to skip a lot?
18:21.03SpaceBasssystem time? or overclocked?
18:21.06bancussystem time
18:21.12pauldyamportal.conf
18:21.13Zeeeknah
18:21.14SpaceBasscan't see why it would
18:21.14bancusseems to be a problem with my motherboard
18:21.31bancusIf left unchecked, it runs about 30-60 seconds fast every 10 minutes.
18:21.39[TK]D-Fender.clear
18:21.49pauldyneat
18:21.56bancusI have it resyncing every minute until I can fix it, but the workaround wouldn't help within each minute.
18:21.58Zeeekwhy not sync it with a time server?
18:22.05Zeeeko
18:22.08tsumeeww amp
18:22.11tsumesuck0rx
18:22.29pauldytsume: why?
18:22.33bancusBut I'm having a weird issue where half of a word or more gets skipped every second or so.
18:22.42bancusIt'll just skip ahead.
18:22.44Zeeekon the local network?
18:22.47bancusyeah
18:22.56bancusHappens when I check my voicemail.
18:23.14bancusI thought it might be a lag issue I'm having with my upstream SIP provider.
18:23.16salviadudalright!!!
18:23.19salviadudhttp://s49.yousendit.com/d.aspx?id=0UGEMIUD5PLD42370SSWF10WLX
18:23.19*** join/#asterisk iGotNoTime (n=iGotNoTi@cpe-65-189-240-199.woh.res.rr.com)
18:23.20Zeeekany IRQ issues?
18:23.20bancusBut then it started happening locally too.
18:23.29bancusI don't believe so, but how do I check?
18:23.47salviadudbasically, im the guy that says hello
18:23.49tsumepauldy: its not a very good interface, also crashes in linux much
18:23.50ZeeekI forgot how!
18:23.53bancusheh
18:24.08Zeeekls /int somebody jump in here
18:24.15Zeeekproc/int
18:24.19Zeeeksomething
18:24.27iGotNoTimesalviadud, hello :)
18:24.31pauldytsume: it crashes?
18:24.43*** join/#asterisk Lino` (n=Lino@i577BC430.versanet.de)
18:24.45bancus<PROTECTED>
18:24.51bancusexcept with two rs
18:24.58Zeeeksounds about right
18:25.09bancusWhat am I looking for?
18:25.10tsumepauldy: yeah, well.. its flash. what to expect
18:25.12Zeeeklook but don't paste
18:25.18*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
18:25.19bancusk
18:25.20iGotNoTimequestion.... I have installed * and it is running now.... isn't there a web GUI for it though? I have tried localhost but it says denied. Am I missing something?
18:25.30Zeeekmuhahaha
18:25.36tsumeall these GUI babies make me sick
18:25.36bancusiGotNoTime: there is no web gui in the default program
18:25.37Zeeek@hole
18:25.40tsumelearn to use vi and linux
18:25.45Zeeekvi sucks
18:25.45pauldyamp is flash?
18:25.47bancussome 3rd party addons might
18:25.53Zeeekno FOP is flash
18:25.58salviadudnot vi necesarily
18:25.59tsumepauldy: oh wait, I'm thinking og flash operator
18:26.02salviadudyou can use pico
18:26.02bancusZeeek: what am I looking for?
18:26.09Zeeeknano is even better
18:26.14pauldyahhh ytea the flash operator pannel is flash
18:26.16Zeeekbancus IRQ shared
18:26.35Zeeekwhat is the Trash Operator Panel then?
18:26.37iGotNoTimetsume, I wish I had your knowledge, but unfortunately it is difficult to lean command line because those who do learn it well you know. :)
18:26.39[TK]D-FenderiGotNoTime : GUI's are a whole other world......
18:26.39bancusHow can I tell when something is shared?
18:26.50tsumeiGotNoTime: it takes 5 minutes to set up ;)
18:26.57iGotNoTimethank you [TK]Fender :)
18:27.02tsumemaybe I should make a GUI based drag n drop and sell it
18:27.02salviadudhey, you guys hear my prank yet?
18:27.08iGotNoTimetsume, yes it is running I said that :)
18:27.08tsumefor asterisk
18:27.18tsumeiGotNoTime: I mean fully setup ;)
18:27.52pauldyI know how to setup via command line and I preffer to use amp just because it makes some of the more complicated stuff rpetty damb simple
18:27.52salviadudits a prank done with asterisk... might i add
18:28.05Zeeekbancus : cat /proc/interrupts
18:28.11bancusZeeek: done
18:28.15iGotNoTimeI agree pauldy, why waste hours trying to learn all the commands when you can have them placed as a button :)
18:28.25tsumeiGotNoTime: hours?
18:28.28Zeeeklook at each line and see if there is more than one device
18:28.30tsumeiGotNoTime: only takes a few minutes
18:28.39tsumereading helps
18:28.43bancusYeah, trying to listen to my voicemail is almost like listening to a tape and repeatedly hitting FF.
18:28.43iGotNoTimetsume type help on your command line :)
18:29.02iGotNoTimethere are alot of commands to remember
18:29.02pauldytsume: no to learn all the commands you need or might need it can take a lot longer than a few minutes
18:29.07tsumeiGotNoTime: I don't need help, I already know how to do everything in asterisk ;)
18:29.13tsumepauldy: not really
18:29.16Zeeekbancus - is there more than one device on lines?
18:29.18bancus21 has ehcd_hcd and eth0, 22 has libata and ohcd_hcd
18:29.21pauldyyea really
18:29.26iGotNoTimetsume then refer to my above observation ;)
18:29.37tsumepauldy: when I first started asterisk, I looked at an extensions.conf and started programming
18:29.49iGotNoTimetsume I have done that
18:29.52pauldyand you learned everything in the first 5 minutes
18:29.52iGotNoTimeit is setup
18:29.59tsumepauldy: yes
18:30.09pauldyhahaha
18:30.10ZeeekI took a slightly different approach
18:30.21salviadudasterisk is great... i took the same aproach as tsume
18:30.24ZeeekI read the old asteriskdocs.org manual about 10 times
18:30.41Zeeekwhat I didn't get, I re-read until I figured it out
18:31.02salviadudthe new book from o reilly kicks mayor ass
18:31.02Zeeekthen I imposed my experiments on two innocent women for one year
18:31.19salviadudit's based on 1.2
18:31.36salviadudkiller book, good jokes, excellent editing
18:31.46Zeeekgive us a joke from it
18:31.46iGotNoTimeI like that Zeeek, I never said I did not read, and I am sure Pauldy has read as well. That does not make us ignorant because I ask about a GUI though does it?
18:31.53salviaduddownload my file! download it in the name of asterisk!
18:31.56salviadudhttp://d49.yousendit.com/F/0UGEMIUD5PLD42370SSWF10WLX/narconon.wav
18:32.00ZeeekI never said that
18:32.10SpaceBassI think Bill O'Riley should do an Asterisk book....Asterisk, No Spin Zone
18:32.10*** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe)
18:32.15Hmmhesayslol
18:32.16Zeeekheh
18:32.20iGotNoTimeSpaceBass, yes he should!
18:32.20SpaceBassim sure it'd be fair and balenced
18:32.35Zeeekfox@home
18:32.52ZeeekBush Control Panel
18:32.56iGotNoTimeLMAO
18:33.03salviadudthey joke about asterisk couldn't get so popular at the start because it wasn't a way to get pr0n much faster
18:33.16Zeeekoh, but it is!
18:33.23Zeeekvoice pr0n
18:33.25iGotNoTimehaha
18:33.36SpaceBassChapter 3 - Legal Wire Taps using FOP
18:33.42Zeeeksomeone has a recording of Allison drunk
18:33.48tsumehmm
18:33.51SpaceBassChapter 4 - if it cashes, blame femao
18:33.53SpaceBassZeeek,  no shit?
18:33.55Zeeeksaying "ok you stupid asshole, I said hit one"
18:34.00salviadudthe voice of digium?
18:34.01SpaceBasss/femao/fema
18:34.07salviaduddo you think she's hot?
18:34.19tsumeand not providing person to person connections but service :D
18:34.21SpaceBasssalviadud,  I have to assume so, its the only thing holding comedian mail together
18:34.29ZeeekI'm more attracted by Vulcan women personally
18:34.41salviadudpointy eers
18:34.50Zeeekyeah
18:34.50asterboyFor some reason, I have to dial a # to get a dial tone from my FXO port, THEN I can dial to another number...how can I make it so that I just have to dial the number end of story?
18:34.51salviadudvulcan women are just the same as zelda
18:35.08Zeeekasterboy what phone?
18:35.16asterboyPolycom IP600/500
18:35.17salviadudmaybe you like the fact that they are very logical
18:35.22SpaceBasssalviadud, http://www.theivrvoice.com/
18:35.25Zeeeklook up digitmaps
18:35.33asterboythnx
18:35.39Zeeekin Polycom
18:35.45salviadud"would you blow me?" - "yes, it is only logical"
18:36.13Zeeek~seen digitmaps in Polycom
18:36.24jboti haven't seen 'digitmaps in polycom', Zeeek
18:37.12salviadudshe's canadian?
18:37.18salviadudshe probably says aboot, instead of about
18:37.26*** join/#asterisk rollergrrl (n=0x3e44d@71-213-20-208.slkc.qwest.net)
18:37.33salviadudnot that i mind
18:38.23*** join/#asterisk brettnem (n=brettnem@nemeroff.com)
18:39.44salviadudi think allison might have some trouble actually talking on the phone, they'd probably think its just an automated service
18:42.09*** join/#asterisk ToTo (n=ToTo@host154-207.pool872.interbusiness.it)
18:42.19tsumehow much would someone pay for a full fledged dial plan creator GUI?
18:42.20*** part/#asterisk oracle^ (n=cam@unaffiliated/cameleons)
18:42.21asterboyhttp://www.freedomphones.net/polycom/files/docs/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf
18:42.28[av]banipolycom has no overlap dialing :<
18:42.28tsumefor win,lin,osx ;)
18:42.30asterboyAdmin guide has some info.
18:42.39tsume49USD?
18:42.53hypa7iatsume: make it flash or ajax, then it's croxxplatform by default :)
18:43.12[av]banicroxx!
18:43.37hypa7iahehe
18:43.39hypa7iayay spelling
18:44.00hypa7iai have one of the weird swoopy microsoft keyboards at work
18:44.07hypa7iatotally throws off the spelling
18:44.19Zeeekasterboy just enter your patterns in the digitmap and the phone will dial as instructed
18:44.35asterboychecking...
18:45.07bancusOkay, I found a stopwatch to verify seconds against my clock
18:45.14bancus"sleep 10" resolves in about 5 seconds
18:45.21bancusso there's definitely something screwy going on here
18:45.42*** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net)
18:45.46Zeeekstuff like [23]xxx to dial 2000-3999 right away
18:45.49bancusyay, bugreport about it
18:45.52Darwin35where are the real time guys
18:46.00jsaundersCan * utilize a multiprocessor environment to any end?
18:46.04jsharpYes
18:46.12jsharpMore processors make it taste better.
18:46.19jsaundersReheally.
18:46.34hypa7iathey improve the texture too don't forget
18:46.44jsharp* is multithreaded, so multiple processors make * happy.
18:47.21jsaundersSo multiple low clocked dual core opterons would be preferable to oh say a single high clocked Athlon.
18:47.23jsaundersBeauty
18:47.41bancusjust under $300 as I recall
18:47.49bancuscheapest dual-core proc I could find
18:48.02justinubeauty, eh?
18:48.15asterboyok, so it looks like Polycom is setup with a default digit map that does not allow standard pickup and dial.
18:48.26jsaundersI'm lookin' at the opteron 240's, two of 'em.
18:48.31Zeeekwhat exactly is the result you want
18:48.33asterboySo I added xxxxxxx|xxxxxxxxxx|1xxxxxxxxxx
18:48.47asterboyI just want to pickup a line a dial, period.
18:48.47ManxPowerasterboy, wow, that was stupid.
18:48.59asterboyhow so?
18:49.02justinuhahah
18:49.06jsaundersheheh
18:49.10ManxPowerasterboy, they are pretty much the same.
18:49.39asterboythe x's
18:49.55Zeeekevery x is the same
18:50.03justinui need my laptop to arrive before I can go on vacation
18:50.26SpaceBassI need a macbook pro
18:50.33Zeeekget a Dingleberry
18:50.49asterboyok, I'm not understanding the idea then.
18:50.52_Paulo_In Brazil collect calls are automated. There are a standard recording like "This is a collect call. To accept the call ...bla..bla..bla". The callee has 5 seconds to let the caller identify himself and accept or drop the call.
18:51.01justinui considered the macbook pro
18:51.02justinutoo big tho
18:51.05asterboywhy does the default in the manual list, 9]xxxxxxxx
18:51.10justinui ordered a thinkpad X60s
18:51.30asterboyI don't want 9 to dial out, just the number part.
18:51.37_Paulo_Is there any application so I can recognise the collect call recording?
18:51.37*** part/#asterisk mko-025 (n=korpim@p5498BD34.dip0.t-ipconnect.de)
18:51.44Zeeekhow many different types of numbers?
18:52.29asterboyie. 123-1234, 123-123-1234 or 1-123-123-1234
18:52.48asterboyactually I just want it to dial no matter what number combo.
18:52.58asterboyJust pickup the god dam line and dial!
18:53.23Zeeektune up the crank
18:53.26asterboyright now, I have to dial something first, then it gives a dial tone, then I can dial whatever I want.
18:53.38Zeeekthat's not the phone, it's asterisk
18:53.51tsume_Paulo_: 5 seconds is much to transmit enough info ;)
18:53.52asterboyThat's what I thought in the beginning.
18:54.06asterboyI've tried every combination in front of the dial plan.
18:54.07tsume_Paulo_: just keep calling back with parts of the sentence :P
18:54.07Zeeekso this is another case of @home?
18:54.18asterboyno @home
18:54.29_Paulo_tsume, It used to be 10 seconds, they shortened to 5 seconds due to abuse...
18:54.37Zeeekwell someone put the logic to wait for a digit in there
18:54.40tsume_Paulo_: still easy to abuse :)
18:54.51asterboy_9.,1,Dial(ZAP/1) - have to dial 9 first obviously.
18:54.59tsume_Paulo_: portuguese are fast talkers :P
18:54.59Zeeekyes?
18:55.08asterboy1,1,Dial(ZAP/1) - gives me a fast beep
18:55.13tsumeyou should hear my motor mouth ;)
18:55.24*** join/#asterisk x86 (n=x86@p3m/member/x86)
18:55.25SpaceBassok.. SIP calls started coming in and looping, so I did a restart gracefully....never came back, so I bounced the box and now when I connect to the CLI i get an error about no realtime
18:55.32SpaceBasswtf...i didn't change a thing!
18:55.39Zeeekasterboy it's impossible for you to have read about dialplans and still be in the place you are now
18:55.56asterboyI know I'm missing something, just not sure what.
18:56.14Zeeekeach technology has a slightly different Dial syntax
18:56.27[TK]D-Fenderasterboy : if you want it to "just dial" then do "x.T" and it'll through EVERYTHING at it regardless of length.
18:56.31_Paulo_tsume, the worse is that the market is distorted...
18:56.32ZeeekZAp is ZAP/chan/Number or something like that
18:56.53SpaceBassARRRRUUUUGGGGGGGG my entire box just shat itself and died for no reason....nothing changed
18:56.54asterboyah
18:57.05Zeeekexten => 18005551212,1, Dial(ZAP/1/${EXTEN})
18:57.22ManxPowerZeeek, don't put a space after the ,
18:57.30Zeeekwrite a fixed extension like that and get it working. Then abstract it to _1800.
18:57.30_Paulo_tsume, in the state owned telco model from the 70s the long distance calls used to subsidize local calls...
18:57.40ZeeekNO SPACES! I'm a bad typist
18:57.42*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
18:57.53ManxPowerZeeek, that's why you should COPY & PASTE
18:58.05ZeeekI was typing but I'm lousy at it :)
18:58.16justinulike copy/paste has never caused any problems :P
18:58.17Zeeekplus my mind isn't a perfect memory
18:58.26_Paulo_tsume, long distance calls are still expensive despite intense privatization in the late 90s.
18:58.29tsume_Paulo_: yikes :/\
18:58.32*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
18:58.39tsume_Paulo_: voip power :)
18:58.41Zeeekthe POINT is, this is all discussed extensively on the wiki and in asteriskdocs.org docs for years
18:58.45ManxPower_Paulo_, you must not be in the USA
18:59.16tsume_Paulo_: voip will take over brazil, then they'll have to look for more ways to make money.. by destroying the rest of the rainforest
18:59.26tsumeobliterating the world, murderers
18:59.57tsumeI wish bush would attack brazil and save the worlds air ;)
18:59.59Darwin35ok I have postgress setup and running and all the tables inthe db made
19:00.11Darwin35now to get asterisk to work with it right
19:00.18Hmmhesaysthis should be interesting
19:00.26Zeeekasterboy you should take the time to read this: it's old but it has the answers to your question
19:00.29Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
19:00.33Zeeek<PROTECTED>
19:00.48ZeeekI said take the time!
19:00.59asterboyThis just sucks: http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
19:01.02tamp4xdebug tells me nothing hmmhesays
19:01.06tamp4xoops
19:01.08asterboycan you make it any more vague
19:01.08_Paulo_tsume, a squirrel could travel from cost to cost in EUA without touching the ground befor colonization.
19:01.08tamp4xanyone here use spandsp? when i load asterisk -vvvvvvvvv  it stops loading when app_rxfax.so loads....any ideas why?
19:01.10ZeeekThis says it all: "The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls"
19:01.16Zeeekduh!
19:01.46*** join/#asterisk jero (n=jero@savoirfairelinux.net)
19:01.55tsume_Paulo_: 35% in the past 20 years is too much
19:02.08tsumeprobably more if I look at the charts
19:02.11Zeeekthen there's this gem: Calling channels with the Dial()  application
19:02.17GerbilNutis it possible to capture everything going to the CLI for a period of time?
19:02.19Zeeekshit, it's all there
19:02.34Zeeekeverything but the final 42
19:02.42Zeeekwhich you then come here to find :)
19:03.06ManxPowerGerbilNut, you mean like as is configured in logger.conf ?
19:03.15ManxPoweror is it logging.conf?  Anyway....
19:03.24GerbilNutprolly so
19:03.54ZeeekManxPower is your madrid presentation online anywhere?
19:04.09ManxPowerZeeek, no.  I don't even know if I have it.
19:04.44ZeeekI think I'll be speaking later this year, I thought I might Stea^H^H^H borrow an idea or three
19:04.48ManxPowerOh!  The astricon people got a copy of it, I think.
19:05.20Zeeekthey must have posted the stuff
19:05.29Zeeekmaybe no longer available tho
19:05.59justinuthe function dial_exec_full is a fucking joke
19:06.05oejI'll make sure the Astricon presentations come back on line
19:06.15*** join/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net)
19:06.27Zeeekthere was some good stuff there
19:06.48ZeeekMatt from digium had some great intro programming stuff too
19:06.52salviadudi wish i could go to astricon...some day
19:06.53ZeeekIIRC
19:06.54Darwin35hey Manx
19:07.29justinuyeah, we were promised an email w/ the presentation info in it
19:07.32justinugot zip
19:07.41Zeeekthe t-shirt!
19:07.46justinui did get spammed about asterisk book camp tho
19:07.51justinuso they obviously have my address
19:07.56justinus/book/boot/
19:08.14ZeeekFWD has been intensly spamming me about von. I had to threaten them
19:08.41ZeeekI told em I was gong to loop through all FWD numbers with a special message
19:08.50justinulol
19:08.51SpaceBassi think my zaptel hardware just died
19:09.06Zeeekgive it a plain pine box, nothing fancy
19:09.33*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
19:09.35Darwin35send it back make them replace it
19:09.44Zeeekthe fastest way to kill it is to plug an fxs into the wall
19:09.48justinutry a different pci slot
19:09.52Darwin35call them everyday ill they do
19:10.13SpaceBassDarwin35,  was a x100p clone
19:10.24SpaceBass$15 off ebay
19:10.36Darwin35toss it in the trash
19:10.38Zeeekalways buy two of them on ebay
19:10.44SpaceBassnot going to lose sleep over it, except that its my offical, company paid for work number...and I'm going on vacation tomorrow and need to have my IVR working before I leave
19:10.49Darwin35buy in bulk
19:10.54SpaceBassmight just forward everything to a SIP trunk for the week
19:11.09Darwin35then get a nmbr threw us
19:11.29*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
19:11.30SpaceBassim not supposed to be doing this kind of stuff...or my real job...im supposed to be working on my ipod mix for skiing
19:11.36[TK]D-FenderSpaceBass : Yup.... I'm about to be dependant on things like that.  Getting Dry-loop DSL next week :)
19:13.04SpaceBasswhen I run ztcfg I get "unable to open /dev/zap/ctl"
19:13.10SpaceBassdoes that sound like hardware failure?
19:13.20SpaceBassdry-loop ?
19:13.21[TK]D-FenderSpaceBass : modprobe wcfxo
19:13.38[TK]D-Fenderdry-loop DSL is DSL without a voice line behind it (no dial-tone)
19:13.44SpaceBasswcfxo not found.....
19:13.48SpaceBass[TK]D-Fender, gotcha....
19:13.58SpaceBassi'd LOVE to get rid of my pots lines all togather
19:14.14SpaceBassi have FiOS for internet connectivity....just need to get my numbers ported
19:14.43SpaceBasscan't do an insmod on wcfxo either
19:14.58hypa7ia``i'm getting dsl on dry copper when i move
19:15.02hypa7iawe have it at my current house
19:15.05justinuoh, i got my dry DSL service about 2 weeks ago
19:15.14justinuwaaay better than the shit cable modem service I had
19:15.15*** join/#asterisk file[laptop] (n=jcolp@sjcc28x192.sjccnet.com)
19:15.23SpaceBassi had 'dry' dsl at one point...didn't know it was called that
19:15.38*** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br)
19:15.39SpaceBasswhat the hell.... cannot even find the zaptel module anymore
19:15.41hypa7iayeah they have started traffic shaping torrents on the cable provider up here
19:15.43hypa7iasucks
19:15.44SpaceBassdisapeared
19:15.50[TK]D-FenderSpaceBass : Because those stuck paying the telo for services they don't want are "all wet" :)
19:16.06[TK]D-FenderSpaceBass : recompile zaptel and try switching PCI slots.
19:16.15SpaceBassthats what I was thinking
19:16.38SpaceBassactually, I'm thinking that I'll finish the 3 hours of work I have, forward that pots number to my sip trunk and not worry about it for 2 weeks
19:17.17file[von]hiya
19:17.18[TK]D-FenderSpaceBass : and they can run on that fora  while?
19:17.32[av]bani[TK]D-Fender: ever figure out how to disable polycom log uploads?
19:17.33[TK]D-Fenderfile[von] : Getting outt my old plac at the end of next week.
19:17.38SpaceBass[TK]D-Fender,  I'll lose call waiting on on those numbers
19:17.46file[von][TK]D-Fender: found somewhere?
19:17.49[TK]D-Fender[av]bani : No way yet it seems short of changing permissions (yuck)
19:17.58iGotNoTime[TK]D-Fender, do you have a link to the manual that Zeeder was talking about for * ?
19:18.03[TK]D-Fenderfile[von] : I was approved at least 2 weeks ago.
19:18.24file[von][TK]D-Fender: excellent
19:18.30[av]bani[TK]D-Fender: that sucks. and still no overlap dialing either.
19:18.31[TK]D-FenderiGotNoTime : there is no "manual", but there are some good books, the first of which is ....
19:18.33[TK]D-Fender~thebook
19:18.34jbotrumour has it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
19:18.43SpaceBassnow to find the zaptel source on a@H
19:18.43[TK]D-Fender[av]bani : Still livable in my books....
19:18.50powerchiphow can I call a ring grup?
19:18.50[av]bani[TK]D-Fender: still violation of rfcs :))
19:18.57[TK]D-FenderSpaceBass : OMG... yeah.. that part could suck :/
19:19.10[TK]D-Fenderthey're RFC's, not RULES ;)
19:19.11SpaceBass[TK]D-Fender,  remember what I said about "when I build my next box...."
19:19.15[TK]D-FenderNO COMMENT!!!!
19:19.28[av]bani[TK]D-Fender: they are rules if you want to interoperate (rfc2833)
19:19.30[TK]D-FenderSpaceBass : When "sooner" meets "later"!
19:19.39[av]bani[TK]D-Fender: sip and rtp are rfcs also.. they are _rules_
19:19.39iGotNoTimethanks :)
19:19.41*** join/#asterisk kuku5 (n=kuku5@c-71-228-21-166.hsd1.il.comcast.net)
19:19.51kuku5sup all
19:19.51SpaceBassahhhhh /usr/src/zaptel
19:19.54[TK]D-Fender[av]bani : Yeah... and how far off is * in SIP compliance all by itself?
19:20.15[av]bani[TK]D-Fender: * does overlap dialing :))
19:20.16[TK]D-Fender[av]bani : no, just increasingly "strong" suggestions ;)
19:20.19SpaceBassnothing a little make clean ; make install shouldn't fix
19:20.41[TK]D-Fender[av]bani : I'm talking SIP support.... not just an aspect of it in reference to other channel types...
19:20.51SpaceBassshit! remember what I did now....installed all the centos updates a few days ago...hadn't bounded asterisk since then
19:21.04SpaceBassdamn, make failed
19:21.09[av]bani[TK]D-Fender: at least * is fixing its problems :))
19:21.09*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
19:21.15[av]bani[TK]D-Fender: sip-b this summer, yay
19:21.29[av]bani[TK]D-Fender: now try to get polycom to implement overlap dial. lollerskates?
19:21.49tamp4xanyone here use spandsp? when i load asterisk -vvvvvvvvv  it stops loading when app_rxfax.so loads....any ideas why?
19:22.05[TK]D-Fender[av]bani : "supposedly".... that's a BIG plus for inter-op for sure and drags things like the SPA's back into the realm of "suggestable" and elevates Polycom further still...
19:22.46[TK]D-Fender[av]bani : not forgetting that ther is an ACD patch for Polycom's signalling that could very well be merged in at that point.
19:22.55*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
19:23.04asterboydone digesting that link Zeek.
19:23.56*** join/#asterisk Vyeperman (n=Vye@ip68-6-130-118.sd.sd.cox.net)
19:24.32asterboyNow the only thing I can see that I may need to do is passing my number via the ${EXTEN} variable, into the Dial plan so I don't have to re type it.
19:25.36asterboyBut how do I get past having to dial a # to execute the application?  use of "s"?
19:26.18asterboynope, just get fast beep.
19:27.06willt[work]was it ulaw that provided the best sound quality?
19:27.31[TK]D-Fenderwillt : the best quality suppoted by the PSTN, yes.
19:27.54*** join/#asterisk ToTo (n=ToTo@host154-207.pool872.interbusiness.it)
19:28.01fourcheeze[TK]D-Fender: surely you can get better down pstn if you compress?
19:28.09[TK]D-Fenderwillt : At the same time as being pretty much the least load on * for converting for that purpose as well...
19:28.18ms345Anyone know the proper timing settings for multiple pri's from the teclo?  Do I set the span=1,1,0,esf,b8zs for my primary pri_cpe span and span=2,2,0,esf,b8zs for my 2nd pri_cpe span?  I assume all my pri_net spans that go to my PBX, routers, and other cpe are  span=<span#>,0,0,esf,b8zs.   Does that sound right?  I'm going off the material here http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax
19:28.26asterboyIf I don't want any numbers, like pressing 9 for outside line, should not "s" execute the application?
19:28.47asterboyor "_."?
19:28.51*** join/#asterisk gambolputty (n=root@64.74.225.131)
19:28.56asterboypattern match anything
19:29.05justinums345: i think you're on track
19:29.06[TK]D-Fenderasterboy : You are not running AMP are you?
19:29.30asterboy~amp
19:29.32jbotrumour has it, amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
19:29.40ms345justinu - thx.  I'll give it a try tonight after-hours.
19:30.01asterboySince I don't know what AMP is, I'd have to say no.
19:30.10[TK]D-Fenderasterboy : You just want to dial a # to place an outgoing PSTN call without a prefix, right?
19:30.18asterboyYES!
19:30.35*** join/#asterisk kamranazeem (i=[U2FsdGV@203.135.40.97)
19:30.38[TK]D-Fenderast, then just use a pattern without a 9 in front.  thats what I've recently adapted mine to.
19:30.49[TK]D-Fenderasterboy : pastebin your extensions.conf and I'll take a look for you.
19:30.55asterboyok thx
19:31.07kamranazeemhello all, is this the right place to ask asterisk@home questions ?
19:31.11Darwin35thats it I quit ..
19:31.15Darwin35I give up
19:31.35file[von]yay giving up
19:31.51Darwin35I am going to startup my own company some how where I dont have to be treated like shit in phone support.
19:31.52justinulol
19:32.02kamranazeemI cannot get a simple setup work, One Asterisk 2.3 on centos, two XP machines. two extensions registered on AMP, cannot get audio on xten-lite
19:32.18justinu~amp
19:32.19jbotsomebody said amp was "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
19:32.44fourcheeze~a@h
19:32.49fourcheezehmm
19:33.06kamranazeemsorry I am a total newbie to IRC as well
19:33.24fourcheezekamranazeem try this: /join #freepbx
19:33.28justinuit's ok, it's just that no one here likes to support AMP
19:33.31justinubecause it's a pain in the ass
19:33.38kamranazeemok thanks very much
19:34.17fourcheezekamranazeem: when you decide that freepbx doesn't do what you want, come back :-)
19:34.20Hmmhesaysi think i found a bug in freepbx
19:34.40Hmmhesaysdial(SIP/<host>/<user>) isn't valid is it?
19:34.48file[von]Hmmhesays: 42
19:35.15Hmmhesaysheh
19:35.18asterboyhttp://pastebin.ca/45839 - look at the [poly] context.  I've changed it to everything with no success.
19:35.25wunderkinfile[von]: 24
19:35.30file[von]wunderkin: negative
19:35.31fourcheezecan I send an arbitrary SIP header out?
19:35.34asterboyIgnore the CALLERID part, taken that out.
19:35.37*** join/#asterisk homebrew-hsv (n=homebrew@mail.kancharla.com)
19:35.37kamranazeemfourcheeze oops. You mean, this chat room is better than freepbx ?
19:35.50Hmmhesaysyou can get to this chat room from freepbx
19:35.56fourcheezekamranazeem: I mean that asterisk is better than freepbx, so choose your poison
19:36.16asterboythe "_9." used in conjuction with ignorepat = 9
19:36.18kamranazeemok I want to ask about asterisk
19:36.28[TK]D-Fenderasterboy : so right no you have to dial 9 + something more, and the "more" is wasted and then you get a 2nd dial-tone and have to dial out....
19:36.47[TK]D-Fenderasterboy : ignorpat only workks on analog FXS on zaptel cars.
19:36.55[TK]D-Fenderasterboy : you are on a SIP phone.....
19:36.55asterboyyes, to at least get it working.
19:36.57[TK]D-Fenderno good
19:37.03kamranazeemI cannot get a simple setup work, One Asterisk 2.3 on centos, two XP machines. two extensions registered on AMP, cannot get audio on xten-lite
19:37.04asterboydidn't know about the only working on FXS.
19:37.09asterboyYes SIP phone.
19:37.13asterboyPolycom IP 600
19:37.39[TK]D-Fenderasterboy : this is what you want to do : exten => _XX.,1,Dial(Zap/1/${EXTEN})
19:37.39asterboyI've also take out the 9 so I can try it without having to dial a prefix first, using "_."
19:37.40kamranazeemthe speakers and mic at both machines are ok, I have checked using the "sound recorder"
19:37.48tzangeris there a preferred POE switch for Polycom IP501s?
19:37.59*** join/#asterisk tzafrir_laptop (n=tzafrir@88.153.166.193)
19:37.59[TK]D-Fenderasterboy : It'll through pretty much everything at your Zap card.
19:38.05[TK]D-Fenderand actually DIAL it for you....
19:38.06asterboyexcellent!
19:38.12asterboytrying...
19:38.33justinutzanger: any 802.3af switch will work, but you'll need the polycom 802.3af power injector
19:38.46*** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net)
19:40.18[TK]D-Fendertzanger : its not so much an injector as a splitter from a PoE supplied cable back out again....  as for switch any 802.3af should do.  D-Link & NetGear are the cheapest these days....
19:40.25Hmmhesayscraziness
19:40.26[TK]D-Fender+/-$500 for 24 ports.
19:40.41asterboyFINALLY!
19:41.06asterboyCan you imagine putting that in as an example SOMEWHERE in the docs.
19:41.11justinufender: what do you know about this "injector/splitter"?
19:42.38vuudCan I have * allow direct dial in from SIP phones?  Well, can I make it work anyway...
19:42.44asterboywhy do you need 2 "_XX." X's in the pattern matching space? or does it matter?
19:43.27asterboyNow I understand the ${EXTEN} passing through.
19:43.28octothorpe_[TK]D-Fender:  what to you mean $500 for 24 ports?  could you post a url?
19:43.43asterboy$500 is kick ass for 24!
19:43.58[TK]D-Fenderasterboy : there is no magic "all answers here" doc... * is just too powerful, but they could give a few better CONCISE samples.
19:44.02*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:44.28asterboywhy do you need 2 "_XX." X's in the pattern matching space?
19:44.35[TK]D-FenderOh, I'm sorry... $400 ... http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1205369&CatId=868
19:44.36fourcheezekamranazeem: is the client registering with * ?
19:45.03octothorpe_[TK]D-Fender:  thanks, that's awesome
19:45.19[TK]D-Fenderoops... Netgear for $335!  http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1697254&CatId=868
19:45.27[TK]D-FenderJust keeps getting better...
19:45.30asterboyand PoE
19:45.31asterboynice
19:45.47asterboythought that was for FXO ports though.
19:46.15[av]banii doubt it can provide 15.4W to all ports simultaneously
19:46.19[TK]D-Fenderasterboy : no, thats for powering your phones :)
19:46.30asterboyyes, see that now.
19:46.39[TK]D-Fender369W tot..... not unbelieveable....
19:46.53[av]baniyou need a big switching PS for >300W
19:46.55[TK]D-FenderI have lamps bigger than that...
19:46.56asterboyIn pattern matching the "_" is for beginning and "." for ending, hence the double X.
19:47.03asterboyok, starting to clue in here.
19:47.16[TK]D-Fenderasterboy : I just make sure not to through single digits at the pstn :)
19:47.16[av]bani[TK]D-Fender: providing >300W power is altogether different from consuming >300W
19:47.32asterboyah, ok.
19:47.33[TK]D-Fender[av]bani : True there is overhead, but lets not get neurotic....
19:47.41octothorpe_<PROTECTED>
19:48.13[av]bani[TK]D-Fender: i doubt it can provide 360W, i'm betting peak is more like 150W
19:48.26[TK]D-Fenderoctothorpe_ : They are.... the prices are just finally plummeting...
19:48.30[av]bani[TK]D-Fender: consider that your average desktop phone eats maybe 5W
19:49.10[TK]D-FenderOops!  Ignor the Netgear one!  Only HALF of the ports are PoE (unlike the D-Link DES-1526 I linked previously)
19:49.36[TK]D-FenderI knew netgear's LOWER products were 50/50, but took a while to find it on the 24 port.
19:49.44[av]baniyep, says there.. 180W max
19:49.46[av]banii was close
19:49.58SpaceBassdoes anyone make a sub $200 24port managed switch?
19:50.27[av]banihttp://www.dlink.com/products/resource.asp?pid=403&rid=1492&sec=0  <- max power consumption 180 watts
19:50.32GerbilNutdo incoming calls just come in as asterisk on the phones if they don't have an incoming callerid?
19:50.56SpaceBassGerbilNut, depends on dial plan....but typically yes
19:51.00*** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-243.dsl.pltn13.pacbell.net)
19:51.08*** part/#asterisk Grizzy (i=Generic@ppp-71-133-231-243.dsl.pltn13.pacbell.net)
19:51.23*** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-243.dsl.pltn13.pacbell.net)
19:52.42*** join/#asterisk sergeus (n=s@195.112.98.13)
19:52.51[TK]D-Fender[av]bani : then again how many heavy load devices are you planning on loading onto it?
19:53.07GrizzyIsn't Quell a medication for lice?  (/me ducks)
19:53.36Qwell[filetop]yes
19:53.41Qwell[filetop]~qwell
19:53.43jbotrumour has it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad.
19:53.59Qwell[filetop]wait, no
19:54.02Qwell[filetop]Kwell is
19:54.08Qwell[filetop]Quell means, basically, to kill
19:54.15Grizzyaaah,  ok.  : o )
19:54.15Hmmhesaysgod i love it when customers say they have an e&m wink t1 thats 5Ess
19:54.33GrizzyI think "to suppress"
19:55.17GrizzyThey know the details, eh?
19:55.55Hmmhesaysanyone on fwd I need to test something on this new install
19:56.14ManxPowerHmmhesays, I believe FWD has an echo test number
19:56.28*** join/#asterisk clive- (n=pirch@dsl-145-31-108.telkomadsl.co.za)
19:56.37[av]bani[TK]D-Fender: well, people look at "15.4w per port" and think OMG 360W
19:56.41Hmmhesaysyeah but then i'd have to open a web browswer and find it
19:56.42[av]bani[TK]D-Fender: it's deceptive at best
19:56.50justinuthere's an available power pool
19:56.52GrizzyI'll be brave:  741907
19:56.54SpaceBassHmmhesays,  use lynx
19:56.58SpaceBass:D
19:56.59justinuand the thing will send out SNMP traps if you get close to exceeding it
19:57.01Qwell[filetop][av]bani: What did we decide, about POE on the ciscos?  Which ones have 802.3af?
19:57.02*** join/#asterisk [Outcast] (n=outcast@222-152-110-218.jetstream.xtra.co.nz)
19:57.18[TK]D-Fender[av]bani : for $400 I think we can afford to STFU and jsut be happy we don't need a motgage for it ;)
19:57.27[Outcast]does asterisk support "P-Asserted-Identity
19:57.35tamp4xanyone here use spandsp? when i load asterisk -vvvvvvvvv  it stops loading when app_rxfax.so loads....any ideas why?
19:57.46homebrew-hsvDoes anyone know how to force  g.711 pass-through when also allowing g729 between two sip peers?
19:57.49*** join/#asterisk jmacz (n=jmacz@201.244.198.113)
19:57.51[av]baniQwell[filetop]: ~phones
19:58.01Qwell[filetop]I don't know how to use this system :p
19:58.08Qwell[filetop]no clue where/what the browser is
19:58.14justinucanadiafied
19:58.17Hmmhesays613 is fwd's echo test
19:59.04Hmmhesaysi think vonage just killed my super secret test account
19:59.19Qwell[filetop]~phones
19:59.20jbotextra, extra, read all about it, phones is at http://bani.anime.net/phones/
20:00.35*** part/#asterisk homebrew-hsv (n=homebrew@mail.kancharla.com)
20:00.53clive-anyone got any advice about setting up raid for the first time
20:01.04[av]banipray?
20:01.10Grizzyjbot_ softphones
20:01.17clive-I am bit confused, it wants a stripe size, but I am using raid 1
20:01.21Grizzy~softphones
20:01.41justinuwrong channel, yo
20:01.53Grizzynice chart, the phones one.
20:02.21[av]banithanks
20:02.44vuudI am trying to connect to my astrisk from XLite on the same segment via direct dial.  peer debug shows that astrisk is getting the invite, but then getting a SIP/2.0 404 Not Found from the calling client.  WTF?
20:03.02GrizzyIt needs an "add" button.  : o )
20:03.02vuudAt least that is how I think it is happening... the sip debug stuff is new to me
20:03.08SpaceBassanyone played wiht the new WIP-300 wifi phone?
20:04.41Hmmhesaysi love it when customers want ME to call their telco and find out their t1 settings for them
20:05.01justinuthat's typically what avendor does :P
20:05.06tamp4xwip 300 sucks
20:05.17Hmmhesaysheh, no... i'm support for the vendor
20:05.18justinuyou can't expect lusers to understand any of the telco jargon
20:05.32Hmmhesaysthen vendor can do the bitch work
20:06.02rollergrrlWhich distro should I go with for a production Asterisk server?
20:06.23fourcheezequeue holy war
20:06.25fourcheezecue
20:06.40NuggetIt doesn't matter in the least.
20:06.46tamp4xwindows xp
20:06.51jsharpWhatever you feel comfortable with.
20:07.05rollergrrlI can't decide on debian or gentoo
20:07.15*** join/#asterisk Noky (n=damian@200.69.211.18)
20:07.16Nokyhi
20:07.25fourcheezerollergrrl: just pick one - I use both
20:07.33rollergrrlI'm just wondering what others are using in production
20:07.33Nokyi want know what is the 'priority' in the dialplan's ?
20:07.35Hmmhesaysi like to run asterisk on your mom
20:07.54fourcheezerollergrrl: one downside of gentoo might be excessive cpu usage if you have to update stuff
20:07.55rollergrrlMy mom is dead
20:08.08rollergrrlahh true
20:08.14rollergrrlcan't have that
20:08.25Vituxgentoo isn't bad...
20:08.32Hmmhesaysrollergrrl: thats hot
20:08.43Vituxruns pretty clean...
20:08.49rollergrrlHmmhesays: and I'm not
20:08.53[TK]D-Fenderrollergrrl : Just make sure whatever you use has the required libraries to compile the components you'll require.
20:09.08tamp4xfreebsd 6.0
20:09.16Vituxbut yah.. updating is a pain in the ass sometimes but if anything I find it easier on gentoo to update then any other distro anyway
20:09.16Nokyhi
20:09.17GrizzyMandrake/Mandriva is a nightmare.
20:09.23rollergrrlIs it very difficult to get spandsp working in debian?
20:09.29*** join/#asterisk zotz (n=zotz@24.231.32.85)
20:09.37tamp4xahh spandsp
20:09.39Grizzyfreebsd 4.6 and 6.1 work; 6.0 wouldn't even boot.
20:09.42tamp4xi have toruble with that
20:09.57Vituxbsd is ok... not much support though last time I used it
20:10.07Vituxmind you that was a long ass time ago
20:10.10dpryorollergrrl: Nope, it's easy
20:10.17Vituxubuntu is also decent..
20:10.28dpryo:)
20:10.31Grizzybsd systems feel faster than linux.
20:10.33[TK]D-Fenderrollergrrl : Debian is a very good base for most things period.  Just keep an eye on the versions between * and SpanDSP.  its the mix that kills more often than not.
20:10.39tamp4xif its easy why did spandsp compile ad when i start asterisk asterisk stops loading when app_rx /tx fax is loaded
20:11.08AlexCTIJustinu: When I did a Update to the new version i have a problem with the module chan_modem.so the UPGRADE.txt said that i need upgrade it to chan_misdn.so  can you explian me how to do that?
20:11.24rollergrrlthanks for the info guys
20:11.55justinujust get rid of chan_modem.so
20:12.44AlexCTIjust remove? and that's it? do I need again the make install?
20:12.58justinuno
20:13.48fourcheezerollergrrl: I tend to build my own *, even on debian
20:14.39tamp4xprob spandsp will be easier sicne they have it as a package
20:14.55*** join/#asterisk dokhench (n=no@adsl-065-080-180-134.sip.bna.bellsouth.net)
20:15.06*** join/#asterisk Katty (n=angela@64.82.232.54)
20:15.07*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
20:15.24kuku5<PROTECTED>
20:15.29Kattyhi lads.
20:15.48justinuafternoon
20:15.58file[von]kuku5: because it's making progress?
20:16.14Hmmhesaysheh
20:16.20jsharpOne channel is passing call progress information to another channel.
20:17.05justinuset verbose 0, problem solved
20:17.20kuku5file[von]: can I stop it ?
20:17.41Hmmhesaysso you can't make calls?
20:18.22SpaceBassok...just had my telco forward one pots to one broadvoice number and my other pots line to another bv number, but they both appear to be coming in on the same bv trunk
20:18.40*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
20:18.51Nokyi want know what is the 'priority' in the dialplan's ?
20:19.06[TK]D-FenderKatty: mew.
20:19.35*** join/#asterisk JerJer[mobile] (n=jj@sjcc28x192.sjccnet.com)
20:19.49[TK]D-FenderNoky : Yes, I'm sure most of us know what "priorities" are....
20:20.00SpaceBassfor instance, mine is getting drunk tonight
20:20.14jsharpA man with a plan.
20:20.21SpaceBassthus exten =>s,1,Drunk(SpaceBass|scotch)
20:20.46JerJer[mobile]exten => s,2,DUI
20:20.56SpaceBassfollowed by exten => s,2,Bitch(about|job)     exten => s,3,Passout()
20:21.08SpaceBassJerJer[mobile],  naw...came too close once, will NEVER risk it again
20:21.19SpaceBasswhen I start I either cab it, make someone else drive or just dont leave
20:21.41JerJer[mobile]lol
20:21.49jsharpexten =>s,4,Wake()   exten=>s,5,StareOverInBed()    exten=>s,6,KnawArmOff()
20:21.58jsharpGnaw, too.
20:22.07SpaceBassI gotta catch a flight tomorrow morning
20:22.25SpaceBasscan;'t be too hung over...then again I can drink on the plane
20:22.58justinubad move
20:23.10justinubooze and fliying == not healthy
20:23.19SpaceBassdisagree.... do it every week
20:23.27SpaceBassnow when I'm the one doing the flying...thats another story
20:23.37justinuthere's a certain physiology to it
20:24.04justinujust drink lots of water with your booze
20:24.07SpaceBasswhat altitude?
20:24.13*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
20:24.22Hmmhesayswow i upgraded * now i can't call out vonage anymore
20:24.25justinuwell, everytime you fly commercial you're at an apparent altitude of about 8500 feet
20:24.33[TK]D-FenderKatty : Mar 25th is the big day....
20:24.50Hmmhesaysbig day for?
20:25.02[TK]D-Fender<PROTECTED>
20:25.03SpaceBassi'm going to be spending the next 10 days sleeping at 9,500 and days mostly over 12k
20:25.04Katty[TK]D-Fender: :>
20:25.27justinuheh
20:25.34justinuamerican beer is probably ok
20:25.37jbalcombtoo many damned Asterisk on Debian HOWTOs to choose from
20:25.43Kattyi wouldn't know, justinu, i don't drink it.
20:25.43SpaceBassand i find that gin martinis go down much better at 9k feet!
20:25.49justinume neither
20:25.59jbalcombAnyone recomend the /best/ guide on installing Asterisk on Debian?
20:26.08[TK]D-Fenderjbalcomb : Just iognore the ones with "apt-get install asterisk" and might do OK :)
20:26.14Hmmhesaysi want a big black jet with a bedroom in it gonna join the mile high club at 37,000ft
20:26.22*** join/#asterisk medusaXX (n=medusaxx@p54A98532.dip0.t-ipconnect.de)
20:26.23justinuyou don't need that
20:26.36SpaceBassHmmhesays,  did that in college on the way to paris once....comercial flight
20:26.37jsharpHmmhesays:  Go business class on Virgin Airlines.   They've got sleepers.
20:26.41fourcheezejbalcomb: just do it, it's not hard
20:26.41[TK]D-Fenderjbalcomb : just install the base from debian, then download * stable from digium's FTP and compile from source
20:26.44jbalcomb[TK]D-Fender thats no good eh? ;)
20:26.49SpaceBassalso...ammmm...took care of that as a pilot once too
20:27.10justinuyou're a rated pilot?
20:27.14SpaceBassyeah
20:27.21jbalcombXXX rated pilot
20:27.25justinui've got a private single, and multi
20:27.25SpaceBassLOL
20:27.28willt[work]LOL
20:27.38SpaceBassjust private single here
20:27.42justinunever did the deed while being PIC
20:27.54SpaceBassand i fly so much comercially I have a hard time keeping it up ....ok...leme rephrase that
20:28.17medusaXXhi
20:28.30jbalcombi did it in a phone booth in a bar once, rather risque i felt.
20:28.36medusaXXi always get this error when starting asterisk: moh_register: Unable to open pseudo channel for timing
20:28.53medusaXXyou know what causes that?
20:29.04jsharpmedusaXX:  You have a zaptel device installed or ztdummy running?
20:29.10medusaXXno neither nor
20:29.16jsharpThere ya go.
20:29.26jsharpNeed one or the other to make MOH work worth a crap.
20:29.42medusaXXuhm ok
20:30.12medusaXXi thought i would have disabled all modules i dont need, but i missed that one obviously
20:30.13medusaXXthanks
20:31.10[TK]D-Fenderjbalcomb : nothing to it...
20:31.10SpaceBassi changed my MOH to Copland's Rodeo....really cracks people up to be put on hold to the "beef, its whats for dinner" comericial
20:31.31*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
20:31.48digg10how can i do macros in extensions from db?
20:31.49jsharpI inadvertantly changed ours to Lords of Acid.  Oops.
20:31.58*** join/#asterisk file[laptop] (n=jcolp@sjcc28x192.sjccnet.com)
20:33.03justinudarling come here and fu...
20:33.06*** join/#asterisk nroej (n=joern@heaven.cyphertext.de)
20:33.07nroejhi
20:33.44medusaXXjsharp: wait :)
20:33.55medusaXXi think i misunderstood that
20:34.06medusaXXmoh is music on hold, right?
20:34.12jsharpYeah.
20:34.14medusaXXwhy do i need a zaptel device or a ztdummy for htat
20:34.27justinugood question...
20:34.27medusaXXi thought moh would be something zaptel related
20:34.35medusaXXsorry
20:34.47jsharpcause it dervives clocking and timing for the audio from zaptel.
20:34.54medusaXXahh
20:35.01medusaXXare there other music on hold modules?
20:35.08*** join/#asterisk the_magic_bean (n=mhermsdo@209.43.15.211)
20:35.39medusaXXi dont like the idea of adding a module to the kernel
20:36.15digg10ne1 knows how to do macros while using sql extensions?
20:36.33jsharpNo other MOH modules.  Either zaptel or nothing.
20:36.48jbalcombWhy isn't asterisk-sounds in the download directory?
20:36.52medusaXXsad
20:37.01medusaXXthen i need that stupid module
20:37.15*** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net)
20:37.52clive-medusa, just install it, it works fine, and wont upset your kernel
20:38.07Grizzymissed the mile high club, though I've got a pilot friend.
20:38.11Noky[TK]D-Fender
20:38.16Nokywhat is the priority?
20:38.32medusaXXit's just that i need to remember that module every time i recompile my kernel
20:40.32[TK]D-FenderNoky : Which priority?  What ARE you talking about?
20:41.10heisondoes anyone know how to make 7960's use RFC2833?
20:41.29heisonafter a recent upgrade from 7.5 to 8.2, inband DTMF no longer works...
20:41.32jbalcombheison RFC2833 for Cisco is referred to as PVT
20:42.00ManxPowerI REALLY hate it when someone releases a Windows only compiled TCL script and won't release the source
20:42.23jbalcombheison or maybe is AVT...
20:43.35justinudtmf_inband: 0
20:43.39justinudtmf_outofband: avt
20:44.21jsharpI have mine set for avt_always
20:45.10jsharpheison:  Are there any new/nifty features to SIP 8.2?
20:45.20jsharpAll I can find are bug fix release notes.
20:45.42heisoni don't know, i was on MGCP and was trying to get the stupid thing to talk SIP, according to Cisco, they recommend 8.2
20:45.47heisonall my other phones are still 7.5
20:45.59jsharpAnd I can't download 8.2 cause my damn smartnet contract expired.
20:47.11SpaceBassjsharp, in the same boat
20:48.42*** join/#asterisk Peaceful (n=Peaceful@70.98.162.62)
20:49.25PeacefulDoes asterisk still require _exactly_ version 0.59r of mpg123, or will newer versions work?  (like 0.59s)
20:49.42*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
20:50.15jsharpNewer should work.
20:51.06Nuggetwhen did that change?
20:51.23NuggetIt's traditionally required that precise version, nothing newer or older.
20:51.33Noky[TK]D-Fender
20:51.38Nokyi'm talking about dialplans
20:51.42jsharpI'm running it with newer with no problems.
20:51.47Nokyextend => name,PRIORITY,...
20:51.54Nokywhat is this priority?
20:51.59jsharppriority is the step number in the dialplan.
20:52.06NuggetNoky: like line numbers in BASIC.  :)
20:52.13jsharpPriority 1 = step 1.  Priority 2 = step 2
20:52.13justinuGOSUB WITHOUT RETURN
20:52.15Nokyahah
20:52.16jsharpetc etc etc
20:52.32Nokyand... this prioritys must have an order?
20:52.45*** join/#asterisk illuy (n=assdf@85-64-194-107.barak-online.net)
20:52.47jsharpThey go in ascending sequence.
20:53.02Nokyok
20:53.04Nokythanks
20:53.21Nuggetand there can't be gaps in the sequence.
20:54.06jsharpAnd that's a PITA.  Anytime you want to insert a step in the middle of the dialplan, you have to renumber everything.
20:54.45Nokyahah
20:55.03Nokyok, thanks for the information :)
20:55.07NokyxD
20:55.13Nokyj# xD
20:59.38*** join/#asterisk Dr-Linux (n=nothing@host202-147-168-130.lhr.dancom.net.pk)
21:00.11*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
21:02.28*** join/#asterisk illuy (n=assdf@85-64-194-107.barak-online.net)
21:02.57*** join/#asterisk heison (n=heison@216.235.9.2)
21:05.17[TK]D-FenderNoky : If you are asking that question you need to read a lot more.....
21:05.23[TK]D-FenderNoky : Try TFOT
21:05.25[TK]D-Fender~thebook
21:05.27jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
21:05.44[TK]D-FenderNugget : sure there can ;)
21:05.48SpaceBasscan I use my modem and asterisk as an answering machine?
21:05.58Dr-LinuxTFOT is very good book
21:06.11[TK]D-FenderSpaceBass : Depends on your modem. NEXT!!@!@!@ (c) BKW
21:06.36SpaceBasswhat about changing my caller ID to prank my friend, can asterisk and my winmodem do that?
21:06.37SpaceBass:)
21:06.41SpaceBasssorry...I'll stop
21:06.45Dr-Linuxanybody knows if SJphone supports g729 ?
21:08.17[TK]D-FenderDr-Linux : nope.
21:08.45[TK]D-FenderDr-Linux : you won't find downloadable OSS soft-phones with it because of the licensing.  Common sense.
21:10.24Dr-Linux[TK]D-Fender: but license is for g729, so i thought may be SJphone support it,  xlite doesn't
21:11.33[TK]D-FenderDr-Linux : Ask yourself how they could control it if you just DLed it from anywhere....
21:11.55Dr-LinuxDLed :S
21:12.13Dr-Linuxhhm.. you are right
21:12.37Dr-Linuxhowever i have EyeBeam for unlimited users
21:12.57Dr-Linuxthat may support g729
21:13.19*** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it)
21:14.02justinueyebeam has g729, indeed
21:14.11markithi, today when I receive a call throught FWD, seems I receive it twice... any clue? anyone else with this problem? I've found an old post about other having experienced this, but not solution in that post
21:14.24Peacefulso, if mpg123 0.59s will really work with asterisk, the "You have the WRONG version of mpg123... you need .59r" message in the makefile should probably be modified
21:15.28_Sam--Peaceful :  you could always use native moh
21:16.12Peaceful_Sam--: "moh"?
21:16.23exonicanyone around here know what the ExtraChannel paremeter is for using Action: Redirect ?
21:16.46_Sam--moh = moooosic on hold
21:17.15*** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-164-89.lsanca.fios.verizon.net)
21:18.41Hmmhesaysto fall in love and fall in debt, to alcohol and cigarettes
21:18.53Hmmhesaysand mary jane to keep me insane sniffin someone elses cocaine
21:19.21exonicHmmhesays, don't for get the leuds
21:19.34exonicludes*.. (spelling?)
21:19.52GamercjmFor GMS format sound, what does the sample rate need to be at? or does it not matter
21:21.44markitGamercjm: aseterisk gsm sounds are at 8000 hz (8Khz)
21:22.13Gamercjmthanks
21:23.42wasimand for that matter, are all other formats as well
21:28.01tamp4xanyone here use spandsp? when i load asterisk -vvvvvvvvv  it stops loading when app_rxfax.so loads....any ideas why?
21:28.22stoffelltamp4x. what version? (of * and spandsp)
21:30.05Darwin35sex
21:30.08Darwin35~sex
21:30.09jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
21:30.58Darwin35~seen jbot
21:31.04jbotjbot <i=ibot@pdpc/supporter/active/TimRiker/bot/apt> was last seen on IRC in channel #debian, 147d 6h 14m 57s ago, saying: 'rumour has it, sarge is Ten-HUT!  Fall in!  Sarge is the code name for the current stable Debian release, version 3.1, released on June 6th, 2005. Ask me about <install debian>, or i guess sarge is the biggest lump of free as in ...
21:31.18De_Moncd?
21:31.23stoffelllol
21:31.49Darwin35cd = change direction
21:31.57De_Monah
21:32.15Darwin35or charge date
21:32.25stoffelllool
21:33.14*** join/#asterisk eivindtr (n=eivindtr@062016241059.customer.alfanett.no)
21:33.57Darwin35ok wich one you bitches stole my wallet
21:34.38stoffellif it's free, it's okay :p
21:34.42Darwin35thnks nugget
21:34.59eivindtrHi all. I have a system (1.2.1 with Sangoma A101) where I experience a longer delay on inbound than outbound voice (in the area 100-200 ms). Anyone got any ideas what might cause it?
21:35.10tamp4xstoffell : asterisk 1.2.5 ; spandsp 0.0.2pre25
21:35.21ManxPowerDarwin35, I'm sure it's nobody that knows you -- if they knew you they would know you are poor 8-)
21:35.40ManxPowereivindtr, we really can't help you with Sangoma
21:35.42gaupestoffell: you've got thomson to update voip-info :)
21:35.46eivindtrI'm trying to learn about the buffering in zaptel, but I can't really figure  it out...
21:35.48stoffelltamp4x, you used the howto like http://asteriskguru.com/tutorials/spandsp.html
21:35.50GamercjmAny one try the: Asterisk Realtime Voice Pitch Changer
21:35.52Darwin35heheh thnks manx
21:36.02ManxPowereivindtr, zaptel basically doesn't buffer
21:36.04stoffellgaupe, yeah, some1 just was faster :p but no BLF yet.. MWI works though..
21:36.10justinueivindtr: echotraining
21:36.22gaupestoffell: will try it later tonight :)
21:36.48eivindtrjustinu: I tried to disable echocancel alltogether, but it didn't really help..
21:36.56stoffellgaupe, it's cool, blf is coming, waiting for email from alcatel to get timing :)
21:36.59ManxPowereivindtr, we really can't help you with Sangoma
21:37.06eivindtrManxPower: Thanks :)
21:37.15Darwin35Manx pvt me
21:37.30gaupestoffell: I was hoping for the norwegian translation to show up too...
21:37.53stoffellgaupe, no no, all norway people I know, speak very well english :P
21:38.06gaupestoffell: I know :)
21:38.38eivindtrIs it right that the ZT_POLICY_IMMEDIATE is only relevant at the setup of the rtp-stream, or is it used throughout the communication?
21:38.39tamp4xyes stoffell
21:38.50stoffellwhat error you get tamp4x?
21:38.57ManxPowereivindtr, zaptel doesn't know anythinhg about RTP
21:39.20tamp4xno error reported stofffel
21:39.35stoffelltamp4x, check /var/log/asterisk/full ?
21:39.38eivindtrManxPower: my bad... I really knew that :/  but the question stands... sortof
21:39.51*** part/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net)
21:39.58Darwin35ok we all need to ppol our money together and open a giant voip provider
21:40.08tamp4xcannot open `/var/log/asterisk/full' for reading: No such file or directory
21:40.30stoffelltamp4x okay, where did you hide your log then? ;)
21:40.32ManxPowertamp4x, stop.  step away from the asterisk server.  Take a class on Linux or Unix
21:41.22tamp4xthe debug log?
21:41.53stoffellyeah, you should find why it doesn't start *
21:42.02tamp4xahh i see  [app_rxfax.so]2006-03-16 09:17:04 WARNING[25479] loader.c: libtiff.so.3: cannot open shared object file: No such file or directory
21:42.43stoffellseems like you don't have libtiff.so.3 , you should install libtiff
21:43.20fourcheezeDarwin35: ok, I have £5
21:43.24tamp4xi have it installed
21:43.36tamp4x/usr/local/lib/libtiff.so.3
21:43.36tamp4x<PROTECTED>
21:43.50fourcheezetamp4x: do you have /usr/local/lib in /etc/ld.so.conf ?
21:43.56stoffelland /usr/local/lib/ is probably not in your ld.so...
21:44.12stoffellwasn't worth typing that for me :)
21:44.19fourcheeze:-)
21:44.54*** part/#asterisk nroej (n=joern@heaven.cyphertext.de)
21:45.11tamp4xyeahaw
21:45.41fourcheezetamp4x: did you run ldconfig ?
21:45.46tamp4xthanks four and stoffell
21:45.48tamp4xyep
21:46.04stoffellhehe, now the fun starts, good luck tamp4x ;)
21:49.52exonici've had nothing but truoble w/ faxes :)
21:50.06jsharpJust the fax, ma'am.
21:50.10exonicthinking of takign it off asterisk all together soon enough
21:50.17stoffellexonic, i only use it to rx, .. works fine
21:50.25exonicstoffell, yeah, rx is money
21:50.34exonicstoffell, tx from a SIP phone on the other hand....
21:50.49exoniceven with T.38 I havn'thad much success
21:50.56exonicabout 10% fail
21:51.05tamp4xhmm sending out a fax seems to crash asterisk
21:51.10stoffellok, good to know..
21:52.09exonicstoffell, how do u send ?
21:52.12exonicstoffell, land line?
21:52.14*** join/#asterisk drs9 (n=DRS@host86-133-127-224.range86-133.btcentralplus.com)
21:52.26*** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com)
21:52.26stoffellexonic, yes, hylafax and a good old usrobotics:d
21:52.44stoffellbut, with an analog->isdn to a BRI ofcourse :D
21:53.11exonicstoffell, does hylafax work w/ any digium hardware?
21:53.11*** part/#asterisk drs9 (n=DRS@host86-133-127-224.range86-133.btcentralplus.com)
21:53.41stoffellexonic, i use analog modem (serial) that connects to a converter to connect it to BRI card
21:53.53stoffellbut i guess the analog cards can use a modem with hylafax, yes
21:54.17exonicahh, I was thinking it'd be awesome if I could use a single port PRI card and use hylafax to get 23 channels for faxing
21:54.34stoffellexonic, hm, no idea on that.. guess not..
21:54.35SpaceBassI still have fax issues....i can get asterisk to recieve fine, if I manually transfer the call
21:54.42SpaceBassbut I all the pages are blank
21:54.54exonicSpaceBass, sending is trouble
21:55.01exonicSpaceBass, i'm in the same boat
21:55.12SpaceBasssending blanks?
21:55.26exonicSpaceBass, i've had blanks, black smears and half filled black pages...
21:55.42SpaceBassthats like the 4th unintentional innuendo ive made today
21:56.01SpaceBassI can send fine usaing an analogue fax connected to an ATA, even over SIP trunks
21:56.02_Sam--for the record, USRobotics sucks with hylafax.
21:56.03Gamercjmanyone try Asterisk Voice Changer 0.3
21:56.09SpaceBassits recieving that sucks
21:56.16exonicasterisk is infertile  when it comes to sending :)
21:56.18_Sam--multitech modems are recommended for hyla :)
21:56.19SpaceBassGamercjm,  I've seen it...havent tried it
21:56.54jsharpGamercjm:  I was playing with it earlier.  Quality kinda sucks.  Sounds *very* processed.
21:57.40Gamercjmoh, I was gonna try it out, Would it be even worth testing?
21:57.58_Sam--why would you want to sound like a girl? :)
21:58.01exonicGamercjm, if you're trying to make a scary movie =)
21:58.11jsharpIts easy enough to set up.  Maybe you'll have better luck.
21:58.28SpaceBassits even better if you change your CID with nufone or something similar
21:58.42SpaceBassbut be careful...I had that prank backfire last weeked!
21:58.46GamercjmDid you have any trouble installing, or did it affect your regular usage
21:58.51jsharpNo and no.
21:58.52Gamercjmi have nufone
21:59.14*** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca)
21:59.32Gamercjmdid you install from CVS?
21:59.46Gamercjmsounds like I would have to reinstall asterisk with the regular patch
22:00.31jsharppatched my work/compiled tree of 1.2.4, then just re-ran make && make install
22:00.51jsharpIt only compiled app_voicechanger.so, since everything else was already compiled and there were no dependencies.
22:01.07Gamercjmu used CVS?
22:01.11jsharpNo.
22:01.14jsharp1.2.4 from ftp.digium.com
22:01.28Gamercjmoh
22:01.40*** join/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net)
22:02.26asterboyWhen another call comes through on a pots line and I want to flash, how do you administrate those kind of calls?
22:02.33*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
22:03.26*** part/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net)
22:03.34jsharpDial a certain DTMF sequence on your phone and the zaptel interface will flash the line.
22:03.40jsharpI forget the sequence, though.
22:04.08Seldon1975* wiki page is down - can someone tell me how to tweak the MOH volume?
22:04.11ManxPower*0
22:04.19jsharpThat's it.
22:04.34jsharpDial *0 and your zaptel interface will flash.
22:04.43*** join/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net)
22:06.17Gamercjm:/
22:06.18Gamercjmchecking for C++ compiler default output file name... configure: error: C++ compiler cannot create executables
22:06.42*** join/#asterisk RoyK (n=roy@213-150-148-98.telenor.se)
22:07.39*** join/#asterisk jeffoatrulez (n=jeffoatr@slim-eth0.horizonlive.net)
22:08.12*** join/#asterisk mroth_imm (n=chatzill@63.65.26.220)
22:09.39mroth_immanyone have any thoughts on irqbalance...should it be running on a high-volume asterisk server
22:09.53jeffoatrulezi have a musiconhold() question: if i'm not sending frames from my client, will it not send music back?
22:10.53*** part/#asterisk illuy (n=assdf@85-64-194-107.barak-online.net)
22:11.00Dr-Linuxjsharp: on what phone? *0
22:11.04jeffoatrulezit's working for me via a phone, but not through a muted iaxclient implementation.
22:12.07ManxPowermroth_imm, asterisk does a pretty good job of IRQ balancing
22:12.26mroth_immManxPower: it's the network interface i'm concerned about
22:12.38ManxPowerjeffoatrulez, if you are not sending frames, astersk won't send frames
22:12.49mroth_immirqbalance is putting each NIC on its own processor in my SMP system (4 CPUs)
22:12.51ManxPowerthat is why Asterisk does not support silence supressiion
22:13.09Dr-Linuxquestion about sip, if the genral section codec is set to g711 , will that work, if i define anyother codec for indivisual user account?
22:13.24mroth_immnet result is that CPU 0 handles ALL interrupts from the interface handling the RTP streams and so it runs at a higher percentage than the others
22:13.42mroth_immmeaning the box will be cpu bound on the processor handling those interrupts
22:13.54jeffoatrulezmanxpower: we do the same with app_conference, though. i don't send frames, but app_conference sends me other folks' frames.
22:14.09De_Monvoip-info down?
22:14.17mroth_immyou can use irq affinity to spread the interrupts via /proc/irq/#/smp_affinity
22:14.35jeffoatrulezi.e. i can listen without sending. or i should say, we send pings, etc.
22:15.00De_Monserver is up, but wiki not talkin
22:15.05mroth_immbut irqbalance tweaks those values as well on a ten second interval...so i'm wondering if I can just turn it off without problems
22:16.00jeffoatrulezi started looking through channel.c and the ast_generator code, but it was non-obvious how to get the generator code to send regardless of incoming frames.
22:17.18asterboyMissed if there was an answer...phones are off the hook literaly.
22:17.21asterboyMissed if there was an answer...phones are off the hook literaly.
22:17.24asterboyWhen another call comes through on a pots line and I want to flash, how do you administrate those kind of calls?
22:17.33jsharpDial *0
22:17.45jsharpAnd your zaptel device will flash your POTS line.
22:18.04asterboythx jsharp!
22:20.04*** join/#asterisk tainted_ (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net)
22:20.37SpaceBasstrying to flash pots is miserable
22:21.00SpaceBassget your telco to call forward on busy to a BroadVoice byod lite account
22:23.17Seldon1975hello all, I've tried playing with mpg123's '-g' parameter to decrease the gain, but when reload MOH and I listen to the on-hold music it's just as loud - can someone tell me how to make this quieter?
22:24.31*** join/#asterisk gtodd (n=gtodd@198.62.158.205)
22:24.43*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
22:24.50*** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)
22:25.29bancusJust for the edification of those who earlier saw me asking if a clock running too fast could affect asterisk playback, the answer is yes.
22:25.40gtoddI have the AMP GUI running in Asterisk@Home VMWare image ... and an idefisk softphone running on the host system
22:26.06bancusUpgrading my kernel fixed the problem.
22:26.11*** part/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)
22:26.28gtodd* is running on 192.168.40.128 .... but I can't get idefisk to register as a peer or friend to use that system
22:26.40rajiv|workhow many voice trunks are recommended per person at a company ? like 1 trunk for eveyr 5 employees ?
22:26.42SpaceBassgtodd,  host or bridgged netowkring?
22:26.50gtoddI've set up an extension ... so
22:26.50SpaceBasssame subnet as your box?
22:26.58gtoddhmm no
22:27.10SpaceBasstrying to even check voicemail while * is running in VMware sucks
22:27.23Seldon1975rajiv|work: if you need a rule of thumb, 1 trunk per 5 employees is as good as any...
22:27.26SpaceBassseriously...ebay a $40 PIII or something...you'll be glad you did
22:27.41SpaceBassgtodd,  can you ssh into your * "box"
22:27.41gtoddthe box is running 172.16.5.* the *@Home is running in 192.168.40.128 ...
22:27.44Seldon1975rajiv|work: WE HAVE 7 TRUNKS AND ABOUT 35 EMPLOYEES, SO...
22:27.48Seldon1975oops sorry caps
22:28.01SpaceBassgtodd, set VM to bridged networking (if its not) and put them on the same subnet
22:28.04SpaceBassbrb
22:28.12gtoddSpaceBass, I have access via vmware so ...
22:28.13gtoddOK
22:28.23rajiv|workSeldon1975: thanks
22:28.39Seldon1975rajiv|work: yw
22:28.42rajiv|workalso, anyone heard the term "3MG" referring to t1 voice/data? what is it
22:31.04Seldon1975rajiv: are you sure they don't mean 3mb
22:31.14Seldon1975rajiv|work: as in 3 megabit
22:31.55rajiv|workthat is what i would think, but they wrote 3MG sevearl times
22:32.38Seldon1975rajiv|work: hmm, I can only thtink the author is using it to mean megabit.  If it's not that I have no idea
22:33.18Seldon1975hello all, I've tried playing with mpg123's '-g' parameter to decrease the gain, but when reload MOH and I listen to the on-hold music it's just as loud - can someone tell me how to make this quieter?
22:33.23fourcheezeis t1 3mbit/sec ?
22:34.52Darwin35lick my balls
22:35.21Darwin35everything is working but vm with pgsql and realtime
22:35.38Dr-LinuxDarkhalf: :S
22:35.41Seldon1975T1 = 1.544 Mbps
22:35.42Darwin35now O have to figure out unixodbc
22:35.55Seldon1975<PROTECTED>
22:36.21Seldon1975or T1c = 3.152 Mbps
22:36.23Dr-LinuxE1 = 2 Mb/ps
22:36.43Seldon1975so maybe they were talking about T1c
22:37.54*** join/#asterisk Raszh (n=Spoon@66.253.253.210)
22:38.07RaszhHas anyone gotten Zaptel 1.2.* to work on Linux 2.4?
22:39.00fourcheezeDarwin35: unixodbc is easy
22:39.32[TK]D-FenderDarwin35 : yup, just download compile and run the GUI config and you're up and running fast....
22:39.36Dr-LinuxRaszh: yes, i'm using
22:39.47[TK]D-Fenderone of the few places I appreciated having a front end :)
22:39.55fourcheezenever tried a gui
22:40.03fourcheezeapt-get install unixodbc
22:40.55fourcheezethen hack on odbc.ini
22:41.21Seldon1975isnt ODBC really slow?
22:41.40Dr-LinuxSeldon1975: what's voice channel and data channel? :S
22:42.03RaszhDr-Linux:  how did you do so without CRC_CCITT support in 2.4?
22:42.43Dr-LinuxRaszh: 2.4 linux kernel right?
22:42.49Raszhyes
22:43.08Dr-Linuxi'm using RHEL, i just update it from RHN
22:43.46Raszhand Zaptel 1.2.*?
22:44.00Dr-LinuxRaszh: my both asterisk box have RHEL
22:44.18Dr-Linuxand both have zaptel 1.2.x
22:44.39Raszhdo you have your kernel config file?
22:44.43Dr-LinuxRaszh: RHEL AS 3  has 2.4 kernel
22:44.52*** join/#asterisk jeffoatrulez (n=jeffo@slim-eth0.horizonlive.net)
22:45.10Raszhfor that matter, can you 'locate ccitt' for me and see if you have a module in /lib/modules ?
22:45.15Seldon1975Dr-Linux: I don't understand your last question to me
22:45.24jeffoatrulezi was asking a question about musiconhold a bit ago, but my irc client exploded, so i lost any responses. if anyone still has responses on their screen, could you copy/paste it to me?
22:45.34*** join/#asterisk Nodren (n=nodren@64.193.95.10)
22:45.41*** part/#asterisk Nodren (n=nodren@64.193.95.10)
22:45.54Dr-LinuxRaszh: sorry right now i'm at home and using win .. i'll do it later
22:45.58*** join/#asterisk Nodren (n=nodren@64.193.95.10)
22:46.03Dr-LinuxRaszh: what OS you are using?
22:46.09RaszhGentoo
22:46.25Raszhbut I'm looking at the linux sources downloaded from kernel.org
22:46.43Raszh2.4 doesn't have CRC_CCITT support
22:46.54Dr-LinuxSeldon1975: as i read E110P T1 digium card can do 12 voice channel and 12 data channel,
22:47.03Nodrenanyone ever experience problems compiling zaptel into centos 4.2?
22:47.16Dr-LinuxSeldon1975: i can understand FXO/FXS, PRI/ISDN/T1/E1
22:47.33Dr-Linuxbut i didn't understand voice/data channels? :S
22:48.08Dr-LinuxRaszh: i'm not much good with kernel, but that works for me
22:48.24Nodreni cant seem to get zaptel to compile, i keep getting errors
22:48.36medusaXXi installed the ztdummy kernel module but i still receive this error when starting asterisk: moh_register: Unable to open pseudo channel for timing...  Sound may be choppy
22:48.38Raszheither you're mistaken or RH has patched CRC_CCITT into their 2.4 kernels
22:48.44medusaXXdo i need additional configuration after loading the module?
22:48.46Dr-LinuxRaszh: can you tell me when did Zaptel 1.2.* released ?
22:48.51*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
22:50.08tsumehmm
22:50.15tsumewhere can I get genzaptelconf?
22:50.19tsumeI'm missing it on this system
22:50.45ManxPowertsume, never heard of it
22:51.12tsume<PROTECTED>
22:51.13ManxPowermaybe you are confused and it's s 3rd party piece of software
22:51.15tsumethis is a fresh install
22:51.31ManxPowertsume, no, you are using some silly gui for Asterisk
22:51.33ManxPower~amp
22:51.36jbotsomebody said amp was "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
22:51.39tsumezapata-channels.conf
22:51.45tsumeManxPower: no I'm not
22:51.51tsumeManxPower: hardcore command line here
22:51.56ManxPowertsume, correct, asterisk does not generate any config files
22:52.07tsumegenzaptelconf takes /etc/zapta.conf and generate the sapta for me
22:52.10RaszhDr-Linux, I don'tk know when it was released
22:52.15tsumeI'm just curious where to get it
22:52.16ManxPowerAnd there are no sample config files with that name
22:52.34tsumewhere did this come from?
22:52.43tsumeI never installed any third party software
22:52.46ManxPowertsume, no idea, but I don't think it came from Digium.
22:52.49medusaXXi found it... the device nodes were missing
22:52.54tsumeManxPower: how odd..
22:53.04ManxPowertsume, so you downloaded Asterisk and Zaptel and then built from source and installed?
22:53.19*** join/#asterisk talljon84 (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com)
22:53.26tsumeManxPower: oh, I know where it came from, nevermind
22:53.33ManxPowertsume, where?
22:53.38tsumeManxPower: my friend copied configs from the other box he was playing with
22:53.44tsumehe was using amp,, heh..
22:54.01tsumeall existing "easy" configs suck
22:54.10*** join/#asterisk menger (n=menger@dsl-125-209-136-232.vic.veridas.net)
22:54.15tsumeI should make one, base it on GTK#
22:54.26tsumea decent one like how oracles db designer is
22:54.56tsumeManxPower: I was just curious if there was a magical way to configure this TDM2400 card :)
22:55.05tsumeTDM2400P (prototype)
22:55.10justinumagic... pay someone to do it
22:55.21tsumejustinu: thats what I'm for ;)
22:55.23talljon84I installed A@H 2.7 and it worked correctly until I rebooted. Now I am receiving the missing /dev/zap error.  I have installed named and dhcpd but everything else is fresh on the box. The udev.rules file has the right lines for zap but I do not understand what is going wrong. Does anyone have ideas?
22:55.26justinuheh
22:55.30justinuthen you're the magic! :)
22:55.41tsumejustinu: I'm not some idiot linuxer. I'm a hard core BSD guy and will make/use scripts to cut down my time
22:55.51justinulol
22:56.10tsumeanything I use will either make me more money, or decrease my time :)
22:56.10justinused, awk, perl, or what?
22:56.23tsumejustinu: whatever works ;)
22:56.28tsumeI know em all ;)
22:56.31justinuas long as it's not linux
22:56.52tsumejustinu: oh I don't care if its linux, but most linux dweebs sit there and do it by hand to be "leet" *chuckle*
22:56.58justinuhah
22:57.26tsumeits just halarious, you know. and they use gimp because they hate adobe making money from photoshop *chuckle*
22:57.31justinui guess there are a lot of linux n00bs
22:57.36justinulol
22:57.40tsumeeven though gimp ois more like PS1
22:57.49justinui came from solaris... *shrug*
22:58.00Nodrenanyone want to help me, i'm having a problem compiling zaptel on a 2.6 kernel
22:58.06Nodrenand i'm using the make linux26
22:58.11justinuyou don't need that anymore
22:58.13justinumake install
22:58.21Nodrenjust make install?
22:58.23justinuyep
22:58.24*** part/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com)
22:58.27tsumejustinu: I favor sun for improving postgres and releasing the code BSD licensed for sol
22:58.35tsume*the postgres code
22:58.40justinui like java...
22:59.03tsumejustinu: java wont run here ;) on NetBSD servers. not well anyway
22:59.08tsumeI'm poritng mono to netbsd
22:59.15justinuyeah... they've gotta release an opensource JRU
22:59.17justinuJRU
22:59.18justinuerr
22:59.20justinuJRE
22:59.21Nodrenit gives me an error
22:59.25Nodrenwhen i do just make install
22:59.41justinusun is being pretty stubborn about the whole JRE thing
22:59.46justinuopensource the bastard already
22:59.49tsumejustinu: once they release an opensource JRE, I'll be all over it. Development tools should be free, applications should be allowed not to
22:59.52justinuyep
22:59.56justinui'm with you
23:00.28tsumeI don't like java because you have to pay a license if you want to include it on embedded platforms
23:00.37tsumeI'd rather work on and port mono
23:00.42tsumemaybe opensource the changes
23:00.58Nodrenjustinu: i tried just doing a make install of zaptel
23:01.01Nodrenit gave me an error
23:01.04Nodrensaid Error 2
23:01.32justinupaste the error
23:01.37justinuhopefully someone will look at it
23:01.37ManxPowerActual error message:
23:01.40ManxPower** (.:11973): CRITICAL **: gtk_pizza_set_size: assertion `pizza != NULL' failed
23:01.44justinuheh
23:02.05justinus/pizza/muffin and file could have written that
23:02.08Nodrenmake[2]: *** [/usr/src/asterisk/zaptel/zaptel.o] Error 1
23:02.08Nodrenmake[1]: *** [_module_/usr/src/asterisk/zaptel] Error 2
23:02.08Nodrenmake[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
23:02.09Nodrenmake: *** [linux26] Error 2
23:02.13justinunot here
23:02.16justinu~pb
23:02.18jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
23:03.27mroth_immanyway to tell if a kernel installed from an rpm (*not* source) has certain constants set such as CONFIG_IRQBALANCE?
23:03.39justinuyou need the srpm
23:04.37mroth_immaaahh...good thinkin'...thanks, i'm a little flustered today : )
23:05.09mroth_immtrying to figure out what the need for a daemon that balances irqs is if the kernel is already doing it
23:05.31mroth_immirqbalance VS irq affinity
23:05.52Nodrenno suggestions?
23:06.29justinunodren, learn how to use pastebin to paste your error output
23:06.33justinusomeone will help you then
23:06.39Nodrenalright i will next time
23:09.20Nodrenhttp://pastebin.com/606425
23:09.49[av]banihttp://www.dell.com/downloads/jp/corporate/imagebank/ecc_1.jpg
23:10.19*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
23:13.00justinu~centosbug
23:13.01jbotcentosbug is, like, a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either.
23:13.22justinuthat's your problem, nodren
23:13.28Nodrenthanks!
23:15.21tsumehmm wow
23:15.32tsumeI need to download the HEAD of cvs for zaptel module
23:15.37tsumeto get this TDM2400 card to work :)
23:16.02ManxPowertsume, you should not need to.
23:16.04ManxPower1.2 should support it.
23:16.17tsumeManxPower: its not working for the channels I have installed
23:16.28ManxPowertsume, maybe you configured it wrong
23:16.38tsume1-12 dont work, the empty slots work. I just get no such device
23:16.55tsumeManxPower: doubt it, the card is full yconfigured
23:17.43tsumeworks fine for theh TDM400
23:17.47ManxPowerWell THAT sucks.  The README in the Zaptel source dir does not mention the TDM2400P at all
23:17.51tsumeso it must be the kernel module
23:18.11Nodrenawesome it worked great
23:18.11ManxPowertsume, does the kernel module show up in "lsmod"?
23:18.20tsumeyes
23:18.27tsumeelse the TDM400 wouldn't be working
23:18.40tsumezttool reports the card as there
23:18.42tsumeand configured
23:18.54ManxPowerand what does ztcfg -vvv say?
23:18.59tsumeanything trying to open a channel which has an installed module fails with "No such device"
23:19.20tsume28 channels configured
23:19.24tsumeI've done all this already ;)
23:19.25ManxPower28?
23:19.37ManxPowerthe card has 24 channels
23:19.49tsumeTDM2400P + TDM400
23:20.05ManxPowerand what kernel module do you have loaded for the TDM2400P?
23:20.08*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
23:20.14tsume1.2.4
23:20.24tsumeI'll just build the kernel module out of head right quick ;)
23:20.26shmaltzhi everyone
23:20.33ManxPowerno.  wctdm, wcfxo, etc.
23:20.45ManxPowergo right ahead, I don't think it will fix anything.
23:20.45tsumeoh
23:21.40tsumewctdm
23:21.46tsumewctdm24xxp
23:21.49tsumeoh wait here
23:21.53tsumeit ssupposed to be working
23:21.55tsume*hrm*
23:21.58justinuso what's so special about a TDM2400P?
23:22.04tsumeI'll try running head
23:22.09tsumejustinu: its a 24port  card
23:22.14ManxPowerjustinu, It has some nifty cool buzzword technology!
23:22.19justinui know what a regular TDM2400 is
23:22.20tsume1 module card == 4 lines
23:22.36tsumejustinu: 2400 is a prototype, period
23:22.41ManxPowertsume, and "lsmod" shows both modules loaded?
23:22.42tsumeits just TDM2400P
23:22.50tsumethis one has echo cancellation built onboard as well
23:22.57tsumeManxPower: yeah it does, strange, eh?
23:23.21ManxPowertsume, which driver loads FIRST?
23:23.35*** join/#asterisk brockj49464 (n=brockj49@63.87.56.236)
23:23.40tsumehmmm
23:23.47tsumethe TDM2400p
23:24.04ManxPowerso then you configure the tdm2400p modules first in /etc/zaptel.conf
23:24.08tsumeno wait
23:24.10tsumelet me check..
23:24.31tsumeManxPower: wht?
23:24.41tsumeManxPower: I just tell it to config 1-25 ;)
23:24.42ManxPowerThe order of the channels configured in /etc/zaptel.conf needs to be the same order as the kernel modules are loaded.
23:24.46tsume1-25
23:24.59justinuzaptel gets really pissy about stuff not matching just right
23:25.01ManxPowertsume, what modules do you have on the cards?
23:25.19tsumeFXS 1-25
23:25.21tsume3 FXO
23:26.07ManxPowerso you have signaling=fxs_ks then channel=1-25 then  signaling=fxo_ks and then channel=26-28
23:26.20ManxPowersorry.
23:26.27ManxPowerso you have signaling=fxo_ks then channel=1-25 then  signaling=fxs_ks and then channel=26-28
23:26.29ManxPowerthat is correct.
23:26.38ManxPowerfxo ports use fxs signaling
23:26.41tsumeyeah
23:26.44ManxPowerfxs ports use fxo signaling
23:29.20twisted[asteria]anyone know how to set ANI with Set(CALLERID(num))?
23:30.57justinuSet(CALLERID(num) = 12135551212)
23:31.00*** join/#asterisk fjean (n=fjean@201009210199.user.veloxzone.com.br)
23:31.03twisted[asteria]right, i know that much
23:31.08twisted[asteria]but that doesn't toggle the ANI flag
23:31.14justinuoh
23:31.38justinuis it in the presentation indicator?
23:31.47justinuSetCallingPres()?
23:32.01*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
23:32.07justinuscreened=yes or something
23:32.14*** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)
23:32.20fjeanhello guys ! ==-  anyone sucessfully installed the unicall patch into asterisk 1.2.5 ?
23:32.39bancusFor some reason, it seems like asterisk is not hearing keypresses coming in over sip. Any ideas?
23:32.49I-MODdtmf setting
23:33.01I-MODin sip.conf
23:33.20X-Rob~centosbug
23:33.27jbotsomebody said centosbug was a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either.
23:33.27I-MODthere are 2 or 3 different settings
23:33.33bancusit says inband
23:34.01I-MODtry the other options
23:34.09Dr-Linux~RHELbug
23:34.12twisted[asteria]justinu, heh...  well, i just used the deprecated application ;)
23:34.15bancusk
23:34.28bancusthe wiki says broadvoice needs inband, but that could be outdated, I guess
23:34.59Dr-Linuxhi justin
23:35.20justinuhello
23:35.40orlockHmmm
23:35.49orlockheya
23:36.01Dr-Linuxjustinu: how data channels work with T1 ? :S
23:36.18justinuyou set aside a certain number of timeslots for data
23:36.22Dr-Linuxi wish to read a doc about my question, but no luck to find some in google :(
23:36.38[av]banidata channels just use timeslots, and no signalling
23:37.25Dr-Linuxyes, but what is this "data" internet from same telco the PRI is coming from? or what
23:37.32justinuit depends
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23:37.44Dr-Linuxi have doubts
23:37.49justinuthe telco will MUX the dataconnection onto the T1 on their side
23:38.19justinuthen you demux the data timeslots to some kind of router
23:38.29justinuand the voice timeslots into some kind of telco switch
23:38.44Dr-Linuxlike T1 card has only port, where we plug the PRI from telco that have TS 24  ,  1.544 Mbps
23:38.48Dr-Linuxbut data :S
23:39.01justinuwell, you need a T1 card that can deal with fractional T1s
23:39.03[av]baniyour router handles it
23:39.09justinuthey make those
23:39.16[av]banior oyu can get a t1 mux
23:39.53Dr-Linuxmux ? :(
23:39.58[av]banimultiplexer
23:39.59orlockhmm
23:40.14[av]banisend some channels out 1 T1 port, and other channels out another T1 port
23:40.29[av]banior just get a cisco router and handle everything in one box
23:40.34justinuyeah, adtran TSU100 can do that
23:40.46justinuT1 out to pbx, V.35 serial to router
23:40.51justinubut that's oldschool
23:41.21rajiv|workmy company is moving to a new space. i suggeted getting a t1 line instead of the 8 pots lines we have now. it will be frac t3 supposedly with some voice and some data
23:41.22orlockStupid question - i need a register=blahblah:blah@voipprodier.com dont i?
23:41.33rajiv|workso i too would get a t1 card for an asterisk box ?
23:42.01justinuyeah
23:42.09rajiv|workrecommendations ?
23:42.21justinuDigium TE110P or Sangoma A101u
23:42.22Dr-Linuxjustinu: but what if i use all 23 B channels for voice?  i can do that right?
23:42.25rajiv|worki think also i do nto want to build the * box. who sells pre-built and tested asterisk boxes ?
23:42.31justinuDr-Linux: yep, then no data!
23:43.03I-MODrajiv|work: i know digium does turnkey systems
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23:44.00rajiv|worktheir bundles ?
23:44.19Dr-Linuxjustinu: data mean here >>  internet/bandwidth  right?
23:44.28justinuyes
23:44.32justinuthat's what you're talking about, right?
23:44.46rpmis this right, i called sprint the other day to get a quote on a PRI, they said they want 500-600$ a month for only 8 available B channels?
23:45.06justinuthat's high
23:45.06[av]baniyep
23:45.12[av]banithats sprint
23:45.14justinubut sounds ballparkish
23:45.23[av]baniyour ILEC can probably do better
23:45.38rajiv|workI-MOD: i see only their bundles on the digium site
23:45.48rpmif i register as a CLEC do you think i can get at-cost rates?
23:46.01[av]banino
23:46.03rajiv|worki just got a quote for $800 / mo for 3mb data + $8 / mon for each line
23:46.15justinuthey'll probably charge a CLEC more!
23:46.41rpm800$ a month for 3mb, from who?
23:46.53rajiv|workconversent, in boston
23:46.53bancusI-MOD: thanks, rfc mode fixed it
23:46.56*** part/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)
23:47.09[av]baniunless you become a facilities based clec, you can only get resale rates, and you can't sell to yourself
23:47.15rpmrajiv|work: is that two bonded t1's?
23:47.18rajiv|workrpm: well i should say that it is "3MG" which i'm thinking is 3mb since thats what we talked about.
23:47.22rajiv|workrpm: frac t3
23:47.29[av]baninow, you can setup a separate corporation and resale to yourself, but thats a lot of work
23:47.33rajiv|workoh. can i plug a frac t3 into a te110p ?
23:47.36justinuis that including the t3 loop charges?
23:47.42justinunot without a ds3 mux
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23:48.00rajiv|workjustinu: supposedly yes, that includes the loop charges
23:48.00[av]banitelcos like to do atm on ds3
23:48.00justinuyou can buy the CAC widebank 28s pretty cheap
23:48.09justinurajiv: that's an amazing price
23:48.17rajiv|work3 year contract
23:48.21justinustill
23:48.31[av]banijustinu: ds3 is pretty cheap actually, its the bw that costs
23:49.01justinuds3 loops charges are like 3k a month around here
23:49.14rajiv|workwhere is that?
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23:49.24justinuLos Angeles
23:49.32rpmi heard you can get a ds3 for about 400-500$ a month
23:49.51[av]banithats just the local loop cost
23:49.57[av]baniyou need services on top of that
23:50.00justinui remember them costing a lot more when we installed them at our facilities in calabasas
23:50.34[av]baniwe pay ~$1500 for our ds3 loop, and then services on that are pretty cheap
23:50.41[av]baniends up much cheaper than buying T1s
23:50.59[av]bani1 meg pvc is only like $15/mo
23:52.19justinuframe relay or ATM?
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23:52.27criptosHi everyone
23:53.07[av]baniatm, though you can get atm->fr conversion too
23:53.13criptosI have a pap2 at a remote asterisk, and when I dial from the pap to another nother asterisk box, I got no sound..
23:53.31[av]baniyou can have the telco route frame relay PVCs to your atm
23:53.33criptosI´m using ulaw all the way, at the iax transport and beweet pap2 and asterisk...
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23:55.21rajiv|workjustinu: good to konw we're getting a deal.
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23:55.55Zodiacalanyone know how to use the one touch recording feature? mines setup for *1, but it doesn't do anything
23:55.59*** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk)
23:56.07Zodiacalit just gives me a fast busy tone.. i tried it during a call and it doesn't do anything it seems
23:56.23I-MODrajiv|work: you'll likely have to call into digium and ask sales to find out about turnkeys
23:56.24Zodiacali read it suposed to store them in /var/spool/asterisk/monitor but it doesn't seem to
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