irclog2html for #asterisk on 20031118

00:00.02pattiejait crawled
00:00.13*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
00:00.15Corydon76-workLoad is defined as the average number of processes in a short wait state.  IO is considered to be a long wait state.
00:00.25pattiejaand after MailScanner killed itself, it got much better
00:01.03cypromisimap servers
00:01.03cypromisabout 40.000 mailboxes
00:01.03cypromisgot sometimes far over 100.0
00:01.03cypromisand still where useable
00:01.20cypromissince than  don't really trust the LOAD value on linux and slowlaris
00:01.42dougheckahaha
00:01.50kapejodjust make sure your * runs as an RTAI softrealtime task ahead of the linux scheduler ;-)
00:02.43kapejodthen you can run make bzImage -j without jittering your calls
00:03.02dougheckahah
00:04.58kapejoddont let the penguin touch your *! ;-)
00:05.02[Barney]am I missing something, or is there a bug in cdr_pgsql ?
00:05.19[Barney]it doesn't seem to be logging the accountcode if specified in config files
00:05.50[Barney]and I can't figure out how to turn on debug logging ... adding debug in logger.conf doesn't show and log entries from cdr_pgsql.c :(
00:07.16Corydon76-workIt's not a bug in cdr_pgsql
00:07.19*** join/#asterisk Noodleman (~tuckerm@adsl-66-72-30-42.dsl.kntpin.ameritech.net)
00:07.35Corydon76-workThat occurs no matter which cdr log you use
00:07.48[Barney]cdr_csv seems to track the account code though
00:08.17Corydon76-workOh, well, file a bug in the bugtracker then
00:09.15[Barney]yup ... going to ... just wanted to check I wasn't missing something before doing so
00:09.32[Barney]will see if I can post a patch at the same time
00:15.07bkw_blah
00:15.08bkw_blah
00:15.20Marlowbkw_ : interesting conversation :)
00:15.28bkw_byte me! :)
00:15.29bkw_haha
00:15.42Marlowbkw_ : that's what the dog is for ..
00:16.03Marlow:o)
00:17.01bkw_dogs can't byte
00:18.41denoncypromis: man, openvpn is pretty cool .. I havent touched it yet ..
00:18.44denonbut I think I may
00:19.26nocnocdamn why does my music on hold sounds like jabba the hut snorring?
00:19.36cypromishmm I want to test it tonight on a really strange setup
00:19.36cypromis;;
00:19.55cypromislinux -> linux/nat -> isp/nat -> public server
00:20.03nocnoci've setup the -r parameter for mpg123 to 8000 and 44100 with the same results
00:20.17denoncypromis: you know, openvpn seems lightweight enough for an iax2 payload encryption
00:20.29denonwonder what latency is like
00:20.40cypromisdenon: I used cipe for tht a lot
00:20.51cypromisworked nicely
00:21.00Connornocnoc: You need to use mpg123 not mpg321, check to see if mpg123 is a sym link to mpg321, if so, download mpg123 make and compile it.
00:23.35nocnocconnor - im sure its mpg123
00:23.48Connordouble, nah.. TRIPPLE check it.
00:23.57stoneflyCan the voltage on a PSTN line help you adjust tx gain and rxgain? I have three PSTN and two of them are 49.6v, and one is 43.7v. ..
00:24.08ConnorI had the same problem... do a which mpg123
00:24.20Connorthen check it.
00:24.57ScaredyCatdoes it say it's mpg123 when you run it?
00:25.17daorkfile `which mpg123`
00:25.25daorkand if its a binary
00:25.39pattiejaAnybody know if there will be any merit to running multiple instances of Asterisk on the same box (if this is even possible) and IAX2 trunk them together to find out whether echo cancellation will get better?  My reasoning is that if the software phones do not do echo cancellation (i.e., X-Lite (not sure of), gnophone, DIAX, etc.), then having Asterisk echo cancel twice might just work.
00:25.49ScaredyCat[root@ASTERISK mike]# mpg123 -v
00:25.49ScaredyCatHigh Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
00:25.50ScaredyCatVersion 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
00:25.53daorkmd5sum `which mpg123` and md5sum `which mgp321` and compare
00:27.02nocnocHigh Performance MPEG 1.0 Audio Player for Layer 1,2 and 3. ..
00:27.03nocnocVersion 0.58 (97/04/10). Written and copyrights by Michael Hipp.
00:27.21ScaredyCatget 0.59r
00:27.27*** join/#asterisk forkenb (~forkenb@dailup-219-90-63-226.appscorp.net)
00:27.30nocnocwill do
00:28.04ScaredyCathttp://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
00:28.24kapejodpattieja: why do you think that iax2 trunking is related to EC?
00:28.25*** join/#asterisk One (~DR@D5776FC9.kabel.telenet.be)
00:28.45kapejodeven TDMoE wont help you, because it has no EC support
00:29.00forkenbI'm trying to handle timeout for a queue by putting a 't' extension in the queues context, I also have single digit extensions which work fine when pressed while in a queue, how come a t is not thrown when the queue times out, am I misunderstanding the functionality of timeout parameter in queues.conf
00:30.35cypromisu are
00:30.38nocnocscaredycat - are you using the default parameters within the res_musiconhold.c file?
00:30.41tclarkforkenb: you need a patch for that sse bug.digium.com
00:30.45cypromisbut there is a patch for that somewhere in bugs.digium.com
00:30.50ScaredyCattimeout just means how long b4 it retries
00:30.52atacommgrrr, guess i should bother reading before I try doing........ where can I pull CPU usage statistics out (per cpu)?  i've tried /proc/stat .... but after finding my bar graph staying steady found out thats number of seconds, whereas i'm looking for a realtime, per second type measurement....
00:30.59ScaredyCatnocnoc: yes
00:31.00cypromisor even talk to bkw cause I think it was his patch :))
00:31.25forkenbi c, I'll take a look at it....thanks
00:32.18ScaredyCat^---- not THE One then, obviously
00:35.13nocnocscaredycat - little bit better now. sounds like jurassic park's velociraptors
00:35.20nocnocim getting closer ;)
00:36.09ScaredyCatbbl
00:36.33ScaredyCat; Music on hold class definitions
00:36.33ScaredyCat;
00:36.33ScaredyCat[classes]
00:36.33ScaredyCatdefault => mp3:/home/mp3,-z
00:36.33ScaredyCatloud => mp3:/var/lib/asterisk/mohmp3
00:36.34ScaredyCatrandom => quietmp3:/var/lib/asterisk/mohmp3/ok,-z
00:37.20ScaredyCatnite all
00:37.27*** join/#asterisk Celtic (Dax@rrcs-central-24-106-64-175.biz.rr.com)
00:37.31hellerw
00:37.33hellerergh.
00:38.04znoGhey all, does Asterisk support MSN? (multiple subscriber numbers)
00:38.35kapejodznoG: it does when combined with capi
00:39.06znoGah, good.. so i can do context routing based on the MSN
00:39.23kapejodsure
00:39.32znoGexcellent
00:39.39*** join/#asterisk killerbee (~Killer@ool-44c1013f.dyn.optonline.net)
00:39.49killerbeebkw_
00:39.50znoGwhen you say combined with capi, you mean with the www.junghanns.net?
00:39.54kapejodznoG: even chan_modem_i4l will do this
00:40.19kapejodwith the
00:40.21znoGi'm going to be using a card that has windows capi drivers, but not linux ones .. i'm hoping www.junghanns.net will work with it
00:40.28kapejod!info kapejod
00:40.28kapejodhmm..where is that bot tonight??
00:40.30znoGkapejod: i mean with the channel capi driver from that site
00:40.33kapejodno, it wont.
00:41.10znoGi think i'm a little confused about how it all fits together
00:41.38kapejodyour winblows capi card will not work
00:41.55znoGso i need actual linux capi drivers for the card...
00:41.56bkw_killerbee yes
00:42.11bkw_killerbee I gave up on cracking that ata via war dialing it.. i'm too impatient
00:42.22killerbeeoh :<
00:42.24killerbeetoo bad
00:42.31killerbeejust let it run...
00:43.00kapejodznoG: unless you find a way to run windoze drivers on linux
00:43.37*** join/#asterisk xantus (~david@208.49.241.98)
00:43.57znoGkapejod: eek, not good. without those capi drivers its gonna have serious echo problems and you mentioned MSN won't work either..
00:44.07*** join/#asterisk NoCarrier (~NoCarrier@copper.voicepulse.com)
00:44.08znoGkapejod: i thought that junghanns site had a generic capi driver
00:44.33kapejodno, they dont.
00:44.45bkw_HAHHAHA
00:44.45bkw_From: Steve Totaro <stotaro@seepu.com>
00:44.46bkw_Reply-To: asterisk-users@lists.digium.com
00:44.46bkw_token: asterisk-users@lists.digium.com
00:44.46bkw_Subject: Re: [Asterisk-Users] Radius on *
00:44.46bkw_looks like critchy is especially bitchy....
00:45.03znoGi had a reply setup but i don't really know this Steve guy so i passed on it
00:45.19bkw_With all his whinging, if i didn't know any better, I'd suspect he was using a 2400 baud modem...
00:45.19bkw_Now I'm off to reply a message and change the subject line....
00:45.19bkw_Andy
00:45.28bkw_ScaredyCat you are funny
00:45.48kapejodi deleted all those radius and unsubscribe mails right away
00:46.13JerJerbkw_: read the next one...  Andrew thinks i run VOnage ?!!?!
00:46.35dougheckaLOL
00:46.38xantusdoh
00:47.15kapejodJerJer: you dont run vonage? ;-)
00:47.36JerJeri don't even wish to run vonage.... i wouldn't have gotten in bed with Cisco
00:48.11dougheckafarfone dev is shut down for the day!
00:48.12JerJerI can tell andrew hasn't acutally tried to deploy a carrier-class VoIP system...
00:48.37dougheckacarrier pidgion?
00:49.18denonbkw_: you gave up cracking the ATA?
00:49.20forkenbanybody know of any patches that break into musiconhold to say a message and then return to the music on hold, for example playing promotions randomly while playing a song when a promotion isn't being played?
00:49.28denonbkw_: Why? you could just let it run all week or somethin right?
00:50.34*** part/#asterisk stonefly (~trillian@toby.stoneflytech.com)
00:52.48bkw_denon it caused drama on my box.. haha
00:52.53bkw_I migth try it again.. who knows
00:54.54znoGkapejod: i guess the only card that has full capi support in Linux is Digium's ones right?
00:56.21pinoznoG: i'm quite happy with AVM's cards
00:56.31mishehubah.
00:56.38mishehuI hope my 7960 arrives soon...
00:56.39kapejodznoG: no digium card has capi support
00:56.52znoGkapejod: do you know any that do?
00:57.11mishehuthe only thing you'd really need capi for is faxing.
00:57.23mishehuI can't think of any other reason for capi
00:57.23kapejodznoG: all AVM card, all active Eicon cards, all primuxisdn cards
00:57.37pinoznoG: if you're australian, you may want to check the mailing list archives...
00:57.38kapejodmishehu: how about BRI?
00:57.44znoGkapejod mentioned other features such as MSN (multiple subscriber numbers)
00:57.57znoGpino: thanks
00:58.13pinoznoG: there are not so many cards with the infamous a-tick certification.
00:58.22znoGexcept for AVM right?
00:58.27znoGAVM!Fritz or whatever its called
00:58.41znoGto be honest i couldn't care much for the A-tick.. :) its only going to be used in a SoHo environment
00:58.44znoGas long as it works basically..
00:59.11pinoi have a fritz and i'm very happy with it. (and with kapejod's chan_capi, of course.)
01:00.16CelticAnybody know if either IAX or Zap channels understand the concept of *called number* as different from *calling number* ?
01:00.53cypromisthey do
01:01.55Celticcypromis: If I forward a number to an 800 number that then comes to * over IAX is there any way to pick that info up in extensions.conf ?
01:02.24Celticcypromis: i.e. detect that the call was originally placed to the first number
01:02.58znoGpino: hmm what sort of faxing support does chan_capi add?
01:03.15pinonone. capi drivers may do that, though.
01:03.18mishehukapejod: I didn't remember if digium had any bri cards or not.
01:03.35pinomishehu -- they don't :)
01:03.41tclarkCeltic: see ${RDNIS}        Redirected Dial Number ID Service
01:03.52kapejodsomebody should make some then.
01:04.32*** join/#asterisk One (~DR@D5776FC9.kabel.telenet.be)
01:04.36pinoznoG: for example, with the fritz (and, i guess, most other CAPI 2 aware cards) you can install capi4hylafax and there you go.
01:04.49znoGpino: sorry for all the questions.. just trying to figure out what exactly chan_capi is. Sounds like there are boards with CAPI support, and chan_capi is just a linux driver to use those features
01:04.53Celtictclark:Do you know if Nufone pass such info onto IAX ?
01:05.08znoGpino: ahh so on incoming faxes it just palms it off to hylafax? is capi4hylafax part of the chan_capi package?
01:05.28JerJerwhat's the diff between DNIS and RDNIS?
01:05.29pinono, capi is a standard interface, and capi drivers support the capi interface.
01:05.30Celtictclark: or do you happen to know if the SIP driver puts anything in there ?
01:05.49pinoboth chan_capi and capi4hylafax use the capi interface to talk to ISDN boards.
01:06.04kapejodcapi == Common isdn API
01:06.41znoGi seee
01:06.50znoGso without capi, how do you normally interface with the ISDN board?
01:06.56pinoif i can make it very simple, think of alsa. some sound boards have alsa drivers, and then all alsa-aware software can talk to them.
01:07.30pinomany non CAPI-aware boards can talk to * through the isdn4linux channel.
01:07.33kapejodznoG: some people use isdn4linux
01:07.35pinoi was not happy at all with it, though.
01:07.45*** join/#asterisk bobman (~bobman@me-sebago-cmts1b-15.agstme.adelphia.net)
01:07.59znoGso capi drivers essentially have full control of the card and depending on the features of the capi driver, you can use them.
01:08.31bevinsIs this a permissions issue?
01:08.31pinoznoG: yes! (or at least I think so)
01:08.31bevinsx=0, open writing:  /var/spool/asterisk/voicemail/local/31/INBOX/msg0010 format: wav49, 0x80dbf48
01:08.57znoGpino: so capi4hylafax talks to chan_capi which talks to the ISDN board... right? :)
01:09.14bevinsI get this when voicemail tries to record a message
01:09.14kapejodno
01:09.28CelticJerJer: DNIS is the number called, RDNIS is the number that passed the call on to you - I think...
01:09.30kapejodcapi4hylafax talks to capi4linux
01:09.37kapejodchan_capi also talks to capi4linux
01:10.08znoGi see.. and the difference between capi4linux and chan_capi is? :)
01:10.20pino(your board) <-> (board-specific drivers) <-> (capi4linux driver) <-> (any software)
01:10.28UnixDawgok who here is on fbsd
01:10.32pinowhere any software may be chan_capi or capi4hylafax, in this case.
01:10.36UnixDawgother then me
01:11.18znoGpino: so board-specific drivers have to support capi as well?
01:11.34bevinshas anyone seen this message?  x=0, open writing:  /var/spool/asterisk/voicemail/local/31/INBOX/msg0010 format: wav49, 0x80dbf48
01:11.49UnixDawgI need the 2 lines that go in sysct.conf
01:11.58pinoznoG: if your own board supports CAPI under linux, then you can use CAPI, otherwise you're stuck with either isdn4linux or nothing :)
01:12.28kapejodand you can still run isdn4linux on top of capi4linux (just to complete your confusion)
01:12.41znoGpino: i see. there is this board, NetJet ISDN Card, which has Windows CAPI drivers but not Linux ones... i thought there may be some generic capi driver for capi-enabled cards
01:12.44znoGkapejod: yep, it sure did.. :)
01:12.55kapejodso every capi user can enjoy the latency and echo of chan_modem_i4l :)
01:13.11znoGi've never had ISDN so this is all new to me.. :)
01:13.16*** join/#asterisk cypromis (~michael@217.11.142.161)
01:13.23kapejodwb cypromis
01:13.33cypromisaloha
01:13.37cypromis:)
01:13.54pinoznoG: active boards usually support capi in hardware; some nice passive boards do that in software, like the fritz.
01:14.14kapejodpino: i disagree
01:14.26kapejodthat is not a nice passive board, after all it's binary only!!!
01:14.56pinokapejod: it is :( could we say it's a nice passive board with a tied-down binary capi driver?
01:15.26kapejodwell...there are much nicer passive boards which unfortunately lack a capi driver...
01:15.33pinobasically, it's not nicer than many other boards, but the folks at AVM - for obvious reasons - developed a software CAPI implementation that runs on, well, just their own passive boards.
01:16.08pinokapejod: someone should make them... ;))
01:16.26kapejodthey could look like this: http://62.8.140.152/bri/
01:16.44jsharpBuh
01:17.27pinopfff, kapejod, that's science fiction. ;)
01:17.44kapejodyeah, the gimp can do wonders ;)
01:17.45*** join/#asterisk MagicMan (~alm971@APointe-a-Pitre-101-1-6-172.w81-249.abo.wanadoo.fr)
01:18.48pinoand in such a fictional board, would the dip-switches for NT/TE or something else?
01:19.03pino*be
01:19.34kapejodthe dip switches near the fictional rj45 plugs would be to enable 100 ohm termination
01:19.41kapejodthe other dip switches would be just for fun
01:19.55denoncue the fanfare
01:19.57denon:)
01:20.23pinoi'd only seen those near the plugs... :)
01:20.41kapejodeverybody has his own imagination ;)
01:21.40denonkram: thanks for fixing that stuff in cvs
01:24.00pinogood night everyone close to this time zone :)
01:24.11kapejodn8 pino
01:24.28denonhmm .. first cvsup in weeks ..
01:26.29CelticHohum - 3 IAX feeds and no DNIS or RDNIS apparent on any of them - anybody getting DNIS or RDNIS info over an IAX feed ?
01:31.48Celtic<PROTECTED>
01:32.20Celticor maybe people have beer and don't care :-) - time for my beer methinks
01:32.41*** join/#asterisk Porta (~sockpuppe@69.1.86.32)
01:34.20*** join/#asterisk vefrra (~vefrra@hoochie.digium.com)
01:39.59*** join/#asterisk espen2k (~espen2k@hoochie.digium.com)
01:40.03*** join/#asterisk your_nick (~Ashmed@193.10.185.3)
01:46.34*** join/#asterisk elusive1 (~konversat@19.chicago-12rh15-16rt.il.dial-access.att.net)
01:49.09*** part/#asterisk elusive1 (~konversat@19.chicago-12rh15-16rt.il.dial-access.att.net)
01:51.47denonthe new IAX phone chipset: http://www.soekris.com/  :)
01:54.39ReG-Hexerhuh
01:54.42ReG-Hexernew phone?
01:54.48denonwas kidding
01:55.02denonpretty hefty for a phone
01:55.18bkw_no its nufone
01:55.21kapejoda bit slow for a phone ;)
01:55.23bkw_NUFONE.. learn it
01:55.39ReG-Hexerlol
01:55.47denonbkw_: why didnt you leave your ata186 crackin?
01:56.08bkw_ADD
01:56.13bkw_lost intrest in the project
01:56.17denonbut you coulda just left it under your desk or somethin
01:56.19bkw_was taking too long
01:56.22denonit was already running
01:56.33bkw_no the * server started to flip smooth out after about 8 hours of wardialing
01:56.47bkw_was scrolling the dtmfsend's faster than it was accually dialing them
01:56.51*** join/#asterisk loko_moko (loko-moko@pool-151-201-225-244.pitt.east.verizon.net)
01:56.53bkw_not sure if it was skiping over some or what
01:56.54knight-bkw, lost interest in what project?
01:57.08denonbkw: huh .. dialing too fast?
01:57.12denonsaw your changed the delay
01:57.15denonyour/you
01:57.26bkw_denon yes.. I might pick it back up next week
01:57.46denonsent myself a test message .. havent gotten the email yet :\
01:58.15denonit uses the same confs and stuff right?
01:58.27denondont need to change anything if I was using voicemail1 before do I?
01:58.35bkw_shouldn't
01:58.54knight-bkw, what project?
01:59.25bkw_wardialing a vonage locked ata
01:59.40bkw_would take about 20 days to crack it
01:59.41denonarrrrrrg
01:59.42dougheckawardialing?
01:59.49bkw_go look up wardialing
01:59.52denonmark said he added those patches
01:59.55dougheckayea, but how?
01:59.57denonwtf
02:00.05bkw_doughecka an agi
02:00.07dougheckayou trying to guess the password?
02:00.09bkw_and a sample.callfile
02:00.12bkw_yes
02:00.18dougheckaon the locked vonage thing
02:00.25bkw_yes
02:00.26bkw_works too
02:00.30bkw_tested it
02:00.31dougheckahmm
02:00.41knight-locked ata?
02:00.42dougheckaso how do you access it?
02:01.19dougheckaI have the resources at work that I can dedicate to it
02:01.41denonbkw_: hrm .. http://bugs.digium.com/bug_view_page.php?bug_id=0000542
02:01.54denonbkw_: and mark shows that its fixed in the changelogs
02:02.05dougheckaonly catche is that I get to keep the ATA once I crack it
02:02.09denonbkw_: and yet: Content-Type: audio/x-wav; name="msg0034.gsm"
02:02.13dougheckacatch
02:02.28blllknight-: wow, everyone really does show up here eventually
02:02.39knight-blll :) indeed
02:02.56knight-whats a locked ata?
02:03.20*** join/#asterisk cfo (~cfo@194.19.190.217)
02:03.36dougheckaa cisco ATA that vonage "gives" away for thier service
02:03.44dougheckathey locked it, and this is the only way to crack it
02:03.56bkw_they dont give it away
02:04.00bkw_you pay for it
02:04.04dougheckayea yea
02:04.05doughecka:P)
02:04.09bkw_YOU PAY
02:04.10bkw_trust me
02:04.22knight-ahhhhhh
02:04.24dougheckaso how are you trying to brute force it?
02:04.41dougheckadoes it have a web interface or anything
02:04.51dougheckatelnet? :)
02:04.55denondtmf
02:05.18denonit has an fxs port
02:05.20dougheckaso the ATA listens for a special command?
02:05.28denonwhen you put it into command mode, yes
02:05.31dougheckaa special number
02:05.45denonmore like a big red button
02:05.46blllhow about calling them up and asking them?
02:05.46denonliterally.
02:05.51blllanyone try that? :)
02:06.04dougheckaso I plug a phone into it and dial a special number, and it turns on a command mode?
02:06.31dougheckablll: hmm, give me the number for an internal tech.. I need to learn social engineering anyway
02:07.39denondoughecka: you push a big red button and it asks for a command
02:08.02dougheckaon the ata its self
02:08.12denonyes
02:08.20denonbut you can loop through lots of times with only one button push
02:08.22ReG-Hexeranyone has any documentation on how asterisk transport layer works?
02:08.26denon(to answer your next Q)
02:08.31dougheckahah
02:08.42dougheckais there a limit on how many times before it resets?
02:08.54denonnot that bkw has seen I dont think
02:08.59dougheckainteresting
02:09.09dougheckaI could let my server do it, since its not doing anything right now
02:09.25dougheckahow is bkw knowing that it let him in?
02:09.44dougheckathe server will keep trying numbers wont it?
02:10.00dougheckahah
02:10.48*** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net)
02:11.32*** join/#asterisk dguisinger (~dan@206.230.56.197)
02:11.51dougheckabkw_: well, if you get tired of wardialing the thing, you can send the ATA to me :)
02:12.28dougheckaI can get the thing have you tried, as the password, the number that it was assigned to?
02:12.53dougheckahave you tried using the number that it was assigned to?
02:18.08*** join/#asterisk marrandy (~marrandy@209.216.76.1)
02:20.20marrandyHello.  I am now trying to move my grandstream sip phone from FWD to the asterisk box.  Despite a number of attempts and changes, I am seeing registration failed errors
02:20.50dguisingerany idea why * isn't listening on port 5060? my config file from my eyes looks right.....
02:21.27marrandyIt is on a static IP 192.168.1.70.  The asterisk is on 192.168.1.1. First, does anyone have a good setup for the sip.conf file
02:28.24UnixDawganyone here have the libpri patch for fbsd
02:33.05thololibpri is completely useless until you have zaptel.
02:33.47*** part/#asterisk jmb-ct (~trillian@solo.staff.chagres.net)
02:34.05*** join/#asterisk cfo (~cfo@194.19.190.217)
02:42.12*** join/#asterisk nassy (~nassy@24-193-228-121.nyc.rr.com)
02:44.20UnixDawgok I need the noinvite lines for a grandstream
02:45.20marrandy<PROTECTED>
02:45.20marrandy<PROTECTED>
02:45.21marrandy<PROTECTED>
02:45.21marrandyNOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from '<sip:206@192.168.1.1>' failed for '192.168.1.70'
02:45.43*** join/#asterisk Celtic (~dax@user-0cdv656.cable.mindspring.com)
02:46.16marrandyWhat is this registration failure all about.  I can call and talk between extensions
02:47.55UnixDawgwhat are the disallow lines for sip grandstream phones
02:48.08UnixDawgI know there are 2 set to no but I cant find them
02:50.41*** join/#asterisk Adam_ (~Adam@pseudo.teragen.com.au)
02:50.45atacommummm.....anyone know why my phone company would be giving me a call cannot be completed as dialed message when sending a 10 digit number out a PRI?
02:51.24Stealth_Manatacomm: try to put "1"
02:51.32atacommtried that too
02:51.44Stealth_Manatacomm: you need to ask them how are they expecting to receive your digits
02:51.51Adam_what's a cool name for soft phone software
02:51.52izoatacomm ; check dialplan
02:51.59atacommizo?
02:52.11izoyou can set in pri message if its national or internatioal format
02:52.19atacommStealth: well they cant expect them in too many different formats
02:52.28UnixDawgok who here has a grandstream working
02:52.34atacommit should be national, i've got the PRI set to National ISDN 2....
02:52.36Connormake sure you have the correct switch type.
02:52.36Stealth_Manizo: is also rigth about Numbering type ...
02:52.40UnixDawgI need 2 lines to get mine back working
02:52.47atacommi can receive incoming just fine
02:52.54izo<PROTECTED>
02:52.57izocheck your zapata.conf
02:53.10izopridialplan
02:53.20atacommswitchtype = national
02:53.20atacommsignalling = pri_cpe
02:53.24UnixDawgI need the disallow invite lines
02:53.25Stealth_ManCan someone tell UnixDawg 2 lines ????
02:53.27izopridialplan= ??
02:53.29atacommi dont have pridialplan
02:53.34izoso you need it
02:53.44izotry pridialplan=unknown
02:53.52izoand if it doesnt work
02:53.57atacommlol, what are the pridialplan options?
02:53.58izotry national
02:53.59Stealth_Manizo: do you know settings for GS phones ? 2 lines for allow.disallow ...
02:54.11izoStealth_Man : I have no idea :-(
02:54.13ConnorYou MUST make sure you have same switch type as telco.. I had a problem, they had NI2 and I had NI.. Didn't work.. We changed to DMS100 and it just worked.
02:55.56Stealth_ManANyone has  GS phones settings ???
02:56.01Stealth_Manizo: thanks :)
02:56.19UnixDawgstealth hol on I almost have it
02:56.53Stealth_Manheh .. unixdawg: I am trying my best :))
02:57.22atacommhmm, cant seem to get either of those to work
02:57.55Connorwhat's the telco using ?
02:58.02izoatacom : do you ahve pri debug span 1 set ?
02:58.09UnixDawgi got it
02:58.09UnixDawgthnks
02:58.17atacommi did, its not now after restarting
02:58.42marrandyStealth_Man: you mean on the web setup ?
02:58.48izowell I could almost bet thats the issue with dialing plan and number format
03:00.41*** join/#asterisk UnixDawg (~UnixDawg@ny-lasalle6c-203.buf.adelphia.net)
03:01.04UnixDawgexit
03:02.32marrandyStealth_Man: you still there ?
03:04.12Stealth_Manyes
03:04.36marrandyO.K then - anyone else using SIP phones with static IP
03:05.01Adam_yea...
03:05.07marrandyStealth_Man: so are you talking about the actual GS config ofr the sip config
03:05.24Stealth_ManAdam: can i ask you about config for sip.conf  for GS phone ?
03:05.34Adam_yea
03:05.45UnixDawgok it should be registring
03:05.50UnixDawghmmm
03:06.01Stealth_Manunix: not really ...  i odn;t know hwy
03:06.03Adam_it's pretty simple
03:06.27Adam_actually, very simple :)
03:06.30Stealth_Man[7000]
03:06.30Stealth_Mantype=friend
03:06.30Stealth_Manhost=dynamic
03:06.30Stealth_Manusername=7000
03:06.30Stealth_Mansecret=testsecret
03:06.32Stealth_Mandtmfmode=rfc2833
03:06.33Stealth_Manmailbox=7000
03:06.35Stealth_Mancontext=sip
03:06.37Stealth_Mancallerid="ext. 7000"
03:06.39Stealth_Mannat=yes
03:06.41Stealth_Manhere it is ...
03:06.43Stealth_Manwhat's wrong here ?
03:06.48Adam_what happens?
03:07.07UnixDawgcontext=admin
03:07.07UnixDawgcanreinvite=no
03:07.07UnixDawgreinvite=no
03:07.10UnixDawgadd those 3 lines
03:07.10Stealth_Manactually i have static ip ... maybe i need to remove host=dynamic ...
03:07.16UnixDawgno
03:07.21UnixDawgleave it
03:07.27Stealth_Manunix: context  is differnet :)
03:07.28Adam_what's the problem?
03:07.40Stealth_Mani have GS with this setting and ATA
03:07.42Stealth_Manon same LAN
03:07.44UnixDawgyes but you set a context for each phone
03:08.02Stealth_Manshell I set context for each phone or not ?!
03:08.15Stealth_Mani can;t dial anywhere :) even from 7000 to 7001
03:08.30Stealth_Man7000 is GS ; 7001 is ATA186 v2.16.1
03:08.32Adam_so it registers correctly?
03:09.08Stealth_Man*CLI> sip show peers
03:09.08Stealth_ManName/username    Host                 Mask             Port     Status
03:09.08Stealth_Man7001/7001        192.168.1.10    (D)  255.255.255.255  5060     Unmonitored
03:09.08Stealth_Man7000/7000        192.168.1.103   (D)  255.255.255.255  5060     Unmonitored
03:09.08Stealth_Manfwd/75574        192.246.69.223       255.255.255.255  5060     Unmonitored
03:09.30Adam_looks good (?)
03:09.38Stealth_Manwhat is (D)
03:09.45Adam_dynamic
03:09.51Adam_i think ?
03:09.57Stealth_Manshit ... maybe i need to change it ...
03:10.01Adam_nah nah
03:10.02Stealth_Manlet me try to remove
03:10.03Adam_just leave it
03:10.08Stealth_Manyou think so ?
03:10.11Adam_just leave it
03:10.17Stealth_Mani have static ip ...
03:10.23Stealth_Manbut it shouldn;t be really matter ...
03:10.31Adam_it's a restriction on who can use it
03:10.58Stealth_Man[sip]
03:10.58Stealth_Man;exten => 7000,1,Dial(SIP/7000,20,tr)
03:11.06Stealth_Maninclude => sip
03:11.11Adam_i see the prob
03:11.15Adam_oh no i don't
03:11.24UnixDawgmine works
03:11.24Stealth_Manno no ..
03:11.27Stealth_Manit is not commented
03:11.44Adam_so exactly what's the problem
03:11.49Adam_(i assume you're doing sip debug)
03:11.54Stealth_Manyes
03:12.03Stealth_Manand i can;t figure out what is the prob :)
03:12.09Stealth_Manjust can;t dial ...
03:12.36Stealth_Manit is like ringing ,, but i nreality i  i have no ringing on secon phone ..
03:12.48Stealth_Mani am trying 7000 (GS) to call 7001 (ATA)
03:13.07Stealth_Manonce I had success from 7001 to 7000
03:13.12Stealth_Manbut not vice versa..
03:13.13Adam_did u try unixdawg params?
03:13.31Stealth_Manyeh... same
03:13.45UnixDawgstealth you need to go back into the phones
03:13.52UnixDawgand reset them up
03:14.08UnixDawgI have a feeling its in the phone where you did not get it all setup
03:14.25Stealth_Manhmmmm
03:14.28Stealth_Manmaybe ...
03:14.31Stealth_Manbut what ?!
03:14.47Stealth_Manyou mean reset config ?
03:14.57Stealth_Manandti start from the scratch ?
03:15.01UnixDawggive me a min I have mine alsmost done
03:15.12Stealth_Manok ok
03:15.36znoGanyone using a grandmaster budgetone?
03:15.55UnixDawgthats what we are setting up now
03:16.00daorkyou mean a grandstream?
03:16.08znoGoops, yep, grandstream
03:16.13daorkim about to buy some.. are they good?
03:16.18znoGthey look good at $75/$85
03:16.23znoGi'd like to know if they're any good :)
03:16.35UnixDawgthey are ok for starter phones
03:16.35daorkznoG: more like 65ish
03:16.49UnixDawgbut whenyou need more power you will have to get a better phone
03:16.56znoGi got an email from @grandstream.com that they are $75/$85
03:17.04daorkznoG: chagres.net
03:17.08daorkand many other resellers
03:17.10znoGUnixDawg: what sort of things can't it do? it seems to do call conference, caller id, call transfer, etc
03:18.02UnixDawgnot all the buttons work it seems the mute does not work send does not work the confrance button I have not got working
03:18.09Stealth_Manznog: i can sell you GS
03:18.12UnixDawgI got the msg button working
03:18.14Stealth_Mantogether with voice services ///
03:18.15Stealth_Man:)
03:18.35znoGStealth_Man: $$ ?
03:18.41znoGdaork: thanks for that, saved $10 already.. :)
03:18.54znoGUnixDawg: eek, thats not so good
03:18.55UnixDawgthey are a good basic phone
03:18.55UnixDawgfor learning with
03:19.27znoGwell as long as caller id, call transfer, call hold, and call conference and hmmm speakerphone work.. im happy
03:19.54UnixDawgspeaker phone works I have not tried confrance yet
03:20.33*** join/#asterisk Gary (Gary@218.19.158.239)
03:20.53Stealth_Manspeaker phone works ...
03:21.26znoGUnixDawg: please let me know how you go with th ephone.. very interested in the feedback
03:21.43UnixDawgok will do
03:21.55UnixDawg<PROTECTED>
03:21.56Stealth_ManznoG: GS works good for sure ... a lot of people are using it ..
03:22.06UnixDawgI am writing my extensions.conf now
03:22.22Stealth_Manznog: you can take a look for some Sipura-2000 adapter maybe ...
03:25.08Stealth_ManCapabilities: us - 524302, them - 281/0, combined - 8
03:25.08Stealth_ManNon-codec capabilities: us - 1, them - 0, combined - 0
03:25.08Stealth_ManLooking for 7001 in sip
03:25.08Stealth_ManTransmitting (NAT):
03:25.08Stealth_ManSIP/2.0 404 Not Found
03:25.17Stealth_Manwhy is that ?
03:26.02Adam_cause the name of the ata isn't really 7001
03:26.28znoGare there any large sites with GS deployed on every desk?
03:26.35Stealth_ManAdam ???
03:27.00Stealth_ManznoG: possible ... but large sites are usually with Cisco phones ...
03:27.19Stealth_ManAdam: what do you mean by ATA name not 7001 ???
03:27.28znoGah okay
03:27.36znoGStealth_Man: how much is the Sipura adapater?
03:27.44Stealth_Man110$
03:27.47znoGouchie
03:27.49Stealth_Mandual port
03:27.51znoGi'll stick to the GS :)
03:28.09Stealth_Man2 FXS port ...
03:28.22Stealth_Manbut you need actual phones to get connecteed
03:28.30Corydon76-homeWow, anybody else notice how _polite_ the Asterisk lists are as of late?
03:28.51Stealth_ManCoryd: yepp ... just very polite and nice talking .. hah ?:))
03:29.04Corydon76-homeIt's the very model of civility...
03:30.41Stealth_ManAdam: what you said about name of ATA ?
03:31.24znoGStealth_Man: so all the features of the GS as advertised should work, right?
03:31.49Stealth_Manznog: HARDWARE PHONE !!!
03:31.50znoGit'll probably become troublesome to have my phone on my laptop :)
03:31.52Stealth_Manit is much better
03:32.03znoGreliability wise or because its a pain in the ass?
03:32.11Stealth_Manznog: I was travelling with Cisco ATA no prob:))
03:32.17znoGnice :)
03:32.23Stealth_ManZnog: for this money you cna try it :)
03:32.29znoGhow much are the ATA's worth anyway?
03:32.30Stealth_Manmany many people are using it
03:32.36znoGusing what? the Cisco ATA?
03:32.41Stealth_Manznog : around 130-150 depends
03:32.42znoGoh the GS
03:32.54Stealth_ManGS are chepest phones
03:33.00znoGyeah i'm gonna buy one to test anyway.. they look good
03:33.09Stealth_Manso you meed to make decision ... to spent 100$ let's  and to check it ...
03:33.11znoGhave you tried any phones from www.sipphone.com ?
03:33.19Stealth_Manznog: same GS phones
03:33.23Stealth_Manas you see on picture
03:33.23znoGhehe okay
03:33.32*** join/#asterisk Mike-69 (~mike@dsl-200-67-40-148.prodigy.net.mx)
03:34.17Stealth_Manznog: get phone, I will get you  some free  Long distance minutes with it ;-)
03:34.50znoGuhhh :)
03:35.01znoGhere's a stupid question - why would you need dual ethernet on a phone?
03:35.07Stealth_Mansimple
03:35.16Stealth_ManGS has 2 versions of phones ...
03:35.26Stealth_Man1 with 1 eth and another with 2 ethr ports
03:35.35Stealth_Manznog : have you seen modem any time ?
03:35.41Stealth_Maninternal modem ...
03:35.43Stealth_Manfor PC
03:35.48Stealth_Mannot for notebook ...
03:35.59Stealth_Manit has 2 ports too rigth ?
03:36.08marrandyStealth_Man: do you have it registering yet ?
03:36.31Stealth_Manmarrandy: i am wlroking on config file .. they are registered , i have jsut mess in extensions.conf
03:36.37znoGStealth_Man: yeah but whats the use of the 2nd ethernet?
03:36.40UnixDawgchan_aix right
03:36.46Stealth_Manznog : same as a telephone port on modem ...
03:36.48marrandycomputer
03:36.53znoGoh ok gotcha
03:37.04marrandyone for the sip phone, the other for the computer
03:37.14Stealth_Man:)
03:37.25znoGhttp://www.grandstream.com/images/BudgeTone.jpg  << they are all budgetone 100 phones but they look different and some have more buttons than others - whats the diff?
03:37.28Stealth_ManLAN connection goes into phone and from phone into PC ...
03:37.34znoGyep i see
03:37.43Stealth_Manznog : you need simple phone .. common stop thinking
03:37.50Stealth_ManBT-100 one thernet
03:38.06UnixDawgok iax is working
03:38.36marrandyStealth_Man: so what is your sip.conf file as you have a static address.
03:39.12UnixDawgin modules.con how do you turn off the iax codec
03:39.15Stealth_Manmarrandy:check email ...
03:39.26UnixDawgnoload=chan_iax right
03:39.55znoGStealth_Man: but does the BT-100 have the hold/transfer/conference features? the second phone in BudgeTone.jpg has the mute feature too. I want that one.. :)
03:40.20UnixDawgok got it
03:40.32UnixDawgman once you learn it comes quick
03:41.41Stealth_Manznog: ye it does  ...  buttons :)
03:43.55znoGStealth_Man: last question, do you know why only the right 2 phones have the mute button in that BudgeTone.jpg pic? they are all BT-100's
03:44.05UnixDawgwell I am almost done with extensions
03:44.06UnixDawg.conf
03:44.08UnixDawgthen to reload and test
03:44.12UnixDawghell all my outbound are gone
03:44.17UnixDawggrr ok this will be fun
03:44.27znoGgood luck UnixDawg .. let us know how you go :)
03:46.34UnixDawgok
03:47.13Stealth_Manunixdawg: yep ... let's make it working ...
03:48.12*** join/#asterisk yifang (~yifang@ip68-9-77-241.ri.ri.cox.net)
03:48.27*** join/#asterisk Rave (~chatzilla@65.123.139.62)
03:48.29znoGso if you want a hardcore phone, you get a SNOM 105 or a Cisco.. right?
03:48.29*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
03:48.43*** join/#asterisk battlej (~battlej@66-65-184-236.nyc.rr.com)
03:49.17Stealth_Manshit
03:49.24Stealth_Manznog: GS too :)
03:49.25Stealth_ManCapabilities: us - 524302, them - 281/0, combined - 8
03:49.25Stealth_ManNon-codec capabilities: us - 1, them - 0, combined - 0
03:49.25Stealth_ManLooking for 7001 in sip
03:49.25Stealth_ManTransmitting (NAT):
03:49.25Stealth_ManSIP/2.0 404 Not Found
03:49.27Stealth_ManVia: SIP/2.0/UDP 192.168.1.103;received=192.168.1.103
03:49.29Stealth_ManFrom: <sip:7000@192.168.1.99>;tag=e9510380-b785-aab9-79b9-608cce612bf1
03:49.34Stealth_Mananyone nay ideas why am i getting this ?
03:52.43*** join/#asterisk frollo (~frollo@d81-211-254-242.cust.tele2.it)
03:52.53frolloHi guys
03:53.12frolloSomeone of you is testing the x.lite phone?
03:54.02frollook no one :)
03:54.40battlejok, frollo, I'll bite. I've played with it some.
03:54.50znoGi have too
03:55.08frollook, did you experienced some issues in sending the mic audio?
03:55.37battlejI've used it on pocketpc and mac platforms, and found it... lacking.
03:55.49frolloeverything is working perfect the only issue is sending mic audio on SIP
03:56.01frollook
03:56.01battlejI'm guessing they codecs haven't really been optimized for those platforms...
03:56.21battlejbut it's just a guess. and I'm not using particularly intense codecs...
03:56.33znoGfrollo: i had that problem, and i did disallow=all, allow=gsm .. and it worked
03:56.34frolloi tested it on windows  and is fine even if i call out trough the fxo of asterisk
03:57.01frolloi'll try just now :)
03:57.04frollolet's see
04:01.57frolloznoG: I LOVE YOU :) IT WORKS!!!!!!
04:02.09Stealth_Manheh
04:02.19battlejexcellent!
04:02.31znoG:-)
04:03.24frolloi was looking for a day in the internet in google asterisk and so on but everyone was saying about the default interface :/
04:03.57frollothe issue was not there, it is strange that there is, no info, nowhere :/
04:04.25znoGyeah, it is strange that the default codecs don't work
04:04.44frollogsm is better i think
04:05.06frolloi heard that g711u takes 20k more bandwidth
04:05.29Stealth_Manfrollo: g711  compare to Gsm ... is big different
04:05.31wasimfrollo: 20k more than what?
04:05.38Stealth_Manwasim: good morning
04:05.39Stealth_Man:))
04:05.41frollomore than gsm
04:05.44Stealth_Manyou are not sleeping ?
04:05.52wasimmorning Stealth_Man, just wakey wakey
04:05.52Stealth_Mang711 =64kbps + overhead
04:06.11Stealth_Manwasim: i have impression that you are not sleeping at all and only working working worling ...
04:06.29frollowhat's the time there? here is 5 am an in a few our i should go to work without sleeping :/
04:06.46wasimits just after 9 am
04:07.12MikeStealth_Man: why is it a big difference?
04:07.30frolloyesterday night I was fighting with a SAP DB upgrade at work and I had to rollback the upgrade (shitty NT4)
04:07.59Stealth_Manfrollo: hehe
04:08.00Stealth_Man:)
04:08.45frolloanyway now asterisk is making me happy now :) Is possible conferencing on asterisk only using SIP phones ?
04:09.19Stealth_Manfrollo : why only sip phones ...
04:09.42frolloI mean is it possible to think an extension as a meeting room for SIP phones
04:10.02frollo?
04:10.08MikeStealth_Man: are you ignoring me?
04:10.18Stealth_ManMike ?
04:10.26Mike<Mike> Stealth_Man: why is it a big difference?
04:11.29Stealth_ManMike: why not ?! G711 is using 64 kbps
04:11.39Stealth_Manmore then twice as a gsm ...
04:12.20Stealth_ManMike: your GS phone
04:12.24Stealth_Manis it working ?
04:12.25*** join/#asterisk sack (~sack@polar.es3.egwn.net)
04:13.14Stealth_ManMike: are you igniring me ?
04:14.04*** join/#asterisk idiot (~asterisk@ip-wv-68-119-143-145.charterwv.net)
04:14.42idiotok.. as my nick says, i'm an idiot :) having two problems, both would be small if i weren't so stupid :)
04:14.53znoGyou idiot, go ahead
04:14.54MikeStealth_Man: sorry phone
04:15.06Stealth_ManMike: :)) no problem
04:15.09MikeStealth_Man: yes my GS phone is working
04:15.12idiotNov 17 23:03:49 NOTICE[139478016]: File chan_sip.c, Line 5074 (handle_request): Registration from '<sip:carolyn@10.0.0.1>' failed for '10.0.0.2'
04:15.14MikeStealth_Man: why?
04:15.27idiotznoG was typing :)
04:15.43Stealth_ManMike:you have one GS behind NAT correct ?
04:15.53Mikewell my asterisk server
04:15.55Stealth_Mancan you put here your sip.conf and extenssions.conf ?
04:15.56idiotanyone got an example for what to put in sip.conf for incoming connections? :)
04:15.58Mikehas the pppoe conextion
04:16.02*** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp)
04:16.06Mikei conect my GS directly to the internal ip address
04:16.25Stealth_ManMike: ok
04:16.31Stealth_Manpost your config please ...
04:16.32idiotany way to get better error message as for what is causing failure?
04:16.40Stealth_Manfor sip.conf part and extensions.conf
04:16.45frolloidiot: http://www.onlamp.com/lpt/a/3956
04:16.49MikeStealth_Man: i got the on the web
04:16.57Stealth_Manok what os the address ?
04:17.21idiotfwiw, carolyn is just a softphone (sjphone) on a laptop
04:18.12UnixDawgok who has the libpri patch for fbsd
04:18.16UnixDawgI need it
04:19.26znoGso, with a mic/speakers headset and X-Lite, is it really that bad to use X-Lite for everyday phone use?
04:19.32znoGif so, whats so bad about it?
04:19.42idiotUnixDawg all you have to do is comment out the ldconfig invokation in makefile, and replace references to endian.h with machine/endian.h
04:19.46znoG(replace x-lite with <any software phone>)
04:19.51idiotznoG I use it daily, works great
04:20.00znoGyeah, but you're an idiot
04:20.03znoGonly joking :)
04:20.07frolloznoG: x-lite is good and it's free :)
04:20.32znoGi'm getting ISDN put in soon.. just wondering if we can survive with X-Lite as phones for a while until we get some grandstream phones in
04:20.36UnixDawgok
04:20.37frollosjphone does not look nice
04:21.13*** join/#asterisk _gorman (~lehmann@pD9E4E223.dip.t-dialin.net)
04:21.17frollowhy buy expensive phones when everyone has a pc :)
04:21.31Mikefrollo: because...
04:21.45Mikefrollo: pcs have bad sound cards or different setting for sound cards
04:21.51Mikefrollo: eco with speakers and mic
04:22.05znoGfrollo: i'll be using X-Lite for everything, not just calling PCs
04:22.05Mikefrollo: you need a compiuter on 24/7 to be accesible
04:22.06frollook the old foggy office assistant does not know PCs :)
04:22.21Mikefrollo: and you dont always listen when people calls you
04:22.36Mikefrollo: using a hardware phone is a standalone system no need for a computer
04:22.44Mikefrollo: hardware works 100% and really good
04:23.04Mikefrollo: you just need a small router in most cases that you wont even notice if its on all day and all night
04:23.05frolloyah it's true :) but may be because i'm used to work in a call center :)
04:23.12Mikefrollo: your accessible 24/7 with your phone
04:24.10frollowe don't have anymore phones :) only VoIP and software everywhere no hardphones :)
04:24.21znoGfrollo: works well?
04:24.40idiotbrb, hacking on this
04:24.44UnixDawgmake: don't know how to make pri.lo. Stop
04:24.57Mikefrollo: that mean you have to be infront your computer to recibe a call
04:25.12Mikefrollo: if your not infront it?? if your sleeping or if your eating
04:25.13znoGso? you have to be in front of the phone to recieve a call :)
04:25.14frollowell of course but in the call center we have AVAYA Definity = million dollars ACD cluster :)
04:25.16UnixDawgnight boys
04:25.21znoGif im sleeping or eating, i dont want to take any calls
04:25.43frolloahahaha znoG :)
04:25.56frolloMike the speakers!!!!!
04:26.14UnixDawgok got it
04:26.15frolloMike the speakers!!!!! They ring when telephone rings!!!
04:26.17UnixDawgcls
04:26.30znoGproblem is i'll have to unplug my headset so i can hear the calls
04:26.38Mikefrollo: eco problem
04:26.45znoGMike: not if you use a headset
04:26.53MikeznoG: hes talking about speakers
04:27.00MikeznoG: and most headset still makes eco
04:27.25znoGi was talking with my headset here and it didnt have the echo problem
04:27.27MikeznoG: good headsets like futball headset motorola are about 200usd
04:27.36MikeznoG: buy a 65dlls hardware phone that it
04:28.42frollook Mike you are right
04:29.01frolloyou convinced me and mi bos too about money :)
04:29.06frolloboss
04:30.04idiotstill not working, znoG
04:30.41idiotREGISTER sip:10.0.0.1 SIP/2.0
04:30.55idiotFrom: <sip:1000@10.0.0.1>;tag=36940285
04:30.59idiotthat looks wrong for some reason?
04:31.16idiotasterisk box is 10.0.0.1, carolyn is 10.0.0.2 presently
04:40.04frollo:) Conferences also work using SIP Phones :) I love ASTERISK :)
04:40.11*** join/#asterisk chayewala (chayewala@adsl-66-122-206-141.dsl.sntc01.pacbell.net)
04:40.17chayewalahello
04:40.56frollook guys. thanks for your precious help. i'll have to go to sleep a few hours before going back to work. Ciao!
04:41.52chayewalaHello Jerjer
04:42.11chayewalais stealthman there.
04:42.31chayewalaI finally was able to setup my grandstream phone.
04:43.20znoGMike: yeah, we're probably going to end up with a hardware phone
04:43.26chayewalahe he h
04:43.44chayewalathere are few nice one now within range of ordinary folks like us.
04:44.11znoGyeah, the grandstreams look ok
04:44.28JerJerbarbietone
04:44.37MikeznoG: 65dlls for a 100% compatible phone
04:44.41MikeznoG: its a good deal
04:47.15chayewalahe he he
04:47.18chayewalaJerjer hello
04:47.48bkw_7960s ROCK!
04:48.33chayewalaJerjer - can I make multiple calls from chan_H323 code. I am haveing a lot of success with your code.
04:48.54bkw_chayewala yes you sure can
04:49.14chayewalaJerjer: What will it take to make your chan_h323 code a gateway instead of an endpoint only.
04:49.21chayewalankw: Have you tried it?
04:49.28chayewalabkw
04:49.30bkw_kill h323 NOW
04:49.33bkw_dont use it
04:49.36chayewalawhy?
04:49.47chayewalaI am routing all h323 calls to *
04:49.57chayewalaa lot of gateways only use H323
04:50.05chayewalaso be considerate.
04:50.08chayewalahe hehe
04:50.21bkw_inbound to * via h323
04:50.25Farooqyes
04:50.38Farooqthose calls we get on H323, I rout them to *.
04:50.44Farooqand then viola
04:50.48Farooqvoila
04:50.55bkw_viola?
04:50.56bkw_haha
04:50.56FarooqJerjer's code is good
04:51.03Farooqhaha
04:51.06bkw_good... DAMN GOOD!
04:51.23Farooqyes but why is it an end_point only.
04:51.34bkw_its not.. it will receive and make calls
04:51.39Farooqyes
04:51.45bkw_you just set a context and it works
04:51.45Farooqwill it make multiple calls ?
04:52.01Farooqhow does it know how many channels.
04:52.08Farooqcan I limit channels?
04:52.19Farooqjerjer?
04:52.28bkw_no
04:52.39Farooqno what?
04:52.47Farooqwill not make multiple calls to and form h323?
04:52.53bkw_I don't think you can limit
04:52.56Farooqok
04:52.58bkw_you can make as many as the box will take
04:53.06Farooqok
04:53.10Farooqand another question.
04:53.29FarooqI am trying to setup a simple generic office setup.
04:53.32Farooqwith voice mail
04:53.44bkw_pay me and I will h00k you up
04:53.49Farooqlike respond with a greeting and etc etc
04:53.51Farooqhe hehe
04:53.54Farooqwith what?
04:54.02Farooqyou mean with jerjer?
04:54.18Farooqalas! jerjer is a guy and me too :-)
04:54.18Farooqhehehe
04:55.00Farooqcan someone provide me some good voice mail + generic office setup
04:55.21FarooqI mean configuratin files
04:55.31Mikebkw_: for some reason asterisk and my gs doesnt like
04:55.35Mikeg723
04:55.44Mikeg729
04:55.45Mikei mean
04:55.49Farooqyeah
04:56.04Farooqcan someone develope g723.1 support using Intell IPP.
04:56.11bkw_nope
04:56.14bkw_still have to lic. it
04:56.15Farooqit is freely available for development
04:56.24Farooqwhy not?
04:56.38Farooqthen who ever wants to use it can pay for it :-)
04:57.04idiothmm
04:58.10FarooqI downloaded IPP and was able to compile the code and save voice files in g723.1
04:58.27Farooqbut I dont know how to add that support to Asterisk
05:00.05bkw_give up now.. you will have to buy the rights to use it
05:00.26Mikeg729?
05:00.27Farooqyes but not to develop it - right?
05:00.39Farooqg723.1 - i am talking about.
05:01.22FarooqCan any one assign 1700 and 1777 numbers or does it require some sort of permission ?
05:04.28atacommbkw: when i enable a macro that is supposed to fix some caller id quirks (i found it else where), Asterisk hangs up the call.....can you look it over quickly?
05:04.30idiotanyone got * working w/ fwd?
05:05.56atacommFarooq: everyone that does it does it internally only.  The PSTN does not recognize them
05:08.34h3x700 is reserved for the IXC's use.  typically 700-555-4141 tells you which ld carrier you have selected as your PIC
05:09.08*** join/#asterisk maharzan (~arun@202.51.76.140)
05:09.18maharzanhi
05:09.24maharzani have a small problem
05:09.37maharzani am currently using netgear router
05:09.45h3x777 is assigned by nanpa as "easily recognizable code"
05:09.51maharzanwhich uses public IP
05:10.12maharzanin my network i.e 192.168.0.x , i have set up my *
05:10.19*** join/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net)
05:10.30maharzani cannot call to US using IP with that configuration?
05:10.37maharzanis it something to do with NAT?
05:10.46maharzani think netgear uses NAT
05:11.09idiotusing SIP?
05:11.16maharzanIAX
05:11.26maharzanusing voicepulse account
05:11.49idioti  dunno about iax but for sip you must forward your RTP ports
05:12.05maharzanok
05:12.11maharzanhow can this be done?
05:12.23maharzanit should be similar
05:12.27*** join/#asterisk lexluther (~admin@80.88.142.221)
05:12.55*** part/#asterisk lexluther (~admin@80.88.142.221)
05:13.21*** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net)
05:14.14mishehua ratings engine?
05:14.18bkw_rating
05:14.24mishehurating what?
05:14.32bkw_calls
05:14.45bkw_i'm sure as hell not talking about movies!
05:14.50idiotmaharzan check your rtp.conf
05:14.57Farooqthanks atacomm
05:15.04idiotand forward that port range to the asterisk box's ip
05:15.05Farooqyou are very knowledgable
05:15.12mishehubkw_: are you recording the conversations to rate them?  ;-)
05:15.20Farooqhow about 1 777 numbers?
05:15.27idiotAnyone using free world dialup with *? :)
05:15.34Farooqhe he he
05:15.37Connorwere do you get the channel h323 code at?
05:15.49Farooqit is part of asterisk
05:15.52bkw_mishehu um no
05:15.56Farooqyou just need to compile it
05:16.02Farooqconnor:
05:16.24bkw_JerJer gots a question
05:16.26Farooqfor that you need to get pwlib and h323 from www.openh323.org
05:17.04JerJeryep
05:17.08Farooqinstructions are in a file in channels directory about h323
05:17.20bkw_haha
05:17.36ConnorHow well does it work with Cisco Call Manager?
05:17.45Farooqbkw : have you tried multiple h323 channels with *
05:17.46Farooq?
05:17.52bkw_Farooq daily
05:17.58Farooqok
05:18.22Farooqcan you provide your configuration files? You can remove passwords etc :-)
05:18.33FarooqI can only use one channel.
05:19.02Farooqwhat type of work do you do bkw?
05:20.02Farooqatacom?
05:20.18decodebkw is an evil hacker
05:20.45decodebkw_ isn't that right?
05:20.52bkw_um no
05:20.54bkw_[general]
05:20.54bkw_port = 1720
05:20.54bkw_bindaddr = 0.0.0.0
05:20.54bkw_tos=lowdelay
05:20.54bkw_allow=all
05:20.55bkw_noSilenceSuppression = yes
05:20.57bkw_dtmfmode=inband
05:20.59bkw_context=autoattend
05:21.01bkw_that my h323.conf
05:21.03bkw_PURE AND FUCKING SIMPLE
05:23.04decodebkw_ sire, what is 'black friday' prices?
05:23.11decodei like the 5$ ups
05:23.24Farooqwell how do you transfer the calls to sip devices?
05:23.26*** part/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net)
05:23.31FarooqBKW?
05:23.38bkw_decode its what bestbuy used the DMCA to quash from fatwallet.com
05:23.41decodeFarooq Call(SIP/port)
05:23.49decodebkw_ Huh?
05:23.53decodewas that bs prices?
05:23.54bkw_I would like to see them try to make me take them down.. because they better have a gun to my head and kill be first.
05:23.57decode5$ for the ups?
05:24.01bkw_haha
05:24.09decodei am confused
05:24.15bkw_not sure.. I just posted the full list from the slashdot links
05:25.50decodeahh
05:25.58decodewas wondering if i have to take 500$ to bestbuy ;)
05:27.44bkw_decode i'm just being evil..
05:27.55decodebkw heh
05:28.07Connorwhere can I find the versons of Open h323 and pwlib that's needed for h323 ?
05:28.08decodebkw happen to use free world dialup w/ *? :)
05:28.41bkw_decode mine works
05:28.43bkw_why?
05:31.48bkw_now what was my fwd number
05:32.01bkw_FWDUSERID=51991
05:33.04decodebecause i cant get it to work on incoming calls? :)
05:33.46bkw_behind nat?
05:33.52decodeyea, sadly
05:34.00bkw_good fuckin luck! god be with ya
05:34.04decodecan punch holes as needed
05:34.05atacommlol
05:34.17bkw_atacomm you got 500 bucks
05:34.32atacomm? i had to borrow money today so i could go park, why?
05:34.40bkw_haha
05:34.42bkw_just askin
05:35.36decodeheh.. i think i'm going to be unemployed soon..
05:35.44decodeone of my gf's came to where i work
05:35.58decodeand my boss made the mistake of calling her a "tasty treat" so i sliced his tires.. heh.. heh..
05:38.01*** join/#asterisk maharzan (~arun@202.51.76.140)
05:40.35tessier__decode: Glad you don't work for me. You're a nut.
05:40.49tessier__"one" of your gf's?
05:41.01decodetessier__ my gf has a gf :)
05:41.08tessier__Sure.
05:41.09decodeand therefore her gf is my gf too
05:43.00Mikeg729 doesnt work!!
05:43.02Mikegrrr
05:43.14Mikecan dial 1800 when i call it rings when they pick up the call gets hang up
05:43.17Mikewhat can i do?
05:45.00*** join/#asterisk Pids (~pids@adsl-67-121-190-78.dsl.sntc01.pacbell.net)
05:45.13*** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p217.telkom-ipnet.co.za)
05:49.45*** join/#asterisk danielq (~danielq@nat154.gw1.bne.webcentral.com.au)
05:52.56danielqanyone awake?
05:53.32Farooqpin drop silence
05:53.44Farooqsprint rules :-)
05:56.10*** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net)
05:56.10*** topic/#asterisk is Asterisk: The Linux of Telephony
05:56.14bkw_blah
05:56.30wasimblah timmah
05:56.56wasimwhat's a timmah, bkw_ ?
05:57.38Connorhey, bkw... In that new vocemail enhancment patch..
05:57.50danielqhaving fun and games getting chan_oh323 to compile
05:57.50*** join/#asterisk cfo (~cfo@194.19.190.217)
05:58.12danielqanyone played with it lately?
05:58.18sizzzungyo momma
05:58.23sizzzungshe likes my TESTICLES.
05:58.26sizzzungaklsjfklajdf
05:58.29Connorwhy the heck did they go overboard with the cmd = ast_waitfordigit(chan, 600) between each and every prompt ?
05:58.32sizzzungHLAGHLAGHLAG
05:58.42ReG-Hexerhi
05:58.42bkw_who is they
05:58.47ReG-Hexeri have a question
05:58.51ConnorWho ever coded the patch. :)
05:58.54sizzzungReG-Hexer: no i am not gay
05:58.57sizzzungReG-Hexer: but blll is
06:09.54*** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net)
06:09.54*** topic/#asterisk is Asterisk: The Linux of Telephony
06:09.56cmani cannot call with * to US using IAX voicepulse
06:10.32cman<PROTECTED>
06:10.33cman<PROTECTED>
06:10.33cmanWARNING[7176]: File chan_iax2.c, Line 1124 (attempt_transmit): Max retries exceeded to host 66.234.228.132 on IAX2[voicepulse]/1 (type = 6, subclass = 1, ts=1, seqno=0)
06:10.33cman<PROTECTED>
06:10.33cman<PROTECTED>
06:11.27cmanmy config files are alrite
06:12.27Farooqwhat is your phone number cman?
06:12.28cmani am using netgear router which has an IP of 202.xx.xx.xxx
06:12.47cmani am in nepal..
06:12.58FarooqOK
06:13.02cmanyou cannot call from there as i have no account
06:13.05Farooqwhat is your voicepulse number?
06:13.10cmani can only mak calle to US
06:13.11cmannot incoming
06:13.17Farooqhmm
06:13.27Farooqwhat type of device do you have?
06:13.31cmani am trying to set up * to call to US
06:13.33FarooqSIP phone?
06:13.39FarooqOK
06:13.49cmani am using X101P and TDM400P
06:13.54Farooqok
06:13.57Farooqwll
06:14.01Farooqwell
06:14.07cmani have normal phone
06:14.15FarooqOK
06:14.21FarooqI dont really know then
06:14.23cmani can make internal  calls, local calls but not to US
15:31.11*** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net)
15:31.11*** topic/#asterisk is Asterisk: The Linux of Telephony
15:32.38opsysPj_ I ran gdb core file then bt and got 'no stack' Any Ideas??
15:33.20RoyKAres: apt-get install icecast-client
15:33.25RoyK...or something
15:33.50Areswhy install a new player
15:33.56RoyKperhaps
15:34.00RoyK*checking*
15:34.16Areslet check
15:34.39Aresbut icecast should normally work well with shoutcast
15:35.02*** join/#asterisk dutch_ (~dutch@a80-126-102-2.adsl.xs4all.nl)
15:35.18AresCouldn't find package icecast-client
15:35.26RoyKdon't mind
15:35.37RoyKit was to feed mpeg streams to an icecast server
15:36.11*** part/#asterisk kapejod (~kapejod@pD9E81940.dip.t-dialin.net)
15:36.28Areswell, do you think possible to use an other player than mpg123 with asterisk
15:36.43*** join/#asterisk Powerkill (~powerkill@195.68.105.195)
15:37.06*** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM]
15:37.06*** mode/#asterisk [+bb *!*@og.latency.net *!*dan@194.158.*.*] by simak.freenode.net
15:37.06*** mode/#asterisk [+q sant!*@*] by simak.freenode.net
15:37.38PowerkillRoyK are you here ?
15:37.52*** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-131-106.NYCMNY83.dynamic.covad.net)
15:37.52RoyKPowerkill: sure :)
15:38.03RoyKPowerkill: I found the entries in a header on a redhat box
15:38.10Powerkilltiff.h:#define     COMPRESSION_CCITT_T4        3       /* CCITT T.4 (TIFF 6 name) */
15:38.10Powerkilltiff.h:#define     COMPRESSION_CCITT_T6        4       /* CCITT T.6 (TIFF 6 name) */
15:38.14RoyKso I just copied the entries over to my header file
15:38.27Powerkillok so it's working now ?
15:38.39RoyKwell - compiled, at least
15:38.48RoyKbut I haven't sent any fax yet
15:39.27*** join/#asterisk cc-reel (~cc-reel@hoochie.digium.com)
15:40.57*** join/#asterisk sjoep803 (~sjoerd@213-84-218-42.adsl.xs4all.nl)
15:41.58cc-reelHas anyone had trouble with GrandStream Budge Tone 100 phones with the new versions of Asterisk?
15:42.25bkw_cc-reel its the barbietones
15:42.27bkw_its not *
15:42.33bkw_barbietones SUCKK
15:42.37bkw_with a double K
15:42.38bkw_:P
15:43.19cc-reelWhat is the cheapest IP phones that you would give a good review?
15:43.42cypromischeap + good review = non existant
15:44.02cc-reelhahaha .... Ok, So what phones do you like?
15:44.16cypromiscicsco 7960
15:44.16cypromisor
15:44.24cypromisfarfon when it will be ready
15:44.52cc-reelWe have had good luck with the Snom 200's.
15:45.42Powerkilli try to fax from tx to rx and it's doesn't work
15:46.35*** join/#asterisk jamie_ (~jamie@softmodem.org)
15:47.24RoyKAres: still here?
15:47.37RoyKbkw_: what's so bad about them?
15:47.39Aresyes
15:48.06AresI was ready to hang  my-self
15:48.08RoyKAres: that jazz station works fine with MP3Player()
15:48.41AresRoyK: that doesn't work
15:48.57Aresprobably because that doesn't work with mpg123
15:49.28Corydon76-workHrm, nobody has commented on bug #543 yet...
15:49.36Aresactually, I m not greazy about this jazz station, but I would like to know which stream I can use and why ?
15:49.46Pj_Ares: if you find another player which support changing output rate and forcing mono output
15:49.57*** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM]
15:49.57*** join/#asterisk daork (~daork@202.89.35.252) [NETSPLIT VICTIM]
15:49.57*** mode/#asterisk [+bb *!*@og.latency.net *!*dan@194.158.*.*] by asimov.freenode.net
15:49.57*** mode/#asterisk [+q sant!*@*] by asimov.freenode.net
15:50.17Areswhy my mother hit me ? and also if there is a solution to make working an other player ?
15:50.25*** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM]
15:50.25*** join/#asterisk daork (~daork@202.89.35.252) [NETSPLIT VICTIM]
15:50.25*** mode/#asterisk [+bb *!*@og.latency.net *!*dan@194.158.*.*] by asimov.freenode.net
15:50.25*** mode/#asterisk [+q sant!*@*] by asimov.freenode.net
15:50.28bkw_Corydon76 its a nice idea
15:50.30tholo_Only mpg123 is supported by *.
15:50.30AresPj_ : I was looking for it
15:50.30*** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net)
15:50.30RoyKhm
15:50.38Aresthere is a version xmms without gui
15:50.50Aresbut I did find way to get the output
15:50.51Pj_Ares: I worked on supporting mpg321 but it doesn't support changing outptu rate
15:50.52RoyKAres: mpg123 can download it, but not decode it afacs
15:50.55tholoBut still only mpg123 is supported by *.
15:51.33Pj_If you find a player that works better, I'll give you a patched version to use with
15:52.23Pj_just mail me at pj@trouarat.com, with the player you've found... Or if you don't find any gimme urls that don't work with mpg123 and works with xmms and I'll investigate
15:52.38bkw_um whats wrong with mpg123?
15:52.41bkw_it works
15:53.08Pj_bkw_: he says some streams don't work with it
15:53.18Pj_and do work with xmms
15:53.27AresPj_: many thanks, I will try to investigate first and give you feedback soon
15:53.36Pj_but maybe it's just a problem of URL malformation, that kinda things
15:53.48Pj_Ares: I already have patched asterisk to support multiple players
15:53.58Pj_just hadn't found any player interesting enough to add ;)
15:54.32bkw_smokin crack trying to use a stream for MOH
15:54.35RoyKbkw_: it doesn't work with ogg
15:54.41AresWe will try to hack noxmms to get the output, i think that would be great
15:54.43RoyKbkw_: that's what's wrong with it
15:54.44bkw_their is a patch on bugs.digium.com
15:54.51bkw_oh ogg
15:54.55*** join/#asterisk olivier_ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr)
15:54.57bkw_why use something that isn't widely used?
15:55.07RoyKAres: just use mpg123 to pipe the output to something else - like sox
15:55.17Pj_bkw_: because it's freeeeeee
15:55.19Pj_;)
15:55.24bkw_so is mpg123
15:55.30bkw_god you people make me laff
15:55.34bkw_never happy
15:55.38Pj_mpeg 2 layer 3 is a proprietary format
15:55.42bkw_always gotta try to break something
15:55.50bkw_its mpeg 1 layer 3
15:55.58Pj_still proprietary ;)
15:56.05bkw_and its hardly far from proprietary
15:56.18bkw_every os on the planet can pretty play them out of the box
15:56.24Pj_(and I don't give a damn about mp3 being proprietary either :)
15:56.24bkw_er /pretty much/
15:56.55bkw_Pj_ in your little world... you might think its proprietary.. but at some level EVERYTHING is
15:57.02Pj_play them yeah, but not encode them, since you have to pay the Fraunhofer institute for hardware compression... something like this
15:57.17bkw_you dont need to encode mp3's to fucking use hold music
15:57.46Corydon76-workbkw_:  why don't you tell us how you really feel?
15:57.50Pj_bkw_: calm down, I have _no_ ogg vorbis on my hd
15:58.01bkw_Corydon76 I might piss people off
15:58.09Pj_It's Ares who wanted to get an ofgg vorbis stream for his moh ;)
15:58.17bkw_oh fuck it.. you're all idiots
15:58.21bkw_:P
15:58.23RoyKhttp://www.apple.com/imac/ <-- 20" new iMac
15:58.37Pj_GPL idiots, please
15:58.56Pj_or BSD idiots, as you like
15:59.05Corydon76-workOooh, that'd be nice to replace my 15" iMac...
15:59.29bkw_corydon you can sed that 15 inch imac my way
15:59.53bkw_me luv you long time^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H
16:00.07*** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM]
16:00.19AresI don't understand what is wrong to try to get asterisk working better
16:00.36*** join/#asterisk anoosh (~anoosh@217.219.56.183)
16:00.45bkw_whats better about ogg?
16:00.48Corydon76-workAres:  if it ain't broke...
16:00.54bkw_we sould just convert our moh to gsm
16:01.02bkw_and then we wouldn't need mpg123
16:01.52AresCorydon76-work: we will try to fix till we get it
16:02.04anooshok, sorry to jump in the middle of the discussion which i haven't read from the beginning ... but
16:02.14anooshwhy not ogg?
16:02.37bkw_ogg isn't standard
16:02.39bkw_its  open
16:02.42bkw_but its far from a standard
16:03.01anooshbkw said whats better with ogg, well, not having a license problem?
16:03.23*** join/#asterisk super_st (~super_ste@hoochie.digium.com)
16:04.19anooshogg's not a standard?
16:05.10*** join/#asterisk Art3d (~Art3d@vickesh01-6569.tbaytel.net)
16:05.16bkw_Stealth_Man STOP IT
16:05.33*** join/#asterisk bkw_ (~brian@ns.bkw.org)
16:05.51dheckaman??????
16:06.06anooshbkw, you were saying ... (about ogg not being a standard)
16:06.12AresPj_: which id have the patch for support the multi player ?
16:06.27*** join/#asterisk mortck (~test@p213.54.146.115.tisdip.tiscali.de)
16:06.53bkw_anoosh its not a standard.. just like iax isn't
16:07.04bkw_its open but far from a standard
16:07.23*** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com)
16:07.31Corydon76-workbkw_, I think you're confused on exactly what makes a standard
16:08.01Corydon76-workGiven that more than one entity agrees on the format and encoding, it's a standard
16:08.08bkw_no it isn't
16:08.13bkw_in that case we have ALOT of standards
16:08.16Corydon76-worksure it is
16:08.19Corydon76-workYes, we do
16:08.33bkw_just because two geeks agree to encode or decode with X format doesn't make it a standard
16:08.36bkw_IAX isn't a standard
16:08.40Corydon76-workSure it is
16:08.55bkw_not it isn't
16:09.05ExomorphMorning Peoples
16:09.14Corydon76-workYou think a standards organization has to approve it before it becomes a standard?  Hogwash
16:09.19bkw_nope
16:09.28anooshwell it really doesn't matter if it's a standard or not, it works and you can convert audio from mp3 format to ogg
16:09.29AresPj_: still here ???
16:09.34Corydon76-workStandards organizations hammer out differing interpretations
16:09.38anooshtherefore
16:09.55anooshanybody that wants to use asterisk could do just that, and get rid of mp3 license problems
16:09.55Corydon76-workActually, while you can convert audio from mp3 to ogg, it's not recommended
16:10.17bkw_Corydon just because alot of people use something.. doesn't make it standard.. wide acceptance does.
16:10.23bkw_and ogg and iax are far from that point.
16:10.45Corydon76-workSo you're saying that because the world doesn't accept T1, it's not a standard?
16:11.01bkw_T1 is a standard
16:11.16anooshbkw_: do you think it matters to use a widely accepted standard in asterisk's case? newbie computer users are NOT the people that are using asterisk
16:11.16bkw_its widly acceptted in many areas
16:11.17ExomorphHe didn't say the whole world accepts something...  He said wide acceptance.
16:11.22ExomorphTwo diffrent things.
16:11.25Corydon76-workBut T1 doesn't have world-wide acceptance, so in your definition, it's not a standard
16:11.35Corydon76-workHow wide is wide, then?
16:11.47bkw_no I did'nt FUCKING SAY WORLD WIDE ACCEPTANCE
16:12.21bkw_I said WIDE ACCEPTANCE
16:12.29bkw_you people dont listen to me
16:12.30Corydon76-workAnd how wide is that?
16:12.37Corydon76-workIAX has wide acceptance
16:12.39bkw_more than a handful of people
16:12.44Corydon76-workOgg has wide acceptance.
16:12.51bkw_still not wide enuf to take off and be viable yet
16:13.05bkw_once we see hardware IAX phones.. and devices.. it will be a step closer
16:13.10Corydon76-workWhoa, you said standard, not viability
16:13.21dheckamaneek
16:13.24bkw_viability as a standard
16:13.45Corydon76-workIt's already viable as a standard
16:14.07*** join/#asterisk doughecka (~rooot@adsl-68-133.lou.bluegrass.net)
16:14.08tholoFrom Webster: something established by authority,  custom, or general consent as a model or example: CRITERION
16:14.10bkw_in your mind maybe...
16:14.31*** join/#asterisk Stealth (~Stealth_M@h-67-101-129-177.NYCMNY83.dynamic.covad.net)
16:14.51Corydon76-workbkw_:  apparently, if you want it to be standard, it's a standard, and if not, it's not.  That's not a viable definition of a standard
16:15.08anooshlol
16:15.34bkw_tholo pasted the def. of a standard
16:15.52Corydon76-workYes, and by that definition, both IAX and Ogg are standards
16:15.54tholo"General consent" is enough for it to be a standard.
16:16.09bkw_I don't feel ogg is
16:16.13bkw_IAX might be
16:16.17bkw_but I don't think ogg is
16:16.22Corydon76-workLike, I said, it's all about what you think, bkw
16:16.28bkw_:P
16:16.28anooshok, could we drop this 'standard' definition, cuz it's really not important in this case (using ogg for moh) doesn't anybody agree with me?
16:16.31bkw_ok NEXT SUBJECT!
16:16.50anooshhehe
16:17.19RoyKare you still discussing mp3 vs ogg?
16:17.26anooshi am
16:17.32RoyKGIF RULES
16:17.47anooshi think any fairly experienced linux user CAN use ogg if s/he wants to
16:17.52Corydon76-workIf you want to create a patch which allows using ogg for playing MOH without breaking mp3 support, I'd be all for it
16:18.00anooshtherefore it's not important how widely it's accepted
16:18.22anooshit's good enough for solving the problem at hand: mp3 license issue
16:18.23RoyKI started to change MP3Player to use sox instead but was never finished...
16:18.27RoyKthat's the solution
16:18.30RoyKuse sox
16:18.48zwijust my .02...Ogg *is* becoming more standard..now that they have implemented an int based decoder there are several HW players avail...that to me would indicated acceptance of ogg in general
16:18.51anooshwhere's bkw_
16:18.57anthmmaybe the standard belongs in the * itself a standardized interface to allow any external music playing app to register itself as a moh agent
16:19.22anooshthat would be nice anthm
16:19.26Corydon76-workWell, you'd need to expand how moh works, then
16:19.36RoyKit'd work with sox
16:19.45anthmbetter to got from 1 to infinity that 1 to 2
16:19.47Corydon76-workRight now, everything is hardcoded to use mpg123
16:19.52RoyKsox can have any sort of plugins
16:20.01RoyKCorydon76-work: I know - that sux
16:20.07*** join/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl)
16:20.35*** join/#asterisk zeta_ (~zeta@207.88.150.254.ptr.us.xo.net)
16:20.39anooshroyk any resources on sox?
16:20.43Corydon76-workAnd speaking from experience, Mark will not accept any patch which breaks existing functionality...
16:21.06RoyKanoosh: man sox
16:21.07RoyK:)
16:21.44anooshoh, lol
16:22.11anooshthanks royk
16:22.35jetsis there any _good_ documentation on AGI?
16:22.53bkw_not really
16:22.53RoyKjets: /dev/zero
16:22.59anthmcould easily make the hard code be a variable that you can fill in form config and have no refrence to those config default back to mpg123 settings
16:23.09anthminstant pick your app support
16:23.10zeta_heh
16:23.50zeta_is there a spot people have submitted code that uses agi? I've seen a few files from voip-info's wiki, but not much
16:26.04RoyKanthm: perhaps you can just use AGI to code a sox interface or something :)
16:29.20Corydon76-workRoyK:  yeah, right.  Create a new fork everytime you need MOH?
16:29.38RoyKso?
16:30.10Corydon76-workI don't know if anybody else does this, but everytime I need something from AGI, I write an application which does the same thing that I need... everything gets done from the dialplan on my machines...
16:30.27*** join/#asterisk TeKP (~TeK@lan.ciberlynx.net)
16:30.30Corydon76-workThreading rocks
16:30.46bkw_yep
16:30.52knight-anyone have any AGI to pass incoming caller id to a script?
16:31.13TeKPwhenever this one particular cisco sip phone trys to register I get this in my asterisk logfile (and it fails to register): File chan_sip.c, Line 5175 (sipsock_read): Recv error: Resource temporarily unavailable
16:31.16Corydon76-workUh, that's what AGI is, knight-... a script...
16:31.18TeKPanyone know what could cause that?
16:31.27GangBangbkw
16:31.42*** join/#asterisk TimRiker (~timr@proxyle02.ext.ti.com)
16:32.02*** part/#asterisk TimRiker (~timr@proxyle02.ext.ti.com)
16:32.20Powerkillsomeone have the patch of app_dial that return the status code (busy answered unanswanred to the agi script ?
16:32.22knight-I know AGI is a scripting language.
16:32.42knight-I need to pass it to a command line program, that will broadcast the callerid to the appropriate machines
16:33.19knight-nufone_rating?
16:33.45bkw_yes i'm abusing the raing_engine jerjer wrote
16:34.01knight-ratings for what?
16:34.19dheckamanwhats a free instant message app?
16:34.42dheckamanI cant find a working jabber server for windows
16:34.50bkw_knight- you know when you get a bill from the phone company.. with a list of calls.. you need a rating engine for that
16:34.51dheckamanthats free
16:35.01knight-bkw, gotcha
16:35.04zoa<zoa> i have 144 instances of asterisk running :(
16:35.09bkw_ouch
16:35.21knight-bkw, so nufone_rating.so queries nufone's database?
16:35.28bkw_no mine boy
16:35.29bkw_mine
16:35.33knight-gotcha
16:36.51*** join/#asterisk olivier_ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr)
16:40.13Corydon76-workknight-:  actually, AGI is not a scripting language.  It's just an interface.  AGI can be used in many different scripting languages.
16:40.37*** join/#asterisk tmoertel_moz (~chatzilla@pa-bethlprk-cad2-grp1-2d-170.pittpa.adelphia.net)
16:41.06UnixDawgok where is everyone getting the latest flash for the grandstreams mine is outof dat compaired to others
16:41.32UnixDawgand the grandstream site seems to be otu of date
16:41.41tholoGrandstream made a special deal for you.
16:42.00UnixDawgcompaired to others getting new phones
16:42.02UnixDawgno
16:42.04tholoJust to ensure less problems w/FreeBSD 5.1...
16:42.08tholo;-)
16:42.16UnixDawgI am back to 4.9
16:42.20Stealth:)
16:42.21UnixDawgafter a hd crash
16:42.31tholoI didn't say better deal, I said special deal.
16:42.52UnixDawgI want the uptodate flash
16:42.55Corydon76-workOh, good, it must have made an impression on you...
16:42.57tholoSeriously, ...3.81 is the last _official_ release.  People running ...4.x are running pre-release stuff.
16:43.11UnixDawgahh
16:43.13Stealthshit ....
16:43.22decodehmm
16:43.41UnixDawgSoftware Version:    Program--1.0.4.17    Bootloader--1.0.0.11    HTML--1.0.0.19  
16:43.42bkw_haha
16:43.47UnixDawgis what I am looking to test
16:43.48bkw_4.17 works for me
16:43.48*** part/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl)
16:44.13UnixDawgthats what I want to get my hands on
16:44.13UnixDawgI am 3.18
16:44.44decode*sighs*
16:44.49UnixDawgbut thats me wanting to push the edge
16:45.21tholoSo ask Grandstream if you can have access to their pre-release stuff, but don't complain when things don't work for you.
16:45.31decodehmm
16:45.41blitzragemorning all
16:47.00UnixDawgwell my pos compiles and works
16:47.04UnixDawgI am happy
16:47.57*** join/#asterisk bobman (~bobman@mube.psouth.net)
16:48.15decodewhat's the command to invoke the user directory?
16:48.18decodeDirectory?
16:48.26*** join/#asterisk rpb (~rpb@clt74-104-173.carolina.rr.com)
16:49.18Mikehey guys the server that has asterisk uses upload bandwidth for remote clients
16:49.28Mikeand remote clients dont use much upload right?
16:50.17decodeI'm really bored..
16:50.20Mikedo i save bandwidth if i have to asterisk servers?
16:50.25decodeanyways, what would be an easy way to access music-on-hold queues from the menu? :)
16:51.46blitzragedecode: you're bored?  Here, take the keys to my car and go and get it e-tested.  I'll go back to sleep
16:52.06UnixDawgok test 2 of new add functions
16:52.21blitzrageI'm out.  Latah
16:52.58decodeblitzrage can i go pick my gf up first? :)
16:53.24decodeUnixDawg how to call musiconhold from a context? :)
16:53.32blitzragedecode: sure, bring her over, I could always use a good romping :)
16:53.42decodeexten => 8,1,MusicOnHold
16:53.59decodeblitzrage only if you're female and hot ;)
16:54.04UnixDawglooks right
16:54.21blitzragedecode: why does it matter, you won't be involved :)
16:58.05UnixDawghold sorry brb workign on 5 thigns at once
16:59.20UnixDawgtrying to add to my pos software and get the pbx intergrated
16:59.34UnixDawgand got a compile error
17:00.30UnixDawgI dont have music on hold workon fbsd yet due to lack of zztdumy
17:00.37UnixDawgI am on on linux
17:00.50*** join/#asterisk nicebub (~bob@216.253.86.210)
17:00.51*** join/#asterisk n3wb33 (~kapejod@pD9E81940.dip.t-dialin.net)
17:02.01UnixDawgnot on
17:07.03decodeheh
17:07.19decodezztdumy?
17:07.26decodei'm using fbsd..
17:07.37decode4.9 to be exact
17:07.53*** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net)
17:08.43ionix-Hey guys, do you suggest Linux or FreeBSD for asterisk. I am fluent in both OS but I want to know the one that works better with drivers and asterisk and it's modules. Thx
17:08.49*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
17:09.05bkw_ionix- use linux
17:09.10bkw_if you wanna keep your sanity
17:09.12denonunfortunately
17:09.23ionix-ok :/
17:09.24AlricI hate tech support.
17:09.27bkw_fluent in both.. like they are that diffrent?
17:09.38ionix-bkw_: Very if you do more than bash :)
17:09.42denonionix: of course, you could use FreeBSD and help the development effort
17:09.52denontalk to UnixDawg
17:10.02denonhe'll give you the lowdown on freebsd support
17:10.14ionix-denon: I will learn then master it before considering developing. But I will probably ending up coding for asterisk
17:10.15bkw_ionix- um no they aren't that diffrent
17:10.40denonionix-: well, by saying help the dev effort, I didnt mean you had to code, just that you could find and isolate bugs
17:11.04ionix-bkw_: Well Linux is a posix kernel with a bunch of GPL software. FreeBSD is a complete base system in the *BSD family. It comes from unix
17:11.20ionix-they are as different as windows and Mac OS9. They both have icons but that's it
17:11.36bkw_ionix- but they aren't functionaly diffrent from the user point of view
17:11.58ionix-denon: I understand but first I want to master it the easy way so I can be more efficient at isolating bugs
17:12.27ionix-bkw_: yes, if you stay in a bash-like environement and run freebsd with a linuxbase :)
17:12.38ionix-also /usr/ports is wonderful :)
17:12.41bkw_ionix- no it isn't
17:12.46*** join/#asterisk jtodd (~jtodd@207.141.153.205)
17:12.49bkw_their  are small diffrences
17:12.54Art3dAnyone ever tried OS X?
17:12.57outtoluncicons?  since when does linux or freebsd have icons? <G>
17:13.19ionix-I won't argue the FreeBSD/Linux discussion anymore we just have different opinions
17:13.45Powerkillwhy accountcode  is not set on my cdr log even if i specify it on my agi ?
17:14.00UnixDawgfbsd works file libpri works and * works we are waiting on zapata drivers
17:14.04bkw_ionix- how long have you been using linux or freebsd?
17:14.15bkw_but gentoo has really impressed me
17:14.17bkw_its more bsdish
17:14.31bkw_:P
17:15.17ionix-bkw_: Like 10 years
17:15.18ionix-never used gentoo
17:15.18bkw_gentoo is very impressive
17:15.37bkw_portage blows freebsd ports away
17:15.54bkw_wish freebsd ports had the same features... but oh well.. i still like freebsd over linux anyday
17:16.04ionix-bbl
17:16.27Powerkillbkw any idea why $AGI->exec('SetAccount',$pin) don't set the pincode in the cdr ?
17:16.35*** join/#asterisk lexluther (~You@212.247.106.52)
17:16.45bkw_Powerkill it makes a new file $pin.csv
17:16.47*** part/#asterisk lexluther (~You@212.247.106.52)
17:16.50bkw_if you use csv stuffs
17:17.13Powerkilland how can I put it on mysql ?
17:17.19bkw_cdr_mysql
17:17.28Mikebkw_: do i save bandwidth if i have 2 asterisk servers???
17:17.33Powerkillmy accountcode colum is empty
17:17.43Mike3 extensions in my server 1 extension in the other server
17:17.44Powerkillbkw_ i'm using cdr_myslq
17:17.47Mikedo i save bandwidth?
17:17.52bkw_Mike I will answer that if you hire me as a consultant... but I and many others have already answered that question.
17:18.06bkw_Powerkill it should be in there
17:18.07decodebkw_ portupgrade is really nice
17:18.22bkw_Powerkill you need a rating engine
17:18.24Powerkillbut it's not
17:18.28n3wb33Mike: you will save bandwidth if you have 0 * servers!
17:18.30Powerkillrating ???
17:18.41bkw_decode portupgrade is nice
17:18.44Miken3wb33: i have 1 asterisk server right now
17:18.48Miken3wb33: 3 extensions
17:18.59Miken3wb33: i want to put another extension on a remote point
17:19.13Miken3wb33: but i could also put a server on the remote point
17:19.20Miken3wb33: the thing is would it be worth?
17:19.28bkw_n3wb33 want the cluebat?
17:19.45n3wb33bkw_: please
17:19.46Mikebkw_: how much per hour/answer?
17:20.11bkw_Mike depends on what you want!
17:20.18bkw_ok that sounded kinda whorish
17:20.32decodehah
17:20.33Mikei just want to know if i would save bandwidth putting another server
17:20.36Mikeon the remote point
17:20.42decode* is cool :)
17:20.49decodebbiab tho
17:20.50Powerkillbkw_ any other idea ?
17:20.51decodecleaning lights
17:20.52bkw_Mike yes you could save bandwidth
17:20.58UnixDawgman I hate when a errror is due to case
17:21.19bkw_Powerkill check your msges
17:21.21Mikebkw_: how much and why?
17:21.43bkw_the ability to use lower bandwidht codecs between the * servers
17:21.59Mikewithout lossing quality?
17:22.43dheckamanilbc isnt too bad quality
17:22.52dheckamangsm is good too
17:23.01dheckamanwhat is it, 30 kb/s?
17:23.06dheckaman~30/8
17:23.073.75
17:23.13Mikenice
17:23.24Mikeand its as good as ulaw?
17:23.26Mikealaw?
17:24.03dheckamannot AS good
17:24.03dheckamanbut not noticable
17:24.03cypromisnothing is as good as ua/law
17:24.03dheckamantry it out
17:24.03dheckamanand see for your self
17:24.32ares_i getting like really bad echo on mine and low volume... increasing the gain increases the echo. :\
17:24.38Mikeulaw and alaw simply are really as good quality as landlines
17:24.49dheckamanyep
17:24.56dheckamanAres: I have the same problem
17:25.01dheckamanexept no echo
17:25.05ares_i read the mailing list...same problem, no solution..
17:25.08dheckamanbut horrible volume
17:25.11Mikeso you guys recommend gsm?
17:25.18dheckamangsm and ilbx
17:25.19dheckamanilbc
17:25.25Mikewhats better gsm?
17:25.31dheckamanilbc uses slightly less BW
17:25.32ares_no solution?
17:25.36dheckamanbut uses more CPU
17:25.45Mikedheckaman: but the quality
17:25.47dheckamanAres: there IS a solution, just havnt found one yet :)
17:26.02Mikewell i dont think 4 extensions in total can use alot of cpu:)
17:26.03ares_did you try increasing the gain?
17:26.03dheckamanMike: cant say, try it out your self and see how it sounds
17:26.12dheckamanMike: indeed
17:26.13Mikeim not really setting up AT&T network or something
17:26.18dheckamanyea
17:26.45dheckamanilbc has better packet loss recovery too, I think
17:26.56Mikethe problem is grandstream phones dont support
17:26.56Mikegsm
17:26.57Miketho
17:27.07dheckamanthey dont?
17:27.21dheckamanoh
17:27.25Mikeplus i have no idea how to link 2 asterisk servers
17:27.27dheckamanthey must support ulaw
17:27.27dheckaman:)
17:27.31dheckamanme niehter
17:27.34dheckamanthats for the higher ups
17:27.35Mikeyes they support ulaw
17:27.41Mikebut asterisk converts it back to ulaw?
17:27.46dheckamansure
17:27.46ares_i used IAX to link, it's so easy ...
17:27.57Mikeares IAX?
17:28.00ares_just use Dial/IAX/EXTENSION
17:28.00Mikeand what codec do you use?
17:28.01dheckamanit can do ilbc to ulaw or whatever
17:28.06ares_yea...read up on that.
17:28.07ares_it's like SIP
17:28.13ares_or you can even use SIP too
17:28.14dheckamanbut better
17:28.15RoyKDial(AIX,ibm)
17:28.28dheckamanif its between 2 * servers, use IAX
17:28.38dheckamanRoyK: hah
17:28.52Mikeok ill use IAX
17:28.59Mikeanyone has any url on where to read
17:29.01Mikean example
17:29.03Mikeor a howto
17:29.05Mikeexplanation
17:29.10UnixDawgok the pos works
17:29.11Mikeor what ever that can help me figure it out
17:29.11Mike?
17:29.18UnixDawgbut not the new addin
17:29.21UnixDawggrr
17:29.21dheckamanread the email list archive
17:29.38dheckamanand digium.com has lots of info in the documentation section
17:30.52UnixDawgwho know the url for the webpage where you type in text and it creates a wav file with a voice reeading the text
17:31.06dheckaman~att
17:31.13dheckaman~rhetorical
17:31.17dheckamangoogle for it
17:31.27bdeUnixDawg: http://www.research.att.com/projects/tts/demo.html
17:31.29UnixDawgI have not finding it
17:31.38dheckamanrhetorical is better
17:31.39UnixDawgthnks bde
17:32.44Powerkill~seen JerJer
17:32.47jerjer is currently on #asterisk
17:32.52PowerkillJerJer are you here ?
17:34.41decodeomg
17:34.50decodei just found a trashbag full of marijuana in my attic :)
17:35.26*** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net)
17:35.54decodedheckaman if it were illegal here :)
17:36.07decodemy hostname is not where i am :)
17:36.41dheckamanheh
17:37.02*** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com)
17:37.23dheckamananyone know of a free instant message server?
17:37.31carrarICQ
17:37.34dheckamanI cant find a free windows server that works
17:37.42dheckamanI said SERVER :P
17:37.47carrarircd
17:37.52dheckamanugh
17:37.57carrarheh
17:38.04dheckamanI want something like MSN or something
17:38.08carrarit's instant
17:38.09carrarand
17:38.18carrarYou can use the windows trillian IM client with it
17:38.21cypromisjabber
17:38.27cypromis:)
17:38.27tzangerjabber kicks ass
17:38.29Powerkilldheckaman jabber
17:38.31anthmjabber
17:38.32tzangerpsi is my client of choice
17:38.53carrarwrite one in perl
17:39.18dheckamanI cant find a free jabber windows server
17:39.21dheckamanthat works
17:39.26dheckamanpsi?
17:40.45tzangerpsi.sf.net
17:40.46tzangerI think
17:40.55tzangerpsi.affinix.com rather
17:41.16bkw_pisy
17:41.20tzangermultiplatform, qt-based and does NOT take up all your screen real estate..  it behaves very much like the old ICQ client, which is IMO the best
17:41.25tzangerhahahha
17:41.33dheckamanhmm, but thats not the server
17:41.34dheckaman:(
17:41.42tzangerdheckaman: well use jabberd then
17:42.05dheckamanon windows?
17:42.29decode"Chong's Bong. Press 1 for sales, 2 for seed catalog orders, or 3 for order information"
17:42.39dheckamanlol
17:42.49decodeChong's Bongs that is
17:43.01dheckamanoh cool
17:43.04dheckamanI found it
17:43.16decodeheh..
17:43.23decodeMy dialplan is all screwed up
17:44.04hellerah. you need the 'unscrewup' command.
17:44.49decodestill cant get fwd stuff working, yet..
17:44.53decodebut that's to be fixed tonight
17:44.58decodeas are incoming fwd calls :)
17:45.28decodebbl, gf and i are off to the shower >:)
17:46.17carrarbf? you kick out your bf
17:46.23carrarerr gf
17:46.27carrarhaha
17:46.28dheckamanLOL
17:53.36*** join/#asterisk zatu (~zatu@6-VALL-X5.libre.retevision.es)
17:55.09*** part/#asterisk mortck (~test@p213.54.146.115.tisdip.tiscali.de)
18:00.47decodeok, question
18:00.58decodei wanna send calls to 411 to 411 via fwd, how? :)
18:01.00carrarsorry no more questions today
18:01.53*** join/#asterisk Simon_ca (~sedgett@h24-81-97-130.vc.shawcable.net)
18:04.20*** part/#asterisk pablo_slw (~zatu@6-VALL-X5.libre.retevision.es)
18:04.54*** join/#asterisk rajo_home_ (~rainer@p508AEFB0.dip.t-dialin.net)
18:06.27*** join/#asterisk ctooley (~ctooley@199.89.146.58)
18:06.46Connor-anyone have problems with compiling pwlib?
18:06.53ctooleyWhich of the VoIP providers that offer an unlimited call plan is most likely to work with Asterisk
18:11.08*** join/#asterisk UnixDawg (~UnixDawg@ny-lasalle6c-203.buf.adelphia.net)
18:16.59RoyKctooley: what do you mean - unlimited call plan?
18:17.10jsharpFlat rate.  No per minute charges.
18:17.24ctooleyRoyK: $19.95/month no per minute fees
18:17.42bkw_all of em work with *
18:17.52bkw_just gotta use an FXO port on some of em
18:18.12ctooleybkw_: :) That's not really what I meant.
18:18.36RoyKunless they're running JHSVOIPP
18:18.37ctooleyI'm trying to avoid the whole analog conversion part.
18:19.55bkw_jsharp ahhahahahah funny
18:19.57*** join/#asterisk rusty (~rusty@65-101-254-85.dnvr.qwest.net)
18:20.27ctooleyjsharp: I've got that!
18:21.46UnixDawggrrrr
18:21.49bkw_we need sip.conf per peer agent strings
18:22.16Connor-need a way to use Radius for sip.conf...
18:22.21zoayes
18:22.25ctooleyI've written an HTML-RPC language, for switch to switch updats, that has all of the negatives of a verbose slow communication protocol without the flexibilty of XML-RPC and thats an "HTML Networking Technology"
18:22.41Connor-I don't give a phuck what JerJer says. :)
18:22.47RoyKConnor-: why the fsck do you want radius?
18:22.54zoatop - 22:22:34 up 8 days, 11:21,  3 users,  load average: 20.98, 20.97, 20.91
18:22.56zoaTasks: 198 total,  22 running, 174 sleeping,   0 stopped,   2 zombie
18:22.56zoaCpu(s):   2.3% user,   0.6% system,   0.0% nice,  97.1% idle
18:23.05RoyKConnor-: jerjer's a jerk. don't bother about  what he says
18:23.33Connor-Because, I've got a very nice frontend billing and controll system I use for my ISP which works with Radius.
18:23.41Connor-No need to re-invinte the wheel for VoIP
18:23.52nicebubi agree
18:23.56nicebubsame for me
18:23.57RoyKagreed
18:24.17RoyKso - what you need is an app_radius
18:24.19RoyKor something
18:24.20Connor-I just got my cisco working for ANI authentication with my radius and it works great.
18:24.31jsharpOr pam_radius.
18:24.35zoatop - 22:24:14 up 8 days, 11:22,  3 users,  load average: 21.10, 21.00, 20.92
18:24.36zoaTasks: 198 total,  22 running, 174 sleeping,   0 stopped,   2 zombie
18:24.36zoa<PROTECTED>
18:24.36zoa<PROTECTED>
18:24.36zoa<PROTECTED>
18:24.36zoa<PROTECTED>
18:24.38zoaMem:   2068704k total,  1261584k used,   807120k free,    17008k buffers
18:24.40zoaSwap:  2097136k total,        0k used,  2097136k free,  1087992k cached
18:24.41Connor-Well.. need a app.. but how would it work to replace sip.cof ?
18:24.44zoaany suggestions ? :(((
18:24.59RoyKzoa: you have too many CPUs
18:25.03Connor-zoa, you using software raid ?
18:25.10Connor-Nah.. he's dual CPU with HT
18:25.15zoanopez, hardware raid
18:25.23*** join/#asterisk adam_gafachi (~diddy@69-55-69-130.da.netsville.net)
18:25.26RoyKzoa: what is it that uses all that?
18:25.30RoyKload avg 21?????
18:25.30zoaasterisk
18:25.37zoa144 instances of them
18:25.41bkw_zoa did you compile everythign to use SMP
18:25.41RoyKshite
18:25.44bkw_zaptel and all the goodies?
18:25.47zoayes
18:25.57RoyKtried disabling smp or ht?
18:26.06zoait normally is max 3% usage
18:26.13zoabut after an hour this is what happens
18:26.23RoyKzoa: after an idle hour?
18:26.36zoanopez, after an hour of 20 simultaneous calls
18:26.37RoyKzoa: connect to it with gdb and see what's going on
18:26.52bkw_zoa get kram that login info and let him look at it
18:26.58zoak
18:28.07Connor-So anyway, I think we need to have a way to do Radius for sip.conf
18:28.08RoyKConnor-: how does radius work? do you have a radius server everyone authenticates to, and then anyone can ask that server 'is tom really tom and should he be there?'
18:28.34Connor-2 ways, Auth only, or Auth and pass attributes needed
18:28.43Connor-For the AUTH side..
18:28.50Connor-then there is accounting, when you
18:29.02Connor-have auth'd, the radius gets a accounting START record..
18:29.05bkw_ewww
18:29.10Connor-when the session ends, you get a stop record.
18:29.22bkw_Connor dont strat the radius flame.. please
18:29.28bkw_if you want it.. write it..
18:29.35Connor-Not wanting it for tracking calls..
18:29.53Connor-I'm wanting it for a replacement for sip.conf
18:29.53bkw_I know but it will lead to call tracking
18:30.09bkw_someone was working on a pam module for *
18:30.11bkw_who was that
18:30.16bkw_then you coud use a pam radius module
18:30.25Connor-That's easy enough anyway.. We do that with or Term Server anyway.. What's the diff?
18:30.44jsharpThat would be me.
18:30.46Connor-That's what Radius was designed for.. Authenticating and Accounting.
18:30.56jsharpI'm hacking on the code right now, in fact.
18:31.03bkw_jsharp what will it allow?
18:31.15jsharpRight now, I'm just using it for authentication.
18:31.26zoakram has been idle 3mins 53secs, signed on Fri Oct 31 23:31:55
18:31.30zoa-kram- I left 14h 4m 40s ago it is 12:29PM right now. info: Not here right now, sorry...
18:31.39bkw_but in theory he could use radius to auth then right?
18:31.41zoayeah right kram :p
18:31.44jsharpAye, he could.
18:31.57h3xzoa: probably anti-idle or something
18:32.04RoyKif someone were to use asterisk to allow 500 users to use it, 120 concurrently... is this possible with a single server?
18:32.29bkw_RoyK should be possible
18:32.35jsharpWith a beefy enough machine, sure.
18:32.43bkw_ya what he said
18:32.55RoyKwhere would the limit go, if using analog phones?
18:33.00zoaroyk: im getting a quad opteron
18:33.05zoaill let you know
18:33.14*** join/#asterisk jtodd2 (~jtodd@207.141.153.205)
18:33.19Connor-Can I ask a question..  Why are so many people against Radius?
18:33.34jsharpCause a lot of people think of it as a really ugly hack.
18:33.39zoai can do 80 with ilbc on a dual xeon max
18:33.56jsharpWill * build and run on an opteron?
18:33.59Connor-jsharp: explain ?
18:34.08zoawhy would it not build and run :
18:34.09zoa?
18:34.13high-rezjsharp: in 32 bit mode for sure.
18:34.14cypromisjsharp: it should
18:34.15jtodd2bkw_: re: your per peer agent strings: see my long "why sip.conf needs to see some re-work" post of a few months ago. It's in there, with a bunch of other issues that probably would improve *'s scalability for sip.
18:34.23cypromisI am just ordering a couple to work as codec translation engines
18:34.55zoacypromos: when will they arrive ?
18:34.57cypromisRoyK: a dual xeon will do 120 concurrent calls
18:35.04cypromiszoa: no idea, I am in poland :)))
18:35.11zoacypromis: depends on the codec
18:35.28cypromiszoa: ok
18:35.33RoyKcypromis: is that a xeon with 1MB L3 cache?
18:35.50cypromisnope
18:35.53cypromiswe use the 512 ones
18:35.58zoathe l3 cache is non relevant according to toms hardware
18:36.34RoyKsounds strange ...
18:36.49Bonbondon't the allow/disallow codec commands work within an iax context?
18:36.50RoyKmore cache usually gives speed on at least databases and such
18:36.52zoathe 1mb version is not a lot faster i mean
18:37.05RoyKat least not for asterisk use...
18:37.08zoabonbon: i have the same impression
18:37.16Bonbonthe don't work
18:37.42Bonboni have disallow=all, allow=ilbc in the general context
18:37.49zoapfft, i cant even do 40 concurrent pass thru calls atm :(
18:37.55Bonbonthen i have disallow=all, allow=ulaw in one of the contexts.
18:38.04zoaand i need 350 concurrent calls
18:38.05Bonbonyet, it still negotiates ilbc
18:38.08cypromisI think you cannot allow a codec per peer that you disallow in the general context
18:38.10cypromisbesides that
18:38.11cypromisit works fine
18:38.13zoabonbon: same here
18:38.14RoyKzoa: 350???
18:38.16RoyKhow many phones?
18:38.19zoa350
18:38.25UnixDawgbusniess
18:38.30*** join/#asterisk ajh (~ajh@c-24-0-23-89.client.comcast.net)
18:38.32RoyKyeah - but that doesn't mean 350 concurrent calls.
18:38.33cypromiszoa: on what box
18:38.41cypromiswe have far far far more than 40 passthrough calls in parallel
18:38.42RoyKthat means some ~150 or so
18:38.43zoathat means 350 concurrent calls :)
18:38.44Bonboncypromis: i've allowed all codecs in the general context
18:38.54cypromisBonbon: strange
18:38.58cypromisbut on the other hand
18:39.11cypromiswe don't use ilbc anymore sinc it is somehow buggy
18:39.12RoyKzoa: what chance is there that everyone in the building are speaking on the phone at the same time?
18:39.24RoyKcypromis: stopped using it?
18:39.25zoaits an outbound callcenter
18:39.35zoaquite high id say :)
18:39.38RoyKoh
18:39.38RoyKok
18:39.41Bonboncyypromis: disallow=all
18:39.41Bonbonallow=ilbc
18:39.41Bonbonallow=gsm
18:39.41Bonbonallow=ulaw
18:39.41Bonbonallow=alaw
18:39.47zoathey want me to go up to 3000
18:39.52decodeblah
18:39.58zoathey are insane
18:40.00Bonboncypromis: in specific context:
18:40.05UnixDawgok now to conver to gsm
18:40.05nicebubdid nufone write their own AAA system?
18:40.06Bonbondisallow=all
18:40.06Bonbonallow=ulaw
18:40.08Connor-Multiple boxes man. Lots of them
18:40.11cypromishmm I have nothing in general and do specific only
18:40.22decodecan't get this damn thing working :(
18:40.47decode(freeworld dialup incoming forwarded to sipphone of name xlite)
18:41.02UnixDawgsox file.wav -??? 8000 file.gsm
18:41.17Connor-sox file.wav file.gsm
18:41.20cypromis3.000 phones ?
18:41.23cypromisor parallel calls ?
18:41.34RoyKon a duron 1200
18:41.55cypromisnah we don't use amd 32 bit stuff since they get too hot
18:41.59zoascxgk3:/var/log/asterisk# lsof | wc -l
18:42.00zoa<PROTECTED>
18:42.19RoyKzoa: wtf?
18:42.47zoadont ask me
18:42.51zoaits asterisk :)
18:43.05RoyKlsof| cut -c67-|grep ^/ | wc
18:43.22RoyKjust want to know how many _files_ there are
18:44.07zoascxgk3:/var/log/asterisk# lsof| cut -c67-|grep ^/ | wc
18:44.07zoa<PROTECTED>
18:44.14zeta_wow
18:44.48*** join/#asterisk Chris_DE (~Chris_DE@p5083053C.dip0.t-ipconnect.de)
18:47.05RoyKzoa: meaning what's a _lot_ is ! files
18:47.38RoyKzoa: lsof| cut -c67-|grep -w pipe | wc
18:48.02zoalots pipes
18:48.05zoalotsa
18:48.13RoyKzoa: lsof| cut -c67-|egrep -v "^/|pipe" | wc
18:48.43zoajust a sec
18:49.26zoascxgk3:/var/log/asterisk# lsof | cut -c67-|egrep -v "^/|pipe" | wc
18:49.26zoa<PROTECTED>
18:49.27RoyKzoa: or netstat --inet
18:49.33RoyKheh
18:49.34*** join/#asterisk simonaut (~simonaut@hoochie.digium.com)
18:49.39RoyKtry netstat --inet|less
18:49.55zoaonly 2 ssh connections
18:50.00RoyKhuh?
18:51.21zoascxgk3:/var/log/asterisk# lsof | grep asterisk | wc -l
18:51.21zoa<PROTECTED>
18:51.44RoyKps ax|grep -w asterisk|grep -v grep|wc
18:51.55RoyK-l
18:52.20zoascxgk3:/var/log/asterisk# ps ax|grep -w asterisk|grep -v grep|wc -l
18:52.21zoa<PROTECTED>
18:52.32UnixDawgnope need sox file.wav -?s? 8000 file.gsm and I cant recall it all
18:52.36UnixDawg~sox
18:52.38somebody said sox was Sound Processing Tool. URL: http://sox.sourceforge.net/
18:52.42UnixDawg~sox info
18:52.52UnixDawg~sox command
18:52.57denon!info sox
18:52.57denon?
18:52.59RoyKUnixDawg: man sox
18:53.08RoyKzoa: how many concurrent calls do you have now?
18:53.09denonhrm, Ainfo is gone
18:53.29zoaroyk: nobody
18:53.34zoait stopped working
18:53.35Bonbonzoa: if a call comes into my server via iax, with no defaulty context or anything then where would it go?
18:53.39zoa:)
18:53.43RoyKgdb -p ...
18:54.08UnixDawgwo knows what the sox output options are to conver from wav to gsm
18:54.14RoyKUnixDawg: sox file.wav -r 8000 file.gsm
18:54.29RoyKUnixDawg: it just looks at the extension
18:54.31RoyKthat's all
18:54.35UnixDawgsounds to slow
18:54.40RoyKslow?
18:54.40UnixDawgok
18:54.54UnixDawgyes sounds liek a 45 on 33
18:54.56zoa0x08051ad3 in ast_sched_add (con=0x4023a1a8, when=0,
18:54.56zoa<PROTECTED>
18:54.56zoa167                     if (SOONER(s->when, current->when))
18:55.03zoais the last thing it gives me
18:56.11tmoertel_mozAnybody know what " File app_dial.c, Line 518 (dial_exec): Unable to create channel of type 'sip'" means?
18:56.22tmoertel_mozI can't bridge a Zap -> SIP call.
18:56.24nicebubrun sip debut
18:56.30nicebubsip debug
18:56.36af_which port I have to open to permit h323 traffic between * and an ip public gnomemeeting?
18:56.36nicebubor reload module
18:56.37RoyKdoes SIP have silence detection?
18:56.42af_ports
18:56.55RoyKaf_: through nat?
18:57.05tmoertel_moznicebub: I am.
18:57.13af_no RoyK * is public, and so gnomemeeting
18:57.27RoyKaf_: the RTP stream is chosen at run time, so you need an agent that speaks h323
18:57.43RoyKer
18:57.49RoyKthe RTP stream's ports ...
18:58.49af_which are those ports?
18:58.50UnixDawgok they sound better
18:59.24RoyKaf_: they are determined at run time
18:59.30RoyKaf_: something > 1024
18:59.50UnixDawgok my inbound iax nmbr is now not working
18:59.50af_mhhh
18:59.53UnixDawggrrr
19:00.04af_udp or tcp, RoyK?
19:00.12RoyKrtp uses udp
19:00.19RoyKvoip over tcp???
19:00.20af_thanks
19:00.21RoyKdon't think so
19:00.32RoyKaf_: what sort of router/firewall?
19:01.00af_iptables
19:01.19RoyKthere's an h.323 iptable conntrack patch in patch-o-matic
19:01.31nicebubconntrack?
19:01.33RoyKbut it's not stable yet, so YMMW
19:01.35tmoertel_moznicebub: there's no sip traffic
19:01.44Bonbonwhat is the [guest] part for in iax.conf?
19:01.51RoyKnicebub: linux connection tracking
19:01.58nicebubrework your config file...try unloading and reloading the module
19:02.12af_I don't do nat
19:02.22nicebubRoyk: reports status on current connections?
19:02.25zoaroyk: do i have to type anything else in gdb ?
19:02.26RoyKBonbon: it allows me to use your PBX to call sex phones
19:02.50RoyKzoa: I don't really know how to debug this. sorry
19:02.58Bonbonroyk: but i can't get it to work. if i make an iax connection to a box, with no username or secret, then wouldn't it go to this context?
19:02.59zoastrace is not giving anything
19:03.09tmoertel_moznicebub: Dial("Zap/1-1","sip/5001|20") causes the problem
19:03.19Bonbonroyk: it seems to go to the first non-guest context that it can successfully authenticate
19:03.25tmoertel_moznicebub: restarting all of * doesn't solve the problem
19:03.33tmoertel_moznicebub: no errors in log, either
19:03.49RoyKnicebub: yes. /proc/net/ip_conntrack
19:04.05nicebubt_moz:when you list your mdules does it show up
19:04.13tmoertel_moz<PROTECTED>
19:04.33RoyKnicebub: with connection tracking enable for a protocol, you can NAT it even if it's 'dirty', like FTP
19:04.41tmoertel_moznicebub: chan_sip.so
19:04.49UnixDawgI have my iax done wrong again
19:04.52nicebubRoyk: i'm checking out the page ,now
19:05.01nicebubRoyk: looks useful
19:05.24Yog-homeis there a difference between Voicemail() and Voicemail2() ?
19:05.38RoyKIIRC Voicemail2 is 'beta'
19:05.40denonYog-home: not anymore
19:05.50denonno ... Voicemail2 and Voicemail are both Voicemail2 now
19:05.53Yog-homeyeh..it looks the same
19:05.55Yog-homein the code
19:05.57denonso just use Voicemail
19:06.01Yog-homewell..from my first glance
19:06.10denonthey are identical as of the other day
19:06.16nicebubare you leacing bindaddr alone?
19:06.37Yog-homedoes anyone have all the "show application XYZ" type documentation collected together in a doc file ?
19:07.08Yog-home<- new to *
19:07.54zoayou can find it on voip-info.org
19:08.27Yog-homeo..the wiki
19:08.55tmoertel_mozgrrr... kernel oops in wcfxo
19:09.13UnixDawgok who has voicepulse in thier iax.conf that point otu what I missed
19:09.36Yog-homeis the X100P pretty much the 'card to get' for home phone line type deal ?  
19:09.49jsharpYog-home Yes.
19:09.54*** join/#asterisk Marlow (~marlow@2001:838:300:70:0:0:0:2)
19:10.12Yog-homeyeh i guess it's good too since it supports *
19:10.43jsharpYah.  And if you buy an X100P, you support the company that wrote asterisk.
19:10.55jsharpfreenode supports ipv6?  Interestin.
19:11.22UnixDawg?
19:11.24Yog-homeyeh!  i just noticed the V6 addy too!
19:12.49jsharpYou'll probably have to get a tunnel through one of the tunnel brokers.
19:13.04Yog-homeIs the X100P made by digium or Zaptel ?  I notice the drivers for it are 'Zaptel'...
19:13.10Yog-homejsharp: yeh probably
19:13.40bkw_Yog-home digium
19:13.40jsharp"zaptel" is the name of the class of hardware that Digium makes and asterisk supports.
19:13.49Yog-homeV6 has nice built in tunneling capability..can be somewhat transparent in some cases
19:14.00Yog-homeah. I see
19:14.34Yog-homeprobabaly from the tunnel broker if your ISP doesn't support it
19:14.41tholofreenet6
19:14.42Yog-homeor ARIN maybe for a portable one
19:14.45jsharpdheckaman:  You can go to any of the tunnel brokers.  freenet...hurricane electric.
19:15.07jsharpYog-home:  But if you get one from ARIN, you still need to find an upstream that you can peer with.
19:15.12Yog-homeyep
19:15.15dheckamanif I have a public ip now, can I just make myself a ipv6 address?
19:15.19dheckamanand I just show up?
19:15.21jsharpNope.
19:15.31dheckamanoh
19:15.32jsharpYou need to peer up with an ipv6 provider.
19:15.38dheckamanwhy?
19:15.41jsharplike freenet6.net or tunnelbroker.net
19:15.49Yog-homethey probably don't give out those to anyone either..I'm sure they're keeping some sort of CIDR type routing in effect so they want everything aggrogated to keep the # of routes down
19:15.52nicebubRoyk: im running 2.4.18-27.0.8 and don't see /proc/net/ip_conntrack
19:16.12jsharpFor the same reason you need to peer up with someone to get IPV4 connectivity.
19:16.14nicebubhmm
19:16.17dheckamanI dont want a tunnel, I just want to connect to freenode on ipv6
19:16.19jsharpYOu can't just magically appear on the network, unfortunately.
19:16.19dheckamanah
19:16.21RoyKnicebub: insmod ip_conntrack
19:16.40Yog-homedheckaman: you need an IP address that's routable to your provider...
19:16.44RoyKnicebub: it's only useful for firewalls, really
19:16.49dheckamanah
19:16.52dheckamanhmm
19:17.02jsharpdheckaman:  Unless your ISP can give you native IPV6 addresses, you must build an IPV4 GRE tunnel to an IPV6 provider.
19:17.02Yog-homedheckaman: and I bet 90% of providers today (ISPs) arn't routing V6 yet
19:17.11jsharpThey're not.
19:17.13nicebubRoyk: got it
19:17.19dheckamanah, ok
19:17.33RoyKnicebub: and upgrade your kernel. 2.4.18's OOOOLD
19:17.51nicebubcan i do it from command line?...
19:17.59RoyKdo what?
19:18.00nicebubthe machine is in mexico..i'm in colorado
19:18.05nicebubupgrade the kernel
19:18.15Yog-homewonder how much addy space the avg broker is giving out...V6 addresses are so huge that I'dthink you'd get a at least 8 bits of address space easily
19:18.30RoyKyou can, but you need to reboot and if something goes wrong, you're looking forward to a rather long trip
19:18.31jsharpI had a /48 from hurricane electric.
19:18.39dheckamanLOL
19:18.42jsharpWhich is a metric buttload of addresses.
19:18.58Yog-homejsharp: haha..nice ... 16 bits ... "class B"
19:19.04jsharpHeh.  Yeah.
19:19.11nicebubi do have someone down there in case something happens
19:19.17dheckamandoes it cost money to get a ipv6 range?
19:19.21jsharpNot yet.
19:19.24Yog-homedheckaman: probably
19:19.24tholoTypically, when you sign up for IPv6, you will get a /48 prefix...
19:19.27Yog-homeo really ?
19:19.27Yog-homehrm
19:19.40dheckamanfor free?
19:19.49dheckamanthen I can wait 20 years
19:19.50Yog-homethe tunnel brokers would charge for the service
19:19.52dheckamanand sell it to MIT
19:19.59tholoDepends where you get it.  freenet6 is free...
19:20.06lecram<PROTECTED>
19:20.23jsharpI wish I could figure out how to get IPV6 here at home.  Damn NAT.
19:20.55UnixDawggrr my inbound is not working
19:21.03UnixDawgmakes no sense
19:21.10Yog-homemost stuff isn't doing v6 native anyway...so it'd wind up being xlated back into a V4 anyway
19:21.48Yog-homeo hold it... /64...that'd be 64 bits of address space ?  V6 is 128 bits if I remember
19:21.53sxpertYog-home: postgresql 7.4 has native v6 support
19:22.42lecramusually from the beginning
19:22.55Yog-homeyeh..that's how they do it w/ V4..i'd expect the same for V6
19:23.23dheckamando I have to have a public ip to use ipv6 on the internet?
19:23.34Yog-homedheckaman: probably not
19:23.42Yog-homedheckaman: just need to be able to tunnel
19:23.54dheckamanhmm
19:23.55Yog-homeyeh... more than you'd ever need :)
19:24.04jsharpYou need either a public IP, or be able to do "stupid routing tricks" with your NAT firewall.
19:24.07dheckamanwhat is the tunnel like? tcp/udp?
19:24.16jsharpGRE.
19:24.27sxpertjsharp: no nat necessary ;)
19:24.31Yog-homeit's just IP in IP
19:24.38dheckamanaah
19:24.39nicebubi love conntrack
19:24.52Yog-homeV6 packed gets put intisde a V4 packet and sent to the tunnel provider
19:24.52Marlowcan anybody tell me, what dtmf fwd accepts ?
19:25.05jsharpExcept I'm running NAT at home and that breaks the GRE encapsulation.
19:25.24Yog-homejsharp: hrm..GRE shoudl work thru NAT
19:25.34Marlownative ipv6 is neat, but not easy to get ..
19:25.35dheckamanmy ipcop has the option for GRE
19:25.41dheckamanI allways wondered what that is
19:25.47sxpertRoyK: lol
19:25.55sxpertRoyK: lol
19:26.07sxpertRoyK: is definitely jealous ;)
19:26.09*** topic/#asterisk by kram -> the tornado sounds like a freight train
19:26.10Marlowhowever .. they are down right now :(
19:26.19denontornado?
19:26.24kram*nods*
19:26.26sxpertRoyK turned into a slapper-bot
19:26.32Yog-home<PROTECTED>
19:26.35sxpertkram: quick, hide in the basement
19:26.37denontornados are kinda cool .. but ive been a little too close to em
19:26.42denonso I get nervous now
19:26.50RoyKkram: I think zoa has some serious problems
19:26.57RoyKkram: needs a new shrink or so
19:27.03zoaaha kraaaaam :)
19:27.19denonsomeone ... quick mirror CVS before digium goes to Kansas :)
19:27.25MarlowRoyK : roligt nu .. det holder han jo ikke til ..
19:27.38denonT400Ps falling from the sky ..
19:27.40denonChristmas!
19:27.44sxpertRoyK: same for you I guess
19:27.48RoyK...svensker...
19:27.49jsharpkram's gettin the same weather system that blew through here yesterday.
19:27.54MarlowRoyK : niksen ..
19:27.58MarlowRoyK : dansker ..
19:28.07RoyKah - ok - ikke så galt...
19:28.14MarlowRoyK : i Sverige :(
19:28.21RoyK:)
19:28.23RoyKjj
19:28.23MarlowRoyK : men ikke længe mere ..
19:28.23Yog-homeo no..more swedes
19:28.28Yog-homethey're everywhere :)
19:28.37RoyKYog-home: he's danish - he just lives in .se
19:28.45cypromisshit happens
19:28.45MarlowYog-home : nope . as i just said to RoyK  ... i'm danish ..
19:28.46Yog-homeah
19:29.03MarlowYog-home : and that i live in .se is very temporary ..
19:29.03Yog-homedanish..swedish...it's liek a 10 mile drive :)
19:29.37MarlowYog-home : nah .. more like 375 miles ..
19:29.47Yog-homeheh :)
19:29.49Simon_cai installed the festivel deb but tsstestasterisk isn't valid - archives seem to suggest a patch.. any pointers?
19:29.53MarlowYog-home : or 60 swedish miles ..
19:30.13sxpertMarlow: what the *** is a swedish mile ?
19:30.18RoyKone 'scandinavian' mile is 10km
19:30.36Marlowyeah .. but only the swedish use that .
19:30.37RoyKwe call it 'mil' - pronounced as in 'meal'
19:30.45RoyKMarlow: and norwegians
19:30.50Marlowok så :)
19:30.57RoyKikke dansker?
19:30.57Marlowin Denmark it's not used anymore ..
19:31.04RoyKdansker er teite
19:31.04*** join/#asterisk costello (~costello@hoochie.digium.com)
19:31.06MarlowRoyK : nej ... faktisk ikke ..
19:31.29MarlowRoyK : eheh ... i don't think so ..
19:31.29RoyK:)
19:31.44voidptrsooo.
19:31.51bobmanSimon_ca: I ended up getting the deb-src and patching it from the /usr/src/asterisk/festival-1.4.2.diff
19:32.31MarlowRoyK : stadig på jobbet ?
19:32.33RoyKja
19:32.41MarlowRoyK : arghh ..
19:32.48RoyKkorrekt
19:32.56RoyKmen - stakk.. bbl
19:33.09MarlowRoyK : god tur
19:35.49*** join/#asterisk nassy (~nassy@24-193-228-121.nyc.rr.com)
19:36.18costelloHi all. Does anyone knows how to configure SIP for using G.723? I understood for G.729 I need license if I want to use.
19:36.43cypromisyeah
19:36.48cypromisfor g723.1 you need a lot of $$$$$$$$$$$
19:36.52bkw_JerJer wake up
19:36.58bkw_I gots to show you something
19:37.47jsharpWith SIP, do you need to send authentication information just if you're going to register or do you need to send it to place a SIP call as well?
19:39.39costellojsharp: I'm using SIP only to do a call from a registered user on another platform and I'm calling asterisk for playing an announcement. Each time only G.711 is used even I configured preferentially on the SIP user G.723
19:39.44Simon_cabobman: where's the patch from?
19:39.57Yog-homebah...voip-info.org DNS server isnt giving up an address for 'em
19:40.27Yog-homeboth DNS srevers down :(
19:40.38bobmanSimon_ca: The patch comes from asterisk's source.  At least, in the CVS checkout I got.
19:41.06RoyK<PROTECTED>
19:41.38costellocypromis: Can you help me to configure G.723.1???
19:41.52Yog-homehaha..voip-info.org's DNS servers are in Hawaii :)
19:42.00Yog-homesomeone must have snipped the line
19:42.09Marlownope ..
19:42.20Marlowthe guy that is running the site is located in Hawaii ..
19:42.22*** join/#asterisk nocnoc (~cvenegas@atlas.ifxnw.cl)
19:42.25Marlowif i remember correctly
19:42.43nicebubtcp        0      0 0.0.0.0:2000            0.0.0.0:*               LISTEN      745/asterisk  
19:42.48nicebubwhat is port 2000?
19:42.51nocnocanybody done this?   ATA ---> Asterisk ---> SIP Provider   using g.729 ?
19:42.56nicebubthis is from netstat -anp
19:43.09MarlowYog-home: but it's allways slow
19:43.19Yog-homeMarlow: yeh.. both name servers are the same IP too :-)
19:43.21MarlowYog-home : :(
19:43.22RoyKPostgreSQL 7.4 Released :)
19:43.24nocnocisnt port 2000/tcp *'s manager?
19:43.29blitzrageyo yo yo
19:43.32nicebubthat's what i was thinking
19:43.40nicebubespecially cause it's TCP based
19:43.44MarlowYog-home : arghh ... i think i have to offer that guy some backup dns ...
19:43.49nicebuband I know the manager API is TCP based
19:43.50bkw_no 2000 is chan_skinny
19:43.54nicebubah
19:43.54nicebubok
19:43.56nicebubtx
19:44.00blitzrageyah.. manager is like 3800 something isn't it?
19:44.02bkw_manager.conf has the ports
19:44.06nicebubok
19:44.08Yog-homeMarlow: looks like lava net having some issues
19:44.12bkw_port = 5038
19:44.13RoyK5038 is manager
19:44.21blitzrageahh.. I knew I got 38 from somewhere ;)
19:44.23Yog-homeMarlow: hehe..maybe volcano :)
19:44.29nocnocforgive my mental diarrea :(
19:44.32MarlowYog-home : good point :)
19:44.34nicebubwhat about 4569 for UDP?
19:44.35blitzrageI find voip-info.org too slow to be useable
19:44.40blitzragethat's IAX2
19:44.41bkw_iax2
19:44.43nicebubudp        0      0 0.0.0.0:4569            0.0.0.0:*                           745/asterisk    
19:44.44nicebubok
19:44.46nicebubthanks
19:44.50pattiejablitzrage: really?
19:44.55nicebubi've only been using sip, so tx
19:45.08blitzragepattieja: yah.. I don't know why.. but I find it painfully slow
19:45.08pattiejablitzrage: I saw some useful information on it
19:45.18blitzragepattieja: oh, I find it USEFUL, just very slow.
19:45.19pattiejablitzrage: well, it is slow, no doubt about that
19:45.30blitzragepattieja: I wasn't speaking of the content at all :)
19:45.37pattiejaok
19:45.48pattiejablitzrage: did you see the latest entries about echo cancellation?
19:45.59Yog-homeso anyone here using SER as a main SIP proxy and * as just a voice mail system ?
19:46.03bkw_netstat -na is faster
19:46.10blitzragepattieja: not yet.  i haven't read much on the wiki in a bit.
19:46.15pattiejaI'm wondering if our problems stem from the fact that we have an analog 2-line phone plugged into the line that goes into the X101P
19:46.35bkw_pattieja could be
19:46.37blitzragepattieja: unfortunatly, I work 45 hours a week at a place an hour away from me.. so my 40 hours a week turn into 65, and I don't have much time to play lately.
19:46.38Yog-homeyen...the "-p" option is great tho..shows the program using the port
19:47.17pattiejathat line is split at a box where it picks up another line to go to the 2-line phone and the other goes to the X101P
19:47.39pattiejabkw_: blitzrage: I don't think we've ever used the system without the phone being plugged in at the same time
19:47.45*** join/#asterisk dobey (~dobey@gateway.ximian.com)
19:47.47pattiejathe phone is not off-hook, mind you, either
19:48.10pattiejabut it is in the circuit, and it's a 900MHz cordless phone, too
19:48.33dobeyare there any API docs for libiax and possibly a specification for the protocol?
19:48.37blitzragepattieja: couldn't you unplug the phone and see if you still have the problem? (I'm not totally sure what is going on, as I just walked in the door)
19:48.58pattiejawhich puts pulsing noises into the earpiece when both lines are in use simultaneously.  It's apparently picking up noise from the LEDs that flash to indicate the lines are in use
19:49.17voidptrsomeone tell me what matt's astgui is really aimed for ;)
19:49.28pattiejablitzrage: well, currently, we've been using it to test the system.  We call from the first line to the second line's number (which is connected to the X101P)
19:49.42blitzrageahhh I see
19:49.44pattiejabut I will attempt to try it without it
19:49.52Yog-homeanyone here using * in an enterprise VOIP environment ?  like 500 or more ppl ?
19:50.00blitzrageyah.  I don't have an X101P, so I have very little knowledge of it :)
19:50.08pattiejablitzrage: and I've been going nuts trying to get the echo worked out of the system
19:50.08blitzragejust the TDM400P on loan from the school
19:50.36pattiejablitzrage: any POTS line cards? (the X101P is just a newer model than the X100P)
19:50.53bkw_not really
19:51.04blitzragepattieja: nope, I haven't used either the X100P or 101P.
19:51.10pattiejaI can never remember stuff like this, but the X10[01]P is signalled with FXO, so I think that means it's an FXS device
19:51.15pattiejablitzrage: ok
19:51.15blitzragepattieja: I only use Cell and VoIP, no landline stuff.
19:51.22voidptrwhatever it is, doesn't look like its very usable... neither operator console, nor softphone
19:51.35blitzragepattieja: the X100P is an FXO device with FXS signalling.
19:51.37bkw_no its signaled with FXS
19:51.42pattiejablitzrage: we're trying to use only softphones (VoIP) in our scenario too, but connected to a single land line
19:51.56blitzragepattieja: I also don't use softphones :) (but I have)
19:52.03blitzragepattieja: gotcha
19:52.05pattiejaahh. ok.  see I told you I can't remember these things
19:52.17blitzragepattieja: yah... always the opposite signalling
19:52.19pattiejabkw_: thanks
19:52.50blitzragepattieja: yah I know.. it's a bit confusing at first until you memorize it pretty much
19:52.56pattiejablitzrage: yeah, I've tried a smattering of X-Lite, DIAX, iaxcomm/client (going to try the latest version here soon), gnophone, linphone, kphone, etc.)
19:53.25pattiejaI even was able to get X-Lite to run under Wine on Linux
19:53.28blitzragepattieja: I've pretty much only used x-lite/pro.  I like it, but find the DTMF support a bit lacking sometimes
19:53.40blitzragepattieja: yah, I've heard it runs under Wine, but never tried so far.
19:53.54pattiejablitzrage: really?  it seems that DTMF works the best for me on X-Lite
19:54.54pattiejablitzrage: you have to have a beefy Linux box, I think, to make it sound normal.  If you have a lower powered machine, it sounds garbled and pulsating, but still understandable.  I haven't tried it on anything faster than the laptop I use which is a 600MHz PIII
19:55.12bkw_MAN the wind is blowing
19:55.34blitzragepattieja: yah, I have it on an AMD 1800.  I am probably using an older version now, and more than likely was just the way I had my * box configured :)
19:55.50blitzragepattieja: but I can use the DTMF now with x-lite and my menu system
19:55.53pattiejabut, if I get iaxclient/comm to work, that will be really sweet.  The main reason is that X-Lite isn't a true Linux app, but it's the only app that works properly with the USB headset I had to get
19:56.06pattiejacool
19:56.26blitzragepattieja: gotcha.  Yah, I think I'm going to try an iax client one of these days.  More than likely will use more of that stuff on my laptop once I get back to school
19:56.40pattiejabkw_: went to bed last night with it raining pretty badly.  Woke up this morning to reports that the basement had at least 2 inches of water
19:57.09jsharpSurfs up.
19:57.12pattiejablitzrage: DIAX for Windows looks very promising and is already at a very useable stage of development
19:57.14pattiejajsharp: :)
19:57.37blitzragepattieja: yah, I've been hearing good things about it.  I'll give it a shot at some point I'm sure.
19:58.02pattiejaalthough, it still crashes every time I exit the application
19:58.02blitzrageWORK!
19:58.08pattiejaoh, sorry
19:58.13pattiejame too
19:59.19dobeyhrmm
20:01.26voidptri dont care what soft phone works, as long as it works and uses (iax|iax2)
20:04.11*** join/#asterisk joe_satriani (~joe_satri@ppp-10-156.28-151.libero.it)
20:04.14joe_satrianihi all
20:04.28*** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-153-89.NYCMNY83.dynamic.covad.net)
20:09.16*** join/#asterisk jtodd (~jtodd@207.141.153.205)
20:09.20*** part/#asterisk ajh (~ajh@c-24-0-23-89.client.comcast.net)
20:10.13jtoddI can't reach my test machines.  Could someone please cut/paste the line(s) that show up in dmesg upon boot when a TE410P is present?
20:10.32joe_satrianiIs anybody who used auto call ?
20:10.42jtodd(i.e.: I wouldn't have to ask this if I could see my normal boxes, but I'm working in an unexpectedly odd network environment at the moment)
20:10.56*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
20:11.57jsharpI get this when I modprobe t4xxp
20:11.58jsharpZapata Telephony Interface Registered on major 196
20:11.58jsharpFound TE410P at base address fc000000, remapped to e087a000
20:11.58jsharpTE410P version c01a003a
20:11.58jsharpFALC version: 00000005, Board ID: 00
20:13.22jtoddjsharp: Thanks.  Do you see anything in dmesg during the PCI probes right at boot (before modprobes have loaded the zap drivers)?
20:13.58jsharpNope.  Nothing shows up about the card till you load the drivers for it.
20:14.21Yog-homeSIGH ->  Look for the condition syntax in examples or documentation.
20:14.21Yog-home{root@fs/pts/7}/usr/local/asterisk/etc# egrep -i gotoif *
20:14.21Yog-home{root@fs/pts/7}/usr/local/asterisk/etc#
20:15.17jtoddHm... OK, I'm used to the T1 cards which I thought I remembered saying "Found a Wildcard" in the dmesg during boot, but apparently I'm confusing that with the modprobe output.
20:16.44jsharp"Found a Wildcard" shows up in dmesg when you modprobe it...be it upon boot or at any other time.
20:18.06*** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-24-2.NYCMNY83.dynamic.covad.net)
20:19.54pattieja:(
20:20.26pattiejaiaxcomm works as a client, even seems to have some very nice features, but it still doesn't work properly with my USB headset.  It seems that only X-Lite under Wine works properly.
20:20.39Connor-what is expat ?? I'm trying to compile oh323
20:20.53pattiejaConnor-: sounds like something to do maybe with XML?
20:21.32jsharpBleh.  This PAM stuff is turning out to be much harder than I thought.
20:21.48atacommdoes * have a problem with knowing when SIP phones are online?  half the time it is saying busy when i dial into Asterisk.....
20:21.59Connor-yea, just found it.. dependancies..
20:24.44Connor-Oh frill..
20:24.46Connor-[chan_h323.so]WARNING[1074395872]: File loader.c, Line 232 (ast_load_resource): libpt_linux_x86_r.so.1: cannot open shared object file: No such file or directory
20:24.46Connor-WARNING[1074395872]: File loader.c, Line 400 (load_modules): Loading module chan_h323.so failed!
20:24.46Connor-Ouch ... error while writing audio data: : Broken pipe
20:25.10jsharpMake sure that library exists somwhere that ld.so.conf can pick it up.
20:25.15jsharpThen rerun ldconfig.
20:26.52Connor-where should that lib be?
20:27.07jsharpMines in /usr/local/lib
20:27.24jsharpThat's where I put it.
20:29.36Connor-what package does libpt_linux_x86_r.so.1 come from.. I can't find it..
20:30.09bkw_um
20:30.14bkw_dude chan_h323?
20:30.19Connor-never mind.. openh323
20:30.21bkw_you must not be following direction?
20:30.25jsharpit comes from oh323.
20:30.29jsharpEr, openh323.
20:30.31Connor-I guess I have to manully move it.
20:30.38bkw_no
20:30.42bkw_you didn't read the readme :P
20:30.45bkw_haha
20:30.49Connor-yes I did.
20:30.53bkw_no you didn't
20:31.01bkw_you don't move it.. you set LD_LIBRARY_PATH
20:31.10bkw_granted you can move it.. but thats not what the README says
20:32.00Connor-well.. It's sitting in /usr/src/
20:32.09Connor-well.. It's sitting in /usr/src/openh323/lib
20:32.18Connor-I don't run ANY lib from /usr/src
20:32.20*** join/#asterisk cfo (~cfo@194.19.190.217)
20:32.29bkw_I don't run h323 on my box anymore :P
20:32.36bkw_One Number Find Me .. now thats an app I need to write
20:32.40*** join/#asterisk Mystique (~bcook@mystique.poklib.org)
20:33.23bkw_dobey what use?
20:34.02bkw_what do you want the information for?
20:34.08bkw_or how do you wanna use it
20:34.40*** join/#asterisk raha (~raha@hoochie.digium.com)
20:35.37anthmecho /usr/src/openh323/lib >> /etc/ld.so.conf ; echo /usr/src/pwlib/lib >> /etc/ld.so.conf ; ln -s /usr/src/openh323 /usr/src/pwlib /root ; rm -fr /usr/src/openh323 /usr/src/pwlib
20:35.57joe_satrianiIs anybody who used auto call ?
20:36.05bkw_auto call?
20:36.15bkw_oh god you people are about as responsive as a rock
20:36.26anthm1 command is not part of the right way but more like what i want to do w h323
20:36.30*** join/#asterisk ktracho (~ktracho@hoochie.digium.com)
20:36.49joe_satrianiyes, if I put a script in /var/spool/asterisk/outgoing it can call a number and if it answer forward to an internal number
20:37.01bkw_easy sample.call
20:37.21gadams666anyone have mysql integrated for cdr?
20:37.26*** join/#asterisk ktracho (~ktracho@hoochie.digium.com)
20:37.34dobeybkw_: i want to notify the user of incoming calls (callerid), and voice mails, and maybe provide a simple app for playing the voicemails
20:37.37bkw_gadams666 cdr_mysql works wondrs
20:37.37joe_satrianibkw_: where i can find that file
20:37.46bkw_dobey it will email the thing
20:37.56bkw_dobey the email can include all that.. and the message too
20:37.59Mikenever sleepet so much
20:38.06bkw_cvs checkout asterisk-addons
20:38.13gadams666that's what i'm comnfiguring right now. have the db creatd, do I need to load a module in modules.conf?
20:38.25bkw_yes
20:38.26gadams666got  it, compiled it, installed it. :)
20:38.39dobeybkw_: i don't want e-mail, i want to do this as a sort-of client
20:38.58bkw_dobey write it.. thats about all you can do
20:39.11bkw_whats so wrong with the MWI on the phone?
20:39.21dobeybkw_: i want to write it
20:39.23gadams666basically a load=cdr_addon_mysql.so
20:39.28bkw_gadams666 yes
20:39.42dobeybkw_: i'm trying to ask aobut the information i can get with the protocols available for contacting the server
20:39.44Mystiquehey all, is anyone around that can answer a few basic questions for me?
20:39.46joe_satrianibkw_: wich is the cvs parameter to obtain the list of module-addon ?
20:39.52dobeylooking in iax-client.h doesn't tell me much of anything
20:39.59bkw_joe_satriani their are only two
20:40.00*** join/#asterisk Shido6 (~shido@d57-81-103.home.cgocable.net)
20:40.08bkw_they are in the same directory
20:40.15joe_satrianion is mysql_cdr and the other?
20:40.15Mikebkw_: care to help me a bit with iax?
20:40.38gadams666getting closer, "failed connect" - betting mysqld.sock not located in /tmp :)
20:40.52*** join/#asterisk bobman (~bobman@mube.psouth.net)
20:41.30ktrachoanyone knows about a full SIP SOFTPHONE for Windoze with HOLD & CONFERENCE features?
20:42.04wasimDEBUG[131081]: File chan_iax.c, Line 3553 (socket_read): Ooh, voice format changed to GSM
20:42.25wasim<PROTECTED>
20:42.29*** join/#asterisk lexluther (~admin@80.88.142.221)
20:42.29dariusktracho: just sjphone have those features?  I don't recall
20:42.30atacommbkw: are there any known problems with SIP and Asterisk where asterisk doesnt even try the phone and returns a busy?  I'm just trying to figure out if it is Asterisk, or if its the X-Lite softphone....want to make sure there aren't any issues before launching
20:42.48*** part/#asterisk lexluther (~admin@80.88.142.221)
20:42.52bkw_atacomm yes.. when you seed from the astdb
20:43.20atacommbkw_: what?
20:43.30wasimAsterlogy: -r
20:44.47ktrachoI'm gonna checkout the SJPHONE! Thx!
20:48.07Connor-okay, that's waked.. I got h323 working.. half way..
20:48.25Connor-I can call.. it connects, and works, but, I get disconnected in the middle of a call.
20:48.29atacommbkw: what do you mean?
20:51.14bkw_do this
20:51.20bkw_database deltree SIP
20:51.21bkw_restart *
20:51.45bkw_the device hasn't re-registered since restart
20:52.31atacommhmm....so its more a question of make the phone register more often?
20:52.45bkw_yes
20:52.57bkw_i have the same problem with my 7960
20:53.18bkw_when you seed the sip registrations from astdb on start.. and the phone hasn't re-registered yet.. it will come up busy
20:53.37*** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net)
20:53.45atacommwhat a pain
20:55.09atacommis that were default ip comes in handy?
20:55.29Yog-homedoes * run chrooted ?
20:55.49Yog-homeor does the System command perhaps ?
20:56.28ManxPowerYog-home: No and No, but I'm sure you can
20:56.33Yog-homeI run a System() command like "System(/bin/echo -e a bunch of text > /tmp/blah) and /tmp/blah is never created
20:56.48*** join/#asterisk WOB (~WOB@ndn-83-105.telkomadsl.co.za)
20:57.18WOBhey all
20:57.32ManxPowerYog-home: Does the *exact same command run at the command line?
20:57.47ManxPowerFor example quotes can be an issue
20:57.52WOBanyone know what would be causeing the problem if I try and compile the chan_capi 0.3.0 drivers and it brings up a [chan_capi.o] Error 1
20:58.02WOBAny help would be great!
20:58.15Yog-homeshoudln't matter in this case
20:58.24rajo_home_jtodd: what's the current state of TRIP with asterisk?
20:58.26Simon_cais it possible to set a var from a system command, ie something like exten => 1234,1,Setvar($DATE(System(date))
20:58.44n3wb33WOB: paste a bit of the error, and make sure you have i4l-devel installed ... or whatever that is called in your distro
20:58.56Yog-homeah..i put quotes around the text and it worked
20:58.57Yog-home:(
20:59.28Yog-homeugh..doesn't solve my problem though.. I'm looking for some variable I can access that references the SIP "To:" field
20:59.34Yog-homeI guess that's not available..ohwell
21:00.37Yog-homeletsee if anyone else has suggestions...
21:00.46rajo_home_~seen jtodd
21:00.48jtodd is currently on #asterisk (51m 32s).  Has said a total of 4 messages.  Is idling for 45m 31s
21:00.59jtoddwhat what?
21:01.06rajo_home_:)
21:01.11jtoddrajo_home: Nothing.  There is currently no progress.
21:01.13WOBn3wb33: Im running redhat 9 .. what is the package called for that?
21:01.32*** join/#asterisk boropest (~boropest@mail.mkda.com)
21:01.43rajo_home_jtodd: ah okay... pity,
21:01.44jtoddrajo_home: Wish there was.  I have a spark of interest from the Austrian guys doing ENUM, which I think would be cool if they put some funding and manpower behind it.  
21:01.54n3wb33WOB: i dont know nothing about red hat.
21:02.50rajo_home_jtodd: yep. we're planning doing enum with * too, but for an interconnection of several * boxes on the universities here TRIP would be handy
21:03.51Yog-homeI have a customer who wants to use * for voicemail... unfortunately, all his phones are set up using SIP "user@domain" style addresses instead of "extension@sipdomain" ... he's using SER as his SIP router/proxy... the # extensions in his dialplan  are simply aliases under SER which map to the "sip:user@domain" style addresses...when I shunt a call off to * for voicemail, asterisk doesn't see the dialed extension, but isntead of "
21:03.51Yog-homeIP invite as the dialed extension...so I either have to include a bunch of usernames as extensions, or somehow get the dialed extension # to map to a * voicemail box...any suggestions ?
21:03.53jtoddrajo_home: Imagine interconnection between twelve different providers, each with daily (hourly!) changing rates and costs.  Without TRIP, it sucks.
21:04.23*** join/#asterisk bobman (~bobman@mube.psouth.net) [NETSPLIT VICTIM]
21:04.31jtoddrajo_home: ENUM is probably sufficient if you don't have a bunch of gateways.  If you're just trying to find end users, ENUM is all you need.
21:04.51*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) [NETSPLIT VICTIM]
21:04.57*** join/#asterisk boropest (~boropest@mail.mkda.com) [NETSPLIT VICTIM]
21:05.04*** join/#asterisk Shido6 (~shido@d57-81-103.home.cgocable.net) [NETSPLIT VICTIM]
21:05.40rajo_home_jtodd: yes, we want to use enum for kind of LCR via voip from the pstn at the university, and having trip to do the connections to other universities here
21:06.28Yog-homeguess not
21:06.32jtoddrajo_home: ENUM is not designed for LCR, and I'm not quite sure how you'd implement that.  ENUM gives you 1 answer, and 1 answer only.  TRIP can give you fifteen different answers, and then costs, and all kinds of other decision criteria, and lets you make the choice yourself.
21:07.08*** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net)
21:07.21n3wb33~seen kapejod
21:07.22kapejod <~kapejod@pD9E82713.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 19h 12m 1s ago, saying: 'a bit slow for a phone ;)'.
21:07.49*** join/#asterisk lexluther (~admin@80.88.142.221)
21:07.58Connor-Why would h323 be terminating the call prematurely?
21:08.02rajo_home_jtodd: well, currently there's kind of an ENUM-hype here in house... and as this is a good way to install some *-boxes, we've taken the chance
21:08.05*** part/#asterisk lexluther (~admin@80.88.142.221)
21:09.55rajo_home_jtodd: obviously ENUM is a bit limited to do so, but it is a beginning. So TRIP is not ready yet... hmm...
21:10.17citatsjtodd: enum can do LCR assuming your DNS server handles it... quite simple actually... either build wildcard routes or do what i did and write a module for powerdns to generate records on the fly
21:10.28jtoddrajo_home: TRIP isn't merely not ready - it doesn't exist.  There is a reference implementation from the Vocal project, but I've been unable to even get it to compile.
21:10.47jtoddcitats: You're not using ENUM, you're using some back end engine that happens to answer via DNS queries.  :-)
21:11.09jtoddcitats: and wildcard routes aren't what I'd call LCR....
21:11.13citatsjtodd: i think thats what it says in websters when you lookup enum :)
21:11.17rajo_home_jtodd: ...that doesn't surprise me... compiling vocal stuff is quite... well... challenging sometimes
21:11.42Connor-Who's doing h323 ?
21:11.42citatsjtodd: if you have multiple gateways in different cities and you setup wildcard routes pointing appropriate prefixes to those gateways then that very much is LCR
21:12.14jtoddcitats: Er... no, I think that it's not quite that simple.  ENUM assumes that you can use the DNS, and a query from any location will get the same response, using the "standard" DNS caching mechanisms.  Your system is just a back end that uses an ENUM answer for what normally is a static answer.
21:12.58citatsjtodd: it doesn't assume that actually... i generate different responses based on the source, timeofday, current call load on gateways, etc.
21:13.23jtoddcitats:  Right.  That's not "pure" ENUM; that's an LCR engine that happens to answer via ENUM, as I said.  :-)
21:14.44*** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-24-2.NYCMNY83.dynamic.covad.net) [NETSPLIT VICTIM]
21:14.46citatsjtodd: ENUM is far from pure no matter how you look at it :)
21:15.35citatsthe LCR engine can be software, or a human looking at a phone book and typing in a few wildcard records
21:16.39iGNAre there any good newsgroups on CTI in general that you'd like to recommend?
21:18.34tzangerwhat is the difference between ANI and CID?
21:19.31tholoCID can be turned off.
21:19.47tholoANI is not supposed to be provided to an end user.
21:20.36*** join/#asterisk joe_satriani (~joe_satri@ppp-10-156.28-151.libero.it) [NETSPLIT VICTIM]
21:20.58denontholo: aren't PRIs really end-users, by the telco's point of view?
21:21.22tholoYeah, and not all telcos will provide ANI on a PRI...
21:21.24Connor-~seen jerjer
21:21.25jerjer is currently on #asterisk
21:21.42denontholo: but if some do .. seems like it defeats the point :)
21:22.12tholoIn some cases (like when you run an 800 service) you can subscribe to it.
21:22.22tholoI.e. when you need it for billing purposes.
21:22.29denonnod
21:23.08tholoIn other situations, if you receive it, you are still not supposed to put it up in the CID data on a desk phone.  Of course, you *can* do it... ;-)
21:24.11denonbuggy script :)
21:24.19denonsent it to caller id instead of billing .. will have to debug that later ;)
21:24.24tholoHonest offiser, it was just a bug!
21:24.29tholoOfficer, too.
21:24.44denonossifer?
21:24.55tholoHm, I like that version better.... 8-)
21:25.07denondrunken assmaster
21:25.14Yog-homehrm... in * voicemail, is the 'greet.gsm' file ever played ?  I've tried calling voicemail w/o the prefix "u" or "b" .. perhaps greet.gsm is just obsolete and the "unavail.gsm" file is used as the 'greeting' ?
21:28.37Mikesomeone can give me an example how to link with iax 2 * servers
21:28.55Mikeusing gsm between them
21:33.07Yog-homehrm..bizzarre .. I see code in app_voicemail to play greet file, but it never plays it :(
21:33.25*** join/#asterisk stonefly (~trillian@toby.stoneflytech.com)
21:37.25*** part/#asterisk rajo_home_ (~rainer@p508AEFB0.dip.t-dialin.net)
21:38.06*** join/#asterisk terry__ (~terry@adsl-68-89-230-153.dsl.spfdmo.swbell.net)
21:38.38Yog-homehm...this was working before.... <sigh>
21:39.26terry__Does anyone have a recommendation for a good (and relatively inexpensive) 4-port FXO gateway?
21:39.49dheckamankmart has some good ones
21:40.06dheckamanthey even have ones with cream filling
21:40.19terry__so helpful.  :)
21:40.23dheckaman:)
21:40.54terry__Does the cream filling cost extra?  I'm on a budget.
21:41.03dheckamanhmm
21:41.20*** join/#asterisk nocnoc (~cvenegas@atlas.ifxnw.cl)
21:41.22dheckamanI think so, its the special, fluffy cream filling...
21:41.31terry__I tried Digium, but their 4-port isn't ready yet and they don't have a release date for me...
21:41.33nocnocis there such thing as a sccp channel?
21:41.54tholochan_skinny
21:44.15Simon_cahow do you route based on callerid?
21:45.04*** join/#asterisk bobman (~bobman@mube.psouth.net)
21:45.12nocnoctholo - is that available on the latest CVS?
21:45.28Mikegui for asterisk?
21:45.42tholoYes.  It is not really production ready, however.  Apparently the funding for it went away.
21:45.52*** join/#asterisk jgaviria (~jgaviria@63.245.86.109)
21:45.55tholoSimon_ca: search for "anti-GF"
21:46.19Connor-what's the dial string for oh323 someone?
21:46.31tholoI believe it is even in the handbook...
21:46.36carrarit is
21:46.38carrarsection 4.4.3
21:46.43carrar4.3.3
21:46.45carrarsorry
21:47.02anthmDial(RTFM/${EXTEN})
21:48.31Stealth_Manis there any special settings in sip.conf for GS phone to be registered ?
21:49.16tzangerwow a lot's changed in CVS
21:50.54bkw_blah blah blah
21:51.29dheckamanhmm
21:51.35dheckamanecho on the sip side
21:51.45dheckamanbut no echo on the POTS side
21:51.56*** join/#asterisk zwi (~chris@216.88.131.43)
21:52.05Stealth_Manwtf wrong with gs and * shit... it is not allowing to me to register GS into *
21:52.07terry__simon_ca:  just in case you actually wanted an answer... route by callerid ... exten => 100/5555551212,1,Congestion would route a call to extension 100 from 5555551212 to the congetstion app
21:52.07Simon_caok, this one isn;t in the handbook. :)  What do you pass to Festival() to get it to readback the callerid, i.e. like an ani readback service.  ${callerid} doesn't work.  i guess another way of asking is how do you get the callerid into a var....
21:52.35citatsSimon_ca: try ${CALLERID} or ${CALLERIDNUM}  special variables are case sensitive
21:53.01anthmhmm maybe wtfm
21:53.02tholoAnd check README.variables in the source directory
21:53.03Connor-I'm having oh323.. someone help?
21:53.11anthmwrite the f^&%^& manual
21:53.16*** join/#asterisk GhostNr1 (~Ashmed@193.10.185.3)
21:53.59voidptrwafm, write a fine manual
21:54.15*** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net)
21:54.19Mikeif my remote server has more bandwidth should i put the iax server over there?
21:54.27n3wb33cat *.c
21:54.51dheckamancat /dev/urandom > /etc/asterisk/extentions.conf
21:54.57dheckaman:)
21:55.10sxpertdheckaman: lol
21:55.13terry__in case anyone new has logged in... Anyone have any experience with any particular 4-port fxo gateways for use with asterisk?
21:55.29tholocat /dev/urandom > /dev/hda
21:56.00dheckamanyea
21:57.15Simon_cacitats: thanks.  I tried ${CALLERID} and doesn;t work.  just tried ${CALLERIDNUM} and it does.  (as does ${CALLERIDNAME)  but ${CALLERID} itself appears undefined...
21:57.38Simon_caso calleridnum will work fine, but curious
21:58.17tzangercitats: you're there!
21:58.37tzangermalloc: dispose_cmd.c:249: assertion botched
21:58.37tzangerfree: called with unallocated block argument
21:58.41tzangerthat doesn't sound good  :-)
21:59.09tzangermalloc: stringvec.c:73: assertion botched
21:59.11tzangernor that
21:59.27Stealth_Manbkw: can you look into my sip.conf  please  ???
21:59.27Connor-Can someone help me for just a minute?  Trying to get oh323 working.. Please??
21:59.44sxpertConnor-: good luck ;)
22:00.56Connor-sxpert: Got it compiled and loading...
22:04.50UnixDawgI dont get why my incoming is not working
22:05.48UnixDawgHost                  Username    Perceived             Refresh  State
22:05.48UnixDawg66.234.228.132:4569   in-PLy97oR  192.168.0.2:60621          60  Registered
22:06.00UnixDawgwhy is it not putting it on normal ports
22:06.37Simon_caanyone know of a way to set a variable with the output of a system call?  ie Setvar(TEMP,System(wget http://localhost/cgi-bin/weather)) - I want pass it to festival...
22:07.32Connor-Well.. I got it to call netmeeting..
22:07.43Connor-How do I get it to work with CCM ?
22:08.31UnixDawgsomething is wrong here
22:08.33bkw_is it me or has all hell broke loose today
22:09.16bkw_Connor you want chan_h323 to work with CCM?
22:09.29twilsonIs there any advantage to using skinny on a Cisco 7960 as opposed to SIP when using asterisk?
22:09.31Connor-no. chan_oh323
22:09.37tholoNo, no, he is using the*other* H.323 implementation!
22:09.42bkw_twilson not yet
22:09.48bkw_chan_skinny needs alot of love
22:09.53Connor-chan_h3h don't work with ccm
22:09.56bkw_nope
22:09.58bkw_was aobut to say that
22:10.08bkw_the rtp stream changes once CCM sets the call up
22:10.17Connor-right, and JerJer won't fix it. :
22:10.19Connor-:)
22:10.22twilsonso far I don't have a lot of love for chan_skinny... :-)
22:10.29bkw_Connor well you could fix it
22:10.40bkw_JerJer has bigger issues to deal with
22:10.42Connor-Don't know a thing about h323 stuff.
22:11.48anthmchan_?h323 + netmeeting has 1% chance of working if you manually set the codec in both to ulaw
22:12.01Connor-Well.. oh323 works with netmeeting.. wonder why it's not working with the CCM
22:12.13anthmh323 and netmeeting both belong together at the bottom of the ocean floor
22:12.25twilsonI have four analog lines that I need to hook up to an asterisk box... I don't have room for 4 FXO cards, and a t1 card/channel bank seems a little excessive.  Anyone have any recommendations?
22:12.31anthmconsider them both voip roadblocks
22:12.32Connor-I don't care about netmeeting.. just trying to get it to work with CCM
22:12.42bkw_CCM is shit
22:12.45bkw_pure shit
22:12.57bkw_poop
22:13.14Connor-come on guys.. I'm not here to debackle or argue of what's hot and what's not.. I just need to get the shit to work.
22:13.37anthmhelping to get it to work prolongs its lifespan , no ?
22:13.51bkw_HAHA
22:13.54bkw_~google bkw asterisk
22:15.12Connor-can't get debugging to work for oh323
22:15.51anthmwhat does CCM stand for?
22:15.59jetscisco callmanager
22:15.59Connor-Cisco Call Manager
22:16.41anthmmost h323 issues come from codec mismatch and nat
22:18.00anthmeveryting seems to speek ulaw so you always get that working 1st
22:18.11anthmbyt making it the only codec in the list
22:20.59*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
22:22.21*** join/#asterisk RickColl (~RickColl@Toronto-HSE-ppp3832260.sympatico.ca)
22:24.15*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
22:25.11bkw_http://www.tric.nl/~cg/asterisk.html
22:25.18bkw_HOLY SHIT.. almost 30k lines
22:25.19bkw_haha
22:25.49tholoYeah -- you just keep on talking, and talking, and talking, and...
22:26.21Mikemike (1593)
22:26.24ExomorphHe also splits his lines up alot. :)
22:26.25Mikedidnt make it at the top
22:26.25Mikelol
22:26.33tholoYeah,
22:26.34tholothat
22:26.35tholois
22:26.36tholoone
22:26.36Mikeim an amateur
22:26.37tholoway
22:26.43tholoof getting lots of
22:26.44tholoentries
22:26.45Mikei think i have to type even more
22:26.46Mike:P
22:27.18Stealth_Man---- >kram is either insane or just a fair op, kicking a total of 3 people!
22:27.20Stealth_Manhehe
22:27.21tholoMe, I prefer to type sensible stuff elsewhere -- typically code...
22:27.31Mikebkw_: thats because i havent been here the 74 days next time ill be on high score
22:27.42bkw_NEVER
22:27.53pbxtechmike who?
22:27.55pbxtech:)
22:27.56bkw_Nobody will be able to top me.. I spent 18 hours + per day in front of this computer
22:27.57bkw_JUST TRY ME!
22:28.00ExomorphLOL     Take a look at this... (from the page bwk sent)
22:28.00ExomorphPoor bkw_, nobody likes him/her. He/She was attacked 27 times.
22:28.15bkw_pbxtech how does it taste?
22:28.17bkw_:P
22:28.20pbxtech:P
22:28.44bkw_Stealth_Man what were you jumping up and down about ?
22:28.47bkw_some sip.conf?
22:28.56bkw_put it on the web.. msg me the URL if you want me to look at it
22:28.59Stealth_Man:))))))))
22:29.13Stealth_Manbkw : yes i will
22:29.22Stealth_Manbecausei have stupied erorrs i can;t figure out anything :( shitt
22:29.30Mikebkw_: they sell lifes for about 3.99 on any pharmacy:P on packages of 3 for use with girls
22:29.31Mike:P
22:30.39Takapawhat time is it at digium's now?
22:30.48bkw_its 4:30 pm @ digium
22:30.49bkw_isn't it?
22:30.54bkw_are they CST?
22:30.54Takapahm.. k
22:31.00bkw_or EST
22:31.24*** join/#asterisk SirGaia (~ircap75@217.Red-80-37-235.pooles.rima-tde.net)
22:31.32malcolmdCST
22:31.46SirGaiaHi all
22:31.57coolhpWould anyone happen to know if Digium is still working on the 5v version of the TE410P?
22:32.28n3wb33mark said a couple of days ago that he had one of those working
22:32.31Stealth_Manbkw:  can i copy/paste sip.conf to you ?
22:32.45coolhpThanks n3wb33.
22:33.54Shido6any news on a Japanese version of the TE?
22:34.45bkw_so help me Stealth_Man you paste me that config  again.. i'm gonna reach thru IRC and smack them teeth out your head!
22:34.50bkw_:>
22:35.38bkw_put it on a website or use a paste bin
22:35.43bkw_we need an * paste bin
22:35.57Stealth_Manok .... hold on :) we will have it ;-)
22:37.59Yog-homehrm...when an AGI script prints to STDERR, where does the results go ??
22:38.23n3wb33Yog-home: to the *CLI
22:38.23Stealth_Manbkw: http://asterisk.xvoip.com/bin/
22:38.26bkw_should go to the cli
22:38.36Yog-homehrm..don't see it...do I have to have debug on ?
22:39.09Stealth_Manbkw: I will  make open access to this, so people will be able to upload conf :) it iwll be Asterisk Paste Bin :))
22:39.11citatsYog-home: they only go to the main * cli, not a remote
22:39.20Yog-homeoh...
22:39.29Yog-homeso asterisk -r dun work
22:39.36citatsuse the AGI VERBOSE command to send stuff everywhere
22:40.07Yog-homeis there any way to get a 'main' CLI up w/o just running asterisk w/ -c ?
22:40.19*** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-152-122.NYCMNY83.dynamic.covad.net)
22:40.22bkw_start asterisk with safe_asterisk then asterisk -r
22:40.25levonnite people
22:40.39citatsYog-home: nope
22:40.47Yog-homehrm...i'm in a "asterisk -r" already and the messages don't show
22:40.49Yog-homeoh..ok...
22:41.05citatsYog-home: because STDERR goes to STDERR, and STDERR isn't duped to remote consoles
22:41.10Yog-homeyep
22:41.13Yog-homemakes sense
22:41.15*** join/#asterisk daecon (~daecon@h-66-167-169-238.DNVTCO56.covad.net)
22:41.18*** part/#asterisk daecon (~daecon@h-66-167-169-238.DNVTCO56.covad.net)
22:41.24*** join/#asterisk daecon (~daecon@h-66-167-169-238.DNVTCO56.covad.net)
22:41.31Yog-homeI could probably redirect STDERR to a file or something when I start * I guess
22:42.06Yog-homecitats: what is the 'agi verbose' command..an AGI command or console command ?
22:42.07*** join/#asterisk dnc (~duncan@213.240.59.220)
22:42.43citatsYog-home: its an AGI command... use 'show agi' in the * cli to see all the available AGI commands
22:43.18UnixDawgok this is weird my outgoing is working fine
22:43.27UnixDawgbut my iax incoming is not
22:44.25anthmI added a new section in musiconhold.conf [custom_exec]
22:44.31UnixDawg66.234.228.132:4569   in-PLy97oR  192.168.0.2:61190          60  Registered
22:44.31UnixDawg*CLI>
22:44.36anthmdefault => /usr/local/bin/ogg123,-q,--audio-buffer,2,/tmp/1.ogg,-d,wav,-f,-
22:44.45anthmthat example lets me play ogg file in moh
22:44.46UnixDawgits not putting me on the correct port
22:45.01anthmand does not disturb existing functionality whatsoever
22:45.10UnixDawghmm
22:47.02*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
22:49.01denoncitats: you around?
22:52.09*** join/#asterisk joe_satriani (~joe_satri@ppp-210-148.28-151.libero.it)
22:56.36joe_satrianito activate cdr_mysql I need to make it only ?
22:57.37*** join/#asterisk mortck (~test@p213.54.146.115.tisdip.tiscali.de)
22:57.46UnixDawgyou need the add-ons I believe
22:58.01bkw_its cdr_addon_mysql.so now
22:58.15joe_satrianiI ve yet cvs it
23:00.37*** join/#asterisk cypromis (~michael@217.11.142.161)
23:04.41*** join/#asterisk Titoon (julien@deepblue.titoon.net)
23:05.24marcus-heheh
23:05.40marcus-so if you only have a credit of $0.50 left of your voicepulse connect account
23:05.49marcus-and you make a call that is longer than $0.50
23:05.52marcus-it wont cut you off
23:06.23bkw_marcus- good to know
23:06.32Stealth_Manheh
23:06.37Stealth_Mannice .. terrible
23:07.00UnixDawggrrr
23:07.06Stealth_Manbkw: so what migth be wrong  in conf ?
23:07.07UnixDawgok somethign is wrong here
23:07.09bkw_Current: $6.05
23:07.10bkw_Pending:  $0.00
23:07.17UnixDawgand I think its on thier side now
23:07.20bkw_I got 6.05 left
23:07.45UnixDawgwif who
23:07.52bkw_voicepulse
23:07.52marcus-woah, thats amazing
23:07.59marcus-i just had a *good* customer service experience with att wireless
23:08.03UnixDawgI have 15 bucks left
23:08.06Stealth_Manbkw: I will give you free service for life !
23:08.10marcus-they must be scared about the whole number portability thing
23:08.10bkw_marcus- OMG really?
23:08.14marcus-my phone broke last night
23:08.15bkw_haha
23:08.21Stealth_Manmake me working :)
23:08.23marcus-i called them today to see if it was still under warrant, and they said it was out by a month
23:08.36marcus-but that i could get a new phone for free if i sign another 1yr contract
23:08.44*** join/#asterisk Gazzas (Gazzas@c211-28-128-147.eburwd3.vic.optusnet.com.au)
23:08.47marcus-of course, i've had attws for over 5 years now, and i really like not being under contract
23:09.05Stealth_Manmarcus: ask them for optional insurance for your cell. phone ... it gonna cost you 1$ i think amonth
23:09.10UnixDawg*CLI> NOTICE[135322624]: File chan_iax2.c, Line 4537 (socket_read): Host 66.234.228.132 failed to authenticate as voicepulse
23:09.10bkw_anthm what did you do ?
23:09.16Stealth_Manafter 1 month you can call and ask for new phone
23:09.19UnixDawgvoicepulse is screwed again
23:09.23Stealth_Manthey allow 3 phone replacements ...
23:09.30anthmI added a new sectinon
23:09.31marcus-so i said "well, if i'm going to have to sign a new contract, i'm going to research what your competitors are offering first"
23:09.37anthm[custom_exec]
23:09.43anthmthat lets you set up your own pipe
23:09.51marcus-suddenly i hear some typing, and then "oh, well, you've been a customer for a long time and we appreciate that.  i'm sure i can work something out with the warranty dept"
23:10.10anthmallowing you to use any app to play back
23:10.25marcus-ud; i just made a 10 minute call via vp
23:10.33bkw_UnixDawg mine works fine
23:10.34anthmhere is an example entry defining a default channel
23:10.38anthmdefault => /usr/bin/mpg123,-q,-s,--mono,-r,8000,-b,2048,http://64.236.34.97:80/stream/1014
23:10.54anthmsticking a shoutcast into moh
23:10.57bkw_anthm their is already a patch on the net to make that possible
23:11.09anthmand here is one doing ogg
23:11.10*** join/#asterisk dmh (~dmh@ool-44c22549.dyn.optonline.net)
23:11.10anthmdefault => /usr/local/bin/ogg123,-q,--audio-buffer,2,/tmp/1.ogg,-d,wav,-f,-
23:11.25dmhdigium come out with 4port fxo yet?
23:11.25Stealth_Manbkw : are taking free long distance from me or not ?:)
23:11.29bkw_http://bugs.digium.com/bug_view_page.php?bug_id=0000413
23:11.30anthmthis makes everything possible not just 1 thing
23:11.42bkw_anthm oh sure make it where stupid people can break it
23:11.53bkw_haha
23:12.12UnixDawgbkw pvt for 1 min
23:12.14anthmyou have a total front end to the exec call
23:12.32bkw_UnixDawg not right now.. just keep it in the channel
23:12.44anthmso you never have to touch src again no matter what you may need any app that can deliver the right format can be loaded
23:13.01anthm<PROTECTED>
23:13.13bkw_anthm better test and provide examples..
23:13.14bkw_do this
23:13.18bkw_post it to bugs.digium.com
23:13.19*** join/#asterisk zonyl (~newbie@CPE-65-31-131-186.wi.rr.com)
23:13.20dmhyou cant compete with the traditional pbx until the 4port fxo becomes available at a lower cost
23:13.23bkw_with documentation
23:13.24dmhi do say
23:13.35bkw_dmh yes you can
23:13.43dmhstate how
23:14.01dmhyou would need a large backplane
23:14.06bkw_most PBX's ie Partner ACS's have only 3 FXO's and 8 extensions
23:14.09anthmhehe i was gonan do that with my extensive app_queue patches till I realized it was more typing than making the hacks
23:14.11dmhand that architecture is not cost efective
23:14.19anthmthis one is pretty easy to document tho
23:14.35dmhbkw, the smallest port CO card you can get for a traditional phone system is 4
23:14.47zonylis there a IAX lib for perl for establish receive calls?
23:14.49dmhand those are like $90
23:14.56bkw_dmh you can still compete just try... 4 lines per box.... or a T1 card and channel bank
23:14.58bkw_still easy
23:15.09dmhchannel banks are not cheap
23:15.19bkw_not too expensive
23:15.21dmhthe 4 port fxo is key
23:15.24dmhit must be made!
23:15.28bkw_it will be
23:15.29dmhcheaply that is
23:15.34dmhi know
23:15.35dmhi cant wait
23:15.36dmhheh
23:15.37UnixDawgits still failing to auth in inbound
23:15.48UnixDawgyet when I register it works fine
23:15.54Stealth_Manunix: and i still have no luck with  even outgoing ...
23:16.25UnixDawgstealth my out going works
23:16.28dmhi heard rumor its under aproval
23:16.31UnixDawgI have tested it
23:16.33dmhor somethin
23:16.57dmhit has been made, just not yet released
23:20.07Stealth_Manehh
23:21.25bkw_blah
23:21.54UnixDawgmakes no sense that its not registering on inbound
23:22.33bkw_inbound is diffrent
23:22.39bkw_bet they need to fixor your stuff
23:23.12Yog-homeok..what are the arguments to the AGI app ?  I expected AGI(script,arg0,arg1,arg2) to show up in $ARGV[0..N] ... but it doesn't..what gives ?
23:23.25bkw_use Asterisk::AGI
23:24.05Yog-homeugh
23:24.21Yog-homeI have to get another damn package :)
23:24.57bkw_Asterisk::AGI is all you need baby
23:25.06Yog-homekinda funny that the agi-test.pl file that comes w/ * doesn't use it
23:25.34bkw_Asterisk::AGI makes it simpler to work with it
23:25.36Yog-homeI'm writing an AGI to 'reverse map' a SIP user name to an extension alias using the SER aliases table :-)
23:25.55Yog-homethen I can map sip:user@domain -> extension :)
23:26.16Yog-homeall this just to get VM working :)
23:26.39Yog-hometoo bad my customer had to use user names instead of just extension numbers for his users :-(
23:27.02Yog-homebut that's kinda what SER assumes anyway... SIP addy doesn't have to be a phone #
23:27.15bkw_doesn't have to be in * either
23:27.41Yog-homeyeh..but I just don't feel like creating a bunch of extensions in the form of "username"
23:27.47Yog-homei could do that I guess
23:28.03*** join/#asterisk TerraByte (~paul@as3.dm.egate.net)
23:28.07Yog-homebut it kinda scares me..what if one of the users is named "s" or "i" or "h" :-)
23:28.18TerraByteIs there some reason when * emails VoiceMails the last 2s seems cut off every time?
23:28.50Yog-homephone is probably cutting off audio when it gets the BYE
23:29.07TerraBytePlaying through VoiceMailMain is fine
23:29.34Yog-homehrm...
23:29.43Yog-homemaybe it's bundling it up wrong
23:30.50TerraByte?
23:32.27Takapais it correct to assume that I would have to define ZAPATA_NET when compiling zaptel to be able to push "raw" data connections over a TE410P?
23:32.57Stealth_Mananthm: what this patch is doing ?
23:33.30*** join/#asterisk ionix (~ionix@MTL-ppp-150366.qc.sympatico.ca)
23:33.37dougheckawow, 256 megs of DDR 2100 crucial memory for 37 bucks
23:33.42Stealth_ManYog-home: are you using SER ??
23:33.46*** join/#asterisk scromp (~bnaylor@haybaler.sackheads.org)
23:33.55anthmthe patch lets you pick the whole exec string to spawn the music player
23:34.08anthmmeaning you can use any music player you can get to work
23:34.28bkw_wav49 is gsm kinda with a riff header so windows will play it
23:34.29dougheckaNOO, WHY do they do this!?!?! I order something, and they freaking mark it down 5 dollars for a one day sale the day AFTER I order it!!!
23:34.38bkw_windows doesn't like it and cuts off the last two seconds
23:34.42bkw_with a nasty error message
23:34.50bkw_TerraByte send in gsm and use quicktime to play them
23:36.02Yog-homedoughecka: your order pushed their profits over the to so they could do the mark down
23:36.08dougheckabah
23:36.11dougheckait was 100 dollars
23:37.47marcus-damn. i'd forgotten just how lame it is to be an oper on efnet
23:38.30TerraBytebkw: I switched to wav
23:38.33TerraByteit seems to work
23:39.17drgalaxymarcus-: what made you remember?
23:39.53dougheckadrgalaxy: they just gave it back to him
23:39.54doughecka:)
23:40.05marcus-oh someone opered me again last night
23:40.09marcus-its been a few years
23:40.56*** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net)
23:42.02dougheckaBAH
23:42.15dougheckaand they are selling my mp3 player for the same price I got it off ebay
23:42.18dougheckawithout shipping
23:42.25dougheckabrandnew
23:45.58marcus-With few options left, ambulance crew members cut the telephone off at the base and took it and Fleming to St. Mary's Hospital.
23:46.00marcus-doh!
23:46.17doughecka?
23:46.50Marlow?
23:46.52drgalaxydoughecka: are you new to being a consumer?  every time you buy a computer it gets cheaper later on
23:46.55Marlowsick phone ?
23:46.55marcus-http://www.cnn.com/2003/US/Midwest/11/18/offbeat.payphone.ap/index.html -- EAST ST. LOUIS, Illinois (AP) -- A man and a pay phone were rushed to a hospital after he got his finger stuck in the coin return slot while trying to retrieve his 50 cents.
23:47.37*** join/#asterisk onixx (1000@CPE0040f47145d1-CM014320024117.cpe.net.cable.rogers.com)
23:47.42dougheckaLOL
23:47.46*** part/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net)
23:48.17bkw_OMG
23:48.20bkw_got his finger stuck
23:48.40dougheckawhat was he doing? reaching up inside the thing?
23:48.49onixxhi all... anyone can explain if possible to park a call with an ata-286 ?
23:48.50*** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net)
23:49.02bkw_onixx # transfers
23:49.04bkw_are the only way to do it
23:50.30marcus-read the article :P
23:50.49onixxanything special to configure ?
23:51.00dougheckayou need spoons
23:51.56onixxbkw_: when I press # nothing happens
23:51.58bkw_anthm do you know how to make unified diff files?
23:52.10bkw_show application dial    optiosn T and t
23:54.00onixxbkw_: ok ... got it. thanks. do you know of any other features you can do with # or * ?
23:55.04*** join/#asterisk FuzzyCat (~ScaredyCa@c44095.upc-c.chello.nl)
23:55.09Yog-homeWarning: Cannot install Asterisk::AGI, don't know what it is.
23:55.10Yog-homeTry the command
23:55.18Yog-homegrr...it's not in CPAN
23:55.30marcus-no, its not
23:56.32Yog-homeok..can anybody tell me where the argumetns that you pass in an AGI command go ?
23:56.45*** part/#asterisk Chris_DE (~Chris_DE@p5083053C.dip0.t-ipconnect.de)
23:57.23Yog-homeanthm: just cat them together, no ?
23:57.44marcus-uhm, they are put in *argv[]
23:57.57n3wb33diff -u -r should do
23:58.02Yog-homeyet, when I print @ARGV it doesn't show
23:58.36marcus-what does it show?
23:58.43marcus-blankness, or a number?
23:58.44Yog-home""
23:58.51Yog-homeblank
23:59.01marcus-oh, strange.  what does your extensions.conf entry look like?
23:59.19Yog-homeAGI(script.agi,arg1,arg2,arg3)

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