IRC log for #asterisk on 20080420

00:11.48jameswf-homewindoze people http://en.poderosa.org/
00:16.34drmessanoOhhhhhhh... latest release: "Before Vista"
00:16.42drmessanoSounds.. maintained
00:21.50*** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au)
00:31.46jameswf-homedont blend me bro http://www.willitblend.com/videos.aspx?type=unsafe&video=stungun
00:34.42tzangerheh I love that company
00:35.09*** join/#asterisk joobie (n=joobie@joobie.org)
00:39.37jameswf-homewe may push some hardware their way... will it blend? your pbx...
00:40.08jameswf-homewill it blend youe sip phone
00:40.21drmessanoIt will blend IAX2 and SIP
00:40.25drmessanoNow THAT is a task
00:44.21jameswf-homethey dont air the videos that fail
01:04.17*** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net)
01:04.56Micko113I just installed trixbox, but can't access GUI, does it needs to be enabled somewhere
01:04.58Micko113?
01:05.54jameswf-home~trixbox
01:05.55jbot[~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
01:06.23Micko113ok, thanks
01:07.53[TK]D-Fenderjameswf-home, that was AWESOME
01:12.32RypPnjameswf-home: I liked the iphone, the marbles and the bic lighter ones, but thnx for distracting me for an hour, very entertaining :D
01:14.48*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
01:14.57*** join/#asterisk drdrain (n=drdrain@cpe-066-057-105-080.nc.res.rr.com)
01:17.17*** join/#asterisk dacs (n=haiger@unaffiliated/dacs)
01:23.29*** part/#asterisk dacs (n=haiger@unaffiliated/dacs)
01:26.10*** join/#asterisk dacs (n=haiger@unaffiliated/dacs)
01:31.18*** join/#asterisk dacs (n=haiger@unaffiliated/dacs)
01:34.48*** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp)
01:52.17Micko113Ok, I'm starting from scratch
01:52.42Micko113What should I download first in order to run Asterisk?
01:54.42*** join/#asterisk unpaidbill (i=bill@66.90.73.20)
01:55.38unpaidbillanyone have some info on using pocketsphinx with asterisk? i have sphinx2 working fine but i read that pocketsphinx has better performance and wanted to try it out
01:55.53jblackMicko113: That's easy.
01:55.56jblack~book
01:55.56jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
01:56.02jblackStart with http://downloads.oreilly.com/books/9780596510480.pdf
01:56.30unpaidbilland the forums appear to be down at the moment :(.  jsut got it all compiled but i was hoping for a crutch!
01:59.51jblackhuh. never heard of sphinx
01:59.52Micko113jblack: Thanks, looks like I need CentOS installed first
02:00.58unpaidbillit's good stuff jblack, sphinx2 is the one you want for simple integration into asterisk, there is/was a really good howto at http://turnkey-solution.com/asterisk-sphinx.html (it's down at the moment but linked off of http://www.voip-info.org/wiki-Sphinx)
02:01.46unpaidbilli have sphinx2 working great on a 500mhz geode but wanted to try out pocketsphinx just because of the performance and accuracy increases
02:18.18_ShrikEaybd436h
02:24.17[TK]D-Fenderunpaidbill, what platform are you running exactly?
02:24.46unpaidbillalix3c3 board running centos 4.6
02:24.50unpaidbilli586
02:29.30*** join/#asterisk jicksta_ (n=jicksta@75-101-5-48.dsl.static.sonic.net)
02:31.40[TK]D-Fenderunpaidbill, yeah, I've started seeing that name around.  I've been looking at the Soekris NET5501 personally.
02:32.46unpaidbillsoekris look interesting too, i have used pcengines boards for a long time though so i stick with them.
02:34.29[TK]D-Fenderunpaidbill, I'm only just getting my feet wet in the "embedded" field.  Soekris has IDE/SATA, 4 nics, and 1 full PCI to its benifit so far...
02:34.55[TK]D-Fenderunpaidbill, looking to build a router out of it (Sangoma S518) and a few other things....
02:35.03unpaidbillwhat's that board cost?
02:35.28[TK]D-Fenderunpaidbill, http://www.soekris.com/net5501.htm
02:35.42[TK]D-Fenderunpaidbill, https://www.soekris.com/shop/product_info.php?products_id=129
02:35.55unpaidbillahh
02:36.29[TK]D-Fenderunpaidbill, yeah, the alix looks a lot cheaper.  Featureset differs of course.
02:36.30unpaidbillyeah.. i paid 170 for my alix3c3 including shipping, but it only has 1lan and 256mb ram
02:36.49[TK]D-Fenderunpaidbill, both seem fair and jsut depends on your needs
02:36.58unpaidbillyup, it's very fun stuff
02:37.10unpaidbillalso, there's a minipci 4port analog board :D
02:37.16unpaidbill2fxo/2fxs
02:37.27unpaidbillalthough it's kind of expensive at the moment..
02:37.49unpaidbillhttp://www.wildix.com/product_info.php?products_id=2704
02:37.50[TK]D-Fenderunpaidbill, sounds neat, but funky for wiring & casing
02:38.17unpaidbillyes, you would definitely need a custom case for it
02:38.22*** join/#asterisk mgdm (n=michael@serenity.mgdm.net)
02:38.34[TK]D-Fenderunpaidbill, $ for that card?
02:40.08[TK]D-Fenderunpaidbill, basically its a split up TDM400P.  Thats one aspect I was thinking about the Soekris for as it has a full PCI slot.
02:40.36unpaidbillcouple hundred i think, not sure.  when you log in it says 'only for resellers' but there was a post on the trixbox forums that said somewhere around 300
02:40.51unpaidbillyeah the alix1c has a pci slot too, but still only 1lan and 256mb ram
02:41.01[TK]D-Fenderunpaidbill, ok, proportionate to the standard version at least.  Good thing
02:41.08unpaidbillim gonna have to go buy one of those boards now
02:41.34[TK]D-Fenderunpaidbill, Which model has the full PCI?  or are you talking about going through an adapter?
02:41.42unpaidbillalix1c has full pci
02:41.44[TK]D-Fenderunpaidbill, thorough the MiniPCI?
02:41.59unpaidbillExpansion: miniPCI + 3.3V PCI + LPC + optional I2C
02:42.01unpaidbillhttp://pcengines.ch/alix1c.htm
02:42.17unpaidbillthat board is 13* too
02:42.37*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
02:43.02unpaidbilli picked one up with a crappy little miniitx case, the bad part is that you cant pxeboot it
02:43.17unpaidbillthe alix3c3 you can pxeboot.. i dont know why the hell the 1c wont do it, it has lan in the boot priority
02:43.28unpaidbilli had to connect a usb cdrom to it for the install :/
02:44.58unpaidbilldigium needs to get their page back up, i need to do some investigating!
02:45.01[TK]D-Fenderunpaidbill, I haven't actually had a chance to play with anything like this..... guess I'll be finding out real soon...
02:45.08unpaidbillit's really easy
02:45.29unpaidbillread up on pxe booting (simple dhcp/tftp configuration)
02:45.34unpaidbillthat's the easiest way to install
02:45.56[TK]D-Fenderunpaidbill, will do, thanks for the tips.
02:45.58unpaidbillif you have a 2GB CF card you can put centos 4.6 on it (centos 5.X doesnt support i586 for install)
02:46.21[TK]D-Fenderunpaidbill, couldn't you just iamge it over?
02:46.24*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
02:46.25[TK]D-Fenderimage*
02:46.26unpaidbillotherwise voyage linux or imedia linux work fine, at least for the pcengines boards.. i assume for soekris since they're pretty much the same thing
02:46.40[TK]D-Fenderunpaidbill, Install base on a real HD and overwrite the CF?
02:46.48unpaidbillyeah you can do that too, i installed linux on a usb thumbdrive using a desktop pc and put it on the alix1c board
02:47.00unpaidbilland yeah you can do that too as long as you remember to bootstrap the CF card
02:47.03unpaidbillheh
02:47.16[TK]D-Fenderunpaidbill, I have plenty of mistakes left to make :)
02:47.22unpaidbillbest way to learn!
02:47.38[TK]D-Fenderunpaidbill, I prefer the "quality guided tour" method myself
02:47.45unpaidbillhaha
02:49.49unpaidbillit's pretty amazing how much power they can pack into the footprint
02:50.00[TK]D-Fenderunpaidbill, http://www.mini-box.com/Alix-2B-Board-3-LAN-3-MINI-PCI_1;jsessionid=0a01074e1f4362a874cd9adf43cf9bcf93cff3469f51.e3eTaxeKbh0Te34Pa38Ta38Sc3j0?sc=8&category=19
02:50.09[TK]D-Fenderyikes, big url
02:50.14unpaidbilli'm waiting for a board with firewire to come out so i can hook up a webcam that doesnt totally suck, usb webcams on linux are horrible
02:50.20[TK]D-Fenderunpaidbill, 3 NIC SUPER cheap
02:50.35unpaidbillyeah, not bad at all
02:50.48unpaidbillthe pcengines guy is actually really cool too, he will respond to your questions if you email him
02:50.50unpaidbillnot sure about soekris :)
02:51.26unpaidbilli asked some dumb questions, too.
02:52.19unpaidbilli think pocketsphinx is going to cause me to lose my mind
03:04.05*** join/#asterisk edwin_quijada (n=macaruch@tdev254-210.codetel.net.do)
03:04.20edwin_quijadaHi!
03:04.51edwin_quijadaI did a simple AGI to connect with asterisk but I dont know to which extension to call for listening
03:05.46edwin_quijadaI saw how to put in extension but I dont know what to do in sip.conf
03:06.46[TK]D-Fenderedwin_quijada, Umm... AGI's don't connect with Asterisk,  your dialplan calls your AGI
03:07.03[TK]D-Fenderedwin_quijada, and AGI has nothing to do with SIP.CONF
03:08.26*** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk)
03:09.19edwin_quijadaI put this in dialplan exten => s,1,Answer
03:09.19edwin_quijadaexten => s,2,Wait,1
03:09.19edwin_quijadaexten => s,3,AGI,script.agi
03:10.09edwin_quijadabut i dont know to which extension dial to use
03:10.27[TK]D-Fenderedwin_quijada, You just MAD an extension, and called it "s".
03:10.34[TK]D-Fenderedwin_quijada, You don't even know WHY you chose that?
03:10.41[TK]D-Fendermade*
03:10.54[TK]D-Fenderedwin_quijada, And what is your AGI supposed to do?
03:11.54edwin_quijada[TK]D-Fender: just read a number,get data, and say this number find in a database and say a quantity
03:12.14[TK]D-Fenderedwin_quijada, well call your AGI somewhere useful in your dialplan then.
03:12.55edwin_quijadaso I change s for extension number like 2030
03:13.13edwin_quijadaso we dial 2030 to get the AGI?
03:13.34[TK]D-Fenderedwin_quijada, that would do it.
03:14.10edwin_quijadai dont do anything in sip.conf?
03:15.00[TK]D-Fenderedwin_quijada, what does any of that have to do with sip.conf?
03:16.00[TK]D-Fenderedwin_quijada, * doesn't care where your call comes from.  Just make sure that it will land in a context that will allow them to call your AGI.
03:18.00edwin_quijadai thought that all extensions must be defined in some place
03:18.24*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
03:18.38*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:19.27[TK]D-Fenderedwin_quijada, Go reread the book and the chapter on Dialplan Basics.  You don't seemed to have grasped much from that.
03:19.41[TK]D-Fenderedwin_quijada, And its the absolutely most important part of *.
03:21.13obnauticusdrmessano you there?
03:26.01edwin_quijada[TK]D-Fender: Ok :(
03:49.09JayTee52~pastebin
03:49.10jbot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
04:02.27*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
04:02.39*** join/#asterisk Raiderman (n=raider@ip-151-155.tricom.net)
04:14.52Raidermanhi guys
04:14.56Raidermanany one can help me
04:15.23*** join/#asterisk Darthclue (n=Darthclu@76-233-19-118.lightspeed.snantx.sbcglobal.net)
04:15.36[TK]D-FenderRaiderman, ask a real question and you might get a real answer
04:16.38Raidermani install slackware 12 and install asterisk and when i try to run it it said me this "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist ?)
04:17.28TJNIIIs asterisk atarted?
04:17.43Raidermandefine atarted?
04:18.02TJNIILije, has an RC script started or have you started it?
04:18.10TJNIIs/lije/like/
04:18.14Raidermanno
04:18.24TJNIIWell, that's your first problem
04:18.26Raidermani do asterikast instalation
04:19.21[TK]D-FenderRaiderman, well I guess you haven't actually started * yet
04:19.39[TK]D-FenderRaiderman, do "safe_asterisk &" and it should read you back a PID #
04:19.55[TK]D-FenderRaiderman, If it does then you can try to reconnect to it.
04:20.15[TK]D-FenderRaiderman, because what you were trying to run is meant to conenct to a running * process, yet there was none.
04:20.55Raiderman[TK]D-Fender, the reply was [1] 2712
04:21.12[TK]D-FenderRaiderman, fine, now try "asterisk -r"
04:21.40Raiderman[TK]D-Fender, ok now it work what should is do to solve the ishue
04:22.10[TK]D-FenderRaiderman, there is no issue, you just have no idea how to RUN *.
04:22.11JTRaiderman: in future start asterisk before attempting to reattache to its console.
04:22.22JTs/reattache/reattach/
04:22.41[TK]D-FenderRaiderman, to start it you do what I first told you to do.  When its running, THEN you can connect to it.
04:23.03Mavviewhat a lousy piece of software...
04:23.11Mavviecan't even start itself when you want it!
04:23.20MavvieDo as I want, not do as I say!
04:24.44Mavviethat reminds me, I have a rewrite of streamplayer which does work properly with streams which are converted in batches.
04:24.57MavvieWonder if people are interested in it.
04:31.26Raiderman[TK]D-Fender, the situation get worse
04:31.37[TK]D-FenderRaiderman, worse?
04:35.19Raiderman[TK]D-Fender, when i execute safe_asterisk -U asterisk -G asterisk i get an error "asterisk ended with starus 1 asterisk died with code 1 mpg123: no prosess killed "
04:35.44[TK]D-FenderRaiderman, well you shouldn't be trying to run it that way TWICE
04:35.57[TK]D-FenderRaiderman, Once its running you don't call safe_asterisk again.
04:36.07[TK]D-FenderRaiderman, you'd just call "asterisk -r" to connect
04:38.56Raiderman[TK]D-Fender, reboot the computer and then i try to run it as the first command that i type
04:39.37[TK]D-FenderRaiderman, try running * directly then. "asterisk -gvvvvvc"
04:44.29Raiderman[TK]D-Fender, to be clear reboot and then run as first command " asterisk -gvvvvvc" ???
04:44.40[TK]D-FenderRaiderman, just do it NOW.
04:44.53[TK]D-FenderRaiderman, I've already explained this TWICE.
04:46.24Raiderman[TK]D-Fender, done
04:47.35Raidermanand keep say a none stop message
04:48.35Raidermanasutomatly restarting asterisk, mpg123 : no preces killed, asterisk ended with exis status 1, asterisk die with code 1
04:49.05Raiderman[TK]D-Fender, sorry for the typeing
04:49.40[TK]D-FenderRaiderman, kill the process completely and run it direct, NOT via "safe_asterisk"
04:50.48Raiderman[TK]D-Fender, asterisk -r then
04:50.58[TK]D-FenderRaiderman, NO.
04:51.06[TK]D-FenderRaiderman, you are not paying attention.
04:51.23[TK]D-FenderRaiderman, "asterisk -r" = connects to a RUNNING Asterisk process
04:51.42Raiderman[TK]D-Fender, asterisk -gvvvvvc then
04:51.51[TK]D-FenderRaiderman, "asterisk -gvvvc" = starts * in NON Daemon mode directly so you can see what's failing.
04:51.57*** join/#asterisk cowmix (n=mmarch@ip68-228-222-163.ph.ph.cox.net)
04:52.05[TK]D-FenderRaiderman, "safe_asterisk" = starts * in daemon mode
04:52.12JayTee52needs more cowbell
04:52.26Raidermanok
04:52.41cowmixanyone using a SPA3102?
04:55.10*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
04:55.41Raiderman[TK]D-Fender, can i test asterisk with a 3com phone ???
04:56.13[TK]D-FenderRaiderman, depends what kind, and what it can speak now doesn't it?
04:56.48Raiderman[TK]D-Fender, i have 3c10401a
04:57.22[TK]D-FenderRaiderman, Thats like saying can a Ford vehicle tow my 40' yaght.  I might say "yes", and then you'll show me a 1984 Ford Fiesta which couldn't tow a rubber duck.
04:58.06[TK]D-FenderRaiderman, Says it supports SIP, so its a definite "maybe".  Go try
04:58.38[TK]D-FenderRaiderman, and show some initiative and go look for yourself : http://www.google.ca/search?hl=en&q=3c10401a+asterisk&btnG=Search&meta=
05:06.56Raiderman[TK]D-Fender, if i dont have a phone to test asterisk can i do it from a program interface ?
05:07.12[TK]D-FenderRaiderman, go install a soft-phone
05:07.16[TK]D-Fender~softphone
05:07.17jbot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
05:15.21Raidermanthanks for the info
05:19.36drmessanoJust got back from my mother in laws.. try #5 to get her internet working
05:19.53drmessanoGo and check the signal on her modem
05:20.00drmessano.4 db
05:20.09[TK]D-Fenderdrmessano, DSL?
05:20.11drmessanoNominal is 3db
05:20.21drmessanoCramcast
05:20.34[TK]D-Fenderdrmessano, Cable?
05:20.41drmessanoComcast, yes
05:20.58[TK]D-Fenderdrmessano, Ok, I'm not 100% familiar with US ISPs
05:21.13[TK]D-Fenderdrmessano, split too many ways?
05:21.33[TK]D-Fenderdrmessano, And shouldn't their techs look at it?
05:22.24drmessanoWell, thats definitely part of it.. They fucked it up even more when the tech installed a gateway for internet/VoIP for her PHONE ONLY and put another I-NET ONLY MODEM on a splitter as well
05:22.35drmessanoThe first box would do BOTH, but he didnt set it up that way
05:23.07drmessanoI started looking at it as a "Why wont this router work on this line" problem
05:23.12drmessanoand it became more and more
05:23.22drmessanoNow I had called Comcast and have a tech coming Tuesday
05:23.30drmessanos/had/have
05:24.33drmessanoI got into the mix because comcast modems are *just* modem, so when she switched from DSL, her PC was plugged direct into her modem unprotected
05:25.15drmessanoI go over and put a router in, get her ATA into my * box set up again.. troubleshoot..
05:25.23drmessanoThen 2 trips later I called comcast
05:25.46Raiderman[TK]D-Fender,
05:26.23Raiderman[TK]D-Fender, can i run asterisk in a X install ??
05:26.29[TK]D-Fenderdrmessano, shouldn't have 2 modems on the line...  And the router should work.  usually you just have to reset the modem so that it recognizes the MAC of the router
05:26.55[TK]D-FenderRaiderman, No such thing as an X install.  We covered this.
05:27.25[TK]D-FenderRaiderman, It is inadvisable to RUN X on a more dedicated * server.  For lower loads you're probably fine.
05:27.56Raidermanok
05:28.06Raidermani have some shues with zatel
05:28.09drmessanoYeah.. one router with OpenWRT won't grab an IP.. another OLD linksys will.. Tested the OpenWRT box several times.. but it wont work over there
05:28.16drmessanoSomething is just flaky
05:28.34drmessanoI tried swapping cables and all, but I think that second modem is just screwy
05:28.59[TK]D-Fenderdrmessano, Well its nuking signal integrity.  Trust = 0.
05:29.37drmessanoYes.. so and when I saw that.. the light bulb went off..
05:29.49drmessano-and
05:29.56[TK]D-Fenderdrmessano, tried isolating the 2 modems to work them 1 at a time?
05:31.44drmessanoTBH.. I was GOING to.. But I pushed comcast hard on provisioning the one gateway for both.. and I cant test that one with I-net due to how it's set up.. So I pretty well left it there as-is.   I suppose I could have checked modem2 with the splitter out of line, but I am not convinced it was worth it, espeically if I get my way
05:33.25drmessanoAfter I got the OpenWRT box working perfectly at home earlier today, I went over there to give it one last shot... and pretty well knew it was going to be a FAIL anyway.  I hit my limit of what I will do FOR comcast to fix THEIR issue ;)
05:34.24[TK]D-Fenderdrmessano, Yeah, time to make them do their jobs...
05:34.47drmessanoIndeed
05:35.36drmessanoComcast is BAD about load balancing.. and they will never get close to taking ownership of that problem
05:35.59drmessanoBut you call and complain that you have a .7 db signal.. they come out.. and suddenly it's 5db without them doing anything
05:36.05drmessanoAccording to them, that is
05:36.21drmessanoyeah.. they know exactly what they're doing.
05:39.23[TK]D-Fenderok, I've had a full day.... off to bed
05:39.24[TK]D-Fenderlater
05:39.28[TK]D-Fenderall*
05:39.32[TK]D-Fender*yawn*
05:51.56*** part/#asterisk jicksta_ (n=jicksta@75-101-5-48.dsl.static.sonic.net)
05:55.34Raidermannight all
05:55.43Raidermanim going to continue tomorrow
05:55.48*** part/#asterisk Raiderman (n=raider@ip-151-155.tricom.net)
05:56.16*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:59.37*** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl)
06:04.04*** join/#asterisk techie (n=techie@adsl-76-214-1-204.dsl.lsan03.sbcglobal.net)
06:04.44*** part/#asterisk techie (n=techie@adsl-76-214-1-204.dsl.lsan03.sbcglobal.net)
06:10.46*** join/#asterisk adorah (n=Michael@87.69.130.248)
06:15.37*** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus)
06:19.05*** join/#asterisk steliosk (n=Stelios@87.203.145.166)
06:24.23*** join/#asterisk jHoNDoE (n=irc@ip68-230-25-116.ph.ph.cox.net)
06:54.10*** join/#asterisk ruied_ (n=ruied@89-180-112-238.net.novis.pt)
06:59.22*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca)
07:08.51*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137)
07:10.42kamanashisroyhi, I am trying to restrict a specific dialplan to use some applications .. how can I do that ?
07:12.48*** join/#asterisk RyanGWU82 (n=ryan@c-24-6-168-141.hsd1.ca.comcast.net)
07:17.21carraronly put it in the dialplan that you want to access it
07:17.29carrarand don't include that dialplan into others
07:19.30*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
07:23.19*** join/#asterisk marc7 (n=marc@S0106001c10243830.gv.shawcable.net)
07:26.30obnauticusdrmessanom honey
07:26.31obnauticusyou there?
07:34.04kamanashisroycarrar: I could not get you ..
07:34.17kamanashisroycarrar: let me tell my goals ..
07:34.56kamanashisroycarrar: I am writing a drupal gui for asterisk that will listen to other client drupal sites that does not have asterisk ..
07:35.40kamanashisroycarrar: I want to give the client drupal sites freedom to create dialplans ...
07:35.54kamanashisroycarrar: but it is a big security issue ..
07:36.01kamanashisroycarrar: what do you think ?
07:36.22carrarI think you should just use existing software instead of re-inventing the wheel
07:37.29carrar& bed
07:37.55kamanashisroycarrar: yes .. but drupal does not have existing software currently ..
07:45.32*** part/#asterisk RyanGWU82 (n=ryan@c-24-6-168-141.hsd1.ca.comcast.net)
07:46.51*** join/#asterisk nirz (n=nir@bzq-79-179-136-15.red.bezeqint.net)
07:56.54*** join/#asterisk marc7 (n=marc@S0106001c10243830.gv.shawcable.net)
08:07.30*** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal)
08:08.31*** join/#asterisk fmueller (n=user@p548F7B41.dip.t-dialin.net)
08:14.07*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:16.58*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.133)
09:17.55*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)
09:23.17*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
09:25.42*** join/#asterisk gmaruzz (n=gmaruzz@88.239.25.99)
09:31.15*** join/#asterisk Asterisk-new (n=Secret@modem-225.40.wayout.net)
09:31.33Asterisk-newGood day All
09:31.48Asterisk-newany help for newbie?
09:32.49puzzledAsterisk-new: don't ask if you can ask. just state your question
09:33.14Asterisk-newthanks puzzled
09:35.43Asterisk-newmy question is : I need to setup asterisk as PBX conencting software SIP-clients to it so they can dial each other and when dialing outside, the call should go to my SIP provider. I knoe the syntax of my SIP provider and I need to define those SIP clients and define that they should use the SIP provider for non local calls ( local means calling each other)
09:35.59Asterisk-newany sample file or links to them?
09:36.40puzzledAsterisk-new: buy the Asterisk book or download it for free
09:36.52puzzledit has all the basic info you need
09:36.53Asterisk-newThe Asterisk is set up on debian and ready for use and I don't need to do calls using the PSTN. I don't need any fancy features for the clients, just call
09:37.15Asterisk-newperfect, would you kindly tell me about the link to download it?
09:37.22*** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
09:39.02puzzledAsterisk-new: http://downloads.oreilly.com/books/9780596510480.pdf
09:39.30Asterisk-newthanks puzzeld I owe you one
09:39.33Asterisk-newhave a good day
09:39.36puzzledyou too
09:52.07*** join/#asterisk friendly12345 (n=friendly@ppp59-167-94-13.lns2.mel6.internode.on.net)
10:43.52*** join/#asterisk quaqo (n=quaqo@85-18-14-38.fastres.net)
10:51.43*** join/#asterisk dr_gogeta86 (n=gogeta@ppp-189-250.32-151.iol.it)
10:51.48dr_gogeta86hi too alle
10:53.06*** join/#asterisk Strom_M (n=strom@208.127.172.112)
11:00.18*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
11:18.14*** join/#asterisk blq (n=Bl@dslb-088-064-135-077.pools.arcor-ip.net)
11:19.19*** join/#asterisk af_ (n=getsmart@88-149-240-183.dynamic.ngi.it)
11:28.37*** join/#asterisk amolojo (n=amolojo@mail.edinburghint.com)
11:43.17*** join/#asterisk axisys (i=iqbala@otaku.freeshell.ORG)
11:45.02*** join/#asterisk computa_mike (n=mike@82-37-220-225.cable.ubr07.dudl.blueyonder.co.uk)
11:47.46computa_mikehello all - got a quick question - anyone able to deal with a bit of a noob question?.....  Just got a trixbox server set up (sort of)... I got a x100p in there to link to pstn..  I can recieve calls, and route to soft phone - however my dial plan doesn't seem to work proerly...  An ideas what I can check... I#ve tried setting up a dial plan like :   Any number starting with 9, with 5 or more following numbers, dial us
11:49.28computa_mikewhen i say trixbox - i actually think it's the now branded as AsteriskNow  product  - I Insalled from ISO
12:05.16*** join/#asterisk af_ (n=getsmart@88-149-241-241.dynamic.ngi.it)
12:18.56*** join/#asterisk computa_mike (n=mike@82-37-220-225.cable.ubr07.dudl.blueyonder.co.uk)
12:19.19computa_mikeHi - i just posted here but I had to reboot...
12:19.49computa_mikegot an issue trying to dial out from newly installed asteriskNow box withx100p card - anyone got any experience of this?
12:21.22tzafrir_home~ask
12:21.22jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
12:21.47tzafrir_homeIt is also recommended to read the topic
12:22.02tzafrir_homethat said, I figure noone will answer you in #asterisknow
12:22.36tzafrir_home(people there are welcomed to prove me wrong)
12:23.52computa_mikeyeah ok air point I am using asterisknow as opposed to asterisk - i'll just go a post in there.... thanks for the advice
12:24.08*** part/#asterisk computa_mike (n=mike@82-37-220-225.cable.ubr07.dudl.blueyonder.co.uk)
12:24.29*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-248-124.bflony.east.verizon.net)
12:34.16*** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com)
12:50.38*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
13:01.18*** join/#asterisk amolojo (n=amolojo@mail.edinburghint.com)
13:10.02*** join/#asterisk ManxPower (n=manxpowe@137.sub-70-223-146.myvzw.com)
13:10.21*** join/#asterisk Raiderman (n=raider@ip-150-163.tricom.net)
13:10.52Raidermanhi all
13:12.15Raidermanany one of you have install slackware on a virtualbox underlinux to install and test asterisk
13:12.40Raidermanany one of you have install slackware on a virtualbox under windows to install and test asterisk
13:20.51*** part/#asterisk gmaruzz (n=gmaruzz@88.239.25.99)
13:21.26*** join/#asterisk Meaw (n=dino@a151-171.adsl.paltel.net)
13:22.10ManxPowerRaiderman: Everything should work fine except Zap cards and there's a good chance your VoIP audio will be screwed up.  Other than that it should work fine.
13:25.31Raidermancan i install the X version aswell during the instalation of the OS?
13:26.16errrwhy would you need X?
13:26.32errr(but yes you could)
13:26.42Raidermani see asterisk now video some where
13:28.07Raidermanwhat osss you recomend me to ger audio aswell
13:28.42RaidermanOS
13:32.06ManxPowerRaiderman: VoIP is very latency sensitive, anything that locks interrupts (VM, graphics, some onboard IDE/SATA/GigEthernet controllers lock interrupts too) will cause you problems.
13:32.40ManxPowerRaiderman: we don't use AsteriskNOW on this channel.  You would want to ask on #asterisknow
13:36.28tzafrir_homeRaiderman, you can use any distro, not necessarily slackware
13:36.54tzafrir_homewonders what's the source for this slackware/virtualbox lately
13:37.36Raidermantzafrir_home: www.asterikast.com
13:37.50tzafrir_homeI would not recommend AsteriskNow if you actually want to build extra software on your own, as its package management is poorly documented
13:39.51Raidermantzafrir_home: what you recomend
13:40.09tzafrir_homeI generally prefer Debian :-)
13:40.27tzafrir_homeBut generally, if you're familiar with a Linux distribution, go with it
13:40.47tzafrir_homeAlso, if you have a separate box to play with, it would probably be nicer than using an emulator
13:44.35Raidermantzafrir_home: for now i have a desktop and a laptop im about to enter a new harddrive into the desktop and make test to it
13:45.13Raidermanbut first i will try to install on a emulator if all goes well then i move to desktop
13:46.22tzafrir_homeDebian has good Asterisk packages. I'm not sure about the ones in SUSE and Fedora
13:46.55tzafrir_homeThat saves you from the need to go through hoops just to install stuff
13:47.42tzafrir_homeWhich is something many of those howtos spend a whole lot of time on
13:48.07Raidermani just need cd1 to install asterisk write ??
13:50.26ManxPowerRaiderman: no, you need to download Asterisk, build, and install Asterisk
13:51.20Raidermani mean for debian
13:51.42errrdisk 1 will be enough
13:52.02errranything else it doesnt provide can be apt-get installed anyway
13:52.25ManxPowerRaiderman: you should use the Distro you are most familiar with.  If you are not familiar with a distro you will have an incredible difficult task.
13:53.18Raidermanminimal booteable cd image will work ??
13:53.22errrRaiderman: ManxPower is 100% spot on.. if you dont know any linux distros you should learn to use one a bit before you try this..
13:53.57errrno not unless you want to do a network install which as I can tell by the questions you are asking will be a task you cant complete
13:54.19Raidermanim familiar with suse but i never use suse without X
13:54.22ManxPowererrr: People think Telecom/VoIP is easy.  Poor sods.
13:54.33errrManxPower: no joke.
13:55.22errrRaiderman: you need to get to know the linux command line, or your life will be a living hell..
13:55.34Raidermani know linux command
13:55.42errrif you say so ;)
13:55.59Raiderman:)
13:56.02Raidermanlol
13:56.17Raidermani just want to get your experinces and dont waste time
13:56.48errr*grin*
13:57.57*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
13:58.31*** join/#asterisk Rico29 (n=Rico@ARennes-256-1-158-101.w86-195.abo.wanadoo.fr)
14:00.14tzafrir_homeRaiderman, if you have a good internet connection , the netinst CD will probably be better for you
14:00.30ManxPowerRaiderman: We are here because we want to be.  None of us are under any obligation to help you be lazy and not read documentation
14:00.38Raidermanone thing that i knowfor sure is that when i done it and while i get experience i will be here sharing an helping others as you are doing with me atm
14:05.18Raidermanmy apologies to all present if i offend you, but i get the task of know and install asterisk this friday and i have to get it done by monday
14:05.58ManxPowerHey!  Thanks for the best laugh of the week!
14:06.28Raidermanthe sad thing is that is tru
14:08.03*** join/#asterisk putnopvut (n=putnopvu@216.207.245.1)
14:08.03*** mode/#asterisk [+o putnopvut] by ChanServ
14:08.44*** join/#asterisk Rico29 (n=Rico@ARennes-256-1-158-101.w86-195.abo.wanadoo.fr)
14:15.11*** join/#asterisk Raiderman (n=raider@200.42.253.91)
14:27.42*** join/#asterisk Raiderman (n=raider@200.42.253.91)
14:30.52Raidermantzafrir_home: devian donloaded susefuly
14:34.01*** join/#asterisk seanbright-home (n=seanbrig@m525f36d0.tmodns.net)
14:34.37*** join/#asterisk bl4q (n=Bl@dslb-088-066-227-143.pools.arcor-ip.net)
14:35.27*** join/#asterisk marc7 (n=marc@S0106001c100a3e7c.gv.shawcable.net)
14:38.13*** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca)
14:49.48*** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194)
15:00.23*** join/#asterisk Raiderman (n=raider@190.94.16.113)
15:02.42*** join/#asterisk Zyna (n=Brainiac@p54BCFE95.dip.t-dialin.net)
15:02.49Zynawhat's this mean? Error 501 - Not Implement
15:02.55Zynawhen connecting the phone
15:03.19Zynabackground: a friend is trying to connect his softphone to my ext IP
15:03.32ZynaI forwarded 10000-20000UDP to asterisk's IP
15:03.45*** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.190.117)
15:03.56Zynaon 5060 I get internrnal collision which I believe comes from aasterisk already using it through NAT
15:06.00*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
15:08.01Zynaanybody?
15:09.34*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:09.34*** mode/#asterisk [+o lmadsen] by ChanServ
15:15.30[TK]D-FenderZyna, what gives you that error?
15:15.46ZynaConnecting X-Lite through the internet through my router to my asterisk
15:16.00Zynafirewall: 10000-20000UDP to asterisk IP
15:16.03[TK]D-Fender~sipnat
15:16.03jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
15:16.07[TK]D-FenderZyna, read this ^^^
15:16.11Zynaah
15:16.13Zynaperfect
15:16.13Zynathx
15:17.17*** join/#asterisk thesyko (n=theSyko@187.222.48.60.klj04-home.tm.net.my)
15:17.25thesykoguys
15:17.27thesykowanna ask
15:17.39thesykoany one here running asterisk on a gentoo machine?
15:24.31[TK]D-Fenderthesyko, plenty of people
15:24.34Zyna[TK]D-Fender, is there a way to use a different port than 5060? I keep on getting collision with internal setup, even though there is no other NAT rule using that port
15:24.50[TK]D-FenderZyna, you trying to run a soft phone on your * box itself?
15:24.57Zynanope
15:25.12[TK]D-FenderZyna, describe the full network path between * and your phone
15:25.29ZynaThere is * and ext softphone and intern softphone, which is not on *
15:25.54ZynaI'm having no issues conencting to asterisk within LAN
15:25.55[TK]D-FenderZyna, pastebin your sip.conf masking only passwords
15:25.59Zynabut my buddy outside gets 501
15:26.51*** join/#asterisk af_ (n=getsmart@88-149-241-241.dynamic.ngi.it)
15:26.54thesykofender, just a q
15:27.08ManxPowerZyna: define "gets"
15:27.15thesykoi have gentoo on a pentium 3 ~ 733 mhz box with 256ram
15:27.26thesykocan asterisk run on it ?
15:27.26ManxPowerThe source port means nothing.
15:27.50[TK]D-Fenderthesyko, sure, everything depends on the load it will be under.
15:28.13[TK]D-Fenderthesyko, software EC for PSTN cards & transcoding can hurt on low-end machines
15:29.09Zyna[TK]D-Fender, http://pastebin.com/ma1d1308
15:30.10ManxPowerZyna: Do you have G729 licenses?
15:30.19[TK]D-FenderZyna, looks good, enable SIP DEBUG at CLI and pastebin the output.
15:30.22ZynaThis is private use at home for testing
15:30.30thesykofender, its just gonna run maybe 4 sip extensions
15:30.32thesykothats all
15:30.37ManxPowerZyna: How is that relevant?
15:30.38thesykoNo physical cards
15:30.44thesykoor PSTN cards
15:30.49Zynaafaik private use is license free
15:30.53[TK]D-Fenderthesyko, with no transcoding it'll be fine
15:30.59ManxPowerThe G729 license does not say "except for personal use"
15:31.08[TK]D-FenderZyna, Thats not how it works, but get me that next pastebin
15:31.26ManxPowerZyna: but regardless, if you don't have a license G729 won't work, so you might as well remove it from your config
15:31.35[TK]D-FenderZyna, * will not transcode g729 without a PIAD codec & activation key download
15:31.39[TK]D-FenderPAID*
15:32.23Zynaok... well I just installd it cause it was part of a howto... they didnot mention that... I'll get rid of it later asa I got this problem solved
15:33.29ManxPowerZyna: Chances are that's an unlicense and unsupported G729 implementation, which the LIBRARY is a free license from Intel for non-commercial use, but you still need a license from the patent holder.
15:34.12ManxPowerZyna: but my buddy outside gets 501
15:34.13ManxPowerManxPower: Zyna: define "gets"
15:34.28ManxPowerI'm still waiting to find out what "gets 501" means
15:34.33ZynaGets: X-Lite display says 501 - Not implemented
15:35.00ManxPowerZyna: Does that message go away when you remove the allow=g729?
15:35.06ManxPowerAs X-Lite does not support G729
15:35.14[TK]D-FenderYeah, there is that too :)
15:35.39*** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola)
15:36.01Zynahe says he is conencted however dialign 101 -> Not available http://pastebin.com/m12de206c
15:36.41ManxPower[TK]D-Fender: according to a quick google search 501 is "METHOD not implemented", as opposed to a codec thing
15:36.58[TK]D-FenderZyna, Apr 20 15:34:05] NOTICE[2623]: chan_sip.c:13952 handle_request_invite: Call from '100' to extension '101' rejected because extension not found.
15:37.33[TK]D-FenderZyna, Clearly your dialplan doesn't match your sip.conf for the #'s you want to dial.  Check your contexts
15:38.17Zyna[TK]D-Fender, ok his x-lite says indisplay: "your username is 101" I did not get any debug output in asterisk though!?
15:38.24ManxPowerZyna: cite your source
15:38.50ManxPowerZyna: do you or do you not have an exten => 101 under [default] in extensions.conf?
15:39.19*** join/#asterisk Telemac (n=cchantep@ANantes-157-1-122-63.w90-31.abo.wanadoo.fr)
15:39.31TelemacHello
15:39.39*** join/#asterisk CVirus (n=GoD@82.201.178.187)
15:39.46Zyna[TK]D-Fender, http://pastebin.com/m498f37f6
15:39.47ManxPower[TK]D-Fender: I'll bet the 501 comes from X-Lite trying to subscribe to a mailbox but no mailbox= line for that peer
15:40.20ManxPowerZyna: you do not have extensions listed for 100 and 101 in extensions.conf.  That is the cause of the problem.
15:40.37[TK]D-Fenderindeed
15:40.43*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
15:40.54Zynaok, so I'll go something like exten=>101,1,DIAL(SIP,100|10) ?
15:41.06[TK]D-FenderZyna, NEVER think of your phones (soft, hard or otherwise) as EXTENSIONS.
15:41.10[TK]D-FenderZyna, Yes
15:41.15Zynaok
15:41.20ZynaI'll give it a shot
15:41.40[TK]D-Fender~g729
15:41.41jbot[~g729] G.729(.a /.ab /.b) is a patent-encumbered ITU-standard voice codec operating at 8kbps offering quality similar to GSM.  For Asterisk to transcode G.729 licenses (per channel) must be bought from http://store.digium.com
15:41.43ManxPowerZyna: Call comes into Asterisk, the call is auth's against sip.conf, then the call goes to the extensions.conf entry for the context and number of the peer
15:42.30ManxPowersorry, for the context if the sip.conf entry of the calling device, and the extension that was dialed
15:42.59flushyo
15:43.08flushanyone got a how to for recording calls with asterisk
15:43.35Zyna[TK]D-Fender, http://pastebin.com/m76d62d3d ?
15:43.35flushis it possible i can start recording whenver i want without the other noticing im pressing a key or something
15:43.57[TK]D-FenderZyna, look where your phones look for where they will send calls <--
15:44.33[TK]D-Fenderflush, "core show application monitor" , "core show application mixmonitor"
15:44.34Zynaoh so I switch [100] dial 101?
15:44.59[TK]D-FenderZyna, No, llok at your sip.conf "context=" for your phones.  THAT is the context that will get used when they dial
15:46.01Zynaoh, so I have to set the context for 100 in sip.conf to 100 instead of default?
15:46.22[TK]D-FenderZyna, No, they only have to MATCH.  * doesn't care if you call them FRED
15:46.30Zynaok...
15:48.53Zyna[TK]D-Fender, still not working, could you @@ mark the ines that need fixes? http://pastebin.com/m54e39284
15:49.27[TK]D-FenderZyna, show me the failed call
15:50.16Zynahttp://pastebin.com/m2820c396
15:51.17Zynalast block I believe
15:53.57[TK]D-FenderZyna : do "core set verbose 10" and redo the call.
15:54.04[TK]D-FenderZyna, You have no dialplan output.
15:54.55Zynahttp://pastebin.com/m29732967
15:56.50[TK]D-Fender#
15:56.50[TK]D-Fender[Apr 20 15:54:28] WARNING[2687]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
15:57.06[TK]D-FenderZyna, looks like [101] isn't registered to your * currently
15:57.07Zynahow can that be? firewall?
15:57.15[TK]D-FenderZyna, "sip show peers"
15:57.18ZynaX-Lite shows it is trhough
15:57.19[TK]D-FenderZyna, possible
15:57.41Zynaok... he isn't
15:58.06Zynabut like I've said in the beginning... I'm having issues setting up NAT forward 5060UDP
15:58.19Zynainternal rule collision
15:58.28[TK]D-FenderZyna, internal on what exactly?
15:58.30Zynaeventhough there is no other rule using that port
15:58.34Zynaon the router
15:58.42*** join/#asterisk BigCanOfTuna (n=chatzill@66.18.226.119)
15:59.06[TK]D-FenderZyna, if the router is refusing you then you've got a real problem.  this shouldn't be too hard to hunt down...
15:59.32[TK]D-FenderZyna, check your ranges in other rules, multiple pages of rules, etc....
15:59.36ZynaI've already deleted all other NAT rules but still
16:00.02BigCanOfTunaI've got jabber configured with my asterisk install. Since I can see messages sent from a client to my Asterisk server, I assume there must be a way to invoke extensions via a text message. Can anyone confirm if this is possilble? I see lots of articles describing asterisk sending notification to gmail clients, but not the other way around.
16:01.04[TK]D-FenderZyna, its your router, go fix it
16:01.19[TK]D-FenderBigCanOfTuna, * does not support IM
16:01.32BigCanOfTuna[TK]D-Fender: Boooooo. =-(
16:01.44ZynaWhat if I can't? It keeps on saying: "Collision with internal rule" what if it's some router internal rule and i cant get passed it
16:01.58[TK]D-FenderZyna, Get another router.
16:02.07Zynafor placing a test cal???
16:02.09BigCanOfTuna[TK]D-Fender: It would be awsome if I could define a general extention that was invoked on every message sent to asterisk via jabber.
16:02.29[TK]D-FenderZyna, Listen up, fucked is fucked period.  It doesn't care how trivial you think what you want to do is.
16:03.03Zynais there no way to assaign asterisk a different port?
16:03.06[TK]D-FenderZyna, So go hunt down the conflict.  If it says its a rules conflict then either it is, or your router has "issues"
16:03.18ZynaI bet it's the router...
16:03.20[TK]D-FenderZyna, go look HARDER.  What are you using?
16:03.38ZynaSpeedprt W 701V
16:03.43ZynaSpeedport W 701V
16:04.59[TK]D-FenderZyna, go read the admin guide
16:05.12ZynaI'm at it,...
16:05.30[TK]D-FenderZyna, only ones I find are in German.  If its interface is in English I may be able to help you out.
16:05.56ZynaIt's in german... I've found some forum posts stating the exact same issue
16:06.40[TK]D-FenderZyna, lovely little unit you've got there then.... DMZ * if you can.
16:07.05Zyna<PROTECTED>
16:07.18Zynatheir standard router you get when signing
16:08.14[TK]D-FenderZyna, go use another then
16:08.22Zynahaha... the speedport has voip settings itself... here goes the actual problem
16:08.33[TK]D-FenderZyna, disable them then
16:08.36ZynaI'm gonna see if I can disable that
16:09.57filehammers are useful for stuff like that
16:13.01riddleboxdoes anyone know why voip providers cannot port cell numbers yet?
16:13.52puzzledafaik cause in .eu they are regulated to only be operated by gsm service providers
16:19.43Zyna[TK]D-Fender, I just did [GENERAL] bindport=5061 and port=5061 and it worx!!! thx sooo much for help
16:20.05ZynaHe can connect and call me now however I cannot call him...
16:22.34Zynahe forwarded all neccessary ports but still
16:23.03[TK]D-FenderZyna, he should not be forwarding anything
16:23.14Zynawell neither worked
16:23.36[TK]D-FenderZyna, did you tell him to set his client to port 5061 as well?
16:23.53Zynajupp... he is conencted and just called me
16:24.19[TK]D-FenderZyna, set "port-5060" in his sip.conf entry
16:24.25[TK]D-FenderZyna, set "port=5060" in his sip.conf entry
16:26.31Zynanope, but doesn't matter... I don't need to test that since external calles will eb going through ISDN on the productive machine
16:27.46[TK]D-FenderZyna, if you only want him to be able to dialOUT and never RECEIVE any calls, sure
16:28.20Zyna[TK]D-Fender, so do I understand this technically correct, tha sip.conf is kinda like the setup file and extensions.conf is the dialplan?
16:29.04Zyna[TK]D-Fender, no, I'm not going to have these issues later, since the productive box will have a ISDN card directly hooked to the telephone weiring ... no NAT problems there ;)
16:29.23[TK]D-FenderZyna, go read.. THE BOOK
16:29.25[TK]D-Fender~book
16:29.26jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
16:29.48ZynaI have... it didn't help me... I'm not much of a reader... I learn best by looking at working conf files
16:30.00*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
16:30.31jblackI know how to solve the "I didn't read the book" problem.
16:30.50jblackWe get leif to redo it as a comic book. With anime french maid lesbians.
16:31.23ZynaI'd still go with the working conf files...
16:31.32ManxPowerjblack: restraints + books on tape
16:31.33jblacklmadsen: How are your drawing skills?
16:33.05ZynaI learned VB6 without ever reading a book... and I can do almost anything up to ADO and winsock programming
16:33.13jblackZyna: Well, the book has config file examples. There's also a number of example config files that comes with the distro.
16:33.15Zynajust from snippets
16:33.26*** join/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au)
16:34.08ZynaI borrowed it from some buddy eith no cd attached and when I read it... I wasn't setting up yet... tima management issue I gues... ;)
16:34.52jblackNo problem. You can use the pdf listed above until you get around to ordering a hard copy.
16:35.18ZynaI already fetched it ;) thx
16:35.20dofearhi anyone here happen to know if I can use sipwise.com to genrate config for openser 1.3.1?
16:36.53jblackJust bear in mind that there's very few questions that aren't answered by the book.
16:41.07drmessanoThat gives me a great idea
16:42.29drmessanoI was farting around one day and decided to take a text file copy of the bible, load it into a bot, have it "read" the book to a channel, one line every 3 seconds.
16:42.45drmessanoOk, so I created it for trolling.... however
16:42.46errrlol
16:43.06drmessanoCould do an #asterisk-book channel
16:43.15drmessanolol
16:43.25drmessanoTFOT 24/7
16:45.20*** join/#asterisk seaq (n=seaq@201.244.27.135)
16:46.46jblackbrilliant! What happened to the channel?
16:48.07*** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net)
16:48.13drmessanoThe bible one?
16:50.02jblackyeah
16:50.38drmessanoI ran it for a few weeks just to tweak how long it played out the whole book.. then shelved it
16:51.07drmessanoI tried to get the time to exactly 7 days, but never got it just right
16:51.12jblackI used to own #warez on dalnet. Well, until I added *!*@* to the aop list.
16:51.22jblackShould have been six days. The seventh day was for rest.
16:51.32*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
16:52.37*** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net)
16:52.45errrheh
16:52.48jameswf-homeRoflmao D&D http://youtube.com/watch?v=joylAVP6v5M
16:52.57errrday 7 could have just printed zzzzzz
16:53.28jblackAnyways, the effect of giving every wanna-be script kiddie aop access in one channel was hilarious. Took dalnet down for 2 hours. =)
16:53.47*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:53.59drmessanojblack: I went down the hall and was going to come back and say just that.. 6 days would have been awesome, then have the bot do a /me rests
16:55.01eric2in sip.conf, how do you deny calls from all ip's?
16:55.20drmessanoIn my years of IRC I spent a LOT of time making scripts.. Trivia games, music scripts, etc
16:55.41*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
16:56.16*** join/#asterisk seanbright (n=seanbrig@m2a5f36d0.tmodns.net)
16:58.58[TK]D-Fendereric2 : deny=0.0.0.0
16:59.22*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
16:59.41[TK]D-Fender~roflmao
16:59.41jbotrumour has it, roflmao is rolling on the floor laughing my arse off, or painful, or http://www.youtube.com/watch?v=iEWgs6YQR9A
17:00.14eric2ah, tx
17:00.58drmessanoHAX
17:02.27drmessanoI remember being accused of hacking someone for doing a CTCP version and telling them they needed to upgrade their mIRC
17:02.54[TK]D-Fenderdrmessano, Borderline
17:03.21drmessanolol
17:04.32drmessanoWhen I ran my trivia channel, and had 50 or 60 people in there at a time.. I HAD to do version checks, especially around the early days of mIRC 6.. otherwise people would get booted off and someone I didn't do enough to stop it
17:13.25*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
17:16.04seanbrightwhen i was in a barber shop quartet in skokie, illinois...
17:20.03drmessanoZOMG
17:20.08drmessanoThat was YOU?
17:20.25drmessano<--- Dangerous Donnie
17:28.55*** join/#asterisk s0lid (n=s0lid@210.213.198.151)
17:31.44*** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
17:32.20ThatKidKelIf my SIP To Header looks like: To: <sip: *99@my.sip.domain;user=18005551212>.   How can I access just the "user" value?
17:41.09*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
17:41.09*** mode/#asterisk [+o angler] by ChanServ
17:41.16*** join/#asterisk Deeewayne (n=dwayne@216.207.245.1)
17:41.16*** mode/#asterisk [+o Deeewayne] by ChanServ
17:46.38*** join/#asterisk igascream (n=igascrea@bzq-84-109-81-197.red.bezeqint.net)
17:48.53*** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola)
17:49.41*** join/#asterisk fish-bulb (n=cstewart@216.207.245.1)
17:50.27*** join/#asterisk solar_ant (n=John@122.164.184.6)
17:55.34igascreamHI all ,I have some problems with hanging up in IL .I use Digium card . Did anybody have the same problem?
18:00.27*** join/#asterisk Testmaster (n=dancole@S010600a0d1757bfb.cg.shawcable.net)
18:01.00Testmasterhello all i am having a problem with asterisk saying a call is connected while its ringing is there anyway around this so it only shows once the call is actually connected?
18:06.58ManxPowerSomethingISODD: FXO or PRI?
18:07.09*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
18:07.31*** part/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net)
18:20.43*** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net)
18:20.53SomethingISODDManxPower its all voip.. sending to a provider via sip
18:29.28[TK]D-FenderSomethingISODD, Then your provider is "answering" the call and there is absolutely nothing you can do about it
18:29.54SomethingISODDOk thank you..
18:30.07SomethingISODDthe weird thing is.. when i connect me sip phone directly to him
18:30.16SomethingISODDit doesnt start counting until its actually answered
18:31.10[TK]D-FenderSomethingISODD, Go look at the sip debug and pastebin it AND your dilaplan output from begining to end
18:33.06SomethingISODDwell there is only one context i use. for the out bound. the reast is the standarded file
18:33.28[TK]D-FenderSomethingISODD, none of what you said means anything.
18:33.34[TK]D-FenderSomethingISODD, PASTEBIN.
18:33.42SomethingISODDgive me a moment let me make a call and send it to you
18:34.18*** join/#asterisk kr-neki (n=e@edo.kr-neki.si)
18:35.44*** part/#asterisk grEvenX (n=even@pc107-130.ktv.no)
18:40.38SomethingISODD[TK]D-Fender i cant check right now seems the provider has blocked me
18:40.56SomethingISODDbut another question do you know of any scripts that uses the API to show active calls with duration?
18:41.13[TK]D-FenderSomethingISODD, BLOCKED YOU?
18:41.22[TK]D-FenderSomethingISODD, From what, SSH-ing into your own SERVER?
18:41.30SomethingISODDno no. from completing a call through him
18:41.56[TK]D-FenderSomethingISODD, So first you don't like the way a call was being handled and now they are just refusing your calls?
18:42.10SomethingISODDya basically
18:42.31[TK]D-FenderSomethingISODD, time to find a clue.
18:42.41SomethingISODDalready looking for a new provider.
18:42.56[TK]D-FenderSomethingISODD, And for your call duration questio, "core show channels concise"
18:43.51SomethingISODDok thanks that will show the duration ?
18:46.04SomethingISODDcool thank you very much [TK]D-Fender
18:50.49*** join/#asterisk akira2014 (n=chatzill@194.Red-88-6-186.staticIP.rima-tde.net)
18:51.10akira2014hello
18:52.35SomethingISODD[TK]D-Fender do happen to know of any php scripts that is built that will login through the manager api to show active calls and duration?
18:52.55akira2014i'm trying to get working a TDM400P
18:53.28akira2014i can't call from the analog phone to a sip phone and i don't know why
18:53.57*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:54.35akira2014from the analog a can heard the demo-congrats, bur when i call the sip exten i only heard MOH and the SIP dont ring
18:54.58akira2014can some one point me to some doc or help me ?
18:55.00akira2014thk's
18:55.36akira2014asterisk is 1.4.17 from sources and zaptel is 1.4.10
19:01.17*** join/#asterisk Raiderman (n=raider@ip-149-141.tricom.net)
19:01.44*** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
19:01.45*** join/#asterisk sulan (n=sulan@89-253-95-143.customers.ownit.se)
19:02.19sulanhm, I'm having troubles when my SPA942 is negotiating codecs: "Found unknown media description format alaw for ID 8"
19:02.30sulanhow come Asterisk doesn't match alaw to G711a?
19:03.00*** join/#asterisk variable_office (n=variable@cerberus.iswan.net)
19:03.04sulan"a=rtpmap:8 alaw/8000" is what the SDP say
19:05.24*** join/#asterisk variable_office (n=variable@cerberus.iswan.net)
19:07.22variable_officemy asterisk box is taking some time to respond to invites, but randomly, sometimes it responds quickly other times not; any ideas as to why this would occur?
19:14.09[TK]D-FenderSomethingISODD, ready made PHP, no.
19:14.10[TK]D-Fender~fop
19:14.11jbotAn XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel
19:14.41seanbrighthmmm
19:14.47[TK]D-Fendersulan, pastebin the sip debug of your failed call.
19:14.55seanbrightthe former seems a little OT
19:15.30[TK]D-Fenderakira2014, pastebin your zapata.conf and your extensions.conf
19:15.31[TK]D-Fender~pb
19:15.32jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:15.33[TK]D-Fender^^^^^^^
19:16.25*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
19:16.42sulan[TK]D-Fender: http://pastebin.org/30957
19:16.46akira2014[TK]D-Fender: i've found why it does not work, a mistake in extensions.conf
19:16.55akira2014thks,
19:18.24[TK]D-Fendersulan, Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x901 (g723|g726|g729)/video=0x0 (nothing), combined - 0x0 (nothing)
19:18.37[TK]D-Fendersulan, dosn't look like the phone is offering G.711
19:18.44sulanwell, it does...
19:18.49[TK]D-Fendersulan, and I've used those phones myself.
19:19.14sulanyes, me too... just not with this asterisk installation
19:19.43sulanalaw works with my cell, but it offers "PCMA" instead of "alaw"
19:19.44[TK]D-Fendersulan, its not *... * is clearly offering alaw.  Go check your phone another 10 times.
19:20.08sulanthe phone offers a=rtpmap:8 alaw/8000
19:20.11[TK]D-Fendersulan, forget that * calls it "alaw", its the codec # that matters.
19:20.33sulanFound unknown media description format alaw for ID 8
19:20.43sulanthat is what makes me wonder...
19:20.47[TK]D-Fendersulan, check it again anyways.
19:20.57[TK]D-Fendersulan, And heck your peer too just because
19:21.25sulanmy "peer" in sip.conf?
19:23.30sulanthe phone says "preferred codec: G711a"
19:24.31sulanit has been knicked from a site that uses asterisk and alaw successfully, so probably the issue is with the config of the asterisk that it is used with now
19:26.24*** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net)
19:28.02*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
19:28.08SomethingISODD[TK]D-Fender are you a php programmer by chance?
19:28.16[TK]D-FenderSomethingISODD, Yes.
19:28.17Raiderman[TK]D-Fender:
19:28.21Raiderman[TK]D-Fender: hi there
19:28.56Raiderman[TK]D-Fender: can you point me to a step by step on how to install asterisk on debian ?
19:28.59*** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com)
19:29.11SomethingISODD[TK]D-Fender i was wondering if you can give me a head I am trying to figure out how to extract just the duration and the telephone number from the core show channels concise
19:29.42[TK]D-FenderSomethingISODD, its nothing but basic string manipulation.
19:30.27SomethingISODD[TK]D-Fender can you tell me what i need to read up on with php to figure out how to do that?? i know basic php very well i am just not great at string manipulation..
19:30.57SomethingISODDmatter of fact i have no clue what i am doing to extract just the numbers i want
19:31.10[TK]D-FenderSomethingISODD, what is "basic PHP" without the ability to do string manipulation?  Go download the handbook and read the function list.
19:31.19[TK]D-FenderSomethingISODD, this isn't Raw-Cat Science
19:31.40[TK]D-FenderSomethingISODD, And go take a programming course
19:32.30SomethingISODDthanks [TK]D-Fender
19:33.32drmessanolol
19:34.13drmessanoI've been trying to write a web based dialer using Ruby, Perl, and PHP
19:34.20drmessanoMainly for 100 or more callcenter agents
19:34.44drmessanoBut I got stuck.. because I know nothing about Ruby, Perl, OR PHP
19:34.48drmessano:(
19:35.32Raidermanany one can help me to install asterisk on debian
19:35.36drmessanoI did get as far as --> nano new.file
19:35.49[TK]D-Fenderdrmessano, So I want to build a skyscraper but know absolutely nothing about archecture.  Chan you help?
19:36.06[TK]D-FenderRaiderman, Download the book, follow the instructions.
19:36.27drmessanoRaiderman: This channel is really not for beginner "I don't know how to install asterisk" help
19:36.32drmessanoIt's for help with the actual working app
19:36.42drmessanoSo you need to get the book, or check some online resources
19:40.58*** join/#asterisk NirS (n=NirS@77.127.78.115)
19:47.39akira2014i'va another problem.... :(
19:47.55akira2014now i can call from sip to zap and zap to zap
19:47.57akira2014but
19:48.32akira2014when i can from zap to sip or from sip to zap the volume is very low, near to 0...
19:49.16akira2014when a call is bridged ( FXS <=> FXO ) volume is ok
19:49.37akira2014i've touched rx and tx gain but no changes
19:53.38*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:58.43*** join/#asterisk rcy (n=rcy@S010600131094a3de.vc.shawcable.net)
19:59.41[TK]D-Fenderakira2014, you have to reload chan_zap for those changes to take effect
20:00.11akira2014i've restarted asterisk
20:07.14*** join/#asterisk CVirus (n=GoD@82.201.178.187)
20:12.16drmessanoWhatever happened to that douche, that was selling cisco, that would condesendingly get on here on Sunday and ask everyone how they enjoyed spending another weekend on IRC?
20:13.15drmessanoI kinda miss his "accidental pastes" of the perversion he was typing to his imaginary IRC girlfriend, his Cisco banter, and his claims of making so much money
20:13.21drmessanoWas a real highlight
20:14.39drmessanoHAHAH -> http://www.traipse.com/upgrade/index.html
20:21.11akira2014must leave, thk's to all
20:21.59drmessanoIs it remotely possible that Firefox, thru Chatzilla, as made IRC more popular again?
20:22.02drmessanohas
20:23.53*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
20:26.16[TK]D-Fenderdrmessano, Doubt it.  My current belief is that apps like * that are not thoroughly documented and can have so many issues leads people to want immediate interactive answers and hence they learn of this place.
20:27.25*** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer)
20:33.20drmessanoI dunno.. I have just seen a lot of Chatzilla use all over the place.. Chatzilla would never be used by any self respecting IRC'er...
20:33.27*** join/#asterisk TTT_Travis (i=d09dac08@gateway/web/ajax/mibbit.com/x-1f92abe90a09b121)
20:33.55TTT_Traviswhat is the difference between an X100 and a TDM800
20:34.21drmessanoYou mean like an X100P $25 card?
20:34.29TTT_Travisyea
20:34.32drmessanoROFL
20:34.36drmessanoNight / Day
20:34.41TTT_Travisi know this is a dumb question but....
20:34.52drmessanoShit / Quality
20:35.10drmessanoHappy customer / Fired for incompetence
20:35.22drmessanoI could go on, if you like
20:35.54drmessanoX100P is like installing Windows 98 for a file server
20:35.56[TK]D-FenderTTT_Travis, TDM800 is a much better PCI design, an multi-port.  The X100P is a low-grade crap as you can get
20:36.16TTT_Travisyea I just need a card to test at home
20:36.25drmessanoOh here we go
20:36.39TTT_TravisI probably should have used a different name
20:36.54drmessanoNo
20:36.55drmessanoIts not that
20:36.58drmessanoBut understand
20:38.01drmessanoEVERYONE N00B has a justification for buying an X100P card.. It would be like if I asked you for help buying a car, and you gave me a recommendation, and warned me about some others.. and I said "Well, the YUGO fits my needs.. and I just need it to get back and forth to work"
20:38.28drmessanoSure.. There is a HUGE gap between the X100P and a good starter PCI card
20:38.38drmessanoThat price gap is a gripe amongst many
20:39.08drmessanoBut in the end, if the X100P didn't exist, people would accept a good starter PCI being a little over $100 and deal with it
20:39.56drmessanoSadly the X100P does exist, and everyone has a justification for getting one as a "card to play with, just to see", and often they get a bad impression of * from it
20:40.08TTT_Travisok but it will work for me to test my asterisk box that has no real purpose
20:40.10drmessanoor they spend hours trying to get help for it
20:40.14TTT_Traviswhats bad? sound quality? setup
20:40.19drmessanoRead what I just said
20:40.26drmessanoIt's crap
20:41.04TTT_Travismmmmmk
20:41.24drmessanoI'm sure he's buying one now
20:41.27C4awaylol
20:41.30C4awayI want one
20:41.30drmessanoAs expected
20:41.42C4awaynever used one, I want to see if they are really as bad as people say
20:42.01mwallingdrmessano: but a Yugo gets good gas milage!
20:42.08drmessanohah
20:42.11mwallingdrmessano: oo! cause you have to push it!
20:42.25drmessanoIt also gets up to 200km/h
20:42.29drmessanoDownhill
20:42.46mwallingi was going to say with several nigerian marathoners pushing
20:42.54drmessanoROFL
20:43.13C4awayhaha
20:43.30drmessanoI would love to get a box of X100Ps... and work on gluing them together
20:43.35drmessanoand make an X100P2
20:43.43drmessanoDUAL HEAD X100P
20:43.46drmessanoGET YOURS NOW
20:43.54mwallingdrmessano: liquid cooling
20:44.05mwallingthat always attracts a crowd
20:44.19drmessanolol
20:45.00drmessanoAll you would REALLY need..
20:45.27drmessanois a standoff between boards. in two places
20:45.32[TK]D-FenderDon't forget "overclocked", passive cooling, LED's,  CCFL lighting, gold-plated, and "Ultra".
20:45.46drmessanoDual head, double passthru
20:46.22drmessanoJust need to get the cards to show up as X100P2
20:46.34*** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com)
20:46.57drmessanoYou know.. I hate to say this
20:47.11drmessanoBut for the price of one of those Chinese $50 knock off cards
20:47.24drmessanoI think the FXO boards are $30 or something like that
20:47.32drmessanoIf you really wanted to be a cheap ass
20:47.36drmessanoand get a "first card"
20:47.44drmessanoThat would be marginally better than an X100P
20:47.56drmessanoAt least theres a chance it will work
20:48.09[TK]D-Fenderdrmessano, .... he left a while ago you know...
20:48.18drmessanoI know
20:48.23drmessanoThinking out loud
20:48.27drmessano:(
20:49.43drmessanoIm trying to remember who that company was
20:49.48drmessanoEven cheaper than openVox
20:52.56drmessanoOuch
20:53.18*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:56.53*** join/#asterisk jmhunter (n=jmhunter@72.14.86.68)
20:56.53*** mode/#asterisk [+o jmhunter] by ChanServ
20:57.03jmhunter~seen slepp
20:57.05jbotslepp <n=slepp@S01060016b6cf3a48.ed.shawcable.net> was last seen on IRC in channel #asterisk, 501d 15h 56m 34s ago, saying: 'and now, it's time for food'.
20:57.31jmhunter~seen kram
20:57.31jbotkram <n=kram@cm56-214-79.liwest.at> was last seen on IRC in channel #debian, 231d 45m 35s ago, saying: 'hey guys, my debian decided to show me parts of gnome and apt outout in half geran since a few weeks, can anyone tell me how to make it speak english eagain?'.
21:08.40drmessanoI found them
21:09.00drmessano$90 for a A400P w/FXO and Octastic
21:09.35drmessanoProbably cheap shit.. but so is an X100P
21:09.52drmessanoAt least it's a better design
21:10.09bitzerobought an x100p knock-off 3+ years ago as his first card to work with Asterisk... I haven't used one since, because they're useless for any kind of production environment... but for a cheap first PSTN test... I still don't understand why people hate them.
21:10.33tzafrir_homedrmessano, that 90$ is only for the base card, without modules
21:10.39drmessanoNo, it's not
21:10.53drmessanoA400P w/1FXO and Octastic
21:10.57drmessano$90
21:11.03drmessano$50 + $32 + $8
21:11.10tzafrir_homeOctasic is software. Use OSLEC instead...
21:11.25drmessanoWell, $82 then
21:11.50drmessanoversus $30 for an X100P that is designed to fail
21:12.37drmessanothey have the A800P for $50 too lol
21:13.56drmessanoIt looks as though you need to patch Zaptel for this 8 port card.. so one of the buy recommendations was the 4 port vs the 8
21:14.33drmessanoHmm
21:14.51drmessanoThey do have patched tarballs of Zaptel for their 8 port card all the way up to 1.4.9.1
21:15.09[TK]D-Fenderdrmessano, Hey I object... the X100P was not designed to fail!
21:15.16[TK]D-Fenderdrmessano, Thats just a "bonus feature"!
21:15.23drmessanoHA!  FTW
21:15.48drmessanoCraptastic Failsure (tm) included with each card (onboard)
21:17.23drmessano[TK]D-Fender: BTW.. My Monster-in-law called this morning and told me her cable was COMPLETELY out now.. so they really screwed the pooch on this install
21:17.40drmessanoActually
21:17.48drmessanoThe internet was out.. the voice was working
21:17.55drmessanoSo i bet that second modem failed
21:17.59bitzeroisn't claiming that the X100P is going to last forever... I don't even know if the one I had would still work today (technically I guess I still have it in a box somewhere)... but it worked fine for my first initial tests with Asterisk... and without it I probably wouldn't have gone on to invest in other more respectable hardware.
21:18.12bitzerowhat I'm saying is I don't understand pissing on home users who just want to give it a try.
21:18.27*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
21:18.29drmessanoFor $82 you can get a knock off of a GOOD DESIGN
21:18.42drmessanoand it may even WORK for HOME use beyond just being a throw away
21:19.24bitzeroIf you've got some knowledge about them having a very low MTBF - tell the home users "It wont last - I'm warning you"
21:19.46bitzerobut why ride them so hard on "IT IS CRAP OH GOD I HATE YOU SO MUCH FOR THINKING SUCH THINGS"
21:19.51drmessanoMTBF is the instant they open the box and it's starts echoing without power applied
21:19.53JerJerwhoohooo  happy 4:20pm central time
21:20.01bitzerodidn'
21:20.09bitzerot have any such issues - maybe I was just lucky.
21:21.32drmessanoThe point is that, if someone wanted to sell me a NEW $3000 automobile, I would pass.. Just because it's dirt cheap and has 4 wheels doesnt mean it's not work spending real money to get something that will last
21:21.43drmessanoworth*
21:22.09*** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com)
21:22.26drmessanoWhats the point in blowing $30 on a card just to go "Yay, I made a zap card work", then tossing it in a box and buying a real card
21:23.16drmessanoIMHO, if you get the $80 card, make it work.. you can probably get some useful life out of it as a home card and have some knowledge of how the big boy cards work if you're going to be selling them
21:23.54drmessanoConsidering that Asterisk + a box to run it on is about $10000 cheaper than the other guy, whats $50?
21:24.10drmessano~cheapskate
21:24.16drmessanobah
21:24.19drmessanoIt WAS there
21:24.19Strom_C~cheap
21:24.20jbotsomebody said cheap was a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
21:24.29drmessanoMaybe that was it
21:24.37carrarcar used to cost less the $3000
21:25.15Strom_Ccarrar: yes, let's just completely ignore the relative actual value of a nominal dollar between 1963 and today
21:25.20carrarinvest in clean fuel, don't buy a car
21:25.30carrargo hydrogen
21:29.38mgdm~book
21:29.39jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
21:29.39[TK]D-Fenderthe kind that persist on searching out validation for the X100P fall into these groups : A. Those that think they need a real card no matter how shit to prove that their system works (BS). B. Those that are actually intended on putting them in production regardless of claims.
21:30.18[TK]D-FenderThose claiming to use X100P's for "testing" is BS, straight up.  You're throwing your money away for nothing.
21:30.33drmessanoIndeed
21:31.08[TK]D-FenderA soft-phone can test your dialplan as well (better in the case of X100P) than anything else.
21:31.23[TK]D-Fenders/than/as
21:32.11drmessanoTheres also some validation that it works on a real phone line, therefore, "I actually built a real PBX".. completely ignoring the fact that you CAN use VoIP for origination and termination
21:33.11[TK]D-Fenderdrmessano, But for anyone saying "we'll get a real card after", is a total shmuck
21:33.28drmessano[TK]D-Fender: Oh, absolutely. I couldnt agree more
21:33.44[TK]D-Fenderdrmessano, thats like buying a Lada when you think you're going for the Honda Accord after regardless.
21:34.06[TK]D-FenderSo all of you, save your money and your whiny excuses!
21:34.22mgdmtook advice and got a Sangoma card straight off
21:35.47drmessano[TK]D-Fender: I'd rather see someone get a knock off card, and at least get some real world experience with a working design than to get an X100P... but then again, theres that same slipper slope
21:35.50drmessanoslippery*
21:37.05*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)
21:37.21drmessanoMy theory is that after spending all this time and justifying buying a PC for learning *, some have to prove value by setting it up as a home PBX so the wife can see it
21:37.45drmessanoSo therefore, I recommend renaming the X100P to the X100Pussywhipped
21:38.22drmessanoX100P: The cheapest card you can get to prove to the wife why in the hell you needed to learn how to build a PBX
21:41.59[TK]D-Fenderdrmessano, The SPA-3102 would do the job and better.
21:42.20*** join/#asterisk znoG (n=gs@host52.190-31-242.telecom.net.ar)
21:44.04drmessanoSure, but it's not a cool card you can install
21:44.18drmessanoand it says "Linksys" on it, so that like -5 to most people
21:44.41drmessanoI love my SPA-3102s.. But I am a lunatic
21:45.02mgdmIs there a good guide to setting up the SPA-3102 anywhere?
21:45.08mgdmIt's a little... cryptic
21:45.21drmessanoSadly, I used to share one
21:45.28drmessanoOne I made up
21:45.38drmessanoBut I got tired of arguing over the config
21:46.43drmessanoThere's a few guides online..  they kinda work
21:47.15mgdmI've done a reasonable amount of Googling
21:47.28mgdmBut then got distracted by sticking * on an NSLU2
21:48.32drmessanoKerry Garrison has an OK guide out there.. Problem with that and a lot of them is they basically dumb down the FXO config on the device to where you throw calls it from * and the SPA-3102 makes any call it gets sent
21:48.37drmessanoNo auth
21:48.50mgdmouch
21:48.51drmessanoIt's meant for being inside a "protected " network..
21:49.12mgdmFor now, I just want to use the SPA as a dumb SIP client to the Asterisk box
21:49.25drmessanoand ANY call that comes in on the FXO gets sent to the IP specified in the config
21:49.45drmessanoI'll give ya some idea what you need to do
21:50.00drmessanoCreate a Peer in asterisk for it.. have the box reg to * like an FXS ATA would
21:50.13drmessanoThen use that peer for your outbound calls
21:50.44drmessanoIf you do it that way, you can set the IP to dynamic and put the box anywhere on earth.. so long as it can register to the * box from the outside
21:51.10mgdmcool
21:51.44mgdmThis might be a dumb question, but are SIP secrets transmitted in the clear, and if so, can that be changed?
21:52.05*** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
21:52.10drmessanoBy committee, I am sure
21:52.18drmessanolol
21:52.21drmessanoNo
21:52.35ThatKidKelTwo Asterisk Boxes, Same Version.  Same upstream Carrier.  One will re-invite, one won't..  What can I look for?
21:52.52mgdmWell, I meant can I run it across SSL, or some such
21:53.16drmessanoYou can run it across anything you want.. but the SPA-3102 doesn't have such built in
21:53.20mgdmok
21:53.23drmessanoVPN, etc
21:56.33mgdmCool. I'm just playing about on my LAN just now, so I won't worry too much for the moment
21:58.20*** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net)
21:58.35*** join/#asterisk Rico29 (n=Rico@ARennes-256-1-158-101.w86-195.abo.wanadoo.fr)
21:59.12drmessanoI've used the SPA-3102 at a remote location as an additional line on my box here, and it's worked well
21:59.35*** join/#asterisk quaqo (n=quaqo@85-18-14-38.fastres.net)
22:00.22[TK]D-FenderWish Linksys wasn't so dumb with their SPA-400 design....
22:01.43drmessanobbiab.. food shopping
22:02.17*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131)
22:02.24ManxPowerThatKidKel: NAT and firewall
22:03.38*** join/#asterisk colinm_ (n=colol@VDSL-130-13-116-41.PHNX.QWEST.NET)
22:03.57*** join/#asterisk xp_prg (n=wgheath@c-24-130-17-9.hsd1.ca.comcast.net)
22:04.39xp_prghi all I just installed pbx in a flash, I want to go into the asterisk mgmt through the web page, it is asking for a login name and password, anyone know what that is on a default install of pbx in a flash?
22:05.02*** join/#asterisk quigon (n=matias@190.3.121.15)
22:07.14[TK]D-Fenderxp_prg, No, and GUI's are not supported here.
22:20.16*** part/#asterisk fmueller (n=user@p548F7B41.dip.t-dialin.net)
22:28.23*** join/#asterisk jackson__ (n=jackson@68-115-108-47.dhcp.roch.mn.charter.com)
22:39.57*** join/#asterisk |Spe[k] (n=pwncakes@CPE-24-160-254-96.wi.res.rr.com)
22:40.20|Spe[k]heyo
22:55.07|Spe[k]anyone here?
22:58.49*** join/#asterisk jmhunter (n=jmhunter@72.14.86.68)
22:58.49*** mode/#asterisk [+o jmhunter] by ChanServ
23:08.43*** join/#asterisk solar_ant (n=John@122.164.184.6)
23:12.17*** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer)
23:18.12*** join/#asterisk craigk (n=craigk@58.174.150.119)
23:21.57*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:22.07xp_prghi all, I am in zoiper and I am confused how to dial what worked in ekiga:  1111@192.168.0.122
23:22.16xp_prgwhat do I type in the phone to dial of zoiper?
23:22.39*** join/#asterisk CCFL_Man2 (i=b7446bf7@71.245.2.119)
23:24.59jblackxp_prg: Presumably zoiper is registering to a sip extension in *.
23:25.18jblackIf so, you can dial any extension in the context that the sip extension is put in.
23:26.39*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
23:33.27*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
23:44.16|Spe[k]so i have an old motorola carphone with an RJ45 Jack, could i theoretically use it to make calls through *?
23:47.26*** join/#asterisk DrDrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com)
23:51.34*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
23:54.24Qwelltheoretically, sure
23:54.29Qwellrealistically, no
23:54.40*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:54.44Qwellyou would have to write a channel driver for whatever protocol it uses - assuming it's even IP
23:54.48drmessanoHoly crap
23:55.04drmessanoRJ45 does not mean ethernet
23:55.08DrDrainAnybody know how to tell what version of zaptel is running
23:55.10drmessanocries
23:55.25drmessanocries harder
23:55.36*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
23:55.43*** join/#asterisk pejo_ (n=hack@89.160.93.224)
23:55.45*** part/#asterisk pejo_ (n=hack@89.160.93.224)
23:56.02*** join/#asterisk pejo_ (n=hack@89.160.93.224)
23:56.02drmessanopejo_ just hacked us
23:56.09pejo_I did?
23:56.12pejo_;)
23:56.17drmessanoyep
23:56.25pejo_By the way, whats the answer to 200 OKs?
23:56.26drmessanoH4X0R3D by pejo_
23:56.35mwallingdrmessano: yeah, cisco thought they would be cleaver and use rj45 for their console connections :/
23:56.44mwallingmoney sucking asswipes
23:56.54drmessanomwalling: DO NOT... DO NOT.. even get me started on THAT one
23:57.08mwallingheh
23:57.12mwallingpushes "START"
23:57.17drmessanoor about dumbasses plugging ethernet cables into console connections on Cisco AP's
23:57.32drmessano"Hmm.. AP7 doesn't work"
23:57.33pejo_... if i send an invite, and recieve a 200 ok, what should i answer back? :O
23:57.51drmessanopejo_: 300 kthxbye
23:58.00drmessanono, not really
23:58.32pejo_drmessano: i would like to make the session setup complete so i can start sending rtps
23:59.08pejo_but i guess i still have some work todo before im there
23:59.16pejo_:~(
23:59.53jblackhttp://www.google.com/trends?q=asterisk%2C+callweaver%2C+trixbox&ctab=0&geo=all&date=ytd&sort=0

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.