IRC log for #asterisk on 20090107

00:00.11`paulyup
00:00.18`paulmodprobe ztdummy
00:00.19jayteeand you don't type /etc/init.d/zaptel start you type service zaptel start
00:00.21`paulreturns
00:00.36`paulFATAL: Module ztdummy not found.
00:00.56jayteethe other command is for debian based systems. Red Hat based systems do it different
00:02.22`pauli remember installing it once.... when it failed it automatically loads ztdummy
00:03.08jayteedid you do a "chkconfig zaptel on" after you compiled and ran make install?
00:04.25`paulchkconfig zaptel on  <--- doesnt return anything
00:04.40jayteeit shouldn't, it just runs
00:05.08`paulok still same error when i start the service
00:05.09`paulLoading zaptel framework:  FATAL: Module zaptel not found.
00:05.12jayteetype locate ztdummy
00:07.16`paulall in /usr/src/zaptel2/kernel/ztdummy.o
00:07.23`pauli mean /usr/src/zaptel2/kernel folder
00:07.30`pauland /usr/src/zaptel2
00:08.06jayteesomething isn't right with your compile/install
00:08.32`paulill try to compile it again
00:08.36jayteewait
00:08.42`paulyes?
00:08.51beek`paul: You did a "make install" as root?
00:09.09`paulbeek: yup
00:09.11jayteebefore you do, go into each of your source folders for zaptel and asterisk and do a make clean
00:10.43`paulok im done ill compile zaptel first ok?
00:11.07`paul./configure returns    configure: *** Zaptel build successfully configured ***
00:11.26`pauli dont have zaptel hardware i just need it for the meetme application
00:11.49jayteethen follow this guide, it's worked for me every time except when I use Asterisk 1.4.22 with the latest zaptel. I'd recommend using Asterisk 1.4.21 and zaptel-1.4.10 or 1.4.11
00:11.52jayteehttp://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
00:12.12`paulim using 1.4.19
00:12.53jayteethe older version of the tarball for zaptel can be found here: http://downloads.digium.com/pub/telephony/zaptel/releases/
00:12.54Talkradiois there a setting in asterisk for how long the message has to be played before extension dialing works?
00:13.20Talkradioyou gotta wait like 5 seconds or so right now before they work
00:13.38*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
00:14.07jaytee`paul, if you use that howto from the wiki after cleaning out your existing stuff in /usr/src/ for asterisk you should be fine
00:14.47`paulis libpri required for zaptel (ztdummy)
00:14.49jayteeTalkradio, what message?
00:14.53Madkisshi all
00:14.55jaytee`paul, no
00:15.02Talkradiothe ivr message
00:15.17jaytee`paul, libpri is only for pri circuits
00:15.32Madkissi am having some iax-problems. I had a setup with one iax-connection that worked nicely. now I added iaxmodems on both sides, and I can't send faxes between the two sides of the iax-link because obviously, the remote-asterisk expects my side to authenticate as iaxmodem
00:15.33jayteeif you have a T1 card you'd need it
00:15.45Talkradiowhen you call in to main number and it starts it's greeting it's like 5 seconds before you can dial an extension
00:17.03jayteeTalkradio, there may be a Wait statement in your IVR code. Could be any number of things. set verbose 10 and pastebin a call attempt
00:17.18Talkradiook
00:17.19jayteeand pastebin the extensions.conf
00:17.23Madkissapparently, the iax-driver tries to connect the incoming connection to one of the hylafax-contexts. which is bullshit, of course.
00:17.35Talkradiolet me remote in brb
00:18.31`paularrrggghhhh
00:18.33`paulLoading zaptel framework:  FATAL: Module zaptel not found.
00:18.33`paul<PROTECTED>
00:18.33`paulWaiting for zap to come online...
00:19.17`paulError: missing /dev/zap!
00:20.44jaytee`paul, cd into your /usr/src/zaptel folder
00:20.49jayteetype make clean
00:21.07`paulok
00:21.20`pauldone
00:21.37jaytee`paul, just thought of something. Is SELINUX disabled?
00:21.52`paulwahts that? how can i know?
00:22.10beek`paul: getenforce
00:22.52`paulgetenforce .... Disabled
00:23.08beek`paul: That's not the problem then.
00:23.16jayteeok, that takes care of that question
00:23.34beek`paul: make clean && make install and then pastebin all of the output
00:23.42jayteewait!!!!
00:23.44joatpaul, did you recently upgrade the kernel?
00:23.47`paulis there any other solution for conference...
00:24.04joatwhat beek said
00:24.15`pauli got kernel-devel a while ago
00:24.28jayteehe should run a make menuselect in zaptel, disable the unnecessary drivers you don't need but you'll need ztdummy and ztdynamic
00:24.31`paulyum install kernel-devel
00:24.34joatwouldn't have mattered unless you use a new kernel
00:24.48joatthen you need to remake zaptel (or dahdi)
00:24.56`paulok ill go
00:24.58`paulmake clean
00:25.01`paul./configure
00:25.05`paulmake menuselect
00:25.18`paulthen select ztdummy and ztdynamic right?
00:25.48joatdon't remember if it's that granular... make zaptel?
00:25.58jayteeyes, and if you want you can comment out any of the other modules like torisa, wctdm, wcfxo etc.
00:26.00joathow old is that version?
00:26.30`paulwhich one joat?
00:26.38jayteejoat, he's got asterisk 1.4.19 but I don't know which zaptel version
00:26.52joatfor conference?  ztdummy if you don't have hardware
00:27.19joats/conference/meetme/
00:27.24jayteewhich fails when he tries to start the zaptel service. it can't find the modules because they haven't been compiled
00:27.43joatodd...
00:27.46`paulok ill paste make output
00:27.55jayteewhen I had him do a locate it could only find the object files in the source folders
00:28.09Madkissas soon as i remove the iaxfax-context-stuff, it's working nicely again
00:28.26`paulhttp://pastebin.com/m2be759ae
00:28.31Madkisscan anyone point me to the right direction? i am out of ideas
00:28.32`paultehre
00:29.18joatyou did "make install"?
00:29.30`paulyup done with make install
00:29.37joatto ask the obvious (sorry)
00:29.49joatlsmod|grep zt
00:29.55jayteemake config
00:30.21`paullsmod grep doesnt return anything
00:30.31`pauli did make config
00:30.31jayteebecause it hasn't been loaded yet
00:30.52jayteeok, now do chkconfig zaptel on and then service zaptel start
00:30.57`pauldid you see the paste bin?
00:31.01jayteeyes
00:31.03`paulanything wrong ther?
00:31.09jayteenot that i can see
00:31.15joatdidn't error out
00:31.37`paulLoading zaptel framework:  FATAL: Module zaptel not found.
00:31.39beek`paul: I don't see any sign of the "make install" in that pastebin
00:32.00`pauli did a make> output.txt
00:32.13`paulso thats only make
00:32.16jayteeyeah, I didn't see that either but he just said he was pasting the make
00:32.18`pauli did make install after
00:32.27beek`paul:  *THAT* was the output I wanted to see.
00:32.27jayteeany errors during make install?
00:32.36`paulok wait
00:32.42`paulill paste make install
00:32.45joatdoes "modprobe ztdummy" error?
00:33.15jayteeif he does a make install he shouldn't need to modprobe the module, it will do it for him
00:33.36`paulhttp://pastebin.com/m389e20a
00:33.47joattrue he's received a module not found error then reinstalled
00:33.48`paulmake install > output.file
00:34.27`pauli mean ive already done this on another server why wouldnt it work on this one
00:34.43beek`paul: do a uname -a
00:34.56beekWhat's the output
00:35.54beek`paul ?
00:36.01`paul2.6.18-8.el5 #1 SMP Thu Mar 15 19:57:35 EDT 2007 i686 i686 i386 GNU/Linux
00:36.07`paulXXXXXXXX.com 2.6.18-8.el5 #1 SMP Thu Mar 15 19:57:35 EDT 2007 i686 i686 i386 GNU/Linux
00:36.40jayteeis this CentOS 5.1?
00:36.50joatsince you've run "make install", try "lsmod|grep zt"
00:36.51`paulhow do i check?
00:37.08beek2.6.18-92.1.22.el5.centos.plus-i686
00:37.26beek`paul: reboot your computer to get the correct kernel loaded.
00:37.38`paulafter reboot
00:37.48`paulill just do nothing?
00:38.18`paulwell i cant reboot the server right now >.<
00:38.26beekYou're mismatched with kernel-devel and what you're running.   Reboot your computer and go through the steps again.
00:38.41`paulok
00:38.53jayteebeek, good catch!!!
00:39.04`paulmaybe ill try that later and be back here tom same time.....
00:39.24beek`paul: It'll take just a couple of minutes... aren't you curious?
00:39.42`paulargh
00:39.49`paulits being used right now
00:40.04beek`paul: I suspect that will take care of your problem.
00:40.06`paulill email you of the result if you like
00:40.08joatpasses a plaque to beek
00:40.09`paul:p
00:40.17jayteeit's odd that he'd have the kernel-devel for a newer kernel but still loading an older one, don't ya think?
00:40.30`paulcause a while ago
00:40.38Kobazbluh
00:40.39`paulits looking for the source right
00:40.44Kobazthese avaya boxes are kinda flakey
00:40.55Kobazsecond tone clock fried in like 6 months
00:40.56`paulkernel source
00:41.00*** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net)
00:41.09`paulso i did an yum install kernel-devel
00:41.28beek`paul: just ensure that you do a yum update so that you have the newest kernel & kernel-devel packages.  Reboot and go through the aforementioned steps.  It will work as advertised then.
00:41.53beekthanks joat for the plaque
00:42.19*** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com)
00:42.24`paulim doing yum update right now
00:42.27`paulthanks guys
00:42.31`paulill reboot it later
00:42.55joatKobaz: don't like the sound of that... we just installed one at work
00:43.47beek`paul: while you're downloading I think that you should take jaytee's advice regarding the versions.
00:43.59Kobazjoat: why not an asterisk box?
00:44.14`paulwhat bout versions?
00:45.10joatKobaz: wasn't our choice, corporate headquarters overrode our asterisk proposal
00:45.14jayteeif you're using asterisk 1.4.19 I'd recommend using zaptel 1.4.10 or 1.4.11
00:45.18Kobazthose punks
00:45.35joatheh
00:45.36Kobazjoat: this is like a 15 year old box, perhaps the new ones are better
00:45.47jayteeI'm using zaptel 1.4.10 with asterisk 1.4.21 and it works fine
00:45.50joatokay, i'm less nervouse
00:45.58joats/nervouse/nervous/
00:46.02Kobazbut the tone clock was new
00:46.08joatcan't type worth a dang tonight
00:46.20Kobazthey seem to be about 5 bucks on ebay
00:46.21joatgremlins then
00:46.28joateatin' the cards
00:46.56Kobazhttp://cgi.ebay.com/Avaya-Lucent-Definity-TN2182B-Tone-Clock-V1-card_W0QQitemZ200289846818QQcmdZViewItem
00:48.37joatthat go in a giant blue cabinet?
00:49.05joator was that the infinity (it's been awhile)
00:49.42*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-22db24a2d8f7d4b2)
00:50.19Kobazgrey
00:52.16Madkissjesus. with a dialstring like IAX2/FOOBAR, why does iax assume that the user who wants to connect is JOHNDOE just because there is a JOHNDOE context in iax.conf as well?
00:52.58jayteenaval grey like Nortel?
00:55.04*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
00:55.37*** join/#asterisk telnettech (n=telnette@199.2.112.71)
00:56.07telnettechok who in here is in a European country and can help me get my zaptel config right?
00:56.18jayteeI remember when I was 5 thinking "when I grow up I wanna be a firetruck" but then later on I let family and friends influence my career choices and look at me now. stuck in IT working on VOIP crap for peanuts.
00:56.43jayteetelnettech, are ya using CCS or CAS?
00:56.49telnettechCCS
00:57.03jayteeand HDB3?
00:57.08telnettechyes
00:57.23telnettechcause of the zaptel problem, my sound files dont play
00:57.58jayteeand this is the ISDN30 link?
00:58.06telnettechi can do a noload =>chan_zap.so in the modules.conf file and then start asterisk without starting zaptel and they play fine
00:58.10telnettechyes
00:58.33jayteewow, that's a weird one
00:58.47jayteewhen the link is up can you make calls?
00:59.24telnettechwell i dont know yet cause the Aruban Telco hasnt installed my ISDN30 yet
01:00.42telnettech;:
01:00.56jayteeso the server is in Aruba?
01:02.48*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-221-124.phlapa.east.verizon.net)
01:02.57telnettechyes with me
01:04.41Madkisswhere is [TK]D-Fender if you need him? *sigh*
01:06.05jayteeyou're back in Aruba? geez, nice gig!
01:06.47Kattyseanbright: :<
01:06.49Kattyhugs seanbright
01:10.10jayteetelnettech, when you've got zaptel loaded and you try to play sound files do you get any console errors?
01:10.30telnettechno errors.....it shows that the sound file is playing
01:11.18jayteesame test phone, same file each time?
01:11.20telnettechit wont play mulitple sond files, ie when you are checking voicemail messages, it doesnt play all the of the menu option files
01:11.25telnettechyes
01:11.43telnettechi even tried to use my xlite to play and i cant hear anything on laptop
01:12.20jayteebut if you unload zaptel it all works fine?
01:12.30telnettechyes
01:12.47jayteeasterisk and zaptel versions?
01:12.50telnettechif i stop zaptel and tell asterisk to not load the chan_zap.so it works fine
01:13.21telnettechasterisk is 1.2.28 and zaptel is 1.2.21
01:13.46jayteehave you upgraded zaptel after installing asterisk?
01:13.54beekGN guys
01:13.58jayteenite beek
01:14.05Zippomancan someone help me out
01:14.23telnettechi tried to use the 1.2.27 zaptel but it doesnt have the zd_ethmf module that I need for the TDMoE
01:15.11jayteemakes sign of the cross and mutters, "fuckin TDMoE"
01:16.44jayteetelnettech, Redfone?
01:16.50telnettechwe didnt have luck with internal cards when they started this and so this is what we have come with
01:16.54telnettechyes
01:17.30jayteesorry man, I wouldn't touch one of those with ....... hmmmm......anything
01:17.44telnettechjaytee: that is where i got the zaptel and libpri tars from
01:17.45jayteebet their tech support is awesome though! :-)
01:18.12jayteeso when  you got those did you recompile asterisk?
01:18.25jayteeafter compiling those tars?
01:18.29telnettechi loaded them from the start
01:18.59jayteeso you went libpri, zaptel, asterisk in that order? with no upgrades to any after the fact?
01:19.06telnettechbefore asterisk...i have reloaded this server 4 times since i startd this install in mid december
01:19.08Kattyjayyyyyyyyyyyteeeeeeeeeeeeeeeeeeeeeeeeeaaaaaaaaa
01:19.28jayteeKatttttyyyyyyyyyyyyyyyyyy
01:19.35telnettechthats the order....i had to look at the workbook from class in the beginning to make sure i did it right
01:19.38jayteethinks of the movie WALL-E :-)
01:19.47Kattyhaha
01:19.54Kattyeeeeeeeeeeeeevvvvvvvvvvvaaaaa?
01:20.01jayteeheheehe
01:20.15jayteeI love the scene where he finds a spork and tries to add it to his collection
01:20.19Katty*blinkblink*
01:20.22Katty*plant on head*
01:20.25Kattyeeevaaa?
01:20.52jaytee"Try blue! It's the new red!"
01:20.56KattyOoooo
01:21.52jayteeok, now i have to watch it again!!!
01:22.18jayteejust moved my Totem player over to my right hand 19" monitor. Nice to have dualies
01:22.20Kattyi like the pizza plant comment at the end.
01:22.47jayteeMy New Year's Resolution: 2880x900 FTW!!!
01:22.53Kattyhehe
01:23.08Kattyi want a chair like that
01:23.40telnettechcool....the singapore development crew just came online and they are lookng at it
01:23.42*** join/#asterisk sack (n=sack@222.Red-83-49-103.dynamicIP.rima-tde.net)
01:24.17drmessanoI have my team in Tunis looking at an issue for me right now
01:24.24jayteelol
01:24.32telnettechwhere are you dr:
01:24.50drmessanoKuala Lumpur
01:24.58telnettechim in Aruba
01:24.59jayteehe's in Georgia being his usual wiseass self
01:25.07telnettechit is tuesday night
01:25.11jayteewhich is why we all love him so much
01:26.27*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
01:26.28carrarEntrapment was a great movie
01:26.35*** join/#asterisk Maliuta_CA (n=biteme@59.167.238.153)
01:26.57jayteeyeah
01:27.06carrar(speaking of Kuala Lumpur)
01:27.07carrarheh
01:27.10jayteeanything with Catherine Zeta Jones is worth watching
01:27.15carraryeah
01:27.25telnettechyeah it was but my favorite is the Bourne trilogy for action
01:27.38jayteeyeah, I got those on DVD
01:27.46drmessanojaytee: Guess what.. my wish is coming true
01:27.53drmessanoAustralia's use of coal and carbon emissions policies are guaranteeing the “destruction of much of the life on the planet”, a leading NASA scientist has written in a letter to Barack Obama. In the letter he says: "Australia exports coal and sets atmospheric carbon dioxide goals so large as to guarantee destruction of much of the life on the planet"
01:27.57telnettechand all the Bond movies except the ones with Timothy Dalton
01:28.13drmessanoWar with Australia!
01:28.24jayteetelnettech, ugh! timothy dalton
01:28.33jayteeI was never a big fan of Roger Moore either
01:28.33telnettechhe was terrible!!!!!
01:29.00jayteeI think Timothy Dalton and Steven Seagal went to the same acting school
01:29.22telnettechroger moore was ok not as good as sean himself but he was better than Timothy Dalton
01:29.35jayteeyes, I'll give him that!
01:29.48telnettechno steven seagal was in the CIA....didnt you know that ?
01:30.03drmessanoCant I Act?
01:30.11jayteefor real? I didn't know that. that explains the lousy acting
01:30.41telnettechwatch TNT sometimes.....they like to have him do action movies for just their channel
01:30.51telnettechhe is a fat guy like us
01:31.26drmessanoOnly Steven Segal would do a direct to video action movie
01:31.31drmessanoLike Under Siege 5
01:31.48telnettechthere is one where he is in vietnamese pajamas
01:31.53telnettechwhat a laugher he is
01:31.56jayteewho you callin fat? :-)
01:32.29drmessanoI always wanted what kind of person it takes to do a direct to video sequel of a hit movie.. Like Home Alone 4
01:32.36drmessanoOr Beethoven's 6th
01:32.41drmessanoThen I look at Steven Segal
01:32.46telnettechwell ok like me....im no spring chicken.....thats what my 13 yr old told me last week
01:32.50drmessanoand I am like "Ah, I see what I did there"
01:35.08drmessanoSomeone needs to make a T.38 fax modem driver for Windows
01:35.54telnettechdr: are you running asterisk on a windows server?
01:36.07drmessanoNo
01:36.40NovceGuruhttp://upload.wikimedia.org/wikipedia/commons/9/9b/T.30_Protocol_Figure_01.jpg
01:36.46NovceGuruthat looks like it was made in mspaint
01:36.59drmessanolol
01:37.20drmessanoUm
01:37.35drmessanoWOW
01:37.41drmessanoSomeone does make one.. $138 per channel
01:37.59drmessanoGood god thats nuts
01:38.33NovceGuruwho would bother buying it?
01:38.47drmessanoFor that price, no clue
01:39.00jayteehttp://www.entertainmentearth.com/prodinfo.asp?number=AU11727
01:39.03NovceGuruya
01:39.09drmessanoFor under $40 I would
01:41.28drmessanoI have an application that is going to be sending weather warnings to hearing impared folks with fax machines
01:41.52drmessanoand right now my answer is Modem >> ATA >> Asterisk >> Provider
01:42.20drmessanoId like to cut the modem and ATA out of the mix and use software to emulate it
01:42.36jayteehylafax?
01:42.48drmessanoHas to be on Windows
01:42.52jayteelearning curve makes climbing K2 look easier
01:45.11drmessanoIm at the point of rebuilding our project and removing points of failure
01:45.28drmessanoand I see the modems and ATA as being a problem
01:46.21drmessano$189 for 2 channels <---- Bargain
01:46.25jayteeso you go from provider to asterisk to ATA to modem to the hearing impaired person's modem and computer?
01:46.42jayteeor their fax machine
01:47.35*** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
01:47.44drmessanoNo, i'm sending a fax over the PSTN to their Fax machine on a PSTN line.. But in order to do it over VoIP, I need modems (which are external serial modems), and a T38 aware ATA
01:47.52drmessanoWhich is a cluster
01:48.23jayteehave you talked to coppice about this?
01:48.35drmessanoIm sure he doesnt like Windows :)
01:49.12jayteeso you're using a Windows host to get the fax info and distribute it?
01:49.34drmessanoThe aggregator server application is windows based
01:50.42drmessanoWithout software, it needs to leave the box as an analog fax call
01:50.43NovceGurudrmessano: could you use a hylafax virtual printer on the windows based aggregator and hylafax ?
01:50.52*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
01:50.55drmessanoHmmm
01:52.21NovceGuruthere's http://winprinthylafax.sourceforge.net/ but I prefer http://whfc.uli-eckhardt.de/ as it lets you dynamically fill out the cover letter info
01:52.25drmessanoI dont believe so.. It would need Windows faxing to believe it's a modem, as the server app ghooks into it
01:52.28drmessano-g
01:52.35drmessanoLemme look to be sure
01:52.38NovceGurudoh :\
01:53.42*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
01:54.40*** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net)
01:54.56drmessanoLooks like Windows Faxing or Winfax.. Which means it would need Hylafax support
01:55.52NovceGuruWinfax....what a piece of complete utter shit
01:56.19drmessanoIndeed
01:56.26drmessanoI knew someone who bought it once
01:56.30drmessanoIt was hooooorrid
01:57.17*** join/#asterisk andresmujica (n=andresmu@190.27.1.233)
01:57.21NovceGurume too! Fought with it for years and then just installed windows fax
01:57.49drmessanoWe have several businesses that live off windows fax.. One of which has a cluster of 6 modems
01:57.52drmessanoand it works great
01:58.44NovceGuruHaha thats awesome. I know a guy that manages clusters of hylafax servers, I think they have > 1000 PSTN lines coming into their setup
01:59.01NovceGuruhmm thats scarry, last release of hylafax was dec 2007
02:00.10drmessanoSeems writing a T.38 modem emulator is more work that writing a printer emulator to hook into App X
02:00.33drmessanoTheres a couple softphones that have T.38, create fax printers
02:00.44NovceGuruhm
02:02.14telnettechwe use procomm plus
02:02.42telnettechthe lady in the admin office uses it to fax all day any work orders and other stuff out of the company
02:03.23*** join/#asterisk Kumbang (n=dsp@167.205.24.69)
02:04.19NovceGuruI have a company I've been trying to get to use paperless faxing for years
02:04.36NovceGuruinstead of printing and sticking in the fax machine...just print to the other damn printer in your list
02:04.50telnettechwe also have a fax server that has adobe on it so incoming faxes are pdf for filing away
02:04.54NovceGuruand :O type in your cover sheet details instead of scratching in with a pen
02:05.12NovceGuruyou don't need adobe to pdf faxes
02:05.56telnettechthat is just what they have setup
02:06.12NovceGuruyeah, it's very nice either way :)
02:06.39NovceGuruespecially with foxit readers annotation/typewriter feature, you can just type write on the fax and file print --> fax printer back to them
02:06.50NovceGururight
02:09.53drmessanoI found ONE commercial product
02:10.03drmessanoNot even something to compare it to
02:11.52f0urtyfiveanyone have a sip connectivity test?
02:11.59f0urtyfiveor something that I can give a sip URI to ring?
02:12.01drmessanoI swear this whole fax thing is so confining
02:12.07f0urtyfivelike a little web app
02:16.22drmessanoAh well
02:17.08NovceGurudrmessano: yeah seems a little convoluted
02:20.35telnettechjaytee: you still in here
02:21.09jayteetelnettech, nope! I'm over there >>>>>>>
02:21.26drmessanoOMFGGGGGGGGGG
02:21.45drmessanoBrooktrout SR140, 2 channel Fax over IP solution for Windows
02:21.45telnettechjust checking....must be a good movie or you are constipated
02:21.49drmessano$1800
02:21.50*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
02:21.50*** mode/#asterisk [+o russellb] by ChanServ
02:21.56telnettecheither way you have been quiet
02:22.09jayteeyeah, watchin WALL-E again
02:22.13drmessano$1800 to emulate a fax
02:22.15jayteeand eating some ramen
02:22.28drmessanojaytee: do you have a ferris wheel on your lawn?
02:22.38jayteeno
02:22.39drmessanoI'm just sayin..
02:22.58jayteeI'm missing a connection here?
02:23.25drmessanoAll the disney movies, the 1981 Chevy Caprice in mint condition, thick rimmed glasses.. Polyester pants...
02:23.30drmessanoPeople talk
02:23.35drmessanoJust sayin
02:24.17telnettechdr: you see that in a movie that americans have ferris wheels in their yards?
02:24.48drmessanoWhat the heck are you talking about?
02:25.02drmessanoI was implying something ELSE.. not that he's an AMERICAN
02:25.36jayteedrmessano, put the bong down and step away from the stash
02:26.24jayteeI wear rimless glasses, don't drive a 1981 Chevy Caprice and hate polyester and would never buy any clothing that has it, even a blend
02:26.25drmessanojaytee: Take the melted milk duds out of your pockets and hand over the Raggety Ann dolls
02:27.16drmessanoActually, telnettech was right.. I was just trying to prove you're an american
02:27.31drmessanoBy asking you if you had a ferris wheel on your lawn
02:27.37drmessanoHa, caught you..
02:27.43*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
02:28.05telnettechwhat movie did you see that in....i have got to find it and watch it
02:28.18telnettechthat is so stereotypical
02:28.51drmessanoOh, I so know
02:30.04jayteewhat movie was that in?
02:30.20drmessano1941?
02:30.25telnettechoh
02:30.40telnettechi remeber the scene now with john belushi flying the plane
02:30.42telnettechlol
02:31.13drmessanoI just remember the uber dorky nerd guy from Wargames on the ferris wheel spinning down the dock
02:31.47*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
02:32.13jayteeMatthew Broderick? or the guy that played Melvin?
02:32.18*** join/#asterisk Defraz (n=T0tal@gump.fuzecore.com)
02:32.31drmessanoMelvin
02:33.01drmessanoThe one fro the "Mr Potato Head" scene
02:33.16drmessanoMR POTATO HEAD, MR POTATO HEAD, BACKDOORS ARE NOT SECRETS
02:33.32drmessanoI think....
02:34.05jayteethe one who's screaming Mr Potato head or the dweeb he's screaming at?
02:34.19drmessanoNo, referring to the dweeb
02:34.30telnettechwas matthew broderick in the movie?
02:34.36drmessanoYes
02:34.39drmessanoHe was the star
02:34.43drmessanoBut you know
02:34.44jayteeHerbie Kazlminsky
02:34.52drmessanoFunny shit here.. we were talking about that today
02:35.02jayteein real life Eddie Deezen
02:35.27drmessanoDid you notice that Norad, with all its government money, had the same voice synthesizer as David?
02:36.16telnettechno i dont normally talk with NORAD
02:36.45drmessanoIM TALKING ABOUT IN THE MOVIE
02:36.51drmessanoGod, keep up
02:37.04jayteeMaury Chaykin played Jim, the screaming guy and also played Sgt Downs in Iron Eagle II and the Major in charge of Kevin Costner who would piss himself and ended up shooting himself in the head in Dances with Wolves.
02:37.35drmessanoGood god
02:37.35telnettechok so you are talking about wargames right DR
02:37.54drmessanotelnettech: No, I am talking about my many visits to Cheyenne Mountain
02:37.55telnettechhow is that related to kevin bacon jaytee....lol
02:38.29telnettechi never have been there but you are saying that the voices in real life Norad are the same as the computer, David, from wargames?
02:38.43jayteeand years later that same voice synthesizer, in spite of all the advancements like Cepstral and other speech engines, is still being used by Stephen Hawking.
02:38.56drmessanoROFL
02:39.01`paulei jaytee
02:39.04`paulbeek
02:39.16drmessanoI would LOVE to meet stephen hawking... Just to get him to voice that
02:39.17jayteeI was at NORAD in the late seventies and we had no voice synth
02:39.22`paulit worked!!!!!!!!
02:39.28`paulremember me?
02:39.40drmessano"Hey man, can you ask me "Shall we play a game".  Please?"
02:40.23jaytee`paul, yeah we remember! so you switched to an older zaptel version and it worked?
02:40.30`paulnope
02:40.35`paulafter the reboot
02:40.45`pauli compiled again
02:40.45*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
02:40.46jayteeah, so the kernel mismatch WAS the problem.
02:40.50`paulthen it worked
02:40.51`paulyeah
02:40.58`paulthanks guys
02:41.17drmessanoThe kernel may be dead, but we're still enjoying his chicken
02:41.40[TK]D-FenderRoad-kill Helper.... makes a great meal!
02:41.41jayteeweird that you'd be booting with an older kernel. You must have run yum install kernel-devel withouth doing a yum -y update first
02:42.01`paulyeah exaactly
02:42.13drmessanoStop with all the Linux talk.. I cant keep up
02:42.22drmessanoLets talk more about DLLs
02:42.26`paullol
02:42.28jayteedrmessano, I recommend adderal or provigil
02:42.45`paulhas the exploit for IE been fixed
02:43.09`paul?
02:43.11jayteeyep, all the holes in Windows have been plugged (smirk)
02:43.23drmessanoSorry, jaytee, I wasn't paying attention
02:43.25drmessanoWhat?
02:43.29drmessanoOh look, purple
02:44.01drmessanoI like how every drug that makes you more sane has to make you fatter too
02:44.05drmessanoMy goal for 2009 ---> Crazy and thin
02:44.19`pauli downloaded the sample in wer a javascript and xml can make the calculator run
02:44.19jayteeif cheese was an operating system's security, Windows security would be Allsace Lorraine swiss
02:44.41drmessanojaytee: A straight person would have stuck with "Swiss"
02:44.44drmessanoBut ok.........
02:45.19jayteedon't forget big fella, I rollerblade! ;-)
02:45.46drmessanoSee that moves the needle in the other direction
02:46.00drmessanoWhen I think "Not straight", I think of standard rollerskates
02:47.00drmessanoI blame Hollywood, and years of my dad calling me ugly names because I wanted to be in the Ice Capades
02:47.01drmessano:(
02:47.31*** part/#asterisk `paul (n=temp_acc@122.55.36.3)
02:48.35drmessanoOk, I got it
02:48.53drmessanoI need a TDMoE modem emulator for windows
02:50.35telnettechdont use TDMoE.....that is my problem tonight
02:50.44*** join/#asterisk brian (n=brian@unaffiliated/brian)
02:50.51telnettechand my development team and figure it out either
02:50.57telnettechi am in big trouble
02:51.38drmessanoI doubt there's a TDMoE anything for windows
02:51.43drmessanoBut I will take your advice
02:52.52telnettechjaytee: development guy is cussing refone...lol
02:52.59drmessanoThing is..
02:53.24*** join/#asterisk mog (n=mog@c-68-62-217-121.hsd1.al.comcast.net)
02:53.24*** mode/#asterisk [+o mog] by ChanServ
02:53.32drmessanoTDM card with 2 FXS to cut the ATA out of the loop, still puts me in the $150 category
02:53.44drmessanoFor a cheap knockoff card, that is
02:53.56drmessanoStill need the modems and cabling
02:54.09drmessanoThat $188 sounds kinda like a bargain lol
02:55.18jayteesorry, I was roasting coffee
02:55.30jayteetelnettech, sounds like par for the course with Redfone
02:55.36drmessanoFrench Coffee?
02:55.43jayteeTDMoE is sucky anyways
02:55.50jayteeFrench Coffee? no
02:56.01jayteeKaui from Hawaii
02:56.20telnettechit looks like they are gonna stay with TDMoE cause they want to use thin rackmount servers
02:56.33drmessanoI only drink Kopi Luwak
02:57.27[TK]D-Fendertelnettech: ludicrous
02:57.28jayteethe pain comes when you have to debug issues on a TDMoE span. Not as straightforward as hardware. I like my T1 PRI on a digium card just fine
02:58.05[TK]D-FenderTDMoE = Retard's Choice... fresh brewed, just add water
02:58.29telnettechwe havent had any problems until this install with them
02:58.34telnettechbut if the zaptel is running, then my sound files play but i cant hear them
02:58.48[TK]D-Fendertelnettech: Stupid dead-end tech that doesn't play with anything else.
02:58.56telnettechthat is the only thing holding me back
02:59.13jayteetelnettech, [TK]D-Fender's right. and the rationale for using TDMoE is flawed. Even a Dell 1750 1u rackmount server has a pci slot in it.
02:59.28telnettechhe is part of a bigger development team.......unfortunately he is caught in a numbes game for dev mgr
02:59.32[TK]D-Fenderjaytee: more than 1....
03:00.11telnettechi know....i do read a little even though i dont play alot with stuff....and like i said I have been dong tis for 4 months
03:00.11[TK]D-Fenderjaytee: And thats before going with an external solution that ISN"T a flaming piece of shit...
03:00.27telnettechim learning networking, Linux, scripts, and asterisk on the fly
03:03.04jaytee[TK]D-Fender, there's a small riser card with two pci slots but I think the one on the right side of the chassis is only for a RAC card.
03:03.24telnettechok guys....calling it a night....gotta get up early to be on a conference call with ops mgr and dev mgr about this site
03:03.40jayteetelnettech, nite guy. good luck!!!
03:03.50NovceGurusounds fun, later
03:04.03*** join/#asterisk brian (n=brian@unaffiliated/brian)
03:04.46jaytee[TK]D-Fender, telnettech's a decent guy. Kind of got thrust into a role he wasn't prepared for though. He and I were in class together back in November.
03:07.52jayteeI'm just glad I got good advice here about staying away from Redfone when I was first investigating using Asterisk with PRI back in late 2006. The Asterisk consultants my boss brought in for a bull session tried to sell us one.
03:08.34jayteeeverything I've read in forums or seen from people in here paint them out to be exactly the nightmare piece of junk that you and ManxPower said they were.
03:09.07NovceGuruI just picked up a TDM404EF, hopefully it was a good choice
03:09.42drmessanoI think [TK]D-Fender is just jaded because he couldn't make TDMoE work
03:09.49drmessanoand its been his arch nemesis
03:10.06drmessanoHence the TDMoExcitement
03:13.58drmessanoUm
03:14.07drmessanolol j/k?
03:15.46*** join/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net)
03:19.43NovceGuruI think you pissed him off
03:21.25*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com)
03:23.02drmessanoI doubt it
03:23.37drmessanoI think [TK]D-Fender is part italian.. He doesn't get quieter when you piss him off
03:25.04russellbdrmessano: i thought it was funny, heh
03:26.38drmessanoI'm sure TK is busy.. I've never not seen him go all drmessano about TDMoE..
03:27.56drmessano[TK]D-Fender: Can you recommend a good Grandstream based TDMoE solution
03:28.02drmessano[TK]D-Fender: I am a bit stuck
03:28.29NovceGuruthrow on the gasoline!
03:30.24jayteea French Canadian from Quebec who's part Italian? Wow! talk about putting nitroglycerin in the blender and punching Puree!
03:31.52*** join/#asterisk cabbey (n=cabbey@24-159-193-106.static.roch.mn.charter.com)
03:32.12*** join/#asterisk Subdolus (n=subby@subby.afraid.org)
03:34.37jayteeoh, god! the environmental whackos are arguing about nuclear power and bashing hippies in #ubuntuforums. looks like a fun time's about to start
03:34.42cabbeyare there any good docs for the end users of an asterisk system? All the docs I'm finding are for the administraitors of the pbx
03:35.06jayteeyou need a manual to figure out how to dial a phone?
03:35.32cabbeytrying to find one that handles simple stuff like "how do I call another extension?"
03:35.44drmessanoOk, I GOT IT
03:36.11jayteeya pick up the handset and you punch the keys and if your admin did their job you shouldn't even need to press a Dial or Send key
03:36.21NovceGurucabbey: wouldn't that partly depend on the phone?
03:36.27drmessanoWindows Serial Modem Emulator >> Serial cable >> Linux Box running PPP >>> Out fax modem >> ATA >>> Asterisk >> ITSP
03:36.36drmessano7 hops, nothing but net
03:36.40NovceGuruHOLY HELL
03:36.45cabbeywell then, aparently our admin didn't :)
03:36.47jayteedrmessano, sounds like a win! :-)
03:36.54drmessanoIm all over that shit
03:36.58jayteecabbey, what kind of phone is it?
03:37.09cabbeypolycom 330
03:37.13jayteePolycom? Grandstream?
03:37.23jayteeah, got one right here. do you know someone's extension?
03:37.28cabbeyyep
03:37.40cabbey706 is the pbx admin :)
03:37.52jayteepick up the handset, press the keys for the extension and press Dial in the upper left above the keypad
03:38.02NovceGurushould be able to dial 7xx and it go go go
03:38.12NovceGurujust sayin!
03:38.23cabbeythis is our first attempt at a remote phone that's not on the same lan...
03:38.42cabbeyheh, that's what I thought, but no go
03:39.15jayteehttp://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html
03:39.16NovceGuruhaven't googled at all...does SIP traverse an openvpn...vpn without any trouble?
03:39.53jayteethere's a link for documentation on the 330. You'd need to know the SIP version to download the correct User guide but the 2.2 version will probably suffice. it's a pdf.
03:40.16cabbeyoh hey, if I let it sit long enough it eventually gives me a fast busy signal three times and then goes back on hook
03:40.17jayteeah, not on the same lan. NAT issues
03:40.23cabbeyyeah
03:40.33jayteeor bad dialplan digitmap on the phone
03:40.50NovceGurucabbey: I'd suggest a vpn
03:41.05jayteei'd suggest a new admin :-)
03:41.07NovceGurualthough I know nothing about your setup
03:41.08cabbeyyeah, that's on the todo list :)
03:41.10NovceGuruthat too, heh
03:41.36cabbeyhmm... tcpdump on my router says a pile of traffic out on udp 5060... but nothing coming back
03:42.05NovceGurumight be an issue
03:43.09*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
03:45.57drmessanoMaybe theres a diode on the line
03:46.02drmessanoEasy to remedy
03:46.17drmessanoUnplug all equipment, and grab a frayed extension cord
03:46.34NovceGuruthe one plugged into the wall?
03:46.37drmessanoThe kind that not only holds up the corner of the end table, but has been powering the lamp on it for 20 years
03:47.00drmessanoCut off the business end, shove it in the smartjack
03:47.15drmessanoThen, plug the other end in a convenient wall outlet
03:47.20drmessanoShould clear that diode right out
03:47.25*** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)
03:47.34cabbeywith a nice pop sound and some blue smoke to boot
03:47.57drmessano"blue smoke" <--- only if you've been a good boy this year
03:48.13cabbeyheh
03:48.41drmessanoMay also be a firewall problem
03:49.06drmessanoIn which case, the criminal destruction of telco property *PROBABLY* didn't solve your issue
03:49.26cabbeywell, I'm starting to think my problem here ain't at my end... thanks for confirming my sanity that dialing an extension really should have been that simple... I"m going to send these traces along and call it a night
03:49.45drmessano"aren't", not "ain't'
03:50.02NovceGuruatleast he used proper punctuation
03:50.11drmessanoAint that the dang truth
03:50.12cabbeyheh
03:50.31*** part/#asterisk cabbey (n=cabbey@24-159-193-106.static.roch.mn.charter.com)
03:51.26drmessanoOh god..
03:58.18jayteeHe's out at the moment, leave a message at the beep and He'll get back to you. BEEP
04:02.50*** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr)
04:14.21x86wtf is astcanary?
04:15.17x86jaytee: when my wife says "oh god" or "jesus christ", I simply reply "Yes?"
04:15.43jayteeastcanary is for asterisk realtime
04:15.55x86what does it do specifically?
04:16.03jayteeas long as the canary is "singing" everything is good
04:16.53x86realtime as in realtime configuration (SQL/LDAP-backed SIP peers, etc)? or realtime like realtime priority?
04:17.10jayteeI don't run realtime but as I understand it it's a process monitor.
04:17.17jayteethe first one
04:18.02jayteeit monitors the database connectivity
04:18.19jayteemight want to Google it for more info
04:18.59jaytee<PROTECTED>
04:20.45*** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net)
04:24.57*** join/#asterisk mosty (n=mosty@eth1426.vic.adsl.internode.on.net)
04:25.38*** join/#asterisk kisu_ (n=kisu@2001:5c0:1100:9900:5d00:b8cf:9894:f1a5)
04:28.04*** join/#asterisk jks (i=jks@193.189.93.254)
04:31.34NuggetI'll stop procrastinating tomorrow.
04:32.15drmessanoI never cease to amaze myself at the utterly useless info I retain
04:34.04drmessanoSitting here plugging in NWS products.. and for some reason I picked them up intentionally..
04:54.30*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
05:07.21*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
05:10.08jaytee[TK]D-Fender, what's up? connection issues?
05:10.31[TK]D-Fenderjaytee: Firefox bouncing like a jack-hammer
05:10.58jayteemine craps out after viewing too many YouTube vids
05:11.27[TK]D-Fenderjaytee: Same here, and BSODs on flash quite a bit lately
05:11.52[TK]D-Fenderjaytee: Which FUBAR's my watching The Daily Show and pisses me off to no end
05:11.53jayteeBSOD? Windows? your running Windows? ugh!
05:12.16[TK]D-Fenderjaytee: yeah... I do have Ubuntu on this PC as well.. just can't quite kick the habit...
05:12.59jayteeyeah, tough habit to break
05:13.22*** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano)
05:17.59*** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-6d67412bed415535)
05:18.48drmessanoHmmm
05:19.30*** join/#asterisk sucituanbo (n=full@c-24-21-121-148.hsd1.wa.comcast.net)
05:19.48*** join/#asterisk dandate (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net)
05:20.29dandatehey guys i am setting up an asterisk machine and was wondering if I could disable DHCP on my router for my cox cable modem and still have functionality
05:22.06*** join/#asterisk hi365_m (n=hi365@bzq-79-182-160-246.red.bezeqint.net)
05:22.38[TK]D-Fenderdandate: Networking is networking.  If things point the right way they work
05:23.02*** join/#asterisk orkid (n=orkid@unaffiliated/orkid)
05:23.42[TK]D-Fenderdandate: Which has nothing to do with *
05:26.44drmessanoO.o
05:27.06drmessanoIf you disable DHCP, it will limit your ability to use DHCP
05:28.41[TK]D-FenderSMRT
05:42.44drmessanoHmmm
05:47.08*** join/#asterisk reneger (n=reneger@dslb-088-078-123-090.pools.arcor-ip.net)
05:47.42LemensTSdandate: if you disable dhcp on the router you will just have to have all the devices under it as static ip's.
05:53.44*** join/#asterisk jtodd (n=jtodd@188.sub-70-214-33.myvzw.com)
05:57.08*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
05:59.18*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
05:59.24[TK]D-FenderW-T-F
05:59.40[TK]D-Fenderturn my back and a random reboot...
06:02.36*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
06:04.39jayteeit's a feature!!!
06:04.53jayteeVista's full of them! might wanna upgrade :-)
06:05.49LemensTSive had vista for 3 weeks, id have to say xp is more stable.
06:06.07jayteemost definitely
06:06.09LemensTSvista isnt as bad as i thought tho.
06:06.12jayteeand less of a RAM hog
06:06.25LemensTSi cant tell this laptop is a haus it runs great
06:06.35jayteenot as bad as it's made out to be if you've got a really good graphics card and enough ram
06:06.57LemensTSyea i havent noticed slow down at all. i got 3gig tho.
06:07.33jayteebut Windows 7 will be WAY better, even on older hardware. MS learned it's lesson the hard way (yet again) and listened for a change.
06:08.03jayteebut in the next 4 years things will evolve much more quickly
06:08.10LemensTSreviews said it was slightly better than vista
06:08.11[TK]D-FenderBringing in my X-10 gear tomorrow to hook my entryway buzzer to *
06:08.38*** join/#asterisk Cherebrum (i=jgarland@209.9.237.93)
06:08.50CherebrumSomeone try this out and let me know if it works ok.... http://www.tollfreetollfree.com/
06:09.07jayteewith the new i7 architecture all the old bottlenecks are gone and if MS codes to take advantage of the architecture's abilities then Win 8 or 9 will be incredible
06:09.15LemensTSdoes x10.com sell cameras or porn videos?
06:09.45Cherebrumx10 home automation stuff
06:10.03LemensTSlol look at their website
06:10.09LemensTSyou'll see what i mean
06:10.35Cherebrumthat site is UGLY with hot girls on it
06:10.43jayteeCherebrum, all I get is a small login box in the upper left corner. What am I supposed to do with that?
06:10.53drmessanoI need a windows app to monitor a parallel port for closures.. I bet its like $1200
06:11.07LemensTSjaytee: login and make a call
06:11.22LemensTSit started asking me to use my camera, i decided to quit
06:11.41jayteeit doesn't even let me login with the creds supplied
06:11.50LemensTShmm..it let me
06:11.53LemensTSie7
06:12.07[TK]D-FenderLemensTS: I always thought that was a lame joke...
06:12.07Cherebrumjaytee: click login
06:12.31Cherebrumjaytee: It doesn't seem to work in linux
06:12.38Cherebrumit works on windows and on my mac
06:12.49Cherebrumit's a flash based SIP softphone
06:12.53drmessanoYEAH JAYTEE.. NOT THAT YOURE STOOPED OR ANYTING.. PUT IN INFO AND HIT LOGIN, NOT ALT-F4
06:13.01[TK]D-FenderLemensTS: Sex sells, but every F-ING image... geez
06:13.13LemensTSi cant get half the sites to work in firefox/safari/iceweasal, etc
06:13.14*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
06:14.16LemensTSCherebrum: is this your site? i can test it if its safe
06:14.31jayteedrmessano, I just signed up on the Atlanta Craigslist using your email and forged creds and placed an ad in Men looking for Men "Bottom seeks dom top to make me his bitch"
06:14.32*** join/#asterisk asciii (n=WinNT@211.25.195.202)
06:14.38*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-97b5e7cd676f852d)
06:15.35jayteewell, I got the phone pad to show up and a call went through but it didn't seem to hear me when I spoke my city and state.
06:15.42jayteeI don't have a webcam on this pc
06:16.18drmessanoJaytee: I dont cam
06:16.28drmessanojaytee: Dream on, rollergirl
06:17.06jayteeok, enough of this bs for one nite. I'm going to bed
06:17.16drmessanoLOL
06:17.17drmessanoNite
06:17.21jayteesee ya round, nite
06:17.23drmessanoNow I want to listen to Skateaway
06:17.50drmessanoPossibly the best Dire Straits song ever
06:18.50*** join/#asterisk murdock_ut (n=chatzill@181.sub-97-153-209.myvzw.com)
06:21.03CherebrumLemensTS: yes... it's my site
06:21.26CherebrumLemensTS: I'm using this softphone: http://code.google.com/p/red5phone/
06:22.15Cherebrumjaytee: you have to allow it to receive audio from your microphone
06:22.28Cherebrumjaytee: it pops up and says, "allow webcam/microphone?"
06:22.42Cherebrumoh.. he left
06:25.13murdock_utI'm having a problem reparking a call that was parked using one step parking and picked up before it times out.  Any ideas?
06:25.27murdock_utUsing 1.6
06:26.02drmessanoHA 4chan hijacked the macworld feed on macrumors.com
06:26.15drmessano"STEVE JOBS HAS JUST DIED"
06:26.23drmessanoThats hilarious
06:26.40drmessanoI wonder how far Apples stock dropped over the 3 mins it took them to catch it
06:26.51drmessanoProbably like 75%
06:27.23*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
06:27.41drmessanoI feel for apple.. All those years of being an underdog, they start kicking ass in their consumer electronics offerings, and even now Steve Jobs coughs and the stock price plummets.. like 1987
06:28.09CherebrumBTW: You can also send tollfree calls via SIP to tollfreetollfree.com  G729 or G711 are accepted. ;)
06:29.17phpboyI wish to goodness I can get this right
06:29.18*** join/#asterisk joobie (n=joobie@joobie.org)
06:29.36*** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net)
06:29.44phpboyI need to speak to tzafrir, I think his script 'dahdi_genconf' doesn't work 100% correctly
06:30.36joobiephpboy, what's wrong with it?
06:32.04CherebrumI'm going to bed
06:32.05Cherebrumnite
06:38.24phpboyjoobie: It doesn't dedect my card
06:38.35phpboyhowever, if I configure it manually it works just fine
06:38.43phpboyin fact, it's still running just fine
06:43.34dandateis there a way to dig up irc chat history? someone here had told me of a few good providers that would charge 1 cent per minute outbound for use with my asterisk call center
06:47.50dandateand shit, how do i use voip with a hard phone?
06:50.29murdock_utdandate: Use an adapter.
06:55.51*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
06:55.54dandatewill my linksys pap2 work?
06:56.23murdock_utmurdock_ut: Yes, but I've seen them do weird stuff before...
06:56.48murdock_utMainly hearing dtmf tones when someone is talking.  Little weird.
06:56.58*** join/#asterisk SlicerDicer (n=SlicerDi@24-119-155-26.cpe.cableone.net)
06:56.59drmessanoPAP2s?
06:57.49murdock_utI think it was the pap2-na not pap2-t  or vice versa
06:57.57dandateyes it is the pap2-na
06:58.04dandatewill that work with a hard fone?
06:58.09drmessanoNever heard of that problem.. and I have used dozens of them
06:58.41drmessanoIf youre hearing DTMF, then something is mixing with it.. and not the adapter
06:59.04drmessanoor your 3rd neck is sitting on the keypad of your $6 walmart phone
07:00.11phpboythis is absolutely lovely :/
07:00.24phpboyHylafax doesn't wanna work anymore :T
07:14.51dandatehow do i tell if a SIP provider is used for inbound or outbound?
07:15.55drmessanoWhat do you mean "Tell"?
07:23.59dandatewell i am trying to configure an asterisk machine for the first time =)
07:24.14dandatei am running an inbound call center www.newlineequity.com
07:25.08drmessanook..
07:25.08dandateso i need to beable to port my sales phone numbers to the machine, then have the machine either dial out the #s to my work at home employees hard phones, or connect directly to their softphone
07:25.31drmessanoum ok
07:26.03dandateso i would need a SIP provider to ...
07:26.06dandateomg i'm confused
07:26.40dandatei know i have to set up a trunk outbound, can i use my comcast digital voice as my inbound?
07:27.15drmessanoSure
07:27.28drmessanoYou can pretty much do whatever you want
07:27.35drmessanoYou need to figure out what you want to do first
07:27.52drmessanoHow many Comcast lines do you have?
07:28.08dandatewell i have 2 comcast lines that i advertise for
07:28.31dandatei need them both to come into the phone system, listen to the music, connect to the next available rep
07:29.09drmessanoOk, so you get a TDM card with 2 FXO ports, connect to the Comcast gateway
07:29.29drmessanoWhere does the SIP provider come in?
07:29.36dandatewoah i have to google tdm and fxo
07:30.01drmessano~book
07:30.02jbotbook is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
07:30.05drmessanoStart reading
07:30.14drmessanoIf youre this lost, you need to do some reading first
07:30.28dandatewell i talked to an expert briefly on the subject, this is what he told me
07:30.32dandate[00:06] <C4thY> porting your numbers to an iax or sip provider ont he internet
07:30.32dandate[00:06] <C4thY> and then making it all true voip
07:30.32dandate[00:07] <C4thY> you dont need ata devices at all
07:30.32dandate[00:07] <C4thY> like bandwidth.com or teliax.com
07:31.43drmessanoum ok
07:32.38drmessanoI would suggest reading the book
07:32.42drmessanoMake your own decisions
07:33.39dandateok i have an FXO port, thats the linksys phone adapter right?
07:34.14dandateit says on it, "linksys phone adapter with 2 ports for VOIP"
07:34.35dandatethe vender says that it was unlocked also
07:34.47drmessanoNo, the Linksys is an FXS
07:34.49drmessanoNot an FXO
07:35.11drmessanoIt can connect to a provider and give you phone lines, like your Comcast box does
07:35.21drmessanoBut does not connect to existing service
07:35.52dandatedo you mean the SIP providers like broadband.com?
07:35.58drmessanoyeah
07:36.50dandatewell i wanted to keep it as basic as possible, so because i have the comcast box i can skip inbound trunks right?
07:37.06dandateand just sign up for an outbound trunk like voipjet?
07:37.25drmessanoAre you gonna use the Comcast for outbound and inbound or not?
07:37.49dandatehmm i'd like it to go both ways
07:38.05drmessanoIF you keep the Comcast box you still need to connect those ports to the PBX.. which means a card with FXO ports
07:38.26dandateok I see
07:38.32dandatecan u recommend a product?
07:38.47drmessanoDigium, Rhino, or Sangoma
07:39.10drmessanoI dont have a specific recommendation
07:40.24dandateok i see that would cost $83, I have unlimmited call forwarding on my comcast lines, so what if i forwarded them to an inbound sip provider instead?
07:40.52drmessanoThat would be stupid.. if youre gonna do that, then port the numbers over and ditch the COmcast
07:41.13dandatewellll it came in a package with the internet lol
07:41.31drmessanoIts still costing you more
07:41.47dandatealright so when u say port the numbers over that means re-lease those numbers from a diff provider?
07:41.54drmessanoyes
07:42.10dandatealright and that provider would be something like bandwidth.com or teliax?
07:42.14drmessanoyes
07:42.22dandateany recommendations on those 2?
07:42.23dandateheh
07:42.32drmessanoI use neither, so no
07:42.49dandatewell i guess i should just jump in and try something
07:43.32*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
07:44.30dandatedoes anyone know anything about broadvoice?
07:44.39The_Lightsidehi all, has anyone had issues with asteriskrealtime on queues with 300+agents
07:45.48The_Lightsidein particular, the MOH stopping playing
07:46.13dandateomg i hit the jackbot! i only advertise in the state of california and broadvoice offers unlimmited inbound and outbound in state for 9.95 per month
07:46.38mchoudandate: how much total minutes of (comcast) phone you use a month?
07:47.09dandateehh its hard to tell but i have 3-5 full time employees handling about 300 calls per day
07:47.21mchoudandate: whoa
07:47.26mchoulol
07:47.41dandatebut the comcast is unlimited
07:47.42drmessanoOn TWO lines?
07:47.55mchoudandate: you a telemarketer?
07:48.01dandateyeah 2 lines, i have to rotate employees and its a bitch, thats why i got this vpbx so i can have as many reps on as needed
07:48.08dandateyes we direct market www.newlineequity.com
07:48.15mchouoh lord
07:48.38mchouastdb blacklist
07:48.41dandatepeople sign up because we show them how to take over a pre-foreclosure loan with no down payment or credit check
07:48.48dandatewhats that mena?
07:48.52mchoulol
07:49.16dandateits possible, but u gotta be a go get em
07:49.29dandateour clients literally have to go knock on the door and buy the house for zero down payment
07:49.44mchoudandate: sont take it personally, I hate telemarketers
07:49.53dandateno we are inbound
07:49.56dandatepeople call us
07:50.14mchoudandate: anything is possible, but the real proof is how LIKELY
07:50.32mchouI may win the jackpot tomorrow....
07:50.33dandateit all depends on the person i guess, we've seen a person get into a home within 3 days
07:50.44mchoubut I'm not counting on it.
07:50.54dandateone out of 5 homes are facing foreclosure so if u went door to door in ur neighbor hood u might find a motivated seller
07:51.08dandatesomeone facing divorce could have too much house, can't sell cuz noone has a loan
07:51.10mchouindeedy
07:51.36dandatebut most of our clients just get signed on and forget about it then get hit up for monthly billing
07:51.44*** part/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1)
07:51.45dandatenow thats telemarketing!
07:52.03drmessanowaits for someone to google the channel logs
07:52.18mchoudrmessano: heh
07:52.35mchoureal estate scams revealed (tm)
07:52.59mchouall those late nite infomercials
07:53.12drmessanohttp://www.myrealwebsite.com <--- BIG SCAM, SHHH DONT TELL THE INTARWEBS
07:53.14dandateactually the process is not a scam, my company provides everything they need, but is often too bogged down to help anyone further after registration
07:53.25drmessanokeep digging
07:53.49dandatebecause when you take over payments on a loan you prevent the original person from having a foreclosure on their record
07:54.22mchouyes.  hit that angle in my investments b4
07:54.40mchoubut there's also opportunity cost
07:54.43dandateand u can contract to provide future payment if necessary, which is some incentive to the owner than losing all equity
07:54.58dandateright u would have to clear up the default to stop the foreclosure
07:55.02dandatebut we tell people its the same as renting
07:55.04mchouall the time knocking on doors cost real $$
07:55.23*** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1)
07:55.32dandateit could, but so could getting outbid at auctions
07:55.44dandatei mean noone said it was easy to buy a house zero down no credit check!
07:55.51dandatewe just said the process was simple =P
07:55.57mchoulol
07:56.00dandateok we did say it was easy
07:56.02dandatethats telemarketing
07:56.22drmessanoYoure a regular Billy Mays
07:56.29mchouhaha
07:56.31dandateshit i told someone its easier than renting and they bought it
07:56.38drmessanoITS EASY!!!!!
07:56.44mchoucount on drmessano with the wonderful retort
07:57.30drmessanoJust break off a small piece of foreclosure putty and squeeze to activate
07:57.39drmessanoThen the magic begins!
07:58.56dandatethats how it has to be, because people facing foreclosure have disconnected phone lines
07:59.08dandateor if they are connected, they don't answer it cuz its prolly the bill collectors
07:59.41dandatethen even worse, if the house is already vacant...u gotta send a letter to the forwarding address, also try to locate their family memmbers
08:00.13dandatebut if u pull it off u could save up to 50% off a home
08:00.23Coolthreadsanyone got any good tutorial on python for asterisk
08:00.56dandatei have a question, i need to have 5 people using my phone system, does that mean i need to purchase 5 seperate sip trunks / phone numbers from broadvoice.com?
08:03.40rob0ask them?
08:04.14rob0they don't care how many users you have, but they probably have a limit on concurrent calls
08:05.16*** join/#asterisk stix_ (n=stix@exchange2003.corporate.billetkontoret.dk)
08:06.35*** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net)
08:15.15dandatei am trying to sign up for broadvoice but it wants the mac address of my pap2-na , how do i get this/
08:18.32*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
08:19.46dandatedo i just provide the mac address of my router?
08:20.58*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
08:21.54mchoudandate: best to ask them which one they want
08:21.58dandatehaha my dumbass it was on the back of the model
08:22.07mchoumost likely they want mac of your pap2
08:22.36mchoubroadvoice is lame to even ask for that
08:22.41mchouavoid
08:23.12mchouthey are going to push some stupid ass configs your way
08:23.33mchouor do some other crap that wont be in your best interests
08:27.27dandateok
08:27.53dandatewell the tutorial i downloaded said to select "generic SIP" in which case they don't ask u for anything at all
08:28.30dandatebut the pricing is really turning me on, i can get unlimmited inbound and outbound in state for just $9,95 a month
08:28.57*** join/#asterisk botox93 (n=botox93@213.221.82.242)
08:30.30dandateis there a better deal than that?
08:30.43rob0Yeah, seems odd that they would need the MAC, maybe they just use it as a customer identifier.
08:30.59dandatewell it said on the website they would create settings for me
08:31.11dandateand in the option for generic it said i would have to customize all the settings
08:31.27drmessanoLOL
08:31.39drmessanoThey're gonna lock your device
08:31.47drmessanoGood luck with that
08:31.55dandateoh shit
08:32.08dandatei gave them my mac address already, but i didn't fill out the credit card ap
08:32.10drmessanoI thought you wanted to use ASTERISK
08:32.21dandatei do i was just confused cuz i have this linksys pap2
08:32.34drmessano....
08:32.49drmessanoIf they asked for your CAR KEYS would you get confused and MAIL THEM?
08:33.05dandatehaha no but u think they might have locked my stuff ??
08:33.19drmessanoI dont see how, unless they have pixie dust
08:33.55dandatek
08:34.00dandatehela scarin me man
08:34.02rob0How can they possibly lock an unlocked device?
08:34.10drmessanoEasily
08:34.23*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
08:34.26drmessanoPop in the provisioning info and turn provisioning on
08:34.28drmessanoDone
08:36.36drmessano<Protect_IVR_FactoryReset ua="na">Yes</Protect_IVR_FactoryReset>  <-- Thats how I roll
08:37.48drmessanoUnlocked is the dumbest fucking term ever, IMO
08:38.10drmessanoTheres several different things to consider with it being "locked" or "unlocked"
08:38.49drmessanoFirst is whether or not there custom code in the firmware that ties it to a provider
08:39.17drmessanoIn the case of the Vonage PAP2s, you dont ever get rid of being a factory reset away from being a vonage box again
08:39.46drmessanoThen theres the PAP2s that were never "locked" and not sold as Vonage devices
08:40.08drmessanoNot "Unlocked".. but NOT TIED TO ANYONE TO BEGIN WITH
08:40.47drmessanoEither one of those scenarios, an "unlocked" or unbranded PAP2 can be locked down with an XML config
08:41.25drmessanoand once I block IVR factory reset and send it an encrypted XML, good luck using it again without me pushing a new config
08:43.24drmessanoActually, the encrypted XML is half true.. Once I put it on a firmware revision (3.1.7 or higher) that can be told to only look for an encrypted XML file, you cant spoof a config to it
08:43.43drmessanoSo yeah
08:44.04drmessanoAbout as useless as that friggin "Brick" term that I can't stand
08:44.34mchouwhat do you mean?
08:44.34drmessanoMakes me want to throw bricks at the next person that "bricks" something just so they can feel what it's really like to be "bricked"
08:44.52mchou'my pap2 is now bricked'
08:45.10drmessanoWell, in my little world, "Brick" means TOAST, DEAD, NON USABLE, DOORSTOP, THROW AWAY
08:45.16mchoui.e. wont boot
08:46.22mchoumaybe I can 'unbrick' it using jtag or whatever
08:46.35drmessanoBut in circles where a product, any product, can have the firmware hacked or changed, and theres various states of "not being 100% usable right now", I hear constantly how someone "bricked" this or "bricked" that, but got it back working
08:46.42drmessanoReally?  How the fuck do you do that?
08:47.06phpboyCan I setup two IAX2 links to the same host, one with gsm only support and the other with slinear only support?
08:47.14mchoubasically reinstall the boot loader, iirc
08:47.22drmessano"I loaded the new firmware and got stuck in diagnostic mode" or "firmware reload mode" is not as cool as "ZOMG BRICKED MAH ROUTER LULZ"
08:47.24drmessanoBut not true
08:48.13drmessanoIf you can make the device usable with a software reload of ANY KIND, you have NOT BRICKED IT
08:48.31drmessanoIf you throw it in the trash or use it as a centerpiece at the dinner table, its bricked
08:48.32mchoudrmessano: yes, that is indeed true
08:48.53drmessanoThat crap pisses me off
08:49.06drmessanoLame ass Digg terminology
08:49.29drmessanoI left my lights on last night.. went out to start my car, and found out I bricked it, LULZOMG
08:49.37mchouhaha
08:50.48drmessano"I tried to get my next door neighbor drunk so I could poke her while her old man was out of town, but I gave her too much vodka and she passed out on my back porch."  "Ha, you bricked her"
08:50.52drmessanoYep
08:50.54drmessanoGuess so
08:51.38mchouthat's what I call opportunity cost :)
08:51.49drmessanoDamn right
08:52.07drmessanoPiss-poor ROI
08:52.15mchouhehe
08:52.48drmessanoBuy a date an $80 meal and she gives you a peck on the cheek.. ROI FAIL
08:53.15mchouthat's prisoner's dilemma
08:53.34mchoudo you byuy her another $80 meal?
08:53.38mchoubuy*
08:53.56mchouand get 3x the return? :)
08:54.35mchouor get zip?
08:54.42drmessanoWell, you can recalculate the ROI... If you buy her another $80 meal, and she was really saving up to give you $300 worth of Nookie if you made the investment in her, then you've got provable returns
08:55.26mchouyeah, but you dont know that in advance
08:55.52drmessanoTrue.. Which I guess is technically required for a calculable ROI
08:56.02*** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il)
08:56.06drmessanoTheres no expectation of a return
08:56.13drmessanoSo really, its more like a lottery
08:56.17drmessanolol
08:56.21phpboyI'm having a serious problem, I'm trying to send phone calls over one IAX2 link to 192.168.0.140 (gsm) and faxes over a second IAX2 link ysing (slinear) and it doesn't seem to wanna work :(
08:56.38drmessanosln?
08:56.44phpboyyes
08:56.48drmessanoFascinating
08:57.00drmessanoWhy are you doing that?
08:57.21phpboyI need to relay my E1 via another box for incomming calls (LONG story)
08:57.26phpboyit's a temp solutions
08:57.30phpboy*solution
08:59.09drmessanoI didnt realize you could even use SLN over a peer
09:02.08phpboyWell, I can do it if I only run sln to the remote host
09:02.16phpboybut I can't seem to get sln and gsm working ;/
09:03.14dandatedoes 5 people holding for a rep count as 5 concurrent calls?
09:11.21phpboyit seems I've learnt something new
09:11.45phpboysetup two diff IAX2 links from the src houst to ONE IAX2 link on the remote host
09:11.54phpboyto force 2 diff codecs
09:16.30*** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net)
09:19.46phpboyBut the question is, does it actually work like this? ...
09:20.36phpboyFUCK!
09:20.40phpboyI was wrong :T
09:22.04phpboyoh wait
09:22.33*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
09:24.43*** join/#asterisk mikkel (n=mikkel@130.226.37.71)
09:28.10*** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za)
09:28.39dandatethe SIP trunk providers i'm looking at do not mention concurrent call limits. Does this mean that there are any?
09:30.35*** join/#asterisk qdk_ (n=qdk@85.235.253.141)
09:31.01*** join/#asterisk Slashman (n=Slash@94.103.140.2)
09:32.49*** join/#asterisk Assid (n=assid@unaffiliated/assid)
09:33.08Assidtakes a number and waits for help on trying to get his cisco 7960 working
09:33.28Assidusiong sccp, can only get 1 way audio, other sip devices same location work fine
09:34.19Assidalso why it doesnt retain the information uploaded to it via tftp is something i dont get
09:41.29*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
09:48.30*** join/#asterisk jas_williams (n=jason@host86-147-181-64.range86-147.btcentralplus.com)
10:04.07*** join/#asterisk Dovid[Laptop] (n=annon@tony09-118-62.inter.net.il)
10:07.08*** join/#asterisk hi365_m (n=hi365@bzq-79-176-35-113.red.bezeqint.net)
10:12.25*** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman)
10:20.34*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903)
10:25.34wonderworldwe have a PRI and whenever something goes wrong at the other side (like "number has changed", "number out of service", etc.) asterisk returns the DIALSTATUS and hangs up. is there a way to make asterisk actually play those original messages from the telco side? thanks
10:26.04BBHosswonderworld: to the user?
10:26.08wonderworldyes
10:26.49BBHossnot sure, but it gets that from the signalling of the pri, no audio is passed i don't think, when its just signalling
10:28.13wonderworldi already tried to set prisignaling to inband in zapata.conf, but i think that would affect incoming calls only, right?
10:28.52BBHossi don't have much experience with pri in asterisk, just know a little about the underlying tech
10:29.31BBHossyour best best is to detect the fail and play your own message probably
10:30.50BBHosswonderworld: what kind of card is it?
10:30.52wonderworldi am already doing this, but there aren't enough different DIALSTATUSes to really tell the user what was going on
10:30.58wonderworlddigium te201p
10:31.27wonderworldfor example the CHANUNAVAIL status can mean 3 different things
10:31.33BBHosswell digium should be able to help you, i would give them a call in a few hours, 4:30 here now
10:31.49BBHossare there no channel vars for the pri info?
10:33.03*** join/#asterisk andresmujica (n=andresmu@190.27.1.233)
10:33.15wonderworldi am not sure, i used DIALSTATUS till now. i'll have to read up. but there *should* be a way to pass the audio thru? when i call a "failed" number from my cell phone, i can hear the message..
10:33.44*** join/#asterisk orTix (i=blaat@j167028.upc-j.chello.nl)
10:33.49orTixhello,
10:33.52BBHossthats probably the switch sending it to you via audio, there should be a way to do it, not sure
10:35.00orTixi got an astrisk server here with SIP on Cisco ip phone 7940 serie but i almost solve all the problems expect this one "W362 No valid names Provisioned" does anyone know howto solve this? I already googled it..
10:35.03BBHosswonderworld: priindication = passthrough maybe?
10:35.11BBHossin zapata.conf
10:35.17BBHosssays you need bristuff though
10:35.50BBHossorTix: cisco SIP firmware is trash, a possible solution is to buy a better phone for SIP
10:36.03orTixrofl
10:36.09wonderworldBBHoss: thanks. gonna read that up
10:36.11BBHossseriosuly
10:36.25BBHosswonderworld: says it doesnt work though so who knows
10:37.25BBHosspriindication = outofband or priindication = inband
10:37.31BBHossinband is apparently default
10:39.01*** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab)
10:39.23BBHossthats in chan_dahdi.conf wonderworld
10:40.03ScriptFanixHi
10:40.39BBHosshello ipv6'er
10:42.06ScriptFanixMy SIP Phone (Siemens C470 IP - behind NAT) sometimes have troubles passing a call through my asterisk server (not on the same network, not natted)
10:42.49orTixBBHoss maby its shit, but i need it to get it working..
10:43.06BBHossorTix: yeah i understand, no idea here though
10:43.11orTixhmm ok
10:43.21orTixtnx iig :)
10:43.53*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
10:44.06ScriptFanixI get this multiple times, then the call finally gets through: http://paste.quarantedeux.net/190
10:48.02ScriptFanixnot a blocking problem, but it's annoying ;)
10:48.34wonderworldhehe BAD! BAD! BAD!
10:48.37wonderworldfunny message
10:48.56ScriptFanix:)
10:49.38ScriptFanixpictures asterisk banging his head on against the wall
10:49.53orTix:o
10:50.03orTixmaby ScriptFanix can react on my question ;)
10:50.22verywisemanwhat is prefer ,fxo fxs ata or fxo fxs pci card?
10:50.25BBHossScriptFanix: probably the phone has a sip quirk
10:50.51BBHossverywiseman: for small installs. atas are fine
10:51.42verywisemancan you suggest for my some brand which have good PCI card?
10:51.58BBHossdigium, openzap, sangoma, etc
10:54.40*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
10:55.55ScriptFanixorTix: what question ?
10:56.18*** join/#asterisk C4colo (n=DJpyro@66.185.107.193)
10:59.38*** join/#asterisk jeffgus (n=jeffgus@static-173-51-181-4.lsanca.ftas.verizon.net)
11:01.31orTix=:11:34:53:=- <orTix> i got an astrisk server here with SIP on Cisco ip phone 7940 serie but i almost solve all the problems expect this one "W362 No valid names Provisioned" does anyone know howto solve this? I already googled it..
11:01.32orTixthat
11:02.19tzafrir_laptoporkid, and you see that error where exactly?
11:05.17orTixw362 No valid line names provisioned
11:05.34tzafrir_laptopWhere is it emmited?
11:05.44tzafrir_laptopcoming from where?
11:05.50orTixOn this cisco phone @ status
11:05.57orTix"status messages"
11:06.14orTixAnd big in the screen it says "phone unprovisioned (sip)"
11:07.59tzafrir_laptoporkid, oops, I meant orTix above. Sorry
11:08.47orTixI know :) np
11:09.07tzafrir_laptoporTix, while I have 0 experince with provisioning those phones, I suppose a pastebin of the provisioned data would help
11:09.34tzafrir_laptop(with passwords and other sensetive information masked out or whatever)
11:10.39*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
11:11.03orTixShort way, you need the data of my cfg, asterisk -vvvvvr and telnet debug ?
11:11.11orTixand the tftp data?
11:11.53*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
11:13.14*** join/#asterisk Subdolus (n=subby@subby.afraid.org)
11:16.28*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
11:17.47*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
11:22.28*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
11:32.07*** join/#asterisk bkruse (n=bkruse@nat/digium/x-093087e253071497) [NETSPLIT VICTIM]
11:32.07*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) [NETSPLIT VICTIM]
11:32.07*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) [NETSPLIT VICTIM]
11:32.07*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) [NETSPLIT VICTIM]
11:32.07*** join/#asterisk Subdolus (n=subby@subby.afraid.org) [NETSPLIT VICTIM]
11:32.07*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk C4colo (n=DJpyro@66.185.107.193) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk orTix (i=blaat@j167028.upc-j.chello.nl) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk hi365_m (n=hi365@bzq-79-176-35-113.red.bezeqint.net) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk Dovid[Laptop] (n=annon@tony09-118-62.inter.net.il) [NETSPLIT VICTIM]
11:32.08*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk Assid (n=assid@unaffiliated/assid) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk Slashman (n=Slash@94.103.140.2) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk qdk_ (n=qdk@85.235.253.141) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk botox93 (n=botox93@213.221.82.242) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk SlicerDicer (n=SlicerDi@24-119-155-26.cpe.cableone.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk Cherebrum (i=jgarland@209.9.237.93) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk orkid (n=orkid@unaffiliated/orkid) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk sucituanbo (n=full@c-24-21-121-148.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk jks (i=jks@193.189.93.254) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:5d00:b8cf:9894:f1a5) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk brian (n=brian@unaffiliated/brian) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk sack (n=sack@222.Red-83-49-103.dynamicIP.rima-tde.net) [NETSPLIT VICTIM]
11:32.09*** join/#asterisk implicit (n=bayan@unaffiliated/implicit) [NETSPLIT VICTIM]
11:32.10*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM]
11:32.10*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM]
11:32.10*** join/#asterisk SkywaIker (n=pirch@58.147.17.166) [NETSPLIT VICTIM]
11:32.10*** mode/#asterisk [+o bkruse] by irc.freenode.net
11:32.10*** join/#asterisk sevard (n=sev@multimedia.dvc.edu) [NETSPLIT VICTIM]
11:32.10*** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk ScarEye (n=scareye@12.27.87.66) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk joesuffceren2 (n=chatzill@srv.fgp.com) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk LemensTS (n=customgt@70.238.154.243) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk beek (n=klinebl@65.211.106.242) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk fogo (n=Paul@69.169.132.35) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk ELBunce (n=erik@kde/developer/bunce) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk NoxIn- (n=noxin@zorlit.org) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk DigitalIrony (n=eric@nat/digium/x-75ef67f885360914) [NETSPLIT VICTIM]
11:32.11*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552227.dsl.bell.ca) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk WHYS (i=lpfm@137.28.94.209) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk rickross (n=rickross@supporter/active/rickross) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk Sargun_screen (n=sargun@64-142-51-251.dsl.static.sonic.net) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk kinnaz (n=nnscript@no.life.ee) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk The_Lightside (n=Lightsid@dsl-241-86-116.telkomadsl.co.za) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk ltd (n=z@pat.transact.net.au) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk sergee (n=serg@voip1.west-call.com) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk rdlang (n=dlangr@84.244.141.35) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-104-80-249.pskn.east.verizon.net) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk phpboy (n=shane@196.36.108.18) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) [NETSPLIT VICTIM]
11:32.12*** mode/#asterisk [+ov Deeewayne phpboy] by irc.freenode.net
11:32.12*** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk maagic (i=maagic@fsck.fi) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk brodiem (n=brodiem@e2.72.1243.static.theplanet.com) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk yang (i=yang@CAcert/Assurer/pdpc.supporter.base.yang) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) [NETSPLIT VICTIM]
11:32.12*** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk tuxx- (n=tuxx@nightshade.nl) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk Skavin (n=kevins@202.37.22.66) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk silentaudience (i=psybnc@194.105.146.253) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk lilalinux (i=e-trolle@fellatio.deswahnsinns.de) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk echinos (n=echinos@H211.C136.B196.A67.tor.heavycomputing.ca) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) [NETSPLIT VICTIM]
11:32.13*** join/#asterisk hawk (n=hawk@l.qw.se) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk r0d3nt (n=astrutt@pinky.ratman.org) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk dongs (n=lol@l212168.ppp.asahi-net.or.jp) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk emist_ (n=emist@unaffiliated/emist) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk nacer (n=God@l.alcolo.a.mpl.pastIX.net) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk jermey_g (n=me@static-213-115-44-90.sme.bredbandsbolaget.se) [NETSPLIT VICTIM]
11:32.14*** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk lotho (n=lotho@valhalla.via-publica.de) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk snaud (i=lp@hades.ds1.agh.edu.pl) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk chazz (n=chazz@173-24-236-97.client.mchsi.com) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk NovceGuru (n=novcegur@server1.jsreedinc.com) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk matt_ (n=matt@mattspc.ipv6.mattstone.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk AMUG (n=junky@modemcable156.137-20-96.mc.videotron.ca) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk x86 (n=x86@p3m/member/x86) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk bmg505 (n=leon@196-209-77-132-tbnb-esr-2.dynamic.isadsl.co.za) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk troubled (n=troubled@unaffiliated/troubled) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk pdfhacker (n=dd@38.104.98.118) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk seb- (n=seb@li30-51.members.linode.com) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk khronos (n=khronos@c-76-110-120-247.hsd1.fl.comcast.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk dthomas (n=darkness@linode.caliginous.net) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk freddyk (n=freddy@host14-144-dynamic.53-79-r.retail.telecomitalia.it) [NETSPLIT VICTIM]
11:32.57*** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM]
11:32.58*** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) [NETSPLIT VICTIM]
11:32.58*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) [NETSPLIT VICTIM]
11:32.58*** join/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) [NETSPLIT VICTIM]
11:32.58*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk [netman] (n=netman@188.Red-88-23-81.staticIP.rima-tde.net) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk slima (i=slima@unaffiliated/slima) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk eppigy (n=Dave@plasticlobster.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk micols (n=micols@scharff.fys.ku.dk) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk oh207 (n=oh207@nylug/member/oh207) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk pa (n=pa@unaffiliated/pa) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk troy- (n=troy@worldnet.tauri.ca) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk exsync (n=mjohnson@pdpc/supporter/active/exsync) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk xuser (n=xuser@unaffiliated/xuser) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk Corydon76-dig (i=seven@pdpc/supporter/bronze/Corydon76-home) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk voxio (i=mark@storm.reedox.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk edoceo (n=edoceo@c-98-247-254-241.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk Takapa (i=vegard@svanberg.no) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk smk (n=scott@cobra.httpd.org) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk carrar (i=tim@osburn.com) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk mnicholson (n=mnichols@nat/digium/x-81b50d14ad7dd2b8) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) [NETSPLIT VICTIM]
11:32.59*** join/#asterisk thehar (i=thehar@thehar.xmission.com) [NETSPLIT VICTIM]
11:33.00*** join/#asterisk jer (n=jtregunn@unaffiliated/jer) [NETSPLIT VICTIM]
11:33.00*** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) [NETSPLIT VICTIM]
11:33.00*** join/#asterisk JT (n=j@unaffiliated/jt) [NETSPLIT VICTIM]
11:33.00*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk ix33 (n=ix@7b.85.b6.static.xlhost.com) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk c4t3l (i=rcallico@equinox.alluvium.com) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk jeff (i=jeff@unaffiliated/jeff) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk drfreeze (n=Jim@207.191.114.82) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk jackson__ (n=jackson@68-115-80-74.dhcp.roch.mn.charter.com) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk twisted (n=twisted@router.asteriasgi.com) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk fnordus (n=dnall@70.71.225.48) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk miloux (n=miloux@213.88.194.123) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk kaii (n=kai@ciphron.de) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk andrewy (n=irssi@cl-53.lax-01.us.sixxs.net) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk scardinal (n=supreme@90.184.100.170) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk srd (i=srd@wombat.frictious.net) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
11:33.01*** mode/#asterisk [+ooo Corydon76-dig twisted angler] by irc.freenode.net
11:33.01*** join/#asterisk justdave (n=dave@unaffiliated/justdave) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk SkramX (i=mark@phalse.2600.COM) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) [NETSPLIT VICTIM]
11:33.01*** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) [NETSPLIT VICTIM]
11:33.02*** join/#asterisk madsara (n=madsara@rtr0.circleeight.net) [NETSPLIT VICTIM]
11:33.02*** join/#asterisk f0urtyfive (n=noone@75.150.130.121) [NETSPLIT VICTIM]
11:33.02*** join/#asterisk C4thY (i=wtf@209.136.127.141) [NETSPLIT VICTIM]
11:33.02*** join/#asterisk mcab (n=mb@mostly-harmless.ca) [NETSPLIT VICTIM]
11:33.02*** join/#asterisk bagpuss_thecat (n=bagpuss_@lodge.glasgownet.com) [NETSPLIT VICTIM]
11:33.02*** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info)
11:33.03*** join/#asterisk jets (n=brian@jets.ownsu.com) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk unpaidbill (i=bill@420nugs.info) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk bpgoldsb (n=bpgoldsb@spatialdata2-gru-gw.customer.gru.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk endemic (n=endemic@orion.onvox.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk tris (i=tristan@camel.ethereal.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk loather (n=loather@damnit.us) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk Talkradio (i=talkradi@linuxgeneration.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk andrewn (n=andrew@76-191-212-233.dsl.dynamic.sonic.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-869df3744e3be6f9) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk awk_r (n=rawk@nat/digium/x-46be58956a1eca8d) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk PanGoat (n=PanGoat@node2.sensoryresearch.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk citats (n=james@mrplow.gnuinternet.com) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk mmattice (i=mmattice@unaffiliated/mmattice) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk derrick_ (n=derrick@204.57.82.166) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk simond (n=simon@syria.uc.org) [NETSPLIT VICTIM]
11:33.03*** join/#asterisk hohum (n=dcorbe@206.71.169.115) [NETSPLIT VICTIM]
11:33.03*** mode/#asterisk [+oo file Qwell] by irc.freenode.net
11:54.52*** join/#asterisk telecos (n=sergio@34.167.219.87.dynamic.jazztel.es)
11:56.24*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
11:56.28*** join/#asterisk mgdm (n=michael@serenity.mgdm.net)
11:58.31*** part/#asterisk mgdm (n=michael@serenity.mgdm.net)
12:01.41*** join/#asterisk path_ (n=path@147-101-21-190.adsl.terra.cl)
12:02.06*** join/#asterisk ITGuru (n=Gabby@82.108.189.20)
12:03.08ITGuruI've got one wierd problem - I have three devices, and they connect to asterisk, but they do not actually fully connect - The ring, but when you dial out, you get instant engaged tone. This is not the same as other handsets configured on the same machine, to do the same thing
12:13.15AssidScriptFanix: you got it working ?
12:13.41Assidi got an issue where my cisco can call a sip phone but only able to send, cant receive audio
12:15.10BBHossITGuru: turn on debug and verbosity and see if you see anything interesting, else pastebin it
12:15.35ITGuruBBHoss, no probs - I'll jump on that right now
12:17.11*** join/#asterisk espent (n=espent@totem.fix.no)
12:17.47espenthi
12:18.00*** join/#asterisk anonymouz666 (n=anonymou@201.19.69.17)
12:18.26espentany agi experts here
12:18.29espent?
12:19.04AssidBBHoss: any idea on this? sccp  audio reaches sip.. but not the other way around.. cant even get the cisco phone to g
12:19.11Assidto hear voicemailmain
12:19.53BBHossAssid: the sccp driver is pretty shitty, there are patched ones floating around i think
12:20.16BBHosstry asking RypPn, i think he was talking about them the other day
12:20.22Assidi actually used some kind of patch
12:20.39Assidyeah he was around last night. but had toleave :(
12:20.58Assidi think im using : http://sourceforge.net/projects/chan-sccp-b/
12:21.52*** join/#asterisk freckle (n=chatzill@195.74.96.119)
12:22.03ScriptFanixAssid: it works, it just sometimes takes ~10seconds to get the call through...
12:22.08*** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif)
12:22.17ScriptFanixAssid: btw, i have no experience with Cisco phones
12:23.40DarKnesS_WolFif i have bindaddr=127.0.0.1 in sip.conf should i be able to register to any external sip proxy/server or not ?
12:25.03espentwhen i call Dial from my agi, how can I continue executing commands, i wont go further before the called up party hangs up
12:25.53ScriptFanix(the more i play with asterisk @home, the more i'm thinking about putting an axe through the 3Com NBX 100 here at work and setting up an asterisk box with appropriate cards)
12:26.20*** join/#asterisk carranca (n=carranca@pampero.itba.edu.ar)
12:26.29AssidScriptFanix: i spoke for more than 10 secs.. nothing
12:28.15carrancaHi, im trying to install a 3rd party app for video gateway purporses, (app_h324m) and it isnt working, ive debuged a little and now i want to see the information that is passing through the B-channel, using dahdi_monitor i can get a raw dump but isnt there a way to get a pcap dump to load it in wireshark like the old ZapDump?
12:28.25ScriptFanixAssid: I have no idea what your problem is, I'm quite new to telephony and thus, asterisk
12:29.13Assidhrmm k.. nvm then
12:29.16Assidwill ask arnd
12:41.56aiksa[LV]is there a way to change zaptel timing source after it has been initialized?
12:53.09*** join/#asterisk qdk (n=qdk@79.138.251.161.bredband.3.dk)
12:57.03freddyki have a question about voice and fax detection on a line
12:57.19freddyki have a fax connected to an ata on a sip peer
12:57.43freddyki would like to hangup a call if no fax is detected in 15 seconds
12:57.46freddykis this possible ?
12:59.50tzafrir_laptopaiksa[LV], you don't control it directly
13:00.28tzafrir_laptopbut generally whenever the zaptel timing master span gets in alarm, there is "election" of a new master span
13:00.52ITGuruBBHoss, verbosity is 3, and core debug is 4 - i'm using asterisk -r at the CLI - but I'm a bit unsure what to look for there doesn't seem to be anything strange
13:01.37BBHossturn sip debug on and try to make a call where it has problems, then pastebin it
13:03.02ITGuruBBHoss, darn, I'm remote - I'll ask someone there to do so
13:03.04*** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net)
13:03.16BBHossITGuru: you cant ssh in?
13:03.22*** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co)
13:03.30andresmujicaHi, good morning!
13:03.38ITGuruBBHoss - I am SSH'd in - you asked to make a call on the affected extension
13:03.43BBHossahh ok
13:03.44BBHossgotcha
13:03.54ITGuruBBHoss, all the other extensions are working fine, just these three.
13:04.03BBHossits when the endpoint attempts to make a call right
13:04.07BBHossbut they ring fine?
13:04.10andresmujicai wonder if anyone knows about a solution for PRI failover ?
13:04.24andresmujicai'm looking this http://www.beronet.com/content/view/83/91/lang,en/
13:04.41andresmujicabut it seems to need 220VAC and is not clear if it supports 60hz....
13:06.15ITGuruBBHoss, That is correct, they ring, but no call goes out
13:06.28BBHosswhat kind of phones are they
13:06.29ITGurucalls*
13:06.55ITGuruMitel 5340 x  2 - and Polycom Sound Station
13:07.01ITGuru@ BBHoss
13:07.23BBHosshmm, mitels aren't really know for standards compliance :)
13:07.41carrancahas anyone configured asterisk with the app h324m?
13:08.45*** join/#asterisk yam_ (i=yam@pdpc/supporter/active/yam)
13:08.50*** join/#asterisk shido6 (n=shido6@209.114.208.111)
13:08.54*** join/#asterisk ipguy2 (n=ipguy2@124-170-238-129.dyn.iinet.net.au)
13:09.02ipguy2hi all
13:09.26ipguy2i just setup my first * server and i'm really excited
13:09.32*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
13:10.59ipguy2currently only using it for one phone line at home with a sip phone
13:11.10*** join/#asterisk anonymouz666 (n=anonymou@201.19.69.17)
13:11.15ipguy2+ vioicemail
13:11.24ipguy2thinking of setting up gtalk
13:11.45orTixwhat for phone are you using?
13:12.10*** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum)
13:12.18ipguy2it's a "GigasetC470 IP"
13:12.33tzafrir_laptopaiksa[LV], hmm... but why do you need this?
13:12.36ipguy2nice phone, cheap, but a little slow
13:12.49orTixhmm oke
13:13.26ipguy2i'm wondering what other people are using * for, like gtalk etc... any other cool things ?
13:14.00ipguy2i'd like to read up about dialplans and extensions so i can create an answering service etc....
13:17.07freddykis there any way to hangup a call if it's not a fax even if it's outgoing or incoming
13:17.07*** join/#asterisk styelz (n=yoohoo@2001:5c0:1100:a00:0:0:0:1)
13:22.28*** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr)
13:25.40*** join/#asterisk telnettech (n=telnette@199-2-116-183.setardsl.aw)
13:26.10PanGoatfreddyk: you mean... just hang up a call?
13:26.21PanGoathangup()
13:26.36freddykpangoat
13:27.14freddykpangoat: i have a line that i need to dedicate to fax but often, people in the office, uses it for personal calls. I just need to hangup if it's a voice call and not a fax
13:27.45jksanybody got an idea how to enable a snom 370 for configuration? - I can access the webinterface, but I'm not allowed to enter anything in the text fields to change SIP info for example
13:31.08PanGoatfreddyk: Ahhh... I see
13:32.25PanGoatfreddyk: can't you just query the extension that the channel is currently on? if it's the extension used for fax, don't hangup().
13:32.50PanGoathttp://www.voip-info.org/wiki-Asterisk+fax <-Asterisk as a fax/voice switch
13:32.59PanGoattowards the middle
13:33.32*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
13:33.42PanGoatI'm guessing though, being a * newbie
13:35.21*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
13:36.25*** join/#asterisk kannan (n=kannan@121.246.242.95)
13:43.29eppigyhello
13:43.32eppigyi am dave
13:44.40*** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za)
13:44.44*** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net)
13:45.37ipguy2open the pod bay doors dave !
13:45.54eppigy:[
13:46.07ipguy2:-)
13:48.09freckleipguy2: I use * from lots of stuff, what do you want to know?
13:51.12*** join/#asterisk ta^3 (n=tacvbo@189.146.181.5)
13:53.33*** join/#asterisk anonymouz666 (n=anonymou@201.19.69.17)
13:57.21*** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co)
14:05.31x86I'm getting one-way audio on inbound calls only (from the PSTN)... outbound calls following the same path work fine
14:06.03x86I've got a 4-port FXO digium card connecting asterisk to 4 POTS lines
14:06.34x86any ideas what might cause that?
14:07.33*** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com)
14:07.36x86Linksys SPA941 SIP phones --> LAN --> Asterisk --> TDM400P --> PSTN
14:07.51*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
14:08.23Mimmushi, I'm having a problem with mail notification of voicemail messages
14:08.32MimmusIt's not the first server I build :-(
14:09.19MimmusI don't see any trace in /var/log/messages
14:11.20Kattyhummm.
14:11.40eppigyTRABAJO
14:11.50Kattymorning
14:12.12eppigygood morning
14:12.24eppigyit is a beautiful day
14:12.46Kattyis it? i've not pried my eyes open yet.
14:13.07Kattyi remember something about it being frosty
14:13.47*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
14:13.53eppigyi am in atl
14:13.55coppicewhat is the frost you speak of? do you mean there are places where the temperature drops to freezing?
14:13.56eppigyit is perfect
14:13.58*** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar)
14:14.06eppigyI will be in portland next week
14:14.12eppigyso it will be a little chilly then
14:14.14eppigy:D
14:15.31jayteeportland maine?
14:15.51*** join/#asterisk TSCDan (n=demord@cpe-69-207-149-216.rochester.res.rr.com)
14:16.40jayteeit's snowing here.
14:16.43Carlos_PHXFrost.  I see some in my freezer.
14:18.01[TK]D-Fender10" of snow coming down....
14:18.09*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net)
14:18.16*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:18.26beek[TK]D-Fender: My world is ice-covered.
14:18.30coppicesnow is for suckers who haven't figured out where the tropics are
14:19.42TSCDanWe are going to be switching over to a VoIP system shortly (internally) that connects to our T1's for POTS lines (True VoIP in the future).  We have a tech support dept that currently has several physical modems sitting in their cubes to dial out to customers. We want to have a modem pool for these users to connect to from their PC instead of physical devices, I was looking for some sort of virtual modem device for Windows machine
14:19.47*** join/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu)
14:21.33[TK]D-FenderTSCDan: Who needs a modem to place a call?
14:21.49eppigyjantman: portland oregon
14:22.05TSCDan[TK]D-Fender: We have support personnel that need to dial into customer's machines.
14:22.05eppigyjaytee: portland oregon
14:22.07jantmaneppigy: huh?
14:22.12eppigyjantman: leave me alone
14:22.31jantman"eppigy: jantman: portland oregon" ???
14:22.35eppigyD:
14:22.42eppigywhy u fakin logs?
14:22.55jantman*sigh*
14:23.05[TK]D-FenderTSCDan: Ok, this is rare stuff, best of luck with that and when you consider going "full VoIP" you can kiss that idea pretty much goodbye
14:23.19jayteePortland, Oregon is nowhere near as "chilly" as Portland, Maine. I've been to both places many times.
14:23.26jantmananyone in here know of a good VoIP forum - something that's active and friendly to non-Asterisk-specific questions?
14:24.15jayteeI much prefer Portland, Oregon but then if I had my preference I'd prefer Ashland on the Cali border in the Rogue Valley. Lived there 5 years and loved it.
14:24.26TSCDan[TK]D-Fender: We will most likely be keeping several POTS lines just for them in the future, but still using the VoIP gateway from their PC's.
14:24.43tzafrir_laptopcoppice, tropics is where the tropical storms come from ;-)
14:25.57coppicetzafrir_laptop: we strategically placed taiwan, and the philipinnes to take the worst of those
14:26.05andresmujicai wonder if anyone knows about a solution for PRI failover ?
14:26.14andresmujicai'm looking this http://www.beronet.com/content/view/83/91/lang,en/
14:26.25andresmujicaanyone has ever used it?
14:30.46LinuturkDigitalIrony: ping
14:30.50x86on inbound calls from TDM400P to Linksys SPA942 phones, getting one-way audio (I can hear caller, caller can not hear me), asterisk CLI shows "chan_sip.c: Can not find address for host '7001'"
14:31.16x86where 7001 would be my SIP peer
14:32.01[TK]D-Fenderx86: Where are you configs?  Where is the SIP debug?  Where's Waldo?  Where in the world is Carmen Sandiego? Who killed J.R.?
14:32.03*** join/#asterisk silentaudience (i=psybnc@194.105.146.253)
14:32.06andresmujicanat enabled on the linksys? maybe the external address field is filled?
14:35.57joesuffceren2>anyone know of a 64bit tsp (tapi) driver for asterisk?
14:39.17jantmandoes anyone in here know of a good VoIP forum - something that's active and friendly to non-Asterisk-specific questions?
14:39.52ITGuruI'm having a seriously crazy time with some MITEL handsets, they refuse to dial out, giving me an instant engaged tone, but they ring when in a ring group - any ideas?
14:40.13[TK]D-Fenderjantman: Here might be good, depending on the question
14:40.36[TK]D-FenderITGuru: Idea : Go look at your SIP DEBUG
14:40.39*** join/#asterisk Shizuo (i=pato@200-171-49-211.dsl.telesp.net.br)
14:40.45ShizuoWhat is the cheapest card for Asterisk?
14:41.01jantmanok... i've got some experience with Asterisk, but only with internal (to the organization) VoIP or analog PSTN, i'm having some issues finding a service provider
14:41.29angryuserShizuo: selfmade clone, a week of soldering 5$
14:41.37*** join/#asterisk awannabe (n=brad@ip98-167-184-247.ph.ph.cox.net)
14:41.49Shizuoangryuser: Clone of what?
14:41.51BBHossShizuo: an x100p clone
14:42.12jantmandeveloping a prototype app for asterisk... just allows employees to call in, go through an IVR and make some selections, and dumps their responses in a database along with CID information... needs to handle about 4 simultaneous calls, and want VoIP service since I don't have multiple analog lines.
14:42.14awannabehello all, how do you set outgoing callerID to be private/anonymous? I think you have to add a SIP header but I cannot find what to put.
14:42.44angryuserdoes x100p able to provide timing btw ?
14:42.52BBHossyeah
14:42.58ShizuoIsn't that extremely hard?
14:43.24angryuserShizuo: i was ironic, you can but a x100p clone
14:43.30angryuser*buy
14:43.36jantmanshould I be looking for IAX or SIP (will probably end up being IAX since I'll probably be on a dynamic IP)? Also, I can't find a definitive answer to the difference between trunk and termination service... what do I need if I want, say, 4 simultaneous calls to the same number (hunt group) and incoming only?
14:44.07[TK]D-Fenderjantman: ...
14:44.10[TK]D-Fender~itsplist-us
14:44.11jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
14:44.13[TK]D-Fender^^^^^^^^^
14:44.23ShizuoI can't seem to find prices, though
14:44.25[TK]D-Fenderjantman: SIP recommended
14:44.28ShizuoI'll keep searching...
14:44.30jantmanok
14:44.32jantmanthanks
14:44.52jantmanwhat about trunks/termination and hunt groups (or whatever they're called in VoIP lingo)?
14:45.05ShizuoIs it hard to code your own drivers for Asterisk? Does this software contains a good driver API? Or is it criptic like Linux?
14:45.08jantmanessentially I need one number that people call, which can handle four simultaneous calls
14:45.22jantmanin the old world, it would be four lines or a PRI with a hunt group
14:45.24[TK]D-FenderITGuru: And no SIP debug does not mean they haven't registered.  You can prove that separately.... and SHOULD
14:45.40ITGuruThanks for the response - I already did a SIP debug, and no joy, didn't spit out a thing, as if the handsets had not registered at all
14:46.01NovceGuruoh FFS time warner business phone that i'm being forced to use does not offer sip trunking, only POTS out of their shitty ass modem
14:46.14[TK]D-Fenderjantman: the idea of "hunt group" is an analog concept for otherwise unassociated lines.
14:46.33BBHossNovceGuru: just dump the firmware and get the sip credentials
14:46.36[TK]D-Fenderjantman: Anywhere else its just "I want 1 DID supporting X channels"
14:46.45angryuserShizuo: ask tzafrir
14:46.47jantmanahhh, thanks!
14:46.49NovceGuruBBHoss: haha that'd be nice
14:46.56BBHosssure its possible
14:46.56ITGuruI guess my next question, would be how to I prove that separately?
14:46.58[TK]D-FenderNovceGuru: You sound surprised
14:47.14[TK]D-FenderITGuru: "sip show peer 12345"
14:47.17jantmanso I guess that would be SIP termination, as opposed to trunk?
14:47.22NovceGuru[TK]D-Fender: well it's "new" service they're offering that comes in over their fiber connection
14:47.29NovceGuruI thought....maybe.....blah
14:47.32BBHosspeople have dumped the magic jack software and gotten the user/pass out of it, unlimited channels and minutes for 20 bucks/year
14:47.36[TK]D-Fenderjantman: Origination actually, but close enough
14:47.43jantmanok, thanks!
14:47.58[TK]D-Fender~itsp
14:47.58jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
14:48.16jantmanahhhh
14:48.27NovceGuruBBHoss: nice!
14:48.32[TK]D-Fenderjantman: What you are looking for is within the scope of "cheap"
14:48.36*** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org)
14:48.38ITGuru[TK]D-Fender, I have 7 registered subscriptions, and that is the 6 handsets, plus softphone for testing
14:48.49[TK]D-Fenderjantman: 4 chan for a single DID... easy to find.
14:49.00jantmanD-Fender: even better, since for now this is just a prototype job
14:49.06[TK]D-FenderITGuru: None of which answers the "what about this ONE peer?"
14:50.21ITGuru[TK]D-Fender, I would like to dump just the debug info for that particular peer, how would I go about that - i seem only to be able to dump for all peers, and this is a live system, with like 3 calls a minute
14:50.49[TK]D-FenderITGuru: "sip show peer 12345" <- PAY ATTENTION
14:51.47*** join/#asterisk chendy (n=chatzill@58.61.197.37)
14:51.55*** part/#asterisk TSCDan (n=demord@cpe-69-207-149-216.rochester.res.rr.com)
14:51.59ITGuru[TK]D-Fender, I feel dumb.... now I got the info I need
14:52.05ITGurugoes to shit in the shame corner
14:52.12ITGuruand by shit - I mean sit! lol
14:52.25*** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com)
14:53.23NovceGurublah this setup is going to fell ghetto
14:53.36*** join/#asterisk moy (n=moy@74.12.127.97)
14:54.47ITGurusip show peer 0301 - This is the peer having the problem - it's output is here - http://pastebin.com/da0cffd9 @ [TK]D-Fender BBHoss
14:55.38[TK]D-FenderITGuru: Reg. Contact : sip:0301@192.168.10.172:5060;transport=UDP
14:55.48*** part/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu)
14:56.09[TK]D-FenderITGuru: Its registered.  If you see no SIP debug on a call attempt then the phone doesn't LIKE what you're telling it to do.  Most likely a local dialplan error
14:58.58*** join/#asterisk jtodd (n=jtodd@nat/digium/x-3e5eaffbbeafa5b5)
15:07.00aiksa[LV]could it be that a massive change in clock +/- 2h might crash asterisk?
15:07.14aiksa[LV]system clock on linux of course
15:07.19ITGuru[TK]D-Fender, I'm guessing that a lot of work to do as well?
15:07.22aiksa[LV]?
15:07.54*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-5766518fd7928974)
15:07.54*** mode/#asterisk [+o putnopvut] by ChanServ
15:08.09tzafrir_laptopaiksa[LV], I'm not aware of such a bug
15:08.11[TK]D-FenderITGuru: ...huh?
15:08.44aiksa[LV]tzafrir_laptop: i just launched ntpdate from one of my timeservers which switched local clock back by two hours
15:09.19tzafrir_laptopaiksa[LV], you do have proper timezone settings?
15:09.39aiksa[LV]and in the very instant asterisk process disapeared
15:09.50*** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org)
15:09.59ITGuru[TK]D-Fender, Each extension should be identical in the dial plan - It's wierd that there is an issue. What if I put in different extension credentials? An extension that I know is working? Should this help me to narrow down where the problem is?
15:10.03aiksa[LV]tzafrir_laptop: - ill check that
15:10.08aiksa[LV]it still seams strange
15:10.24aiksa[LV]and might explain a few issues i have had in the past
15:10.38aiksa[LV]/var/log/asterisk/full gives no hints as to why the process died
15:10.47[TK]D-FenderITGuru: You're saying the PHONE can't dial.  I said LOCAL DIALPLAN.  Your PHONE doesn't like what you are dialing.
15:10.51[TK]D-FenderITGuru: not ASTERISK.
15:11.16[TK]D-FenderITGuru: And stop calling devices "extensions"
15:12.01aiksa[LV][TK]D-Fender: "Each extension should be identical in the dial plan " - make your head spin doesnt it?
15:12.19aiksa[LV]just a bunch of identical lines in extensions.conf
15:13.35ITGuruOkay, Thanks for the lesson, I just wanted to narrow down where the problem is - The phone rings, but that is it - when you pick it up, you hear nothing - if it was just to do with what your dialling on the handset, then it should be able to take incoming calls - it rings, but when you pick it up, nothing
15:14.05aiksa[LV]tzafrir_laptop: just checked - the timezone is set correctly, so it could be wrong settings of system clock. Ill change them in BIOS tonight
15:14.07*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
15:14.13*** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx)
15:14.25[TK]D-FenderITGuru: And now you're talking about audio issues, not a "can't place calls to *" issue.  Let me know when you've got your head screwed on straight.
15:14.51[TK]D-FenderITGuru: EVERYTHING appears to be bad and we're hearing it in bits and pieces and I really can't waste time on a runaround now.
15:15.27tzafrir_laptopaiksa[LV], hwclock --systohc
15:15.46aiksa[LV]sets the hwclock to current "date"?
15:15.59jtoddDon't know of anyone else has asked this recently, but does anyone know of a pre-built program (based on Asterisk) that will call a list of customers and do automated surveys?
15:16.08ITGuru[TK]D-Fender, Easy with the language, I understand that I'm not getting the terminology 100% correct, and I understand that I'm probably not conveying my issues as well as I could, but seriously, no need for insults, I'm obviously not as competent as you are.
15:16.12verywisemancan i use Grandstream  FXS/FXO ATA with asterisk
15:16.14jtoddI know those are annoying, but assume that it will be used for good instead of evil.  :-)
15:16.38jdnWESTjtodd: YOUR THE GUY! (everyone get him!)
15:17.06jtodd"Your automobile insurance has expired!  Press 1 to rene<gaaaaack>"
15:17.13[TK]D-FenderITGuru: There was no insult there.  I'm just not wasting time trying to sort out your now apparently numerous issues with it.
15:17.42[TK]D-FenderITGuru: When you've got things straightened out I might offer to help with some specific point once progress is made.
15:18.20NovceGuruso thoughts on vitelity for business class (5 channels) sip trunking?
15:18.23[TK]D-Fenderverywiseman: ...
15:18.24[TK]D-Fender~gs
15:18.25jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
15:19.15ITGuru[TK]D-Fender, I understand, I'm sorry - I'm just trying to get through the issues - I may be wrong, and it could be all related, and be just one issue. What is confusing me, the fact that all the endpoints can be seen by the asterisk server - so the problem must be the phones - but the phones are all identical, and just 2 of them do not work, the rest do
15:19.18aiksa[LV][TK]D-Fender: btw I have had experience (cough, cough) with BT102 in the past. Is every product from gs THAT bad?
15:19.43[TK]D-Fenderaiksa[LV]: YMMV, and I would never risk it.
15:19.57*** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr)
15:20.15aiksa[LV]ITGuru: those phones can make calls (ring appears on the other end) but no audio goes through?
15:20.47aiksa[LV]is the network topology the same for all of the phones vs. asterisk?
15:21.17ITGuruaiksa[LV], the phones in question ring when they are dialled, but cannot call out - and the topology of the network is the same for all phones
15:21.56aiksa[LV]ok. when those phones ring, can you lift up and hear audio?
15:22.10*** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6)
15:22.21*** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br)
15:23.09SQLDarklyGood Morning.
15:23.50kannanwill "reload " command in * CLI reload manager.conf?
15:24.15SQLDarklyjust do asterisk -rx "restart gracefully"
15:24.24SQLDarklywith the quotes
15:24.26[TK]D-Fenderkannan: Should
15:25.50*** join/#asterisk mog (n=mog@nat/digium/x-42ccadcb380d1df3)
15:25.50*** mode/#asterisk [+o mog] by ChanServ
15:27.29Shizuotzafrir_laptop: Hi
15:27.54tzafrir_laptopShizuo, hi
15:28.00ShizuoWhy do you guys hate Grandstream? Did they denied you guys funding?
15:28.11Shizuotzafrir_laptop: Is is easy to develop drivers for Asterisk?
15:29.22tzafrir_laptopShizuo, I mostly have exprience with chan_zap/chan_dahdi and the Zaptel/Dahdi drivers
15:29.48Shizuotzafrir_laptop: I'm thinking of building a device
15:29.50orTixDo or did you guys also use or used softphone?>
15:29.54tzafrir_laptopdo you have any prior experince with C? Linux? Drivers for what?
15:30.02[TK]D-FenderShizuo: Perhaps making shoddy products has something to do with it.
15:30.02Shizuotzafrir_laptop: Lots of C experience
15:30.10Shizuo[TK]D-Fender: I don't agree
15:30.33*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
15:30.33*** mode/#asterisk [+o russellb] by ChanServ
15:30.39Shizuo[TK]D-Fender: This looks like groupthink to me
15:30.52*** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net)
15:30.56*** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-531e82755baeb200)
15:31.14[TK]D-FenderShizuo: I guess you are grand judge of that then, aren't you....
15:31.17Shizuotzafrir_laptop: The thing is: Some apps have really shoddy device driver APIs
15:31.19coppiceif making shody products was a reason to not use stuff, nobody would use any VoIP kit at all
15:31.25Shizuocoppice: Exactly
15:31.44eppigyBOOYA
15:31.49Shizuocoppice: I bet Grandstream rejected to fund the project, making the group leaders tell everyone to hate the company
15:32.07[TK]D-FenderShizuo: Wow, and that isn't paranoia.
15:32.18Shizuo[TK]D-Fender: That's just normal OSS behaviour
15:32.23coppiceoh, grandstream stuff is nasty, but is it really worse than most other stuff?
15:32.28*** join/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net)
15:32.46[TK]D-FenderShizuo: No, the constant complaints from users complaining about audio issues flakey firmware, etc don't count
15:32.48Shizuo[TK]D-Fender: I coexist with lots of people from lots of project and that behaviour is not only normal, but it has become the rule, recently
15:33.17ShizuoAudio issies where? Only at Asterisk? Using any app?
15:33.31[TK]D-FenderShizuo: Fine and if you want to walk in with that hypothesis here then you are applying your theory blindly.
15:33.47Shizuo[TK]D-Fender: It's safe to do that nowadays
15:33.56jtoddShizuo: No, that is not the case.  I believe most of the people on this channel have made their own opinions through direct experience, as most of the people that you're speaking with have been doing VoIP for >5 years.
15:34.14SQLDarklyFender are you knocking Theory again ;)
15:34.24SQLDarklyj/k
15:34.26Shizuojtodd: I own 2 racks full of GXW4024 gateways
15:34.30coppicepeople are working around problem in Sonus, Audiocodes, Cisco and all the biggest vendors. the industry standard for VoIP firmware is utter crap
15:34.30Shizuojtodd: They're fine
15:34.41[TK]D-FenderShizuo: So you're ok to make sweeping statements walking in the door blind yet those who've dealt with people with actual complaints here for YEARS don't get to judge based on experience?  That is retarded
15:34.45jtoddShizuo: I have no beef with Grandstream currently, though my past experiences with their initial consumer products were less than favorable.  Recently, though, I have no idea how well they work.
15:35.07Shizuo[TK]D-Fender: Unfortunately, groups are fast to pick enemies
15:35.30Shizuo[TK]D-Fender: That behaviour is called groupthink for a reason
15:35.46eppigyoh man
15:35.49eppigyhere we go
15:35.54jtoddShizuo: Sometimes a cigar is just a cigar.
15:35.56ShizuoHaving the official bot badmouth a vendor because is stupid
15:35.58rob0I was dissuaded from Grandstream by this channel, but I think the opinions sounded fair. Asterisk isn't out begging for funding, Digium is a profitable company.
15:36.09eppigyShizuo: do you have a degree in psychiatry or some shit
15:36.09[TK]D-FenderShizuo: yes, so are "holocost-deniers"
15:36.15eppigysociology
15:36.20eppigyenglish major?
15:36.26Shizuo?
15:36.34eppigy[TK]D-Fender: the holocaust is 100% exagerated
15:36.44eppigywell I guesss 7-%
15:36.48eppigy70%
15:36.57*** join/#asterisk maddog01 (n=minotaur@d221-91-175.commercial.cgocable.net)
15:36.57ShizuoHolocost just looks like Holographic Cost to me
15:37.02SQLDarkly60% of the time is works Everytime!
15:37.03[TK]D-FenderShizuo: And all the psychology you can must doesn't undo a known checkered past.
15:37.17[TK]D-Fendermuster*
15:37.21Shizuo[TK]D-Fender: Ok, if you're happy with the boundaries given to you by your masters, ok
15:37.26ShizuoI guess most people tend to live that way
15:37.41SQLDarklylol masters.
15:37.41jtoddnotes the sign beside the bridge that says "Do not feed the trolls."
15:37.41[TK]D-FenderShizuo: Masters huh?  And who would those be?
15:37.44rob0Shizuo is coming across like a fool, with a faulty assumption leading to faulty conclusions.
15:38.06rob0Pipe up with some evidence about the "denied funding".
15:38.19Shizuorob0: Sorry, but the official (# and not ##) channel for Asterisk should never badmouth hardware vendors like that
15:38.31Shizuorob0: It's childish and makes the whole thing look amateur
15:38.41rob0~sangoma
15:38.41jbotsomebody said sangoma was a Canadian based company that makes PRI and Analog cards. See their site at http://www.sangoma.com/
15:38.46SQLDarklyThe channel isnt badmouthing
15:38.51[TK]D-Fenderrob0: No, he only layed funding as "suspect".  He has absolutely no factual basis, only vauge psychology theory.
15:38.55rob0That's a Digium competitor.
15:39.01SQLDarklyits an opinion
15:39.21ShizuoIts the official Asterisk channel saying that company X is crap
15:39.24[TK]D-FenderShizuo: O RLY?
15:39.50SQLDarklyIf I crap then guess whats its crap. Sometimes companies really make crap
15:39.54[TK]D-FenderShizuo: Since when does this channel have to stifle people who happen to frequent it?
15:40.00SQLDarklyCisblows Call Manager for one
15:40.00ShizuoNo problem...
15:40.17ShizuoIn fact, it's not my channel and not my product, so
15:40.19jjshoewelcome to #bicker full of whiney bitches since 1996
15:40.29Shizuo:D
15:40.42SQLDarklyGreat here comes the Chuck Norris puns
15:40.45rob0My "groupthink" was reinforced by this analysis.
15:40.50eppigybottom line 6 million jews did not die in the holocaust
15:41.02SQLDarklyStarting to sound like WoW Barrens Chat in here
15:41.05[TK]D-FenderShizuo: the channel isn't saying anything.  Just a shared optinion from many prominent members who have no particular affiliations with any hardware vendor.  That's a diverse bunch.
15:41.17SQLDarklyBingo Fender
15:41.24Shizuo[TK]D-Fender: Sure, yeah, sure
15:41.31*** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca)
15:41.57[TK]D-FenderShizuo: YOU'RE the newb here, and want to tell us "how it is"?  Sorry, nobody is buying that line.
15:42.23ShizuoWhatever, men
15:42.28rob0jtodd was right, I'm done.
15:42.56SQLDarklyOk boot the fool and lets get back to doin what we do. "Trashing Vendors"
15:42.59timeshellIs there an oslec patch that will work with dahdi?
15:43.07*** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
15:43.09Shizuo"Oh, we are not a hate group, our leaders share a opinion that these people should be shot and killed, but that's not the group's opinion"
15:43.14*** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com)
15:43.38ShizuoIt's a good line to avoid being jailed, but fails at common sense
15:43.57*** join/#asterisk voipnet-tech (n=voipnett@216.195.128.62)
15:43.57clyrradDo any of you guys know of a way to generate a ring tone for X number of seconds?
15:44.12eppigyclyrrad: Shizuo is our resident expert
15:44.18ShizuoThanks
15:44.18clyrradI know about "ringing" but cant specify an interval
15:44.20voipnet-techis this the right channel for asterisk support?
15:44.30eppigyvoipnet-tech: yes please direct questions to Shizuo
15:44.58voipnet-techthanks, and hello.  Shizuo is this correct?
15:45.04SQLDarklyYes it is. Just an off-topic conversation going on.
15:45.09*** join/#asterisk arpu (n=arpu@chello080109017021.12.14.vie.surfer.at)
15:45.12[TK]D-FenderShizuo: You're right and noone should advise others against products that are the product of mass recalls like Bridgestone tires whose blowouts have killed people.
15:45.41[TK]D-FenderShizuo: And of course nyou're also comparing our opinions to those advocating MURDER.  NIIIICCCE
15:45.44SQLDarklyPeople kill people not tires
15:45.48SQLDarklyor is it guns
15:45.51SQLDarklyi forget
15:46.01SQLDarklyno doubt lol
15:46.03[TK]D-FenderSQLDarkly: No.  Guns don't kill people, *I* do :p
15:46.16voipnet-techI'm getting a Segmentation fault when I try to dial console/dsp.  chan_alsa is dying.  this is a new linux install.  have not had this problem on other servers. what should I try
15:46.23*** join/#asterisk DarkRift (n=dark@65.92.169.136)
15:46.24Shizuo[TK]D-Fender: Hey, I don't want to stand between companies and your inconditional hatred for them
15:46.33Shizuo[TK]D-Fender: Feel free to keep acting like that
15:46.38clyrradHrm, anyone have any suggestions for me?  Im all eyes :)  I just need to play a ringing sound to the caller for X number of seconds....
15:47.41[TK]D-FenderShizuo: unconditiona?  No, being plaged by handset echo is a GREAT basis for dislike of a product <-
15:47.46jjshoeclyrrad did you search for "play ringing" on voip-info.net ?
15:47.53NovceGurus
15:47.58*** join/#asterisk mintos (n=mvaliyav@123.236.161.151)
15:48.20Shizuo[TK]D-Fender: Sure! That sure allows for eternal badmouthing of the said company, under the official bot of the official channel
15:48.43Shizuo[TK]D-Fender: And I really mean eternal! No more Firestone tires for me or Greatsuck gateways for my company!
15:48.59Shizuo[TK]D-Fender: I feel much safer now after hearing your advice
15:49.13[TK]D-FenderShizuo: You, being a user of their products are therefor unbiased as well and couldn't possibly be suffering from "fanboy-ism" <-  Want to play the psych game?  goes both ways.
15:49.14jjshoeis this a good time to do the following?
15:49.17jjshoe~grandstream
15:49.18jbothmm... grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
15:49.27SQLDarklyclyrrrad why do you want to play a fake ring? Just record a sound and play it back......... I do not see the point but I am not in your shoes so have at it ;)
15:49.33Shizuo[TK]D-Fender: Sure! I'm a fanboy of VoIP gateway hardware!
15:49.50jjshoeSQLDarkly I've used it in cases where a t1 picks up before the telco can even get a ring in
15:50.00[TK]D-FenderShizuo: Not possibly even a specific brand which is under contention!.
15:50.09[TK]D-Fenderload chan_hypocrisy.so
15:50.12coppice[TK]D-Fender: when a snom or many other handsets have echo problems, people say turnt he gain down. When a grandstream does it they say grandstream sucks. realistically every VoIP phone out there is pretty mediocre
15:50.16Shizuo[TK]D-Fender: They'll be releasing new products next week, I think I'll sleep on the street, in front of the store, to be the first to get them!
15:50.24voipnet-techi don't mean to interupt your wonderful offtopic discussion here, but I have 4 servers to get working today and I can't get console audio to work, and i need it for external paging.  is there anyone here that cares to help me figure out why asterisk gives me a Segmentation Fault and exits?
15:50.24[TK]D-FenderERROR : chan_hypocrisy.so is already loaded
15:50.27SQLDarklyjjshoe good point. I stand corrected
15:50.28clyrradSQLDarkly: its for a special application I am developing, I background (Silence/3) for example, but I dont want the caller to think the call was dropped and have dead air, so instead I would like to play a ringing sound so the caller knows they are being connected
15:50.50Shizuocoppice: Too bad that GS never opened their deep pockets to the project, like some other companies do
15:50.57Shizuocoppice: That means eternal badmouthing of the brand
15:51.40[TK]D-FenderShizuo: and I don' work for Digium, or code for *, nor do many of the people who've used it for years.
15:52.01SQLDarklyShizuo hush lol this is not a channelf or your bullcrap. I HATE many companies like CISBLOWS but their 79XX phones rock balls. Its just about what works nothing more.
15:52.06jjshoeclyrrad record ringing and play it back, as SQLDarkly suggested
15:52.27Shizuo[TK]D-Fender: There is no need to convince me, ok? Your masters have teached you right: GS sucks... I'm with you on that one;
15:52.28SQLDarklyA collection of opinions is all your reacting to. Not some Digium conspiracy
15:52.34Shizuo[TK]D-Fender: No more GS or Firestone for me
15:52.55jjshoeShizuo I agree, can you take this to #teenchat with your boyfriend?
15:52.57[TK]D-FenderShizuo: You fail to understand : I *HATE* no masters here, and no paid loyalty to ANY BRAND
15:53.10[TK]D-FenderShizuo: You are talking out your ass.
15:53.15[TK]D-Fenderhave*
15:53.19Shizuo[TK]D-Fender: They're your masters, you obviously don't hate them
15:53.21clyrradjjshoe: I just thought there would be an eaiser way using Playtones or something similar
15:53.39[TK]D-FenderShizuo: "they"?  Who the hell is "they"?
15:53.53Shizuo[TK]D-Fender: The group is one of them
15:53.57Shizuo[TK]D-Fender: Master n 1
15:54.11ShizuoYou can name the others, as I don't know you that much :D
15:54.16jjshoeQwell ping
15:54.27SQLDarklyShizuo go take your NerdRage elsewhere please. People have work to do and you obviously write for the a tabloid
15:54.34[TK]D-FenderShizuo: Pshychology double-skeap.  You are so completely full of shit, its laughable.  Take your entire psych background and toss it right out the window because ti is worthless here.
15:54.35jjshoeclyrrad hrm, looks like you might be able to, did you research playtones on voip-info.org ?
15:54.37[TK]D-Fenderspeak*
15:54.37*** join/#asterisk XnOSX (n=XnOSX@212.145.172.127)
15:54.46jjshoeclyrrad I hit an article which pointed to indications.conf and seemed to make it sound plausable
15:55.00ShizuoA very people group of people :D
15:55.02clyrradjjshoe: its what I have been doin gthe last 30 minutes, just wanted to check here to see if anyone else has done this before
15:55.04ShizuoA very polite group of people :D
15:55.17[TK]D-FenderShizuo: "Secret Masters"... go off Illuminati-hunting... I hear its all the rage now
15:55.24jjshoeclyrrad what link are you at right now on voip-info ?
15:55.27ShizuoAnd that's only because I complained of an official bot badmouthing a company
15:55.39clyrradjjshoe: Asterisk+cmd+Playtones
15:55.41ShizuoThe excessive reaction to that showed that I was correct on my assumptions
15:55.47jjshoeclyrrad full url please.
15:55.54SQLDarklyDO you work for GS?
15:55.59[TK]D-FenderShizuo: Official bot?  Wow you continue to assert I have any particular affiliation....
15:56.17clyrradjjshoe: http://www.voip-info.org/wiki/view/Asterisk+cmd+Playtones
15:56.20Shizuo[TK]D-Fender: Oh crap, you're stupid as a doorknob
15:56.33SQLDarklyGS simply puts out garbage compared to excellent phones like the 79xx
15:56.39Shizuo[13:20] <jbot> GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
15:56.48SQLDarklyand THAT cannot be denied even by GS themselves
15:56.51[TK]D-FenderShizuo: No, you are master of the blind accusation that those of opinion must be paid to think or "controlled"
15:57.01[TK]D-FenderShizuo: So much for self-determination.
15:57.12ShizuoNo, you're stupid. I mentioned BOT several times and you still think I'm talking about YOU
15:57.18ShizuoIt can not get more stupid than that
15:57.20SQLDarklyShizuo you your windows still dont you?
15:57.35ShizuoSQLDarkly: You your doors, don't you?
15:57.37[TK]D-FenderShizuo: Do clarify exactly what bot statement and I will enlighten you.
15:57.57Shizuo[TK]D-Fender: Pasted on the channel, Einstein
15:58.16[TK]D-FenderShizuo: Feel free you repaste it now specifically :)
15:58.22[TK]D-Fenderto*
15:58.24ShizuoNo I won't, it's 8 lines away
15:58.30ShizuoIf you're that limited, sorry
15:58.33SQLDarklyShiz have you even used GS with Asterisk? Yes they do work but they have to be replaced a lot
15:58.34[TK]D-Fender~gs
15:58.34jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
15:58.37[TK]D-Fender^^ this guy here?
15:59.05SQLDarklyQuality phones like 79xx by Cisblows vs a Garbage GS phone that is fine for testing but not enterprise level deployment
15:59.07voipnet-techi'm checking here for someone that might be able to give me some direction to why i can't get console/dsp calls to work
15:59.07voipnet-tech[10:58] <voipnet-tech> initially chan_alsa wouldn't compile because make menuselect had it XXX out says: depends on asound(e)
15:59.07voipnet-tech[10:59] <voipnet-tech> i think that's because i ran .configure before i installed alsa
15:59.07voipnet-tech[10:59] <voipnet-tech> now alsa is installed and working (alsamixer and play/aplay work)
15:59.08voipnet-tech[10:59] <voipnet-tech> and after rerunning ./configure and make menuselect
15:59.10voipnet-tech[10:59] <voipnet-tech> i can now select chan_alsa
15:59.12voipnet-tech[10:59] <voipnet-tech> and make, make install
15:59.14voipnet-tech[10:59] <voipnet-tech> and asterisk runs
15:59.16voipnet-tech[11:00] <voipnet-tech> if I start asterisk with safe_asterisk, i always get a busy/congested calling console/dsp
15:59.19voipnet-tech[11:00] <voipnet-tech> if i start it as just asterisk, i can call it, but then asterisk gets a segmentation fault
15:59.25rob0yikes
15:59.26[TK]D-Fendervoipnet-tech: STOP SPAMMING
15:59.35LemensTS~pastebin
15:59.36jbot[~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
15:59.43rob0Now we have a new focal point for our wrath. :)
15:59.45[TK]D-FenderShizuo: So, that one?
15:59.52[TK]D-FenderShizuo: Lets be clear....
15:59.57Shizuovoipnet-tech: Stop trying to get support here! This is the place to tap people on the back and badmouth companies!
16:00.04Shizuovoipnet-tech: Not for support!
16:00.08voipnet-techthis is why asterisk sucks, you guys
16:00.25voipnet-techi'm asking legitmate questions for support
16:00.27SQLDarklyWrong Shiz. You are getting this channel off topic.
16:00.29LemensTSvoipnet-tech: pastebin your problem then post the link
16:00.36voipnet-techand you guys are spamming the room with the most offtopic crap
16:00.46SQLDarklyYes voip but your spamming it. Address someone and start from the beginning
16:00.52SQLDarklywe dont even know what * version you have
16:00.55SQLDarklywhat distro your on
16:00.56SQLDarklyetc
16:01.06voipnet-techand i ask legit question all be-it lengthy, and i just get shit for it
16:01.08jjshoeclyrrad do you see where it says to look for what tones are defined?
16:01.16SQLDarklyKey things in order to help you clear your problem
16:01.17[TK]D-FenderShizuo: Who are you "helping" now troll?
16:01.24LemensTSvoipnet-tech: ask it or leave
16:01.33clyrradjjshoe: yea I am still checking on it and watching all the nonsense happing in this channel :(
16:01.33[TK]D-FenderShizuo: So answer the previous question.
16:01.35voipnet-techi have asked
16:01.41voipnet-techthat's what the paste was
16:01.47voipnet-techi'm explaining my situation
16:01.50voipnet-techi'm checking here for someone that might be able to give me some direction to why i can't get console/dsp calls to work
16:01.50voipnet-tech[10:58] <voipnet-tech> initially chan_alsa wouldn't compile because make menuselect had it XXX out says: depends on asound(e)
16:01.50voipnet-tech[10:59] <voipnet-tech> i think that's because i ran .configure before i installed alsa
16:01.50voipnet-tech[10:59] <voipnet-tech> now alsa is installed and working (alsamixer and play/aplay work)
16:01.51voipnet-tech[10:59] <voipnet-tech> and after rerunning ./configure and make menuselect
16:01.53voipnet-tech[10:59] <voipnet-tech> i can now select chan_alsa
16:01.54SQLDarklyvoipnet-tech you have yet to tell anything about your specs dude
16:01.55voipnet-tech[10:59] <voipnet-tech> and make, make install
16:01.55LemensTSvoipnet: im not reading that you need to pastebin it. sob.
16:01.57voipnet-tech[10:59] <voipnet-tech> and asterisk runs
16:01.57jjshoeclyrrad when you follow that link, do you see ring?
16:01.59voipnet-tech[11:00] <voipnet-tech> if I start asterisk with safe_asterisk, i always get a busy/congested calling console/dsp
16:02.02voipnet-tech[11:00] <voipnet-tech> if i start it as just asterisk, i can call it, but then asterisk gets a segmentation fault
16:02.05voipnet-tech:-)
16:02.17LemensTSim not reading 3 words on each line 20 lines long
16:02.18voipnet-techgod i hate this room
16:02.24voipnet-techBroadsoft rules, asterisk blows
16:02.24SQLDarklyvoipnet-tech we need infor about your rig.
16:02.26clyrradjjshoe: I see in indications.conf there is a tone defination for ring, I am going to try that
16:02.26Shizuo[TK]D-Fender: That issue is as clear as it can get. You're trying to make the discussion bigger as it fills your trollish void, but I'm not in the mood of cooperating
16:02.30jjshoevoipnet-tech sweet
16:02.33jjshoeclyrrad perfect :)
16:02.39SQLDarklyvoipnet-tech how can we diagnose a problem with no information remotely?
16:02.42SQLDarklyanswer that?
16:02.50voipnet-techi gave you the info
16:02.53[TK]D-FenderShizuo: Was that, or was that not the "bot" comment you're basing "rooom" bias on?
16:02.58voipnet-techyou're just refusing to read it unless i put it in pastebin
16:03.04LemensTSexactly
16:03.09SQLDarklyvoipnet-tech wrong you didnt you gave symptoms of your problem
16:03.22lowtekHey TK, good call on Telephony Depot, they rock!
16:03.26SQLDarklyvoipnet-tech I asked 3 times already for versions and distro
16:03.28Shizuo[TK]D-Fender: Check what I pasted back there. Until then, I won't pollute the channel by giving you attention
16:03.29jjshoeclyrrad let me know how that works out
16:03.35clyrradjjshoe: yup
16:03.43*** join/#asterisk errr (n=mike@fedora/errr)
16:04.10SQLDarklyvoipnet-tech You are insisting on insulting us saying the room sucks. Do not take out your anger on us. We are trying to help but certain information is needed.
16:04.29jjshoewelcome to #teenchat look out, everyone is on the rag right now!
16:04.42SQLDarklyvoipnet-tech You obviously came here for some help the least you can do for the free support and our time is help us understand a bit more about your setup would you not agree?
16:05.05voipnet-techyes, i'm using all the latest stable builds of everything
16:05.08voipnet-techi said that
16:05.23voipnet-techcentos, *, zap
16:05.37[TK]D-FenderShizuo: Well you don't seem to have confirmed that as the "room bias" bot statement.
16:05.40jjshoelatest stable build of which release? 1.2 1.4?
16:05.41zoid_99I need a CLI command to move all messsages in someone's INBOX to someone else's INBOX.
16:05.43SQLDarklyThat wasnt in your spam ;) Dont get snapping we are here to help we simply like to read organized information
16:05.46voipnet-tech14
16:05.55jjshoezoid_99 asterisk cli?
16:06.43SQLDarklynow let me see what I can do for you mr meanie :)
16:06.43zoid_99jjshoe: yeah
16:06.43jjshoezoid_99 I don't know of any, but I'm not the most familiar
16:06.43zoid_99voicemail forward new 1111 to 2222
16:06.43[TK]D-Fenderzoid_99:
16:06.43SQLDarklyvoipnet-tech can you please pastebin the console output when you console dial for me please?
16:06.46*** join/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:06.59KOCATEPEhi all
16:07.02SQLDarklyEnsure debug is on
16:07.05KOCATEPEany one from Turkey ???
16:07.07voipnet-techsure
16:07.08*** join/#asterisk Sargun (n=Sargun@66.151.148.225)
16:07.15SQLDarklyyou can turn on debug in the console via logger.conf
16:07.24SQLDarklybe sure to turn it off before you go back into production
16:07.53zoid_99jjshoe:  I've started writing it, but it's going to require a little more work than I thought.
16:08.02SQLDarklyafter you change the logger.conf you will need to asterisk -rx "restart gracefully"
16:08.23jjshoezoid_99 oh? how come?
16:08.28*** part/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:08.30*** join/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:08.36SQLDarklybrb while your doing that I am in need of coffee.
16:08.48*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
16:08.51*** part/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:08.52*** join/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:08.59zoid_99jjshoe: because I suck at C :)
16:09.12[TK]D-Fenderzoid_99: You're coding a CLI command to do this?
16:09.19zoid_99yes
16:09.24jjshoezoid_99 ahh, why not write an agi in a language you know?
16:09.32jjshoezoid_99 or doesn't asterisk have a system() ?
16:09.32*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
16:09.39[TK]D-Fenderzoid_99: And yes, it is quite challengiing give the way the files get numbered, etc
16:09.48jameswfHEY #ASTERISK your OT
16:09.48clyrradjjshoe: no joy, didnt do what I need, after the 3 seconds are up, there is no ringing when the actuall phones are ringing, looks like its back to the drawing board
16:09.50jameswf:)
16:09.57SQLDarklyFYI for all the naysayers. This room is a great resource just understand people come in here and start yellin and need to address their issue in an organized manner. You have to think how many questions get asked here and how many are repeat. THis room had taught me a lot over the years and my hat goes off to that surly bastard fender lmao he subjects himself to helpdesk 24/7
16:10.20jjshoeclyrrad hrm, the console showed it trying to play the tones? honestly I've always just played back a ringing gsm :)
16:10.22[TK]D-FenderSQLDarkly: Almost :)
16:10.27zoid_99[TK]-Fender: I thought about doing it externally but I need the locking capabilaties of app_voicemail to keep from having a race condition
16:10.43voipnet-techwell... it appears the last reboot fixed it
16:10.45voipnet-technvm
16:10.47jameswfSQLDarkly: point of order I believe [TK]D-Fender's parents were married
16:10.49voipnet-techyou may resume your offtopic bs
16:10.54clyrradjjshoe: so you recorded a ringing sound for how every many seconds you needed?  If so, how did you acomplish the recording of the ring sound?
16:10.55voipnet-techthanks for trying SQLDarkley
16:10.58voipnet-techgold star for u
16:11.08[TK]D-FenderSQLDarkly: I love how people feel I'm affiliated with XYZ.  With my loyalty lasts until something better comes along or something I liked more than something else goes south :)
16:11.09zoid_99jjshoe: yes it does but I need to ensure that the voicemails are locked while I do this
16:11.11jjshoeclyrrad just a single ring, then you can decide how much to play it
16:11.23clyrradjjshoe: how did you record the single ring?
16:11.23jjshoezoid_99 so why not implement the lock algo. in another language?
16:11.57zoid_99jjshoe: because asterisk will be twiiddling with the files behind my back
16:12.07[TK]D-FenderSQLDarkly: Even Asterisk itself :)  Every piece of hardware I run here at the office can play nice with PLENTY of alternative PBX solutions besides *.  Not one piece of my solution "owns me".  Masters they say... LOL!~!!!!
16:12.10*** join/#asterisk riddlebox (i=80fc9a6f@gateway/web/ajax/mibbit.com/x-f3c0fdef7df0ff2b)
16:12.31[TK]D-FenderSQLDarkly: No F-ING clue :)
16:12.37jjshoezoid_99 that doesn't make much sense...
16:12.43SQLDarklyD-Fender I am surprised he isnt destroying bloggers for their opinions and experiences
16:12.55SQLDarklyok back now voipnet-tech
16:12.55jjshoezoid_99 are you suggesting app_voicemail.c ignores locks by any application other then itself? O.o
16:13.04SQLDarklyDo you have your output ready?
16:13.15LemensTSSQLDarkly: it fixed when he restarted
16:13.22clyrradjjshoe: Did you see my last msg?
16:13.28jameswfjjshoe: I am pretty sure you touch .lock it will cause issues
16:13.38SQLDarklyLemensTS ah excellent
16:13.54LemensTSYou got a gold star for your help too if you read up heh
16:13.54[TK]D-Fenderlowtek: You're welcome
16:14.11[TK]D-FenderlowSo far I an all those I have referred there have been very happy with them
16:14.14zoid_99jjshoe:  I'm saying that app_voicemail  numbers files and renumbers files.  If I move a file while someone is leaving a voicemail then things will get out of whack
16:14.22jameswfsometimes if an asterisk system crashes we have to go in anr rm .lock files
16:15.00jjshoeclyrrad show application ringing
16:15.11jjshoezoid_99 presumably not if you lock it no?
16:15.52clyrradjjshoe: This is what you used to record the ring sound?  Or I am not following what you mean?
16:15.57zoid_99jjshoe: old files may be locked but it doesn't prevent app_voicemail from creating a new voicemail file
16:16.00voipnet-techsql problem fixed
16:16.03timeshellLooking for some detailed instructions to use OSLEC with DAHDI
16:16.03voipnet-technew problem
16:16.08voipnet-techwho supports safe_asterisk ?
16:16.13zoid_99jjshoe: and the new file will be numbered wrong
16:17.28*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
16:17.38ShizuoOh Asterisk...
16:18.03zoid_99but in app_voicemail I can use AST_LIST_LOCK (I think)
16:18.15[TK]D-Fendervoipnet-tech: its part of the core
16:18.38jjshoeclyrrad type that in asterisk
16:19.04jjshoezoid_99 ah, it's been a long time since I've looked at app_voicemail.c
16:19.08clyrradjjshoe: yup I did, I am aware of this command, I am just not following what you are trying to tell me
16:19.11*** part/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:19.14jjshoeclyrrad use it.
16:19.16*** join/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:19.42clyrradjjshoe: When I use that, after its doen the backgroud(silence/3) you get no ring when the phones are ringing
16:20.07jjshoeclyrrad how long do you ring for?
16:20.17clyrradjjshoe: 3 seconds
16:20.28jjshoeclyrrad while you play silence for 3 seconds at the same time..
16:20.46clyrradjjshoe: not sure what you mean?
16:21.06clyrradI call Ringing() first, then background(silence/3)
16:22.07jjshoeclyrrad ah
16:22.12*** part/#asterisk KOCATEPE (n=admin@88.247.133.123)
16:23.03clyrradjjshoe: but it does not work
16:23.07jjshoeclyrrad add a wait after the ringing
16:25.50*** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr)
16:25.50[TK]D-Fenderclyrrad: Ringing WILL stop for audio...
16:26.05clyrradjjshoe: the problem is as soon as you do Background(silence/3) the ringign stops
16:26.18[TK]D-Fenderclyrrad: And why wouldn't it?
16:26.19clyrradjjshoe: I need to record the ring sound as you initialy said, how did you manage this?
16:26.24[TK]D-FendercyYou're telling it to play audio
16:26.46clyrradjjshoe: To record the ring sound seems to be the only solution here since Playtones didnt work for me
16:26.52lowtekI didn't catch the first part of this, but why not just Wait(3) after Ringing()?
16:27.14jjshoeclyrrad an english definition of your call flow, and a pastebin of your context would sure help
16:27.19clyrradlowtek: I have not tried that approach, let me give it a go
16:27.24*** join/#asterisk hetii (n=vircuser@193.159.172.162)
16:27.27hetiihi
16:27.36*** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com)
16:27.47jjshoeclyrrad I just told you to try that and you didn't bother??
16:27.48lowtekclyrrad: This is what TK was trying to say, Playing() the file silence/3 still engages the playback function and will terminate the ring function.
16:27.59jjshoeblah.
16:28.10[TK]D-Fenderclyrrad: So as you've been told by others, jst "Wait(3)
16:28.39clyrradWait(3) does "not" work, it wont allow the user to press keys, this is why I had background(silence/3)
16:29.17[TK]D-Fenderclyrrad: Then go make an audio recording of 3 seconds of ringing and background THAT
16:29.19lowtekclyrrad: Then find or create an audio file that sounds like a ring at whatever length you need and Background() it.
16:29.34[TK]D-Fenderlowtek: tag, you're "it" :)
16:29.36clyrradI need to have a bit of a pause where the user can hit a hidden feature before the phones start to ring, that is what I am trying to acomplish.... now having said that..... I cant just background(silence/3) becase the user will think the call was dropped, hence the reason I am trying to play a ringing sound....
16:29.42lowteklol
16:30.05clyrrad[TK]D-Fender: Yes, thats my question....... HOW to record that ring sound?  I need the ring sound to match Asterisk ring sound so its seemless
16:30.24[TK]D-Fenderclyrrad: Go monitor a call to "ringing"
16:30.29clyrradlowtek: Yes, thats my question, I not sure how to record that rinning sound
16:30.35[TK]D-Fenderclyrrad: this ain't Raw-Cat Science
16:30.58clyrrad[TK]D-Fender: I understand, but I have not done it b4, its easy to be an expert once you've done it once, I have not done this before
16:31.07[TK]D-Fenderclyrrad: MONITOR <-
16:31.16lowtekclyrrad: You could record it right from asterisk with Record() if you want.   Do a Answer(), Monitor(), Ringing(), Wait(3).
16:31.16clyrradI am checking on that now
16:31.28[TK]D-Fenderclyrrad: Start recording.  Cal ringing, wait a few sec, hang up.
16:31.32clyrradlowtek: thanks thats what I was looking for
16:31.36lowtekclyrrad: then just find the file it created, and Background() it.
16:31.44*** join/#asterisk Alric (n=Alric@masq.hyperusa.com)
16:31.44[TK]D-Fenderclyrrad: We're taling what, 4 lines of dialplan with 3 apps we hand-fed ?
16:31.51clyrradlowtek: yup I see what your saying now, thanks
16:31.59clyrradmuch apprecated :)
16:32.14[TK]D-Fenderlowtek: No, not "record" unless he's doing a 3-way to bridge from a phone, etc
16:32.46lowtekclyrrad: right, use Monitor(), not Record(), my bad.
16:33.15clyrradlowtek: no probs, I was just looking for the approach, what you defined was great, I can research based off that, thanks again for your help
16:33.36*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
16:35.25ShizuoLol, Asterist
16:36.43voipnet-techi'm gonna slit my asterist
16:37.19clyrradhaha
16:37.45[TK]D-Fender~emo
16:37.46jbot/wrists
16:37.55Shizuo~badmouthing
16:38.38jjshoe~grandsuck
16:38.54[TK]D-FenderFAILURE :)
16:39.06voipnet-techi've got one of those that i can't get to place outward calls
16:39.15jjshoeone of what?
16:39.19voipnet-techa gateway
16:39.22voipnet-techfrom grandsuck
16:39.26voipnet-tech4 port fxo
16:39.33jjshoeoh, yeah, use it to level your desk off, or throw it away.
16:39.39riddleboxmy grandstream phones work fine
16:39.40[TK]D-Fendervoipnet-tech: What is an "outward call"
16:40.04[TK]D-FenderShizuo: OMG, a testimonial, quick, squash him, his "masters" are clearly controlling him!
16:40.05voipnet-tech[TK]D-Fender, referring to calls where the gateway would dial out on the pots line
16:40.27jjshoeriddlebox I've used many a grandstream that work ok, I've also used many aastra's and polycom's that work great :)
16:40.30riddleboxouch there are 10 foot poles
16:41.13*** join/#asterisk CunningPike (n=arodgers@204.239.10.119)
16:41.17carrancahas anyone configured asterisk with the app h324m?
16:41.25[TK]D-Fendervoipnet-tech: well if you provide debug and shot of your configs maybe someone might have idea of some settings to change to get it working.
16:41.36[TK]D-Fendervoipnet-tech: Pasterbin is your friend.
16:41.47voipnet-techit's not running with *
16:41.52voipnet-techdifferent softswitch
16:42.12voipnet-techand i've taken full debugs and sent them to GS support and even they can't figure them out
16:42.25voipnet-techthey said that the debugs i sent to them are too confusing
16:42.30voipnet-techi laughed
16:42.31voipnet-techand hungup
16:42.44*** join/#asterisk UQlev (n=kvirc@91.184.220.73)
16:43.43voipnet-techi'll be able to fix it myself i'd bet, that's what always ends up happening, i just need time to sift thru the wireshark
16:43.50*** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70)
16:45.01kb3ienanyone care to praise a softphone for windows, ideally a Free one, but a free (as in zero-monies) is okay too.
16:45.10*** join/#asterisk DarkRift (n=dark@65.92.169.136)
16:45.13voipnet-techx-lite
16:45.32kb3ieni never touch win32 with a *long* pole, its too krufty...
16:45.41kb3ieni'll pass the suggestion alon.
16:45.46UQlevkb3ien: phonerlite
16:45.46*** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net)
16:46.10[TK]D-Fenderkb3ien: All of the "free" ones have shortcomings that frustrate one group or another.
16:46.21ShizuoOmg, kb3ien: You're soo leet
16:46.41pta200Any idea when Asterisk 1.4.23 will be officially released? Seems like there are a lot of fixes since 1.4.22
16:46.55[TK]D-Fenderkb3ien: Zoiper is "OK" and supports SIP/IAX & native call transfer... so that might do.  X-Lite is a bit more stable, but more crippled
16:47.04*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
16:48.19UQlev[TK]D-Fender: does zoiper support proxy?
16:48.50[TK]D-FenderUQlev: No idea, I've only used it sparingly and direct to *
16:49.26clyrradlowtek: That solution worked thanks bud much appreciated
16:49.27*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
16:49.28UQlevI have not found any proxy settings in zoiper
16:49.34clyrradjjshoe: thanks awell for your assistance with this :)
16:50.48*** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net)
16:51.01*** part/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net)
16:52.46NovceGuruanybody care to recommend an ITSP for business sip trunking? (5 channels)
16:53.44*** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk)
16:55.06clyrrad~ITSP
16:55.06jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
16:55.13*** join/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za)
16:55.38clyrrad~itsplist
16:55.44NovceGuruI was thinking about actual experiences/input instead of that which i've seen 100 times :)
16:55.59timeshellAsterisk hacked?  http://www.ic3.gov/media/2008/081205-2.aspx
16:56.09russellbugh
16:56.16russellbplease read blogs.digium.com ...
16:56.26ShizuoLulz, owned
16:56.45riddleboxold news
16:56.48russellbspecifically: http://blogs.digium.com/2008/12/06/sip-security-and-asterisk/
16:56.54timeshellGuess that was a while ago
16:57.35*** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk)
16:58.00*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
16:58.55NovceGuru~itsplist-us
16:58.56jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
16:59.49NovceGuruI wonder if voicepulse ever picked up any more proxies to help with ping times
17:02.05NovceGuruthey do have a shiny new site, i'm sold!
17:02.24lowtekNovceGuru: Try FlexPulse -> http://flexpulse.com
17:02.31[TK]D-FenderNovceGuru: I've had clients who've been quite happy with them.
17:02.59ShizuoSeriously... Why?
17:03.02DarkRiftAnyone knows a good Canada Telco for Voip services or DID termination I can access through an asterisk installation I host ?
17:03.04NovceGuruI used them years ago when tinkering...they seemd decent
17:03.57[TK]D-FenderDarkRift: les.net & unlimitel.ca
17:04.03DarkRiftThank you
17:04.04NovceGurumy only issue is im in BFE and most ITSPs can't port my number
17:04.08[TK]D-FenderDarkRift: both very solid
17:04.52DarkRiftAlright, I'll have a look, thank you
17:05.13*** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
17:06.12*** part/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
17:06.18*** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
17:06.36[TK]D-Fenderlowtek: boingBOING
17:07.33lowtekAT&T sucks.
17:07.42theharindeed
17:07.48ShizuoJust like GS
17:07.57ShizuoAt least that's what the local bible says
17:09.03NovceGurublah voicepulse can't port me :\
17:10.19NovceGuruI have one POTS line that I am keeping for fax, I wonder if I can have 2 numbers attached to it, but only foward one of them
17:10.29NovceGuruI doubt it
17:10.41[TK]D-FenderNovceGuru: Have any providers been able to offer to?
17:11.17NovceGuruI don't believe so, but possibly. I know sometimes their website says they can't but they actually can
17:11.57[TK]D-FenderNovceGuru: Dirty trick, but ask Vonage.  If they say yes, then you can do a 2-stage swap to swpie it from them once they've got it.
17:12.11[TK]D-FenderNovceGuru: Sometimes you just need to take it out of the first guy's grip
17:12.24NovceGuruyeah, I think p8 was able to
17:12.54*** join/#asterisk Assid (n=assid@unaffiliated/assid)
17:12.59pta200Any idea when Asterisk 1.4.23 will be officially released
17:13.21NovceGuruer, they can't, shooooooooot
17:13.23[TK]D-Fenderpta200: Never
17:13.32pta200riiiiiiiiiight
17:13.38NovceGurualthough time warner is able to, but I don't want to use their shitty analog lines
17:13.50[TK]D-Fenderptathis is an OSS project.  they don't have deadlines and this kind of question will never get a solid date handed out.
17:14.20[TK]D-Fenderpta200: and that wasn't "never" as in "will never be released", just "Never going to get an answer"
17:14.34NovceGuruvoange can't also...hell
17:14.44[TK]D-Fenderpta200: Know how long people were asking "when is 1.6 noon RC going to be released"?  MONTHS
17:14.53pta200I don't doubt it
17:15.12pta200I just thought I would ask after reading through the change log
17:15.20[TK]D-Fenderpta200: So don't hold you breath on an answer for 1.4.23
17:15.27pta200go it
17:15.29pta200got it
17:15.50[TK]D-Fenderpta200: if there is something particularly important to you in there upgrade to a newer SVN release
17:16.40*** join/#asterisk murdock_ut (n=chatzill@64-42-64-98.atgi.net)
17:17.33timeshellAny inside news on the skype channel?
17:17.47[TK]D-Fendertimeshell: Nope
17:17.55[TK]D-Fendertimeshell: Still closed beta
17:18.04NovceGuruohhh broadvoice can xfer it!
17:18.42*** join/#asterisk outtolunc (n=me@adsl-76-211-236-48.dsl.pltn13.sbcglobal.net)
17:20.50*** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net)
17:20.50*** join/#asterisk bbryant (n=Brett_Br@adsl-068-016-200-248.sip.chs.bellsouth.net)
17:23.20carrancaHi, i have a weird scenario for you, i have an asterisk 1.4.22 with libpri and dahdi installed, i have a Digium TE205P with 2 Pri connected with a crossed cable (loopback) and the following configuration http://pastebin.com/m32269f20. As you can see i have 2 sip phones (402 and 403) and if i make a call with the prefix 201 I use one of the spans and with 202 the other span (continues)....
17:24.08*** join/#asterisk theHub (n=theHub@69.177.93.21)
17:24.12*** join/#asterisk qdk (n=qdk@79.138.230.80.bredband.3.dk)
17:24.19carrancaboth isdn have different context, if i call 2011234 i hear 1,2,3,4 and hangup... if i call 2021234 i hear the monkeys.... so far so good
17:25.12verywisemancan i use Grandstream  FXS/FXO ATA with asterisk?
17:25.37carrancabut if i call 201402 (i should be able to talk to my other SIP phone aka gateway) the phone rings but when i answer i get no sound
17:25.42lowtekverywiseman: Yes, but it will probably blow up or let notify child molestors of where your children go to school.
17:29.18murdock_utusing 1.6 and one step parking, I am unable to repark a call.  Any ideas?
17:32.28verywisemanlowtek, what is you mean?
17:33.16lowtekverywiseman: Nothing, lol, I just don't like Grandstream.
17:33.34verywisemanwhy?
17:34.08[TK]D-Fendercarranca: where is SIP/402 relative to your * server?
17:34.46NovceGuruhmm, so to have multiple lines with broadvoice...you have to have an account for each line
17:35.01[TK]D-FenderNovceGuru: Remember what I suggested : 2 stage porting
17:35.24verywisemanlowtek, why?
17:35.25carranca[TK]D-Fender, what do you mean? SIP/402 register itself like a regular SIP phone
17:35.26[TK]D-FenderNovceGuru: I never said you should jsut leave it in the first place that can grab it
17:35.40NovceGuruyeah, I called voicepulse about porting from braodvoice and they said no :\
17:35.50[TK]D-Fendercarranca: It is a NETWORKING question.  Describe the path between * and your phone
17:35.54NovceGuruSeems silly
17:36.00*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
17:36.03[TK]D-FenderNovceGuru: This is just so much BS...
17:36.06NovceGuruonce it's off the POTS it should be able to go anywhere? </noob>
17:36.18NovceGuruI know!
17:36.33[TK]D-FenderNovceGuru: This shouldn't be any kind of issue and they are just playing games with you...
17:37.02NovceGuruthey said since they dont have a rate center in my area even if broadvoice has my number there's nothing they can do
17:37.15[TK]D-FenderNovceGuru: IIRC number portability is supposed to be ensured... the enforced part is another matter however.
17:37.40[TK]D-FenderNovceGuru: telcos usually get away with being assholes in this regard
17:37.47NovceGuruyeah :\
17:37.56NovceGuruI've had issues with every port i've done
17:38.11NovceGurulast one, the telco keep reactivating the number
17:38.22[TK]D-FenderNovceGuru: FUN...
17:38.33NovceGuruyeah
17:38.51carranca[TK]D-Fender, well, i connect myself to the company switch with a physical SIP phone (Zultis). That is the SIP/402, my machine is also connected to the phone (so the phone is also a switch) and in there i have a SIP/403 softphone. Asterisk is in the server room connected to the company switch
17:38.52NovceGurubroadvoice accounts come with 2 channels so I *guess* I'd only need 2 accounts
17:39.10NovceGuruor I could keep the main number with the local telco and drop long distance and just use it for incoming
17:39.29[TK]D-Fendercarranca: And your other phone?
17:39.30NovceGuruand not go with 4 "unlimited" lines with time warner and STILL probably save money
17:39.57carranca[TK]D-Fender, thats it, i have two, a physical one (402) and a soft phone 403
17:40.04[TK]D-FenderNovceGuru: Assuming your telco doesn't try to rape BV as well
17:40.12[TK]D-Fendercarranca: Both local LAN?
17:40.21carranca[TK]D-Fender, yes
17:40.37[TK]D-FendercarraAnd the 2 can call each other just fine through *?
17:40.46NovceGuru[TK]D-Fender: how could they do that?
17:41.09carranca[TK]D-Fender, have you read my whole question? the problem is not connecting the phones themself, that was the easy part, the problem is passing throug dahdi in a loopback
17:41.10[TK]D-FenderNovceGuru: Same way you said they kept reactivating that other number
17:41.24thedonvaughnHey, is there a way to turn off hold music for queues and just have the caller listen to ringing?
17:41.38[TK]D-Fendercarranca: Yes, but I want to rule out that your endpoints don't have any possible issues themselves.
17:41.46[TK]D-Fendercarranca: you need to validate the whole path
17:41.53NovceGuruwell in theory we'd be "staying with" the local telco for the main number and fax, then having an account with voicepulse for rollover and long distance outbound
17:41.59[TK]D-Fendercarranca: And the enpoints are the first thing you prove
17:42.06carranca[TK]D-Fender, from the phone 403 if i call 402 its all good, but if i call 201402 the phone rings but i have no sound
17:42.56[TK]D-Fendercarranca: through your loopback, go use another function like voicemail and see if it records,e tc
17:42.59[TK]D-Fendercarranca: set up a small IVR and test DTMF, etc
17:43.30carranca[TK]D-Fender, ive tested a playback, thats 2011234 and it works correctly
17:43.37*** join/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk)
17:43.38carranca[TK]D-Fender, thats the weird part
17:43.47[TK]D-FendercarraNot jsut playback, recording <-
17:44.16carranca[TK]D-Fender, i will try, give me a min ;)
17:51.41*** join/#asterisk carranca (n=carranca@pampero.itba.edu.ar)
17:54.19*** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net)
17:56.37riddleboxI dont think there is a ITSP that will ever port my number
17:57.09carranca[TK]D-Fender, recording and playback worked ok
17:57.26lowtekriddlebox: Who have you tried?
17:58.13riddleboxlowtek: everyone in the ITSP list from jbot
17:58.31riddleboxI even tried vonage just to see and they wont port it either
17:58.56riddleboxflexpulse
17:59.18*** join/#asterisk rene- (n=renemend@200.34.66.137)
17:59.35rene-is there a way to increase the rx volume on an IAX trunk?
17:59.42rene-hello all
18:00.22carranca[TK]D-Fender, this is the new conf http://pastebin.com/ma7327bf
18:00.44lowtekriddlebox: I work for FlexPulse, we can usually port anything.  Odd.
18:01.59riddleboxI msg'd the number to you maybe you can check but the site says no
18:03.15*** join/#asterisk sasargen (n=chatzill@174-152-144-174.pools.spcsdns.net)
18:03.16hardwireUSAC sent us a bill for 2 cents
18:03.29lowtekriddlebox: Nope, we can't either.  Must be some off the beaten path clec.
18:03.30hardwirethey spent like $0.50 just to send us the bill.
18:03.40lowtekriddlebox: Who's your phone company?
18:03.41riddleboxit is a number from Charter Communications
18:03.52carranca[TK]D-Fender, also tried voicemail and it works fine
18:04.01lowtekDo you know what rate center your area code is in?
18:04.11riddleboxnope
18:04.48carranca[TK]D-Fender, i think it has something to do with redirection
18:05.06LemensTSHey is the guy who posted the flash phone in here from last night
18:05.07*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
18:05.20ShizuoFlash phone?
18:05.21ShizuoWhat?
18:05.38LemensTSYea someone posted a link in here last night of one they made.
18:05.50ShizuoWas it ok?
18:06.03LemensTSIt looked nice
18:06.20ShizuoNice...
18:06.34ShizuoDoes it work with my nice GS gateways?
18:06.36LemensTSHad a job for them if they wanted it.
18:06.45LemensTS~gs
18:06.46jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
18:07.00ShizuoGS works for me
18:07.04LemensTSheh sorry lol
18:07.05ShizuoNever had a problem
18:07.11ShizuoNever used it asterisk, though
18:07.27LemensTSIm not sure I never used gs gateways.
18:07.34ShizuoThey're nice
18:07.47ShizuoTheir hybrid circuit outperforms most Cisco stuff I have
18:07.57LemensTSWhats it do
18:08.18Shizuo"The hybrid" is the circuit responsible for avoiding echoes at the interface
18:08.36ShizuoWhen you listen to the line, you'll also listen to whatever you're injecting at it
18:08.51*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ee8e9f46d0ff5b07)
18:08.52*** mode/#asterisk [+o Deeewayne] by ChanServ
18:08.55ShizuoSo POTS handsets and equipments need to contain a "hybrid"
18:09.10LemensTSoh is it a big ata
18:09.29ShizuoIt subtracts outcoming speech from the incoming data
18:11.04*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
18:11.08ShizuoLemensTS: Most cisco stuff lack an hybrid
18:11.18ShizuoSo it needs to be performed in software
18:11.24ShizuoWhich is not nice
18:13.20*** part/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za)
18:14.20maddog01having problem with call files. i create a file move it to /var/spool/asterisk/outgoing and asterisk dosn't process it. any ideas??
18:16.29*** join/#asterisk Assid (n=kvirc@unaffiliated/assid)
18:16.42Alricmaddog01: Are you connected to the console when you move that file into the outgoing directory?
18:16.42Assidstupid machine
18:16.59maddog01no
18:17.15murdock_utmurdock
18:17.20maddog01not the asterisk console
18:17.31Alricmaddog01: Try that, I believe it will show an error/notice if its rejecting the call file for some reason.
18:17.42murdock_utwow, I can type my name.  I'm so proud of myself.
18:17.55maddog01thanks
18:18.56coppicealmost all modern hybrids are implemented in software, like a short echo canceller. even FXO/FXS chips like the ones used on the digium cards, which appear to implement that simple DSP in hardware and probably doing it in small embedded processor
18:20.00Assidwants to move his desktp to a mac.. and soon
18:21.04*** part/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk)
18:21.32Qwellmurdock_ut: hey, that's like 400 points on the SAT
18:21.45*** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com)
18:22.09murdock_utQwell: Hey I can use all of the help I can take.  Do you have any ideas on my one step parking issue with 1.6?
18:23.00Assidanyone wanna trade my core2quad 6600 - for a mac pro
18:24.13*** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com)
18:25.11*** join/#asterisk dieguito84 (n=diego@host118-190-dynamic.12-79-r.retail.telecomitalia.it)
18:25.24AlricAssid: How about a Mac Pro gutted and re-filled with PC components, running Vista? ;)
18:25.56Assidget away from me
18:26.04*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
18:26.11tzafrir_laptopmac pro can run Debian just fine. Though there are simpler alternatives
18:27.17Assidtzafrir_laptop: im just tired man.. really freaking tired.. it crashed on me right now.. i had soo many open browsers.. it just froze
18:27.30AlricAssid: You ever see the pics of the person that did that to a PowerMac when they first came out?
18:27.33Assidwas updating a cisco 7960 phne
18:27.43tzafrir_laptopAssid, it == ?
18:27.46AssidAlric: no.. and honestly i wouldnt want to
18:27.51Assidtzafrir_laptop: xp :|
18:28.12iEatChildrendoes [DID_span_1] specify where my calls go when they come in?
18:28.12Assidin the middle of someone helping me.. getting this phone to work
18:28.16Assidcan you imagine that..
18:28.19Assidembarassing
18:28.43AlricDid it save your tabs at least?
18:29.01AssidAlric: npe.. hung.. nothing moved
18:29.15AlricYuck
18:29.24Assidimagine if i dont save my codes everytime i edit them, or a db task
18:29.32Assidor even a CC transaction..
18:29.47*** join/#asterisk kotique (n=picachu@78.129.232.75)
18:29.48Assidsorry.. .im just annoyed
18:30.06kotiquehey guys. How do I use C - like stares cases ? -
18:30.11kotiquecase _XXX:
18:30.17kotiquecase _XXXX:
18:30.32kotiqueplayback();
18:30.38kotiqueI'm talking about AEL.
18:30.59*** join/#asterisk axisys (n=axisys@155.70.141.45)
18:31.10LemensTS<PROTECTED>
18:31.12*** join/#asterisk Assimilate (n=Assimila@72.22.242.66)
18:31.20kotique*stairs, not stares :)
18:31.39Corydon76-digLemensTS: probably -addons not installed
18:32.55*** join/#asterisk Assid (n=assid@unaffiliated/assid)
18:33.53maddog01Alric: i tried that it dosn't show anything in the console when i move the call file.
18:35.29[TK]D-Fendermaddog01: check the TIMESTAMP
18:35.46maddog01it's in the past i checked
18:35.59maddog01and i checked the permissions
18:36.14[TK]D-Fendermaddog01: remove the file, max out your debug and move the file back.
18:37.20*** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk)
18:37.47LemensTSCorydon: do i need to restart asterisk after installing asterisk-addons
18:38.11errrI think so
18:38.16[TK]D-FenderLemensTS: Should be able to load the module live
18:39.16LemensTSYep got it working now. thanks.
18:39.25*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
18:39.27*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
18:40.10*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
18:41.26iEatChildrencan anyone tell me whats wrong here? exten = _XXXX,1,Goto(sip/2002|2002|1)
18:41.32*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
18:41.36iEatChildrenkeeps telling me invalid extention 2002 in the CLI
18:42.11kannangoto,context,priority,data
18:42.13Qwellsip/2002 isn't a context
18:42.26kannansorry
18:42.29loatheryou probably want Dial(SIP/2002) instead
18:42.34iEatChildrenill try that
18:42.38kannangoto,context,exten,priority
18:42.46iEatChildrenwhat happened is a user deleted something via gui and our asterisk guy is out of town
18:42.48maddog01[TK]D-Fender: nothing again. I also check to see if the module is loaded and it dose show up when i use the show modules command
18:43.20[TK]D-Fendermaddog01: Can be permissions, bad file, etc.  Please pastebin EVERYTHING
18:43.31loatherThat exten line will also match any four-digit pattern, so if someone dials 2956 or 0987 it'll go to that extension as well
18:43.40iEatChildrenso something like exten = _XXXX,1,Dial(SIP/2002)  ?
18:43.44loather... which might not be what you want
18:44.11iEatChildrenits fine for now, im just putting on a band-aid fix untill i have time to redo the voice menu
18:44.23loatheryou also want => instead of =
18:44.44iEatChildrenok, ill try those things. thank you!
18:44.49loatherwelcome
18:45.00*** join/#asterisk lanning (n=lanning@66.151.128.195)
18:46.31*** join/#asterisk soa2ii (n=soa2ii@i59F56CE2.versanet.de)
18:47.18*** part/#asterisk Shizuo (i=pato@200-171-49-211.dsl.telesp.net.br)
18:47.34soa2iiHi there. I'm looking for people that are familiar with ISDN... maybe from germany so they coild tell me if this setup http://www.lagom.de/misc/setup.png would work (:
18:49.28soa2iiThe VersatelBox is this here: http://www.festnetz24.com/images/versatelboxkonfiguration.jpg combined isdn splitter with two rj11, telephone, fax and rj45 for your internet.
18:50.23soa2iiSo I simply want to plug my server into the second S0-Bus and make it a VoIP-ISDN-Gateway.
18:51.28iEatChildrenloather - how would i make that transfer to a voice menu instead of a sip ext?
18:51.34verywisemanhow connect analog phone to Rhino R24FXS PCI card?
18:53.33[TK]D-Fenderverywiseman: with an RJ-21 patch panel or break-out box
18:53.47[TK]D-Fenderverywiseman: And that is a fugly solution for FXS.
18:54.31iEatChildrennevermind loather...i got it woot woot!
18:54.37iEatChildrenthanks again for your help earlier though
19:06.20LemensTSall you can do with the MYSQL cmd in the dialplan is query? no insert?
19:06.55[TK]D-FenderLemensTS: Yes, you can
19:07.01murdock_ut[TK]D-Fender: Do you use one step parking?
19:07.19[TK]D-Fendermurdock_ut: Nope.
19:07.32LemensTSeven in 1.2
19:08.32*** join/#asterisk ta^3 (n=tacvbo@189.146.186.167)
19:09.49[TK]D-FenderLemensTS: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
19:11.33*** part/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net)
19:11.43*** join/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net)
19:12.28LemensTSTK: yea thats what i was looking at, i only seen queries. then i get this in my cli:
19:12.29LemensTS-- Executing MYSQL("SIP/12345678-099e84d8", "INSERT questions SET questionnum=1, pollid=22") in new stack
19:12.30*** join/#asterisk mchou (n=mchou@unaffiliated/mchou)
19:12.38LemensTSJan  7 14:09:58 WARNING[10338]: app_addon_sql_mysql.c:411 MYSQL_exec: Unknown argument to MYSQL application : INSERT questions SET questionnum=1, pollid=22
19:12.56LemensTSI run that cmd in mysql just fine.
19:13.47lowtekLemensTS: post your MySQL() line from your dialplan
19:14.55LemensTSexten => s,n,MYSQL(INSERT questions SET questionnum=${questionnum}\, pollid=${pollid})
19:15.05LemensTSmaybe i dont have enough \  in there
19:15.07lowtekLemmensTS: You're missing some stuff.
19:15.57lowtekLemensTS: Should be somethign like: MySQL(Query resultid ${connid} INSERT\ questions\ SET\ questionnum=${questionnum}\, pollid=${pollid})
19:16.36lowtekLemensTS: afaik you have to specify "Query" follwed by the resultid (even if there isn't a result) and the connection id you've opened in a previous MySQL() "connect" statement.
19:17.57[TK]D-FenderLemensTS: INSERT into table set field=value
19:18.09*** join/#asterisk reneger (n=reneger@dslb-088-078-123-090.pools.arcor-ip.net)
19:18.15[TK]D-FenderLemensTS: improper base SQL syntax and improper escaping and MySQL() app syntax
19:18.22[TK]D-FenderLemensTS: Go read that page again
19:19.53sasargendoes anyone know how to configure a Mitel 5302 ip phone to register with asterisk?
19:22.55LemensTSThanks that worked. Confused myself. One last question, in php after i insert a reccord that has an auto_incrementing id, i can use mysql_insert_id() to find what that id was that it last inserted. Do you know of a way of doing this in the dial plan after i insert it?
19:23.29lowtekLemensTS: insert the uniqueid channel variable and use that.
19:24.36maddog01trying to get call files working. I move the file to the outgoing folder and nothing happens. I have put a bunch of folder dumps / CLI info here (http://pastebin.com/d147ab9a3) can someone help me.
19:25.49lowtekmaddog01: check your permissions on the file created, change them to something asterisk has permissions to r+w+x to before copying.
19:25.53*** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net)
19:25.56lowtekmaddog01: er.. moving, rather
19:26.49*** join/#asterisk StephenF (n=none@198.144.201.106)
19:26.55maddog01i checked permissions there set to read asterisk:asterisk
19:27.19lowtekmaddog01: Asterisk has to have permission to remove the file after it parses it.
19:27.26StephenFIM quoting a small office (16 phone, 5 POTS line) asterisk system what is everyone's favorite server hardware for this type of confg
19:27.27jayteefile: are you here?
19:28.13rob0Steve, never skimp on hardware.
19:28.48[TK]D-Fendermaddog01: * needs to be able to WRITE.  Give ti FULL perms
19:28.57maddog01k
19:29.06StephenFThats my point what works well with asterisk, Dell, HP, etc...
19:29.31Corydon76-digIt's lmadsen's birthday, today.  Everybody wish him a happy birthday!
19:29.46rob0Name brands? Uh ... I don't buy those. I build my own.
19:30.01StephenFAlso they currenty have what looks like an integrated T1 with voice and data. There is a box that splits the T1 into data and simulated POTS lines. What is that called? Would that be a PRI?
19:30.05jayteeI've got Asterisk running on a Dell PowerEdge 2950 with 4GB of RAM using a Digium TE212P dual port T1 card and 53 sip phones. it never breaks a sweat.
19:30.16rob0Assemblers like Dell & HP are going to cut corners on things.
19:30.37filejaytee: moo
19:30.38jayteeStephenF, the "box" is called a T1 add/drop mux
19:30.43StephenFahh ok
19:30.45jayteefile, quickie question
19:31.04jayteeI've got my main grammar file here: http://www.pastebin.ca/1302924
19:31.46jayteewhen I use speech the response is quick, when I use dtmf there is about a 4 to 5 second lag before the SpeechBackground app executes the option
19:32.05StephenFso if the customer already has that mux, I would need to spec an analog digium card like the AEX808E
19:32.21maddog01[TK]D-Fender: asterisk dose have write permissions
19:32.23StephenFrob0 I see your point
19:32.46filejaytee: set the SPEECH_DTMF_MAXLEN dialplan variable to the maximum length of DTMF you want and it'll return immediately once it reaches it
19:33.07rob0Note, I've never bought their server-grade hardware, so it might be worth more $$$ to save some time.
19:33.39StephenFrob0, thats kinda what I'm thinking. Plus the warranty support is helpful to maintain the device
19:33.48jayteefile, maximum length of DTMF? is that set in the grammar or the lumenvox.conf file? I'm confused.
19:33.55filejaytee: dialplan
19:34.12jayteeah, it's a system global var?
19:34.15filejaytee: Set(SPEECH_DTMF_MAXLEN=2) before calling SpeechBackground, type in 2 DTMF digits and boom it'll return immediately
19:34.23fileit's a dialplan variable
19:34.24filenothing special
19:34.31rob0well, if you need warranty service, you (your customer) is screwed. They can't do anything with a Linux system.
19:34.48jayteefile, cool, I'll give that a shot. thanks!!!
19:35.19StephenFjaytee, So with that MUX in place I would need to spec an analog telephony card such as the AEX808E correct?
19:36.03[TK]D-FenderStephenF: Or jsut get a digital card and replace their MUX & data hardware and connect directly to the smartjack
19:37.55StephenF[TK]D-Fender right, Im not sure their provider would be happy with that. But anyway if I did what would their data network connect to then? If the smartjack connected to the digital tel card, where would the data connect?
19:38.15[TK]D-FenderStephenF: Your * box
19:38.18StephenFDo I use a 2 port card, one for smartjack connectivity, and the other for data network
19:38.21StephenFahh
19:38.27[TK]D-FenderStephenF: UNIX has a long history of networking you know...
19:38.30StephenFlol
19:38.50StephenFok, so I just need a 1 port card, and then * box connects to data network over standard NIC
19:38.57StephenFduh
19:38.58eppigyUNIX for networking?
19:39.03eppigyprepostrous
19:39.18StephenFwho knew?!
19:39.33StephenFso in a sense I would be replacing the Mux with an * box
19:40.04StephenFvery cool
19:40.54jaytee[TK]D-Fender, so with your proposal you're having him use a T1 card in * and zaptel to do the demux by defining the b channels for voice in one group and the data group as another?
19:42.17[TK]D-Fenderjaytee: Individual timeslots, yes.  However I'm pretty sure it isn't PRI, just FXSLS & data
19:42.40[TK]D-Fenderjaytee: Whicht he card can bod off as an ethernet device and route accordingly.
19:42.51[TK]D-Fenderbond*
19:43.11jayteeyeah, that's kinda where I figured you were going with that. smart choice and efficient as hell
19:43.38[TK]D-Fenderjaytee: eggs in a basket as well but yes, efficient
19:43.56[TK]D-Fenderjaytee: jaytee And better o the telephony side.
19:44.18jayteedon't know what brand of equipment their using but the Kentrox Add/Drop muxes are pricey as hell compared to a 2 port digium card.
19:44.42StephenFMux is existing, provider is cbeyond
19:45.10StephenFthough once their contract is up they will be dropping cbeyond
19:45.47StephenFSo i think I will recommend the analog card since once they drop cbeyond they will be using us as their data provider and I doubt will want to pay for an PRI when they are only using 5 lines
19:46.17[TK]D-FenderStephenF: partial PRI w/ data is a great option.
19:46.32[TK]D-FenderStephenF: rAnalog sucks
19:46.43[TK]D-FenderStephenF: Mixing voice & data is cost-effective
19:47.06StephenFdefinetly, but this is kind of a weird situation. We are an ISP and they are moving into our building. So we can provide data with no local loop. Just a cat5 down to their suite
19:47.39[TK]D-FenderStephenF: If you're an ISP why not just go for an ITSP?
19:47.51*** join/#asterisk Greek-Boy (n=greek@41.222.89.77)
19:48.07StephenFIm contemplating that as well. We could do that
19:48.17StephenFJust not sure we are ready to take that on
19:48.51[TK]D-FenderStephenF: Levy againsth the cost of any card.  then again against the woes of analog
19:49.10timeshellI found this useful to make oslec work with dahdi:   http://www.freepbx.org/forum/freepbx/installation/asterisk-1-6-dahdi-oslec
19:49.20jayteefile, worked like a champ!!!! no more lag. where do i find this stuff documented? I've been using sample apps from lumenvox and just trying to make things work.
19:49.49StephenFI've never done an analog * box before, is configuration that much worse? Or are you talking about quality of analog lines?
19:49.58LemensTSlowtek: this works to find the last auto_incrementing id - exten => s,n,MYSQL(Query resultid ${connid} SELECT\ last_insert_id())
19:50.19*** join/#asterisk cvnet (n=dahitler@24.156.136.205)
19:50.26cvnethello
19:50.43lowtekLemensTS: Assuming that another record from another call isn't inserted between the time you do your insert and the query.
19:50.45*** join/#asterisk adminguru (n=atze@p57BD7B49.dip.t-dialin.net)
19:50.47[TK]D-FenderStephenF: Qualify, hassle, call setup delay, lack of DID functioning, line contention, etc.
19:50.50lowtekLemensTS: So it won't *really* work.
19:52.15StephenF[TK]D-Fender so what kind of ITSP setup would work best?
19:52.27*** join/#asterisk km2 (n=x@68.161.155.223)
19:52.27lowtekLemensTS: Unless, of course, it's specific to that connection, in which case it might work.  I wouldn't trust it, just insert the uniqueid and you won't even have to make another query.
19:52.37StephenFhosting the *, providing a sip connection to our *, or something even more complex?
19:53.15StephenFor just have an extablished ITSP provide their PSTN termination?
19:53.39*** join/#asterisk alrs (n=lars@cpe-76-174-43-54.socal.res.rr.com)
19:53.41LemensTSlowtek: as soon as it inserts, the next line is checks for last_insert_id(). And i think it is doing it based on the current mysql connection.
19:54.04LemensTSill have to test and see if i can screw it up.
19:54.36lowtekLemensTS: Well you can always fall back to inserting ${UNIQUEID} if it doesn't, good luck!
19:54.57[TK]D-FenderStephenF: Well this is for YOU, right?
19:55.07jayteefile, never mind that last question. I just finally found the motherlode on lumevox's site. couldn't find much 8 months ago.
19:55.18StephenF[TK]D-Fender what do you mean for ME?
19:55.46Greek-Boylol
19:55.56[TK]D-FenderStephenF: nvm.  Just pick an ITSP and run your own * box locally.
19:55.59LemensTSYea I will keep that in mind, i should be able to do that too. Only reason i want to use the table id is if they end up doing this thru php in a browser, then i wont be able to ${uniqueid} but i can reference the last id still.
19:56.03StephenFwe have our own * box, and we are using another ITSP for PSTN termination
19:56.38StephenFyou said if we were an ISP, why not gor for an ITSP. Did you mean set-up this customer with an ITSP?
19:56.44*** join/#asterisk maddog01 (n=minotaur@d221-91-175.commercial.cgocable.net)
19:57.00[TK]D-FenderStephenF: First you're talking analog off a T1 voice circuit, then PRI, not you already have an ITSP.  Please clarify this mess
19:57.12StephenFlol ok
19:58.00StephenFI have a customer that currently has a T1 add/drop mux with 5 voice channels and the rest for data
19:58.03StephenFwe are specing an * box for them
19:58.26StephenFThey will be dropping their current T1 provider when the contract is up, and then using us for data connectivity
19:59.02StephenF(we are an ISP, they are moving into our building. This allows us to provide data connectivity without a local loop fee)
19:59.38StephenFSo I was saying I dont think I would want to replace the Mux with an * box, because once they change providers they will most likely just order 5 POTS lines
20:00.21StephenFUnrelated to this customer, we also have our own * box that uses an ITSP for PSTN termination
20:00.26StephenFand thats the story of the day
20:00.33*** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net)
20:00.47[TK]D-FenderStephenF: if your data is separate then no, PRI isn't probably a great way to go.
20:00.59StephenFyea prbly too expensive right
20:01.13[TK]D-FenderStephenF: Have they considered going right over to VoIP?
20:01.53StephenFthat is an option but I really want this to be rock solid and I'm a little weary of doing that
20:02.09iratikI'm getting "funny" dtmf issues (like not recognizing digits pressed accurately) when playing an audio file and waiting for input from the remote extension .... this is on outbound calls where the remote party being called is entering digits ....  the docs seem to indicate that relaxdtmf is a primarily a setting for tweaking problems with SIP, however this is on a PRI on a digium T1 card ... any ideas?
20:02.16*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
20:02.37[TK]D-Fenderiratik: it is primarily for Zap, not SIP
20:02.53[TK]D-Fenderiratik: and IS the first option to look at.  Then look at your gains
20:03.19*** join/#asterisk cuco (n=diego@DSL217-132-187-5.bb.netvision.net.il)
20:03.46iratikso if i'm getting inaccurate results ... go ahead and set relaxdtmf=1 ?
20:03.52cucotzafrir_laptop, ping
20:04.17[TK]D-Fenderiratik: relaxdtmf=yes
20:04.41[TK]D-Fenderiratik: Don't go using numbers for these kinds of fields.  Sets a bad precedent.
20:04.42StephenF[TK]D-Fender So i shall brave the woes of Analog
20:04.52[TK]D-FenderStephenF :why not...
20:04.56iratiksounds like relaxdtmf .. would make dtmf trigger more easily ... but shouldn't it be more strict?
20:05.10StephenFSounds good to me, we'll get it done
20:05.26StephenFThx for listening to the noob by the way ;)
20:05.40tzafrir_laptopcuco, hi
20:06.35cucotzafrir_laptop, hi, i have a nice game for you. wanna play? :)
20:07.39tzafrir_laptopdepends how nice it is
20:07.39cucotzafrir_laptop, pmsg?
20:08.23*** join/#asterisk af_ (n=getsmart@88-149-240-130.dynamic.ngi.it)
20:09.50*** join/#asterisk denon (i=denon@synapse.subneural.net)
20:09.50*** mode/#asterisk [+o denon] by ChanServ
20:12.38[TK]D-Fenderiratik: does RELAX sound STRICT to you?
20:12.57iratikno ... thats the point
20:13.09iratikwell.. if it is getting inaccurate results...  it needs to be more strict about dtmf tones... not more releaxed
20:14.26verywiseman[TK]D-Fender, are you prefer "Grandstream HandyTone HT-286 ATA" for using with Asterisk?
20:14.27eppigyRELAXEDFITDTMF
20:15.10eppigyiratik: usually the results are innacyrate because it is not interpreting certain keypresses as valid dtmf tones
20:15.15[TK]D-Fenderverywiseman: No.
20:15.16eppigytherefore you want its interpreation
20:15.20eppigyto be relxed
20:15.24eppigyrelaxed
20:15.38verywiseman[TK]D-Fender, can u give me reasons ,pls?
20:15.50iratikwell.. its interpreting them either inaccurately or not at all ... let me try it
20:15.52[TK]D-Fenderverywiseman: History of being flakey shit.
20:16.13eppigyiratik: well there you go
20:16.18eppigywhat i said verbatim
20:17.05StephenFany problems with asterisk and AMD processors?
20:17.38verywiseman[TK]D-Fender, what about sangoma PCI card?
20:18.03[TK]D-Fenderverywiseman: before you go running around like a headless chicken, describe your precise NEEDS.
20:19.14iratikWhen using MixMonitor, then DTMF tones the caller pressed aren't heard in the recordings... is there anyway to make it so you can hear them?
20:22.30*** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr)
20:22.55verywiseman[TK]D-Fender, i want to find suitable hardware for SOHO,medium business and large business
20:24.05seb-can Skype uses talk to Asterisk SIP servers? ...or are they locked out?
20:24.07[TK]D-Fenderverywiseman: perhaps you do not understand the word PRECISE
20:24.13seb-users*
20:24.20[TK]D-Fenderseb-: 3rd party only stuff now, all sucky
20:24.24jjshoeseb- closed beta going on right now for a connector
20:24.25eppigyTRABAJO
20:24.37verywiseman[TK]D-Fender, maybe , sory E is not my native language
20:24.48jjshoeskype had their own pbx on their website, for a few weeks, dunno whatever happened to it
20:24.56[TK]D-Fenderverywiseman: DETAILS
20:24.58seb-jjshoe: what is cheapest/easiest way for people to connect to Asterisk? I was guessing Skype
20:25.04Assidokay so i managed to get  a cisco 7960 on SIP, however i dont see it registering to asterisk.. sip debug doesnt show me a connection.. anyone here played with cisco+sip?
20:25.11eppigyseb-: xlite
20:25.36jjshoeseb- what eppigy said
20:25.37jjshoehttp://www.prettymay.net/index.htm
20:25.48seb-eppigy: i believe that is a Windows app?  then they just need to buy a plantronics headset and pray the docs are clear
20:26.03khronos<PROTECTED>
20:26.06eppigythey have windows and *nix
20:26.45StephenFwhat is everyone's opinion of Switchvox? any good?
20:26.49seb-eppigy: i'm doing an online calculus class..i hope the little students can figure out xlite set up then..thanks
20:27.03jjshoeStephenF why not download the free version and try it?
20:27.11StephenFI mean the hardware
20:27.13jjshoeseb- it's fairly straight forward
20:27.28jjshoeseb- you could of course ultimatly build an installer for it if you truly cared :P
20:27.33[TK]D-FenderStephenF: Its a PC.  What is there to say?
20:27.47StephenFlol ok, has anyone used it?
20:27.47seb-jjshoe: in that case you've made my day!! buying a phone # from Teliax for conf calls was *butt* expensive
20:28.03iratikWhat is relaxdtmf set to by default?
20:28.17StephenFnvm way to expensive
20:28.33seb-jjshoe: yes! i could make a custom xlite that automatically connected to my SIP address! brilliant!
20:28.42jjshoeStephenF yeah, especially for a buisness phone system, cheap is always better!
20:29.01StephenFno but i can build a better dell box for half the price...
20:29.04jjshoeseb- or use one of many auto programs to install configs for it, or automate filling out, like autohotkey
20:29.14[TK]D-Fenderok, leaving early to go home.  Later all.
20:29.19StephenFCya
20:29.28jjshoeStephenF they provide you with more then just hardware.
20:30.21verywiseman[TK]D-Fender, i am now need fxs/fxo equipment from 1-250 fxs port and 1-48 fxo port , and there are 2 options (PCI card and ATA), and when i ask in this channel , somebody prefer ATA for small business
20:30.27eppigyseb-: if they hope to comprehend calculus im sure they will have no trouble with xlite setup
20:32.11*** join/#asterisk jtodd (n=jtodd@nat/digium/x-970ebba76f676edd)
20:32.30seb-XLite is not open source...is there a quality open source SIP client?
20:33.05*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
20:33.22LemensTSif i have lots of s,n, lines, how can i send a GoTo(s,n) to a particular one?
20:33.29LemensTSive always wondered that...
20:33.49Alricn ends up translating to a number.
20:34.04LemensTSoh reall?y so if its the 5th n, i can do GoTo(s,5) ?
20:34.07AlricJust manually count it out, or you can probably use labels or something.
20:34.21LemensTSawesome
20:34.28jjshoeseb- what does open source matter? where you hoping to modify the code?
20:35.30mogseb-, there is ekiga
20:35.34AlricLemensTS: I haven't bothered with labels, so I couldn't say if that really works, I've used the manual counting method in my Gotos.
20:35.35mogwhich is awesome
20:36.29LemensTSyea labels would help incase you insert a line in the middle of it
20:36.50LemensTSthink they are s,n(labelhere),
20:37.35*** join/#asterisk iceyp (n=icepick@60.234.68.250)
20:37.38jjshoegoto and labels.... awwwwwwwww. so cute!
20:37.51iceyphey guys, does anyone know of a good free web based sip client that I can host on my web page
20:38.06iceypI don't want a remote hosted product
20:38.25LemensTSjjshoe: yea...
20:38.55*** join/#asterisk kerx (n=kerx@adsl-69-104-78-246.dsl.irvnca.pacbell.net)
20:39.18kerxHi, is there any way to have SIPAddHeader() when you make a originate call from AMI?
20:42.21*** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net)
20:44.52Assidi dropped my polycom handset once too many times.. it now has a screw loose inside or something
20:45.34*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com)
20:50.00kerxdamn :-(
20:50.08kerxPolycom handset's are expensive
20:50.21kerxYou have warranty on it?
20:50.36*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
20:50.46Assidnope :(
20:50.53flujanhello guys, i am creating a simple agi script to give it a try
20:51.09kerxTo give what a try?
20:51.10flujanit needs to write in the stderr, so it will be redirected to asterisk
20:51.13flujanonly it
20:51.15flujana hello world
20:51.19kerxOh, cool
20:51.21flujanbut i am not having success
20:51.39*** join/#asterisk awk_r (n=awk_r@nat/digium/x-b96d8a553644a429)
20:53.04*** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net)
20:54.13*** join/#asterisk sekil (n=sekil@80.93.247.26)
20:57.03*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
20:57.06flujanhttp://pastie.org/354993
20:57.10flujanhere goes the error message
21:02.06*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
21:02.49iceypAnyone know of a web based sip client I can use on my webpage?
21:04.16lowtekiceyp: It would have to be java based or flash based so you could use sockets and access the computers mic.  Probably better off just using a soft-phone.
21:04.39iceypI'm looking for something to allow people to call me direct from our website
21:04.46iceypnot to make inbound / outbound calls
21:04.56iceypjust to dedicated numbers
21:05.03lowtekiceyp: Oh, so you want the ability to originate a call from a web page?  Use php + an AMI call.
21:05.25iceypumm, yeh possibly?
21:05.36jjshoeiceyp many of us do php + ami
21:05.46jjshoe"Enter your number and we'll call you!"
21:06.07iceypi.e. people go to our contact us page, and click "Contact us now via VoIP" then when clicked, it enables their mic and speakers and they can talk to us
21:06.11lowtekiceyp: Yep, it's actually quite easy.  PM me your email address and I'll give you a really good AMI wrapper I wrote a while back.
21:06.50*** join/#asterisk jasonwoot (n=some@bookit-dev.com)
21:06.55jasonwootwoot
21:06.57iceypbrb ciggy
21:07.05lowtekiceyp: That's back to what I originally said.  It would have to be java or flashed based and wouldn't work very well due to default security settings that prevent said practices on end-users computers.
21:07.38lowtekiceyp: Better off just initiating regular phone call.
21:08.04*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
21:08.08AlricDoes the average user even have a mic on their computer statistically?
21:08.21lowtekAlric: I wouldn't think so, laptops maybe.
21:08.22[TK]D-FenderAlric: No.
21:08.37AlricThat would have been my guess.
21:09.03lowtekiceyp: The problem taht will be the show stopper is that Java (not javascript) and Flash can't access the computer's resourcs or TCP/IP sockets without special permissions.
21:10.05aiksa[LV]lowtek: all of ar still waiting for pac.ifica ?
21:10.16aiksa[LV]all of us are *
21:11.45iceyplowtek damn, it's just for family use, people that dont know how to install sip clients ;/
21:11.54lowtekiceyp: skype?
21:12.03iceyplowtek can you still email me your wrapper, would be good to take a look
21:12.12iceyplowtek can skype dial sip:// urls?
21:13.00jjshoeiceyp no.
21:13.01aiksa[LV]iceyp: if only it could
21:13.03aiksa[LV]:P
21:13.24aiksa[LV]but there was a word a month or so ago that digium might have made deal with the devil
21:13.46[TK]D-Fendericeyp: Zoiper can be distributed pre-configured in a drop-in install mode
21:14.05*** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4)
21:14.08[TK]D-Fendericeyp: And supports IAX which can help in some cases
21:14.18lowtekiceyp: The short answer is that you're not going to be able to do browser based sip calls.  Go to plan B, a softphone or maybe just an IM client if it's in-family stuff.  Or Skype.
21:14.21iceyp[TK]D-Fender thanks let me take a look
21:15.30iceyp[TK]D-Fender the free version allows a pre-configuration ?
21:15.47aiksa[LV]iceyp: actually there is a config xml file
21:15.54[TK]D-Fendericeyp: Don't know all the fine points.  Go look
21:16.03aiksa[LV]which contains everything that interface does and a little bit more
21:16.29aiksa[LV]the launch of the binary doesnt require the installation
21:16.43*** join/#asterisk grantm (n=grant@68.142.138.4)
21:16.56Kattywoah
21:17.01aiksa[LV]but user will be prompted in windows to allow this application to access the network by default inbuilt windows firewall
21:17.01iceypmmmm
21:17.03Katty[TK]D-Fender: sodium acetate sculptures
21:17.25iceypaiksa[LV] you talk of zoiper?
21:17.31lowtekiceyp: In my family, I'd rather just pick up the cost vs trying to help them configure *anything* on their f*cked up computers and home networks.  I just host a conference bridge and a toll-free for everyone to call in on.
21:17.34aiksa[LV]iEatChildren: yes
21:17.55aiksa[LV]iceyp: yes
21:18.08iEatChildrenty
21:18.14iceyplowtek I have this in a number of countries though not all have toll free's, just trying to do the extra step :)
21:18.16carrancahow can i hardcode dahdi to always make calls in ulaw?
21:18.35aiksa[LV]iceyp: additionaly - there was one russian comapy which did develop a prototype flash sip phone. flashphone.ru or something.
21:19.06iceypaiksa[LV] so the xml is hosted on a web page, and the user just needs to input the XML URL?
21:19.10[TK]D-FenderKatty: Sodium tri-iodide > sodium acetate
21:19.43aiksa[LV]carranca: in /etc/zaptel.conf (or its dahdi.conf now): add this line alaw=25-86
21:19.45[TK]D-Fenderis Iron Chef of the Anarchist's Cookbook
21:19.57aiksa[LV]in your case that would be ulaw=chansfrom-chansto
21:20.23aiksa[LV]iceyp: no you just leave the ziped folder with all the files of an installed zoiper
21:20.32aiksa[LV]and the config file
21:20.43*** join/#asterisk DarkRift (n=dark@65.92.169.136)
21:20.47Katty[TK]D-Fender: oh? does that also make dry ice?
21:21.00iceypahh and the user downloads the zip and then runs the binary out the zip
21:21.07[TK]D-FenderKatty: "Not Quite"
21:21.12aiksa[LV]back to a timing hacking
21:21.20Katty[TK]D-Fender: what's that do then?
21:21.22carrancaaiksa[LV], you mean /etc/dahdi/system.conf or /etc/asterisk/chan_dahdi.conf?
21:21.32[TK]D-FenderKatty: GOOGLE :)
21:21.35Kattyk
21:22.01aiksa[LV]carranca: the file which a while ago was named /etc/zaptel.conf  not sure of the naming for the dahdi
21:22.14aiksa[LV]carranca: the one driver not asterisk related
21:22.45Katty[TK]D-Fender: is this just regular iodine?
21:23.03carrancaaiksa[LV], ok :), its now /etc/dahdi/system.conf
21:23.15Katty[TK]D-Fender: not much documentation--wikipedia is not big on the practical application
21:23.21[TK]D-FenderKatty: No, a sediment-based contact explosive :)
21:23.22aiksa[LV]btw, [TK]D-Fender if a span in zttool is shown as "internally clocked" although has timing source priority of "1". Does that mean that telco doesnt provide the timing on this digital line at all?
21:23.33Katty[TK]D-Fender: ah, i see.
21:24.02[TK]D-Fenderaiksa[LV]: I don't know zttool really
21:24.04aiksa[LV]sorry for HL you, but i thought you would probably now
21:24.37aiksa[LV][TK]D-Fender: I have a suspicion that i dont get the timing from an E1 line and I am not sure how to check that
21:24.39[TK]D-Fenderaiksa[LV]: pastebin your config
21:24.53aiksa[LV]a sec.
21:24.55aiksa[LV]~pb
21:24.56jbot[~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, or , http://bin.cakephp.org/
21:24.59SargunCan someone translate the SIP tos_(audio/video/sip) fields in sip.con?
21:25.03SargunCan someone translate the SIP tos_(audio/video/sip) fields in sip.conf?
21:25.30FinboySlickSargun: To which language?
21:27.14aiksa[LV][TK]D-Fender: http://pastebin.com/m4ec5db9e
21:27.45aiksa[LV]iceyp: exactly
21:27.47SargunFinboySlick, nevermind, I found a document elsewhere.
21:28.00[TK]D-Fenderaiksa[LV]: Indeed you are not taking timing from ANY of your ports
21:29.06[TK]D-Fenderaiksa[LV]: span = PORT,TIMING,LBO.  0 = ACT as timing source.  Bad for use with telcos
21:29.27[TK]D-Fenderaiksa[LV]: 1= act as primary, 2,3,4, = secondary, tertiary, etc
21:29.48[TK]D-Fenderaiksa[LV]: If all to telco : span=1,0,0,ccs,ami should be span=1,1,0,ccs,ami
21:29.55aiksa[LV][TK]D-Fender: i want to take timing from that dynamic statement there
21:29.56[TK]D-Fenderaiksa[LV]: span=2,2,0,ccs,ami
21:30.07[TK]D-Fenderaiksa[LV]: as I've now shown you
21:30.20*** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr)
21:30.23aiksa[LV][TK]D-Fender: all of the non dynamic spans ar not connected to telco but to a legacy pbx
21:30.32aiksa[LV]#
21:30.33aiksa[LV]dynamic=ethmf,eth1/00:50:c2:65:d3:92/0,31,1
21:30.33aiksa[LV]#
21:30.33aiksa[LV]dynamic=ethmf,eth1/00:50:c2:65:d3:92/1,31,2
21:30.45[TK]D-Fenderaiksa[LV]: OH FFS, TDMEO
21:30.51[TK]D-FenderTDMOE
21:30.54[TK]D-Fenderruns
21:31.43*** join/#asterisk shinao1 (n=shinao1@62.173.48.42)
21:31.49aiksa[LV][TK]D-Fender: like run and hide?
21:31.58aiksa[LV]or like going to bed finally?
21:32.13aiksa[LV]explain: FFS ?
21:32.48shinao1hi all... has anyone ever tried using the Cisco 3911 "SIP" phone with asterisk?
21:33.15[TK]D-Fender~ffs
21:33.16jbotffs is probably for f**k's sake, or for fine's sake.  UCB's Fast File System
21:33.39aiksa[LV]i suspect the first ..
21:34.19*** join/#asterisk Trionnis (n=rboggs@253.ptr.ifbyphone.com)
21:35.47iratikWhen using MixMonitor, then DTMF tones the caller pressed aren't heard in the recordings... is there anyway to make it so you can hear them?
21:36.26aiksa[LV][TK]D-Fender: nevertheless is there a way to check if a span is being used as a timing source
21:36.30aiksa[LV]?
21:36.57[TK]D-Fenderaiksa[LV]: None of your fixed ones are, and I am not sure how TDMoE play into this
21:38.53aiksa[LV][TK]D-Fender: fixed cannot be used as timing sources as they are not connected to telco
21:39.01aiksa[LV]only TDMoE is
21:39.16aiksa[LV]and according to documentation of ztd_ethmf
21:39.39[TK]D-Fenderaiksa[LV]: As Isaid I have little to advise in configuring TDMoE.
21:39.39aiksa[LV]the last parameter in the "dynamic=" line refërs to timing priority
21:39.55aiksa[LV][TK]D-Fender: ok. wont bother you then
21:40.21aiksa[LV]I just thought that there could be a tool which would show which span is used currently as timing source
21:41.23*** join/#asterisk jtodd (n=jtodd@nat/digium/x-3f01ceaf57ff08af)
21:43.12Trionnisis anyone aware of a functional (with 1.4.21.2) backport of the bridging abilities?  either app_bridge, or the built-in functions from 1.6?
21:46.18Trionnisthe silence is deafening... ;-)
21:46.31[TK]D-FenderTrionnis: From what I remember having been mentioned about how they did it in 1.6 it took some major core changes the likes of which I don't believe couldbe back-ported....
21:47.01Trionnisyeah, I kinda figured the 1.6 "version" would be a no-go
21:47.08[TK]D-FenderTrionnis: And yes, you wated a max of 3 minutes for an answer to an oddball question.  Your patience is nothing less than ASTOUNDING
21:47.25Trionnisit is kinda saddening to see that the patch that enabled it back in 2005 didn't make it into 1.4 at any point
21:48.13[TK]D-FenderTrionnis: so what stops you from going to 1.6?
21:48.34SargunQuestion: Does the RTP stream's source IP have to be the same as the SIP IP?
21:48.38Trionnisthat it took 18 point releases of 1.4 to be considerable as production quality :)
21:49.04jjshoeTrionnis touche
21:49.28aiksa[LV]there is another "I want" feature of 1.6, I would really (I mean - REALLY) like to have in 1.4
21:49.38aiksa[LV]Transfer event in AMI
21:50.01aiksa[LV]Trionnis: i suppose it could go in same changes bag with app_bridge
21:50.39[TK]D-FenderSargun : Of course not.
21:50.42aiksa[LV]as the event model is totally redone as regards the old "Link" events.
21:51.15*** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net)
21:52.21Sargun[TK]D-Fender, I assume in the SIP stream, the source is hinted at.
21:52.58[TK]D-FenderSargun : the entire point of SIP is to setup RTP and and handle call flow
21:54.05jasonwootI thought the entire point of SIP was to ruin my life
21:54.29[TK]D-Fenderjasonwoot: no, thats in the "Other Cool Features" column
21:54.31aiksa[LV]jasonwoot: it's his alter ego
21:54.37Trionnisdunno about ruining my life, but it's certainly caused its fair share of migraines
21:55.07*** part/#asterisk adminguru (n=atze@p57BD7B49.dip.t-dialin.net)
21:55.38Sargun[TK]D-Fender, Well, I was wondering if I can do layer 3 routing for SIP.
21:55.42aiksa[LV]Anyone fiddling with Sip is  much like Oracle admins - they dont need wife to get guaranteed sexual life.
21:55.59Sargun[TK]D-Fender, Err, I mean RTP. So, I can send RTP via a different source IP.
21:56.17SQLDarklyMy users keep getting dropped from my conferences.... I have looked at messages but I see nothing at the time of the drop. What could cause this? Its happened 4 times in 2 days
21:58.18[TK]D-FenderSargun : Might be an idea... if it wasn't Layer 4
21:58.52Sargun[TK]D-Fender, Well, I can set the ToS to SIP to N, and then route everything with the ToS value of N through line 2.
22:00.04[TK]D-FenderSQLDarkly: I'm sorry, could you be a little more vague please?
22:00.15SQLDarklyI can if you like ;)
22:00.50SQLDarklyMy apologies. I am runnign 1.4.22 on 2 boxes. I have RT Arch setup and Meetme is pulling its rooms from a DB
22:02.08SQLDarklyOnce the meetme is started people can join the conference but random people seem to drop
22:02.24*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
22:02.30SQLDarklyI do not see anything in the messages log as to the cause.
22:02.42SQLDarklyI was not monitoring the console during the drop
22:03.04jasonwootSQLDarkly: happens to my meetme's constantly with zap callers, but not with SIP
22:04.16SQLDarklyThere must be a cause....
22:05.19aiksa[LV]SQLDarkly: not even anything in debug log?
22:05.32aiksa[LV]verbose sometimes is not verbose enough
22:05.41aiksa[LV]off for a smoke
22:05.53*** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net)
22:13.47aiksa[LV]back
22:19.27*** join/#asterisk Assimilate (n=Assimila@72.22.242.66)
22:25.49Assiddoes polycom 301 do hinting?
22:26.40errrI think so
22:26.47[TK]D-FenderAssiiViewable in Buddies, but not on line-keys
22:27.27Trionnisok, slightly different track... anyone know if the Bridge application/manager command will work if you give it the Call-ID of each leg, or does it require the actual channel name?
22:27.28*** part/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com)
22:27.57verywiseman[TK]D-Fender, i am now need fxs/fxo equipment from 1-250 fxs port and 1-48 fxo port , and there are 2 options (PCI card and ATA), and when i ask in this channel , somebody prefer ATA for small business
22:28.05Assid[TK]D-Fender: so same thing as a directory?
22:28.41[TK]D-FenderTrionnis: Bridge bridges CHANNELS.  Call-id is a PROTOCOL-SPECIFIC thing.  This has been abstracted
22:29.11[TK]D-FenderAssid: similar
22:29.27*** join/#asterisk joobie (n=joobie@mx01.anric.com.au)
22:29.35[TK]D-Fenderverywiseman: AUdioCodes MP-124 <-----
22:29.35Assidtheres more?
22:29.49[TK]D-FenderAssid: More what?
22:30.13Assidisnt the only difference between hinting and directory is online keys?
22:31.06[TK]D-FenderAssid: huh?  Buddies screen just lists the contacts you have and their status
22:31.23[TK]D-FenderAssid: And I have no idea what you're getting at with "online keys"
22:31.37Assidlike online/offline (registered/not registered) ?
22:31.59Assid"on-line" keys
22:32.48[TK]D-FenderAssid>"on-line" keys <- drop this term
22:32.49*** join/#asterisk awk_r (n=awk_r@nat/digium/x-b2bceb54c5fa3021)
22:33.04[TK]D-FenderAssid: and you will see their STATUS
22:33.29Assidokay like if they are on call and such?
22:33.43[TK]D-FenderAssid: Yes
22:34.05Assidsweet.. i need to learn this..  but do OTHER buddies have to have polycoms?
22:34.20[TK]D-FenderAssid: No
22:34.50*** join/#asterisk ManxPower (n=manxpowe@router.asteriasgi.com)
22:35.09Assidbrilliant.. everyone else on ATA anwyays
22:35.19*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
22:35.41Assid[TK]D-Fender: oh, is it possible to check for the status of a phone on another asterisk server?
22:35.56Assidlike i have 2 boxes, in 2 locations
22:41.05[netman]~nat
22:41.06jbotsomebody said nat was Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
22:41.06aiksa[LV]hmm I was wondering perhaps someone did a backport of new manager events to 1.4.22 ?
22:41.06[netman]~ports
22:41.06jbotmethinks ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm
22:41.16*** join/#asterisk sucituanbo (n=full@c-24-21-121-148.hsd1.wa.comcast.net)
22:41.51*** join/#asterisk pittstains (n=frank@mx1.distributivenetworks.com)
22:43.08pittstainsI have a question about passing variables from the AMI to the dialplan: is it possible?  If so, how?  Here's what I'm trying without success:
22:43.12SQLDarklywould MOH still be played with zap/pseudo even if telephony hardware is installed?
22:44.52pittstainsin the AMI...
22:44.52pittstainsAction: Originate
22:44.52pittstains[other necessary params]
22:44.52pittstainsVariable: FOO=bar
22:45.53pittstainsin the dialplan:
22:45.53pittstainsexten => 0,n,Verbose(Test: ${FOO})
22:48.44*** join/#asterisk Jembanin (n=Ben@193.103-84-212.ippool.ndo.com)
22:50.00pittstainswhat i get when i run it is "Test: " instead of "Test: bar"
22:50.23Kattyyeah we should probably see some pastebin
22:50.25Kattyand some cli output
22:50.28eppigyno
22:50.29Kattymmmkay
22:50.33eppigylet me read your thoughtd
22:50.36eppigythoughts
22:50.40Kattyeyes eppigy
22:50.43KattyWHAT AM I THINKING
22:50.45eppigy8[]
22:51.13Kattytaps fingers
22:51.21eppigyhold please
22:51.51Kattyhttp://mfrost.typepad.com/photos/uncategorized/2008/03/16/overslurp.jpg
22:52.08pittstainswhat do you need me to show you?
22:52.37eppigyyou are currently wondering what is taking me so long to come up with a witty response?
22:52.41eppigyyou are currently wondering what is taking me so long to come up with a witty response
22:52.57drmessanoHA
22:53.01drmessanoNow thats funny
22:53.01eppigyYEAH SON
22:53.07drmessanoA cat without a caption
22:53.15drmessanoOLD SKOOL
22:53.40Kattyeppigy: http://pastebin.ca/1303076
22:53.45Kattyeppigy: what am i thinking?
22:53.49drmessanoThat cat is all like "I just got licked".. not like "ZOMG, HE R TASTED MEH"
22:54.24eppigyTRUE
22:54.27eppigyTIME
22:54.28eppigyTO
22:54.29eppigyDIP
22:54.30eppigyOUT
22:54.34eppigyyeah it is
22:54.58eppigyi just ordered take out
22:55.02Kattyhot
22:55.05eppigyso i gotta give them about 10 minutes
22:55.21Kattyyeah i'm on hold.
22:55.25Kattytaps fingers
22:55.34eppigyKILL EVERYTHING
22:55.40Deeewaynewaves to Katty
22:55.46Kattydeeeeewayneee!!!!
22:55.51Kattyhugs Deeewayne
22:56.07eppigyimitates a third wheel
22:56.07Deeewaynegives Katty a Happy New Year hug
22:56.11Katty:>
22:56.16Kattyhappy mew years to you too!
22:56.37Kattyi learned so much today.
22:56.46Kattybioluminescent dana octopus squid
22:56.52Kattyand sodium acetate hot ice.
22:56.59pittstainshmmmm, seems to be working now
22:57.18aiksa[LV]i just did a thing I thought is never possible!
22:57.23pittstainsthe only thing i changed was to change my variable name from THIS_IS_A_TEST to TEST
22:57.34pittstainsi guess maybe underscores in variable names are bad
22:57.58aiksa[LV]I managed to trick zaptel to choose correct timing device by calling ztcfg twice in a row with two seperate configuration files
22:58.04aiksa[LV]yeah! :)
22:58.21Kattywoo! time to go
22:58.22Kattydarts
23:00.11aiksa[LV]if only tzafrir_laptop was here, he would say I am nuts
23:00.13aiksa[LV]:P
23:01.23eppigypittstains: underscores denote inheritable variables
23:01.32eppigyi dont know if that has something to do with i t
23:02.51pittstainsinheritable variables?!?!  never heard of it -- time to reread the manual!
23:02.54ix33is going from zaptel -> dahdi difficult or am i just retarded?
23:03.53*** join/#asterisk Assimilate (n=Assimila@72.22.242.66)
23:06.22aiksa[LV]ix33: it depends on what modules you are using
23:08.14drmessanoix33: Its very easy.. There 162 steps you need to follow in readme1.txt thru readme14.txt, but beyond thats it all gravy
23:09.03*** part/#asterisk Alric (n=Alric@masq.hyperusa.com)
23:09.32cvnetanyone know of any softphone where you can change the sip port?
23:09.56SQLDarklyNo match Their Call ID: 17f9@10.43.160.24 Their Tag 0x17f9-1owxe1 Our tag: as5908261c | any idea what that message means. THis is in my debug log at the point of drop
23:10.04jasonwootmost that I've used... cvnet
23:10.06jasonwootzoiper
23:10.31cvnetjasonwoot thanks
23:11.09*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)
23:12.18aiksa[LV]drmessano: not to mention that some modules havent ported to dahdi yet
23:12.25cvneti changed the port for sip to 2222 on *, yet i can not connect to it, I have the firewall off, is there way to find out if that port is open?
23:12.48drmessanoWhich ones?
23:13.47_ShrikEcvnet: nmap
23:13.59cvnetnmap is a unix command?
23:14.38_ShrikEyes, you may need to install it though
23:14.54cvnetyap nmap?
23:15.08cvneti have it installed
23:19.07cvnetif i want to use a different por than 5060 for sip, is there any rules ?
23:19.22jasonwootnmap localhost
23:19.47jasonwootsip.conf must be configured to bind to that port however
23:20.09jasonwootrunning a tcpdump on the appropriate interface will tell you if that port is responding
23:20.18cvnetthanks
23:21.21*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
23:22.10*** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
23:22.33cvnethum, when i do nmap, and sipconf binds to 5060 it doesnt show it there
23:22.44cvnetyet i can connect via softphone
23:23.06aiksa[LV]drmessano: not the ones from digium of course
23:23.14aiksa[LV]third party modules
23:23.23drmessanoObviously
23:23.29drmessanoStill doesnt answer
23:23.33*** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com)
23:23.53aiksa[LV]xpd_usb
23:24.00aiksa[LV]which relies on bristuff
23:24.07aiksa[LV]but I might be wrong
23:25.26aiksa[LV]cvnet: there are two meanings of a port in sip clients
23:25.52aiksa[LV]i hope this is all ABC to you, but stil not to confuse
23:26.26aiksa[LV]the SIP port option in Zoiper can realte to either a) The port on which zoiper is listening to b) to which port of the SIP server to connect to
23:27.18*** join/#asterisk MrNaz (n=mrnaz@124-168-62-116.dyn.iinet.net.au)
23:27.21aiksa[LV]drmessano: I just finished a day long of butsex with zaptel :)) but in the end i managed to break its spirit ;)))
23:28.15*** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net)
23:28.57jjshoecan sox "display/print" what "volume" a sound file is?
23:29.06jjshoefor example, when trying to normalize all your audio files to be the same level
23:31.01*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
23:31.22aiksa[LV]jjshoe: sorry cant answer this one
23:34.03drmessanoSo.. google is working on building its own routers
23:34.14drmessanoThey are using their own switches already
23:34.27drmessanoPretty soon: Google will make it's own users
23:34.39drmessanoSoylent green <--
23:35.24beekdrmessano: you said "make", not "eat" its own users.
23:37.29[TK]D-Fenderbeek: We're relatively sure that SoylentCo didn't eat their own product ;)
23:37.34*** join/#asterisk c017 (n=c017@85.232.120.248)
23:40.21*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:42.23*** join/#asterisk oh207 (n=oh207@nylug/member/oh207)
23:44.43cvnetaiksa[LV] in Zoiper I only see one port option, Sip Option -> Port, I thought thats the port it connects to the server.  have you ever tried to change the SIP port and have a softphone connected to ?
23:48.17aiksa[LV]cvnet let me open the configuration menu of my zoiper
23:48.31cvnetthanks
23:48.59cvnetwhen i click on Advance Option then i get the Sip Option
23:49.03aiksa[LV]i have that under protocol options
23:49.17cvnetya and sip option right?
23:49.20aiksa[LV]thats a port Zoiper listen for SIP packets on
23:49.28cvnetthat is the port which it connects to server right?
23:50.11*** join/#asterisk Kumbang (n=unknown@125.163.83.153)
23:50.27aiksa[LV]if you wanted to change a port on the server it connects to, i suppose you could simply add it with server.host.tld:5061 in specific account options
23:50.31cvnetso its not the port it connects to server? hum then why would it be 5060 by default
23:50.44cvnethum, let me try taht
23:50.49aiksa[LV]cvnet: because 5060 is a default SIP port
23:50.59aiksa[LV]Zoiper is acting as both server and client
23:51.11cvnetok let me try that
23:52.18cvnetoo boy
23:52.32cvnetthanks, I waisted 3 days to figure it out, man I feel dumb
23:52.41cvnetaiksa[LV] thanks a bunch
23:58.32*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
23:58.57*** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)

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