IRC log for #asterisk on 20090901

00:13.38*** join/#asterisk dysinger_ (n=tim@71-20-35-99.war.clearwire-wmx.net)
00:15.22ayesoraden_work: yes it is in /etc/logger.conf... it should allready be logging to /var/log/asterisk/messages
00:17.04*** join/#asterisk netpro25_ (n=mmanning@fl-71-0-164-16.sta.embarqhsd.net)
00:18.51netpro25_hello, can someone tell me more about this GSM bug and if it is in Asterisk 1.4.21.2~dfsg-1ubuntu3
00:19.03KyleKwhat gsm bug? :)
00:19.16russellb~gsmbug
00:19.17infobot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read :  http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39
00:19.26netpro25_thanks russellb
00:19.30netpro25_;-)
00:19.33russellbinfobot knows all
00:19.34infobotand don't you forget it
00:19.39russellbO.O
00:19.40netpro25_hah
00:20.22netpro25_russellb: do you know if that bug still exists in the most recent release
00:20.46russellbnot AFAIK
00:21.06russellbI haven't heard complaints about it in a long time
00:21.30netpro25_yea Ubuntu 9.04 uses an older 1.4.21 version
00:21.37russellblammmmme
00:21.44netpro25_hah
00:22.58netpro25_okay so I have another ubuntu server with 1.4.17 and I did not have these gsm problems
00:23.01KyleKcompile and package a newer version and send it to them?
00:23.31netpro25_well I was looking to see if there was a repo that I can just add usually there are bleeding edge repos
00:23.38netpro25_for things like this
00:23.40netpro25_did not see one
00:24.06KyleKwell asterisk is a bunch of "compile your own" people
00:24.17carrarYEAH!!
00:24.23netpro25_heh, yea... It just makes things less standard
00:24.52netpro25_debs are so much easier to deal with. If i could package my own then I would, just dont know how
00:24.54KyleKwell if you really want a standard, step up and start packaging asterisk for debian
00:25.15KyleK(im assuming ubuntu just autocopies that package from debian)
00:25.23netpro25_possibly
00:26.39netpro25_I assume most asterisk people use other distros as ubuntu is just too easy to use
00:26.49netpro25_based on your earlier statement
00:27.08KyleKi use ubuntu
00:27.20netpro25_cool
00:28.07grandpapadotdear god, it takes like 5 minutes to install asterisk from source
00:28.30grandpapadotif you want to upgrade via deb you gotta jump through about 30 hoops and waste half a day
00:28.31netpro25_grandpapadot: okay I am gonna take the plunge
00:28.47netpro25_is 1.6 stable?
00:28.58netpro25_nm
00:28.58netpro25_rc
00:29.01KyleKlol
00:29.01grahamsaanetpro: yes
00:29.02grandpapadotPossibly, I run 1.4.26.1, very stable *in most cases*
00:29.11KyleKnetpro25_: theres lots of versions of asterisk right now
00:29.19netpro25_yes there are
00:29.26KyleK1.6.2 is the svn trunk, im using 1.6.1 svn
00:35.47carrarOh my
00:35.55fileactually trunk is trunk, and 1.6.2 is 1.6.2 - they are different
00:36.26carrarmore junk in your trunk
00:38.15theharopens file's trunk
00:38.34netpro25_lol
00:38.34russellbcartwheels
00:38.45thehartosses cheese its at russellb
00:38.55russellbnom nom nom nom
00:39.33theharindeed, sir.
00:39.41netpro25_Hey kinda off topic but what do you guys think about the LPI Certs for linux
00:39.50netpro25_anyone heard of it or done it?
00:39.52thehari think certs are crap.
00:39.59netpro25_So do I
00:40.06carrarThey are not Scottish!
00:40.11theharmy workplace also thinks the same
00:40.11netpro25_but i have a BS in CS and I cant find crap for work
00:40.23theharqwest is hiring locally, lol
00:40.24netpro25_(just graduated)
00:40.29theharah
00:40.37thehareBay is hiring a local telcom engineer as well
00:40.46theharbut that is avaya or cisco
00:40.48netpro25_yea local as in cali?
00:40.52theharsalt lake city
00:41.03netpro25_yea I am married
00:41.16theharoh here i can fix that
00:41.18carrarand mormon?
00:41.20theharcuts off the ball and chain from netpro25_
00:41.22netpro25_kinda hard to just pick up and go
00:41.25carrarheh
00:41.26theharis not mormon
00:41.45*** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net)
00:41.46*** mode/#asterisk [+o Deeewayne] by ChanServ
00:41.55netpro25_Now I would rather do a LPI cert then a Microsoft cert anyday
00:42.11carrarInternet Certified!
00:42.16theharyou are probably having a hard time going against people with work experience with your brand new degree.
00:42.27theharexperience > degree w/out exp.
00:42.42netpro25_yea exactly. I do have 5 years experience though
00:42.42netpro25_as a network admin
00:42.43theharprevious to your degree?
00:42.45theharor concurrently
00:42.46netpro25_yea
00:42.49theharwell there you go
00:42.50netpro25_well concurrently
00:42.52theharah.
00:43.06thehartalk to your university. they can probably help =)
00:43.14netpro25_lol yea a masters
00:43.23carrartell them you want your money back
00:43.38netpro25_hah i knew that was coming
00:43.44theharmost universities will help find placements or jobs.. internships, something.
00:44.25netpro25_yea I have been applying for shit jobs there
00:44.31netpro25_$10 hr
00:44.33netpro25_and no go
00:44.48*** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do)
00:44.52theharwelcome to the economy =)
00:45.09netpro25_yes, you are kinda blind to it when you are in school
00:45.13netpro25_then you get out and bam
00:45.14thehari help hire techs in our company and we got thousands of resumes last job posting
00:45.23netpro25_damn
00:45.50netpro25_yea I have been doing contract work for the time being. Trying to build up my business. Maybe it will turn out for the better
00:46.45carrarget yourself on linked-in
00:46.49netpro25_so yea when I compile from source on like ubuntu, I should first uninstall any repo packages right?
00:46.54netpro25_carrar: I am on there
00:47.00carrarnetwork!
00:47.06netpro25_heh. I have.
00:47.15netpro25_Actually I think I am in the asterisk group
00:47.35carraralways stuff on CL too
00:47.44netpro25_yea thats hit or miss
00:48.02netpro25_only thing I have not tried is temp agencies
00:48.11netpro25_saving those for last resort
00:48.25carrarrecruiters are nice to know when you are looking
00:48.38carrarannouying as hell when you're workin
00:48.46netpro25_hah yea
00:48.47theharnah..
00:48.52theharit's nice to know you're wanted when working
00:50.10carrars/when working//
00:50.35*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
01:05.33*** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do)
01:12.57netpro25_hey before building asterisk from scratch should I uninstall the existing debian package?
01:13.26jayteesure, why not
01:16.23netpro25_lol
01:16.49*** join/#asterisk TJNII (n=TJNII@207.189.199.58)
01:16.50*** join/#asterisk ming_zym (n=ming_zym@124.127.101.0)
01:19.56netpro25_sweet asterisk ascii art
01:21.23drmessanonetpro25_: yeah, love it.. if only the effin devs spent their time fixing bugs and adding features instead of working on Asterisk ASCII art, imagine what we could have
01:21.48netpro25_hah... Yea that had to have taken a good amount of time
01:21.59netpro25_its original though
01:22.03netpro25_have not seen that before
01:22.23drmessano"hey, we should fix that bug drmessano reported"  "Nah, fuck him, lets have some Dr Pepper"  "RIGHT ON!"
01:22.23netpro25_maybe I have not been compiling from source enough to notice
01:22.25netpro25_heh
01:22.42netpro25_lol
01:23.15netpro25_drmessano: what phones do you use
01:25.41netpro25_any ideas of the top of your head for this error: Unable to open pid file '/var/run/asterisk.pid': Permission denied
01:25.50drmessanoWell, Linksys and Polycom
01:25.51netpro25_when trying to start asterisk
01:25.58drmessanoYeah
01:25.59netpro25_yea I have linksys its nice
01:26.05drmessanoWhich Linksys?
01:26.21*** join/#asterisk cviniciusm (n=cviniciu@189.27.12.215.dynamic.adsl.gvt.net.br)
01:26.28drmessanoYou need to set perms .. chown that dir
01:26.29netpro25_linksys SPA941
01:26.41drmessanoI have a few SPA941s.. they're nice
01:26.42cviniciusmHello.
01:26.50netpro25_yea
01:26.51drmessanoHi circumcision
01:26.55drmessanoWassup?
01:27.00netpro25_lol
01:27.08netpro25_circumcision
01:27.14cviniciusmHAHAHA.
01:27.25drmessanoSorry, <TAB> key is broken
01:27.28drmessanohad to paste from google
01:27.35*** join/#asterisk freakazoid0223 (n=knoppix@pool-71-246-17-206.phlapa.fios.verizon.net)
01:27.37drmessanoDid you mean: Circumcision
01:27.38drmessano:(
01:27.39drmessanoSorry
01:28.07netpro25_drmessano: chown the run directory?
01:28.54cviniciusmWhat´s the Sg8obX2Q
01:29.01drmessanouseradd -c "Asterisk PBX" -d /var/lib/asterisk -s /bin/false asterisk
01:29.02drmessanoopenssl rand -base64 10 | passwd asterisk
01:29.02drmessanomkdir /var/run/asterisk
01:29.02drmessanomkdir /var/log/asterisk
01:29.02drmessanochown -R asterisk:asterisk /var/run/asterisk
01:29.02drmessanochown -R asterisk:asterisk /var/log/asterisk
01:29.07drmessanoThats from my playbook
01:29.08netpro25_ah
01:29.44netpro25_ah same shit
01:29.52netpro25_let me try something
01:29.56cviniciusmError...error...
01:30.04cviniciusm...sorry.
01:30.06drmessanoYou have /var/run
01:30.15drmessanoCheck your asterisk.conf
01:30.20netpro25_okay
01:32.29*** join/#asterisk Kumbang (n=whazzup@125.163.83.153)
01:33.18netpro25_drmessano: sweet. For some reason the default ubuntu asterisk.conf had just directories and not other info
01:33.23netpro25_changed that
01:33.30cviniciusmI have installed Asterisk 1.6.1.4, but I don't known what's the port udp/5000 ?
01:33.48netpro25_not sure what you mean
01:33.53netpro25_what it is used for?
01:33.59drmessanoudp 5000?
01:34.20drmessanoDid you get the tarball from The Pirate Bay?
01:34.25netpro25_lol
01:34.44netpro25_yay no more GSM bugs
01:34.45cviniciusmYes, netstat -lunp shows port udp/5000 for the process asterisk.
01:35.18cviniciusmNo, I get one from Digium site.
01:35.20drmessanoDoes it have any open connections to russellbryant.net?
01:35.36netpro25_lol
01:36.49netpro25_drmessano: okay so I have some packages that are haning out in the auto-remove from the old asterisk ubuntu install. Do you think there is any possibility that the new asterisk install is dependent on those?
01:37.18drmessanoThey're probably old, obsolete shit.. So, doubtful
01:37.27netpro25_okay gonna auto-remove
01:38.09drmessanoFriends dont let friends install from apt-get
01:38.09cviniciusmrusselbryant.net is a domain from godaddy.com .
01:38.25*** part/#asterisk FoxValley (n=foxvalle@Elgn-Mlnm-WiFI.FoxValley.net)
01:38.52drmessanoAwesome, you're just one step away from sleeping in the bushes outside Russell's house
01:39.01drmessanoNote: bring blankets
01:39.21drmessanoand lots of cat food
01:39.38netpro25_drmessano: I still have some sound hickups
01:39.50netpro25_and I installed the most recent trunk release
01:40.01drmessanoBribe the cats, and maybe you'll get your hands on his toothbrush too
01:40.05netpro25_any suggestions?
01:40.08netpro25_lol
01:40.12drmessanonetpro25_: Yeah, you dont want trunk
01:40.20netpro25_you are scaring circumcision away
01:40.27netpro25_okay so I will recompile regualr
01:40.43netpro25_how can I uninstall trunk
01:40.57drmessanomake uninstall ?
01:41.12netpro25_lol
01:42.08*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
01:42.47netpro25_yay more ascii art
01:43.26netpro25_so I assume if you delete the source directory then you are screwed
01:43.31netpro25_which I did not
01:43.36netpro25_but have in the past
01:43.43KyleKscrewed for what
01:43.46netpro25_uninstall
01:43.57KyleKoh i installed to /home/asterisk for that reason
01:44.21netpro25_yea I want to make debs to avoid that
01:44.29KyleKwell --prefix=/home/asterisk --localstatedir=/var because I'm crazy insane
01:44.29netpro25_but have not found an easy way to do it
01:44.38netpro25_hah
01:44.48KyleKfind a debian packaging guide?
01:45.01netpro25_yea have not looked into that
01:45.09netpro25_guess I have some time know that I am unemployeed
01:45.16netpro25_;-)
01:45.25netpro25_asterisk is a neat toy
01:45.44netpro25_i find joy in designing IVR's
01:45.48netpro25_and extensions
01:46.40KyleKwell basically you just need to hijack the make install part
01:46.47cviniciusmIs there a patch from 1.6.1.4 to 1.6.1.5 ?
01:47.04netpro25_whoa 1.6
01:47.09netpro25_sounds risky
01:47.48drmessano1.6 pwns
01:47.57netpro25_any major improvements?
01:47.59cviniciusmHow to move from Asterisk 1.6.1.4 to 1.6.1.5 ?
01:48.50drmessano1.2 was like Windows 95.. 1.4 was like Windows 98.. 1.6 is like Google OS
01:49.00drmessanoIts a nirvana eutopia
01:49.10netpro25_hah, so we went from a full pc to a netbook with a limited OS
01:49.18netpro25_nice
01:49.22netpro25_sounds like a downgrade
01:49.39drmessanoCommunist
01:49.51netpro25_hah
01:50.01drmessanoYou know how I convince people to use 1.6
01:50.02drmessano?
01:50.03netpro25_no I like google, no wait, I love google
01:50.04*** join/#asterisk TJNII (n=TJNII@207.189.199.58)
01:50.19netpro25_drmessano: no because if I did I would probly be using it
01:50.39drmessanoI call them on the phone, talk up 1.6 a little.. then I abruptly end the call with "hey, sorry, we have 1.2 on this system here, and I need to reboot".. then I hang up
01:50.50netpro25_hah
01:50.56netpro25_sweet
01:51.04KyleKis google os out yet?
01:51.11cviniciusmhahaha.
01:51.18drmessanoNope
01:52.08netpro25_i am waiting for the day they take over the world
01:52.14netpro25_those jerks
01:52.24KyleKi could use some google back pain relief
01:52.28netpro25_hah
01:52.38drmessanonetpro25_: No need to wait
01:52.47netpro25_google eye doctor
01:52.56KyleKgoogle eye doctor beta
01:52.58KyleK:D
01:54.07netpro25_hah
01:54.07drmessanoGo to google
01:54.15netpro25_they reinvented the beta
01:54.18netpro25_term
01:54.22netpro25_whats the url?
01:54.23drmessanoEnter: "What date will google take over the world?" in quote, and hit "Im feeling lucky"
01:55.30netpro25_is that your site?
01:55.31netpro25_lol
01:56.36netpro25_do you know if the new google voice uses asterisk?
01:56.57drmessanoNo
01:57.13netpro25_no you dont know?
01:57.21drmessanoI dont know
01:57.27netpro25_I doubt anyone will ever know
01:57.34netpro25_except the engineers
01:57.36KyleKwhats the svn command to check for updates but not do an update
01:58.04netpro25_speaking of google
01:58.22drmessanosvn donothing
01:58.56russellbi wish i knew ... I think grandcentral used it
01:59.13russellbbut they took enough time between acquisition and launching google voice to rebuild the whole thing
01:59.33netpro25_yes
01:59.40netpro25_are you using it?
01:59.45russellbgoogle voice?  yeah.
01:59.52netpro25_I find it kinda a pain
01:59.59netpro25_cause you end having two numbers
02:00.05cviniciusmI think Google uses libjingle.
02:00.08russellbKyleK: there isn't really a good command for that, actually.  :-/
02:00.54russellbI suppose what you could do is run "svn info" to see what revision you're at, and let's call that <current_rev>
02:01.03russellbsvn log -r <current_rev>:HEAD
02:01.38KyleKk i'll give that a shot
02:05.10cviniciusmSo, the development cycle of Asterisk is evolution based?
02:05.37*** join/#asterisk denon (i=root@synapse.subneural.net)
02:05.37*** mode/#asterisk [+o denon] by ChanServ
02:05.42russellbI'm not terribly sure what you mean by that ..
02:06.30mk12picklei preferr the orbital palm sander to the face
02:06.54netpro25_okay
02:06.56netpro25_?
02:07.00russellbblinks
02:07.04drmessanorussellb: FYI, the bushes need a little pruning, and the spicket in the back of the house leaks a little
02:07.26drmessanorussellb: You'll be happy to know I put all you mail back, after I finished smelling it
02:07.38drmessanoyour*
02:07.50*** kick/#asterisk [drmessano!n=russellb@asterisk/digium-open-source-team-lead/russellb] by russellb (+17 creepy)
02:07.51*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano)
02:07.59drmessano[21:38] <cviniciusm> russelbryant.net is a domain from godaddy.com .
02:08.04drmessano[21:39] <drmessano> Awesome, you're just one step away from sleeping in the bushes outside Russell's house
02:08.12drmessanoahhaha
02:09.11drmessanorussellb: Do you own any polydactyl's?
02:09.33cviniciusmAt the end of each evolution cycle, we get a new code ready for use, then the several iterations are patched apart.
02:10.22russellbpatched apart ... is that like picked apart or torn apart?
02:10.29russellbor is that something different :-)
02:10.34*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
02:11.17russellbwe have feature frozen releases that we maintain with bug fixes, and we release bug fix releases on a fairly regular basis
02:11.37netpro25_any ideas on this error
02:11.38netpro25_Unable to connect to remote asterisk (does /var/run/asterisk//var/run/asterisk.ctl exist?)
02:11.48russellbFor additional information on asterisk releases: http://www.asterisk.org/node/48602
02:11.57cviniciusmThe Asterisk has several branches 1.6.0.x, 1.6.1.y, 1.6.2.z and so on.
02:12.09russellbYeah, see the link I just posted.  That will explain it.
02:12.15russellbnetpro25_: that path is seriously messed up.
02:12.19netpro25_hah
02:12.21netpro25_it is
02:12.32netpro25_where is it pulling that from?>
02:12.48russellbasterisk.conf?
02:12.50drmessanoYou can also ask leif madsen ... he can into it in great detail, and I have his home phone number
02:13.04netpro25_bingo
02:13.18netpro25_sweet
02:13.26netpro25_even more gratifying then google
02:13.49russellbthat will cost you $9.95
02:14.07drmessanoYes
02:14.15netpro25_damn and I thought adwords was a rip off
02:14.18russellbdrmessano: did you get your SfA license btw?
02:14.31drmessanoLeif will cost you.. if you want something free, we can give you file's.. nobody ever wants to call him
02:14.48russellbharsh
02:15.04drmessanorussellb: Yeah.. it never came.. I e-mailed customerservice and it was resent in about an hour
02:15.14russellbcool, glad it got resolved.
02:16.02drmessanoThanks.. Now I just need some time to play with it
02:16.40*** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru)
02:18.59netpro25_well okay so yea what are some other causes of jittery sound
02:19.13raden_worki want to be able to dial *67 before a number and have it set the caller id to unavailable or unknown how would i go about that
02:19.38*** join/#asterisk KuASha (n=ahmed@203.189.242.181)
02:19.39KyleKI hope skype gets its wideband codec sorted out soon
02:19.50KuAShahi guys.
02:19.57drmessanoKyleK: What do you mean?
02:20.02KuAShai am a newbie, looking for an asterisk expert to work with me on a project :)
02:20.09KuAShais anyone here for paid work ?
02:20.17russellbI'll do it for 1 million dollarz!
02:20.30netpro25_lol
02:20.37drmessanoguesses its probably illegal
02:20.53KuAShaaww :(
02:21.04russellbdrmessano: ha
02:21.19KuAShaillegal to work in some paid project for opensource software ?
02:21.56KuAShai belive guys are in thousand of data center where linux are being used are doing illegal job then, such as system administrators :P
02:22.20KyleKdrmessano: my interest in skype is its wideband codec, so before i plunk down for SFA it needs to support wideband, probably have the asterisk server transcode to G.722 or somesuch
02:22.25manxpower~answers
02:22.25infobotit has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
02:23.44russellb~questions
02:23.45infobotremember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html>
02:23.53russellboic.
02:24.34russellbKyleK: do you have wideband endpoints in use already?
02:24.47*** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com)
02:25.08drmessanorussellb: it never fails.. i defend 5 or 6 guys who need help with high call volume startups using Asterisk that get accused of being spammers, and number 7 comes right out with "I am new to asterisk... I want to spam millions with auto warranty extension calls.. can you help me with my dialplan"
02:25.12drmessano:( FAIL
02:25.20ifluxanyone here know of some cheap ip kvm solutions?
02:25.25cyberfab007I LOVE ASTERISK!!!!!!!!!!
02:25.34ifluxI know that there was one made by dlink for awhile that you could get for like $50
02:25.37cyberfab007just had to get that out
02:25.41ifluxbut it was EOL'd
02:25.43russellbcyberfab007: <3
02:25.45drmessanoASTERISK ROCKS
02:25.50drmessano... probably
02:25.56russellbdoes it also roll?
02:26.16cyberfab007Man you know I have been configuring and building asterisk systems for 7 years and this is the first time I have been in the freenode room
02:26.27KyleKrussellb: nope i just have 3 SPA3102's in use right now
02:26.29russellbdoes the math ...
02:26.32*** join/#asterisk blkry (n=chatzill@96.37.27.72)
02:26.34russellbcyberfab007: since 2002, really?
02:26.49cyberfab007yegh , like the first system I install were sooooooooooooo basic
02:26.55drmessanoI try not to claim asterisk is the best shit since sliced bread or else the devs get a bad case of LEAD BOTTOM
02:27.07drmessano"ASTERISK IS.. REALLY NOT BAD"
02:27.25netpro25_russellb: any ideas on what other things I should do to diagnose jittery sound? Or anyone else.
02:27.28KyleKthe idea of using a single board x86 for a phone has crossed my mind :)
02:27.32drmessanocyberfab007: Long time listener, first time caller?  Named your first kid Qwell?
02:27.44mk12picklejitter is variation in latency
02:27.49ifluxsup drmessano
02:27.55drmessanoI've been using asterisk for 11 years now
02:27.57mk12pickleif all the latency is the same.. no jitter
02:28.00russellbnetpro25_: I suggest a hammer
02:28.08russellb(my helpfulness after hours goes down drasitcally)
02:28.13cyberfab007Actually First kid is a girl Named Nebula LOL
02:28.21netpro25_russellb: let me find one
02:28.24drmessanocyberfab007: Put down the weed
02:28.31cyberfab007I try to ,
02:28.52netpro25_mk12pickle: latency as in the internet connection latency?
02:29.01cyberfab007but yegh man I did my first asterisk install in fall 2002 , it was a small business  like 5 employees
02:29.13drmessanoWhat version?
02:29.14cyberfab007was my junior year at ASU
02:29.17KyleKnetpro25_: internet, wifi, overloaded lan
02:29.26drmessanoAugusta State University?
02:29.29drmessanowowow
02:29.41iflux***D00D!*** <-- me using asterisks like 20 years ago on efnet
02:29.45raden_workanyone have a called id script to block calls with like *67 they could share
02:30.16netpro25_damn see ya
02:30.16drmessanothrows a rope over the divide
02:30.17cyberfab007I think it was 0.2.0
02:30.31ifluxya know.. if I hadn't sent +++ instead of ***.. I totally would have thought that netsplit was suspicious
02:30.37KyleKraden_work: something like that should be easily available
02:30.38cyberfab007Arizona State man
02:30.42ifluxr/hadn't/had/
02:30.46KyleKodd that it isn't
02:30.57cyberfab007home of the SUN DEVILES WE ROCK !!!!!!!!!!
02:31.15KyleKraden_work: you make an *67 extension that sets caller id and then Waitexten(10)
02:31.18drmessanocyberfab007: Marko was sending me Betas over ICQ back when it was called "Star"
02:31.23cyberfab007well 0.3.0 did not come till the end of the year there
02:31.27drmessanocyberfab007: I guess that was around 97 or so
02:31.40raden_workKyleK, how can i do it so they can dial the whole thing  like *671234567890
02:31.51KyleKoh god not more pasturbating
02:31.54cyberfab007nagh man 0.3.0 was around jan -feb 2003
02:31.59raden_workstrip first 3 digits
02:31.59KyleKraden_work: regular expressions match it
02:32.05drmessanopasturbating?  <--- <3
02:32.19cyberfab007LOL , I guss that is some kind of computer sickness
02:32.20cyberfab007lol
02:32.24KyleK${EXTEN:3} or something
02:33.00cyberfab007anyways I thought you all should know my company is making a Joomla extension that will easily provide a front for customer thorough joomla off a elastix box
02:33.28drmessanoYou know you've been using Asterisk too long when someone accuses you of taking something out of context and you insist "No, I used a Goto!!!"
02:33.29cyberfab007I think 0.4.0 cam like late april early may or somthing like that
02:33.37cyberfab007LOl
02:33.53russellboh yeah, well, i actually taught mark spencer how to code
02:34.00russellbso you know, no big deal
02:34.07KyleKholy crap i may have gotten that syntax right
02:34.27cyberfab007Haha ,
02:34.32ifluxya know.. the asterisk box in my house is one of the two things I've put in that have extremely high wife acceptance factor
02:34.45ifluxmy wife loves it because I figured out how to use it to save us $50/mo
02:34.50drmessanorussellb: I helped him code Gaim while chatting with him over early Gaim alphas
02:34.51cyberfab007who remember blowing the soder off the old 56k  modem boards?
02:35.10cyberfab007man I was like 17 back then , how time flies
02:35.12manxpowerIf you never got the wife I bet you'd be saving much more than $50/month
02:35.18ifluxcyber: you kidding.. I still have an original unsoldered Apple
02:35.25cyberfab007LOL
02:35.33drmessanorussellb: It was like the picture in "Back to the future".. i would feed him a bad line of code, and one of my buttons would disappear
02:35.35cyberfab007man thats cool
02:35.37KyleKmanxpower: hookers are cheaper in the long run? ;)
02:35.38drmessanorussellb: good times
02:35.47manxpowerKyleK: friends with benefits
02:35.55KyleKeven better
02:36.03cyberfab007So listen guys I am rallying the troops soon
02:36.07*** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) [NETSPLIT VICTIM]
02:36.07*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) [NETSPLIT VICTIM]
02:36.07*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) [NETSPLIT VICTIM]
02:36.08*** join/#asterisk m477au (n=m477au@60.241.150.14) [NETSPLIT VICTIM]
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02:36.09*** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk)
02:36.09*** join/#asterisk fnordus (n=dnall@70.70.0.215) [NETSPLIT VICTIM]
02:36.09*** join/#asterisk skymeyer (n=skymeyer@mailout.itconnect.be) [NETSPLIT VICTIM]
02:36.09*** join/#asterisk Zhad (n=tom@server30261.uk2net.com) [NETSPLIT VICTIM]
02:36.16ifluxcyber: it's in my parents attic.. my mom was like.. "hey can we throw out this super old computer that you're never taking home?"
02:36.19drmessanoTHE REVOLUTION IS HERE?
02:36.29drmessanoGREAT, I HAVE THE GRENADES
02:36.30ifluxI was like "*cough*sputter*NO!"
02:36.33KyleKI guess i gotta stop putting off this caller id view
02:36.37cyberfab007any good C programmers in here with php experience and asterisk hard-on?
02:36.44russellblooks around
02:36.52KyleKi dont have a hardon
02:36.54drmessanocyberfab007: you want to code a GUI for Asterisk in PHP?
02:36.58ifluxcyber: got a chan you want developed?
02:37.00cyberfab007No , that is done
02:37.08KyleKyou want someone to fix it?
02:37.13drmessanocyberfab007: you want to code a CLI for Asterisk in PHP?
02:37.17cyberfab007No , fixing is for morons
02:37.23cyberfab007coding is for pros
02:37.32cyberfab007No,
02:37.46KyleKif it needs fixing the pros weren't pro enough
02:37.49cyberfab007I am taking the elastix system and making moduals for it
02:37.57cyberfab007excatly
02:38.04drmessanoAHAHHAH
02:38.08drmessanoElastix?
02:38.12ifluxelastix is another centos based distro, right?
02:38.16cyberfab007Elastix
02:38.17cyberfab007yes
02:38.21cyberfab007but more than that ,
02:38.30cyberfab007has fram work for the ultimate god box
02:38.41drmessanoElastix has the worst asterisk binaries ever
02:38.41cyberfab007did I say god box
02:38.46ifluxif it were debian or ubuntu based I'd probably be slightly interested but if it's just centos based then I'd have no reason to not use piaf
02:38.56drmessanoor AsteriskNOW
02:38.59drmessanoCOUGH COUGH
02:39.06cyberfab007people , it is more than that ,
02:39.18cyberfab007any asshole can install and use elastix , this is important
02:39.26cyberfab007Right?
02:39.36drmessanoAny asshole can code and package elastix too, apparently..
02:39.38ifluxI just don't like the package management system that all the fedora based systems use
02:39.55drmessanoI ran Elastix here for a week
02:40.10cyberfab007I have about 40+ business running elastix , for 2 years now
02:40.14drmessanoThe next weekend, I put that box to better use and installed Vista on it
02:40.26cyberfab007OHHH MAN THAT IS HARSH TALK
02:40.30cyberfab007HARSH TALK
02:40.57cyberfab007i have elastix cluster that i have not updated in a year cause I am scared and it still runs like a rock
02:41.26drmessanoIt's a cluster alright
02:41.34drmessanoba-dump CHING
02:42.01cyberfab007Actually I am gonna upgrade with elatix 2.0 that will be running asterisk 1.6
02:42.11drmessanoSeriously, Elastix is slowsterisk.. They compile their binaries on a 286
02:42.49ifluxdrmessano: and they don't even use -O3 I bet
02:42.56raden_workam i doing something wrong caller id still coming through http://pastebin.com/d20e819f5
02:43.02iflux-O3 -funroll-loops, etc
02:43.19drmessanoiflux: Damnit, you went over my head.. Now I need to google
02:43.34cyberfab007But it is a solid distro , better than any other out there unless you install your own asterisk box running freepbx on you own distro
02:43.35KyleKraden_work: my itsp has a setting in the web interface to force caller id
02:43.46cyberfab007But this is the call
02:43.51cyberfab007Free networks ,
02:43.52KyleKor you're setting the caller id at *67 and in @to-callcentric ;)
02:43.53ifluxdrmessano: compiler flags that optimize the hell out of the code..
02:43.56drmessanocyberfab007: AsteriskNOW and PIAF are far more stable.. Elastix is slower than Trixbox
02:44.04KyleKpbx in a fire
02:44.16KyleKwho burns a pbx?
02:44.25ifluxI should make a super optimized version some time.. I bet you could squeeze a few hundred more calls out of an optimized system
02:44.25cyberfab007Humm , before I ran asterisk @ home
02:44.31cyberfab007than trixbox
02:45.06raden_workKyleK, trying to set it to something anonymous
02:45.06ifluxi hate the green machine
02:45.10cyberfab007than they went F***ed , and went astray of GPL , I could never leave trix box running turn the key and come back to delete logs , Elastix let me do that
02:45.26KyleKraden_work: 6042809000
02:45.36cyberfab007Anyways ,
02:46.21KyleKthen set the name to be Dialup Modem Pool ;)
02:46.28drmessanoYou never had to worry about Elastix being exploited because the boxes were too slow with their unoptimized binaries to make them worth using, even over dialup
02:46.50raden_workKyleK,  ?
02:46.53cyberfab007I am making some software that will do alot of things , one of them is letting someone install a joomla extension that will give them a customer frontend in joomla from a elastix box with a gateway to DIDX.net , set up a complete phone business in less than a day I say , but thats where it starts I pay a development team to do this
02:46.55KyleKjoking
02:47.02raden_worklol
02:47.03*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
02:47.13drmessanoAHHHHHH
02:47.15KyleKi always put in the number for my old dialup isp when sites ask for my number
02:47.17drmessanoYoure an idea guy
02:47.22drmessanoWell
02:47.28drmessano~happyclownpbx
02:47.29infobot[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone
02:47.38drmessano^^^ Bidding starts at 1 BILLION dollars
02:47.58raden_workand i cant dial out with vitelity for some reason :(
02:47.59raden_workhttp://pastebin.com/d3bdfe579
02:48.02KyleKerm
02:48.11raden_work<PROTECTED>
02:48.15raden_workwithout caller id
02:48.16cyberfab007i love vitelity , 2 years and not one issue
02:48.25KyleKraden_work: why are doing a Dial to local btw
02:49.26cyberfab007But hey who would like to take and asterisk box , and provide a front end for people to govern them selfs with asterisk running as the communications engine?
02:49.38drmessanocyberfab007: You remind me of this guy I used to talk to on Undernet years ago.. he could tell you 100 different household items you could grind up and throw in a hookah, and would finish with "Its all legit, man"
02:49.45raden_workSIP dont work cause the rest of the dial plan in outbound context
02:49.49raden_worki use SIP it dont work
02:50.15KyleKcant use goto?
02:50.20ifluxdrmessano: hahahahaha I remember someone like that
02:50.27KyleKi guess it doesn't matter
02:50.34drmessanoDamn, what was his nick
02:50.59drmessanoWas a food item, I think
02:51.01KyleKdrainbamaged?
02:51.12cyberfab007Funny
02:51.37ifluxit wasn't ratsalad was it?
02:51.41*** part/#asterisk mk12pickle (n=johaan99@dhcp063-129.navigonet.com)
02:51.48drmessanoFlapjack
02:51.52drmessanoThat was it
02:52.03drmessanoThat guy was... burned
02:52.55*** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com)
02:53.37cyberfab007I am working not only on the joomla app , but also on a smart house app that lets you use Luminvox to speak commands to your house and lets asterisk runs scripts that produce results such as food status in your fride , IM contact on line and anything else you can think of to ask your asterisk box to figure out for you LOL 6 months in development this package is
02:54.34drmessanocyberfab007: Working as in "writing code" or "idea"?
02:54.56ifluxcyber: do you get a 5D6 savings roll for 'forced out of parents basement'?
02:54.57raden_workanyone have any idea why i cant dial out with vitelity ? http://pastebin.com/d3bdfe579
02:55.03cyberfab007Imagin being on your laptop and asking your asterisk system to bring up your survalince equipment , whether you need eggs and milk or your cousin recently has posted on facebook
02:55.05KyleKI'm not an idea man but i come up with enough ideas to keep me busy
02:55.06iflux:-P
02:55.22KyleKraden_work: incorrect password?
02:55.45drmessanocyberfab007: Id prefer a freePBX facebook application so I only have to keep one browser window open at home
02:55.47raden_workI can register inbound  works fine
02:56.03*** join/#asterisk mumtazah (n=mumtazah@203.82.91.103)
02:56.03cyberfab007There is code, but I have investors and stupid policeys and stuff , it will be GPL but not till it is released
02:56.23drmessanoSo what percentage is done.. Estimated lines of code?
02:56.30drmessano"Napkins do not count"
02:56.31KyleKdrmessano: facebook uses xmpp doesn't it?
02:56.39drmessanoKyleK: Not currently
02:56.45KyleKoh
02:56.46cyberfab007raden-work , if you have half a brain and cant get vitelity workinf with there auto config script on their site , it is porbley the Distro your using
02:57.13drmessanoKyleK: They talked a big game about XMPP interop, but it turned out to be bullshit vaporware
02:57.27cyberfab007but what I am here for is to rally the troops
02:57.46ifluxdrmessano: hahaha.. it'd be fun to have your asterisk box post up facebook wall messages about who called you
02:57.49ifluxI should do that
02:57.49drmessanocyberfab007: Sorry, fresh out of Kool Aid, and my sneakers are in the wash
02:58.09drmessanoiflux: Easy.. Twitter.. > Facebook
02:58.30cyberfab007drmessano do you believe in GPL , like fucking religion man ?
02:58.32ifluxdrmessano: yeah that'd work..
02:58.34drmessanoTwitter API is super simple.. ive played with Twitter + Asterisk quite a bit
02:58.56cyberfab007gforce.ichorcom.com
02:59.09ifluxcyber: personally.. I love fucking religion.. those catholic chicks are always shouting Oh God!
02:59.20cyberfab007my public devel site
02:59.53cyberfab007Ohh they are the best , here in Toronto I lived in Forrest Hill , man those girls were out of contorl
03:00.23KyleKhey can i access cdr unique id from within app_voicemail?
03:00.34drmessanoSeems like the only project is a week old and it has one dev
03:01.07KyleKhehe gforce
03:01.10iflux"All connections are made at the rear" - adder.us
03:01.49cyberfab007Well they are the openprojects
03:01.53drmessanohans paperbag is the only dev, and that Jeremy guy keeps fucking blogging.. I bet hans is bitter jeremy cant be bothered to commit one stinkin line of code
03:01.55cyberfab007I have about 8 that work for me
03:02.15drmessanohttp://gforce.ichorcom.com/gf/project/elajoom/
03:02.25raden_work<PROTECTED>
03:02.39cyberfab007You should install the extension fuckign awsome
03:03.07cyberfab007next week , most of the basic pageviews will be done inside of joomla
03:03.39drmessanoBy hans?
03:03.40cyberfab007the week after joomla, vtiger, a2billing will be synced
03:03.48drmessanoWhen do we see jeremy commit?
03:04.10KyleKraden_work: you're probably calling out via the sip account thats attached to right?
03:04.11cyberfab007Well no Hanes actually has had to quite becasue of personal emerganices with his lady friend
03:04.12drmessanoHans already seems bitter.. It wont be long now before total meltdown
03:04.22drmessanoOh, only dev quit?
03:04.28drmessanoBRIGHT future
03:04.38cyberfab007Well he is a Joomla core developer
03:04.47cyberfab007so he has much on his plate already
03:04.47drmessanoand your only coder
03:04.56cyberfab007well no as I said this is my open site
03:04.57cyberfab007lol
03:05.05raden_workKyleK, http://pastebin.com/d49c0b549
03:05.14drmessanoand now you're here, because you have blog posts and need someone to write 10,000 lines of code to go with them?
03:05.18drmessanoRight?
03:05.28ifluxbahahaha drmessano.. that's not nice
03:05.49cyberfab007Actually , most the code you see on that site I made and Hannes put it in Joomal fram work
03:06.01cyberfab007and no for those project I have lots of developers
03:06.09cyberfab007about 10 that I pay
03:06.31drmessanoOh?
03:06.39drmessanoPublic repos we can look at?
03:06.56cyberfab007for this project I am thinking a E- Democracy Project ,
03:07.13*** join/#asterisk geneticx (n=geneticx@70.146.116.10)
03:07.18*** join/#asterisk shinao1 (n=shinao1@41.219.205.197)
03:07.50KyleKraden_work: so vitelity-outbound is configured to be a different sip peer than vitelity-inbound?
03:07.55drmessanoWhich project?  The one with no devs, or the mutlimilliondollar 10 dev coding tank we can't get a link to?
03:08.06raden_workKyleK, yeah thats what they sent me
03:08.08cyberfab007a project that can only be done by people for people
03:08.37raden_workbut callcentric is my 866 -452-3565 number that keeps getting forbidden i dont understand whats going on
03:09.08drmessanoYou're recoding Obama?
03:09.09raden_work[Aug 31 21:47:20] WARNING[10802]: chan_sip.c:15268 handle_response_invite: Received response: "Forbidden" from '"866-452-3565 x101"
03:09.21raden_workim x103 so im really confussed
03:09.30cyberfab007Ichorcom is Deleware corporation with 20 million shares ammended and has a million dollars in funding. Is registerd with the SEC and is filling it FORM D offering for you ino
03:09.32cyberfab007info
03:09.58KyleKraden_work: who is 69.179.99.17
03:10.07KyleKoh its you
03:10.09raden_workWAN connection
03:10.28raden_workWTF the aastra phone set the called id itself
03:11.06KyleKwell getting that forbidden response is separate from dialing vitelity
03:11.24KyleKbefore i checked your ip i thought vitelity was that ip
03:11.44raden_workvitel-outbound/tanning     64.2.142.29                 5060     OK (45 ms)
03:11.44raden_workvitel-inbound/tanning      64.2.142.15                 5060     OK (45 ms)
03:11.45raden_workcallcentric/17772445766    204.11.192.38               5060     OK (45 ms)
03:12.08*** join/#asterisk ketema (n=ketema@ketema.net)
03:13.17cyberfab007http://www.edgarcompany.sec.gov/servlet/CompanyDBSearch?page=detailed&cik=0001465570&main_back=2
03:13.21cyberfab007if you were looking
03:14.01drmessanowww.gigllc.com was more interesting
03:14.12drmessanowww.gigpllc.com was more interesting
03:14.14drmessanoMy bad
03:15.14cyberfab007HA i know the joomla template the are using LOL
03:15.59drmessanoGLOBAL INVESTMENT GROUP, LLC, owner of telenity.net, which points to your site?
03:16.20raden_work[Aug 31 22:16:04] WARNING[10802]: chan_sip.c:15268 handle_response_invite: Received response: "Forbidden" from '"103" <sip:tanning@69.179.99.17>;tag=as2f4e3703'
03:16.21cyberfab007if they do I dont know about it LOL
03:16.36drmessanoAre you serious?
03:16.46drmessanohttp://telenity.com/gf/
03:16.52*** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net)
03:17.24drmessanotelenity.com, rather
03:17.32drmessanowhich, BTW.. redirects to /gf
03:17.48raden_workhttp://pastebin.com/d47cb8712
03:17.49*** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net)
03:18.13drmessanocyberfab007: Have you ever used G726?
03:18.16cyberfab007LOL ,
03:18.29hescois it unusual for multiple calls to have the same channel ID?
03:18.41cyberfab007I get this VPS through ultrahosting , they must own this IP before or somthing lol
03:18.49KyleKhesco: at the same time? yup
03:18.58hescosequentially
03:19.19KyleKI've had some of that
03:19.28hescoI'm doing a series of tests and I keep getting the same Channel ID
03:19.51hescoI would have thought that each would be unique
03:20.09KyleKwell CDR(uniqueid) should be unique
03:20.33*** join/#asterisk zerko (i=zerko@srv1.techality.com)
03:20.49cyberfab007will only for very low bandwidth situations is has the backwards compatibility with asterisk 1.2 but I am no expert
03:21.11cyberfab007it is old codec from like 90 or 91
03:21.15cyberfab007one of the first
03:21.40KyleKDECT uses G726
03:21.54drmessanoActually, its not backwards compatible with 1.2.. The implementation in 1.4 and later changes, requiring a unique codec name
03:21.56KyleKso I'd think its use is on the uprise right now
03:21.59*** part/#asterisk mumtazah (n=mumtazah@203.82.91.103)
03:22.24drmessanoG726 in 1.2 != G726 in 1.4
03:22.56*** join/#asterisk yidiyuehan (n=yidiyueh@bb116-14-76-211.singnet.com.sg)
03:23.32cyberfab007yes you are right , it has NO back wards compatiblity in 1.2
03:23.47cyberfab007but is older codec
03:24.02ifluxI don't get why companies don't just spend the money on the g.729 license.. is it really THAT expensive?
03:24.47cyberfab007yes , it is when you get to a call load of like 30,000 and they want like 250,000 for a license fee ,
03:25.04*** join/#asterisk netpro25_ (n=mmanning@c-71-226-86-184.hsd1.fl.comcast.net)
03:25.06cyberfab007be different if you had milliions of customer but for start up  makes no sense
03:25.22zerkoanyone here have servers in dallas?
03:25.24cyberfab007hard to pitch to investors
03:25.43zerkoi have a few if anyone needs one
03:25.45netpro25_Anyone ever use Colopronto?
03:26.47cyberfab007drmessano pm me your skype you are obiousily the regular here
03:26.49ifluxsee.. it seems like you could build something so that g.729 is an addon module that is easily purchased by the customer.. basically enabled via a license key
03:27.06ifluxcyber: drmessano's first day here was like 3 days ago..
03:27.10cyberfab007g729 should be opensource ,
03:27.22cyberfab007your shitting me it is my first day he is knowlageable
03:27.23ifluxg729 is open source.. for um.. europe
03:27.31cyberfab007g729 should be opensource
03:27.36netpro25_lol
03:27.45ifluxcyber: yeah.. I've been here like 7 years and I've only seen him this last week
03:28.08KyleKcyberfab007: its patent encumbered til 2012 or 2014
03:28.09cyberfab007really , I have avoided this room for some years but I finally need some help
03:28.37cyberfab007its bull shit and another attempt by the power that be to be greedy and not publish GPL , will they ever get it?
03:28.40ifluxcyber: yeah.. drmessano is actually 3 guys.. you never know which you're talking to at the time..
03:28.50ifluxdrmessano: are you bill, rich, or harry tonight?
03:29.06cyberfab007I think he may be flake
03:29.46drmessanoiflux: LOL
03:30.24cyberfab007Kidding , goign for a smoke , brb
03:30.35drmessanocyberfab007: i am the flake?  Your company had one post in March about your "big ambitions, then nothing until 2 weeks ago, followed by spamming on every forum known to man.  You guys get out much?
03:30.59*** part/#asterisk SimplyZero (n=drewyate@pool-96-238-62-45.prvdri.fios.verizon.net)
03:31.04drmessanoLemme guess
03:31.23drmessanoRan out of quarters at Kinkos.. needed time to print the 20,000,000 shares?
03:31.36raden_work<PROTECTED>
03:31.36raden_work<PROTECTED>
03:31.36raden_work<PROTECTED>
03:31.39ifluxdrmessano: basically.. he made his savings throw..
03:32.01drmessanoD20000000?
03:32.27drmessanoWhat I dont understand is
03:32.57ifluxgoodnight all
03:33.12drmessanoIf hes going to code/steal/beg for help on developing Joomla and Vtiger integration with Asterisk, why even use the Elastix GUI?
03:33.19drmessanoNot welll thought out
03:33.22drmessanoNight iflux
03:34.21cyberfab007No that is paid for alreday
03:34.23cyberfab007lol
03:34.53zerkoanyone need a dedicated in the infomart? (dallas, tx)
03:35.26cyberfab007you will see new release in about a week that will be complete except vitger,joomla, and a2billing intergration ,
03:35.31cyberfab007I need no help with this
03:35.42cyberfab007What I want help on is another project
03:35.53cyberfab007one that is just planing ,
03:36.51cyberfab007could Asterisk be the engine that is the basis for one online community that runs government? maybe!
03:37.27raden_workexten => _*67.,1,SET(CALLERID(number)=0000000000)
03:37.27raden_workexten => _*67.,1,Dial(LOCAL/${EXTEN:3}@outbound)
03:37.31cyberfab007could asterisk lead the way in American Democracy 2.0?
03:37.31raden_workam i doing something wrong ?
03:37.45drmessanoyou will see new release in about a week that will be complete except vitger,joomla, and a2billing intergration ,  <-- Joomla, Vtiger are the entirety of what youve been spamming.. if those are not in there, then what the hell will be complete?
03:38.12cyberfab007well buddy , no one code extensions like that overnight
03:38.19cyberfab007you think I am GOD or somthing?
03:38.59drmessanoIf the integration of none of those items will be complete, what will this release be good for.. screenshots?
03:39.09cyberfab007first lets get the extension manging the elastix services , than we will set up the billing and CRM intergration as I said that project is funded and on its way
03:39.13raden_workis there a way to disable caller id with vitelity ?
03:39.21KyleKdamn i cant find the delcaration of ast_channel
03:39.23cyberfab007Not here for that
03:39.25yidiyuehanhi, anybody knows why if I originate two calls through asterisk manager interface, the second exten won't be called unless the first call has been answered?
03:39.38KyleKraden_work: its only setting it to your 866 number?
03:39.42drmessanoIf the integration of none of those items will be complete, what will this release be good for.. screenshots?
03:39.52netpro25_question about latency. Is tracert a good way to determine if I have a latency problem. Obviously I do since I am getting jitter.
03:39.56raden_workKyleK, no its always unavailable even when i dont touch it
03:40.36drmessanocyberfab007: what exactly will be in this next release if not the aforementioned features?
03:40.54KyleKodd
03:42.12raden_workhow do i make asterisk try to reregister a peer if that makes sense
03:42.21drmessanoI have this great idea.. its a search engine where instead of putting in keywords and it producing search results, you put in the search results and it produces the search terms.  I'm not sure if it's even useful, but we already hired the sales and marketing teams, so who gives a shit
03:42.27raden_workvitel-outbound now unreachable : (
03:42.50cyberfab007Well a customer in joomla will be able to purches packeges with a number of extensions, view and edit the extensions in the packeges , view the CRD's on the account , purches and manage trunks , purches and manage 50+ dids from didx.net and Have the Joomla admin set all this up in joomla in less than a hour
03:42.51zerkonetpro25_, do you see any delays in the traceroute?
03:43.20netpro25_Okay so when I tracert my sip providers server
03:43.23cyberfab007I will be wraping the ARI in this extension of course
03:43.36cyberfab007for extension based management
03:43.42netpro25_I get anywhere from 30 -90 ms
03:43.57drmessanoWy not just rewrite the ARI.. its crap, and I am sure the freePBX team would love a rewrite
03:44.23cyberfab007ahh but it works and I wiill not re-invent the wheel investors dont like that
03:44.31cyberfab007if it is there use it and move on ,
03:44.38cyberfab007ARI servers its purpose
03:44.39netpro25_zerko: http://pastebin.com/m6bce59bb
03:44.44drmessanoWhat happened to GPL democracy utopia?
03:44.57drmessanoOh, that shit only works if they're coding for you, right?
03:45.15netpro25_drmessano: hah
03:45.44netpro25_is this like a public board room or something
03:45.55raden_workKyleK, yeah caller id just came uo unavailable
03:45.56*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
03:45.58netpro25_where are the sharks?
03:46.30drmessanoYour model of a GPL democracy fails if the utopia only exists inside your ecosphere.. we call that a CULT, not a democracy
03:47.04cyberfab007You obviously have no clue what I am hinting at
03:47.29cyberfab007this is not a virtual world buddy get your head out of your ass , I am no virtual world freak
03:47.53drmessanoCode for shares in this so-called company, I get it
03:47.55cyberfab007I am talking a facebook dedicated to government powerd by asterisk for communications
03:48.02drmessanoPaid in hours of work, at the end of the month
03:48.39cyberfab007well if someone writes code the least least least I can do is give them shares , especially if paid positions are all filled up 10 of tehm
03:48.42cyberfab007them
03:49.12cyberfab007better shares than nothing , what other project is offering this?
03:49.15drmessanowhat is the estimated value of these shares?
03:49.30cyberfab007well right now about 75 cents
03:49.56cyberfab007I look to go on a market next april , maybe pinksheets
03:49.58drmessano20 million shares, 10 million dollar investment.. thats 50 cents an hour to code for you.. Thats cheaper than india
03:50.17cyberfab007its not about the money dude
03:50.25drmessanoOf course not
03:50.31drmessanoHow could it be?
03:50.56cyberfab007people who get paid by be as a job get 15 minum a hour and 30 max if your good
03:51.11*** join/#asterisk brian (n=brian@unaffiliated/brian)
03:51.17drmessanoand the others code for 50 cents an hour in their spare time?
03:51.59cyberfab007well if they are helping in the community hopfully there code will do well and my stock 3 years from now will be at 20-30 per share that would be nice for everyone
03:52.49cyberfab007it is gratitude to the community that I am doing this not because I want to attract people you have alot to learn my friend
03:53.05drmessanoHmm.. which would require the company to be valued at 40 million to 60 million?
03:54.05cyberfab007Well I hope I do well , if I have people eventually writing code for me for free the LEAST I can do is give them a part of the company , better than all the other project out there
03:54.51cyberfab007for instance I dont see kevin flemming or mark spencer giving us a part of Digium?
03:55.03cyberfab007for writing all this code do I ?
03:55.21cyberfab007What if they did ,
03:55.23drmessanoYeah, no shit.. Digium owes me like $18.92 for all the debugging I have done for Asterisk, at 50 cents an hour
03:55.48cyberfab007buddy read your shit right 50 shares an hour
03:56.07cyberfab007which mean if you give me 20 hours a month that 1000 share
03:56.33cyberfab007if my stock goes to 10 dollars two years form now that 10,000 dollars a month you just got
03:56.53cyberfab007I hope it goes to 40 or 50
03:57.10raden_workexten => _*67.,1,SET(CALLERID(number)=0000000000)
03:57.10raden_workexten => _*67.,1,Dial(LOCAL/${EXTEN:3}@to-callcentric)
03:57.14cyberfab007based on dividen of course you want investment package I send out ?
03:57.19cyberfab007ha you have no money
03:57.23raden_worktat wont even let me dial out :(
03:57.47cyberfab007minum but in is 13,000 dollars ,
03:58.28KyleKraden_work: 1
03:58.29KyleKthen 2
03:58.38drmessanoI did my math wrong.. For you to hit 20 to 30 dollars a share, that means you're looking a valuation of 400 million to 600 million
03:58.39KyleKexten => _*67.,2,Dial(LOCAL/${EXTEN:3}@to-callcentric)
03:58.53drmessanoand $10 a share would be $200 million
03:59.06raden_workKyleK, omfg i thought i had n there been working on this 2 long lol thanks lemee try
03:59.08cyberfab007or a dividend of about 40 - 60 cents a share
03:59.20KyleKraden_work: I occasionally do exten => 123,1,Noop() then exten => 123,n,etc
03:59.39cyberfab007as I said I have the high market cap because I plan on giving out lots of stock
03:59.48raden_workKyleK, what does that do ?
03:59.59KyleKnoothing operation
04:00.04*** join/#asterisk rizwank (n=rizwank@pool-71-118-51-238.lsanca.dsl-w.verizon.net)
04:00.21KyleKhttp://en.wikipedia.org/wiki/Noop standard term
04:00.46drmessanoWell, good luck.. I think your first order of business should be to find a developer, because is hans paperbag quit, and no one else is commiting any code, you're fucked
04:00.54rizwankHi there. I'm looking for a way to have Asterisk take an incoming SIP call, and connect it to the PTSN (so the call effectively contains some extra data with a DID, and when Asterisk gets it, it completes the call via the PTSN.) Is there a name for this feature?
04:01.09raden_workKyleK, works with vitelity now but not callcentric
04:02.10cyberfab007HA this week there will be lots of code commited , by 4 differnt developers , no worries
04:02.28drmessanoDude
04:03.09drmessanoIm not the one that should be worried.. you got a 10 million dollar venture cap investment by a company that includes in their south african operations, the mining of blood diamonds
04:03.13cyberfab007what you do sit here on IRC trying to demorilize people , ha you want pic of my 53 story penthouse over lookign the lake all because of asterisk ? Get a life buddy and stop trying to demorilize others
04:03.14drmessanoYOU should be worried
04:03.41*** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net)
04:03.52cyberfab007I have nothing to do with those peopel other than I got an IP address they used to have my friend
04:04.53cyberfab007Let me roll a joint now ,
04:05.31geneticxlol
04:06.31cyberfab007so anyone in here interested in having there own online asterisk business for free?  This is the extension I am making in joomla I need no help , only people to report bugs to my developers and test, but I am not here for that , I am here for asterisk and E Democract
04:06.34drmessanoInteresting
04:07.08drmessanoCEO of a multmilliondollar corporation talking about rolling joint on a logged IRC channel
04:07.10cyberfab007What is E- Democracy?
04:08.16cyberfab007Well thats the point socitey is wrong in wanting the fake professional face of a well seasoned politicant , my netwok accepts people for who they are , and in canada weed is accepted my friend
04:08.43cyberfab007I am not a mulitmillion dollar coproatino
04:08.47cyberfab007corporation
04:09.02cyberfab007jsut a guy with a few project and some cash thats all ,
04:09.02raden_workKyleK, thanks for eveything vitelity working at least I have to set my Call ID but thats ok :)
04:09.08drmessanoYou said you had a 10 million dollar investment.. did you smoke it already?
04:09.51manxpowerAt least tonight's drivel is different than most night's drivel.
04:10.19manxpowerStarting your own business is far overrated.
04:10.19cyberfab007I said a million that was last year
04:10.34rizwankhmm.
04:10.40*** join/#asterisk dongs (n=blogger@l212047.ppp.asahi-net.or.jp)
04:10.51dongshi, what setting in sip.conf controls "Expires: " header going out
04:10.58cyberfab007now I do a reg D offering that lets me rais another million between 133 investors dude get your shit stright , I feel I am wasting my fingers
04:11.08dongsi mean i would just grep the sores for this,but y'know, in 2009 i was hoping I didn't have to do that.
04:11.11manxpowerdongs: if it can be set, it would be listed in sip.conf.sample
04:11.26cyberfab007Dongs that cant be set I dont thin k
04:11.29dongs...
04:11.38dongswell on 1.4 its 3600 and on 1.6 its 120.
04:11.57*** part/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com)
04:12.06dongs1.4 can register to my silly ip provider, 1.6 cannot.
04:12.14cyberfab007dongs I have never used that before ,
04:12.16dongsso i'm going comparing 1:1 and seeing the differences in headers.
04:12.25manxpowerdongs: it might be a lot of work, but you should check UPGRADE*.txt and the changelog
04:12.29cyberfab007usally I set the device to determain that in the device config
04:15.23cyberfab007I think in asterisk 1.6 you may be able to define the expire time in sip.conf but do not quote me #:^)
04:15.25*** join/#asterisk errotan (n=errotan@62.201.122.164)
04:16.22cyberfab007http://lists.digium.com/pipermail/asterisk-dev/2003-September/001568.html
04:16.29cyberfab007search expire on the page ok
04:16.39dongs1.6 is missing "Event: registration"
04:17.42cyberfab007twords the bottem of that link it talks abou that
04:17.57cyberfab007We add a few extra features in each sip peer.
04:17.57cyberfab007;  expire = the number of seconds that this registration should
04:17.57cyberfab007;    indicate for expiry.  Default is 500.
04:18.42dongscan this go in global.
04:18.45cyberfab007Why is this so important , is this high security server or somthing ?
04:18.45dongsthis has nothign todo with peer.
04:19.15cyberfab007do you have 1000000 phone connectiing to this server and need to save resources?
04:19.25cyberfab007or somthing
04:19.34dongsnoidea, imguesingits some half-assed "lunix" solution and breaks at even slightest difference from what it expects.
04:20.14cyberfab007so why bother? download asterisknow or elastix or somthing?
04:20.22dongs... ?
04:20.35dongsi'mconnectingto provider's sip junk.
04:20.46cyberfab007ohh use vitlity man
04:21.00cyberfab007vitelity man they are best out there till i get my new site up !
04:21.15cyberfab007complete with elajoom LOL
04:21.33dongs??????????????????????????????????????????????
04:22.04cyberfab007people are so worried about saving their half a cent with someone else , just pay good rate for good service with no hassels
04:22.36dongslooks like im grepping the sores after all.
04:22.51cyberfab007yegh man
04:22.56drmessanohalf a cent is 33% in some cases
04:23.12cyberfab007yegh if your doing 100,000 minuets
04:23.16cyberfab007but your not
04:23.24cyberfab007so worrie when you have the volume man ,
04:23.32cyberfab007not now
04:23.51cyberfab007now you worrie on getting traffic
04:23.55drmessanoLast time I checked, a half cent difference between one provider and the other was still a half cent at 1 minute or higher
04:24.47*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
04:24.54cyberfab007ok , half a cent on 10000 minuets is 50 buckes man
04:25.02cyberfab007are you doing 10,000 minuets
04:25.20cyberfab007half a cent a minuet is 5 dollars on 1000 minuets
04:25.47cyberfab007if you worrie bout that this early in the game your screwed trust me your worries need be else ware
04:26.02*** join/#asterisk Tim_Toady (n=moi@adsl242-72.kln.forthnet.gr)
04:26.08drmessanoSo paying 30% more for anything is ok, because its stupid to worry about it?
04:26.13dongswhy 1.6 doesn't print Event: registration?
04:26.15cyberfab007no
04:26.19dongsin initial header
04:26.26cyberfab007you have no pie ,
04:26.34cyberfab00733% of no pie is nothing
04:26.55cyberfab007when you have pie of 100,000 minuets a month than worrie about 33% till than forget it
04:27.48drmessanoI'll tell that to my mother in law when she asks me why her phone cost her 60 a month instead of 40.. "Fuck you bitch, youre not doing 100,000 minutes here"
04:28.01cyberfab007LOL
04:28.16*** join/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com)
04:28.33cyberfab007I do like 750,000 a month , in long distance
04:28.40cyberfab007minuts that is
04:29.22cyberfab007maybe 60% is us the rest is international
04:29.50cyberfab007it is about 4k a month in my pocket out of my canadian company
04:29.55drmessanoI average about 5000 a month until around December.. then hit about 3,000,000 a month for December
04:30.12drmessanoSanta gets a lot of calls
04:30.20rizwankHi there. I'm looking for a way to have Asterisk take an incoming SIP call, and connect it to the PTSN (so the call effectively contains some extra data with a DID, and when Asterisk gets it, it completes the call via the PTSN.) Is there a name for this feature?
04:30.30cyberfab007as I said  get the pie first that worrie about how much your makign off it
04:31.26drmessanoI thought your business was Elastix plugins
04:31.29KyleKrizwank: being a service provider?
04:31.30manxpowerrizwank: Accepting a call and sending the call back out is a basic feature of Asterisk.  I dunno about "extra data".
04:32.15KyleKrizwank: whats the application? are you trying to redirect rizwank@example.net to your cellphone?
04:32.22rizwankI effectively want to be able to deliver SIP calls via my land line.
04:32.40rizwankNot sure of the best way of doing that.
04:32.42manxpowerrizwank: standard stuff.  That's what a PBX does.
04:32.45KyleK"ip terminiation"
04:32.51KyleKbrb
04:32.53cyberfab007http://www.doingitwrong.com/wrong/20070515-003247.jpg
04:33.36manxpowerrizwank: Asterisk is a toolkit for building a business phone system.  It's not simple or easy.
04:34.00rizwankmanx - wanted to make sure it was doable, and some terms for that feature set so I can do more research
04:34.09drmessanohttp://neoavatara.com/blog/wp-content/uploads/2009/04/l2_46730.jpg
04:34.11cyberfab007no its not and anyone who has tried to write thier own dial plan will tell you that
04:34.25rizwankand I didn't know if Asterisk interfaced with analog modems or not.
04:34.32manxpowerrizwank: It's such a basic features that it doesn't have a name.  It's like asking for the name of the feature to describe a car moving.
04:34.42cyberfab007LOL
04:34.59manxpowerrizwank: it does not interface with analog modems.  Access to analog phone lines requires a hardware card designed for that.
04:35.01manxpower~boot
04:35.02infobotrumour has it, boot is what you get when you act like a DalNet user, or #debian-boot
04:35.03manxpower~book
04:35.04infobot[~book] 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://astbook.asteriskdocs.org or see ~buybook
04:35.07cyberfab007dude anlog modems is what we used before digium cards man
04:35.14rizwank=)
04:35.15manxpowerrizwank: read the first part of the book.
04:35.58rizwankcheck.
04:36.04rizwankThank you.
04:36.18rizwanktrying to find out about delivering calling card calls via POTS
04:36.33rizwankso I wasn't show how the destination number was processed and dialed manually.
04:36.38cyberfab007a2billing ma n
04:36.41cyberfab007are you a noob
04:36.41rizwankdidn't realize it was built in.
04:36.44rizwankyes.
04:36.53cyberfab007it is like bread and water  man
04:37.02rizwankwe've got our own billing platform already...
04:37.10rizwankbut I see your point =)
04:37.17manxpowerThe book will give yo a good inro
04:37.44rizwankthanks. FXO cards interface with POTS lines, so that's what we're looking at hardware wise...
04:37.48rizwankdownloading it now.
04:38.26manxpower~fxofxs
04:38.26infobotfrom memory, fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it.
04:39.10*** join/#asterisk jjshoe_ (i=jjshoe@75.85.173.90)
04:39.11*** join/#asterisk d00gster (n=doughant@77.31.106.234)
04:39.43cyberfab007Ok peoples , going to bed , good first nigh in the room here , wish I did not have to spend it arguing with Dr. Fag boy aka mulitple personality boy
04:40.06cyberfab007see yall in the morning EST
04:40.34jjshoe_if you're arguing with someone you're doing it wrong
04:40.39jjshoe_use your client's ignore feature
04:41.03manxpowerJoel: I wish my client had such a feature.  Tonight was the first time in a while that I missed that feature.
04:41.34cyberfab007jjshoe your right , but I give people much more often than I should the benifit of the doubt
04:41.57cyberfab007everyone had their opinon and I shoudl deal with it
04:42.03manxpowerBut if nutjobs like cyberfab007 keep coming around I may just switch to a client that has that feature.
04:42.42cyberfab007nutjobs sound like somthing your mom does
04:43.15rizwankthanks.
04:44.40*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
04:44.59drmessanoDr Fag Boy?  Multiple Personality boy?
04:45.18drmessanoName calling.. you're a keeper
04:45.41drmessanoI'll have you know, I have no personality and I don't smoke
04:45.53drmessanoIf you can't be smart, be accurate
04:48.03cyberfab007member:iflux
04:48.03cyberfab007cyber: yeah.. member:drmessano is actually 3 guys.. you never know which you're talking to at the time..
04:48.03cyberfab007member:ifluxx member:drmessano: are you bill, rich, or harry tonighy
04:48.18cyberfab007Haha
04:48.48drmessanoApparently you're as dumb as I figured
04:48.59cyberfab007so are you bill rich or harry tonight ,
04:49.03drmessanoiflux is my cousin
04:49.18cyberfab007Yegh waiting for that confirm
04:49.25drmessano[23:30] <iflux> think I had him going
04:49.30drmessano:(
04:49.42drmessanoDont believe everything you read on IRC, n00b
04:51.09cyberfab007well what ever ,
04:51.13drmessanoTime to hop in the Bentley and run to Taco Bell.. I will grab a Chalupa and throw it at your window on the way by, cyberfab007.. Still the 3rd basement window on the left, right behind your moms car?
04:51.24cyberfab007you guys have a good night ,
04:51.44cyberfab007ha I live in a million dollars condo , with a view of the lake
04:51.46drmessanoYou too sweetheart
04:52.08cyberfab00753 story
04:53.30*** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net)
04:54.14cyberfab007well 53 floor 47 stories
04:55.26drmessanoPalace Pier?
04:55.29*** join/#asterisk frk2 (n=faraz@zivios/member/fkhan)
04:55.50Joelmanxpower it helps if people arn't retarded and would just simply not engage the idiots
04:56.12cyberfab007Montage
04:56.17cyberfab007City place
04:56.36*** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-rahfpmsqbfuitiif)
04:56.42cyberfab007all bought and paid for by Asterisk !
04:57.12cyberfab007http://www.cityplace.ca/montage_le/
04:58.27cyberfab007I have the south west penthouse , 2000 squear feet of niceness
04:59.08Zuchmir2which ATA is descent that handles multiple SIP accounts (ideally also multiple FXS ports)?
04:59.19drmessanoToo bad about the real estate market I guess
04:59.55drmessanoAll those listings for condos at the Montage for 200k - 500k
05:00.09drmessanoYawn
05:00.15drmessanoGoogle harder
05:01.27cyberfab007Yegh live in your trailer
05:01.36cyberfab007goodnight yall
05:02.43*** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker)
05:03.57cyberfab007I finally ignor that guy
05:04.13cyberfab007should have done it sonner but had a few frinks tongiht
05:04.15drmessanoscratches his crotch, burps, takes a swig of bud light and slaps the wall of his trailer..
05:04.51cyberfab007drinks , totaly distracted me from what I came her for tonight what a ass
05:05.11Tim_Toadylol, wtf
05:05.37drmessanoTim_Toady: You can code.. cyberfab007 can hook you up
05:06.15cyberfab007Tim_Today ever have someone like that ? I dont chat much in IRN
05:06.17cyberfab007IRC
05:06.44Tim_Toadylike u? nah
05:06.59drmessanofrinks his fanta
05:07.16drmessanosells that ad campaign
05:07.46drmessanoMy goal in life is to pressure coke into making Watermelon Fanta
05:08.32drmessanoThen move next door to cyberfab007
05:08.42Tim_Toadylol
05:08.42drmessanoSo we can be asterisk buddies
05:08.46dongsok why am i getting ;rport added at end of Via:
05:09.23dongsnat=never
05:09.32*** join/#asterisk maour (n=gnu@unaffiliated/maour)
05:09.35drmessanoI honestly have no idea why Windows comes with any disk utilities at all
05:09.41drmessanoThey dont work
05:09.48dongsno, you just don';t know how to use htem
05:09.51cyberfab007Tim what you mean like me , I let that guy totally drag me in , what a drunk Iwas tonight to take me off topic
05:09.55drmessanolol
05:10.09drmessanoI'm well aware how to use them
05:12.27dongswhy in the hell is default asterisk build option -g3???
05:12.52drmessanoM$ needs to hire the guy who wrote volrepair for netware and put him in charge of chkdsk
05:12.54dongsdo you enjoy linking 40megs ofdebug info into 4 megs binaryor something?
05:14.03dongsanyone?
05:14.15dongsmanxpower: what is the reason for default compile option for asterisk being -O6 -g3?
05:14.32dongsespecially in tarballs designed for end-users?
05:14.53KyleKwhich tarball is this?
05:15.00dongsanything off asterisk.org.
05:15.37cyberfab007http://support.sas.com/documentation/onlinedoc/sasc/doc/changes/z0090339.htm
05:15.49dongswhat.
05:16.44dongsLOL got it to work after removing ;rport= shit
05:17.06cyberfab007fucking with you lol
05:17.13KyleKhrm
05:17.38cyberfab007it is old IBM mainframe options , confuses the shit out of people but it drove the answer out of ya hugh?
05:18.01drmessanoHe called you Hugh, frank
05:18.04dongsfunny how someuseless option has made it in by default, justto support "buggy firmware" on some random uniden phone nobody uses.
05:20.18dongscyberfab007: i'm not sure what your purpose in this chat is.
05:20.33dongsso far you've been nothing but an annoying self-righteous stuck up piece of opensores shit.
05:20.43dongs(but thats what a typical opensores chat is filled with, so im hardly surprised).
05:21.44dongsmeanwhile im waiting for 'ld' to figure out how to link 40MiB*30 .o files linked with -g3 into a 40meg asterisk.bin because apparently turning off -g3 for release builds shipped to customers (lol, customers in opensores, rite) was too hard.
05:22.09*** join/#asterisk miloux (n=KVIrc@milu.rit.se)
05:23.05Tim_Toadydongs ure compiling on a 486 or smth?
05:23.11dongsnope.
05:25.10dongsk blank ;rport= washte problem that was confusing the remote side.
05:25.13cyberfab007dongs that sound like some complicated stuff , I am not familiar with to be honest
05:26.05KyleKhrm -g3 -O6 -O0
05:26.12KyleKyea that is odd
05:26.15drmessanoHe's a coder, but doesn't understand compiler options.. move along
05:27.19cyberfab007cyberfab007 is a bit offended by the use of "opensourses shit"
05:27.51drmessanoor "opensores".. which ruins his allusion
05:32.32*** join/#asterisk oej (n=olle@ns.webway.se)
05:37.07Zuchmir2which ATA is descent that handles multiple SIP accounts (ideally also multiple FXS ports)?
05:37.19drmessanoHow many ports?
05:37.26Zuchmir22
05:37.39drmessanoLinksys PAP2-T
05:38.22Zuchmir2does that support multi accounts per FXS?
05:38.48drmessanoThat supports 1 account per each FXS.. it has 2
05:39.20[TK]D-FenderCheckout time, later all
05:39.42drmessanoI dont think you'll find a device that supports multiple accounts per port
05:39.49Zuchmir2ok, i need 2 SIP accounts, 1 for incoming, 1 for outgoing, and both use the same 1 acct for out
05:40.14drmessanoif youre using asterisk, you can do this in the dialplan
05:40.23Zuchmir2i thought i read somewhere there's a device with multi accounts
05:40.46drmessanoATA's are not meant to be that smart.. make it work in Asterisk, which is easy
05:40.54Zuchmir2i know the Prestige 2302R can do it
05:41.18*** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc)
05:41.21Zuchmir2but am having trouble with it
05:41.21drmessanoIm guessing youre not even using asterisk
05:41.51Zuchmir2i have an asterisk server, but it keeps losing reg, and CID doesnt display name
05:42.40hescoI'm writing a REST application which interacts with Asterisk.  sometimes its methods are called by apache, sometimes by asterisk through agi scripts embedded in the dial plan.  Asterisk is running as root.  It seems completely incapable of unlinking call files from the staging queue putting a copy of them into the outbound spool.  Can't fugure out why, though.
05:45.11Zuchmir2hmm, the PAP2-T has GPL code, does that mean i would be able to mod this thing to add multi ports (or is this only partial GPL)
05:46.18hescops aux says asterisk is running as root.
05:46.27drmessanoNo, you can't mod it in that way
05:46.43*** join/#asterisk thansen (n=thansen@76.27.110.194)
05:46.54hescoI assume that means that the agi scripts called from the dialplan will run as root as well.  Does that make sense?
05:47.27cyberfab007well I am gonna shoot some people on xbox live before I go to bed, I have lots of meetings tomorrow. Let me leave you people with one thing. OpenSource is our existence,  never forget that almost like a god father was stallman. You have the dreamers and workers. Which are you? Which ever you are , we need both, as success-full as linux has become there are much larger things in the future. Asterisk could be
05:47.28cyberfab007Engine which drives the future , I have been working on this design since I was about 17 years old about the time asterisk came around, an asterisk box in every living room that is free, opensource, user owned with encrypted data, has really been the goal since the beginning, Well imagine asterisk doing more than that, running AI in households via voice recognition, social networking, mass private telecom
05:47.28cyberfab007networks over a neutral net, Self government. All of this requires the flexibility of the advanced communications technology that asterisk offer. For christ sake you can run advanced c and php scripts right from the asterisk dial plan those commands can be determined by voice recognition which can lead way to a whole variety of smart-house technology and make it an appliance in every home. Self government, h
05:47.33cyberfab007Asterisk can already provide advanced communications , but can we make that communications smart? Combine it with social networking technology? Let people governthem selfs with the help of communicaitnos technology? Well my investors expect this of my company...
05:47.41KyleKomg spam
05:47.46*** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl)
05:47.46cyberfab007nope
05:47.49cyberfab007worse
05:48.00cyberfab007drunken ramble , from the heart !
05:48.10cyberfab007xbox time , you people think about that
05:48.27*** join/#asterisk ming_zym (n=ming_zym@124.127.101.0)
05:49.26cyberfab007cool link
05:49.38drmessanohttp://www.xbox360updates.com/uploaded_images/Xbox-360-Updates-729048.jpg <--- One can only hope
05:49.44cyberfab007http://areyouadavinci.com/
05:49.53cyberfab007I bet 9/10 of you are
05:51.57drmessanoI scored 100%.. I smell shenanigans.. I've always been a C+ or B- kinda guy
05:52.20cyberfab007a stunning silence
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05:54.15drmessanoHmm.. the Davinci Method.. But if I already have the Davinci Code.. do I need this?
05:54.19drmessanoDoubtful
05:54.33KyleKdraw lots of pretty pictures?
05:55.15drmessanoI'm gonna write a book called the Davinci Cheat Code
05:56.02drmessanoU, U, D, D, L, R, L, R, B, A, B, A, Start <--- The secret to success in life
05:57.08dongsoh you're kidding.
05:57.18dongsdid make clean just wipe my menuconfig settings.
05:57.38dongswhat  the fuck people. clean removes OBJECT CODE ONLY.
05:57.44dongsdistclean = including configured settings.
05:58.32dongslunix. its all about surprises. you don't know what's gonna fuck you over next.
06:03.12KyleKpost some bugs
06:03.18dongsno point.
06:03.25dongsnobody reads bug reports.
06:03.32dongsit took them over 6 years to implement sip session timers.
06:04.18dongsinfact ifi remember correctly (6 years and all), the bug i opened on that was closed with "wont fix, nobody fucking uses that shit".
06:05.03drmessanoebay is selling skype
06:05.07drmessanoYay
06:05.14KyleKhow much
06:05.48drmessanoNot officially announced yet, will be today
06:06.12dongsgood riddance.
06:06.55drmessanoRiddance?
06:08.01drmessanoSkype will be much better off without ebay
06:08.37KyleKhopefully they sell it for less than 4 billion
06:08.53KyleKI'll be pissed off if marvel is worth less than skype :)
06:08.57drmessanoEbay has been shopping it out at 2 billion
06:10.06drmessanoWho knows.. Guess I need to wait for Scott Fulton to proclaim Skype is dead over on Betanews around lunchtime
06:10.31drmessanoUntil then, i am out.. GOODBYE KIND PEOPLE
06:10.39*** join/#asterisk TimToady_ (n=moi@77.49.29.202)
06:10.41drmessanoFIGHT THE GOOD FIGHT
06:11.01drmessanoI R NOT DRUNK BUT TEH OPEN SOURCE IS TEH OPEN DOMOCRASY
06:11.19drmessanoU R SIP, I R SIP.. WE R ALL SIPS
06:11.27drmessanoI SIP U, U SIP ME
06:11.52KyleK:)
06:12.01TimToady_i missed something good eh?
06:12.06KyleKI just ignore the channel when teh crazies re in
06:12.31drmessano01:47
06:12.34drmessanoET
06:12.45drmessanoCheck the log on riker's site
06:12.47drmessanoGood stuff
06:12.52drmessanoAnyway
06:13.10drmessanoSIP AWAY MAH GUD PEOPLE.. COMMUNISTS USE THE H323!
06:13.21cyberfab007crazies are we , what do you do ? what is your life style like?
06:13.22drmessanoDONUT BE COMUNISM
06:13.46*** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net)
06:14.17drmessano(Hes all yours KyleK.. Just make sure if you cant get the hook from his mouth, you keep smashing his head with the stick.. he'll let go)
06:14.17cyberfab007I live sex, drugs , music , code, engineering , lushness, I live this everyday
06:14.42cyberfab007am I crazy , most say yes , am I sucessfull, most to to much ,
06:14.42drmessanoWUT R COMPILER FLAGS?
06:14.50drmessano\\\\GONE
06:15.51cyberfab007I thought University was a joke , I was to busy making money I got by with my C
06:20.11cyberfab007crazys hugh ? crazyones.org
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06:37.46hescops aux says asterisk is running as root.
06:37.47hescoI assume that means that the agi scripts called from the dialplan will run as root as well.  Does that make sense?
06:38.07hesco$> and $< seem to indicate that is the case.
06:38.36hescoHowever, my AGI script seems unable to remove a file in the queue.
06:38.46hescoAny ideas why that might be?
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06:42.37KyleKwhats the error its getting?
06:43.49hescoI'm not seeing any error messages, only a deleted file persisting in the pending queue directory
06:44.06j_kroonhttps://issues.asterisk.org/view.php?id=14577 - what's my options to get a fix for this?
06:44.06KyleKwhat lang is the agi
06:44.33hescoperl
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06:49.05hescoI've been using print $FH to build the call file. File::Util->write_file to copy it over to the spool; and unlink to clean up behind myself.  but the unlink has not been cooperating.
06:49.58hescoI just tried rename $queue/$call_file, $spool/$call_file.  and that did not seem to work either.
06:52.37KyleKwhy File::Util->write_file instead of open write close?
06:54.07KyleKwell i guess open; print $handle; close
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06:57.47KyleKfor rename is $queue on the same drive as $spool?
06:58.03KyleKwell filesystem actually
06:58.10hescosame drive, different partition
06:58.26hescolatest attempt is: rename "$queue/$userid/$call_file", $spool/$call_file;
06:58.34KyleKmove the queue
06:59.02hescoperl's `mv` is rename
06:59.55KyleKmove is like this
07:00.08Doctehunless its on a different FS
07:00.11Doctehthen it does this
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07:01.09KyleK:)
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07:04.23KyleKbut anyways if thats a callfile and you want it to work properly the queue and the spool directory have to be on the same file system
07:04.39KyleKor you'll want to copy to spool/.. and then rename it
07:05.33hescothe queue is at /tmp/app_name/queue/userid/  and the spool is at /var/spool/asterisk/outbound/
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07:06.52KyleKis that the same filesystem?
07:09.34hescoyes, though distinct partitions
07:10.22hescomy /tmp is on /dev/hda2; while my /var is at: /dev/hda6
07:10.28KyleKuhh
07:10.53KyleKfilesystems sit in partitions which sit in drives
07:11.16hescook, so I guess they are different file systems
07:11.31hescoI though a single filesystem was rooted at /.
07:11.36hescothought
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07:12.01KyleKwell in windows each drive is a file system
07:12.21hescofstab says they are all reiserfs
07:12.22KyleKbut you CAN mount into a drive such that C:\windows is E:
07:12.30cjkhi, does anyone know an app for nokia phones that is able to call a http url from the contacts. in order to develop click2dial for nokia
07:12.38KyleKhesco: so they're both like ntfs
07:12.50hescohow so?
07:13.05KyleKI'm throwing windows terminology at you
07:13.17KyleKI figure it might help you get it :)
07:13.56hescoreiserfs is an open source journaled filesystem, ntfs is proprietary so no one really knows what it is, or at least that is what I've been told.
07:14.07KyleKwell yea
07:14.22KyleKwell what are you better at
07:14.26KyleKWindows or Linux?
07:14.27hescowindows terminology is most likely to only confuse me
07:14.42hescohaven't owned a windows box since 2002 or so
07:15.15hescothough I launched a windows instance in AWS' EC2 last night to test an install of an app I have to support the end of this week
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07:15.50KyleKalrighty so i wont try throwing windows terminology at you
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07:16.07hescoI'm pretty easily confused at this hour as it is
07:17.00hescoso do you think this might work better if I moved my queue to the partition with /var on it?
07:17.06KyleKyea
07:17.19hescook, I'll try that then
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07:28.20hescook, now it is really broken.  And I'm feeling the need for sleep.
07:29.05KyleKhopefully its not in production then ;)
07:31.16hescono, on my test bed.
07:31.45hescoI just did an svn revert on the broken module and restored it to its previous state of brokenness
07:32.21hescoso it now generates the call, but fails to remove the call file from the queue.
07:33.34hescoI guess less broekn is better.
07:33.38hescobroken
07:34.00hescook, enough then, to bed with me.
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08:54.52tzafrir_laptopZX81, this starfish-pbx appears interesting at first glance. Sadly they use a non OSI-compliant license for their software
08:55.11ZX81yeah saw that
08:55.36syntheticsudo apt-cahce search headers
08:55.36ZX81they say it's Open Source, but after reading the license I'd say it's pretty much a no-go
08:55.44syntheticerr oops
08:55.48ZX81:D
08:55.56tzafrir_laptopsynthetic, apt-cache can be run as non-root
08:55.57syntheticdamn pidgin doesnt scroll
08:56.12syntheticthought iw as helping someone
08:56.19syntheticmy typing doesnt help people
08:57.16ZX81man I wish Adium wouldn't read out the friggin time of a message if I'm online - I just want it to read me the text of messages so I can watch TV and be contactable at the same time :D
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08:59.48ZX81I've update the news article to point out non compliance just in case people don't check first
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09:08.08Guggeanyone know if a feature like this one was ever integrated, https://issues.asterisk.org/view.php?id=7771
09:08.54Guggethe patch seems to apply cleanly, and using SIPCHANINFO(ruri,field) works fine ... but i would rather use a unpatched asterisk :)
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09:17.20KyleK~mysql
09:17.21infobotSQL (Structured Query Language) database server. URL: http://www.mysql.com/
09:17.27KyleK~cdrmysql
09:17.36KyleKeh i'll ask tomorrow
09:17.49ZX81~areski
09:17.55ZX81~cdrstats
09:18.00ZX81meh
09:18.01ZX81:)
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09:29.55KyleKoh its in addons
09:30.14ZX81yeps
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10:29.30garymcHi anyone know how to setup the polycom ip330?
10:29.46garymcjust got 2 of them, one seems to have booted itself? and another is just sitting there
10:30.24kaldemarmine don't boot themselves unless i tell them to.
10:30.48garymcyeah weird hey
10:31.02garymcso i get the phone no instructions like default passwords etc
10:32.48kaldemarPolycom:456 iirc
10:33.35kaldemarhttp://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip330_320.html
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10:37.35ValenHeres a question, I have a number of users and they often play "musical desks" I was wondering if there was a "good" way for them to be able to login to a phone such that their extension number will follow them around the system?
10:44.47garymckalemar: I cant seem to find the default passwords etc in those guides you gave me the link too
10:47.29Sandheavergarymc: the default passwd for the phone is 456.  If you config via the phone's web interface, the user is Polycom (with a captial P) and the password is 456
10:47.38Sandheaverthat stuff is in the docs
10:47.47kaldemaryou didn't do much lookin then. it's in the admin guide, and i told the default username and password a few lines above already.
10:47.59garymcwell i didnt get any docs with it, only putting the parts together doc
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10:48.13garymcright ok i didnt know what that was
10:48.15garymcsorry
10:48.30kaldemari have you the link to the guides. use them.
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11:07.55garymci have to say the docs are not very clear :(
11:08.02garymcnot to me anyways
11:08.39garymcalso as soon as I go to dial extension 1000, the phone only lets me dial 10 and says the number is not in use
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11:16.57AndyMLso, with our old nortel we recieved the CALLERID(name) from the telco, but now it never seems to make it into the system. Here are my dahdi configs. http://pastie.org/601524 - i even tried sendcalleridafter=1 but that seems to be for outbound.
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11:36.21garymcwhen im just setting up 2 polycom phones to test, so i can call extension to extension? Do i need to do this bootfile stuff?
11:40.15*** part/#asterisk Valen (n=jake@ppp226-148.static.internode.on.net)
11:40.21dongsso uh
11:40.22kaldemarextensions and boot files have nothing to do with each other
11:41.49garymcright, well ive setup the phones, they got an extension name on them, but they are not registering with freepbx (which im asking about in #freepbx) also the time is just flashing, cant seem to get that to work
11:42.29garymcso im not sure if its my asterisk/freepbx setup not letting them work or the phones
11:42.30AndyMLi've added a Wait(1) before it processes the call and now I'm getting the name in asterisk but it isn't getting sent properly to another device connected via PRI.
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11:47.15Naikrovekgarymc: you have to tell the phones themselves which extension they are, and that's done via configuration file
11:47.25Naikrovekor via the web interface
11:47.31garymcyeah i done that
11:47.52garymctheres that many diff options though, not sure ive done all i need too
11:48.15Naikrovekalso, you have to set up the Digitmap, so it knows what kind of dial pattern you're going to give it.
11:48.35Naikrovekif it's completing the call before you've finished dialing, that's the digitmap
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11:49.07garymcwhere can i find an explanation on that?
11:49.13dongsi got provider sip -> asterisk -> sip/pstn adapter x2 -> analog to ISDN converter, and connecting ISDN out from that to a keysystem. In 1.4, caller ID worked fine -> cid from provider was passed over to voip adapter, which in turn passed to ISDN. but now for some reason all callerid-aware devices I tried show the phone # of voip adapter, regardless of who's calling. what cahnged?
11:49.18garymcI can see my digitmap
11:49.24Naikrovekgarymc: just use mine: 0T|011xxx.T|xxxxxxxxT|[2-9]xxxxxxxxxxT|[1-9]xxxxT|xxxT
11:49.43Naikroveki need to fix mine, actually
11:49.45Naikrovekbut that works
11:49.45dongsi tried doing the obvious of like Set(CALLERID(num)) or whatever, no change.
11:49.54dongs${CALLERID(num)} shows correct number.
11:50.52Naikrovekdongs: then use that?  I'm not really familiar with caller id
11:51.10dongsi am, its not passing it along to the sip/pstn adapter like it used to.
11:51.15dongsor it does, but wrong way from 1.4.
11:52.03Naikrovekyou're using 1.6 now?
11:52.05dongsyeah.
11:52.11Naikrovekthings are different in 1.6
11:52.17leifmadsenNaikrovek: how so? :)
11:52.22dongsdifferent to the pointof breaking?
11:52.36Naikrovekyes, dongs.  that's why it's a different version number
11:52.44Naikrovekwhy do people never see that coming
11:52.55dongsum
11:53.00dongscaller ID is a basic function.
11:53.09garymcNaikrovek: giving that digitmap a try
11:53.13Naikrovekyes, but how it is handled changed from 1.4 to 1.6
11:53.23leifmadsendongs: you'd have to provide some sort of console output and relevant configuration output in a pastebin in order for people to give you anything other than vague answers
11:53.43leifmadsenNaikrovek: in which way?
11:53.54Naikrovekleifmadsen: in lots of ways, you know this
11:54.10garymcNaikrovec: that digitmap only lets me dial a single digit
11:54.25Naikrovekgarymc: then google a better one.  it works perfectly for me
11:54.29Naikrovekand i'm using a polycom phone too
11:54.40leifmadsenNaikrovek: I don't know this -- I'm curious to know what changed because I've obviously missed that part
11:54.42Naikrovekand you ahve to keep dialing or it'll assume you're done
11:54.56leifmadsen(in terms of CallerID specifically)
11:55.40Naikrovekleifmadsen: i don't know specifically, but people come in here every day with callerid issues and they have all upgraded from 1.4 to 1.6.  I use trixbox so i'm not much help with that
11:55.43dongswell, i see what the problem is.
11:55.44dongsFrom: "realnumberthatscalling" <sip:siptopstnadapternumber@lanip>;tag=as1c8d2f42
11:55.55dongsof course that'll never work.
11:56.03leifmadsenSIP?
11:56.07dongsyes.
11:56.14leifmadsendon't rely on the From headers -- use RPID
11:56.25leifmadsentrustrpid=yes, sendrpid=yes
11:56.25dongssorry, tell that to the device manufacturer.
11:56.34leifmadsenget a better device manufacturer
11:56.38dongsnice try.
11:56.39Naikrovekdongs: get some things up on pastebin and wait for [tk]d-fender to show up.  the VERY FIRST question he'll ask is for some pastebin links.  he is smart and can help you
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11:57.32leifmadsengive a snarky answer, get one back
11:59.10dongsleifmadsen: i bet your answer is same reason broken code for adding a improperly-formatted ;rport= atthe end of every SIP invite was kept in release version of code intended for end-users, right?
11:59.36dongs'work around bugs in <some uniden softphone nobody ever heard of'
11:59.43leifmadsenI'm just telling you what works for me on every device I've used -- I've not used every single device
11:59.52dongs(but break everything else meanwhile, i mean, nobody cares right.
12:01.49Naikrovekdongs: you're the one that upgraded without testing.  i'm not saying that things should have changed, but you can't just upgrade and assume everything will continue to work exactly as before
12:02.06Naikrovekfeatures come and go, some things get deprecated
12:02.15dongsi didnt upgrade anything. i retired the box and rewrote configurations implementing changes needed for 1.6
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12:02.41Naikrovekthere was a guy in here the other day complaining that 1.6 remove the ability to use | as a parameter separator in extensions.conf.  he had to change everything
12:03.13verywisemanwhat hardware i need to run openbts with asterisk?
12:03.21Naikrovekidles
12:06.13garymcanyone tell me why my phones are not registering?
12:08.09garymcI check my phone status, it says not registetred. I look in Freepbx it says no phones Reged! HELP!
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12:09.07leifmadsengarymc: what information have you given that doesn't require wild guesses?
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12:09.35garymc??
12:09.57leifmadsengarymc: console output, sip debug, relevant configurations?
12:10.25garymcim using FREEPBX, not sure how to get thoses things
12:10.43leifmadsenIt's like me telling my mechanic that my car isn't working, and not telling him what part is causing me issues
12:10.58kaldemargarymc: go ask in #freepbx then.
12:11.00leifmadsengarymc: ahh -- you may want to try #freepbx then
12:11.08garymcim in freepbx too
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12:11.47garymcmust have to wait a couple of hours for people to come in or something
12:11.47leifmadsenmost people here just use vanilla asterisk, thus can't help you with your GUI based system
12:11.51garymcMaybe youd know wy im getting Cronmanager encounted 1 errors
12:11.54leifmadsenif you can ssh into the box, you can get into the asterisk console via 'asterisk -r' and then run various console based commands to get output
12:12.01garymcCould not reload FOP server
12:12.06leifmadsenhas no idea
12:12.11garymcok
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12:17.42leifmadsen[TK]D-Fender: I don't want to meet your Mom! I just want...
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12:19.31verywisemanwhat hardware i need to run openbts with asterisk?
12:19.34[TK]D-Fenderleifmadsen: ! ! !
12:19.41leifmadsen\o/
12:20.08leifmadsenverywiseman: not sure -- the openbts website seems to have the information for getting it configured -- I had never even heard of it till this morning
12:20.18leifmadsenthere was even a youtube video
12:20.51Chainsawverywiseman: Mostly, you need a spectrum license or a dummy load.
12:21.50[TK]D-Fenderverywiseman: http://www.kestrelsp.com/OpenBTS.html <- go ask them
12:22.13AndyMLhas anyone solved the hum/buzz issue with polycom ip phones and plantronics headsets?
12:22.40EiNSTeiN_verywiseman: I suggest you watch the talks that Harald Welte gave at 25C3 and HAR2009
12:22.48[TK]D-FenderAndyML: I've seen this when your power supply is flakey
12:22.54[TK]D-FenderAndyML: NO OTHER TIME
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12:23.59AndyML[TK]D-Fender: so like, PoE probably solves it out of hand? I'm seeing it on every brand new IP 331 with new Polycom power supplies right out of the box all over this company (~45 phones)
12:24.45[TK]D-FenderAndyML: Well I don't know if you've been messing with gains, or if you're running an amp if you have it on the right position, etc
12:25.21[TK]D-FenderanyRunning 45 polycom phones on bricks instead of PoE was definitly a mistake :)
12:25.44dongsallright
12:25.50dongslemmereblog my question for [TK]D-Fender
12:26.02dongsi got provider sip -> asterisk -> sip/pstn adapter x2 -> analog to ISDN converter, and connecting ISDN out from that to a keysystem. In 1.4, caller ID worked fine -> cid from provider was passed over to voip adapter, which in turn passed to ISDN. but now for some reason all callerid-aware devices I tried show the phone # of voip adapter, regardless of who's calling. what changed?
12:26.44dongsobviously i'd like the phone number of hte caller from incoming sip connection to be passed to pstn/isdn.
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12:27.10AndyML[TK]D-Fender: I've moved the headset tx gains down 3db to no effect, and we're using headsets plugged right into the polycom's, using their internal amplifier. the customer was given the option of replacing all their switches with PoE which they turned down. They don't have enough network drops to simply add-on. (Every phone's switch port is in use.)
12:27.12dongs${CALLERID(num)} shows correctthing.
12:27.26[TK]D-Fenderdongs: pastebin the complete call attempt with SIP debug and include your sip.conf
12:27.51[TK]D-FenderAndyML: they could replace their core switch
12:27.52dongsor you can just tell me what to look for , since i'm not spending 30 minutes cutting out my provider call info out of it
12:28.04dongsthat'll save YOU 30 minutes oflooking through stuff.
12:28.38AndyML[TK]D-Fender: agreed. they decided against it. I will bring it up as a solution to this problem and ask again if they're willing to replace it.
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12:29.35[TK]D-FenderAndyML: I'd test it first.  Isolate a phone as well and put it behind a UPS and see if the power conditioner clears up the hum
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12:30.53dongs[TK]D-Fender: so, what changed in cid handling that this is happening?
12:31.34[TK]D-Fenderdongs: Nothing I can think of, and I'm not going to start randomly pointing fingers.  Show the debug & configs
12:32.02dongsi insist neither are relevant to answer the question.
12:32.19[TK]D-Fenderdongs: Bullshit.  They guy who doesn't look doesn't find the problem.
12:32.35dongs[TK]D-Fender: i'mstill waiting for your advice on exactly WHAT to look for.
12:32.49[TK]D-Fenderdongs: Insist all you want, I've worked with asterisk from before it hit 1.0
12:33.09dongs[TK]D-Fender: glad to see in 2009 they finally added sip session timers.
12:33.14[TK]D-Fender[08:27]<[TK]D-Fender>dongs: pastebin the complete call attempt with SIP debug and include your sip.conf <- I already told you
12:33.17dongsyou might remember me asking for htem in 2001.
12:34.05[TK]D-Fenderbrb
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12:43.08dongslol. it was 'fromuser'.
12:43.39*** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.162)
12:43.43dongswhat fucking ridiculous documentation. first, change username to 2 possible (and insane) choices. then make it totally fucking unclear WHY that was done, and leave no explanation to use one or another.
12:43.54dongs[TK]D-Fender: problem was 'fromuser' in sip/pstn device config.
12:44.29[TK]D-Fenderdongs: Yes, that will force the CID to that of fromuser unless you use sendrpid=yes and trustrpid=yes and they support those headers
12:44.43[TK]D-Fenderdongs: Which I'd have pointed out the moment I'd have seen it
12:44.46dongs[TK]D-Fender: how about adding  that to the documentation.
12:44.59dongsso that you wouldnt have to deal with "dumb" questions like these.
12:45.09[TK]D-Fenderdongs: Cool it already
12:45.09dongs;fromuser=yourusername            ; Many SIP providers require this!
12:45.12dongsthis is super useful!
12:45.32[TK]D-Fenderdongs: Yes, and many providers also don't allow you to set your callerid.  Should we document this as well?
12:45.38dongsi mean anyone converting old username= crap to newstuff wouldjustgo and change this, cuz u know?? most providers requirethis, let's add it and thenforget what it was for.
12:45.43dongshuh.
12:45.57dongsthis is callerID from asterisk to sip/pstn adapter.
12:46.01dongsnothingtodo with my provider.
12:46.07dongsasterisk originates hte call.
12:46.14[TK]D-Fenderdongs: and username is in place of just using the section header name etc
12:46.29dongsusername = auth username.
12:46.30[TK]D-Fenderdongs: SIP is SIP.  Everything depends on what you're talking to
12:46.37dongsnow you have defaultuser and fromuser.
12:46.46dongshow does that makesense without proper docs.
12:47.18dongsoh i have another rant. i had to edit sores of chan_sip and recompile removing that retarded ;rport= hack
12:47.47dongsbecause while it was *probably* possible to disable it by some combination of nat=whatever, it was easierto just take it out completely.
12:47.59[TK]D-Fenderdongs: </monologue>
12:48.08dongswhy something that only affects ONE random device was left in release code to be used by users?
12:48.20dongs(breaking random number of other devices in hte process)
12:48.21[TK]D-Fenderdongs funny, we all seem to get by just fine
12:48.43dongsblank ;rport= crashed my provider's sip proxy with 400/bad request.
12:48.44*** join/#asterisk afink (n=afink@204.26.87.226)
12:49.24[TK]D-FenderHe's got a might weak proxy then :)
12:49.25dongsnow you can whine about "get a better provider" all you want, and that would be completely irrelevant.
12:49.34*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
12:49.35oejdongs: Part of the problem was that everyone misunderstood "username=" and did not get that it was used together with defaultip=
12:49.45oejIt was a poorly named configuration option from start
12:49.55oejusername= has nothing to do with the name of the device at all.
12:50.03[TK]D-Fenderoej: The man!
12:50.15afinkmorning everyone,  I am trying to get streaming moh working and having a hard time getting it to work.  I have been reading this and I have tried quite a few of the examples on here http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf and haven't got anything to work.  Anyone have a working example they can show me?
12:50.21dongsoej: both defalutuser and fromuser are just as poor. wouldnt something making sense like "authuser" be better , especailly properly documented.
12:50.35oejYes, that's something I would like to add.
12:50.44oejI hate config options with multiple uses
12:50.50*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
12:50.55oejfromuser is separate from username/defaultuser
12:51.05oejfromuser and fromdomain belongs together to form the From: header
12:51.13kaldemarmultiple different comments for them in samples are just as fun.
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12:51.22[TK]D-Fender\o/
12:51.22dongsuhhuh
12:51.37oejso now we have "defaultuser" and "defaultip" on one side and "fromuser" and "fromdomain" on the other so you see which belongs together.
12:51.43oejWe're still missing authuser I believe
12:51.44kaldemarthe best one in sip.conf.sample is "Many SIP providers require this!"
12:51.59oejauthuser is best set with the realm auth config
12:52.03dongskaldemar: thats why i added it in my shit when i was moving the settings over!
12:52.12dongsi was like omg, most require, i must add it!
12:52.20kaldemartwice!
12:53.16[TK]D-Fenders/must/many.
12:53.32[TK]D-FenderKeep on writing between the lines...
12:55.06afinkdrr, wrong path to mpg123
12:58.30eliasphi
12:59.02*** join/#asterisk dysinger_ (n=tim@71-20-35-99.war.clearwire-wmx.net)
12:59.18eliaspi have this extensions.conf: http://pastebin.com/ffadebdf  incoming calls on the mISDN trunk are now rejected with this message:
12:59.19eliaspWARNING[13643] pbx.c: Channel 'mISDN/3-u0' sent into invalid extension '955327180' in context 'default', but no invalid handler
12:59.31eliasphas anyone a hint what i did wrong/were to look at?
12:59.38eliasps/were/where/g
12:59.58leifmadsenexten = o,1,  <-- definitely wrong
13:00.29leifmadseneliasp: basically you're sending the call from MISDN into the default context, which doesn't contain a pattern match, or that number, which matches
13:00.42leifmadsenand you have no invalid handler (i extension) which means you get that message
13:01.02leifmadsensend it to a context that does match
13:01.06eliaspah, ok...
13:01.12[TK]D-Fendereliasp: And you have no exten to match that number
13:01.19[TK]D-Fendereliasp: just like it says.
13:01.27leifmadsenin the dialplan you provided, nothing would match that
13:01.37leifmadsendon't confuse 's' with a catch all extension either, because it isn't
13:01.59leifmadsenactually nevermind -- I just looked again, and there are a couple contexts it would match on
13:02.02eliaspyes, and 's' is on the SIP trunk anyways...
13:02.04leifmadsensend it to one of those
13:02.40leifmadsenruns off for... a run
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13:25.47kindyroothello, I am a new (happy?) user of asterisknow, I have everything working
13:26.04kindyrootbut I don't know how to make a sip call from point A to point B
13:26.10kindyrootin my local network
13:26.16kindyrootcan anyone help?
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13:29.06fiddurkindyroot: try #asterisknow
13:29.25[TK]D-Fenderkindyroot: #freepbx <-
13:31.43eliasphmm i'm quite new to Asterisk + dialplan ... just trying to understand all this while reconfiguring an existing plain asterisk setup... i'm still confused by all these terms in the dialplan... 2 questions: what does DID mean?  and where in a dialplan is the actual "send call with extension FOOO to context BAR" done? in a "exten =" line and the following command?
13:32.16afink~book @ eliasp
13:32.22afink~book
13:32.23infobot[~book] 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://astbook.asteriskdocs.org or see ~buybook
13:32.32kindyrootfiddur: thanks
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13:37.08eliaspafink: just reeding the freely available PDF version... looks nice.. let's see how far i come...
13:37.36dongs[Sep  2 01:42:22] NOTICE[874]: chan_sip.c:18520 handle_request_invite: Call from 'siptopstnadapter' to extension 'numberimcalling' rejected because extension not found.
13:37.37dongsnow what.
13:37.45dongsthey're both in same contexts.
13:37.46dongsetc.
13:37.49dongswhat am i missing? (1.6)
13:38.10Hatrixhello, i could compile the debian package before, but today i get:
13:38.10Hatrix<PROTECTED>
13:38.15kaldemarsiptopstnadapter's conext is something else
13:38.20Hatrixwhat's wrong? this is driving me nuts!!!
13:38.52dongskaldemar: rly, i just checked and they're all in 'sip' context.
13:39.42[TK]D-Fenderdongs: Clearly you don't have an exten to match that number in the context its looking in
13:39.46kaldemarhow about "i don't believe you until you show it"?
13:40.01dongs[TK]D-Fender: printing WHAT context it looking in would be super.
13:40.04[TK]D-Fenderkaldemar: You learn quickly Padawan :D
13:40.05oejdongs: run "sip show peer" on siptopstnadapter
13:40.15oejTo check the context it receives calls in
13:40.16dongskk
13:40.17*** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca)
13:40.22dongsits not reciving
13:40.24dongsoriginating
13:40.26[TK]D-FenderdongsYes, and like many years before it shows up when you pay attention to SIP DEBUG
13:40.46kaldemar[TK]D-Fender: quickly? lack of cluebat has made me this slow.
13:40.55Skeeter-anyone knows how to setup call routing with a distinctive ring??
13:40.57dongsContext      : sip
13:41.17[TK]D-Fenderreaches for his ClueBat (tm)
13:41.49[TK]D-Fenderdongs silngle lines like that are meaningless.  You continue you insist you must be right.  Well if you did everything perfectly it'd work.  You aren't looking
13:41.54Maliuta[TK]D-Fender: the one with the big spikes and H.E. tips?
13:42.14dongs[TK]D-Fender: sip debug obviously does not show anything about searching for extensions.
13:42.16[TK]D-FenderMaliutaNo, straight ironwood :)
13:42.26[TK]D-Fenderdongs: Yes it does.
13:43.04[TK]D-FenderSkeeter-: depends on the phone
13:43.09dongsLooking for numberimcalling in sip (domain mydomain)
13:43.21dongsdid _ become some different meaningin 1.6?
13:43.25[TK]D-Fenderdongs : I'm not seeing debug & configs...
13:43.36[TK]D-Fenderdongs: No, and you continue to deflect
13:43.37Skeeter-Fender: wanna make it work for the whole system
13:43.52Skeeter-Fender: inbound distinctive ring
13:43.59[TK]D-FenderSkeeter-: Where is the distintive ring coming from?
13:44.05[TK]D-FenderSkeeter-: Inbound over what?
13:44.45Skeeter-Fender: 2 land lines, 3 phone number, 1 of the phone number is using a distinctive ring, and cannot be process
13:44.59[TK]D-FenderSkeeter-: what is it coming IN on?
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13:45.24Skeeter-Fender: can you be more specific
13:46.13[TK]D-FenderSkeeter-: What kind of friggen wire, plugged into what HARDWARE?
13:47.02dongsokay this is ridiculous.
13:47.21dongsextension IS in [sip] and i even got rid of _ and just replaced with 0
13:47.25dongsstill same shit, not found.
13:47.35Skeeter-Fender: regular land phoen wire, into the sangoma card, which is into the asterisk server
13:47.37dongsand itis lookingin sip context for it.
13:47.53[TK]D-Fenderdongs: And you're still not showing the backup
13:48.00Skeeter-Fender: i want 2 phones lines, to work on 1 zap
13:48.22dongs[TK]D-Fender: what backup?
13:48.28*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
13:48.28*** mode/#asterisk [+o russellb] by ChanServ
13:48.30[TK]D-FenderSkeeter-: if its ANALOG, then go look on the WIKI for "distinctive ringing"
13:48.37[TK]D-Fenderdongs: SIP Debug & dialplan
13:49.11dongsi did that.
13:49.25dongsalready pasted relevant parts.
13:49.25eliaspwhat does ${EXTEN:0} or ${EXTEN:5} mean? i know what ${EXTEN} is, but i can't figure out, what the :$NUMBER means.....
13:49.28*** join/#asterisk s14ck (n=s14ck@190-76-92-153.dyn.movilnet.com.ve)
13:49.34dongseliasp: first n chars of extensioncut off.
13:49.48*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
13:49.50*** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com)
13:49.57eliaspdongs: aaah, this explains a lot.. thx!
13:50.14dongs[TK]D-Fender: Looking for numberimdialing in sip (domain mydomain)
13:50.16[TK]D-Fendereliasp: Chop off X digits
13:50.17dongsthis is in sip debug.
13:50.18dongsgetting 404.
13:50.25[TK]D-Fenderdongs: PASTEBIN <-
13:50.31dongsohfucksake.
13:50.47[TK]D-Fenderdongs: You are still showing only worthless tiny bits and masking what little could be of any value at all in them
13:51.07dongshttp://bcas.tv/paste/results/cxBInl92.html
13:51.11dongshappy now.
13:51.24*** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file)
13:51.24*** mode/#asterisk [+o file] by ChanServ
13:51.34dongsor do you want to know what audio format the call was palced in?
13:51.44[TK]D-Fenderdongs: I don't see the number or your fucking dialplan
13:51.52[TK]D-Fenderdongs: You are showing precisely jack shit and being a whiny bitch about it.  This is simple dialplan
13:52.03dongsprecisely why I expect it to work.
13:52.07[TK]D-Fenderdong and hasn't changed in a DECADE
13:52.19beekis surprised that [TK] hasn't exploded yet.
13:52.31[TK]D-Fenderdongs: Now stop being a dumbass and PASTEBIN the actual call debug and your friggen dialplan
13:52.41dongshttp://bcas.tv/paste/results/auLIGh53.html
13:52.45beekthinks it's getting close.
13:52.46dongs^
13:52.57dongsvery simple.
13:53.05eliaspwee, got it working... i think i finally got it, how the dialplan works ;-)
13:53.09eliaspthx a lot guys
13:53.24[TK]D-Fenderdongs: Again you're wasteing time masking everything
13:53.36dongsi didnt mask anything.
13:53.46[TK]D-Fenderdongs:  numberimcalling
13:53.48[TK]D-FenderO RLY
13:53.51dongsyes. its a number
13:53.52dongsfixed.
13:54.02dongsit had a _ before it
13:54.11dongsbecause my fucking sip/pstn adapter requires 0 beforeitdials out
13:54.17dongsso i replaced that with 0.
13:54.20[TK]D-Fenderdongs: Stop masking things and show your actual code, I can tell you are hacking this shit up at every turn
13:54.26[TK]D-Fenderdongs: Stop being a moron
13:55.08kaldemar_ doesn't match a digit or character. it tells asterisk that the extension is a pattern.
13:55.17Maliuta[TK]D-Fender: I think that's like telling it not to breathe ... assuming it's actually alive
13:55.18Katty:<
13:55.32[TK]D-Fenderindeed doesn't match anything, only indicates that what follows is a pattern
13:55.33Maliutawaves at Katty
13:55.52Kattyhi :<
13:56.56MaliutaKatty: 'sup?
13:57.03Kattystuck in toshiba meeting :<
13:57.37beekKatty: Are you being punished for something?  ;-)
13:57.42MaliutaKatty: could be worse ... it could be Dell ;)
13:57.45Kattysometimes i wonder.
13:58.04Kattywe don't sell asterisk based systems anymore
13:58.36dongskaldemar: thx, works.
13:59.09beekKatty: that's unfortunate.
13:59.24Kattywe also sell Samsung
13:59.29beekSamslug
13:59.35Kattywe're whores.
13:59.44Katty:<
13:59.47beekWhy no *?
13:59.53Kattyno one wants it
13:59.57Kattythey don't recognize the Brand
14:00.37beekThat's depressing.
14:00.41Kattyyes.
14:00.58*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
14:01.14beekI actually just installed one at a site where the president of the company said "why are we replacing one proprietary piece of shit with another proprietary piece of shit."  They have an Asterisk box now and are quite happy with it.
14:02.19[TK]D-Fenderdongs: So, got something real to show for a change?  Or have you found your mistake?
14:03.24Kattybeek: well. i still use asterisk here for our phone system. we still have people that use it. i have one at home.
14:03.42Kattybeek: just more stuff for the resume.
14:03.51beekGotta look at the bright side...
14:04.01Kattynods
14:04.14Maliutaconsiders bed
14:04.17[TK]D-Fenderbeek: "WARNING : Do not look into the laser with your remaining good eye"
14:04.34beekGood advice!
14:06.11Skeeter-how can i see dring in the Asterisk CLI
14:06.21Skeeter-what verbose i should use
14:06.40Kattyi always use 10
14:08.44Kattythis toshiba guy is all sales and no tech :<
14:09.40*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4df0:d9ce:287a:92ea)
14:10.17Skeeter-still cant see it
14:10.27Skeeter-must be some : show .... command
14:10.39[TK]D-FenderSkeeter-: show us you've configured it right
14:11.04beekKatty: Have some fun with him and ask him some nice techie questions and make him squirm.
14:11.13Kattyseriously considering it.
14:11.45beekYou may as well enjoy yourself and it's some much better when it's at someone else's expense.
14:12.03Kattyyeah but there are 3 sales reps in here with me
14:12.18*** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com)
14:12.18Kattyit'd probably cause drama
14:12.21beekyou really are being punished
14:14.34*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
14:14.37Skeeter-http://pastebin.com/m56696506
14:15.07Skeeter-i just need to add the dring=
14:15.21Skeeter-with the 3 numbers, but i cant find those numbers in the CLI
14:15.51[TK]D-FenderSkeeter-: well you HAVEN"T done it.  You certain won't see anything unless * is told to look for it
14:16.02[TK]D-FenderSkeeter-: you aren't going to see NUMBERS there
14:16.24Kattywell. it could be worse. it oculd be 65F in here.
14:16.45Skeeter-Fender, the wiki says to look at the CLI to find the dring number, how am i suppose to find them then
14:17.30*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
14:17.49*** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1887:6a0:2ac0:762d)
14:17.53cuscohello folks
14:18.55*** join/#asterisk maour (n=gnu@unaffiliated/maour)
14:19.07Kattyhello
14:19.09[TK]D-FenderSkeeter-: core set debug 10
14:19.46cuscowe recently changed our service, clients calling us now have to press 1 if they have client nr and access code
14:20.00cuscoseveral things happening:
14:20.30cuscowhen we press 1, sometimes jumps straight to asking access code, not asking client nr
14:21.01cuscosometimes when we press keys, they are not interpreted as a dftm? not taken by the system
14:21.20dongswell looks like after some cockups my 1.4->1.6 migration is ok.
14:21.35Kattycusco: have you considered hiring a consultant to have a look at the system for you?
14:21.39dongsnothanks to [TK]D-Fender and some thanks to kaldemar ^_^
14:21.51[TK]D-Fenderdongs: Yeah you got what you gave.
14:22.04dongsnow i just gotta make a call longer than 5 mins to see if session timer shit actually works.
14:22.15[TK]D-Fenderdongs: And we know the mistake was on your end, and no doubt something embarassingly simple
14:22.28[TK]D-Fenderdongs: But glad you found it
14:23.20dongs[TK]D-Fender: ? i was assuming _ = 0 and thus it was looking for -1 sized extension and notfindingit. i forgotthat particular fixed-number extension was for a long-unused softphone that didn't require 0 before dialing.
14:23.32*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
14:23.32*** mode/#asterisk [+o putnopvut] by ChanServ
14:23.40dongsstupid? yes.
14:23.46[TK]D-Fenderdongs: Yup, embarassingly simple..
14:23.49dongserrors should have been more helpful? yes.
14:23.57[TK]D-Fender~assume
14:23.58infobotAbout assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight&quote=assume.txt&file=assume.wav  It makes an (ass) out of (u) and (me)
14:24.01[TK]D-Fender^^^^
14:24.15dongshaving to turn on debug and sift through potentially hundreds of lines to track down a misnumbered extension  = lol
14:24.20dongsbut iguess you get what you pay for wiht opensource.
14:24.33Kattybe happy there's a log to dig through (=
14:24.37[TK]D-Fenderdongs: the error WAS extremely clear.  You didn't have a match.  It won't GUESS that you should have done something in one exten differently.  It isn't PSYCHIC
14:24.59[TK]D-Fenderdongs: it told you the number and the context.  It is completely your own fault for not knowing dialplan 101
14:25.04dongs[TK]D-Fender: im dialing 0123456
14:25.11dongsits printing "cannot find extension 0123456'
14:25.15[TK]D-Fenderdongs: And never has "_" meant "0"
14:25.16dongsnot very helpful.
14:25.41cuscoKatty: I guess something is wrong with dftm, I would liek somebody to point me in the right direction
14:25.44*** join/#asterisk Take (i=take@nerd.fi)
14:25.58[TK]D-Fenderdongs: It is helpful.  means you should look to see if you have an exten to match.  Only thing the basic debug didn't say was the context.  Then again your exten was wrong regardless
14:26.08*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
14:26.19TakeHello.
14:26.29[TK]D-Fenderdongs: But you made a big fuss and masked shit left right and center and whined
14:26.39Kattycusco: alright. fair enough (= I would be patient tho. this room is filled with volunteers.
14:26.52TakeI tried (again) to search via google if asterisk could be used as PoC PTT -server (used on Nokia phones atleast), but I didn't find an definitive answer.
14:26.54Kattycusco: it may take some time to get an answer.
14:27.06TakeDoes anyone know anything about the issue and if it can be done?
14:27.19*** join/#asterisk manxpower (n=EWieling@69.73.94.162)
14:27.57*** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel)
14:28.05*** join/#asterisk friartuck (n=pmccary@66.162.90.56)
14:28.31KattyTake: i don't know. perhaps someone else does tho.
14:29.10TakeKatty: there's several conversations available and everyone has heard that it can be done, but no references :-/
14:29.12Kobazhttp://www.reuters.com/article/newsOne/idUSTRE57U02B20090831
14:29.29KattyTake: i would recommend being patient and waiting to see if anyone has an answer.
14:29.44TakeKatty: I'm not going anywhere ;)
14:29.46Skeeter-FEnder: how can i see the debug log after that
14:29.48Katty(=
14:29.49leifmadsenI am!
14:29.50*** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:29.56Kattylol.
14:30.59*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
14:31.00*** mode/#asterisk [+o Deeewayne] by ChanServ
14:31.00[TK]D-FenderSkeeter-: that is CLI DEBUG
14:31.06dongsKobaz: i just saw that on the news this evening.
14:31.32dongsgood thing I got my hybrid before the chinks figured it out.
14:31.47Kattyhi Deeewayne!
14:32.05Deeewaynehugs Katty
14:32.07Deeewaynehello :-)
14:32.13Kattyhugs on Deeewayne
14:32.58Deeewaynerough morning.... I had to turn around halfway to work to rush to the dentist to hog tie my kids (3 & 5) on the dentist's bench
14:33.00Skeeter-Fender: then it didnt work
14:33.06[TK]D-Fenderdongs: careful on your wording there... racial commentary like that can send you skidding out of here.
14:33.16KattyDeeewayne: eesha :<
14:33.40KattyDeeewayne: teeth cleaning?
14:33.52Skeeter-http://pastebin.com/m6a61d36b
14:33.55Deeewayneyes.  no cavities :-)
14:34.04Kattyexcellent. those are never fun.
14:35.23KattyDeeewayne: your kids would probably like the new strawberry colgate.
14:35.36Kattythey also carry watermellon now too (=
14:35.57Deeewaynethey had bubblegum today
14:36.04Deeewaynebubblegum toothpaste, that is
14:36.14Kattytrust me, the strawberry is much much better!
14:36.16Kattymm!
14:37.10*** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136)
14:37.44cuscohow come DTMF's that arrive trough the primary channel, may be missinterpreted by Asterisk?
14:38.38*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
14:38.43Kattyhey Naikrovek (=
14:38.58Naikrovekhola
14:39.30Kattyscowls.
14:39.46Kattyi asked the toshiba rep a tech question, and one of my company's sales reps tried to answer me.
14:39.49Kattyhow insulting.
14:39.57jaytee"Wildfires are a normal part of nature, cleaning out underbrush from forest floors, allowing more room for the larger flame-retardant trees (like sequoias) to grow. These Los Angeles fires are no different, getting rid of undesirable two family houses and dense suburban areas so large celebrity mansions can take their place."
14:40.17Kattylol
14:40.33beekmorning jaytee
14:40.40jayteemorning beek
14:40.54beekKatty: try again -- and this time ask your company's sales rep to politely STFU
14:41.00*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
14:41.03Kattythat would cause drama.
14:41.15Kattyand i have to work with these people all day.
14:41.19beekThat would make a boring day interesting.
14:41.21manxpowerAsk his boss when that person started working for Toshiba
14:41.38beekmanxpower: nice touch
14:41.39Kattyher boss, is sitting right in front of me.
14:41.44Kattythis is a very small company.
14:41.53Kattythe drama would come from both.
14:41.56Kattytrust me.
14:42.04manxpowerKatty: in that case I think poison is the solution of choice.
14:42.11Kattymanxpower: i am forced to agree.
14:42.24Kattyto tell you how bad it is, i had to make a public and a private facebook
14:42.31Kattybecause drama spawned from my political preferences.
14:42.41manxpowerKatty: sounds like it's time to find a good place to work.
14:42.50Kattysighs.
14:42.56Kattyi keep thinking the same thing.
14:44.01*** join/#asterisk dmz (n=dmz@244.sub-75-209-248.myvzw.com)
14:44.08cuscolol @ "Female voice are known to once in a while trigger the recognition of a DTMF tone."
14:44.09Kattyreally it's not too bad. it's just one sales rep here that causes drama.
14:44.28*** join/#asterisk wwalker (n=wwalker@72.249.1.66)
14:44.38Kattybecause if she feels she's been Wronged, she'll go tell everyone in the office and drag the owner into it.
14:45.38wwalkeranyone have a reference to the actual speed one can expect from asterisk for originate requests through AMI?  I seem to be getting an average of about 10 calls/sec with maybe 18 calls/sec peak.
14:45.39TakeKatty: properly used laxatives after drama queen -acts work as a great lesson ;)
14:45.42jayteeif someone doesn't like my political preferences they are free to fuck themselves and die at any time, in the workplace or not.
14:46.12*** join/#asterisk Jymm (i=jim@unaffiliated/jymmm)
14:46.30Kattyjaytee: i agree.
14:46.35*** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426)
14:46.40Kattyjaytee: everyone here is right winged conservative.
14:47.07Kattyjaytee: and i disagree on somethings, obviously, but i don't appreciate 5 different people trying to lecture me on my political views and how it may adversely affect the company's Reputation
14:47.18jayteeugh, I hate those people. they all smell of death, decay and having drank the blood of small children
14:48.02Kattyour customers don't need to know what my political preferences are.
14:48.05Kattynor do my co-workers.
14:48.26Kattyso i just kicked everyone from work off my facebook, and put them on another one. if i'd just dumped them completely there'd be drama.
14:48.34Kattywhy did you dump $coworker? did they upset you? blahblahblah
14:49.02JymmIn X-lite, all unidentifed calls are going directly to VM, any suggestion? I can't seem to find the filter to allow them.
14:49.51dongs<Katty> our customers don't need to know what my political preferences are. < neither does anyone on irc. plz blog this crap either in privmsg to someone who cares (nobody) or off-line somewehre.
14:50.23*** join/#asterisk k4tanaLINUX (n=k4tanaLI@190.196.70.189)
14:50.51dongsi thought in 2009 opensores channels would be past getting a hard-on the moment someone wiht a female /whois joins a channel. i guess not.
14:51.00dongsbbl
14:51.12*** part/#asterisk dongs (n=blogger@l212047.ppp.asahi-net.or.jp)
14:51.52Jymm"dongs" is a femine nick, aint it?
14:52.29*** join/#asterisk netpro25_ (n=mmanning@c-71-226-86-184.hsd1.fl.comcast.net)
14:52.32Kattyi'm guessing they're having a bad day.
14:53.07manxpowerA lot of nut cases have been joining this channel recently.
14:53.14Kattyit's okay.
14:53.20Kattythey just want help.
14:53.23eppigyello
14:53.23JymmMaxxed: Hey now, I resemble that!
14:53.25eppigyI am dave
14:53.31Kattyhello, sir!
14:53.32Kattylet's hug.
14:53.32jayteeTRABAJO!
14:53.45Kattyhugs on eppigy
14:53.48Jymmmanxpower: Hey now, I resemble that!
14:53.58eppigyhuggles Katty
14:54.05k4tanaLINUXhi people ... somebody know  how to convert mp3 or wav in g729 ?
14:54.15eppigyi passed my CCNA :D
14:54.23jayteecongrats!
14:54.29theharcaps meatpaws
14:54.31theharclap
14:54.33hescoI'm having an issue with an AGI script described here: http://perlmonks.com/?node_id=792672  Any ideas?
14:54.34Jymmeppigy: totally useless cert
14:54.49Jymmeppigy: Kidding, congrats!
14:55.10Jymmeppigy: So, how many times did it take?
14:56.56JymmMy old job would pay up to 5 times, after that, you were on your own
14:56.56hescok4tanaLINUX: there is a command line tool that handles most audio conversions, not sure if it does that one or not.  sox and flac do many conversions.
14:57.32manxpowerk4tanaLINUX: you will need commercial software to convert anything to/from g729.  You can do it in Asterisk 1.6, but you will need a g729 license ($10/per channel) from digium.  For just conversion, you should only need 1 channel.
14:57.57Jymm~softphone
14:57.58infobot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
14:58.04*** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint)
14:58.11Jymmdoh
14:58.21k4tanaLINUX~asterisk
14:58.22infoboti heard asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall
14:58.25*** join/#asterisk |Cybex| (n=John@80.100.126.176)
14:58.30k4tanaLINUX:)
14:59.24Jymmnobody uses xlite?
14:59.31*** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com)
15:00.09k4tanaLINUX~a2billing
15:00.27k4tanaLINUXno description
15:02.21k4tanaLINUX€1500 for MOR ..... :(   a little expensive for me
15:02.32manxpower~manxpower
15:02.32infobotManxPower has been using Asterisk in production since late 2001.  Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX.  http://www.nyigc.com
15:02.52*** join/#asterisk momelod (n=smelo@CPE001f3a8fe859-CM0012c91df0bc.cpe.net.cable.rogers.com)
15:03.17Jymmmanxpower: a lil self promotion there, huh?
15:04.01*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
15:04.07manxpowersome people seem to think I work for Digium
15:04.25Jymmmanxpower: Ah
15:04.39giovanianyone have CNAM providers to recommend?
15:04.53giovanicnam.info hasn't been accepting customers for a while now
15:05.13jayteeit's the long hooded cloak and the sheleighly that confuse people
15:05.49Jymman irish monk?
15:05.58Skeeter-http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DetectingDistinctiveRingonIncomingCalls
15:06.03Skeeter-this is clearly not wokring
15:06.09Skeeter-cant find the numbers
15:06.18*** join/#asterisk theHub (n=theHub@69.177.93.21)
15:06.34*** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.162)
15:06.43JymmEbay dumps skype... http://money.cnn.com/2009/09/01/technology/ebay_skype/index.htm?postversion=2009090108
15:06.56Kattyhi fender.
15:07.15momelodgreetings channel
15:07.23[TK]D-FenderJymm: and plenty of people use X-Lite
15:07.53momelodhow do i list my configured trunks from the CLI?
15:08.02manxpowerSkype: The AOL of VoIP
15:08.16[TK]D-Fendermomelod: SIP show peers
15:08.44*** join/#asterisk Naikrovek (n=jjohnson@63.252.251.77)
15:09.46manxpowermomelod: "zap show channels"
15:09.52momelod[TK]D-Fender: sorry i should have been more clear.  im trying to setup a te122 card and i want to see if the channels are referred to as Zap/* or DAHDI/*
15:10.49*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
15:10.49momelodmanxpower: right, but 'dahdi show channels' gives me the same output
15:11.00[TK]D-Fendermomelod: either can work if you configured it as such
15:11.02*** join/#asterisk xpot-mobile (n=james@mx0.synergyconsultant.net)
15:11.12momelodWhen i dial in, i see the channel in use is Zap/1-1 and these calls work.  But when i dial out, i see DAHDI/g1 and i get an all circuits are busy message.  Now when i edit the trunks, in the trunk identifier i put in g1 but how do i force Zap over DAHDI
15:11.13manxpowermomelod: then the only way to know is to know what you have installed.  zaptel or dahdi
15:11.36[TK]D-Fendermomelod: that message is BS.
15:11.41momelodim using dahdi
15:11.54momelodbut im trying to get it working with freepbx
15:12.00[TK]D-Fendermomelod: Go look to see if chan_dahdi.so is even loaded, and pastebin "dahdi show channels"
15:12.05momelodand i set it to run in zap compatability mode
15:12.47manxpowermomelod: I wish you the best of luck
15:12.50manxpower~freepbx
15:12.51infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
15:14.23momelod[TK]D-Fender: http://pastebin.ca/1550385
15:14.52[TK]D-Fendermomelod: ok, looks fine so far
15:15.10momelodyeah, incoming is working.. just having problems with outgoing
15:16.08[TK]D-Fendermomelod: And I don't see the problem
15:16.16*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
15:16.52Naikroveksomeone say something
15:16.57[TK]D-Fendersomething
15:16.59Naikrovektesting my network
15:17.00Jymmwoof
15:17.01Naikrovekperfect :)
15:17.12eppigyJymm: 1
15:17.18[TK]D-Fender~areyouadog ?
15:17.19infobotBark! Bark!
15:17.22[TK]D-Fender~botsnack
15:17.22infobot[TK]D-Fender: :)
15:17.27Naikrovekfinally got his Cisco router ACLs fixed up
15:17.31[TK]D-Fenderinfobot: Good Boy!
15:17.31infobot[TK]D-Fender: aw, gee
15:17.36Kattyprods eppigy
15:17.39momelod[TK]D-Fender: that makes two of us :)
15:17.40eppigy:>
15:17.52[TK]D-Fendermomelod: Yeah, You aren't SHOWING us.
15:18.00eppigynow I am ordering my ccnp stuffs
15:18.07Kattyorly
15:18.10eppigyyesh
15:18.19Kattyhave you ordered a tree for your computer room yet?
15:18.20eppigywhich will be a lot more interesting
15:18.28eppigysince it actually applies to what I do daily
15:18.35eppigyKatty: negative :[
15:19.01Jymmeppigy: Since when is sitting on your butt playing WoW on a test?
15:19.24*** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt)
15:19.36Jymmeppigy: I'm just asking  =)
15:19.38KattyJymm: be nice to eppigy
15:19.42KattyJymm: don't be rude.
15:19.57eppigyJymm: lol
15:20.00KattyJymm: life's too short to run around being a mean person.
15:20.06eppigywow is a huge black hole
15:20.10eppigyfor your time and money
15:20.11Jymm\ignore Katty
15:20.16*** part/#asterisk manxpower (n=EWieling@69.73.94.162)
15:20.56Jymmeppigy: So, what did you think of the ccna test? what you expected?
15:21.19eppigyit was a lot easier than I thought it was going to be
15:21.34eppigyI would have gotten and almost perfect score
15:21.44eppigybut I made a mistake on a five part question
15:21.47Jymmeppigy: which study material did you use?
15:22.13eppigyhttp://www.amazon.com/Official-Certification-Library-640-802-Guide/dp/1587201836/ref=sr_1_2?ie=UTF8&s=books&qid=1251818527&sr=1-2
15:22.46eppigyI also purchased some switches and routers
15:22.48eppigyfor a home lab
15:22.54eppigythen our netadmin left
15:23.00Jymmeppigy: cisco's typical dry reading?
15:23.01eppigyand I took up the position
15:23.10eppigyJymm: of course
15:23.38eppigyI mean if you could convey binary math and distance vector loop prevention measures
15:23.43eppigyin a light and colorful manner
15:23.46eppigyi'd love to see it
15:25.27JymmActually, I probably could. But authors dont make all that much money.
15:25.44eppigyyeah dude screw the arts
15:25.58*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:27.09*** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell)
15:27.09*** mode/#asterisk [+o Qwell] by ChanServ
15:27.17*** join/#asterisk Talirk81 (n=David@rrcs-67-78-39-22.sw.biz.rr.com)
15:27.34Jymmeppigy: (My tech writing usually focuses on a broad audience of all/no skill levels)
15:28.17Talirk81I am running "AGI Script Executing Application: (VERBOSE) Options: (SET VARIABLE __SetCallerID_Name 'Level Call'"  but when i  use NoOP I see Executing [s@DialCallCenter:14] NoOp("SIP/vjzwsq-b7500468", "Level") in new stack      why is it not capturing the full  contents of the variable even though they are in quotes to prevent  confusion by the system?
15:28.38eppigyI cannot write anything without it reading like hunter s thompson
15:29.58Jymmeppigy: LOL, so YOU'RE the one that wrote that manual!
15:31.16*** join/#asterisk bmoraca (n=chatzill@66.242.174.254)
15:32.16*** part/#asterisk Katty (n=asterisk@mail.copi-rite.com)
15:34.13*** join/#asterisk ruben23 (n=RPL@122.55.48.243)
15:34.17kindyroothello, I have asterisk not listening to port 5060, and I can't fix that
15:35.01[TK]D-Fenderkindyroot: Show us
15:35.52kindyroot<[TK]D-Fender> http://pastebin.ca/1550382
15:36.23[TK]D-Fenderkindyroot: now DON'T filter it
15:36.56kindyrootok
15:37.29*** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net)
15:38.03wwalkeranyone have a reference to the actual speed one can expect from asterisk for originate requests through AMI?  I seem to be  getting an average of about 10 calls/sec with maybe 18 calls/sec peak.
15:38.32kindyroot<[TK]D-Fender> http://pastebin.ca/1550411
15:39.21[TK]D-Fenderkindyroot: not with a BRAIN : netstat -an
15:39.24[TK]D-Fendernow*
15:41.15*** join/#asterisk oej (n=olle@ns.webway.se)
15:47.09*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4df0:d9ce:287a:92ea)
15:48.06kindyroot<[TK]D-Fender> http://pastebin.ca/1550416
15:49.37QwellIn case people hadn't seen it - http://www.reuters.com/article/pressRelease/idUS144421+01-Sep-2009+BW20090901
15:49.58[TK]D-Fenderkindyroot: Now show me that * is running
15:51.06JymmQwell: Did you see that Ebay is dumping skype and Marc Andressen is buying it?
15:51.34QwellI hadn't seen that
15:51.43mort_gibhow do I enable ilbc
15:51.49JymmEbay dumps skype... http://money.cnn.com/2009/09/01/technology/ebay_skype/index.htm?postversion=2009090108
15:51.53QwellWhy is that name familiar?
15:52.00JymmQwell: Netscape
15:52.05Qwellright
15:52.08Qwelland something else
15:52.39JymmQwell: I forget his other post-netscape comany name, though I drove past the bldg the other day and I see an HP sign out front.
15:52.59Jymmdoesn't mean much
15:53.31rene-wwalker: have you read about nir simionovich work?
15:54.13[TK]D-Fendermort_gib: There are libs to install that are in the docs...
15:54.33mort_gibHi TK, just found it, is ilbc any good??
15:55.20[TK]D-Fendermort_gib: I have no purpose for it
15:55.54mort_gibI have a VOIP provider saying "this is the way to go"... I'm testing them out...
15:56.10kindyroot<[TK]D-Fender> [root@localhost asterisk]# ps -A | grep asterisk
15:56.11kindyroot<PROTECTED>
15:56.11kindyroot<PROTECTED>
15:56.58[TK]D-Fenderkindyroot: now connect to * cli and do "sip show peers"
15:57.48kindyrootas a sip client I have Ekiga, is that okay?
15:58.03kindyrootwhat should I do?
15:59.44kindyroot<[TK]D-Fender> is that what you mean by * cli?
15:59.50raden_workok our fall over internet connection would have been handy this morning :(
16:01.17[TK]D-Fenderkindyroot: If you don'tknow how to log into * CLI you have a lot of basic reading to do...
16:01.24[TK]D-Fenderkindyroot: asterisk -rvvvvvvvvv
16:01.42raden_work[TK]D-Fender, morning
16:01.47wwalkerrene-: no, will read up after lunch, thx
16:02.12[TK]D-Fenderis off to lunch
16:02.48*** join/#asterisk CunningPike (n=CunningP@204.239.8.157)
16:07.06kindyroot<[TK]D-Fender> honestly that's the first time I hear about asterisk client :s
16:07.13kindyrootgoogling right now
16:07.49kindyroot[TK]D-Fender> ah you mean like FreePBX?
16:08.23Talkradiolooking for suggestions on phones you guys like to use, i'm using polycom 330 and volume level is an issue for the old ladies working there
16:08.42*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
16:08.48JymmTalkradio: buy new and improved ladies?
16:09.01Talkradioi wish i had that option lol
16:09.47kaldemarkindyroot: no, not FreePBX. FreePBX is a GUI to configure asterisk. you connect to asterisk CLI (command line interface) when asterisk is running.
16:10.49kindyrootkaldemar: haaaaaa!
16:11.00kindyrootthis must be what I am missing
16:11.15kindyrootdoes that cli come with *now?
16:11.57*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
16:11.57kaldemarit comes with asterisk, so yes.
16:12.13kaldemarit's not a client software, it's a built-in CLI.
16:12.29kindyrootwhat's the name of the command ?
16:12.34hardwiresigh
16:12.37hardwirewhy do we have bosses?
16:12.48kaldemarkindyroot: < [TK]D-Fender> kindyroot: asterisk -rvvvvvvvvv
16:13.10kaldemarkindyroot: are you using asterisknow?
16:13.27kindyrootkaldemar: yes a fresh install of it
16:14.21kindyroothaaa I was dropped into the cli mini-shell
16:14.37kaldemarthen go ask in #asterisknow. people here use plain asterisk, not GUI's. if someone tells you how to fix the problem from console, the GUI will just screw it up again sooner or later.
16:15.53kindyrootkaldemar: I can drop freepbx if I can solve my problem by hand
16:16.07kindyrootmy purpose is to get things working and to learn
16:16.38kindyrootbtw I have already been to asterisknow and they couldn't help
16:18.23carrartime to switch to Asterisk compiled from source then!
16:19.04*** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com)
16:19.32kindyrootcarrar: I'd love to, I am just still kept back by fear
16:19.44hardwireanybody have a multi-associatable bluetooth headset?
16:19.48kaldemarkindyroot: what does command "sip show settings" show you?
16:19.50hardwirethat's a word.. I checked.
16:20.04kindyrootcarrar: even the fully configured distro didn't work for me
16:20.12*** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br)
16:20.37kindyrootno such command
16:21.07kaldemarkindyroot: what about "module load chan_sip.so" ?
16:21.17bpgoldsbIn my dialplan, I make lots of use of goto.  However, goto doesn't support arguements.  The community/docs say that using Gosub/Macro without returning is bad.  What are my options for setting arguements for the context I'm going to?  Currently I try to Set(VAR) before jumping, then inside the jumped-to context, check that VAR is set.  Does anyone have a better idea?
16:21.28*** join/#asterisk seanmh (n=johndoe@207.114.199.107)
16:22.03kindyrootit passed, means worked?
16:22.12bpgoldsbAnother idea is to use the extension '_!' and key off ${EXTEN}, but that posses multiple problems.
16:22.57kindyrootkaldemar: what is this supposed to do?
16:23.11JymmWhat's the purpose/benefit of having asterisk on a lil linksys router? I just don't "get it"
16:23.16kaldemarbpgoldsb: if i were you, i'd refactor my dialplan to use GoSub.
16:23.37kaldemarkindyroot: load the sip channel module, since it seems that it's not loaded.
16:23.45kaldemarkindyroot: now pastebin the output
16:23.49raden_workis there a way to set a unavailable message up with vitelity in case we loose registration
16:24.03*** join/#asterisk qdk (n=qdk@195.242.194.41)
16:24.18kindyrootkaldemar: I am afraid there was no output, it just took it silently
16:24.57*** join/#asterisk haryv (i=lanny@174.1.123.38)
16:25.04kaldemarkindyroot: "core set verbose 10" and try again
16:25.10bpgoldsbkaldemar, I was under the impression Gosub was meant to do something, and _return_.  Is that not the case?
16:25.27kaldemarbpgoldsb: yes
16:25.35kaldemarit is the case.
16:26.06kaldemarif you want a mere goto and values to variables, keep on using Set and Goto.
16:26.14*** join/#asterisk jkroon (n=jkroon@dsl-240-169-69.telkomadsl.co.za)
16:27.11kindyrootkaldemar: Verbosity was 9 and is now 10
16:27.24kindyrootmodule load chan_sip.so
16:27.35kindyrootthen again, took it silently
16:29.37kaldemarinteresting. try "restart now" and the module command again.
16:30.32garymcim installing sox on my asterisk server now and I think i need mpg123
16:30.46garymcdo i just yum install them and hey presto?
16:31.26*** join/#asterisk nohup_ (n=nohup@chef6.nohup.nl)
16:31.30nohup_good afternoon
16:31.31kindyrootDisconnected from Asterisk server
16:31.31kindyrootExecuting last minute cleanups
16:31.41*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
16:32.43kindyrootkaldemar: then I go through the cli again and load the module?
16:32.50kaldemaryes
16:33.16nohup_i am thinking about 'upgrading' my power-slurping server to an intel atom pc, without any PCI slots. In my current server i have a ZAPATA PCI card, that won't work anymore.. so my question is if there would be any USB solutions i could use instead?
16:33.54kindyrootkaldemar: still goes silently
16:34.01*** join/#asterisk sjobeck (n=Adium@64.122.41.37)
16:34.06kindyrootI think it's loaded now
16:34.39kaldemarwhat kind of solutions? analog? BRI? PRI?
16:34.41kindyrootbut netstat still shows that It's not listening to the right port
16:34.55kaldemarkindyroot: why do you think it's loaded?
16:34.58*** part/#asterisk sjobeck (n=Adium@64.122.41.37)
16:35.03kindyrootkaldemar: all software for the moment
16:35.49kindyrootkaldemar: because otherwise it would thrash garbage at my face, just a thought
16:35.53nohup_kaldemar: sorry.. i have my landline (analog pstn) connected to my asterisk box
16:36.55haryvdid asterisk ever resolve the line presence issue?
16:37.41Qwellharyv: you're gonna have to be a bit more specific...
16:38.26kindyrootkaldemar: what are my chances if I reinstall?
16:38.27kaldemarnohup_: you could use an ATA
16:38.32haryvQwell, say a employee is on line 1 in the other room. It should show that line presence on all phone that this employee is on that line.
16:38.50[TK]D-Fenderharyv: what is "line"?
16:38.54nohup_kaldemar: you have those for imcomming pstn too? and they'recompatible with asterisk ?
16:38.57*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
16:39.01[TK]D-Fenderharyv: And what sounds like stuff I've done already
16:39.09kaldemarnohup_: it converts analog to SIP for example
16:39.11nohup_i have one of those linksys with two 'outgoing' pstn lines already...which works pretty good i might ass...
16:39.21bpgoldsbIn Asterisk 1.6, is it still 7 levels of nesting before Asterisk explodes?
16:39.35haryvline1,line2,line3 on polycom phones. a local IP provider could never get it to work.
16:39.56cuscoWe also noticed sometimes extra DTFM comming trough DAHDI. Clinet types only, say a phone number and instead of 9 chars, we get 10 or 11
16:40.06[TK]D-Fenderharyv: then you aren't using the phone right
16:40.19haryvyour not reading my message.
16:41.00[TK]D-Fenderharyv: "a lcoa IP provider"?  Who? Get what to work?  What are they doing on their end?
16:41.07haryvA local ip provider did not get it to work properly. Mine was never setup that way.
16:41.23kindyrootwell thank you <[TK]D-Fender> and <kaldemar> very much for your help, I think I will continue tomorrow, have a nice day/night
16:41.28[TK]D-Fenderharyv: So what have you now done which still doesn't work?
16:42.14haryvTK, never configured it to work in that way. But does it even with any sip channel in use ?
16:42.40nohup_kaldemar: what is the name of such a thing? (so i can go ebay for some stuff :) )
16:42.47[TK]D-Fenderharyv: Yes they support presence
16:43.34haryvI like to disprove them wrong but then again, thay are leaving the voip biz and entering into a different model
16:43.51*** join/#asterisk frieze (n=frieze@pool-74-101-21-2.nycmny.fios.verizon.net)
16:44.20friezeanyone know if someone makes an SIP cordless phone with a PoE charger? Or failing that one with a charger that has a wall mount
16:44.25[TK]D-Fenderharyv: Sounds like they never knew what hey wer doing
16:45.40haryvI dont know the companies history but did have a active working biz. Now it is just down to the owener and linux developer.
16:46.14haryvI think Shaw communications and telus killed off most voip providers
16:46.17kaldemarnohup_: for example linksys spaXXXX ATA
16:46.20[TK]D-Fenderharyv: Sounding more accurate by the moment
16:46.31haryvknocked down there prices so low, the small companies could not compete
16:47.11nohup_kaldemar: thank you! :) i'll be ebaying then :)
16:47.48[TK]D-Fendernohup_: Don't  With your luck you'll end up buying a locked POS and waste more money on shipping
16:48.07JymmLOL, that's cold man, cold.
16:48.31nohup_[TK]D-Fender: yeah.. i noticesd some of them having a tag saying "UNLOCKED", so i'll be looking for only those :)
16:48.53[TK]D-Fendernohup_: An playing the odds on it all the while
16:49.10Jymmbtw, is PAP2T still in production? I can't seem to find any at retailers. Seems like they might have been transitioned to Cisco Small Business
16:49.28*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com)
16:49.57JymmWell, PAP2T-NA specifically
16:51.38QwellJymm: they've been replaced, yes
16:51.57JymmQwell: oh noes! With what?
16:52.03Qwellchickens.
16:52.15Qwellno, SPA2102 I think
16:53.05JymmI can't find that either it seems. Frys has ZERO non-branded VoIP gear - any brand. MicroCenter seems simular.
16:53.20Jymmsame goes for costco online
16:53.59[TK]D-FenderQwell: No, SPA-2102 was in production at the same time, and they are different
16:54.39JymmWell, Staples Online has PAP2T - bastidges
16:55.07[TK]D-Fenderstaples = LOCKED
16:55.15[TK]D-FenderRetailers don't carry unlocked shit
16:55.21Jymm[TK]D-Fender: To whom?
16:55.31[TK]D-FenderBecause they don't want to deal with idiot end users who have no clue what they're doing
16:56.08Jymm[TK]D-Fender: Who or what are they locked to/againest?
16:56.34*** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net)
16:58.17*** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan)
16:58.37*** join/#asterisk xuser (n=xuser@unaffiliated/xuser)
16:58.51bmoracawhatever provider staples sells
16:59.16bmoracawww.voiplink.com sells unlocked PAP2Ts
16:59.33Jymmhttp://www.staples.com/Linksys-PAP2T-VoIP-Internet-Phone-Adapter-with-2-Ports/product_766549
16:59.47JymmNo provider listed
17:00.52*** join/#asterisk dymaxion (n=dymaxion@89.248.128.108)
17:01.06[TK]D-FenderJymm: "Good luck" all I can say
17:02.15Jymm[TK]D-Fender: No, I wasn't doubting you, I just dont know and I'm asking. I know that PAP2T-NA *IS* unlocked, but when a provider isn't stated and no -NA, what would be locked?
17:02.16theharwe get our pap2ts from 800voipstore
17:03.20Jymmmaybe I need the definition of "locked" in this respect.
17:03.55JymmIf I tried to hack a vonage box, sure I get it.
17:04.11Zuchmir2just called staples: they say it's unlocked
17:04.32garymcanyone here know why i cant add music mp3 to my music on hold? its freepbx i know but nobody there knows why
17:04.32dymaxionHi,  is there anyone here usingi the   Atcom IP04 for their asterisk PBX ?
17:04.57JymmZuchmir2: cool, ty
17:05.39[TK]D-FenderZuchmir2: Yes, and Stapes are real techies...
17:06.28*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
17:06.28*** mode/#asterisk [+o angler] by ChanServ
17:06.46Jymm[TK]D-Fender: Ok, what could it be locked to?
17:08.48[TK]D-FenderJymm: Vonage Broadvoice, etc
17:09.04Zuchmir2on the webpage they don't mention it being locked, the rep i spoke to spoke to a techie before responding
17:09.26Zuchmir2wouldn't they want to specify that on their website?
17:09.47[TK]D-FenderZuchmir2: because many are lazy with their descriptions
17:09.57[TK]D-FenderZuchmir2: And many "techies" are morons.
17:11.16*** join/#asterisk dysinger_ (n=tim@71.20.35.99)
17:13.02Zuchmir2is the PAP2T better than HT-502?
17:15.43*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
17:15.54*** join/#asterisk dysinger_ (n=tim@71-20-35-99.war.clearwire-wmx.net)
17:16.12*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:16.26k4tanaLINUX~billing
17:17.11[TK]D-Fender~gs
17:17.12infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
17:18.44Zuchmir2quality of: build, fw, sound? or all of the above?
17:19.31[TK]D-FenderZuchmir2: AOTA
17:20.07Zuchmir2so the PAP2T is better
17:23.53*** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net)
17:27.36*** join/#asterisk ariel_ (n=chatzill@63.214.236.169)
17:29.14ariel_Afternoon folks
17:30.25momelodgreetings
17:35.51cuscoasterisk is taking 100% cpu
17:35.56cuscohow can I find out why?
17:36.09cuscoplease
17:37.34ariel_Maybe via tops you might find out which process is taking up your cpu.
17:37.56cuscoI found
17:38.05cuscohtop shows 2 processes taking huge amounts
17:38.28*** join/#asterisk seanmh (n=johndoe@207.114.199.107)
17:38.41dustybini think its time to get asterisk up and running
17:39.23hardwiredo it do it now
17:39.41dustybinok
17:39.44cuscoall calls are having cuts
17:40.02dustybinthere are so many compile optios that im lost in what i need / dont need
17:40.06dustybin*options
17:40.07ariel_all calls are having cuts, oh due to 100% restart now
17:40.20[TK]D-Fenderdustybin: Start with the defaults
17:41.11dustybinok
17:41.58dustybin[TK]D-Fender: i will just do a ./configure && make && make install
17:42.02cuscoasterisk machine got really slowwwwwwww
17:42.07ariel_is a bit happy, he may have to take a trip to setup install/fix a system overseas.....Athens...I have not been there yet...
17:42.14cuscoand now i coud reattach a cli ai could read LOTS of:
17:42.15cuscoutils.c:1126 ast_carefulwrite: write() returned error: Broken pipe
17:43.51dustybinits compiling :D
17:44.03dustybin1.6.1.5
17:44.41dustybinim tempted to setup a softphone until i get a proper voip external
17:44.44*** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com)
17:46.48[TK]D-Fenderdustybin: should do that regardless
17:47.38*** join/#asterisk [jmc] (n=[jmc]@93-45-200-133.ip104.fastwebnet.it)
17:47.39dustybin[TK]D-Fender: are softphones used for testing?
17:47.59[jmc]hi guys :)
17:48.01[TK]D-Fenderdustybin: They're used for whatever you use them for
17:48.06dustybinace
17:48.16dustybinfeels excited
17:48.18*** join/#asterisk fofware (n=Client@host3.190-230-38.telecom.net.ar)
17:49.25dustybinmy god, asterisk required no deps
17:50.04russellbjust libncurses-dev
17:50.04russellbheh
17:50.32Qwelland gcc
17:50.33Qwell:p
17:50.38dustybini just did: make samples
17:50.45[jmc]and libc6-dev
17:50.49dustybinthere is another one, i missed it, something like make progdocs?
17:50.59Qwelldustybin: you don't need that
17:51.13dustybinok
17:51.23dustybinwell thats it, asterisk is installed
17:51.32dustybinnow i need to find a decent softphone what runs on os x
17:51.51Qwellx-lite or whatever they call it these days
17:51.58[jmc]hmm
17:52.20[jmc]QuteCom?
17:53.33[jmc]hey does anybody know how a Linksys 3102 works?
17:53.46[jmc]there's something I don't get
17:54.04ariel_3102 there are allot of info on that unit it's a good fxo/fxs ata
17:54.24[jmc]yes
17:54.32[jmc]but I'm not expert with its configuration
17:54.42[jmc]and I'm doing something bad I think
17:54.45dustybinx-lite os x installed :D
17:55.07[jmc]I mean, I have an analog phone attached to the "Line" entrance, the FXS one
17:55.21dustybinnow i need to buy a pay-as-you VOIP phone account
17:55.23[jmc]and the phone cable in the FXO entrance
17:55.24QwellFXS isn't a line...
17:55.46[jmc]I said entrance
17:55.51[jmc]how do you call it? :D
17:55.53Qwellyou also said "Line"
17:56.04Qwelldoes it say "Line" on the jack?
17:56.11dustybinim going to buy some VOIP credits from these people: http://www.voiptalk.org/products/IAX+PSTN+Call+Credit
17:56.25[jmc]err, "Phone" I'm sorry
17:56.40[jmc]"Line" is the other one, you're right
17:57.12[jmc]however, what I'm trying to get is that I want to call on the phone line from my Asterisk PBX
17:57.28[jmc]and have my phone working just like it did before
17:57.33[jmc]I receive calls perfectly
17:57.45[jmc]but I can't call anymore
17:58.16[jmc]when I dial something from the analog phone, it tries to call an extension of my PBX
17:59.20cuscois it possible to compile asterisk with proffiling enabled?
18:00.33spckhi guys
18:00.43[jmc]here's what the 3102 says on my syslog
18:00.46[jmc]http://www.nopaste.com/p/a69Yad8RU
18:01.23[jmc]obviously 192.168.0.2 is the IP of my Asterisk PBX
18:02.05[TK]D-Fendercusco: Profiling what?
18:02.14[TK]D-Fender[jmc]: What does ASTERISK say?
18:02.28spcki'm running a redundant setup with two asterisk boxes and redfone fonebridge between, currently i have an issue where one box will provision the spans and bring them up just fine, but the other box refuses too
18:03.17[jmc][tk] I'll tell you in a moment, but the point is that asterisk should NOT say anything, I'd like it to dial on my PSTN line, not via  VoIP
18:03.24[TK]D-Fenderspck: almost noone uses TDMOE.
18:03.40*** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
18:03.46[TK]D-Fender[jmc]: Oh... nothing to do with us... go check out www.voxilla.com 's forums
18:03.48dustybini have signed up for a free acount with a VOIP provider, i now have a incoming number, a SIP ID and a password, is that all the details i need for when i configure asterisk?
18:03.51*** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:04.22[TK]D-Fenderdustybin: the host to send calls to/ auth calls from.  Allowed codecs, etc
18:04.29dustybinok
18:04.57spcki'm starting to realize why
18:05.18dustybin[TK]D-Fender: i have [SIP NUMBER]@voiptalk.org
18:05.38[jmc][tk] I've been doing that for two days, and I can't get it to work, I'm not here to bother you all... as a last resort I tried to see if someone had my same experience. If you know a dedicated IRC channel I'll be happy to go bother them instead! :)
18:06.16spckis there anyway to force the spans to come up?
18:07.05ariel_spck: red-fone has a good support department
18:07.20*** join/#asterisk oej (n=olle@ns.webway.se)
18:08.04*** join/#asterisk wayne_r (n=wayne_r@rrcs-24-173-187-234.sw.biz.rr.com)
18:08.09spckariel: debateable
18:08.19[jmc]thanks, anyway
18:08.24[TK]D-Fenderdustybin: Go look up some config samples for them
18:09.17ariel_[jmc]: trick is the line (fxo)  don't register that with the asterisk box.
18:10.31dustybin[TK]D-Fender: aye thanks
18:11.03[jmc]ok ariel_, thanks, I'll do some tests and see what happens ([tk]'s right, this is not really the right place to talk about it, I'm sorry)
18:11.07[jmc];)
18:11.24dustybin[TK]D-Fender: http://www.voiptalk.org/products/iaxconfig.html
18:11.40hescoif called from an AGI script, where does a print STDERR 'debug message'; go?
18:12.03dustybinextensions.conf v iax.conf
18:12.15dustybinlikes the sound of iax.conf
18:12.37*** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser)
18:13.16*** join/#asterisk cesar_CR (n=cesar@201.192.86.30)
18:15.03dustybinjesus lord, i have never seen so many .conf files in my life
18:15.42dustybinanybody would think asterisk is _serious_ stuff :P
18:15.56*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:16.11wayne_rI'm trying to spec out what I need for my first attempt at an asterisk installation.  What I'd like to do is use four Linksys SPA3102s for a total of 4 FXO and FXS ports -- is it possible to route incoming calls on the FXO ports to asterisk and route them to the appropriate FXS port (on a different SPA3102)?
18:16.32bmoracayes
18:16.36bmoracathey're all just sip peers
18:16.50wayne_rbmoraca: that's good news.  :)
18:17.02bmoracai personally would never do that...but it's possible
18:17.03[TK]D-Fenderwayne_r: and thats a horrible thing to do for a business
18:17.05cusco[TK]D-Fender: so I can use gprof
18:17.19[TK]D-Fenderwayne_r: I highly recommend your getting a decent FXO solution
18:17.20cuscoto tell me where it may be spending cpu
18:17.51wayne_r[TK]D-Fender:  why is the 3102 not a good solution?
18:18.08[TK]D-FenderSPA-3102 is ok for personal lo usage and failover, not primary business
18:18.46[TK]D-Fenderwayne_r: Poorer EC, no fine grained interface control, more devices to configure, inability to pool channels, etc
18:18.48wayne_rhmmm well that's no fun
18:19.30bmoracawayne_r: 4 of those is 2/3rds of the way to a proper FXO card anyway
18:19.57Qwellwayne_r: really aught to consider SIP phones too
18:20.00wayne_rthe goal is running it in vmware and avoiding buying a new machine
18:20.13*** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net)
18:20.27wayne_ris there something similiar that is better and would work without needing to be physically installed on a box?
18:20.44bmoracai hope you're not trying to use something like VMWare Server or Workstation...
18:20.56wayne_rnah esxi
18:20.59Qwellbmoraca: player ;)
18:21.59bmoracawayne_r: i've used those SPA3102s in that context, and it was a pain in the ass.  i'd very much advise against it.  use a sip provider and SIP phones if you absolutely have to run it in VMware.
18:23.02bmoracathey work well SIP to FXS, but the FXO to SIP is clunky at best
18:23.10*** join/#asterisk moy (n=moy@bas1-unionville55-1177733667.dsl.bell.ca)
18:23.42[TK]D-Fenderwayne_r: AudioCodes or Mediatrix analog gateway
18:23.56wayne_rbmoraca: fair enough
18:24.02wayne_r[TK]D-Fender:  i'll have a look, thanks
18:25.58bmoracait definitely works, but i'd prefer not to have to set that up again...hosted PBX customer had two analog lines at their location that they could not get rid of or move (they were extensions on a hospital's phone system) so i had little choice...but i definitely don't want to do it again
18:30.44giovanihas anyone here used Metrostat for their CNAM service?
18:35.45*** join/#asterisk bluOxigen (n=asad@static-host119-73-69-38.link.net.pk)
18:36.30*** join/#asterisk De_Mon (i=de_mon@fl-69-34-134-91.dhcp.embarqhsd.net)
18:39.17*** join/#asterisk dymaxion (n=dymaxion@89.248.128.108)
18:41.37linageeif i'm using a fixed SIP jitter buffer, how do I know where to set the buffer length?
18:42.14garymc[TK]D-Fender : http://pastebin.ca/1550604
18:42.21garymcI sussed out what you was saying
18:42.43[TK]D-Fendergarymc: except for "SHOW ME THE FAILURE"
18:43.23carrarNo such command 'SHOW ME THE' (type 'help' for help)
18:43.31garymcyeah no such command
18:43.34garymc:P
18:43.41*** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
18:43.55carrarhahah
18:44.37carrarRussel really needs to implement a command like that
18:44.40carrar+l
18:44.49garymcI dont know how to show the failure
18:44.52garymc:(
18:44.58carrarthen nothing is broken?
18:45.26garymcbut it aint working
18:47.39garymcdoes [TK]D-fender just blank you when hes had enough of you?
18:47.39carrarHow do you know it'
18:47.43carrars not working
18:47.55garymcCos it says an error in the browser
18:48.03garymcalso when i hold a call theres no music
18:48.12garymc:(
18:48.22carrari don't see any error
18:48.25garymcIm thinking its a freepbx issue
18:48.41carrarDid you ask the people in the channel that deal with your setup?
18:49.03garymcI pressed reload config after it and i get loads of info so much so that the first lot is missing from the terminal window
18:49.08garymcYes
18:49.16garymcNobody seems able to help me
18:49.18garymc:(
18:49.24garymcits nearly my bedtime too
18:49.32carrarinstall asterisk from source
18:49.39carrartoss that freepbx
18:50.39garymcothers seem to be doing ok with freepbx
18:50.44garymcjust not me
18:52.58carrarBut this is not a FreePBX channel
18:53.08[TK]D-Fender[14:48]<garymc>also when i hold a call theres no music  <-- why am I not seeing a call with MoH failing?  Why don't I see a dump of your MoH folder?
18:53.36[TK]D-Fendergarymc: And why have you spilled into here?
18:53.42carrarheh
18:53.51carrarNeed a bigger dam!
18:54.01garymcim usin chatzilla on pc easy to do by accident
18:54.32carraryeah I sometimes accidently get stuck in the microsoft windows 95 channel
18:54.37carrarno idea how
18:54.38[TK]D-Fendergarymc: I use Chatzilla as well.  Doesn't pose any problems
18:54.56garymcyeah im quite new to irc
18:55.00*** join/#asterisk bluOxigen (n=asad@static-host119-73-67-71.link.net.pk)
18:55.06*** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de)
18:55.07garymcim also using a mouse pad, pain in the arse
18:55.17carrarI use a desk
18:56.19garymcwell i put a call on hold and nothing else appears in the cli
18:56.22garymcI like CLI
18:56.59*** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net)
18:57.03carrarWhat did you set your verbosity too?
18:57.33bmoracayeah, you should have a message in CLI even for successful MOH
18:59.06IBC_jkenneyhey i need some assistance if you all have the inclination. I have a problem with a Wildcard AEX2400 Board 1 when someone dials into the card and we have the card plugged into a bank of modems we get a RX TX line problem
18:59.17IBC_jkenneyis there away to better set this card up for data
18:59.23IBC_jkenneywe do not really do any speech on it
18:59.56*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)
19:00.00*** join/#asterisk jong2 (n=chatzill@65.100.10.89)
19:00.22bmoracaIBC_jkenney: no.  and you're probably not going to get much support doing that.  if you need a fax server or a dialin aggregator, get one.  used AS5200s are pretty cheap on ebay.
19:00.24IBC_jkenneythe inbound calls come from a PRI
19:00.26*** join/#asterisk sasargen (n=chatzill@173-124-140-145.pools.spcsdns.net)
19:01.05IBC_jkenneybmoraca our inbound calls are coming in on a PRI and we are using the AEX2400 for pots line breakout
19:01.18IBC_jkenneythere is no sip or "voip involved"
19:01.28bmoracadid i say there was?
19:01.57bmoracaeither way, though, they're routing through Asterisk, which is not really equipped for what you're trying to do.  that's why I recommended getting a real access server, such as an AS5200
19:02.15*** join/#asterisk oej (n=olle@ns.webway.se)
19:02.31joakoI know this is off-topic as hell, but: does anyone know where I can contact to find a LOCAL Nokia service center?
19:02.48carrarGOOGLE!
19:03.20carrarhttp://tinyurl.com/nax7xj
19:03.20rene-hehe
19:03.56joakoHell I called Nokia, the rep told me there were local service centers but that they could not give out that info
19:03.57carrarYou can play with the working
19:04.02carrarmaybe change stuff
19:04.03carrarmaybenot
19:04.08carrarwording
19:04.13carrarheh
19:04.44joakocarrar: That took me to a page that says I need to enable JavaScript... why would i need a script in my coffee? I am confused.... :)
19:05.20[TK]D-FenderIBC_jkenney: Could have echo, clocking issues, gain, etc... all sorts
19:05.41[TK]D-FenderIBC_jkenney: is has always been recommended to keep data devices as far away from * as possible
19:06.52bmoracaIBC_jkenney: one possible solution is to go entirely Sangoma...as they do support FAX services from PRI to FXS with a timing sync cable between two cards with echo cancellers.  as far as I know, no such similar claim exists for Digium cards.
19:07.04*** join/#asterisk d00gster (n=doughant@77.30.9.36)
19:07.09bmoracathat is manufacturer supported, so you'd have some recourse when things don't work
19:07.14bmoracabut i still wouldn't recommend it
19:07.35*** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk)
19:07.44carrarI use PRI -> Digium T1 card -> ADIC600 channel bank -> FAX
19:07.46carrarthat works
19:09.22*** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87)
19:09.38FlaPer87hey guys, is it possible to write C++ modules for asterisk?
19:09.54bmoracait hasn't been made clear exactly what kind of modems are being dialed, so it's plausible that the person's attempting to terminate dialup calls
19:10.20FlaPer87where can I find tutorials for creating Asterisk modules?
19:10.29garymcanyone know why the CLI is not showing my internal clal getting put on hold?
19:10.35garymc*call
19:11.11bmoracagarymc: core set verbose 10
19:11.21garymcyep i did
19:11.33joakoFlaPer87: take a look at app_skel.c it should be part of your Asterisk source code
19:11.55FlaPer87joako ok, thanks
19:12.04bmoracagarymc: and you still haven't pastebinned an actual call.
19:12.14[TK]D-FenderFlaPer87: And there a few googleable tutorials out there as well
19:12.18garymci have in another channel
19:12.25[TK]D-Fenderbmoraca: I saw a (useless) sample
19:12.26joakogarymc: FWIW I don't ever recall any output (well except perhaps SIP debug) that shows a call being placed on hold, but if you set your verbosity > 2 it should indicate when the music on hold is started
19:12.36bmoracathat doesn't particularly help me very much, now does it?
19:12.49[TK]D-Fenderbmoraca: Didn't help me much either
19:12.49Naikrovekchecks out starfish pbx
19:13.02[TK]D-Fender~happyfunclownpbx
19:13.15[TK]D-Fender~happyclownpbx
19:13.16infobot[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone
19:13.19[TK]D-Fender:)
19:13.28*** join/#asterisk jtodd (i=uancftet@ns.fox-den.com)
19:13.28*** mode/#asterisk [+o jtodd] by ChanServ
19:13.58*** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72)
19:14.05bmoracaNaikrovek: looks too much like a linksys interface.  eck
19:15.06garymcok
19:15.21garymchttp://pastebin.ca/1550636
19:15.46*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:15.52*** join/#asterisk Chromis (n=millsu2@mail.serverplus.com)
19:16.04garymcsee what you make of that one, as I made an internal call and put it on hold. Had them on Loud speaker and could hear myself coming through other phone. But no music when on hold. Red light was flashing on Ip330 phone
19:16.21garymcwhen was holding as supposed too
19:16.36garymcthat was also a verbose setting of 10 or higher
19:16.53bmoracagarymc: for the love of god don't leave the freepbx "status" page up when you're doing these debugs...it's difficult enough already to debug it without that crap in the way
19:17.19[TK]D-Fendergarymc: How did you put them on hold, and what device are you using?
19:17.25garymcwhats that, its only a couple of lines, just thought id let you see my verbose setting etc
19:17.32garymcPolycom Ip330
19:17.34*** part/#asterisk korihor (n=korihor@190.77.83.180)
19:17.37garymci pressed the hold button
19:17.48garymcI can also press hold then pick another line and phone them again
19:17.59[TK]D-Fendergarymc: What's the other device?
19:18.03*** join/#asterisk maour (n=gnu@unaffiliated/maour)
19:18.03bpgoldsbWait, someone is actually making a product named happyclownpbx?
19:18.05garymcPolycom IP 330
19:18.11[TK]D-Fendergarymc: both?
19:18.14garymcyes
19:18.22[TK]D-Fendergarymc: Something is very wrong
19:18.30garymchuuurraaahhh
19:18.37spckheh, vs starfishpbx?
19:18.37garymcI knew there was something weird going on
19:18.47[TK]D-Fendergarymc: do another call.  Place on hold, place 2nd call
19:18.53garymcok
19:20.20FlaPer87is it possible to compile an asterisk module without compilling the whole asterisk?
19:20.29*** join/#asterisk rickross (n=rickross@supporter/active/rickross)
19:21.22garymcok here we go, so much stuff i cant get to the beginning
19:21.35garymchttp://pastebin.ca/1550642
19:21.39*** join/#asterisk jkroon (n=jkroon@dsl-240-169-69.telkomadsl.co.za)
19:23.28[TK]D-Fendergarymc: while on hold do : sip show channels
19:23.44garymcthe phones?
19:23.44[TK]D-Fendergarymc: and prior do : core set debug 10
19:23.51[TK]D-Fendergarymc: and prior do : core set verbose 10
19:23.56*** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
19:23.58garymcit was verbose 10
19:24.05garymcand the phones show channelas
19:24.06[TK]D-Fendergarymc: Confirma dn do the other
19:24.13garymc??
19:24.27[TK]D-Fendergarymc: Confirm and do the other
19:24.42garymci set verbose to 10
19:24.47garymcits already been done
19:24.56garymcand the phones show the channels
19:25.11garymc??? if you mean something else i dont know what that is
19:26.59ChromisI keep getting errors reading: "pgsql_log: cdr_pgsql: Reason: ERROR:  column "calldate" specified more than once". It has worked fine for over 2 weeks and just started doing this a few hours ago.
19:27.22garymcok well i should be getting home to the missus now, been here for hours and hours. its 20:30 here so[TK]D-Fender you reckon you could have a think about it for me and i will grab you 2morow?
19:28.22garymcyeah as soon as i pres hold nothing happens
19:28.25garymcin the cli
19:29.14ChromisAsterisk is trying to insert duplicate row into the dtabase for some reason. "INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid","calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid") VALUES . . ."
19:30.19rickrossis there an easy way to watch/log commands that asterisk receives via the manager interface on port 5038?
19:30.28*** join/#asterisk netpro25_ (n=mmanning@64-238-176-253.ksg.apt.gru.net)
19:30.36[TK]D-Fenderrickross: there is not AMI debug mode
19:30.49[TK]D-Fenderrickross: Best I think you can do is spy on the port via wireshark, etc
19:31.12garymctk?
19:31.15rickross[TK] - thanks
19:31.15garymcany thoughts?
19:31.37[TK]D-Fendergarymc: I just asked you to show me 3 more things, and I've got nothing.
19:31.39rickrossI guess maby ngrep would let me see this stuff?
19:31.41[TK]D-Fendergarymc: there's a though
19:31.43[TK]D-Fendert
19:31.51garymci showed you them
19:31.53[TK]D-Fenderrickross: packets are packets
19:32.07rickrossI don't know wireshark
19:32.13[TK]D-Fendergarymc: no, you didn't
19:32.19garymci thought i did
19:32.29garymcyou asked if i set it to verbose 10 which i did
19:32.32[TK]D-Fendergarymc: Sub-contract <-
19:32.49garymcthanks
19:33.05[TK]D-Fendergarymc: I also asked for a "sip show channels" while ON HOLD, and enable CORE DEBUG
19:33.28[TK]D-FenderSwear to God some people need to learn to #*&$ing read
19:33.45KyleKodd
19:33.54KyleKsmartypants asterisk addons ignores includedir
19:36.19*** join/#asterisk dominic_ (n=chatzill@207.61.107.242)
19:36.46*** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
19:37.46dominic_anyone know why I would be getting this warning :[Aug 31 14:48:30] WARNING[22463] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4)
19:39.42dominic_looks like a codec issue... but 'frame type 64' ??
19:40.18Chromishesco: it looks like about a year ago you had the same problem I am having. Did you ever find a solution?
19:40.29*** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br)
19:40.50carrarhttp://www.voip-info.org/wiki/view/Asterisk+codecs
19:41.36dominic_ahh thx
19:41.47*** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.162)
19:42.10[TK]D-FenderStupid laptop
19:42.14carrarYEAH
19:42.23carrarNo so SMRT
19:42.57dustybinwhat port will my x-lite phone use to auth my asterisk server?
19:43.13KyleKto auth?
19:43.16carrarUDP 5060?
19:43.27dustybini think my firewall is blocking, so im going to telnet
19:43.54dustybin<PROTECTED>
19:43.54dustybineeek
19:43.54KyleKUDP != TCP
19:43.55QwellYou can't telnet to UDP
19:44.00dustybinoh yes of course
19:44.06lirakisdust netcat
19:44.12carrarnmap -sU -p 5060 1.2.3.4
19:44.12lirakis= friend
19:44.45dustybini will check on this asterisk server using, netstat -natp | grep 5060
19:45.02dustybinnothing open
19:45.19carrar*sigh*
19:45.33lirakisdustybin,  drop the t
19:45.37dustybinok
19:45.41lirakisdustybin, netstat -nap | grep 5060
19:45.53dustybinyep, nothing open
19:46.08dustybinthere are 100's of .confs
19:46.12lirakisdustybin, ... uhh... i think you are ahead of yourself here
19:46.14lirakisgo read
19:46.17lirakis~read
19:46.18infobotACTION reads Lord of the rings
19:46.24lirakiserr damn
19:46.27lirakis~thebook
19:46.28infobotsomebody said thebook was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org
19:46.38dustybini bought the o-reilly book
19:46.45dustybini will take a read i think
19:46.47lirakisdustybin, Thats Awsome !! ME TOO
19:46.51lirakisgo read it
19:46.52dustybinasterisk wis not a 5 minute job....
19:46.55dustybinits a 5 month job :D
19:47.00dustybin*year
19:47.01*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
19:47.11lirakisdustybin,  after you read a bit - you will be fine
19:47.14dustybinok
19:47.22carrarasterisk/linux/tcpip
19:47.37*** join/#asterisk WHYS (n=drumm@137.28.94.209)
19:48.22jong2if you have dial plan nightmare, try http://www.safisystems.com/
19:49.55[TK]D-Fenderdustybin: cat /etc/asterisk/modules.conf
19:50.35*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
19:51.15dustybin[TK]D-Fender: http://paste.debian.net/45490/
19:51.42[TK]D-Fenderdustybin: netstat -an|grep 5060
19:51.55[TK]D-Fenderdustybin: then at * cli : sip show peers
19:52.21dustybinohhh it is open
19:52.27dustybinudp        0      0 192.168.1.10:5060       0.0.0.0:*
19:52.33*** join/#asterisk oej_ (n=olle@ns.webway.se)
19:53.21dustybinim not sure how to execute the * cli command
19:53.32[TK]D-Fenderdustybin: Fine, its listening.
19:53.42[TK]D-Fenderdustybin: Congrats.
19:53.45dustybin:D
19:53.47lirakislol
19:53.50[TK]D-FenderNEXT!@!@!!!@ (c) BKW
19:54.06bmoracaasterisk -rx 'command'
19:54.51jong2gee gmail down..
19:55.12dustybinhttp://paste.debian.net/45491/
19:56.11dominic_carrar: [Aug 31 14:48:30] WARNING[22463] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4)
19:56.39dominic_would mean that someone is asking for slin and I am providing ulaw?
19:56.52*** join/#asterisk jcape (n=jcape@209.120.251.81)
20:01.51KyleKtype 64 is sln? then yea it sounds like it
20:02.46[TK]D-Fenderdustybin: OK, 1 SIP device configured, not yet registered and SIP is listening
20:05.40dominic_I trying to find out who is asking for slin
20:06.10dominic_I put disallow=all
20:06.17dominic_allow=ulaw
20:06.23dominic_everywhere
20:06.46dominic_I googled allow=slin and only got bodybuilding stuff
20:07.08KyleKturn on sip debugging?
20:08.05dominic_theres lots of traffic on the server
20:08.12dominic_so I wont see the output
20:08.15dustybinnc -u 192.168.1.10 5060   <-- is this the correct command to check a udp port using netcat?
20:08.35lirakisdustybin, no
20:08.51dustybinnc -u -s 192.168.1.10 -p 5060
20:08.54lirakisdustybin, save yourself and a lot of people here some headache ... go read the first few chapters please
20:08.57*** join/#asterisk TimToady_ (n=moi@adsl209-7.kln.forthnet.gr)
20:09.08dustybinok
20:09.30dominic_yeah I will turn it on and try to see if I find something
20:10.56kfifeDTMF doesn't break DISA dialtone?  Exten => *67 invokes DISA() with null CLID.  Connected to asterisk via private-loop PRI from a legacy PBX.  Asterisk 1.6.  Any ideas?  Do I need to relax DTMF?
20:11.10KyleKis gmail down?
20:12.14bmoracaappears to be
20:12.29dustybin[192.168.1.10] 5060 (sip) open  :D
20:12.40dustybinnow reads book
20:12.57dustybinand sips some tea :D
20:13.29kfifeThe GMail IMAP interface seems to be online however.
20:14.00dominic_thx everyone
20:14.30timeshell_atworkHey, I got a new thing...
20:14.31bmoracaseems to be back up now
20:15.14timeshell_atworkI have a Polycom IP 601 that freezes up and reboots when  call comes in on a specific analog trunk port that is forwarded to it using 1.6.0.14.
20:15.28timeshell_atworkThis didnt happen before my upgrade from 1.6.0.9
20:15.46timeshell_atworkAny ideas?
20:15.52kfifetimeshell_atwork: is your polycom firmware up to date?
20:16.02timeshell_atworkkfife Yes
20:16.20timeshell_atworkMost recent boot and sip software for the 601
20:16.38timeshell_atworkUpgraded it to 3.1.3 from 3.1.2 a month or so ago.
20:17.01kfifetimeshell_atwork: Waiting for epiphany
20:17.32[TK]D-Fenderkfife: epiphany is already installed, but Firefox is selected as default.  Is that OK?
20:18.30kfife[TK]D-Fender: LOL.
20:18.50[TK]D-Fenderis DONE for the day....
20:18.52[TK]D-FenderBBL
20:18.53kfife[TK]D-Fender: Epiphany=webkit?
20:19.41*** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil)
20:20.22KyleK~dev
20:20.23kfifenope--I see now- mozilla based.
20:21.03jong2no starfish channel?
20:29.46*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
20:31.27*** join/#asterisk heit0050 (n=andy@mail2.heitkeconsulting.com)
20:32.27*** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net)
20:34.48drclue<PROTECTED>
20:34.49drclueAnything folks might want to put on a wish list?
20:36.06drclueThe existing code and documentation I've posted as open source at http://code.google.com/p/fastagi-php-drclue/
20:36.27KyleKFastAGI is AGI over sockets?
20:36.36drcluePretty much
20:37.05drclueI also toss in a persistent AMI connection
20:38.01drclueThe way I set it up my FastAGI.php runs as a daemon and loads ones script at dial time based on whatever you specify in the dial plan
20:38.10*** join/#asterisk Caplain (i=shayne@caplain.loves.thraen.fbi.gov.silverelitez.org)
20:38.55drclueThe FastAGI.php even has a command line option to generate the init.d script to start it at boot
20:39.47KyleKcool
20:40.42*** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167975979.dsl.bell.ca)
20:41.01drclueI found that having the FastAGI.php load the scripts at dial time that PHP development/debugging cycle is pretty painless edit,save, dial
20:43.03*** join/#asterisk denon (i=denon@synapse.subneural.net)
20:43.03*** mode/#asterisk [+o denon] by ChanServ
20:43.11*** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com)
20:43.19*** join/#asterisk rgavril (n=rgavril@89.120.4.153)
20:43.32*** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com)
20:43.37drclueI have a companion script called FastAMIevents.php , that does event monitoring but I have not really figured out what I want to do with it yet. I was sorta thinking of having it use shared memory and output XML and JSONized XML depending upon how it was called
20:44.32*** join/#asterisk netpro25_ (n=mmanning@64-238-176-253.ksg.apt.gru.net)
20:45.27netpro25_Can someone point me to some info regarding the # shortcut
20:45.43netpro25_to transfer the call
20:46.00*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
20:47.14drcluenetpro25_: what kind of phone are you trying to dial the "#" from?
20:47.39netpro25_drclue: well I have a spa941 just wanna know how to go from desktop to cell if I gotta run
20:49.01*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
20:50.04carrarAre you listening for a trasfer option when you call out or receive a call?
20:50.13netpro25_nm just figured out how to do it with the transfer button
20:50.21netpro25_thats sweet
20:54.20*** join/#asterisk tRSS (i=tRSS@140.192.34.2)
20:54.53tRSSquick question: can I have multiple periodic-announce messages for the same queue?
20:59.21*** join/#asterisk ZX81 (n=Matt_Rid@121.74.14.197)
21:00.21tRSSquick question: can I have multiple periodic-announce messages for the same queue?
21:00.56*** part/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net)
21:01.54netpro25_is it safe to have asterisk use the tmp folder for record cache
21:02.19KyleKI'd only worry about it if you have shared hosting on the same server
21:02.27*** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net)
21:02.33netpro25_okay
21:02.37*** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net)
21:02.48netpro25_well in that case where is a standard place to put it
21:02.57netpro25_/var/lib/asterisk/tmp?
21:03.51KyleKI dunno actually
21:03.59netpro25_well that sounds good to me
21:04.00netpro25_heh
21:10.45*** part/#asterisk ZX81 (n=Matt_Rid@121.74.14.197)
21:12.47kfifeDigits 1,2 and 3 don't break DISA dialtone?  Strange.  DTMF is generated by legacy PBX, connected to Asterisk via private-loop PRI Any ideas?  1.6.0.13
21:13.14kfife4,5,6,7,8,9,0,#,* register properly
21:17.51*** join/#asterisk voxter (n=voxter@76.77.95.2)
21:19.31*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:20.31*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
21:22.46Naikrovekall sip traffic is UDP, yes?
21:23.12uqlevRTP is UDP only
21:23.17Naikrovekokie dokie
21:23.25Naikrovekthe SIP signalling is UDP as well?
21:23.40netpro25_is it possible to include a whole directory in extensions.conf
21:23.49netpro25_like #include "exts/*"
21:24.12[TK]D-FenderNaikrovek: SIP supports TCP & UDP.  UDP is standard with *, TCP is 1.6 option
21:24.41NuggetSIP TCP is wonderful if you're stuck with ucky NAT.
21:24.50Naikroveki'm setting up some ACLs on my router and whenever I enable the proper ACL, my asterisk system no longer hears any incoming sound from outside callers.  even though my VOIP provider is given full UDP connectivity to my * server
21:24.55Naikrovekthis is terribly frustrating.
21:25.03Naikrovekwill add a TCP allow for them as well
21:26.33dustybini have communication!!!
21:26.42[TK]D-FenderNaikrovek: TCP won't help your RTP issue
21:26.51Naikrovekthis is starting to piss me off
21:27.09Naikrovekwhere could the traffic be coming from, if not from the trunk
21:27.48kfifeWhat's the benefit of SIP over TCP?  Just authentication or some other beneift?
21:27.50dustybin-- Registered SIP '1000' at 192.168.2.20 port 26342
21:27.52dustybin:D
21:28.13Naikrovekwhen i have the ACL in place, and I dial from my cell phone, I can connect to my own IVR and hear the voice.  but I can't dial an extension.  My phone system hears nothing
21:28.17netpro25_kfife: two different animals
21:28.22netpro25_you mean UDP TCP
21:28.25Naikrovekwtf is up with that...
21:28.31[TK]D-FenderNaikrovek: kfife Bypasses need for NAT keep-alives
21:28.43[TK]D-Fenderkfife: ^
21:28.53Naikrovek?
21:29.01kfifeBut transport is still over UDP no?
21:29.20kfife...media transport I mean
21:29.36kfifeTCP just for sip signalling--that's the idea, right?
21:29.48Naikrovekkfife: it's all UDP as far as I can tell
21:30.00kfifeI was goign to say: otherwise katy bar the door!
21:30.02Naikrovekbut [TK]D-Fender said that TCP is optional
21:30.13kfifeThere's latency, and then there's LATENCY
21:30.23[TK]D-FenderNaikrovek: No, I didn't
21:30.26netpro25_kfife: TCP would help reduce packet loss and thus jittery sound
21:30.41Naikrovek[TK]D-Fender: what did you mean, then
21:30.50kfifenetpro25_: I believe that's incorrect.
21:30.57[TK]D-FenderNaikrovek: Exactly what I said and nothign more.
21:31.01Naikrovekoh that you can use TCP SIP in asterisk 1.6?
21:31.08[TK]D-Fendernetpro25_: Completely incorrect
21:31.14netpro25_kfife: then I guess I dont fully understand
21:31.43netpro25_I was thinking in the lines of UDP having to acknowledgement packet
21:31.48kfifeIf you had TCP guarante delivery, you'd end up with HUGE latency-seconds, minutes, etc as packets lined up to be delivered
21:31.50*** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net)
21:31.55raden_workthere anything special i should have todo to make my procurve work with asterisk ?
21:32.04netpro25_kfife: right slower
21:32.12*** join/#asterisk manxpower (n=EWieling@69.73.94.162)
21:32.13kfifeEver done a VIDEO stream over TCP VPN?   See what happens.
21:32.14dustybinthe book says asterisk can also be used with analogue phone lines with the correct card, how cool is that?
21:32.15[TK]D-FenderNaikrovek: rarely do you realy have need of TCP for SIP.  I can be helpful if you have multiple phones behind a remote NAT as the inbound is taken care of due to a persistent connection.
21:32.21manxpower~answers
21:32.22infoboti heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
21:32.38Naikrovek[TK]D-Fender: the phone server is not behind NAT; it's on a public IP, so I don't think NAT is the issue
21:32.53[TK]D-FenderNaikrovek: I also didn't say this was your issue.
21:33.04netpro25_So how do sip providers like vitelity resolve nat issues, because I do not need to open ports for them
21:33.04Naikrovek[TK]D-Fender: i know, just ruling NAT out
21:33.12[TK]D-FenderNaikrovek: I never did get a solid description of it or SIP debug of a failed call
21:33.18kfifenetpro25_: stun, ice, turn
21:33.24[TK]D-FenderNaikrovek: if * is behind NAT then you might have an issue
21:33.34netpro25_kfife: all three or one of those
21:33.43Naikrovek[TK]D-Fender: it isn't
21:33.44kfifetake your pick
21:33.50kfifenetpro25_: take your pick.
21:34.02kfifenetpro25_: any or all.
21:34.03Naikrovek[TK]D-Fender: fetching sip debug for call now.  few moments, please
21:34.05[TK]D-Fendernetpro25_: * needs ports forwarded to it otherwise no audio, and probably no calls
21:34.28netpro25_[TK]D-Fender: yes, thats what I had to do
21:34.41netpro25_otherwise I had no connection from my phone to server
21:34.57raden_work[TK]D-Fender, is there something keeping me from registering with vitelity or is there a network issue  ? http://pastebin.com/d5e269648
21:35.45kfiferaden_work: I just dialed my vitelity DID, and forwarded the call to my mobile phone via vitelity trunk.  All seems to work.
21:35.45*** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
21:36.04kfiferaden_work: try reloading your sip.conf
21:36.12raden_workdid
21:36.18raden_workdid a graceful restart
21:36.21raden_workas well
21:36.24netpro25_seems like vitelity is the choice of a few people in here
21:36.36netpro25_I like them, have had no probs
21:36.47netpro25_anyone using them for business lines?
21:36.58kfifeI reallly like their user interface for managing/adding/deleting etc.  I also like their ticket system.
21:37.00dustybini know you guys dont approve of guis, however, is there some kind of asterisk stat / simple gui system, so i can check things from html if im at work, etc
21:37.15netpro25_kfife: yes and they respond so diligently and quickly
21:37.19netpro25_very helpful
21:37.25raden_workwell i have a route to them just for some reason seem to be having issues
21:37.32kfiferaden_work: netpro25_:  AMEN
21:37.36Naikrovek[TK]D-Fender: http://pastebin.com/d5716c0ae
21:37.52kfifeAt the moment they
21:37.57raden_worki like vitelity  never said i didnt
21:38.13netpro25_raden_work: did you forward 5060?
21:38.20kfifeAt the moment they're one of the only ITSP's that can set CNAM for your DID's.  A mere $10 no MRC.
21:38.23[TK]D-FenderNaikrovek: Do not specify a specific IP and that PB is a complete waste
21:38.27kfife^ VITELITY IS
21:38.33Naikrovekugh
21:38.36Naikrovekk
21:38.48Naikrovekwhat is your pastebin preference
21:38.57raden_worknetpro25_, yeah that all done on the router all i did was throw my hp procurve in and yeah every other provider registering
21:39.08raden_workvitel-outbound/tanning     64.2.142.17                 5060     OK (83 ms)
21:39.08raden_workvitel-inbound/tanning      64.2.142.15                 5060     UNREACHABLE
21:39.08raden_workcallcentric/17772445766    204.11.192.31               5060     OK (46 ms)
21:40.16*** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be)
21:40.17netpro25_raden_work: strange
21:40.25raden_worknetpro25_, i know
21:40.32netpro25_raden_work: did you file a bug
21:40.37raden_worktraceroute shows same route to both
21:40.39netpro25_raden_work: err ticket
21:40.45raden_worknot yet
21:41.05netpro25_raden_work: as I said earlier they are very helpful and nice
21:41.11raden_workyeah they are
21:41.24raden_workcall centric support better but i like vitelity overall better
21:41.38Naikrovek[TK]D-Fender: http://pastebin.ca/1550844
21:42.10netpro25_raden_work: ah they have unlimited at call_centric
21:42.16raden_workour ISP been down 5 hours in last 16 hours and 42 min so it could be something todo with that as well
21:42.22kfifeI had a ticket in which Vitelity set my outgoing CLID to some number in New Jersey for certain terminations.  Seems fixed now, but I'm sure it's because some cheap route in their 'Rate Deck' is is some half a$$ provider.  Made me a bit nervous.  Never had that problem with anyone else.   A vexing probelm because you'd rarely be in a position to notice all the times it happens.
21:42.32raden_worknetpro25_, there are advantages to both
21:42.48netpro25_raden_work: like single channel with unlimited
21:42.57raden_workyeah :)
21:43.13raden_workor 3 channel unlimited inboung and 9 dollar a channel extra
21:44.54netpro25_raden_work: ah personal unlimited
21:44.57netpro25_$6
21:44.59[TK]D-FenderNaikrovek: Looks ok, verify your WAN IP and confirm what you have forwarded
21:45.00netpro25_thats good deal
21:45.12*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:45.31Naikrovek[TK]D-Fender: there's no NAT here, what do you mean forwarded
21:45.53[TK]D-FenderNaikrovek: Confirm your firewall settings.  Also, yuo only reach an IVR.  Still no audio direct?
21:46.07*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
21:46.56Naikrovek[TK]D-Fender: when i turn off the firewall it work, turn it on, and it doesn't work.  I already know the firewall is the issue.  I just don't know why.  I can hear the IVR on my cell phone, but dialling numbers does nothing when the firewall is up.  The DTMF or DMTF or FTDM or whatever it is doesn't get to my IVR
21:47.21Naikrovekturning the firewall on renders * deaf for me
21:47.22raden_workNaikrovek, you have your ports forwarded ?
21:47.31Naikrovekraden_work:  no NAT
21:47.34raden_workNaikrovek, router / firewall model ?
21:47.40NaikrovekCisco 2811
21:47.59raden_workmust be something with SPI
21:48.11Naikrovek~SPI
21:48.12infoboti heard spi is serial peripheral interface.  Software in the Public Interest
21:48.12manxpowerNaikrovek: Audio uses UDP ports 10,000 - 20,000 by default.  CISCOs use port 16384 - 32768 by default for audio.  change /etc/rtp.conf to use those ports
21:48.34Naikrovekmanxpower: the trunk is wide open to my provider.
21:48.42manxpowerNaikrovek: apparently not.
21:48.49manxpowerif it was you would not have this problem.
21:48.50raden_workNaikrovek, pastebin debug of call
21:48.59Naikrovekwith or without firewall, TCP and UDP are wide open to the * server
21:49.04raden_workmanxpower, are you the one that gave me contact for vitelity ?
21:49.08manxpowermake sure you turn off the sip fixup in the router.
21:49.08Naikrovekaccording to the Cisco ACL they are
21:49.16manxpowerNaikrovek: do you have canreinvite=no?
21:49.21[TK]D-FenderNaikrovek: Your firewall is blocking the wrong stuff then.  You need to allow 5060 & 10000-20000 all UDP
21:49.30manxpowerotherwise the phone will try talking directly to the provider
21:50.27manxpowerNaikrovek: on your router issue the config command "fixup protocol sip 5060 "
21:50.28Naikrovekwhat EXACTLY does this ACL block between these two hosts?  surely there must be some Cisco folk in here.
21:50.30manxpowerthen save the config
21:50.33manxpower..er...
21:50.33Naikrovek<PROTECTED>
21:50.33Naikrovek<PROTECTED>
21:50.34manxpowerwait!
21:50.38manxpowerno fixup protocol sip 5060
21:50.38dustybinbefore i go to bed, my softphone has a telephone number of 500, how can i get asterisk to call that number so it rings?
21:51.17manxpowerNaikrovek: You understand that by default Asterisk gets out of the audio path and the phone and the provider talk directly, right?
21:51.26TapoutI'm getting service for a reverse-cell calling , do you guys think the communications are 'private'?  In that, I should be able to give out my credit card numbers over phone...?
21:51.41Naikrovekmanxpower: no
21:51.49Naikrovekhow do recordings happen if that's the case
21:51.56*** join/#asterisk alecdavis (n=sivad@202-78-149-14.cable.telstraclear.net)
21:51.58Naikrovekthat must be what you mean by NOT default
21:51.59KyleKwhats cdr(userfield) for?
21:52.20Naikrovekk let me try something else here then
21:52.22netpro25_dustybin: google asterisk dial command
21:52.24manxpowerNaikrovek: no, there are features that can be enabled that will prevent reinvites, but those are not DEFAULT.
21:52.36dustybinthanks
21:53.10Tapoutanyone?  are the services you buy from voip providers 'private' in that, I should be ok giving out my credit card number over the phone?
21:53.32TapoutI'm getting reverse cell phone calling .. so I get free north american calling on my cell, i wanna be able to give my credit card number over the phone without issues... what do you guys think?
21:53.40manxpowerTapout: your question is silly.  Do you pay by credit card when you eat at a restaurant?  Phones are more secure than that.
21:53.52manxpowerTapout: I don't even know what "reverse cell phone calling" is.
21:54.19manxpowerTapout: do you give your credit card number out over a land line?  Easy to tap that too.
21:54.22Naikrovekjesus fucking christ i hate this
21:54.38Naikrovekmy network is wide fucking open and phone calls work, or i secure it and calls break.  what a choise
21:54.39Naikrovekchoice
21:55.11Tapoutmanxpower, i'm paying a company so taht when I dial out on my cell, it rings busy.. i hang up, and it calls me back, and then I can dial out for free
21:55.13manxpowerNaikrovek: no, listening to what I said about reinvites is also another choice.  turn them off.  happy Naikrovek
21:55.14raden_work[TK]D-Fender, any idea why i cannot register ?
21:55.23Naikrovekwill never be happy
21:55.27Naikrovekso lose that idea now
21:56.43Naikrovekand
21:56.53Naikroveki'm still talking to asterisk when audio stops working
21:57.06manxpowerI guess it sucks to be you.
21:57.09Naikrovekso i haven't handed off to a phone yet
21:57.26Naikrovekthe traffic has to be coming from somewhere other than the trunk
21:57.37Naikrovekmy router isn't voip-aware so i don't think it woudl be that
21:58.42*** join/#asterisk flujan (n=flujan@189.111.254.251)
22:00.31Tapoutanyone doing at-home asterisks boxes in canada?  I wonder if there is a voip provider that gives blocks of minutes cheap near calgary.. my voip is a bit shitty
22:00.41raden_workI have everything plugged into a hp procurve switch and then uplinked to netgear at moment could this be causing any problem with registration  ?
22:01.20*** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com)
22:01.32*** join/#asterisk shinao1 (n=shinao1@41.219.218.218)
22:01.46KyleKTapout: who are you using right now?
22:02.11Tapoutmagic jack lol
22:02.16dustybinhow odd, im missing the dial command
22:02.21KyleKah
22:02.26Tapoutfor home, but i wanna setup my own box.. i'm paying telsair right now for cell phone, but i'd rather do it myself as this is awesome
22:03.23KyleKah
22:04.13KyleKright now I'm using les.net, from BC it seems like his servers are colocated in BC too even though the company is manitoba based
22:04.21Naikrovekis there a way in linux to see where UDP packets are coming from
22:04.36Naikroveknetstat doesn't seem to be doing it
22:04.36manxpowerdustybin: not odd at all.  The CLI dial command is only available if you have the alsa headers and libraries installed when you build Asterisk
22:04.45dustybineeek
22:05.08KyleKNaikrovek: tcpdump
22:05.21dustybinthanks for help this evening, time for bed
22:05.34TapoutKyleK, what are you paying a month?
22:06.38Naikrovekah hah
22:06.45NaikrovekMFer
22:06.51KyleKI'm paying 1.5 cents a minute
22:08.18MaliutaKyleK: what if you just wanted it for inbound calls?
22:08.45MaliutaKyleK: I have been looking for a .ca DID (my parents live in Fort McMurray)
22:09.04*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
22:09.35Naikrovekokay i figured it out
22:09.42Naikrovekwhat in the holy piss is going on
22:09.58Naikrovekmy incoming data is not coming from my trunk.  funny thing
22:10.09Naikrovekwill be calling about that tomorrow
22:10.20MaliutaNaikrovek: reinvite?
22:10.24Naikrovekno
22:10.26Naikroveknot reinvite
22:10.37Naikroveknot fixup protocol sip 5060
22:10.39Naikroveknot NAT
22:10.49Naikroveknot anything anyone described fixed it
22:11.06KyleKMaliuta: I still have POTS service i'll look at what it is at les.net
22:11.15KyleKbut they might not offer any dids in fort mcmurray
22:11.24Naikrovekthe packets carrying audio INTO my PBX from my PSTN provider do not come from the trunk I have configured
22:11.34MaliutaNaikrovek: I know my provider use a bank of machines to handle stuff ... it pisses me off
22:11.59MaliutaKyleK: they have Edmonton .... which I figure is close enough (and cheaper than them calling my .au number)
22:12.01Naikrovekapparently mine does too
22:12.26Naikrovekwhen configuring the primary access list at my internet connection, I neglected to imagine that anything other than my effing trunk would be sending me voiip data
22:12.27Naikroveksilly me
22:12.36KyleKMaliuta: true dat
22:12.37Naikrovekso when I put the ACL in place, that data was blocked
22:13.35MaliutaNaikrovek: I had that too. I put a qualify on the sip trunk config and then used some whois foo to figure out what to do on the firewall
22:13.58Naikroveki'm not opening up two /16 networks, which is what my voip provider has, apparently
22:14.12Maliutaick
22:14.15Naikroveki just opened the /24, and i'll be on the lookout for problems
22:14.50MaliutaI'd contact them and ask for info on exactly which servers/ip's you need to allow
22:14.59Naikrovekwell that's fixed.  only took three years to semi-secure that network.  I can't BELIEVE we weren't hacked.  about 8 machines were full open to the internet for years
22:15.05Naikroveknot even firewalls on the machines
22:15.19Naikrovekis going to reinstall everything anyway
22:15.30Naikrovekslaps the previous admin
22:15.46Naikrovekwith a cricket bat
22:15.49Naikroveksideways
22:15.53Naikrovekidiot
22:15.59Naikrovekshesus
22:16.01Naikrovekleaves
22:16.37Maliutayou don't want to use a cricket bat sideways ... you want the full face of the blade to deliver force in a more destructive manner
22:16.51Naikroveksideways means more PSI
22:16.56Naikrovekpounds per square inch
22:17.12*** join/#asterisk jcape (n=jcape@209.120.251.81)
22:17.24Naikrovek... if pound is abbreviated "lb" why is "pounds per square inch" abbreviated "PSI"
22:17.36Naikrovekshouln't that be "lbSI"
22:17.38Naikrovekugly.
22:17.41Naikrovekokay leaving now
22:17.48Naikrovekstill mad even though i solved it, manxpower
22:17.50Naikrovektold ya :)
22:19.53*** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker)
22:20.10KyleKMaliuta: go with the flat rate DID
22:20.22*** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com)
22:22.15*** join/#asterisk Tim_Toady (n=moi@adsl209-7.kln.forthnet.gr) [NETSPLIT VICTIM]
22:22.43MaliutaKyleK: will look into it, have to talk to the 'rents and see if it's worth it given their phone plan
22:23.14KyleKso they dont have broadband?
22:23.25MaliutaKyleK: I can already call all of .ca (including mobiles) for $0.08AU untimed
22:23.52KyleKhaha including mobiles
22:24.11MaliutaKyleK: they have shaw cable, if I can get my shite together and take the right bits with me at xmas then it'll go to a * trunk
22:24.15KyleKmobile/not mobile is only noticable outside north america
22:25.02Maliutawhich is most of the world :P
22:25.14KyleKtrue
22:25.43Qwellpfft, only by volume and by population
22:25.54Qwells/volume/land mass/
22:26.29Maliutaand every other meaningful measurement
22:26.30KyleKcdr_sqlite.so is deprecated? :(
22:26.32Qwell:D
22:28.48KyleK$0.08AU untimed as in 8 cents for the whole call?
22:29.30MaliutaKyleK: yeah
22:29.39TapoutKyleK, i don't see how you signup to les to use it at home... seems like you gotta pay X amount of dollars for colocation
22:29.42MaliutaKyleK: to a bunch of places
22:30.06KyleKTapout: ?
22:30.35*** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net)
22:30.43KyleKTapout: i guess thier website is kind of hard to navigate
22:30.44TapoutKyleK, i want to setup voip at home, using my adsl .. thru les.net , how much initially should it take to use 'em?
22:30.53Tapouti'd guess, unless you know what you're looking for
22:31.00Tapoutwhich, obviously i don't ;)
22:31.07KyleK:) just signup
22:32.05KyleKI threw in $20 via paypal, they sat on it for 14 days since I dont trust them with my CC info and I don't trust paypal with my banking account (unverified paypal account) :) but like $5 would do
22:32.24bmoracaKyleK: "mobile/not mobile is only noticable outside north america" <=== that's not true.  mobile in the US is a different OCN than even the same providers' copper service...as such, they will have different negotiated rates.
22:34.54KyleKbmoraca: well i dont have access to any such negotiated rates, I'm just a sucker paying a flat rate :)
22:36.42Maliutameh. like I care about $0.08AU to talk to mum for 2 hours
22:36.52bmoracamost people are far down the ratescale and their upstream providers have extracted enough markup from them that they don't have to account for it...but, for me, i know that calling AT&T/Cingular mobile costs roughly double what it costs to call AT&T copper...that is to say that it costs ~$0.005/min to call AT&T Wireless as opposed to ~$0.0026/min to call AT&T landlines
22:37.41KyleKgot any numbers for canada?
22:37.56Maliutabmoraca: how much volume do you have to do to get those rates?
22:38.21bmoracano volume commitments...i collocate in the CLEC's data center
22:38.38rene-bmoraca: sweet
22:38.55*** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
22:39.29bmoracathey negotiate rates with various providers and i get the benefit of THEIR volume committments
22:40.04rene-bmoraca: do you resell?
22:40.06MaliutaI guess that's the issue with collocating in my loungeroom in Brisbane
22:40.19bmoracai charge $40/mo for hosted PBX trunks...which on average cost me about $3.50 in usage fees
22:40.29rene-bmoraca: sweet
22:40.32bmoracarene-: yes.  currently only to California
22:40.35*** join/#asterisk dshap (n=samhel55@cpe-024-211-245-021.nc.res.rr.com)
22:41.24bmoracaregular trunks are $30/mo...but they either have to buy the equipment from me or lease it for another $75/mo...Adtran TA900s aren't cheap!
22:42.44dshaphey can someone help me out?  all of a sudden my IVR menu is less responsive to DTMF
22:42.52dshapi have to press the button a couple times
22:42.54dshapor hold it down
22:42.59dshapto get it to switch to the extension that i want
22:43.02*** join/#asterisk propellerhead (n=yogurt2u@190.210.1.37)
22:43.05dshapthe weird thing is
22:43.10dshapwhen i run asterisk in the console
22:43.14dshapwith output and everything
22:43.15dshapit works fine
22:43.25dshapit's only when i have it running as a daemon
22:43.27dshapany ideas?
22:43.57KyleKbmoraca: so they give you a good rate and you just need your own equipment to go from T1 to VoIP?
22:46.54*** join/#asterisk voipmonk (n=voipmonk@65.14.229.26)
22:48.26dshapKyleK: you have any ideas about why running as daemon vs. console would make a difference?
22:49.54KyleKlike -d?
22:50.16KyleKit shouldn't make a difference
22:50.22KyleKtry running it in screen
22:50.39dshapwhen I do "asterisk &"
22:50.43dshapi'm havin weird issues
22:50.45dshapbut when i do
22:50.47dshap"asterisk -cvvv"
22:50.50dshapi don't have those issus
22:51.25dshapi haven't made any changes to asterisk during the period that this started happening
22:51.35dshaphowever i have started using the linux box extensively more as a web server
22:51.47KyleKoh
22:51.50dshapdo you think it gets more processing priority when it's a console?
22:51.52KyleKwhat linux
22:51.52dshapor something like that
22:51.57KyleKubuntu?
22:51.58dshapCentOS 5.3
22:52.02dshaprunning on a VERY old box
22:52.11dshap(surprisingly capable though, haven't had any issues what so ever until now)
22:52.27KyleKsounds like the computer is allocating more cpu power to tasks it considers interactive
22:52.35KyleKwhats the current load
22:52.40KyleK(uptime)
22:52.41dshaphow do i check
22:53.34*** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87)
22:54.26FlaPer87hey guys, This is the vm_exec method of app_voicemail.c : http://pastebin.com/d79f3c03a  don't understand when is recorded the message
22:55.02FlaPer87can anybody tell me what part of the method execution creates the audio file and save the message?
22:55.20KyleKdshap: whats uptime say?
22:55.59KyleK<PROTECTED>
22:56.05voipmonkFlaPer87: you're looking at the wrong part of the code.. keep looking
22:57.06FlaPer87voipmonk hmm, could you please help me to find what part does the thing? this part is executed by VoiceMail(123@anything)
22:57.21FlaPer87It answers the call
22:57.30voipmonkwhat do you want to do when you find it?
22:58.32*** join/#asterisk TimToady_ (n=moi@adsl16-233.kln.forthnet.gr)
22:58.34dshapKyleK: checking now
22:58.51dshapKyleK: 23 days
22:59.03dshapKyleK: seems about right...back when my dog accidentally unplugged the server
22:59.17dshapoh
22:59.18dshapaps.facebook.com/photodigger/logs.php
22:59.20dshapoops
22:59.24dshap15:58:41 up 23 days,  5:59,  1 user,  load average: 0.02, 0.02, 0.00
22:59.35FlaPer87voipmonk I need to process some vocal messages, So, The call will be answered and the user will say something, I need to save the things the user says in a file
22:59.52dshapwhat does that mean? 0.02, 0.02, 0.00
23:00.40KyleKits how many processes are usually waiting to run
23:00.51KyleKfirst is average is over a minute, then 5 then 15
23:01.56*** join/#asterisk jjshoe (n=jjshoe@wsip-70-183-82-162.sd.sd.cox.net)
23:02.19KyleKalso check top it displays cpu usage
23:03.00dshapnone are over 1% cpu
23:04.22dshaphow can i get a list of every program running on my server
23:04.44dshapis that it?
23:04.45dshaptop?
23:04.56dshapwhat about running services?
23:05.21KyleKservices are programs
23:05.45dshapwhat's the difference between ones you have to type "start" in front of though?
23:05.57dshaplike u don't just type mysqld &
23:06.00dshaplike i can for asterisk
23:06.15KyleKthe init script picks up settings
23:06.30dshapah so "start" means use an init script
23:06.50KyleKi believe so
23:06.59KyleKI've been ignorint centos more than i should
23:07.07dshapalso how come sometimes when i start my FTP server which is "pure-ftpd", i type "pure-ftpd &"
23:07.08dshapand it works fine
23:07.12dshapbut when i close the SSH terminal
23:07.16dshapit stops the server
23:07.24KyleKits still attached to the tty
23:07.25dshapi just want it to always be running
23:07.35KyleKnohup pure-ftpd
23:07.45KyleKor use an init script
23:08.32dshapand if i wanna kill it later i just use top to get the PID and then "kill [pid]" ?
23:08.33Maliutathe "&" simply backgrounds it in the current shell, it's not how to run a daemon proper
23:08.48dshapMaliuta: so I shouldn't be doing "asterisk &" ?
23:09.02KyleKuhm
23:09.06Maliutaif you use init scripts you can start and stop things at will
23:09.07KyleK"asterisk"
23:09.13KyleKbackgrounds itself that i've noticed
23:09.38MaliutaI think someone needs to go take *nix admin 101
23:09.55KyleKgotta start somewhere
23:10.04netpro25_Anyone every used the call centric office unlimited seems they don't want you to use it with an IVR
23:10.14FlaPer87where can I find docs about asterisk functions?
23:10.15netpro25_can anyone confirm this?
23:10.20dshapMaliuta: i do, lol. u gota  good book to recommend?
23:10.23Maliuta~book
23:10.24infobot[~book] 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://astbook.asteriskdocs.org or see ~buybook
23:10.32KyleKMaliuta: thats not nix admin 101
23:10.33dshapi meant a *nix book
23:10.37dshapi've read the * book
23:10.37FlaPer87Maliuta thanks
23:10.39netpro25_In other words can I not use call centric with asterisk
23:10.44MaliutaKyleK: that was for FlaPer87
23:10.45dshapand i have a working * server that i've been programming
23:10.47KyleKoic
23:10.48dshapoh
23:10.55KyleKhrmm
23:11.05dshapKyleK: well it seems that this issue is pretty spontaneous
23:11.05FlaPer87for example, what should this function do? ast_stream_and_wait
23:11.09dshapprobably due to processor load
23:11.10KyleKdshap: check out o reily books
23:11.18KyleKFlaPer87: doesn't the code have comments on what it does?
23:11.21dshapKyleK: should i read something generic or CentOS specific?
23:11.34KyleKdshap: read the generic book first
23:11.40FlaPer87KyleK not the one I'm reading
23:11.45dshapunix or linux?
23:11.54FlaPer87I mean, I'm watching how they use it
23:12.27KyleKdshap: generic linux yes
23:12.46Nuggetdshap: technically you're asking bash questions, which isn't specific to any unix or linux or posix os.
23:13.05NuggetI'm assuming you are running bash because that's what people run if they don't know what they're running.  :)
23:13.07KyleKpshaw
23:13.20KyleKhe just needs some background information on *nix administration
23:13.28KyleKthen read the centos book
23:14.40dshapwill do
23:14.41dshapthanks
23:15.09dshapbut very quickly
23:15.14dshapwhat is the proper way to run a daemon in centos?
23:15.19dshaphow should i be starting asterisk?
23:15.27dshapas opposed to my current method: "asterisk &"
23:15.38KyleKmy asterisk backgrouns without the &
23:15.43KyleKso its no difference
23:15.51netpro25_doent asterisk run in daemon mode by default?
23:15.58netpro25_doesnt^
23:16.02KyleKyou could auto start it in /etc/rc.local or look around the dir for an init script
23:16.17KyleKnetpro25_: I dont think it'll install a script into /etc/init.d/ by default
23:16.28KyleKI'm assuming hes smart and installed from source;)
23:16.40KyleKsqlite3-dev
23:16.50netpro25_KyleK: when you compile from source?
23:17.36netpro25_seemed to do so in ubuntu, unless it never removed the old when when I uninstalled the package
23:17.52KyleKaye
23:18.23netpro25_KyleK: cant remember if I was talking to you earlier about callcentric
23:18.28netpro25_or if it was someone else
23:18.52KyleKprobably someone else
23:19.11netpro25_do you use it?
23:19.14KyleKdshap: look in contrib/init.d the redhat one should work
23:19.30KyleKnetpro25_: just les.net, I currently call only 2 people long distance
23:19.42KyleKbeen thinking about getting a voip phone for one of them :)
23:19.56*** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
23:20.44dshapok cool thanks!
23:21.00netpro25_Yea CC has strange packages, the $9 office one says it cant be used for IVR
23:23.41KyleKthey just intend people to use it for incoming calls to peoples
23:23.57*** join/#asterisk voipmonk (n=voipmonk@65.14.229.26)
23:24.18netpro25_yea with a ATA I am assuming
23:24.21netpro25_not asterisk
23:24.21KyleKI'd be asking them lots of questions before choosing that package
23:24.30KyleKno you can still use asterisk
23:24.34netpro25_KyleK: yea I sent an email
23:25.02KyleKIVR services probably means like you cant sell tickets but you can have a menu for extensions
23:25.08*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
23:25.08*** mode/#asterisk [+o leifmadsen] by ChanServ
23:25.24*** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk)
23:25.26netpro25_KyleK: what do you mean sell tickets
23:25.53KyleKif you're 1800buyporn they want you to get the $20 package
23:26.01netpro25_ah
23:26.03netpro25_lol
23:28.13netpro25_Personal unlimited seems tempting also
23:37.22netpro25_Anyway to use a regex on file includes?
23:37.31netpro25_or just an asterisk
23:37.45KyleKfor what?
23:37.58netpro25_#include sips/*.conf
23:38.20KyleKoh in sip.conf
23:38.24KyleKI haven't tried even *
23:38.31KyleKso probably no for regex
23:38.32[netman]you can also include a directory, I think
23:38.41netpro25_let me try that
23:50.03bmoracawhy would you need to use regular expressions to include files?
23:55.00Corydon76-digBecause some people don't know the meaning of the word "overkill"
23:55.22Corydon76-digfile includes support globbing, though
23:55.59manxpowerI always buy a quad core server with 8GB of RAM so I am SURE it will handle 8 analog calls.
23:56.00manxpower8-)
23:59.04linageecan anyone go to this URL?  http://www.voip-info.org/wiki/view/Asterisk+QoS
23:59.22linagee"Invalid Response error was encountered while trying to process the request:"
23:59.58manxpowerLiNeTuX: I noticed it was broken the other day

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