IRC log for #asterisk on 20090208

00:02.11ZippomanJust so you guys know I didn't mean asterisk web GUI
00:02.20*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
00:02.39root52So what on earth do you mean?
00:03.05drmessanoIf this involves porn, I am all in
00:03.12Zippomanhahaha
00:03.24drmessanoUnless it's pedo crap.. in which case, I need to punch you in the throat
00:03.27ZippomanIm making a site so I can have text to speech calls
00:03.40drmessanoSo out with it, Loose Zipper Larry
00:03.44drmessanoHmm
00:04.00StanManCani had quite the grin on my face when i saw my grandma's call registering on the asterisk server
00:04.01StanManCanlol
00:04.14StanManCangoing through the steps "enter pin" "thank you" "enter number"
00:05.49StanManCannow the quality was good, but a little fuzzy, i'm using ulaw, any other's i can try that may give a better result ?
00:12.39[TK]D-FenderStanManCan: Its likely your provider.  ulaw is the highest practical quality codec.  it then comes down to network conditions and your provider
00:16.46StanManCanthought so.. oh welll
00:21.11carrar... and then as quickly as he came into the room, he was gone!
00:21.54drmessanocarrar: I hate it when you act "like that"
00:22.18carrarYou hate everything when this time of the month!
00:25.54[TK]D-Fendercarrar: Actually he lasted far longer... and listened and compied and stuff too!
00:26.24carrarheh
00:26.36[TK]D-Fendercarrar: Last time he tore off when I told him his attempt to mask everything in debug was fucking over attempts to fix the problem.
00:28.23drmessanoIm glad his grandma can call him now
00:28.28drmessanoMission accomplished
00:28.50drmessanoI would love it if my grandma called me, but Asterisk doesnt have that feature yet
00:29.10carrarI'd be a little scared if mine called
00:29.30[TK]D-Fendercarrar: I'm suspecting your reasons are the same.
00:29.38carrarmodule load app_afterlife.so
00:29.57[TK]D-Fenderdials 1-800-THE-LORD
00:30.17carraror perhaps thats chan_afterlife.so
00:31.11carrardebugging that could be hell
00:31.51drmessanoba dump
00:31.53drmessanoCHING!
00:32.32[TK]D-Fendertakes his giant cane and yanks carrar off the stage
00:32.40drmessanoI hear they just installed a large CCM network in hell..
00:33.43carrargood dd, good dd, good dd errr Thats All Folks!
00:34.02drmessanoMySQL hates me
00:34.14carrarPostgreSQL love you
00:34.28carrarloves even
00:34.39[TK]D-Fendergoes running for his asbestos suit...
00:35.40carrarSUN says MySQL 5.1 is ready for production!! :)
00:36.21drmessanoMySQL 5.1 is giving me a tumor
00:36.34[TK]D-Fendercarrar: http://arstechnica.com/open-source/news/2009/02/unsatisfied-with-direction-mysql-creator-leaves-sun.ars
00:36.57carrardrmessano, be glade thats all MySQL 5.1 is doing too you
00:37.13jupetersondoes anyone know of a good web voicemail application?  So people can listen to voicemail via a browser.. login and check messages
00:37.35carrarheh yeah I read that on /.
00:40.57*** join/#asterisk ManxPower (n=manxpowe@75.251.218.255)
00:41.23[TK]D-FenderMy company uses MySQL for a few small things for which I could probably faily easily switch to PostgreSQL if things ever turn on us
00:41.58carrarjupeterson, did you look at the source?
00:42.03carrarjupeterson, asterisk/contrib/scripts/vmail.cgi
00:42.46carrarTK, no time like the working present! :)
00:42.48jupetersonno, I didnt look at that yet.. thanks.. I'll check it out
00:42.59ManxPowerAs I understand it vmail.cgi is not maintained and has not been for years
00:43.30carrargoes back to googling for jupeterson
00:44.24*** join/#asterisk neurosys (n=vinix@c-71-196-9-142.hsd1.fl.comcast.net)
00:45.26[TK]D-Fendercarrar: ... presently I have MySQL and it is working :)
00:46.07[TK]D-FenderManxPower: As I understand it *'s voicemail file storage hasn't changed in years either :)
00:46.27ManxPower[TK]D-Fender: I'm working with FreeBSD.  I've not had to recompile a Linux kernel in *years*.  I've recompiled the FreeBSD kernel like 5 times already.
00:46.27[TK]D-FenderManxPower: And apparently the CGI has been updated with references to ODBC...
00:46.39ManxPower[TK]D-Fender: in that case I was wrong.
00:50.39carrarjupeterson, I setup Asterisk to use UW Imap for vm and that works slick
00:50.54carrarpeople just add the VM imap server to their existing mail program
00:50.56rob0One time, I thought I was wrong, but as it happened, I was mistaken.
00:51.48jupetersonok.. yeah, I've got it emailing attachments right now.. wanted a point and click method via the web.. I found this iPhone app for Asterisk but havent gotten it working yet
00:51.56jupetersonwas wondering if there was something turnkey out there
00:52.04carrarthis is not "emailing attachement"
00:53.05carrartheres a few web apps out there
00:53.21carrarthere is a nice one that comes with SwitchVox
00:54.43[TK]D-FenderAnd i'll only cost you your immortal soul! muahahhaahaha *cough*
00:54.47carrarhahha
00:54.50carraryeah
00:55.00carrartoo bad SwitchVox isn't open source
00:55.07ManxPower[TK]D-Fender: I sold that recently.  I don't seem to miss it all that much.
00:55.33[TK]D-FenderManxPower: I'm not sure who got gypped on that deal ;)
00:55.36ManxPowersorry, I thought you said "Immoral soul", not "immortal soul"
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01:03.46*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
01:05.45neurosysManxPower:  are you running asterisk on fbsDF?
01:05.48neurosys*fbsd?
01:06.06drmessanofbsDF, christ, another distro
01:06.12neurosysheheh
01:08.14ManxPowerneurosys: no
01:08.21carrarI used to only use freebsd
01:08.28ManxPower*heart* Linux.
01:08.55ManxPowerBut Linux has some issues with high speed usb-serial, so I'm using FreeBSD.   I'm starting to hate their packet filtering stuff
01:09.17neurosysfbsd is my platform of choice.
01:09.24neurosysManxPower:  Which one are you using?
01:09.55ManxPowerneurosys: I use Linux for everything except my gateway, which is FreeBSD
01:10.31neurosysManxPower:  What are you using? IPW, IPFW, PF?
01:11.24ManxPowerneurosys: ipfw  I found that I needed option IPFIREWALL_FORWARD  So I'm recompiling again
01:12.16ManxPowerneurosys: think "captive garden",  all connections to port 80 from the local network with a destination that is not the local network be redirected to 127.0.0.1, port 81, where a web server takes over
01:12.28neurosysManxPower:  you really should try PF. It is by far the best. It was ported over from OpenBSD. Far superior.
01:12.30ManxPowerI need to allow ssh and irc thru the box with just basic nat
01:12.48ManxPowerneurosys: I've spent days learning about ipfw, I would rather not throw out all that work.
01:13.08ManxPowerneurosys: and PF is far, far, far too vague for my taste.
01:13.43neurosysManxPower:  Vague? How so?
01:13.44ManxPowerallow web not to me and to not me nat to local 1.2.3.4
01:13.55ManxPowersorry, that is not a filter that is "fisher price my first firewall"/
01:14.21ManxPowerBut I come from the iptables world where you are expected to understand networking before you try to set up a firewall.
01:14.30carrarIf you can deal with the bright colors it works great
01:14.57neurosysManxPower:  heh PF is much more robust than that.
01:15.03carrarred circle can only fit through the red circle fw rule
01:15.09*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu)
01:15.29carrarunless you paint the outside blue
01:15.31rob0grabs the red circle from carrar and runs off!
01:16.04[TK]D-Fendercarrar: But Miss the square peg DOES go in the round hole! *WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham
01:16.22ManxPowerI can configure iptables in my sleep but ipfw and the rest are just kicking my ass
01:16.58jayteeso just take a nap and use iptables :-)
01:16.59carrar[TK]D-Fender, nothing a little a soldering iron can't melt away some protection!
01:18.41jayteetries to calculate the risk factor in eating a Reese's peanut butter cup.
01:18.55*** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200)
01:19.21ManxPowerjaytee: Verizon EVDO on Windows = 1.5Mbps more or less.  Verizon EVDO on FreeBSD = 0.6Mbps   Verizon EVDO on Linux = 0.014Mbps.
01:19.22carrar1000:1
01:20.18jayteeManxPower, what's up with the shitty performance of EVDO on BSD and Linux? crappy drivers?
01:20.45[TK]D-FenderManxPower: Think their throughput math is bad.... jsut wait'll you look at their accounting ;)
01:20.48ManxPowerjaytee: that is my thought.  0.6Mbps is OK for my use almost all the time.
01:21.05[TK]D-FenderManxPower: just my .02 cents worth :p
01:21.47carrarpenny for your thoughts?
01:22.05ManxPowerjaytee: Almost all the USB Hight Speed (EVDO Cellular) internet devices act like serial devices.  The Linux usb/serial drivers are pretty horrid with some devices.
01:22.47drmessanoWhy not just use an EVDO router?
01:22.54carrarMy AT&T HSPDA works pretty good in the city
01:23.05carrarconsistant 1 Mbps
01:23.06ManxPowercarrar: on what OS?
01:23.10carrarOSX
01:23.13jayteeManxPower, what does your EVDO modem show up as in linux?
01:23.20carrarI should test it again
01:23.23carrarbeen a fewmonths
01:23.32drmessanoOSX isn't an OS.. it's a lifestyle
01:23.33ManxPowerjaytee: /dev/ttyU0.0 using the option.ko driver
01:23.37carrarhahah
01:23.39carrartrue
01:23.47carrarbut makes a smoken desktop
01:24.05drmessanoWell
01:24.10jayteeManxPower, what distro?
01:24.13ManxPowerWe'll be right back, after this reboot.
01:24.16drmessanoWindows XP does one thing OSX will never do
01:24.18ManxPowerjaytee: centos
01:24.23drmessanoRun on my PC
01:24.34carrarYou can put OSX on a PC
01:24.47drmessanoNot out of the box
01:25.23jksManyone using the siemens gigaset SIP-phones and got direct call _to_ them working?
01:27.49carrarok I'll try another speed test here
01:29.49carraryeah still the same
01:29.59carrar1 meg
01:30.48jayteeManxPower, what brand of modem is it?
01:31.41carrarMerlin XU870 3G HSDPA 7.2 ExpressCard
01:32.07jayteethat's his card?
01:32.14carraroh
01:32.15jayteeor yours
01:32.16carrarsorry
01:32.18carrarmine
01:33.00carrarGoes up to 7.3 Mbps
01:33.11carrartoo bad AT&T will never let me do that
01:34.06carrarpeaking out at 1.0768 Mbps
01:37.12[TK]D-Fendercarrwhat's your upstream?
01:37.20carrarwas just gonna test that
01:39.08carrarheh
01:39.14carrarnot so good :)
01:39.26carrarso a wget down does 1.096 Mbps
01:39.37carrara scp up is about 400 kbps
01:40.23carrar100 meg file
01:40.40carrarso it has time to level off
01:42.20carrarwe'll try scp down to see if they favor the web traffic
01:42.37carrarnaw looks the same down
01:43.20carraracceptable speeds
01:43.59[TK]D-Fenderwaits for ADSL2 to roll out in his area...
01:45.09carrarbonded ADSL2!!
01:45.26[TK]D-Fendercarrar: 1 at a time :)
01:45.33carrarheh
01:45.36[TK]D-Fendercarrar: I was considering bonded ADSL1
01:45.48carrarwe do bonded DSL
01:46.00carrarworks pretty slick
01:46.03[TK]D-Fendercarrar: I'm looking to get a better bang/buck on T1+ overall speeds
01:46.32[TK]D-Fendercarrar: anything >= to 1.544 up/down (sync not req
01:46.33carrardual 7meg DSL customers see about 12 megs down
01:46.42[TK]D-FenderCarUPSTREAM is what I care about
01:47.06carrar1.5 up
01:47.18carrartops
01:48.09[TK]D-FendercarrGAH!
01:48.17[TK]D-Fendercarrar: dang autocomplete
01:48.27[TK]D-Fenderanyway... off to play pool for a while.... BBL:
01:48.39carrardrink more!
01:48.42carrarhave fun
02:08.23*** join/#asterisk neurosys (n=vinix@c-71-196-9-142.hsd1.fl.comcast.net)
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03:15.44drmessanohave I mentioned in the last hour how much I hate mysql
03:16.05ekatwhys that
03:16.06ekat?
03:18.48jayteeno, you haven't mentioned it in at least two or three hours.
03:18.48drmessanomysqldump needs to be renamed to "dump some crap.non-restorable"
03:18.59ekatlol
03:19.14drmessanoIt doesnt replicate well
03:19.17ekatare you trying to restore a .sql file to a different mysql version?
03:19.20drmessanoand you cant back it up and restore
03:19.20*** join/#asterisk ipguy (n=ipguy@124-170-148-202.dyn.iinet.net.au)
03:19.21drmessanoNo
03:19.37jayteeI've used mysqldump to move data from a database on one server to a database on another with no problem.
03:19.40drmessanoIm trying to dump from one server to another
03:19.44drmessanosame version
03:19.50drmessanoSame OS
03:19.51ekatyea, thats weird them
03:19.54ekater then
03:20.09drmessanoSo I am unpacking a new, blank SQL DB
03:20.09ipguystupid question time, how do i change the  theme is trixbox ?
03:20.21jayteeyou ask in #trixbox
03:20.27drmessanoor go to their forums
03:20.45ipguyyep, just did
03:20.48*** part/#asterisk ipguy (n=ipguy@124-170-148-202.dyn.iinet.net.au)
03:21.06jayteeand how'd that work out for you? better than asking in here?
03:21.16ekattoo late
03:21.19ekatalready gone
03:21.50ekatdrmessano: does the new server support the same type of table types? innodb?
03:22.07ekatit should say where the .sql file breaks
03:22.10ekatthat should give a hint
03:22.16jayteetrixbox users, meh! they'd friggin glue sequins on the interface if they could
03:22.26ekatol
03:22.28ekatlol
03:25.48drmessanoI think I got it now
03:26.06drmessanoempty db for the win
03:26.26drmessanoI had 2 servers.. which I wanted to set up to replicate
03:26.29drmessanoand it never worked
03:26.39drmessanowell, worked for a month, then stopped
03:26.45drmessanoThen worked, stopped.. worked, stopped
03:26.53drmessanoDamn relay log corruption
03:27.19drmessanoSo I gave up.. its a hot standby.. just want to dump the DB and restore it once a day
03:27.24drmessanoBut couldnt even get that working
03:27.38ScribbleJWhat DBMS?
03:27.50ekatahh
03:27.53jayteewhat about using a cronjob to stop the daemon, rsync the files to another server and then restart the daemon?
03:27.56*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
03:27.58ekatyea i haven't messed much with replication lately
03:28.33drmessanoThats not a bad idea, jaytee
03:28.44drmessanoI had considered it.. and it would work well
03:29.16jayteebut I think you'd have to stop the daemon on the target server too but I think you could use rexec for that or a scripted ssh command line instruction.
03:29.34jayteemysql replication definitely sucks ass
03:30.09ScribbleJjaytee, hey, I wanted to mention to you... I built a 'north' 'south' 'east' 'west' grammar in my system.
03:30.15drmessanoNothing like feeling like you never REALLY have a backup
03:30.27ScribbleJjaytee, and... wouldn't you knwo it, it /can't/ get 'south' most of the time.
03:30.50ScribbleJBut now I'm on a mission to solve it and I've got some help from the CMU guys.
03:31.57jayteethere's just something "odd" about south.
03:32.35jayteeI guess I should be glad that it's not just Lumenvox
03:32.47ScribbleJYou're not kidding.  I even added several different pronunciations to try out
03:32.59ScribbleJSpeaking of which - I had better luck if I made a point of pronouncing the 'th' at the end.
03:33.08ScribbleJBut removing it from the pronunciation didn't help too much.
03:33.39drmessanoOk, got my MySQL dump moved over and inserted.. so thats good
03:34.32jayteeaccording to Lumenvox's analysis of my sre response files from test calls it seems that most of the time it was missing the S at the beginning or picking up noise. I think maybe the recognition engine mistakes sss sounds for noise and dumps it.
03:34.52ScribbleJHuh
03:34.57drmessanoDid you try OUTH
03:34.59ScribbleJI wonder if I set u...
03:35.08ScribbleJjust what  was going to say, basically, drmessano .
03:35.32ScribbleJ"OW TH" maybe.  I went the other way and tried "S OW" and "Z OW"
03:35.46drmessanoDump the S if its getting clipped as noise
03:36.02ScribbleJWell, I'll try that myself in a little while.
03:36.04jayteedrmessano, yep, that was Lumenvox's suggestion. I put in an outh and added phonetics for "AW TH" which is what their phonetic list suggests. Made recognition worse instead of better.
03:36.13drmessanoOW TH
03:36.23drmessanoAW TH is way different IMO
03:36.25jayteenot according to their phoneme list
03:36.36jayteeI'd agree with you though
03:36.37ScribbleJdrmessano, it depends on their phonome set..
03:36.40ScribbleJright.
03:36.46drmessanoheh
03:37.03drmessanoAW TH would be appropriate for NORTH, if youre from up NAWTH
03:37.17drmessanoCAW FEE
03:37.32jayteethe odd thing was after I added that to the grammar it wouldn't recognize east or west, just north and they came after south in the grammar list. I couldn't find any typos so I'm not sure what was going on.
03:37.38drmessanoGOT A CAW TER FOR SOME CAW FEE
03:37.43ScribbleJjaytee, that does sound odd, though...
03:37.58ScribbleJjaytee, I'll try it without the 's' on mine and see how it looks.
03:38.05drmessanoThat was our answering machine message my sister put on the home machine years ago
03:38.30drmessano"Sorry you wasted a CATWER on the CAWL but we're out getting CAWFREE right now"
03:38.35drmessanoFunny shit
03:38.51drmessanoCAWTER
03:38.53drmessanobah
03:39.30jayteeI just made a test call, if I stress it by saying sss oww thhh it recognizes it.
03:40.00drmessanoMaybe it the accent
03:40.02drmessanoits
03:40.28drmessanoSATH?   heh
03:40.30jayteeyeah, ya mean cuz I pahk my cah in the hahvahd yahd?
03:41.06drmessanoHey, you ova dare.. lemme come ova dare and wrap a wrench arand ya neck, ya little joik
03:41.26jayteeso for a southern drawl I'd do "sss ay ow th"
03:41.54drmessanoNah, you just leave off the South..
03:42.03drmessano"Over there in Carolina"
03:42.06drmessano"WHICH ONE?"
03:42.29drmessanoSad but true
03:42.36drmessano"Which direction will you be coming from?"
03:42.42jaytee"for directions please say the direction you are coming from unless it's south. If you're coming from the south say, "BUMFUCK!!!"
03:42.43drmessano"We're coming up there"
03:42.58drmessano"We're headed up yonder"
03:43.01drmessanoHA
03:43.06drmessano"down yonder"
03:43.08drmessanoThere you go
03:43.28drmessano"We're coming from down yonder"
03:43.50drmessanoI love the system Comcast has
03:43.57carrarWHAT
03:44.03jayteeI bet if I added "Fuck!" to the grammar but left it out the prompt after the 2nd repeat the average user would say, "Fuck!" and it would then go, "Coming from the south, take I-65 to exit 17 and head left....."
03:44.13drmessano"Ok.  Tell me what sort of problem you're having so I can direct your call"
03:44.18carrarSuch a POTTY mouth!
03:44.24drmessanoSo theres me at work
03:44.30drmessano"Shits broke"
03:44.39drmessano"Im sorry, i didnt understand your response"
03:44.43drmessano"Tell me what sort of problem you're having so I can direct your call"
03:44.47jayteehahahaa
03:44.50drmessano"Um, damn internets broke"
03:44.53drmessano"Im sorry, i didnt understand your response"
03:44.56drmessano"Tell me what sort of problem you're having so I can direct your call"
03:45.05drmessano"Cant surf, damnit"
03:45.16drmessano"Ok, let me connect you to someone that can help"
03:45.53drmessanoIt doesnt understand when I try something like "cant connect to the internet"
03:45.56drmessanoSo screw it
03:45.57ScribbleJI used to get the IVR at Comcast and say "fuck" and it would put me through to a real person immediately.
03:46.06ScribbleJDUnno if it still does, but it was a useful trick.
03:46.22drmessanoWe tried that
03:46.44ScribbleJIt's actually understood me every time... but it annoys the living shit out of me.
03:46.46drmessano"Customer service" doesnt work either.. nor does "representative"
03:46.51drmessanoor the name of the competition
03:47.02jayteejust say stuff like, "switching to U-Verse". bet that goes through really quick
03:47.10ScribbleJi suppose it's kind of funny I've been working on this speech API plugin when IVRs annoy me so much.
03:47.14drmessanoAlthough yelling "Knology" or "AT&T" over and over is fun
03:47.30ScribbleJI'd buy U-verse right now if they offered it in my area. :(
03:47.33drmessano"Tell me what sort of problem you're having so I can direct your call"
03:47.45drmessano"I hate you.. all of you... I want to... BURN you...."
03:48.09carrarjaytee is becoming the enimey!
03:48.15carrarerr
03:48.20carrarScribbleJ is
03:48.22drmessano"The world is burning.   RUN"
03:48.23carrarsorry jayteee
03:48.25carrarheh
03:48.36drmessanoThats one of my fav XKCDs
03:48.55drmessanohttp://xkcd.com/78/
03:48.57jayteemy favorite is the stove ownership one
03:49.32drmessanoMy all time fav is A minus minus
03:49.38jayteehttp://xkcd.com/418/
03:49.49drmessanoROFL
03:50.04drmessano"instead of office chair, package contained bobcat"
03:50.09drmessano"would not buy again"
03:50.37jayteehahahaaaa
03:50.37drmessanoI have that taped to the wall
03:50.53drmessanohttp://xkcd.com/325/
03:51.15drmessanoblack hat guy is always = WIN
03:51.32jayteeI loved the clip of an Ebay feedback page that had one entry that said, "If we were in prison together, I'd protect you in the shower! Highly recommend!!!"
03:52.27drmessanohttp://xkcd.com/538/   <-- This reminds me of Mr "I encrypt my config files, my hold music, AND my .call files /just in case/"
03:53.01drmessanoIts like "what if i walk up the console with a $6 thumb drive?"  "Oh, shit.. BRB"
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03:53.53jayteehttp://xkcd.com/276/
03:54.36drmessanoI love that one
03:54.51drmessanohttp://xkcd.com/440/
03:54.54drmessanoThat one rocked
03:55.25drmessano"she'd be alive if it weren't for you"  "Oh, god"
03:56.48jayteehttp://xkcd.com/111/
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03:57.47drmessanohttp://xkcd.com/350/
03:58.37drmessanoWho's a good virus?
03:58.39drmessanoYes you are
03:59.05jayteehttp://xkcd.com/186/
04:00.50phixxkcd <3
04:01.41jayteeok, this is from another channel: temugen> isn't it also the summation from 0 to infinity of (4/2n+1)(-1)^(n)
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04:01.53jaytee<jdong> I don't want to figure out what arctan(1) is in maclaurin series :)
04:02.04jayteewhat are the odds that these two have never had sex?
04:03.58phixlol
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04:18.30drmessanoROFL
04:19.04drmessanojaytee: dungeonmasters do not have levels
04:19.53giovanidrmessano: how would you know? ;)
04:20.36jayteelol
04:21.43drmessanoI have a +20 against your sarcasm
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04:43.16drmessanojaytee: Check my facebook status
04:43.59drmessanonevermind.. shit
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04:44.15StanManCanFender....
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04:45.19StanManCanI borked something and I can't figure it out for the life of me.... I just get silence when I dial out, verbose 3 doesn't show any errors, it's just...silence... Let me know if anybody's here and i'll pastebin my confs and debug
04:45.24jayteedrmessano, what about it?
04:45.53drmessanohang on.. I had a good status update and it borked it
04:47.29drmessanodanny defines "database" as "binary storage of important data in an unstable, and deletion prone format".
04:47.53drmessanoShould add "See also: MySQL is the MySUQ"
04:48.24drmessanoThe world is burning.  RUN.
04:50.10StanManCansymlink /usr/src/asterisk/contrib/init.d to /etc/init.d/
04:50.17StanManCanis that the right way to make asterisk run on boot ?
04:50.44jayteeStanManCan, what distro?
04:51.09StanManCandebian
04:51.42jayteeprobably, but if I were you I'd google it to make sure. On RHEL or CentOS it's chkconfig asterisk on
04:54.53neurosysCan asterisk handle SMS to cellular carriers?
04:55.04StanManCanhmm, excepting i don't have a /usr/src/asterisk..
05:04.49drmessanojaytee: on debian, depends how you hold your mouth
05:05.41jayteeI don't run * on debian or ubuntu or any of the debian based distros, just RHEL and CentOS
05:05.58sipy~book
05:05.59jboti guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
05:06.23jayteeand if you don't have a /usr/src/asterisk it probably means you installed from packages.....
05:06.30jaytee~wglwat
05:06.31jbotrumour has it, wglwat is well, good luck with all that
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05:08.08_abc_is there a tiny iso with asterisk on it ? the 50mb business cd kind ?
05:10.41rob0If so it would be tight, since asterisk alone would probably be more than half of that.
05:11.03rob0well, I'm not sure, I installed some extra sounds
05:11.46rob0mine is 26MB
05:12.35StanManCanyea i installed by hand
05:12.48StanManCandownload packages and installed
05:13.03StanManCaner.. the tarballs
05:13.43_abc_i heard of asterisk running in a wrt54gl. so there must be a way to squeeze it
05:14.04drmessanoApples and oranges
05:14.25jayteepackages and tarballs, two different animals
05:14.46drmessano1 is talking about putting the install on a 50MB CD, I think.. the other is talking about putting a compiled binary asterisk on a WRT54GL
05:15.00drmessanoNot even close to the same comparison
05:15.08_abc_awe and wonder, and this is not the one i saw, this is what i found now: http://www.kvaes.be/unix-linux/installing-asterisk-on-a-linksys-wrtg/
05:15.35_abc_i was talking about a live iso with asterisk on it
05:15.50_abc_the wrt54g has at most 16MB flash afaik, usually 4 or 8
05:16.25drmessanoand is also a completely different platform
05:16.30_abc_please put _abc_ in the line if you want me to react i am busy otherwise
05:16.46drmessanoNo, I cant be bothered
05:16.52_abc_drmessano, yes, mips binaries are larger
05:17.05_abc_drmessano, just _a TAB please
05:17.19rob0Um, I think better ettiquette is that people who ask questions are expected to read the channel.
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05:17.30drmessanorob0: No shit
05:17.35rob0and did I spell that right?
05:17.44_abc_i read the channel of course but /whois _abc_ will tell you i have a lot of channels to read
05:17.55drmessanoheh
05:18.17rob0http://en.wikipedia.org/wiki/Etiquette
05:18.21rob0I guess not
05:18.30_abc_netiquette :)
05:18.30rob0one of the few I always get wrong
05:18.49drmessanorob0: Sorry, if you are gonna post hyperlinks, can you please VNC to my workstation and type them in my browser for me
05:18.53drmessanorob0: Very busy here
05:19.02rob0Sorry drmessano, I will.
05:19.53neurosysI guess Email gateways are the best way to get my system to text cell carriers :P
05:20.05drmessanoYes
05:20.10drmessanoless hassle
05:20.25drmessanoThey're actually very reliable
05:21.15rob0How about inbound SMS, do you need a special service for that? I can email my cell, but it can't reply (no SMS-to-email gateway)
05:21.34_abc_is there a gateway from asterisk iax to S7 ?
05:21.42drmessanoYeah, Asterisk
05:22.11drmessanoIAX <> ASTERISK <> libss7
05:22.19_abc_asterisk does S7 ? i need to go read on it. i haven't touched a new version in 1 year
05:22.32_abc_ok, thanks
05:22.32QwellSS7, not S7.
05:22.36_abc_sure
05:23.23_abc_i wonder how hard it is to backport the wrt54 asterisk install to x86 tiny bootable iso. probably not hard.
05:24.02QwellO.o
05:24.31drmessanolol
05:24.33_abc_by backport i mean compile natively instead of cross and graft it on grml or similar ( http://grml.org/ )
05:24.55_abc_what is so funny
05:24.57drmessanoBy backport I mean, not using backport in any sense of the word
05:26.02QwellSo you want to install it normally
05:26.24_abc_well wrt54gl oopen source firmware is a cross compiled set of packages that should be compilable for x86 as is just leaving out the platform specific tools. the core is busybox based anyway.
05:26.39_abc_and no, it is not so normal imho.
05:28.15neurosysdrmessano:  smsm email gateways seem to be very quick too. Curious how quick duing peek hrs.
05:29.08drmessanoneurosys: About the same.. I get a weather bulletin out to phones in about the same time as e-mails hit standard inboxes
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05:54.15_abc_there is even a version for nslu2 apparently
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05:54.47[TK]D-Fender_abc_: Don't expect performance after something nick-named a "slug"
05:54.58_abc_hehe
05:55.05_abc_just reliability would be nice
05:55.38[TK]D-Fender_abc_: Buy a 10$ phone at the drug store and run it off your telco's analog service.  Ther you go
05:56.19_abc_lol
05:56.23_abc_ok, i get the spirit
05:56.56drmessanoMake it look like a sunflower
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05:57.54drmessanoEffin Steve Jobs
05:58.44drmessanoSad they diagnosed him with Cancer.. That just shows me how crappy medical science is.  So much they had to have missed.
06:01.18[TK]D-Fender"We don't what's wrong" = cancer
06:07.26neurosysHe's been officially diagnosed with Cancer?
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06:11.28drmessanoLike 2 years ago
06:11.43drmessanoHe's had pancreatic cancer
06:11.52neurosysOh yeah. but didnt that go into remissio or something? Is that what the problem is again?
06:12.44drmessanoRumor has it he's HIV positive.. but so far just a couple liveleak documents that may be forged
06:12.56neurosyshehe
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06:24.13baliktadAsterisk 1.6.0.5 is crashing when I park a call: http://pastebin.ca/1330559
06:24.59neurosysdrmessano:  LOL you still awake?
06:25.15drmessanoyes
06:25.22neurosyshttp://www.aim-med.org/library/articles/1192638115/
06:25.27neurosysfounded in 2006 hehe
06:25.50neurosysthose leaks say that company tested him in 04' :P
06:28.34drmessanoThe company has been around for years.. the claim is that they are a reissue of old results
06:30.05drmessanoIm not saying I necessarily believe the documents to not be fake, but issue of the age of the company is the weakest argument
06:31.07neurosysOh, saying the documents are simply under a new letterhead?
06:31.24neurosysi gotcha. well.. i hope for his sake it's not true :P
06:33.10drmessanoWell, I honestly hope it's not.. Him covering up having AIDS with a pancreatic cancer story would tell me he's still living in the 1980s..
06:33.43drmessanoHe may as well at that point issue a press release "I have AIDS... and I am not gay"..
06:33.53drmessanoHe'd get the drmessano WTF of the year award
06:34.06neurosysToo right. :)
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07:03.02Zippomananyone ever used any sms gateways...to recommend?
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07:53.58DelphiWorldhi my friends. how are you ? please, try to give me a book (PDF or CHM format) about asterisk PBX. Thanks!
07:54.16carrar~book
07:54.17jbotbook is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
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08:18.45DelphiWorldmy friends: please give me a asterisk book in CHM format if pocible
08:24.00MaliutaLap~bool
08:24.01jboti heard bool is bool, nothing else need be said.
08:24.04MaliutaLap~book
08:24.05jbotrumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
08:24.12MaliutaLap'tas all
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08:37.00DelphiWorldMaliutaLap: are you able to download the PDF book ?
08:37.52MaliutaLapyes, I have copies everywhere
08:39.34DelphiWorldplease, is it pocible to send to me a copy ?
08:39.45DelphiWorldbicose i'm no able to use it ofline
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09:37.41estr4ng3dHow do you enable zap debug?
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09:48.54ScribbleJI zap debug with de bug zapper.
09:53.09*** join/#asterisk DelphiWorld (n=Miranda@41.201.96.184)
09:53.14DelphiWorldhi my friends
09:53.51DelphiWorldplease, what is the realy best linux distribution to run asterisk ? (simulare to TrixBox)
09:54.06Maliutathere are no friends, only people screaming RTFM!
09:55.07DelphiWorldMaliuta: i love open source users. then this is my friends.
09:55.19ScribbleJBSD, this is the best linux distro for Asterisk.
09:55.26MaliutaDelphiWorld: run slackware
09:55.33ScribbleJ<- real jerk.
09:55.34ScribbleJHeh
09:55.37Maliutaor build your own distro
09:56.51DelphiWorldScribbleJ: BSD dont have asterisk by default
09:57.03DelphiWorldMaliuta: please ho to build my hone ?
09:57.20DelphiWorldi'm next to Linux from scratch but is no easy for me
09:58.27Maliutadistro is all about preference
09:59.36carrarDelphiWorld, you should use LFS (Linux From Scratch)
09:59.38DelphiWorldMaliuta: please try to help my friend to bring me to the open source community
09:59.43DelphiWorldi'm using only a windows PBX
09:59.49DelphiWorldbut i want to switch to linux
10:00.21ScribbleJIn this case, I suggest something user-friendly, like Ubuntu, or something with paid corporate support, like RedHat.
10:00.51carrarCentOS works very well also
10:01.00DelphiWorldScribbleJ: you dont understand me
10:01.13ScribbleJCentOS is the paid of RedHat without the support.... :(
10:01.14DelphiWorldScribbleJ: i want a distro simulare to trixbox (if you know it)
10:01.18ScribbleJpaid = pain
10:01.29ScribbleJDelphiWorld, Sorry, I don't.  Why this requirement?
10:01.37carrarCentOS works awesome
10:01.56carrarWho cars about paid
10:01.58carrarcares
10:02.01DelphiWorldi want to start a very very small PBX a my home
10:02.05ScribbleJI'm sure it works great... I have personally issues with Redhat's package management and naming scheme.
10:02.15ScribbleJDelphiWorld, I do this.  I use Ubuntu.
10:02.15carrarlike you are gonna get commercia support for opensource Asterisk
10:02.40ScribbleJcarrar, I believe you can get that all day long from Digium.
10:02.46carrarnope
10:02.55carrarThey will not support open source
10:03.02DelphiWorldcarrar: do you know trixbox ?
10:03.04carraronly their Buisness eddition
10:03.08ScribbleJOh, I see.
10:03.20ScribbleJHrm
10:03.34carrarDelphiWorld, you might try #trixbox
10:03.39ScribbleJThat makes me wonder if we bought Business Edition for our IVR.  Guess we must have.
10:04.14DelphiWorldi dont wabnt to use trixbox
10:04.20carrarthen stop talking about it
10:04.24DelphiWorldui'm looking for other distro simulare to it
10:04.31carrarthen use it
10:05.39DelphiWorldcarrar: realyty: i want to use a distro based on debian no CentOS (RedHat)
10:05.57carrarI use Asterisk OpenSource on CentOS
10:06.01ScribbleJDelphiWorld, so what was wrong with my Ubuntu suggestion?
10:06.04carrarworks great
10:06.14ScribbleJI use it on Ubuntu, and Debian both, works fine.
10:06.15carrarUbuntu works great too
10:06.21carrarthey all do
10:06.29DelphiWorldScribbleJ: your ubuntu suggestion is realy cool. but asterisk is no pre configured!
10:06.30carrarDelphiWorld, it's personal pref
10:06.37ScribbleJDelphiWorld, wrong, it is too.
10:06.51carrarif you go with pre configured why bother with Asterisk open source?
10:06.52ScribbleJDelphiWorld, just click to install it in the package manager, comes complete witha  working same config.
10:06.56ScribbleJsame = sample
10:07.05carrarYou are missing a lot of things
10:07.37DelphiWorldScribbleJ: but include a web interface (frontend) ?
10:07.37carrargoogle for building asterisk from source
10:07.50carraragain, wrong channel
10:08.16DelphiWorldcarrar: then what you suggest to me ?
10:08.28carrarwhat are your requirements?
10:08.45ScribbleJThere is no web interface for ASterisk in Ubuntu, but honestly configuring it from the config files is not rocket surgery.
10:08.51DelphiWorldcarrar: please heare me:
10:08.57DelphiWorld1. a small linux distro
10:09.00carrardon't waste your time with web front ends
10:09.11DelphiWorld2. setting up asterisk for home usage
10:09.25DelphiWorld3. a web interface (frontend) to configure my asterisk
10:09.32ScribbleJ1. Ubuntu Server.  2. aptitude install asterisk
10:09.38ScribbleJ3.  Doh!
10:09.43*** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com)
10:09.51carrarDelphiWorld, why do you need web?
10:09.59carrarWhy can't you learn to you command line?
10:10.04carraruse
10:10.10brunnerI just got a phone call through IPKall, but they haven't told me my phone number yet!
10:10.26ScribbleJbrunner, so, who was calling?!? Ar telemarketers THAT good now?
10:10.33DelphiWorldcarrar: probably you dont know me
10:10.41DelphiWorldcarrar: i'm a blind user using a screen reader
10:10.45brunnerthe call was from sip:4153385516@66.54.140.46
10:10.48ScribbleJWhat!
10:10.50ScribbleJWhat!
10:10.53DelphiWorldi want to access / configure asterisk from windows
10:10.55ScribbleJI'm sorry, I'll say it once more.
10:10.56ScribbleJWhat!
10:11.01DelphiWorldthis is the frontEnd Usage reason
10:11.17ScribbleJDelphiWorld, you're the first blind person I ever met who would prefer a web browser to a commandline.
10:11.26brunnerI can't call them back through IPKall, of course, but I tried with my cell phone and it said it was an invalid pager number
10:12.07ScribbleJUsing a web browser with a screenreader or braille terminal is a nightmare.
10:12.37DelphiWorldScribbleJ: i'm using only TTS no brail terminal
10:12.43carrarssh into your asterisk box from your windows box that has a screen reader
10:12.52carrarproblem solved!
10:13.39ScribbleJDelphiWorld, yeah, I'm sighted, but I notice my blind friends prefer TTS to the braille terminal.  It's just faster, I suppose.  Do you use somehting like FEstival only with the speed cranked up to chipmunk-speed?
10:13.43DelphiWorldcarrar: SSH i know only the populare "PUTTY" SSH client but no accessibkle to me!
10:14.39carrarTry SecureCRT
10:14.45DelphiWorldScribbleJ: i know a screen reader for the GNOME Desktop named "orca" that use ESpeak
10:14.50ScribbleJDelphiWorld, there are plenty of ways to set up Linux for blind folks; like I mentioned, FEstival can be used for TTS.  You might try it all sometime, it's a better intro to linux than ASterisk, and once you go tthat down, ASterisk'll seem way easier.
10:15.26ScribbleJSorry, "once you got that down" - I bet parsing typos on a TTS is a real bother.
10:15.30DelphiWorldcarrar: ho to download it ?
10:16.28DelphiWorldScribbleJ: i want to use ubuntu server, but dont have a screen reader included
10:17.04ScribbleJDelphiWorld, I can imagine getting it set up initially might be a bitch.  Maybe get a sighted linux nut to help out.  I'd think it would be an interesting project.
10:18.55DelphiWorldScribbleJ: what you thank if i start a new linux distro that have the GNOME desktop and a pre included ORCA screen reader to setup it ?
10:19.16DelphiWorldbicose ubuntu have orca, but not working if i start the setup program
10:19.35ScribbleJDelphiWorld, I think that would be cool, but possibly missing the boat.  If I were blind, I would imagine I would like the fact that I can accomplish everything in linux from a non-graphical environment.
10:19.43*** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net)
10:20.11DelphiWorldScribbleJ but no easy
10:20.42ScribbleJDelphiWorld, I wonder how easy it would be; in theory it couldn't be much more complicated than piping your tty through festival.
10:21.04ScribbleJDelphiWorld, or you mean, the linux commandline isn't easy? Heh
10:21.19ScribbleJDelphiWorld, All I can say to that is, it's easier than Windows once you know it as well.
10:21.25DelphiWorldScribbleJ: faistival is only a TTS
10:21.32DelphiWorldrequire a program to use faistival
10:21.55ScribbleJDelphiWorld, yeah, I'm surprised one isn't out there already for that though; seems like an obvious application of it.
10:22.36DelphiWorldorca use ESpeak by default but is pocible to setup faistival and configure it to use it
10:22.57ScribbleJEspeak might be better for all I know, I'm not familiar with it.
10:23.24DelphiWorldScribbleJ: no
10:23.29DelphiWorldfaistival is best
10:24.10ScribbleJDelphiWorld, I heard 'cepstral' recently, which is a clone of festival, but with better data, and it costs money to buy... but holy cow does it sound nice, almost like a real person.
10:24.47DelphiWorldScribbleJ: but probably that dont have a driver for GNOME
10:24.56ScribbleJDelphiWorld, Yeah, probably, I have no idea.
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10:25.35DelphiWorldScribbleJ: what about asterisk billing ?
10:25.38DelphiWorldhow is work ?
10:26.20ScribbleJI don't know much about it, actually.  My involvement with Asterisk so far has been limited to a little bit of SIP and a lot of TTS and Speech Recognition.
10:27.15DelphiWorldthen cool!
10:27.23DelphiWorldi dont know any informations about asterisk
10:27.28DelphiWorldi'm only a new user
10:29.24DelphiWorldScribbleJ: i try trixbox and tel you later
10:29.30ScribbleJGood luck!
10:31.03DelphiWorldScribbleJ: thanks
10:47.16DelphiWorldplease what is a zapcart ?
10:54.30brunnerdoes anyone have an IPKall number I could use for testing?
10:56.01DelphiWorldbrunner: do you want to make a call ?
10:56.33brunnerDelphiWorld: no... I just to be able to receive a call on my asterisk box
10:56.59DelphiWorldbrunner: then you want to i call you ?
10:57.47brunnerDelphiWorld: well it would make testing easier if I could call it myself more than once as needed.  I'll just wait for IPKall to process my application or sign up with a voip provider.
10:57.54DelphiWorldsend to me your sip uri
11:03.04ScribbleJBrunner, I tried voicepulse and broadvoice... broadvoice'll turn you on the minute you subscribe and if you never sign their 911 form they'll autocancel your account after a month and give you a refund.
11:03.28ScribbleJI was pretty impressed with them to be honest... but I stuck with voicepulse instead.
11:05.08brunnerScribbleJ: so this is unlimited in-bound, right? http://www.broadvoice.com/rateplans_unlimited_state.html
11:05.34ScribbleJbrunner, both voicepulse and broadvoice are unlimited inbound - the niice thing about broadvoice is they also offer unlimited /outbound/
11:05.49brunnernot to be picky, but is there some low-quality provider that offers even cheaper inbound service that I could use for testing?
11:06.10ScribbleJNo, that's not picky.  I do not know personally, those are the only two I have used.
11:06.35ScribbleJHow long do you need it for, brunner?
11:06.48brunnera week?  maybe a month?
11:07.00ScribbleJAh, yeah.  Well.
11:07.30ScribbleJI signed up for broadvoice, paid the fee, never signed their 911 form they require, and they refunded me the full amount.
11:07.31brunnerIPKall is probably the only thing out there.  if they don't respond in another day or so, I'll just use broadvoice
11:07.40ScribbleJSo... doesn't get much freer than that.
11:07.48brunneryeah, that's cool
11:07.55brunnerI appreciate the heads pu
11:07.57brunnerup*
11:08.04ScribbleJIT's kind of taking advantage to do that on purpose I guess, but... it is what it is.
11:08.17brunnerI'll just pay the $10, heh
11:08.29brunneror do they require a contract?
11:08.56ScribbleJI don't remember if they want you to sign up for long-term, I think it was month-to-month.
11:09.03ScribbleJI never made it past the first month myself, of course. :)
11:10.17brunneryeah
11:10.34brunnerwell I don't mind paying the $10 for a month of service, as long as there's no contract
11:10.36ScribbleJI'm going to get a 2nd number from VoicePulse tongiht, actually, to put on my website as a demo of this Speech Rec system I'm working on.
11:11.20brunnerhell, I might just hack my roommate's vonage adapter and get him to switch so I can make use of the sip line at night to test asterisk
11:11.39brunnercool. what's your website?
11:12.20ScribbleJIt's scribblej.com - there's nothing there yet but check back in 6 hours and you'll either find a free replacement for LumenVox.... or... I'll be dead, I suppose, could happen anytime.
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11:13.01brunnerk
11:13.05ScribbleJVonage adapter, hah.  I /just/ got a Linksys PAP2T-NA today, haven't had time to do more than hook it up.
11:14.14brunnerwoah, perfect: http://www.broadvoice.com/rateplans_byod.html
11:14.37ScribbleJOh yeah, that's the plan I signed up for I guess.
11:15.00brunneroh, I didn't see all the fees
11:15.15ScribbleJOh, except I did BYOD-Unlimited-World, I think
11:15.17brunnerthat sucks.  I wish IPKall would just hurry up and approve the application
11:15.54ScribbleJHuh
11:16.15ScribbleJI haven't heard of IPKall but I've been looking for some way to get a SIP termination for free...
11:16.20ScribbleJHow can they do that??
11:16.40brunnerterminative minutes
11:16.58brunneron the PSTN, whoever the call terminates with gets paid
11:17.03brunnerby the other telcos
11:17.07ScribbleJOh, I see.
11:17.17ScribbleJSo basically they are gambling on more inbound thn outbound?
11:17.18brunnerso they actually make money on it
11:17.28brunnerthey won't do outbound at all
11:17.31ScribbleJThat's pretty clever.
11:17.31brunnerexcept toll free
11:17.32brunnerbbr
11:17.57ScribbleJHUH!
11:18.05ScribbleJI wonder what it takes to set up a system like that.
11:18.14ScribbleJThat would be a fun service to provice and 'free money'
11:18.40brunneryeah
11:18.48brunnerthey don't offer customer support at all
11:19.02ScribbleJRight, that kind of thing, you can basically set it up and forget it.
11:19.06ScribbleJAutomate the whole thing.
11:19.20brunneryeah, until the spammers discovered it
11:19.21ScribbleJI'd have no idea where to start on the PSTN side though.
11:19.29brunnernow there's a long waiting period
11:19.54ScribbleJThere's ways to try to automate some of that anyhow... we automate background checks and things for our business, but of course that gets cost prohibitive.
11:20.26brunnera company in Pennsylvania is basically letting me use a quad xeon asterisk box for the same reason
11:20.30ScribbleJStill, you could do a some simple tests... and ACH is cheap, depending on how profitable it is you could ACH a coupel pennies to each person to verify them.
11:20.39brunnerjust so they make money on the inbound calls, I mean
11:20.49ScribbleJOf course that takes at least a day.
11:20.52brunnerand it's not costing me a penny
11:21.24brunneryou wouldn't need to do ACH
11:21.41brunnerjust require they confirm they have a real alternate phone number in the US
11:21.54ScribbleJWell, you could automate /that/ easy with ASterisk.
11:21.58brunneryou call them, they enter a code online, and it's done
11:22.02brunnerright
11:22.02ScribbleJGive them a code # to punch in, cl... yeah
11:22.07ScribbleJHeh
11:22.38ScribbleJI don't get how you have to be hooked into the phone network to a) get phone numbers to give out and b) get on the dole for incoming calls.
11:23.45brunneryeah, I don't really understand it, either... I'll have to ask my friend about it again sometime
11:24.01brunnerhe works for a phone company that makes money just on that
11:24.14brunnerthey offer free conference services and such
11:24.14ScribbleJI'd guess you have to pay someone something for the block of #s up front, I wonder if you could optimize that though - people are getting a free service, maybe they'd be cool with sharing numbers and only getting an /extension/ on your automated phone tree.
11:24.38brunnernah, numbers are super cheap wholesale
11:25.32brunnerthere'd be no need for sharing
11:25.45ScribbleJHrm...!
11:25.50*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
11:25.52ScribbleJI'm going to have to find out more about this.
11:25.58*** join/#asterisk estr4ng3d (n=sm4rt@196.219.96.141)
11:26.13ScribbleJIf it didn't require too much capital I might do it myself.... and if it did I know some people I could pitch it to.
11:27.18brunnerI really don't think it's worth it
11:27.37brunneryou'd have to physically exist in a state with screwed up regulations
11:27.37ScribbleJNot that much money in the incoming calls?
11:27.41ScribbleJHah
11:27.44brunnerand then you'd have fraud out the ass
11:28.02brunnerit just really wouldn't be worth it
11:28.12brunnerotherwise, there'd be a lot more people doing it
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11:42.51estr4ng3dHow do you enable zap debug?
11:49.04MaliutaLapyou zap de-bug wiss de bug zapper
11:49.25MaliutaLaptakes loose interpretations
11:56.31ScribbleJDude.
11:56.39ScribbleJI so made that joke hours ago.
11:57.08MaliutaLapScribbleJ: I improved on it
11:57.15ScribbleJ2 hours ago: <estr4ng3d> How do you enable zap debug?  <ScribbleJ> I zap debug with de bug zapper.
11:57.26ScribbleJHah, sheesh
11:57.40MaliutaLapScribbleJ: I _could_ have told him the compiler options for gdb
11:57.58MaliutaLapScribbleJ: bah! _your_ telling was weak :P
11:58.03ScribbleJI suppose actually being helpful is beyond either of us.
12:01.09MaliutaLapfor something documented?
12:01.37MaliutaLapif people actually read some manuals and tried things on the CLI ...
12:02.22*** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com)
12:02.29SparFuxregular phones suck! I wish I only had IP phones :-( All a mess here in my stupid house!
12:02.59MaliutaLaphugs his cisco
12:03.24MaliutaLapI scared a troll off on #linpeople!
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12:24.33markitI'm looking for a gsm plugin for kaffeine or other player in GNU/Linux, any idea where to find it?
12:25.02markit(debian sid)
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12:29.36[gnubie]anyone knows if there is a repository on binary debian etch package for asterisk 1.4.23.1?
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12:54.15nvrpunkwhich sound is the ringback sound in /var/lib/asterisk/sounds ?
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13:02.05Dovidis the presentation number on a PRI the same as CID on callerid(num) on sip ?
13:09.24nvrpunkhow come when I have a voice prompt context for inbound calls to dial an extensin, the ring isn't passed to the caller?
13:09.35nvrpunkie it's blank until the person answers their extension
13:12.33Maliutanvrpunk: because you didn't pass the right option to dial?
13:12.44Maliutanvrpunk: because you don't have any tones on the system?
13:13.00Maliutanvrpunk: because the planets are out of alignment?
13:13.16Maliutanvrpunk: you haven't given us anything to base our advice on
13:13.29nvrpunkit
13:13.42Maliutalets start with some dialplan action\
13:13.48nvrpunkits and AGI Script setting the voice prompt system
13:13.53nvrpunkit's not just a config
13:14.04Maliutathen we need to see the script
13:14.14nvrpunkif you would like to see 10 pages of php sure
13:14.15nvrpunk:)
13:14.22Maliutait's that simple, you give us info an we'll see if we can help
13:14.25Maliuta~pastebin
13:14.26jbot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
13:14.44nvrpunkill be back, im going to struggle with it for a bit
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13:14.47nvrpunkbefore I give up
13:14.48nvrpunk:P
13:15.18nvrpunkI do have one question though
13:15.21nvrpunkthat should be simple
13:15.22nvrpunk[Feb  8 16:06:35] WARNING[20997]: channel.c:2930 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
13:15.25Maliutalook at the manual for dial
13:15.35Maliutanvrpunk: that's simple
13:15.36nvrpunkI don;t know what it's trying to transcode or why
13:15.43nvrpunkthere should be nothing
13:15.48Maliutanvrpunk: you don't have g729 installed
13:15.48nvrpunkto rings have to be transcoded?
13:15.54nvrpunkof course not
13:15.59Maliutaeverything has to be transcoded
13:16.02Maliutawhy not?
13:16.02nvrpunkim using g729 all the way through
13:16.04nvrpunkno
13:16.13Maliutawhat codec are the files in?
13:16.19nvrpunkg729
13:16.20nvrpunkall of em
13:16.21nvrpunk:P
13:16.29Maliutaobviously not
13:16.41Maliutaotherwise it wouldn't need to transcode
13:16.42nvrpunkbut its not pointing me to what its looking for
13:16.47nvrpunkthats not g72
13:16.51nvrpunkwhich is bullocks
13:17.10nvrpunki cant fix something without proper debug messages!
13:17.11nvrpunk:P
13:17.43nvrpunkso how am I supposed to find which file is not g729?
13:17.44nvrpunk:p
13:20.16Maliutaand the output of 'module show like sln' is???
13:20.43Maliutait is a useful message if you use you brain
13:21.07nvrpunk1 modules loaded
13:21.19nvrpunkraw sign linear audio support
13:21.25nvrpunksigned*
13:21.48nvrpunkif I wanted to use my brain I would never use a computer :)
13:24.53MaliutaI don't have time for this, I have to be at the hospital in 7 hours for my hip replacement
13:28.02*** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com)
13:30.29ScribbleJVery beta version of the free Asterisk Generic Speech API engine I've put together: http://scribblej.com/svn/
13:31.56ScribbleJUh, that's not actually a subversion repo... maybe I should rename that directory.
13:55.19estr4ng3dHow do you enable zap debug?
14:02.29nvrpunk- Executing [11076@makecall:1] Dial("IAX2/jackal-8426", "SIP/192.168.2.8/11076||S(6000)r") in new stack
14:02.37nvrpunkit's still not rnging back to the caller
14:03.19*** part/#asterisk markit (n=marco@88-149-177-66.static.ngi.it)
14:04.15Daejeonvrpunk: if you want use computer, you also need to use your brain
14:05.00nvrpunkhmm
14:05.18nvrpunkso if i have a gotoif that passes a call to another context that has a dial
14:05.22nvrpunkand using the r option
14:05.26nvrpunkwill it pass back the ring?
14:05.49nvrpunkwhat I am really wondering is what stops the ring from passing back to the caller
14:05.52nvrpunkin general
14:05.58nvrpunkwhat specifically stops it, ever
14:06.41nvrpunkDaejeo, and as I don't have a lot of time to do all the things I need to do in a day and considering my salary is quite high, this is just a side thing to figure out
14:06.44nvrpunka minor annoyance
14:06.58nvrpunkand in the end, I could just pay someone to do it :)
14:07.18nvrpunkmoney > brain power
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14:22.19[gnubie]tzafrir_laptop: hello.. are you already freezing to version 1.4.21.2?
14:22.34tzafrir_laptophi
14:22.37tzafrir_laptopprobably
14:23.11[gnubie]tzafrir_laptop: what about the fixes made from the upstream from 1.4.22 and the current one?
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14:45.36masusHi all, Newbie Question. Is it possible to get the dialstatus if we place a call to /var/spool/asterisk/outgoing ? Thanks All.
14:50.47masusOr how to handle outgoing.call files ?
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15:38.16Dovidanyone use the $ams-> "function" with phpAgi ?
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15:55.33markithi, I'm updating italian suond set for asterisk 1.4.x (previous was 1.2.x). I need some help in understanding the meaning of some sounds. conf-adminmenu-162.gsm, what is the meaning of "extend the conference"? let more people enter, or last more time?
15:55.42markit(never used conferences myself, sigh)
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16:07.24*** join/#asterisk elux (n=pak@24.114.233.2)
16:07.26eluxhey guys
16:07.58eluxim trying to understand what asterisk does in the voip-stack. i understand its at the lowest level, which is like an SIP-daemon?
16:08.01eluxam i getting that right
16:12.02[gnubie]waves to all.. gtg.. thanks.. ;-)
16:14.10*** join/#asterisk LtScarr (i=benno@palm.hoeg.nl)
16:15.02LtScarrey everyone
16:15.10eluxhi
16:15.36LtScarris this a right channel to ask end-user questions about asterisk?
16:15.51LtScarror should i do that somewhere else?
16:16.12eluxbeats me. first time im here. but im sure your free to ask questions but it doesnt seem like anyone is around
16:16.37LtScarroh heh :)
16:16.47LtScarrah well
16:16.51LtScarri can try then
16:17.19markitwhat is a "representative"?
16:18.15LtScarri can seem to be able to "dial" from an analog phone
16:18.20nvrpunkdevs are in here
16:18.21nvrpunk:P
16:18.22nvrpunkso yeah
16:18.29LtScarri can make calls to it
16:18.32LtScarrbut not from it
16:18.36nvrpunkthere are people who know an awful lot about asterisk
16:18.38LtScarri do get a dial tone
16:18.46nvrpunkasterisk -r
16:18.49nvrpunkcore set verbose 9
16:18.52nvrpunkcheck the output
16:18.55LtScarrbut when i dial a number
16:18.57nvrpunksee where it's hanging
16:19.04LtScarrit just keeps giving a dial tone
16:19.22nvrpunksounds like your contexts aren't right
16:19.32nvrpunkand that it's not grabbing the number at all
16:20.01LtScarrwell what i did
16:20.09LtScarri backupped the default config
16:20.14nvrpunkpastebin your configs
16:20.17nvrpunkand post the url here
16:20.19LtScarrand then stripped everything
16:20.20nvrpunkill look at em
16:20.30LtScarrmaybe a bit too much :)
16:20.40LtScarrokay hold on a sec
16:21.18nvrpunkoddly, I can't get a ring to passback between openser and asterisk
16:22.05eluxwhat kind of provider do i need to make/accept multiple calls with asterisk?
16:22.20eluxi guess there are many channel protocols supported, im looking at SIP or IAX2
16:22.33nvrpunkpstn gateway
16:22.35nvrpunkis the norm
16:22.48LtScarrhttp://pastebin.com/d62563ece
16:22.56LtScarrthose are my extensions
16:23.09eluxso i hook up * to a pstn, and then enable access to my services (ie. adhearsion) to be able to connect via SIP?
16:23.27LtScarrhttp://pastebin.com/d498c3f5
16:23.34LtScarrand that's my SIP config
16:23.41eluxthe PSTN will take my asterisk exchange and link it to the global telephony system?
16:23.51eluxso i can make calls through it etc.?
16:24.34LtScarrow and i configured my router to be the ATA
16:24.52LtScarrand i registers with asterisk
16:28.26eluxwhat is the difference between having my clients connect directly to a SIP using some provider or making my own asterisk box that connects to a PSTN? .... i guess its just cost?
16:30.16markittrying to translate asterisk sound files: is there some subtle differences between agents and representatives?
16:30.33nvrpunksorry back
16:30.37nvrpunklooking at your configs
16:31.07LtScarrthanks
16:31.10eluxim trying to wrap my head around how voip/asterisk works
16:31.16eluxnvrpunk: have what i been saying makes sense?
16:31.51nvrpunkone min
16:31.53nvrpunklemme read
16:31.57nvrpunkdoing multiple things at once
16:32.15eluxthx
16:32.17nvrpunkelux, the differences vary
16:32.24nvrpunka pstn allows you to have a DID
16:32.33nvrpunkyou can do what a pstn does with additional cards
16:32.42nvrpunkand a trunk into your home
16:32.54eluxwhat is a DID?
16:33.01nvrpunkDirect Inward Dial
16:33.07nvrpunkcheck out voip-info.org
16:33.13nvrpunkit has a lot of useful info
16:33.17nvrpunkthat you should read through
16:33.26eluxi simply want to use make/accept multiple calls using voip through a SIP/IAX2 client
16:33.49dweryhello. with a sip channel with call-limit = 1, I need to be able to st the cause code that goes back to chan_lcr to busy. is there a way to do it?
16:33.54nvrpunkyou would think this helicopter above my building is looking for my rogue radio station
16:33.55nvrpunkhmm
16:34.19LtScarr:)
16:34.24nvrpunkLtScarr, where is your outbound calling context?
16:34.28nvrpunki dont see one at all
16:34.40nvrpunkor are you trying to make calls from phone to phone
16:34.40LtScarrthere isn't any :)
16:34.45LtScarrjust internal calls
16:35.29LtScarri'm trying this as an experiment for future implementations
16:35.40nvrpunkok so you have extension 1 2 and 3
16:35.44LtScarryes
16:36.33*** join/#asterisk hi365_m (n=hi365@85.130.230.240)
16:36.37LtScarrtwo of them are softphones
16:36.40nvrpunkLtScarr, lemme point you to http://www.asteriskguru.com/tutorials/
16:36.41nvrpunk:)
16:36.47nvrpunkgreat place to start
16:36.55nvrpunkalso, as I said asterisk -r
16:37.00nvrpunkcore set verbose 9
16:37.16nvrpunkwatch the asterisk server handle the call :)
16:37.24nvrpunkit's better than just giving an answer hehe
16:37.36LtScarrok :)
16:37.52nvrpunki dont see your sip.conf either
16:37.57nvrpunkwhich i would need to see
16:38.12LtScarrwell i did paste 2 urls
16:38.16nvrpunkoh
16:38.17nvrpunksorry
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16:39.33LtScarrone note though about the tutorials online
16:39.33eluxso i can only get a phone number and accept inward calls from the global telephony network if i have DID?
16:40.05LtScarrthey don't start with zero configuration
16:40.42LtScarrso it's hard to see what is needed and what isn't
16:44.01LtScarrokay i think this problem isn't asterisk related
16:44.17LtScarri think my ATA isn't functioning the way it should
16:44.41LtScarrso thanks for the time :)
16:46.57*** join/#asterisk jtodd (n=jtodd@jetblue.colorbroadband.com)
16:46.57*** mode/#asterisk [+o jtodd] by ChanServ
16:49.28*** join/#asterisk yondaime (n=Yamato@unaffiliated/yondaime)
17:01.48*** join/#asterisk joako (n=joako@99-153-162-33.lightspeed.miamfl.sbcglobal.net)
17:05.51nvrpunkhaha, its fun to call from a cell phone in iraq to our pstn in the states which then routes to our asterisk box in sweden and then to our asterisk box in iraq
17:06.01nvrpunkwalkie talkie ftw
17:06.04dweryis there a way to force a sip peer to use plaintet authentication?
17:07.03*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
17:07.17*** join/#asterisk Khratos (n=Khratos@190.166.111.76)
17:07.18nvrpunkhmm, http://books.google.com/books?id=9_wRFy5OGw4C&pg=PA140&lpg=PA140&dq=plaintext+authentication+sip+asterisk&source=web&ots=7hPikKrL-d&sig=ulh9sDMViL7sq3AB3066Fpen3_A&hl=en&ei=IBGPSdSVI8H7tgecy7CrCw&sa=X&oi=book_result&resnum=4&ct=result
17:07.45nvrpunkthat will answer your question, somewhere in it :P
17:08.25dwerynice, that mean shttp://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone is completely wrong
17:08.53*** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64)
17:09.19dweryI have the problem where my sip phone generates a busy tone after the first digit when in early dial mode.
17:09.35dweryit will however keep transmitting the digits and the call will go thru
17:09.40dwerybut the bsy tone is annoying
17:09.47[TK]D-Fenderkill off early dial
17:10.08*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
17:10.21dweryI can't, I need asterisk to provide a dial tone when I press the key for the external isdn line
17:10.26dweryad without early dial
17:10.33nvrpunkI have a problem with the "r" option not creating a ring and don't know why
17:10.34nvrpunk:(
17:10.35dweryI would have a 4 second timeout
17:11.04nvrpunkon the bright side people are still paying for calls
17:11.07nvrpunkwhich is a good thing
17:11.57*** join/#asterisk estr4ng3d (n=sm4rt@196.219.96.141)
17:13.30[TK]D-Fenderdwery: what key for external line?  Show us the dialplan & CLI output for your call attempt
17:14.25nvrpunkanyone care to tell me if there's anything wrong with these two contexts?  http://www.pastebin.ca/1330837
17:14.41nvrpunkit's not generating a ring to the caller
17:16.13giovaninvrpunk: try specifying the ring time?
17:16.21giovaniI've never tried using double-commas like that
17:16.24[TK]D-Fendernvrpunk: you need to ANSWER teh call first
17:16.49nvrpunk[TK]D-Fender, its an IVR
17:16.59nvrpunkthey dial in the extension
17:17.03nvrpunkit rings the extension
17:17.05nvrpunkhmm
17:17.08nvrpunkah
17:17.09[TK]D-Fendernvrpunk: [makecall] <- isn't
17:17.12nvrpunkAnswer before it dials
17:17.12nvrpunkduh
17:17.51*** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br)
17:17.52dwery[TK]D-Fender:  the actual dialpan has only one line: exten => _2XX,1,Dial(SIP/${EXTEN})
17:18.21[TK]D-Fenderdwery: What does that have to do with ISDN?
17:18.51dwerythat come later. I plan to have the 0 key to access the first free line. and I need to signal that w have one using a dial tone
17:19.12*** join/#asterisk ikevin_ (n=kevin@ANancy-256-1-53-94.w90-26.abo.wanadoo.fr)
17:19.18ikevin_hello
17:19.23dwerySo I want the user to press 0 and have a dial tone immediately. and that works only with early dial
17:19.55ikevin_anyone know how can i have sound quality in moh?
17:20.00ikevin_i use mpg12
17:20.04ikevin_mpg123*
17:20.08[TK]D-Fenderdwery: fine record a continuous dialtone and background it in an IVR
17:20.20[TK]D-Fenderikevin_: HUH?
17:20.41*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:20.41Maliutaikevin_: you are aware that it doesn't transmit all the frequencies
17:20.55ikevin_while i try to setup music on hold the sound is bad, cutting, ...
17:21.01ikevin_yes
17:21.33dwery[TK]D-Fender: there must be an easier way
17:21.37ikevin_so i think that can be played in the good speed and without cutting?
17:21.44[TK]D-Fenderikevin_: Stop using mpg123 and use Native MoH
17:22.08[TK]D-Fenderikevin_: For MP3's you also have to make sure they are non VBR and have no ID3 tags
17:22.12ikevin_i have tryed to play an mp3 width 'mp3' or 'quietmp3' that are not played
17:22.35nvrpunk[TK]D-Fender, even with exten => _XXXX.,1,Answer and then the Dial, it's still not ringing back
17:22.37ikevin_how can i check vbr?
17:22.38[TK]D-Fenderikevin_: See above
17:22.46[TK]D-Fenderikevin_: Get a good audio program
17:23.05ikevin_k
17:23.19ikevin_does the bitrate limit is 8kb/s ?
17:23.40[TK]D-Fenderdwery: If there is it has to be offered by the phone... * can't slowly pass early dial info on for you to your ISDN.
17:24.00[TK]D-Fenderdwery: * will pass the first thing it can directly on and you're DOA from there
17:24.06[TK]D-Fenderikevin_: No.
17:24.22[TK]D-Fenderikevin_: standard bitrates are fine.  Use mode=files
17:24.31dwery[TK]D-Fender: even without ISDN, I want early dial to work for SIP without the bsy tone on the first digit
17:24.37ikevin_ok
17:24.48dweryill try playtones
17:24.57[TK]D-Fenderdwery: Show us what is happening.
17:27.00dwery[TK]D-Fender: sure. what you need me to show? sip set debug ?
17:27.23[TK]D-Fenderdwery: A complete call attempt and what it's doing
17:27.50*** join/#asterisk hi365_m (n=hi365@85.130.230.240)
17:28.03dwery[TK]D-Fender: ok, logging now
17:28.22ikevin_what kind of software are you using to encode mp3 for moh?
17:28.38ikevin_all i test are putting tags on the file
17:30.24dwery[TK]D-Fender: http://privatepaste.com/820oxi3sgf I elieve the problem lies on line 130. * trasmits 404 instead of 484
17:30.45dweryI have probably configured somethig wring
17:30.49[TK]D-Fenderikevin_: If you encode it yourself don't use MP3
17:31.43[TK]D-Fenderdwery: Looking like * itself dos not support early dial
17:31.50ikevin_what format is bester?
17:31.56ikevin_can i use ogg? ^^'
17:32.15[TK]D-Fenderikevin_: use the codec your CHANNEL will use
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17:33.48dwery[TK]D-Fender: that's strange, I'm pretty sure I read it was supported
17:34.03[TK]D-Fenderdlewis: People always are
17:34.12dweryouch!
17:34.17dweryallowoverap=yes
17:34.54dwery[TK]D-Fender: now it's prefect. thaks for helping out
17:35.12[TK]D-Fenderdwery: glad you found it
17:36.02ikevin_[TK]D-Fender, i tryed to make a wave and using mode=files, sound is already cuted, slow, ...
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17:38.48[TK]D-Fenderikevin_: have you fixed your FILES?
17:38.59[TK]D-Fenderikevin_: what are you testing with?  Describe the call
17:40.13ikevin_i use ekiga as sip client, i use codec gsm for the call, i have make a wave file (bitrate 8k/s) and using asterisk native format
17:40.48[TK]D-Fenderikevin_: Try Ekiga with ULAW or ALAW instead and test
17:41.14ikevin_ok
17:43.15ikevin_ekiga still don't have this one
17:44.04sipyifconfig
17:44.13sipyaaaarrrrggghhh
17:46.18dwery[TK]D-Fender: now I have to tell * to buffer the digits when using the ISDN line, bcause it does not support early dial
17:47.57[TK]D-Fenderikevin_: yes it does
17:48.06[TK]D-Fenderikevin_: g711u, g711a
17:48.28[TK]D-Fenderdwery: that is the part I figured you would get screwed by
17:48.37ikevin_i only have g721
17:49.01[TK]D-Fenderikevin_: there is no way you only have GSM & G721 <- what is that?)
17:49.10[TK]D-Fenderikevin_: look harder
17:49.43ikevin_gsm seems the bester availlable, i have put a lowest priority on, and he's used
17:49.43[TK]D-Fenderikevin_: they are listed as PCMA and PCMU in Ekiga
17:50.00ikevin_ok, found
17:50.12ikevin_so, there are the first activated in the list
17:50.26ikevin_do i need to declarer anything on asterisk to use them?
17:50.38[TK]D-Fenderikevin_: your allow/disallow by peer
17:51.09ikevin_i don't have any entry allow or disallow
17:51.27[TK]D-Fenderikevin_: Then go set them
17:51.38ikevin_ok
17:51.48[TK]D-Fenderikevin_: "disallow=all" followed by "allow=ulaw" (for PCMU)
17:53.43ikevin_ok i try
17:54.48*** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net)
17:54.54ikevin_waou, there are a big difference :D
17:55.05[TK]D-Fenderikevin_: Improvement?
17:56.47ikevin_yep
17:56.52ikevin_thx very much :)
17:57.19[TK]D-Fenderikevin_: this is likely your problem :
17:57.21[TK]D-Fender~gsmbug
17:57.43jbot[~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
17:57.46[TK]D-Fender^^^^^^^^
17:57.50*** join/#asterisk estr4ng3d (n=sm4rt@196.219.96.141)
17:58.03ikevin_yep, 13k/s is maybe too short for what i need :)
17:58.13[TK]D-Fenderikevin_: No.... its a transcoding error
17:58.28[TK]D-Fenderikevin_: if ULAW came out OK, this is almost guaranteed to be why
18:02.41*** join/#asterisk cesau (n=cesau@207.58.225.93)
18:03.31*** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it)
18:03.56markithi, conf-extended... extended means "more duration" or "more people allowed in"?
18:08.42[TK]D-Fendermarkit: A sound file alone means nothing
18:10.37markit[TK]D-Fender: there are some commands to "extend" the conference. I have to translate sounds related, but since I don't use conference, don't know the meaning of "extend" a conference
18:11.03[TK]D-Fendermarkit: usually it would imply TIME, not # of participants.
18:11.17markit[TK]D-Fender: thanks a lot :)
18:11.40markit[TK]D-Fender: also can't understand the difference, if any, between "agent" and "representative"
18:12.03*** join/#asterisk ManxPower (n=manxpowe@11.sub-70-214-34.myvzw.com)
18:12.10markitdo you have some hint about it?
18:12.18markithi ManxPower
18:12.18[TK]D-Fendermarkit : Their meaning is usually the same.
18:12.54[TK]D-Fendermarkit: "representative" is more gentle & welcoming.  Agent sounds a little harsher
18:13.16markitok, fine. Last question: there are some minor errors (very trivial) in core-sounds-en.txt, do I have to use the bugtracker or try the -bugs channel?
18:13.43ManxPower[TK]D-Fender: I prefer "Your worst customer service nightmare"
18:13.45ikevin_does sound at 32kb/s@16khz is better than a sound at 64kb/s@8khz ?
18:14.11ManxPowerikevin_: very few devices support wideband (16Khz)
18:14.31ikevin_ok
18:15.41ManxPowerPolycom calls wide band "HD voice"
18:16.04ManxPowerthat's the only company that supports it as far as I know, and only a few polycom models even support it
18:16.19Corydon76-digbut in answer to your question, yes, 16kHz almost universally sounds better than 8kHz
18:16.43[TK]D-FenderHalf the bitrate and better quality.... odd
18:17.11ManxPowerCorydon76-dig: Even at 32K?  I would think since you have more data to compress the same bit rate would sound worse at 32K compared at 8Khz?
18:17.22Corydon76-dig[TK]D-Fender: what matters is not the number of samples, but the depth of each sound
18:18.27ManxPowerquickly parents a wide band extension to the LPC coded!
18:18.30Corydon76-digAbove a certain number of samples per second, you won't be able to tell a difference anyway
18:18.41ManxPowerand paTents it too!
18:19.40[TK]D-FenderCorydon76-dig: Yeah.. there looks to be a # missing.  sample frequency, audio range, bitrat... those seem to include only 2
18:19.42Corydon76-digThe only advantage LPC10 ever had was the size of the resulting bitstream
18:21.31Corydon76-digIt's like the gamers who try to get 110fps... anything above about 30fps is wasted CPU
18:22.05Corydon76-digIn this case, we're talking about the perceptual limitation of the ears, not the eyes, but the same principle applies
18:22.58ScribbleJGood morning friends.  I promised to post a version of my Generic Speech Engine API plugin yesterday, even though I am ashamed of the code at this point.  And I did - you can get it at http:/scribblej.com/svn/ and there is a phone number on that page too where you can try it.
18:26.11[TK]D-FenderCorydon76-dig: 60 fps :)  the 30 deal well... thats a whole story
18:26.44[TK]D-FenderCorydon76-dig: nifty trick how movies get away with 24 because of the concept of persistence .\
18:30.38*** join/#asterisk hi365_m (n=hi365@85.130.230.240)
18:30.53ScribbleJ24 fps is how Hollywood makes it's money
18:30.57*** join/#asterisk taso (n=hfaohh@c-67-180-35-251.hsd1.ca.comcast.net)
18:31.01ScribbleJYou thought they were only gouging you on concession stand prices!
18:31.12ScribbleJBut no, not only that, but they /keep/ 6 frames out of every second that YOU PAID FOR!
18:32.10tasohey guys, quick question.  Suppose I have a cell phone w/ an active #... Let's say the # is 228-492-2249 ... can I have that # forwarded to a Asterisk server that acts as an answering machine?
18:33.15ScribbleJYes, but if your next question is 'how' the answer is 'ask your cell phone provider' - they have to arrange the forwarding.
18:33.53tasookay, so, that's possible great.  Next question.  Let's say that I forward my calls to an Astierk server, then, if it's a number I want , it calls me, is that possible or no?
18:34.29*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
18:34.36ScribbleJI'd assume so, althugh it might be impossible for your provider to forward some calls and not others (i.e. when you call yourself from asterisk, it has to /not/ get forwarded, right?()
18:34.39tasoI would think no since it would has to call you back.... and when it went to call you back...it would be calling itself
18:34.43ScribbleJRight
18:34.46ScribbleJI think no also.
18:34.58ScribbleJHere's how I'd solve it.
18:35.03taso2 #'s ?
18:35.05tasosame phone?
18:35.07ScribbleJGEt a SIP provider, hand out /that/ phone number to my frineds.
18:35.12tasoah I see
18:35.20ScribbleJThen when my friend calls they are calling directly to ASterisk
18:35.24tasoyeap
18:35.35ScribbleJAnd from there I canf orward to the cell # they do not know, if they meet my quality standards.
18:35.47taso:)
18:35.49ScribbleJThey don't, by the way -
18:36.01taso:P
18:36.04ScribbleJI would not be friends with anyone who would have a loser like me for a friend.
18:36.19tasounfortunate
18:38.27*** part/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
18:41.37tasoso here's another question, can you move your cell phone # to a land line? or to any line for that matter?
18:42.12tasoI know you can move a home phone # to a cell phone
18:42.17tasobut can you do the inverse
18:43.23ScribbleJYou can in the US
18:43.37ScribbleJYOu can even move it to  SIP provider if you do not want a real phone line
18:44.22tasowild
18:45.49*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:48.29ManxPowerPorting a number to a SIP provider usually takes 10 - 14 days.
18:48.41tasoyea, I just got off the phone with my provider
18:48.46tasothey said it was possible
18:48.57tasois there anyway to automate this service?
18:49.14ManxPowerdefine "automate"
18:49.42ManxPowerIt takes as long as it takes.  Nothing you can do about it.
18:49.52tasoi.e. , let's say you were a telco, and you had people signing up for your SIP service, and it said "Please tell use the # you would like to use" , and you enter in your cell phone # and it automatically starts the switch over
18:50.01tasoor send an email to your provider
18:50.06tasoetc
18:50.18ManxPowertaso: not a chance.  Telcos need LOTS of paperwork to prove you authorized the change
18:50.25tasoah
18:50.49ManxPowerAND your old telco has to accept the loss of the customer.
18:51.45*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
18:51.49ScribbleJYeah, if they want to, your old telco can bitch about it and require papers submitted.
18:51.57ScribbleJAlthough it turns out that sometimes they do not.
18:52.11tasowell, it actually wouldn't be a loss, you would get a new # assigned to your cell phone, and your old cell phoen # would go to the SIP provider
18:52.18ScribbleJI transferred a number to AT&T from Sprint; all AT&T needed was my word, Sprint just trusted them.
18:52.53ManxPowerIt generally works like this.  Customer requests a number port from their new telco.  Their new telco gets all the required information for your old account, then contacts the new provider for a number port, the old provider says something like "REJECTED! You did not dot that i", the process repeats once or twice and then your number is (usually) ported.
18:53.14ManxPowerScribbleJ: cellphone to cellphone ports are different
18:53.34ManxPowerI was referring to porting from a land line.
18:53.48ManxPowerScribbleJ: how long did the port take?  a few days?
18:54.13ScribbleJAre they different in some real way, or are they just different int hat the cell phone companies tend to know eeach other and go easy on the requirements?
18:54.13ScribbleJManxPower, as I recall it took about 4 hours, actually.
18:54.17*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
18:54.17ManxPowerScribbleJ: I don't think there is an anti-slamming law for cell companies.
18:54.31ScribbleJOh, I see.
18:54.58ManxPowerWith land lines if you move your number and then claim you did not authorize it -- if the telco can't prove you authorized it there can be big fines.
18:55.20ManxPowerfor slamming long distance service it's $10,000 per incident if I recall correctly.
18:55.22tasoit's a law that you have to state "This phone call with be recorded, correct ? "
18:55.47ManxPowertaso: call recording within a state is goverened by the laws of that state.
18:55.48tasoif you're going to record a call that is
18:55.54tasointeresting
18:56.12ManxPowersome states only require ONE of the parties to consent to recording.
18:57.04tasohrm
18:57.37tasomeaning that if there is Person A and Person B... and Person A calls person B, Person A can be okay with it, without person B knowing
18:57.40taso?
18:59.58tasohrm reading up on it now, that's pretty wild
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19:11.39giovanitaso: wild?
19:11.49giovaniit just allows you to record your own calls ... reasonable to me
19:12.25ScribbleJIn my opinion, it's easy to just always hav your system say "this call will be recorded" - people don't even pay attention to this anymore.
19:13.01giovaniI think most companies go with "may be recorded" for legal reasons, so that they're not obligated to provide it, should someone, say, a lawyer, request the recording
19:13.04[TK]D-FenderScribbleJ: "may" be recorded :)  Less forceful and equally legitimizing :)
19:13.44giovanimaybe obligated was the wrong word -- so that they can claim they never recorded it :)
19:14.08tasobut if it was a personal phone call people would be thinking wtf?
19:14.29tasoyou call your friend Dave, and it says "This call will be recorded"
19:14.31tasothat's different
19:14.32giovaniif it's a personal phone call ... you won't be facing any litagation unless you're publishing the recordings
19:14.45giovaniso, it's really a non-issue other than a hypothetical, and pedantic legal one
19:14.52tasoheh
19:15.42giovaniif, however, you call up some company, like say, customer support, and plan on recording it for some reason, say, to embaress the company on your blog ... I'd announce it to ensure no legal trouble :)
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19:25.44jayteeusually the standard phrase in the states is "This call may be recorded for "quality assurance" purposes."
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19:33.09[TK]D-Fenderjaytee: AND "training"
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19:48.26pseubodothi there, I'm trying to get asterisk running on debian with a Wildcard TDM400P REV E/F Board 1, running thru the examples in the oreilly book. seems no matter what I do, I always get "4 channels to configure." even if zttool reports the card as 'OK'
19:49.05ScribbleJI don't suppose any helpful person wants to tell me if I set things up properly for incoming sip calls, if you tried something like sip:2000@home.scribblej.com you /might/ get my Speech Engine PLugin
19:50.13ScribbleJOr you might get nothing if I screwed up. Heh.
19:54.57[TK]D-Fenderpseubodot: that message is fine.
19:55.27[TK]D-Fenderpseubodot: it is not negative. Its saying that Zaptel is initialize and ready for * to allocate channels agains
19:55.47[TK]D-Fender+t
19:55.56pseubodot[TK]D-Fender: thanks
19:56.25pseubodoti get to the point where my dial plan looks like what it says in the book, and 'dialplan show' seems to come out right
19:56.38pseubodotbut I'm not getting dialtone on my phone (plain phone)
20:01.11*** part/#asterisk bgmarete (n=marebri_@196.201.217.80)
20:01.39[TK]D-Fenderpseubodot: What modules do you have on your card?  Pastbin your configs.
20:01.41[TK]D-Fender~pb
20:01.52jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/
20:01.55pseubodotthanks
20:02.31pseubodothttp://paste.debian.net/27970
20:04.00pseubodot2FXOs, 2FSOs
20:04.12[TK]D-Fenderpseubodot: I see.  Now for your configs...
20:04.28*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
20:04.32pseubodot[TK]D-Fender: which specifically?
20:04.59[TK]D-Fenderpseubodot: zaptel/zapata, dialplan, etc
20:06.03pseubodot[TK]D-Fender: http://paste.debian.net/27971/ <- all here
20:06.48[TK]D-Fenderpseubodot: Your zaptel doesn't match your ztcfg
20:06.54[TK]D-Fenderpseubodot: You commented the FXO's out
20:07.15[TK]D-Fenderpseubodot: Sorr, 1 each
20:07.19pseubodotstandby
20:08.13[TK]D-Fenderpseubodot: And you are defining your channels twice because of that include
20:08.14pseubodot[TK]D-Fender: http://paste.debian.net/27972/ (fixed zaptel.conf file, did ztcfg -vv)
20:08.26pseubodot[TK]D-Fender: I should remove the include?
20:08.44[TK]D-Fenderpseubodot: include is fine but you can see it doing them twice
20:08.53[TK]D-Fenderpseubodot: fix tha
20:09.21[TK]D-Fenderpseubodot: Also not that you do not have anything you can dial in your [default] context.  Not good...
20:09.31[TK]D-Fenderpseubodot: Also ensure that you have the molex connected to the card
20:10.32*** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
20:10.40pseubodotNot sure I follow - I'm not sure that I can see it being added twice
20:11.08pseubodotthe molex is on the correct port (afaik), cord from wall goes in 3rd receiver from the top of the card
20:11.34pseubodotcord from the telephone on my desk is going into the 1st receiver from the top of the card.
20:16.59pseubodotthis is the third time I'm trying to set this up
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20:19.35[TK]D-Fenderpseubodot: No, the molex POWER connector from your power supply to the PCI card
20:19.45pseubodot[TK]D-Fender: ah yes, I do have power connected
20:19.54pseubodotto the card
20:20.10[TK]D-Fenderpseubodot: Only connect an analog phone and try each port.  IIRC they are not sequential on the back of the card
20:20.32pseubodot[TK]D-Fender: what should I be listening for?
20:20.52[TK]D-Fenderpseubodot: dial-tone, and watch * CLI for "starting simple switch
20:21.52pseubodotfound one port where I seem to be getting the echo
20:22.53pseubodotand when I lift the receiver I see in zttool the Total/Conf/Act line change from 4/4/0 to 4/4/1
20:23.08pseubodotwoo! progress!
20:23.22[TK]D-Fenderpseubodot: Get out of zttool and startup *
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20:23.42pseubodot[TK]D-Fender: debian*CLI>
20:24.02[TK]D-Fenderpseubodot: "zap show channels"
20:24.14masushi all, can anybody say me the possibility to get the dialstatus of a call maded by /var/spool/ast.../outgoing directory ..
20:24.29pseubodot[TK]D-Fender: "No such command 'zap show channels'"
20:24.53[TK]D-Fenderpseubodot: pastebin "show modules"
20:25.05pseubodot[TK]D-Fender: will do. standby
20:25.17masus;(
20:26.09pseubodot[TK]D-Fender: http://paste.debian.net/27973/
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20:26.58[TK]D-Fendermasus: for the "Channel:" you aren't going to be able to for sip/zap/iax, etc.  You need to be dialing a LOCAL channel and you can process the dial in there.
20:27.24[TK]D-Fenderpseubodot: "module reload chan_zap.so"
20:27.34[TK]D-Fenderpseubodot: the do "zap show channels" again
20:28.31pseubodot[TK]D-Fender: http://paste.debian.net/27975/
20:28.53masus[TK]D-Fender: Thank you. I'll try .
20:29.16[TK]D-Fenderpseubodot: Looks like your modules are configured backwards
20:29.31pseubodot[TK]D-Fender: I would change these in /etc/zaptel.conf?
20:30.33[TK]D-Fenderpseubodot: Seriously double check the module order on the card physically and see how they really map to the port #'s.  IIRC its simething like 4,1,2,3 or something like that... then make sure zapata & zaptel both match
20:31.19pseubodot[TK]D-Fender: two green modules on left (towards card bracket), two red modules on right (away from bracket)
20:31.47[TK]D-Fenderpseubodot: Read your card docs for the functional order, not just left-right
20:32.27cesauok - i have a box with no dialers, just astersik -- im trying to make an outbound call via my itsp and play a file -- my itsp tells me i dont need to register with them -- when i originate a call, would i use Local/somerandomeext@context extension SIP/provider@context?  if so, which context is being performed, the local or the providers?
20:34.00pseubodot[TK]D-Fender: okay, thanks
20:34.09pseubodotI have no docs, so I'll have to figure that out.
20:34.35[TK]D-Fendercesau: SIP/provider@context is not an EXTENSION.  Your DIALPLAN holds your extensions.
20:34.41[TK]D-Fenderpseubodot: www.digium.com
20:34.54[TK]D-Fenderpseubodot: should have the install guides, etc
20:36.46cesauok, if i have a simple extensions (like _X) setup, would it then be: originate sip/provider@context extension numbertodial <-- with the dial command in the first context? (this seems like it should be really easy and im overcomplicating it)
20:37.34[TK]D-Fendercesau: "go read the instructions for originate
20:37.38masus[TK]D-Fender: i have do it with local channel , but there is something wrong its calling the number twice , here are some debug and config files -> http://rafb.net/p/jMyto215.html if u can take a look please . :)
20:38.26cesauthe instructions seem to imply that i have a hard phone to originate from... =/
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20:38.52[TK]D-Fendermasus: Of course it is... YOU are telling it to call that local channel and dump it into the SAME EXTENSION
20:39.23pseubodot[TK]D-Fender: found it, according to this my card should be as I expect.
20:39.38pseubodot[TK]D-Fender: should and is are two different things tho
20:39.41masushmmm , i dont understand the logic :s
20:40.25[TK]D-Fendermasus: http://pastebin.com/m7ac11036
20:41.27[TK]D-Fendermasus: And you didn't specify the context
20:42.28masus[TK]D-Fender: Thank you i'll try a little more
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20:44.15badcfei have a question actually related more to sip than to *
20:44.26badcfesince UPDATE is a non rfc3261 it should be treated as OPTION and thus not fork.
20:44.31badcfea proxy may have parallelly forked an INVITE and now receives an UPDATE on that early dialog.
20:44.34badcfemy question is: what should this statefull proxy then do?
20:45.43pseubodothmm.
20:46.01pseubodotremoves 1xFXO and 1xFXS to make figuring this out simpler.
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21:00.33badcfeno SIP experts around huh
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21:04.02ManxPowerbadcfe: try the SER/OpenSer/OpenSIPS channels.
21:04.42*** join/#asterisk EagleXDR (n=EagleX@bzq-82-81-124-249.red.bezeqint.net)
21:06.25cesauassuming "Contact: <sip:asterisk@172.18.55.20>" should really be a class C ip address, which setting in sip.conf would change that?
21:06.47[TK]D-Fender~sipnat
21:06.54jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:06.55[TK]D-Fendercesau: READ ^^^
21:07.16badcfeManxPower: thanks.  never heard of opensips by the way..
21:07.16cesauD-Fender, thank you! =D
21:07.17ManxPower1) there are really no "classes" anymore, and anything like that would need you to know the netmask
21:07.33ManxPowerbadcfe: It is a fork of OpenSER
21:07.45ScribbleJDoes anyone have a second to try calling sip:2000@home.scribblej.com and let me know if they get an answer?  Or is there some clever way I can test my SIP externally myself?
21:07.50[TK]D-FenderManxPower: its a whole friggen cutlery set by now...
21:08.16ScribbleJMaybe if I route my own call through my itsp that'd work
21:08.23[TK]D-FenderScribbleJ: Dial(SIP/2000@home.scribblej.com)
21:11.24ScribbleJ[TK] my only * box is the one serving that, and I also need to know if I got my nat and routing correct to let in calls from the universe.
21:11.43ScribbleJSo I have to originate the call from someplace other than my own LAN, and * server.
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21:12.07HermesNetoHi
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21:24.16ghenryany progress on the best way to run asterisk on an xen guest without a patch i.e. ztdummy or dahdi_dummy
21:26.15hardwireinstall. use asterisk 1.6
21:26.21hardwireHEAD
21:26.33hardwirerejoice
21:28.29ghenryis anyoen using 1.6 yet? I think it's way too early in the cycle
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21:29.36hardwireghenry: you're right.. sorry I mentioned it.. continue having issues please.
21:30.14ghenrywill do hardwire. Thanks ;-)
21:30.33hardwireNot accepting my word as the golden law of the land has made you an enemy of Ra..
21:30.47hardwireBe prepared to deal with the consequences.. My Anubis army is already on it's way.
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21:30.47*** mode/#asterisk [+o russellb_] by ChanServ
21:30.56[TK]D-Fenderghenry: Careful or hardwire will go nova on your ass!
21:31.12phix:D [TK]D-Fender!
21:31.24ghenryI got a pretty thick ass ;-
21:31.45ghenrywill try iy
21:31.47ghenryit
21:31.57hardwiregood boy
21:31.59ghenryI havent tried since workign on the ldap stuff
21:32.16ghenryI've got a few tickets I need to attend to when I can
21:32.23hardwirewhat project?
21:32.30ghenryres_ldap
21:32.52hardwirewhat functions does that expose?
21:33.27hardwireI rarely use LDAP so I'm curious how it's integrating.
21:33.34ghenryall the noraml realtime stuff
21:33.41ghenrythat can be put in a directory server
21:33.48hardwireweird
21:34.02ghenryfor some things, not for sip users etc.
21:34.09hardwireso contexts are ldap groups?
21:34.16hardwirehow are dialplan entries sorted?
21:34.24ghenryexactly
21:34.30ghenryI've got to update the docs
21:34.35ghenrytake a peak
21:34.58ghenryhttp://bugs.digium.com/view.php?id=13660
21:35.00hardwireis there a priority kv in your schema?
21:35.22hardwireah.. so for endpoints only?
21:35.51hardwirerussellb: ..
21:35.57russellborly?
21:36.03hardwireyou're spamming my hosstom
21:36.05hardwirehosstop
21:36.12hardwireI demand sacrifice.
21:36.44hardwireis an agent of the sun god today.
21:36.46russellbo.o
21:36.52russellboffers file
21:36.58hardwirewoot
21:37.21hardwireI'd like to thank whoever left a half eaten bag of doritos on my desk
21:37.32hardwiremuch obliged.
21:37.38hardwireyour offering is acceptable
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21:38.29[TK]D-Fendersacrifices... HARDWIRE :p
21:38.37[TK]D-Fenderactually... I'm not sure that would count ;)
21:38.41hardwireoh i'd like to see you try!
21:38.53hardwireis a fast lil bugger
21:40.07hardwireI should have gone home long ago
21:40.18hardwireI just did /etc/init.d/networking stop on a remote machine
21:40.18ghenryhardwire: so with 1.6 dahdi_dummy is not needed
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21:40.35[TK]D-Fendergrabs his athamé and chases after hardwire
21:40.38hardwirewhat's funny is there is a serial cable enroute to site.. but nobody is on site today to plug it into the backup admin modem.
21:40.40hardwirelul
21:41.08hardwire[TK]D-Fender: for shame.. making me google
21:41.26hardwireooh shiny.
21:41.46hardwireghenry: you know.. I don't actually know.
21:42.03hardwireI just keep hearing how it's being/been/possibly/wishlisted into the most current code.
21:42.16*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
21:42.29hardwirebut setting youself up to accept a 1.6 update sounds like a fancy idea
21:42.52ghenrywill try for now then.
21:43.13ghenryif timing doesn't need dahdi then for meetme etc. then good to go I guess
21:43.33ghenrysuppose so; http://www.jeremy-mcnamara.com/2008/06/17/a-new-timing-api-for-asterisk-silencing-digium-critics/
21:43.33hardwirerussellb: ? know anything about that?
21:43.48hardwirehears a groan in the distance.
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21:44.20russellbknow anything abotu what
21:44.26russellbthe timing API?
21:44.26russellbyes.
21:44.40hardwireGood.. ghenry that should answer that.
21:44.43hardwiresomebody knows about it.
21:44.56ghenryI'm doing asterisk with xen with no pci cards etc.
21:45.07russellbthe timing API is only in 1.6.1, not 1.6.0
21:45.08ghenryjust wondering if I need dahdi at all
21:45.15russellbghenry: depends what you want to do.
21:45.25russellbif you want conferencing, yes, but that is the only reason.
21:45.36ghenryeven in 1.6.1?
21:45.39russellbyes.
21:45.46russellbthe timing API does not replace the need for DAHDI for conferencing
21:45.49ghenryfor dahdi_dummy right
21:45.53russellbas DAHDI contains the entire conference mixing engine.
21:46.00russellbit's not just used as a timing source
21:46.04rob0I thought the dahdi_dummy was needed as a timing source, but only for conferencing?
21:46.12rob0I didn't know that part
21:46.12ghenryso, should be DAHDI, sorry
21:46.20russellbHowever, we have code that replaces the need for it for conferencing, as well
21:46.27russellbit is on the roadmap for inclusion in 1.6.2
21:46.33ghenryok
21:46.48hardwirewoot
21:46.55hardwireI use asterisk under openvz
21:46.57hardwiretiming works fine
21:47.03ghenryso with 1.6.2 you can, if no hardware needed say for use in xen, no DAHDI is needed.
21:47.04hardwireit's actually somewhat evil
21:47.30ghenryI mean, no need for analgoue or digital connections etc.
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21:49.57ghenryI could just add a pri card to the box for timing to save all this I suppose
21:50.28ghenryrussellb: can Digium Business Edition run under xen?
21:50.45russellbit does not have additional functionality that is not open source
21:51.00ThreeSeventhshello all, I am trying to implement dundi to connect two asterisk boxes together. One is on the internet with a public IP and iptables. The other is behind a cisco/linksys hardware firewall with a private IP. UDP port forwarding for port 4520 has been added on both systems. Each system will successfully perform a query for the other's entity ID. only the system which is publicly available can receive a lookup from the other one behind the
21:51.22ThreeSeventhsdoes anyone have any idea what the problem may be? Any help or hints would be appreciated
21:51.32ghenryrussellb: I mean is the rpath xen kernel patched or is zaptel?
21:52.14russellbghenry: i'm not sure what you mean ...
21:52.16ghenryif it can, that might be quite interesting
21:52.17russellbthe issues are dahdi/zaptel issues
21:52.31russellbThreeSevenths: if you have port forwarding on, it should work ...
21:52.49ghenryyeah, but I thought you can patch ztdummy etc. Maybe I googled wrong
21:52.57russellbThreeSevenths: do you have registration turned on from the private box?
21:53.08russellbghenry: yeah, you can i think
21:53.15russellbghenry: i think there might be a patch on bugs.digium.com for it ...
21:53.23De_Monlol. xkdc is great.
21:53.30De_Monoops xkcd
21:53.32russellbThreeSevenths: I mean the "register=yes" option
21:53.38ghenryAnyone tried the 51i Aastra phone and experience after a couple of mins "no service" even though it registers first time?
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21:53.43ghenryrussellb: OK, thanks.
21:54.44ghenryhttp://bugs.digium.com/view.php?id=9592
21:54.47ThreeSeventhsrussellb: no yet
21:54.50ThreeSeventhslet me try that now
21:55.34ThreeSeventhsrussellb: register=yes has not helped
21:56.30russellbThreeSevenths: have you done a packet capture to see if the query from the public box is even getting there?
21:56.40russellbthat is where i would look to see if it is a network problem or an asterisk problem
21:56.49ThreeSeventhsrussellb: yes we have, both boxes are communicating
21:57.16russellbAlright, next thing to look at is the "dundi debug" output, to see what Asterisk does with the query
21:57.21ThreeSeventhswhen i do a lookup from one to the other, we receive a reply with HINT DONTASK|UNAFFECTED
21:57.34russellbAh, so it's a configuration issue of some kind.
21:58.13ThreeSeventhsyes, and i'm just not sure where it is, because the lookup works one way, but not the other, and we have the exact same config at each end
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21:59.49russellbThreeSevenths: I can sanity check the configuration if you'd like.  just pastebin it.
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22:01.50alfa202hello
22:02.07ThreeSeventhshttp://pastebin.com/d2a12889
22:02.14russellblooks
22:02.57ThreeSeventhswe are using the e164 context while we debug and plan to move to the priv context later
22:03.40russellbnods
22:04.45russellbThreeSevenths: and in extensions.conf, you have the extensions you want to be able to lookup available in dundi_local, right?
22:05.26ThreeSeventhsnope
22:05.33ThreeSeventhsactually, you just pinpointed the issue
22:05.35russellbthat would be the problem, then.
22:05.37russellb:-)
22:05.43ThreeSeventhsit's dundi-local in ext.conf
22:05.48russellbah ha!
22:06.14*** join/#asterisk kink0 (n=xchat@212.170.176.86)
22:06.16kink0hello
22:06.28ThreeSeventhset voilla, it works
22:06.32russellbyay!
22:06.47kink0there any way to return NO DURATION and/or preseted q931 cause code from Background() ?
22:06.58ThreeSeventhsrussellb: thanks alot, pebkac
22:07.06russellbyou're welcome.
22:07.46kink0i.e. I want to play a message "the number is wrong..." and returns Cause 27 or some other cause, at same time not duration to avoid billing
22:08.35russellbkink0: well, I know you can control the cause code.
22:08.43russellbkink0: as an argument to the Hangup() application.
22:09.01russellbI don't think there is anything you can do beyond that
22:09.34*** part/#asterisk badcfe (i=cso@78.156.5.233)
22:12.12kink0russellb  I tryed Playback(wrong_number + Hangup(27) , but I faced these problems...
22:12.36kink0first the call has duration , seconds the Cause seems ignored
22:13.04russellbkink0: did you Answer() before the Playback()?
22:13.08russellbif so, don't.
22:13.11*** join/#asterisk oej_ (n=olle@81.253.94.194)
22:14.48russellbkink0: also make sure you set the 'noanswer' option for Playback()
22:16.06*** join/#asterisk styelz (n=yoohoo@2001:5c0:1100:a00:0:0:0:1)
22:16.18kink0russellb, no, I have not Answer()
22:16.41kink0I did not put "noanswer" option for playback ... let me try it now with noanswer option
22:16.49russellbk
22:17.12russellbwithout it, Playback implicitly answers
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22:24.03kink0I am trying.... but with noanswer option file is not played, just fast busy
22:25.00russellbok, one more thing to try ...
22:25.10russellbrun Progress() right before the Playback() using 'noanswer'
22:25.41russellbafter that I give up, as I need to go to bed soon :-)
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22:27.05[TK]D-Fenderrussellb: Ah yes... you're in Sprouts-land right? :)
22:27.39russellbyup
22:31.15kink0sprouts ?
22:32.16WilliamKsprouts are good, didn't your mother teach/tell you that? :)
22:32.39russellbI'm in Brussels, heh
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22:34.12WilliamKsame as -
22:34.14WilliamKThe Brussels (or brussels) sprout (Brassica oleracea Gemmifera Group) of the Brassicaceae family, is a Cultivar group of wild cabbage cultivated for its small (typically 2.5–4 cm or 1–1.5 in diameter) leafy green buds, which resemble miniature cabbages.
22:34.55WilliamKsorry russell - no escaping!
22:35.46kink0ahhhh here Spain
22:36.09kink0How is dressed the Petit Juilane today ? :)
22:36.58russellbI didn't make it over to see that ..
22:37.50russellbi'm out.  good night, everyone!
22:37.55kink0good night
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22:42.38wonderworldis there a dedicated linux server hoster that offers an E1 / T1 as well?
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23:09.41keith4_I'm converting a system from cdr-csv to postgres, and I want to push the existing CSV records into the database. The fields aren't in the same order... is there documentation on the CSV format somewhere?
23:10.01keith4_most of the fields are obvious, but there are a few that i'm not sure about
23:10.59[TK]D-Fenderkeith4 : Yes.. in your TARBALL
23:13.15keith4_my fault. i asked the wrong question...
23:13.24keith4_is there documentation anywhere *other* than in the source code?
23:14.05seanbrightdo you have to enable something magical in zapata.conf to get CID name over PRI?
23:14.23seanbrighti don't even see the IE coming over on an inbound call, so i'm assuming it's on the telco side
23:14.46[TK]D-Fenderkeith4 : Maybe somebody cut & paste it on the WIKI... until then go look in the damn source tarball :p
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23:16.14cesauin sip.conf, should it be "[general]" or "[global]" ?
23:16.18cesauv1.4
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23:18.31jaytee[general]
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23:26.30keith4_[TK]D-Fender: despite your best efforts, i think you pointed me in the right direction
23:26.46keith4_sure, the validity of this is questionable... but it looks right to me http://www.voip-info.org/wiki/view/Asterisk+CDR+csv+conversion+mysql
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23:49.24joakoseanbright: Yes, but first and foremost you need the provider to send the name over the PRI:
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