IRC log for #asterisk on 20070817

00:01.12OlobolaI wonder if it's possible to rewrite grammar files dynamically for lumenvox apps?
00:03.43*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)
00:03.52*** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.49)
00:04.57*** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net)
00:07.53rbdis it possible to have the playback application support barge-in (e.g. premature stopping of prompt playback by the user pressing a digit)? I see there is a ControlPlayback application that has a 'stop' DTMF button, but it doesn't look like the rewind, ff, etc controls can be disabled (I just want stop for barge-in)
00:11.36*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
00:11.56Yourname`Hey fujin, I don't know if you were there when I told you about the ulaw conversion thing... but yeah, I converted 'em all to ulaw, and damn, the load was LOW!
00:12.53JTnice
00:18.03fujinyeah
00:18.04fujinhehe
00:18.07fujinI did another load test today
00:18.16fujin256~ concurrent calls, all in queue with ulaw musiconhold
00:18.45fujinload of about 0.24 :)
00:18.45JTon what?
00:18.45Yourname`Hmm, then I must've done something wrong today.
00:18.45Yourname`I did about 300~ and my load was around 3.
00:18.46JTerr
00:18.47Yourname`But I thought that was normal.. :S
00:18.51fujin300 with two-way rtp?
00:18.54JTyou're assuming your hardware is the same
00:18.58Yourname`Yes, sir.
00:18.59fujinthis is true
00:19.05fujinwhat kind of gear?
00:19.08Yourname`ah, true JT.
00:19.28Yourname`Intel Core2Duo, 3GB RAM, 250HDD
00:19.48JTso not quite server grade
00:19.53fujinmm
00:20.00JTfujin: and you?
00:20.11Yourname`Hmm.
00:20.15fujinnot that good
00:20.21Yourname`WHAT!
00:20.27Yourname`I've surely done something wrong.
00:20.31Yourname`Man, this sucks.
00:20.55fujindon't sweat it
00:21.04fujinI was only using SIPP with a captured rtp stream
00:21.06CoffeeKidAnyone here worked with PlayDTMF from the asterisk manager commands before?
00:21.10fujincalls were going in and out quite fast
00:21.19fujindidn't actually have 300 *real* callers.
00:21.42Yourname`Oh, that was the difference then..
00:21.52fujinI'd say so :)
00:21.56Yourname`phew
00:21.58fujindid you use that exact sox command I gave you?
00:22.02fujin1 channel, 8000bits?
00:22.07Yourname`Yup!
00:22.17fujinyep. Probly the real caller fact.
00:22.19Yourname`It was just hard to find sox's dependencies, but got it in the end..
00:22.37Yourname`Hmm, now I'm going to try a benchmark test with sipp
00:22.47Yourname`Just so I can compare to you
00:22.55fujindid you build from asterisk from source?
00:23.02Yourname`Yeah..
00:23.05fujinsweet
00:23.05Yourname`Is there another way?
00:23.12fujinlol
00:23.16fujinyes, you can run binary packages
00:23.19Yourname`Oh, wait, did you say did you build FROM asterisk?
00:23.32Yourname`Meaning if I built sox from asterisk?
00:23.51Yourname`I think I rpm'd sox from someone who said it had mp3 support builtin.
00:23.54coppicewe build sox for calls of wool
00:24.07fujinno, I meant build asterisk from source
00:24.08fujinnot sox
00:24.08fujin:P
00:24.12Yourname`Because the sox I originally compiled didn't have mp3 support..
00:24.18Yourname`Yeah, lol, asterisk was from source.
00:24.22fujinI apt-getted sox, or it was preinstalled, I forgert.
00:24.29JTwhat's the CPS rates on your benchmarks, Yourname`, fujin ?
00:24.43fujin9.483
00:25.01fujin10(10000ms)/1s
00:25.07Yourname`I didn't check those yet, JT :(
00:25.13fujinjust dropping them into a queue
00:25.24fujinwaiting for 10000ms, then hanging up
00:25.26fujinrinse repeat
00:25.35Yourname`I think I stopped trying to figure out sipp earlier today, it just got annoying, lol
00:25.41fujinheh yeah
00:25.43fujinit's not very intuitive.
00:25.53fujinI updated the SIPp page on voip-info
00:26.01fujinwith a little no-brainer SIP volume test (no rtp)
00:26.10fujinhttp://www.voip-info.org/wiki/view/Sipp
00:26.56*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:27.38Yourname`When did you do that? I don't think it was there earlier, damnit.
00:28.59fujinoh, about an hour ot two ago?
00:29.20fujinLast modification by AJ on Fri 17 of Aug, 2007 [11:11]
00:29.39Yourname`hah, nice. I should have found it earlier..
00:29.41Yourname`Let me try now.
00:29.49Yourname`Once I figure this out, it's off to tx_fax
00:31.34fujingood luck getting that working.
00:31.38fujinI've heard most success with IAXfax
00:31.42fujiniaxmodem*8 rather.
00:32.02fujinwouldn't catch me _dead_ doing FoVoIP
00:32.15Yourname`heh, hence the try.. :P
00:32.35Yourname`iaxmodem over VoIP?
00:32.38[hC]hylafax+iaxmodem is much better than rx/txfax
00:32.48Oloboladoes anyone know if it's possible to load lumenvox grammar info dynamically?
00:32.50[hC]havent tried it over voip though.
00:32.54Yourname`I wanna try to do asterisk+somefax
00:32.58Yourname`..over VoIP
00:33.41Sweeper[hC]: got a howto handy? I need to get that going
00:34.14[hC]there are a bunch on google
00:34.20Yourname`[hC]: I'm trying the asterisk+spandsp+rx/tx_fax approach over VoIP, maybe ulaw..
00:34.21[hC]search for iaxmodem hylafax
00:34.43[hC]Yourname`: the issue with that is not that it wont work, its that it wont work reliably enough to count on.
00:34.58JTerr
00:35.06JTiaxmodem IS FoVoIP
00:35.10[hC]Yourname`: you'll likely see that you have about an 80-90% success rate doing that... Which may or may not be acceptable to you.
00:35.13*** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au)
00:35.25[hC]JT: He means FoIPWAN
00:35.34Yourname`[hC]: ah.. I want to see how good we can actually maybe even be able to send, like a faxblast environment.
00:35.52JTright
00:35.58Yourname`[hC]: I think that's a bearable rate..
00:36.05JT[hC]: for tx or rx?
00:36.11Pengguhi all. on digium cards, once asterisk dials a number and bridges a POTS zap channel - can it know if the person actually successfully made a call, or if it rung out?
00:36.11[hC]JT: havent got that far yet. :)
00:36.18JThaha, 10-20% failure os pathetic, not bearable
00:36.29[hC]Yeah, thats not what i'd call bearable, in my business
00:36.36JTs/os/is/
00:36.45[hC]but i guess it depends on wether or not you really care that much about the fax. heh.
00:37.12JTiaxmodem+hylafax still uses spandsp, just in a different way
00:37.37Yourname`Yeah, it's more like "can it?" :D
00:37.39Pengguis ast 1.4 stable enough for production?
00:37.47Yourname`Send as many successful faxes as possible, etc
00:38.31Pengguwhich would be better... spandsp+asterfax or the iaxmodem/spandsp/hylafax combo?
00:38.33[hC]Yourname`: you know what... since what you're going for here is fax spam, yes, use iaxmodem over wan IP.
00:38.42[hC]Yourname`: your 10-20% failure rate means less spam for the world.
00:38.43[hC]:)
00:38.55Yourname`haha
00:39.01Sweeperbut hey, at least it'll be high quality spam
00:39.01Pengguanyone actually use asterfax?
00:39.02Yourname`I'm not going to though.
00:39.10Sweepernone of that blurry shit you get these days
00:39.29Yourname`I'm trying to see what things are really possible with asterisk, and what we can make out of it..
00:40.33Sweeperwell, if you have control of both endpoints, might as well go fax -> tif -> email -> tif -> fax
00:41.21Yourname`I was thinking about that, like an email2fax gateway.. where all you do is spam emails, heh. Seems easy.
00:41.25*** join/#asterisk Cresl1n (n=matt@12.150.240.229)
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00:41.46Nivexnmap the internets for things listening on port 9100 and send postscript on that port ;)
00:42.03*** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net)
00:42.23knarflywhy does asterisk-addons have to include mysql?
00:42.37knarflyis there a way I can install add-ons without mysql?
00:42.43Yourname`haha Nivex, sounds neat
00:43.19*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
00:43.27Nivexthe number of printers sitting open to the world on college campuses might surprise you
00:43.28Sweeperknarfly: you can disable it in modules.conf
00:44.20knarflySweeper, cool, but what about avoiding installing it in the 1st place? is that possible...
00:45.04Sweeperoh, I see what you mean, it requires mysql and you don't have it
00:45.09Sweepermmm, take a look at the makefile, I suppose
00:45.42knarflyno, I just don't want to clutter up my server with mysql
00:46.00Pengguwith bindaddr in manager.conf, how would i specify multiple interfaces? eg bindaddr = 10.0.0.10, 127.0.0.0
00:46.14jcolpmysql isn't required... if it's available it'll be used to build the module(s) that use it
00:46.49knarflybut when you do a build with the add-ons it installs by default
00:47.40Yourname`fujin: Got sipp to work!
00:47.47jcolpif the dependencies are met yes, if you are using 1.4 then you can use menuselect to disable building/installing of it
00:47.56Yourname`fujin: I think I'm at 300 and the avg is at 1.37
00:48.30*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
00:48.31knarflyjcolp, cool, but I run FreeBSD not linux...so it's a different method for my server...I'll keep checking
00:49.44jcolpthe build process is the same unless you are using ports
00:50.50knarflyjcolp, yes, asterisk won't really install on FreeBSD unless you're a programmer with all the skills to do what the ports guys did...but it puts me at their mercy...with releases and how it's installed.
00:52.27jcolpwhat? it should install fine, provided all the minimum dependencies are met... I will perhaps check that tomorrow
00:55.11*** join/#asterisk asterisknerds (n=asterisk@66.7.122.93)
00:55.26knarflyjcolp, that would be cool. I tried buiding * from source on FreeBSD and but with so many dependencies it's not easy...and the state of the ports/packages right now makes it difficult. lots of upgrades won't work with the older versions of dependencies and I have to wait until the techies get it all just right.
00:55.41asterisknerds<PROTECTED>
01:01.09Yourname`fujin: Now at 3.63. Totalcalls: 2125. CPS 10.0(120000 ms)/1.000s
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01:07.11Pengguquestion: with rxfax()... when a phone call is answer()'ed... how does rxfax() deal with non-faxes? for e.g i want to pass the call to a queue() or bridge() if it isnt a fax
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01:28.38stefmtlsomeone knows what's happening with fwdout ? It's not working anymore since a few week...
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01:40.30Pengguanyone know if echo cancellation can be turned off in the dial plan for a specific call?
01:40.46JTfax calls turn it off
01:40.52Pengguor if TDM2400s turn of echo cancellation if/when a fax is detected?
01:40.57Pengguhmm
01:41.08Penggueven for hardware echo cancellers?
01:41.11JTyes
01:41.18Pengguexcellent, thanks
01:41.27JThardware echo cancellers are controlled by software
01:43.51*** join/#asterisk Cresl1n (n=matt@12.150.240.229)
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01:46.25Penggudigium + asterisk for faxing = nightmare
01:47.30JTespecially FXO
01:47.47*** join/#asterisk SgtDitt (n=sgtditt@adsl-074-236-020-203.sip.mia.bellsouth.net)
01:47.54Pengguwould've thought that this would be a walk in the park
01:48.29JTweird thought
01:48.40stefmtlsomeone use fwdout with asterisk here ?
01:48.41Penggulike, simple, plain old faxing...
01:49.13JTsending modem signals though a PC based soft PBX, yes very simple...
01:49.19JTwant simple... buy a fax machine
01:50.38Penggui think this would be a nice setup: have aster, and fax machien on the same line
01:50.42Penggui mean
01:50.44Penggufax-modem
01:50.49Pengguand get aster to answer
01:50.49*** part/#asterisk SgtDitt (n=sgtditt@adsl-074-236-020-203.sip.mia.bellsouth.net)
01:50.52Pengguand detect fax
01:51.01Pengguif it is, activate the modem manually to pick up the line
01:51.05Pengguand then hang up
01:51.13JTon the same line? yeah, nah
01:51.37Pengguonly problem is... aster won't know if the line is busy or not to be used for outgoign calls... may be the fax app contorlling the modem could report status back somehow
01:51.54JTyou can't use asterisk like that
01:52.02fujinheh
01:52.05fujinget an analogue line
01:52.06fujinor stop using faxes
01:52.07fujinimho
01:52.07JTyou don't share lines for your pbx with something else
01:52.13fujinotherwise use iaxmodem
01:52.22Pengguyeh im just compiling iaxmodem now..
01:52.52*** join/#asterisk SgtDitt (n=sgtditt@adsl-074-236-020-203.sip.mia.bellsouth.net)
01:52.56Pengguwas trying asterfax, too much messing about.. i get lost when the process is too complicated
01:53.58Pengguthe other day our fax lines died (telco problem)
01:54.07Pengguso i was scrambling for something
01:54.14Penggudidnt get very far
01:54.25Penggushould know next time not to scramble..
01:54.42JTshould've just got the line diverted
01:55.09Pengguthats what i had in mind
01:55.14*** join/#asterisk asterisknerds (n=asterisk@66.7.122.93)
01:55.14Penggudiverted to our pbx lines..
01:55.24Pengguand set up a soft-modem..
01:55.43asterisknerds<PROTECTED>
01:55.46JTyeah, more like diverting to another fax line ;)
01:56.44Pengguwell, we had 2..
01:56.46Pengguand both went
01:57.32Penggubtw
01:57.40Pengguwhen i do "restart when convenient" in the asterisk cli
01:57.45*** join/#asterisk Jabeeds (n=jabeeds@075.d.011.mel.iprimus.net.au)
01:57.48Penggui get no output after that
01:57.58JTyeah it will restart when there's no calls
01:57.58Penggui have to exit and go into it again
01:58.05Penggu(this is when it doesn't die yet)
01:58.25Pengguit's like it mutes all the output
01:59.06fujinmm, I've seen that too.
01:59.15fujinSome other commands cause it too.
01:59.16Penggui need a command asterisk*CLI> Announce(get y'all boody off the phone i need to restart)
01:59.55*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
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02:02.34Pengguwhat's the best tos= to use for iaxmodem?
02:03.31Jabeedstos=reliability ?
02:05.26*** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au)
02:07.47Penggui love this url: http://www.pctools.com/guides/password/?length=8&phonetic=on&alpha=on&mixedcase=on&numeric=on&nosimilar=on&quantity=27&generate=true
02:13.03*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
02:13.17SgtDittanyone using asterisknow with fwd?
02:16.47*** join/#asterisk SgtDitt (n=sgtditt@adsl-074-236-020-203.sip.mia.bellsouth.net)
02:17.30SgtDitthello
02:18.23SgtDitti'm trying to get asterisknow to work with FWD, and no luck. anyone here been succesful?
02:18.35*** join/#asterisk pruonckk (n=pruonckk@201-95-164-150.dsl.telesp.net.br)
02:18.42pruonckkhello,
02:18.48SgtDitthi
02:19.03pruonckkplease, somebody have an asterisk 1.2 with unicall working ?
02:20.21pruonckkim trying to compile, but, later compile, i have a problem with a unknow symbol (dtmf_put)
02:21.07Pengguis there any way to tell that asterisk is still waiting on a "restart when convenient" ?
02:24.40JabeedsSgtDitt: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
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02:28.55SgtDittthanks
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02:50.56JabeedsAnyone know how to discard digits in a dialplan??
02:51.20JT:1
02:51.43JabeedsIe. User Dials 6155001 -> Dial(SIP/55001)
02:53.17JabeedsCan it be done?
02:54.59JTJabeeds: ${EXTEN:2}
02:55.03JTread the book :)
02:55.05JT~thebook
02:55.05jbotsomebody said thebook was a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
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02:56.04asterisknerds<PROTECTED>
02:57.17JabeedsThanks JT
02:59.52J4k3facebook
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03:05.05blitzrageJ4k3: nerd
03:05.09blitzrage:)
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03:13.53J4k3haha
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03:30.13*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
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03:49.47fujinso, anyone run any queue analysis software?
03:53.55[TK]D-Fenderfujin, No, we all run "faith-based call centers" where everything is running at maximum efficiency, because we melieve it must :)
03:54.51ManxPowerWe have faith in Asterisk!  Praise Mark!  All hail Allison!
03:54.55fujinlol.
03:55.05fujinI have faith, queuemetrics just seems to be a buggy piece of shit
03:55.09fujin+it's written in java
03:55.11*** join/#asterisk asterisknerds (n=asterisk@66.7.122.93)
03:55.11fujin+it requires tomcat
03:55.17fujin+it parses queue_log (this is dumb)
03:55.28fujinI'd rather not write my own parser, but that's a possibility
03:55.41asterisknerds<PROTECTED>
03:55.47[TK]D-Fenderhttp://www.cafepress.com/buy/democracy/-/pv_design_prod/pg_1/p_storeid.46696249/pNo_46696249/id_10979029/opt_/fpt_/c_360/
03:55.47[TK]D-Fender^^^^^^^^^^^^^^^^^^
03:56.03fujinlol
03:56.05fujinamerica fails at life
03:57.19ManxPowerI'd prolly get my ass kicked by the locals if I put that on my truck.
04:00.23[TK]D-FenderManxPower, What state are you in again?
04:01.26x86omg... I've got a real problem
04:01.59x86I've got 17 PSTN lines coming into a rhino FXO channel bank, hooked up to a sangoma A102DX dual-span T1 card
04:02.21ManxPower[TK]D-Fender: Alabama
04:02.24x86I've also got 24 stations hooked up to another rhino FXS channel bank, also going to the same sangoma A102DX card
04:02.37x86I can't figure out why asterisk will not start!
04:03.03[TK]D-FenderManxPower, Figured they be more receptive to that kind of brandishing....
04:03.24[TK]D-Fenderx86, PASTEBIN <---------
04:03.53ManxPower[TK]D-Fender: They take the 11th Commandment very seriously here.
04:04.12[TK]D-FenderManxPower, "Thou shalt not waaannnnnaaaaa?"
04:04.35ManxPower11) If you are not a patriot you are a traitor.
04:05.18ManxPowerI'm an atheist.  It's like being back in the closet around here with my views on religion.
04:05.36[TK]D-FenderManxPower, Where "patriot" = "whatever THEY are thinking should be done about everything".
04:06.11x86[TK]D-Fender: was working on it! :P
04:06.17x86http://www.pastebin.ca/660612
04:06.18[TK]D-Fenderreligeous zealots of any faith piss me off......
04:06.29ManxPowerI'm a devout atheist.
04:06.29x86that's the "full" error log
04:06.39JTisn't patriot a misspelling of "parrot"
04:06.59x86http://www.pastebin.ca/660614
04:07.06x86that's /etc/zaptel.conf
04:07.12[TK]D-FenderJT : Synonym or homonym : you be the judge!
04:07.26JT:)
04:07.34x86# cat /etc/asterisk/zapata.conf
04:07.34x86[channels]
04:07.34x86; Sangoma A102 port 1 (channels 1-24) (internal T1 to stations)
04:07.35[TK]D-Fenderx86, And verification of wanpipe, ztcfg, etc?
04:07.42x86yessir
04:07.47[TK]D-Fenderx86, Show me the card is sane
04:07.51x86hold on, didnt mean to paste that there...
04:08.00x86<PROTECTED>
04:08.04x86that's zapata.conf
04:08.36x86zttool was telling me OK on both interfaces a second ago, now it's telling me UNCONFIGURED
04:08.39x86what the hell?!
04:09.13x86is there something wrong with my zaptel.conf or zapata.conf?
04:09.30[TK]D-Fenderx86, go verify WANPIPE and ZTCFG *now*
04:10.43x86ztcfg told me changing from "unused" to "fxo" and "fxs" a bunch of times, now zttool is showing me OK again instead of UNCONFIGURED
04:11.02x86ok, so asterisk is running now, but it's acting crazy...
04:11.12x86it's showing all my extensions going off and back on hook
04:11.46ManxPowerx86: your signaling is screwed up
04:11.46x86ah
04:11.46x86lemme reverse it
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04:11.51*** mode/#asterisk [+o codefreeze] by ChanServ
04:11.53[TK]D-Fenderx86, Perhaps your cards are initializing out of order...
04:12.19ManxPowerchannels going to the telco should be fxo ks, channels going to phones should be fxs ks
04:13.07JTManxPower: don't you mean the other way around?
04:13.16x86ManxPower: dont FXO lines use FXS signalling, and vice versa?
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04:17.19[TK]D-Fenderok, I'm done with the night... later all...
04:17.52ManxPower~fxofxs
04:17.55jboti heard fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
04:17.58ManxPowerLet's ask jbot
04:18.06ManxPowerargh
04:18.15ManxPoweryes, fxo ports use fxs signaling, etc.
04:18.20ManxPowerit's been a long day
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04:27.45x86ok, so i swapped the cables going into the sangoma
04:28.13x86now the console is not going crazy with zap channels going off/on hook, but when i pick up a station i get nothing but dead air
04:28.17x86any ideas?
04:31.24JTyeah, give us some errors and current zaptel, wanpipe and zapata.conf in pastebin, preferably a single pastebin
04:32.30x86no errors at all
04:33.56x86http://www.pastebin.ca/660644
04:37.37x86any ideas JT
04:39.08x86when I dial out via Zap/g2, it answers the call immediately (I hooked up a Polycom IP301 for testing)
04:39.14x86never rings
04:40.20tzafrir_laptopx86, zaptel now comes with nice tools such as xpp/utils/genzaptelconf and xpp/utils/zapconf . I wonder how they fare with Sangoma hardware...
04:41.00tzafrir_laptopnever mind
04:41.03JTtzafrir_laptop: sangoma generates its own configs
04:41.59tzafrir_laptopyeah, and what happens if you have sangoma + something else?
04:42.05JTno idea
04:42.24x86i'm only using sangoma hardware
04:42.42tzafrir_laptopI believe that their scripts attempt to configure that "something else" as well. Just as genzaptelconf attempts to configure sangoma
04:44.14x86ok, so any ideas here?
04:44.23x86I'm out of ideas
04:44.28tzafrir_laptopx86, do you have /proc/zaptel/1 ?
04:44.51JTx86: does everything look good when you run ztcfg -vv ?
04:45.11*** join/#asterisk saftsack (n=saftsack@pD9E07502.dip.t-dialin.net)
04:45.37tzafrir_laptopwhat is the output of:  ls /proc/zaptel
04:45.58Pengguwhats this mean: "chan_iax2.c:5159 register_verify: Peer 'iaxmodem' is not dynamic (from 10.0.0.10)" ?
04:46.55x86JT: yessir
04:47.36x86http://www.pastebin.ca/660654
04:47.50x86# ls /proc/zaptel
04:47.51x861  2  3
04:48.05x86as it should be
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04:48.57x86i've tried swapping the channel banks to different ports on the sangoma
04:51.26x86ok, got the SIP phone to make calls out Zap/g2!
04:51.53x86analog stations get dialtone, but that's about it
04:52.47x86i try to dial, and the dialtone doesn't go away
04:52.53x86and eventually i get a fast busy
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04:55.49x86cool! stations can dial out!
04:55.52asterisknerds<PROTECTED>
04:56.41Pengguwhat's this: ast_sched_del: Attempted to delete nonexistent schedule entry 15721 ?
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05:01.18x86some of my analog stations work fine
05:01.25x86others are f*ed up
05:01.32x86i think the wiring guy messed up badly
05:13.49x86ok, I can call _all_ of my analog stations from my SIP phone
05:13.57x86but only some of my analog stations can dial out
05:14.28x86most of them never stop giving dialtone when you start dialing, and give you fast busy after about 5 seconds
05:14.50x86asterisk sees when they go off hook, and when the fast busy starts, asterisk CLI reports hangup
05:15.00x86so what could be the problem?
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05:36.50Penggufinally got fax working
05:37.21x86is there anything with timing or something that could be causing Asterisk (or perhaps the Rhino channel banks) from not seeing digits pressed?
05:37.40x86I can take a good phone, move it to a location I thought was bad, and it works perfectly fine
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05:44.58booraydoes anyone have any particular wireless handsets that they like that work well with asterisk?
05:46.59JTno wifi ones
05:47.07JTlook at DECT units
05:48.05booraygoogling...
05:50.44boorayany favorites just the same?
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06:59.03boorayanyone play with the aastra 480i ct?
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07:10.47creativxbooray: gn netcom 9330
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07:45.13CoolGuy21hi, is there a good toolset that will allow me to run asterisk as a hosted pbx solution for customers?
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08:06.02RsaMan# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
08:06.03RsaManfxols=1
08:06.03RsaManfxsks=2
08:06.03RsaManfxsks=3
08:06.03RsaManfxsks=4
08:06.04RsaManoops
08:06.06RsaMansorry
08:06.26RGi_26does asterisk still work with BRI ISDN lines ?
08:06.46JTRGi_26: yes, with certain driver sets
08:09.02x86BRIStuff :)
08:09.32JTyeah
08:09.54RGi_26like the old HFC card ? like asuscom etC ?
08:10.37JTif they're hfc, they're doable
08:10.46tzafrir_laptop~thebook
08:10.47jbothmm... thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
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08:11.58RGi_26hmm bristuff.. hmm I have to look into that..
08:12.10RGi_26but bristuff has to be compilled into asterisk ?
08:12.17JTyes
08:12.55RGi_26hmm.. then I cant use Trixbox.. :(
08:14.10JTawesome
08:17.23*** join/#asterisk incorrect (n=incorrec@host81-138-107-161.in-addr.btopenworld.com)
08:17.36JTRGi_26: trixbox sucks
08:18.01incorrecthello, is it possible to use asterisk as a text based chat as well as voip?
08:18.09JTnot really
08:18.46incorrectshame
08:19.09*** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
08:19.12RGi_26JT :well.. I like it because it easy to use and support...
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08:19.33RGi_26JT: editing the extension.conf etc is crap.. hehe :P
08:25.20RGi_26JT : hmm looks like trixbox have a install script for  zaphfc :)
08:25.30RGi_26wondering if its going to work.. hehe :)
08:25.49JTno, it's actually a good way to do things
08:26.00JTRGi_26: if you use trixbox, you cannot get help here
08:26.02JT~trixbox
08:26.03jbot[trixbox] a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
08:26.18JTthe dialplan trixbox produces are utter garbage
08:26.58sergee~jbot
08:26.58jbot[jbot] a hack!, or known to have only said one useful thing.
08:27.23RGi_26JT : hehe.. yeh.. I dont understad to much of it.. hehe :)
08:27.30JT~zeeek
08:27.31jboti heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."
08:28.26RGi_26JT : but you support asterisk and freepbx ?I like the gui..:)
08:28.30UatecLOL
08:28.31JTno
08:28.32Uatecthat's amazing
08:28.37Uateci'm going to use that quote!
08:28.39RGi_26doh.. hehe :)
08:28.41JTRGi_26: we absolutely DO NOT support them AT ALL here?
08:28.47RGi_26okaydokay.. :)
08:28.48JTs/here?/here./
08:29.23JTfreepbx, trixbox, almost the same thing to us
08:30.55Uatecinfact
08:31.02Uateci'm going to email that quote to my customer
08:31.02RsaManhello guys
08:31.25RsaMani have run genzaptelconf
08:31.32RsaManit sets all teh channels there
08:31.40RsaManbut when i run ztcfg i get no output
08:31.45RsaMannot even an error
08:31.46RsaMan:(
08:32.46jeremy_ggood morning
08:32.48tzafrir_laptopztcfg or ztcfg -v
08:32.50jeremy_gamigoes
08:32.59tzafrir_laptopztcfg (without -v) produces no output
08:33.11RsaMani see
08:33.11RsaManthanks
08:33.16tzafrir_laptopIt gives errors if there are. But if it is silent, then al is well
08:33.22jeremy_g:D
08:33.30jeremy_gvery interesting
08:33.42jeremy_gtzafrir cohen :) how do you do
08:34.09RsaMansweet
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08:39.10RsaMan:q
08:41.13tzafrir_laptopjeremy_g, hi
08:41.42tzafrir_laptopbreaking zaptel, as usual
08:42.13RsaManwhat do i use to dial on my tdm400 now
08:42.22RsaManmy zapat.conf seems to be in order
08:42.43RsaMando i just call Dial( zap/1/${EXTEN})
08:42.44RsaMan?
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08:49.07jeremy_gis it possible to compile * without ssl?
08:49.29UatecI hated the phase everybody when through of having custom backgrounds to their IE4 toolbar. >:(
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08:51.21joelsolankiHi room
08:51.45yidiyuehanhi,if i download the asterisk-1.2.24.tar.gz, do i still need to apply the asterisk.24-patch.gz etc?
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08:57.15RsaManhi i am back
08:57.19RsaManExecuting [111@internal:1] Dial("SIP/alain-08b6a9e0", "Zap/2/0824111334") in new stack
08:57.19RsaMan<PROTECTED>
08:57.19RsaMan<PROTECTED>
08:57.19RsaMan<PROTECTED>
08:57.29JTback and flooding
08:57.36RsaManit seems i can dial with my tdm400, but ... the damn thing answers
08:57.39tzafrir_laptop~pb
08:57.40jbotA Pastebin is a place to paste your stuff 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
08:57.42*** join/#asterisk implicit (n=implicit@210.16.55.38)
08:57.53RsaMansorry i thought 4 lines was pushing it
08:58.09RsaMani dial using the sip client and it answers immediatly
08:58.16RsaManbut i am trying to call my cellhpone
08:58.49tzafrir_laptopso, to which context does it go? you should see that in 'zap show channels'
08:59.03tzafrir_laptopnow look for the extension 's' in that context
08:59.12tzafrir_laptopshow dialplan CONTEXTNAME
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09:03.56hi365tzafrir_laptop: hi
09:04.08tzafrir_laptophi
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09:04.13RsaMan2hi
09:04.18RsaMan2sorry my adsl reset
09:04.51hi365tzafrir_laptop: you have/use/know of a script that uses externotify to send you an sms when you get a new VM?
09:05.48*** join/#asterisk kkn088 (n=kikoun@88-138-6-52.adslgp.cegetel.net)
09:05.50RsaMan2i am confused
09:06.09tzafrir_laptopsend you sms via what service exactly?
09:06.20tzafrir_laptopsendsms?
09:06.28RsaMan2i plugged my phone line into my fxo port,
09:06.38hi365you tell me :) im open to suggestion
09:06.47RsaMan2when i dial to it using dial/1/0993, it just rings
09:06.55*** join/#asterisk guillote_GNU (n=guillote@host35.200-117-217.telecom.net.ar)
09:07.04RsaMan2but its not the phone i will calling that rings
09:07.36RsaMan2it does not seem to be dialing
09:08.06*** join/#asterisk qdk (n=qdk@213.150.62.32)
09:08.27RsaMan2i am not sure what is ringing and what is not ?
09:08.49RsaMan2exten => 111,1,Dial(Zap/1/0824111334)
09:08.56RsaMan2my dialplan looks like this
09:09.12RsaMan2and when i read is, it wil use channel 1 zaptel device which is my fxo channel
09:09.16RsaMan2and dial that number
09:09.20RsaMan2right?
09:09.46tzafrir_laptopright
09:09.48*** join/#asterisk implicit_ (n=implicit@vc241211.vpn.uci.edu)
09:09.52RsaMan2as i dial
09:10.00RsaMan2it rings immediately
09:10.05RsaMan2but my cellphone does not ring
09:11.50*** join/#asterisk dominic1 (n=dob@213.221.82.242)
09:11.56[hC]RsaMan: you should be hearing a ring as though you would hear it if you dialed that pstn line directly
09:12.03[hC]not a superbly clear asterisk ringing tone
09:14.15RsaMan2aaaaah
09:14.26RsaMan2i have 3 fxo modules and 1 fxs
09:14.37*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
09:14.57RsaMan2but its confused
09:15.19RsaMan2http://pastebin.ca/660794
09:15.28RsaMan2it detected it the opposite way around
09:15.40*** part/#asterisk dominic1 (n=dob@213.221.82.242)
09:16.37RsaMan2:(
09:18.33hi365tzafrir_laptop: you tell me :) im open to suggestion
09:19.47tzafrir_laptopthere's a sendsms script that uses the icq interface, or sme celular
09:20.11tzafrir_laptopif you have a port to Bezeq, you can send SMS messages through there
09:20.51tzafrir_laptopwith the utility smsq
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09:22.14DavieyCan anybody tell me why this isn't working?
09:22.15Davieyhttp://pastebin.com/d40655b77
09:22.25DavieyThe callerid doesn't seem to get set?
09:22.57DavieyBut works fine when a call is made, with the callerid set per extension in users.conf
09:23.11tzafrir_laptophi365, Nadav Har'El's script: http://nadav.harel.org.il/software/sendsms/
09:23.25Davieyie callerid per extension works, but not when done this way...
09:24.08tzafrir_laptopthe version there is from 205. I believe it got broken since (or at least parts of it)
09:25.47hi365tzafrir_laptop: that looks very cool, going to have a deeper look at it. do you have a script that gets caleld when a caller leaves you a new voicemail?
09:25.52s0ckRemoving zaptel module: ERROR: Module zaptel is in use by zttranscode
09:26.09s0cki guess that is stopping me from running genzaptelconf
09:26.10s0ckany ideas...
09:26.26tzafrir_laptops0ck, which version of genzaptelconf?
09:26.37tzafrir_laptopthe one that comes with zaptel?
09:26.51s0cknot entirely sure, ill --version it now?
09:26.57tzafrir_laptopnot the one in /usr/local/sbin in trixbox
09:27.01tzafrir_laptopgenzaptelconf --help
09:27.33s0ckit is indeed in that path
09:28.02s0ckand --help is an illegal option apparently :P
09:28.24RsaMan2guys
09:28.32RsaMan2that genzaptelconf is confused
09:28.46tzafrir_laptopyeah, /usr/local/sbin/genzaptelconf of trxbox
09:28.47RsaMan2it mixed up my fxo and fxs channels
09:28.59RsaMan2that cant be good
09:29.00RsaMan2:(
09:29.01s0ckam i forked
09:29.03tzafrir_laptoptry: /usr/sbin/genzaptelconf --help
09:29.09s0ck-h the flag is
09:29.11tzafrir_laptophmmm.....
09:29.13s0ckand it doesn't give a version number
09:29.36tzafrir_laptopecho   ZAPATA_FILE=/etc/asterisk/zapata-auto.conf >>/etc/sysconfig/zaptel
09:29.44tzafrir_laptopbefore you run it
09:29.48tzafrir_laptopand also:
09:30.19tzafrir_laptopecho   context_lines=from-zaptel >>/etc/sysconfig/zaptel
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09:30.49s0ckzapata-auto being the blank one?
09:30.52tzafrir_laptopthis should make the output more like what trixbox expects
09:31.37tzafrir_laptopyou probably already have #include zapata-auto.conf in /etc/asterisk/zapata.conf
09:31.40RsaMan2wait i am an idiot
09:33.01RsaMan2http://pastebin.ca/660809
09:33.18RsaMan2does this setup tell u that there are 3 fxo channels
09:33.21RsaMan2and 1 fxs channel?
09:33.31RsaMan2the signalling is confusing me
09:34.17RsaMan2because when i dial on channel 2...Zap/2-1 answered SIP/alain-08b6a9e0
09:34.22RsaMan2something answers
09:34.29RsaMan2but it does not dial on my phoneline
09:34.38s0cktzafrir_laptop: cheers for that
09:34.45RsaMan2any ideas?
09:34.46s0ckim not entirely sure it hasn't worked
09:34.51s0ckzttool shows me the card
09:34.56RsaMan2i am so close to calling my phone
09:34.58s0ckbut i get chanunavail when trying to dial etc
09:35.08tzafrir_laptops0ck, what card is that?
09:35.19s0ckx100p (yes, yes)
09:36.11s0ckwould you expect it to show up in zttool if genzaptelconf had not worked
09:37.07tzafrir_laptopDo you have a zaptel trunk for Zap/g0 ?
09:37.12tzafrir_laptopIf so: just use it
09:37.18tzafrir_laptopIf not: add one
09:39.25s0cki added zap/g0, no joy tho
09:40.05tzafrir_laptopzap show channels
09:40.15tzafrir_laptopmaybe you'll need to restart asterisk
09:40.19tzafrir_laptopneed to go now
09:40.47s0ckhmm
09:40.51s0ckonly pseudo
09:42.05s0ckzapata-auto remains unpopulated
09:43.40s0ckthanks anyway :)
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09:55.52asterisknerds<PROTECTED>
10:00.57*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
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10:10.48tzafrir_laptops0ck, ls -l /etc/asterisk/zapata*.conf
10:11.01RsaMan2is it possible to store sip users in a database?
10:11.33RsaMan2i want to make it easy for users without installing asterisk-gui
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10:15.33jeremy_gRsaMan2:yes its possible
10:15.47RsaMan2sweet
10:15.55jeremy_gRsaMan2: i am using it and i love it
10:17.15jeremy_gis it necessary to use spandsp
10:17.24jeremy_gcant asterisk get compiled without it
10:18.07RsaMan2trying to think of a good solution to add new sip users
10:18.12RsaMan2the easy way
10:18.18s0cktzafrir_laptop: genzaptelconf just kinda hangs
10:18.19RsaMan2for newbies
10:18.27s0cki left it ages, it doesn't seem to do anything on this box
10:19.35jeremy_gRsaMan2:you should my gui
10:19.57jeremy_gRsaMan2:i developed it for a client that runs everything from db
10:20.05jeremy_gno extensions.conf or sip.conf
10:20.07tzafrir_laptops0ck, hangs?
10:20.13RsaMan2jeremy_g: where can i get hold of it ?
10:20.25RsaMan2jeremy_g: sounds like something i would like
10:20.30tzafrir_laptopWhat does 'ps fax' shows? is a sub-process of it hung?
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10:21.00jeremy_ghmm
10:21.09jeremy_gi should ask my client to open it
10:21.17jeremy_gclosed source suxs
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10:22.18masusi need a voip provider in germany. anyone know a good company ?
10:23.04jeremy_gmasus:voipcheap.com :p
10:23.04RsaMan2bleh
10:23.10jeremy_gits supposed to be german based
10:23.12jeremy_gnot sure
10:23.21RsaMan2jeremy_g : are there any other open source guis that do just that ?
10:23.32RsaMan2i dont want the thing to take over my asterisk
10:23.41RsaMan2just add sip clients ,
10:23.41jeremy_ghehe
10:23.43RsaMan2i dont want trixbox
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10:23.46RsaMan2of asterisk gui
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10:23.54jeremy_gi  am not sure
10:24.00jeremy_gnot to my knowledge
10:24.10RsaMan2asterisk gui , uses users.conf, which is not what they say in ~thebook
10:24.16RsaMan2so i dont believe in it
10:24.18RsaMan2i am a noob
10:27.07Uatecis there a gui in asterisk Business edition 2.2.1?
10:27.37RsaMan2probably
10:27.42RsaMan2u pay soo much
10:28.09Uatecthat's not really a helpful answer
10:28.20RsaMan2lol
10:28.21RsaMan2sorry
10:28.24RsaMan2i am not sure
10:28.32RsaMan2but i would probably say yes
10:28.42Uatecwell i wonder how to find it
10:32.29jeremy_gRsaMan2:what features do you want on a gui
10:35.38RsaMan2jeremy_g : add new sip users and allocate extensions
10:35.57RsaMan2jeremy_g : i might move from sip to IAX but it will do the same thing
10:36.29RsaMan2is there an asterisk command to check who my iax users are ?
10:36.32RsaMan2like sip show users
10:37.16masusjeremy_g: i need a real company i want to pay ;)
10:37.23masusnot free
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10:39.24jeremy_gRsaMan2:how much would you be willing to pay for such a gui
10:39.56RsaMan2jeremy_g : not willing to pay , sorry, i can make my own, just wondered if such a thing exists
10:39.57RsaMan2:)
10:40.25jeremy_ghehe
10:41.14RsaMan2hehe
10:41.24RsaMan2i just dont want to reinvent the wheel
10:41.46jeremy_ggood,me too
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10:44.05jeremy_gRsaMan2:can you figure out whats stopping from my * from getting compiled http://www.pastebin.ca/660887
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11:12.02CoolGuy__hi
11:12.10CoolGuy__1.2.17 is a good stable version?
11:13.10Uatecgood stable and old
11:17.35styelzanyone know if its possible to specify a User Ring tone on the tftp cfg file for a Linksys SPA942 ?
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11:26.26UatecI'm sure it must be
11:26.28UatecI have a 922
11:26.31Uatecbut it's the same firmware
11:26.39Uateci don't have A 922
11:26.41Uateci have about 20
11:26.56Uatechave you looked in the administration guide?
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11:45.12flujanhi guys, I am with a terrible problem with asterisk.
11:45.23flujanI am having a lot of this messages on my CLI.
11:45.25flujanChannel 0/28, span 1 got hangup request, cause 111
11:45.28flujanChannel 0/28, span 1 got hangup request, cause 112
11:45.33flujanChannel 0/28, span 1 got hangup request, cause 16
11:45.50flujanis is a example of a specific channel.
11:46.18flujanbut others are having trouble will all my T410P channels.
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11:51.14dagrimare there any incoming IAX providers that allow you to bill the caller per call directly to their phone bill?
11:51.19dagrimlol
11:53.43Uatec?
11:56.17dagrimLets say Im offering support of any kind.. how can I bill the customers via phone bill.. is this even possible with *?
11:58.00MindTheGapmorning all, management needs monitoring abilities but also would like some privacy on selected channels. Ive set up 2 zap groups. group 1 can be monitored, g2 cannot. Im forcing outgoing calls from management to go through g2, but how do I prevent them being monitored if a call comes in on g1 and gets transfered to one of their extensions?
12:00.15creativxgreat
12:00.21creativxour itsp just fell off the internet
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12:09.50styelzUatec: the only way ive found i can do it is with using a direct call to the phone web server, or using the sipura tone generator tool.
12:10.06styelzbut nothing in the way of auto config via tftp
12:16.35styelzlike.. http://phone-ip/ringtone1?tftp://tftpserver-ip/ringtone1.dat
12:16.45styelzbut i need to do that manually
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12:17.21dagrimso nobody knows of providers that offer toll DIDs?
12:17.42Nivexdagrim: have you checked the wiki>
12:17.42Nivex?
12:17.50dagrimyes
12:17.54dagrimlol
12:18.03dagrimno luck
12:18.38dagrimonly thing i could find was one that gives you a free DID but charges your caller, and they keep the fee
12:18.53*** part/#asterisk dominic1 (n=dob@213.221.82.242)
12:19.21Sweeper~book
12:19.22jbotbook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
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12:34.10GalerasSirs, which gateway do you recomend for E1 -> asterisk?
12:38.17tzafrir_laptopGaleras, gateway? you can also use a E1 card
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12:38.51Uatecstyelz, manually edit the onboard addressbook
12:38.58Uatecand then dump the file to disk and see what's up
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12:42.54cy_`hello
12:44.53cy_`i would like to just try out asterisk without much of installation.. and i saw there are several modified knoppix around with asterisk pre-installed.. a list at http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM ... which one would you recommend me to try?
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12:46.23lirakismorning
12:47.13Mavviehow do I with linux see the interrupt assignments again?
12:47.19Mavvieaha, /proc/interrupts
12:49.20jeremy_gcy_`:try astbill
12:51.14cy_`jeremy_g, thanks
12:51.42jeremy_gtzafrir_laptop: http://www.pastebin.ca/660983
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12:53.15tzafrir_laptopthose are the last lines. But where does the error begin?
12:53.23tzafrir_laptopa missing include or something?
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13:14.12datachompercy_` Asterisk isn't that hard to install, imho
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13:18.33hankhi
13:19.09hankautologoff is only for agents not for queue members, right? is there some aequivalent for queue member?
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13:23.21Airwolf-how do i know the address of a SIP phone to be dialed on incoming call ?
13:24.27Airwolf-i want to pass the address to AGI
13:24.58hankhttp://www.voip-info.org/wiki-Asterisk+config+agents.conf On the first line it says: "Agents are configured in the file queues.conf" and at the bottom: "this last section contains the definition of the agents." Which of those two statements is correct now?
13:25.59Airwolf-it's a matter of preference if i'm not mistaken
13:26.04[TK]D-FenderAirwolf-: referring to IP address?  "to be dialed"?!? huh?
13:26.38[TK]D-Fenderhank: agents.conf defines the agents, queues.conf choose which QUEUES they are a member of
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13:27.05Airwolf-[TK]D-Fender: when a SIP phone register itself on asterisk, i assumed that asterisk store the address that phone. correct me if i'm wrong
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13:27.16[TK]D-FenderAirwolf-: Yes.
13:27.44Airwolf-how do i get that SIP phone address from AGI ?
13:28.25Airwolf-(pass that to AGI)
13:31.51[TK]D-FenderAirwolf-: "show channel [channelname]" in your AGI would probably have that
13:33.11Airwolf-ok
13:33.16Airwolf-thank you [TK]D-Fender
13:33.40s0ckanyone dealt with toshiba ctx's? :P
13:33.42knarflycan anyone tell me what will happen if I set USE_MYSQL=no in the Makefile for asterisk-addons?
13:34.03s0cktrying to set the number presented on the outbound calls
13:34.10s0cktis the wrong ddi atm
13:34.43jeremy_gtzafrir:http://pastebin.com/m547ff16 <--- thats the error
13:34.48jeremy_gcomplete config.log
13:34.54jeremy_gtzafrir_laptop :http://pastebin.com/m547ff16 <--- thats the error
13:35.03hank[TK]D-Fender: yes i finally found http://www.voip-info.org/wiki/view/Asterisk+Agents which explains it better
13:35.07hank[TK]D-Fender: thanks though
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13:38.44tzafrir_laptopjeremy_g, sorry. I really have no clue there
13:40.50jeremy_gtzafrir_laptop:thanks for the effort, its really weird.
13:40.53flujanguys. someone already have problems with asterisk and linux framebuffer? I think my digium board is droping calls because of that,
13:41.41*** part/#asterisk hank (n=hank@leonardo.netwichtig.de)
13:43.30[TK]D-Fenderflujan: As in you turned it off and now everything is fine?
13:44.31flujan[TK]D-Fender: nops... i am recompiling the kernel...
13:44.45flujanif I restart the machine the board stop the dropping...
13:45.09flujan[TK]D-Fender: some problem I had sometime ago, remember? The timing one? I resolve it changing the board to another server...
13:46.09flujanops
13:46.12flujansame problem. :D
13:48.30[TK]D-Fenderflujan: wHAT BOARD?
13:49.13flujan[TK]D-Fender: digium TE410
13:50.10[TK]D-Fenderflujan: ....my condolences :/
13:50.46flujan[TK]D-Fender: what this board sucks?
13:51.03[TK]D-Fenderflujan: You know my M.O already..... :)
13:51.35flujan[TK]D-Fender: sangoma right?
13:51.44flujan[TK]D-Fender: we are installing a sip/trunk here...
13:51.54flujan[TK]D-Fender: I am burned a lot by the digium boards... BAH.
13:52.28flujanthe first board the guys sold with the wrong jumpper... not working on a E1 but in a T1 env.
13:52.39flujanI needed to discover about the jupper right here...
13:52.59flujanbah... digium + Brazilian's vendors SUX!!!!
13:53.50coppiceI didn't know anyone bothered with the jumpers
13:54.43jeremy_gcoppice:lol
13:54.43coppicehuh?
13:55.29flujancoppice: yeap...
13:55.46flujanlater I suffer to configure the r2/d2 protocol... You helped me a lot coppice...
13:55.59flujanin the end the problem was with the board, not with the software...
13:56.01coppiceI've no idea why they put the jumpers there, since you don't need them
13:56.06Corydon76-digflujan: you can override the jumper with kernel parameter t1e1override
13:56.22flujanI exchange the board for this one...
13:56.37flujanthis new board start to make problems and hangup ALL calls on a P4 machine
13:56.54flujani switch the digium board to another server... It worked up to now... BAH
13:57.01flujandammed.
13:57.43flujanthe result is that i don't trust the digium hardware anymore... We bought a 412P
13:57.54flujanI think i will just drop the board and use audiocodes.
13:58.18Sweeperflujan: bahaha
13:58.20Sweepergood luck
13:58.49flujanSweeper: do you had bad experiences with audiocodes?
13:58.55russellbflujan: have you talked to tech support to give them a chance to help you?
13:59.06Sweeperflujan: yes.
13:59.16flujanrussellb: I tryed but it is difficult from Brazil to speak with the guys...
13:59.26flujanthe reseller gives NO support at all.
13:59.31russellbyou can also do it via email
13:59.41flujanSweeper: please tell me...
14:00.22flujanrussellb: I know, but when a card start to drop all 90 channels we got a bit desesperate... I don't even think about the digium support.
14:00.32russellbwell that's what they are there for
14:00.42russellband are experts in figuring out your problem
14:00.42Sweeperflujan: mmm, fxo box I worked with refused to register properly, was a general pain in the ass, and has far too many config options
14:00.51russellbcoming in here and complaining and talking about how much it sucks doesn't help you
14:01.02russellball i ask is that you give us a chance to fix your problem before giving up
14:01.02Sweeperimo, you're better off with the digium than with audiocodes
14:01.20flujanrussellb: but what could they do if they need to exchange the board? The don't have a official reseller here... :(
14:01.31russellbyes, we have a distributor in Brazil
14:01.37Sweeperonly problem I've ever had with digium is the whole irq thing, but if you've got it working once, you'll be fine there
14:01.38flujancommlogik?
14:01.39Mavvieis Raj Jain here?
14:01.45jeremy_gSweeper:audicodes and digium, they are two different vendors of different kinda products
14:01.57jeremy_gSweeper:weed?
14:01.57flujanrussellb: commlogik?
14:01.59Sweeperjeremy_g: I concur!
14:02.01russellbflujan: maybe, not sure
14:02.18russellbflujan: but they will ensure that you get a new board if that's what you need
14:03.03flujanrussellb: well, this guys just sucks.. I discover the jump issue before their techs. :( If today I am a newbie about asterisk, that time I had never heard of asterisk ...
14:03.20russellbwell i'm sorry to hear that
14:03.37flujanrussellb: good... where can I find the digium support? Do I need to buy a license of something?
14:03.38russellbthat's why i'm asking you to give the digium support department a chance to fix your problem before you give up
14:03.43russellbno, it is free support
14:03.47russellbyou can email support@digium.com
14:03.55russellbor you can call in ... info is on digium.com
14:04.41flujanrussellb: thanks for the tip I am updating zaptel/libpri/asterisk to the 1.4.10.1 current and disabling the kernel framebuffer... Let sse if its fix the board. :)
14:04.49russellbcool
14:04.51russellbyou're welcome
14:04.58russellbthanks for supporting digium ...
14:05.06flujanthanks russellb
14:05.41flujanHum, I also bough 10 licenses for g729 lets hope i works... so I will need at least 120 licenses... We are installing a sip trunk here.
14:07.29[TK]D-Fenderflujan: If you need that many licenses you might be better off with the TC400B
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14:09.24Mavviejeremy_g: see http://bugs.digium.com/view.php?id=10481 for the issue I described yesterday to you.
14:09.33Mavvie(if you remember)
14:09.41Mavvie(if you don't, don't worry)
14:09.44flujan[TK]D-Fender: thanks FOR THE TIP... I will check it... COOL I dunno about it. :D
14:10.00nacerhi
14:10.20nacercan you explain me on what case a need FXS Or FXO module ?
14:10.35[TK]D-Fenderflujan: ~fxsfxo
14:10.36nacerFXS analog phocall FXO data call like  FAX ?
14:10.40[TK]D-Fender~fxsfxo
14:10.41jbotextra, extra, read all about it, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it.
14:10.47[TK]D-Fendernacer: ^^^^^^^^^^
14:10.51*** part/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net)
14:10.59nacertks
14:11.02nacer:)
14:12.25nacer~fxspowersupply
14:12.29nacer:p
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14:13.14flujan[TK]D-Fender: here we are using softphones.... Let me describe the environment.
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14:13.51flujan[TK]D-Fender: We contract a sip trunk that supports both, ulaw and g729
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14:14.26flujanWe also have 4 E1, 80 X-lite softphones and 30 Eyebeams(for g729 support)
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14:14.45[TK]D-Fendernacer: You use an FXO port to plug in a LINE from your telco.  You use an FXS port to plug in a PHONE.
14:15.09nacerok
14:15.12Mavvies from server, o from originator.
14:15.14naceror fax
14:15.17Mavvielousy trick, but it works.
14:15.30nacertjs fender
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14:16.20[TK]D-Fendernacer: yes, or fax.  However if you know whats good for you you will keep faxing as far away from * as possible.
14:16.50tsuk[TK]D-Fender: what makes you say that? that was my next project :/
14:16.51Corydon76-digWhat's wrong with fax and *?
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14:17.36Sweeperwell, it's fax
14:17.58[TK]D-FenderCorydon76-dig: I *like* getting them all at an acceptable %
14:18.23Corydon76-digYes, there's something wrong with certain boards, but there are other boards that don't have that problem
14:18.51[TK]D-FenderCorydon76-dig: Ok, get specific then.  Which are "good", and which are "bad"?
14:19.01*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
14:19.08flujanhere goes the sip.conf configuration: http://pastie.caboo.se/88605
14:19.08Sweeper"digium" and "people who don't play licensing fees"
14:19.16Sweeper*pay
14:19.21Corydon76-digThe TDM400 is bad.  The TDM800 and TDM2400 are good
14:19.31Sweeperosnap
14:19.42Corydon76-digTE405, TE410, TE205, TE210 are all good
14:20.44flujanWhen I configure eyebeam to use the g729 and only the g729 protocol and dial using the trunk all goes well
14:20.47anonymouz666don't say TDM400 I ordered one right now
14:20.53anonymouz666is bad
14:20.59Corydon76-digTDM400 is fine for voice
14:21.12Corydon76-digbut don't try to use it for fax
14:21.14[TK]D-FenderF-I-N-E ;)
14:21.25flujanWhen I add the alaw/ulaw codec to the eyebeam in order to support the other extensions and dial throught the trunk... the call is completed but it statys muted.
14:24.04tsukon the topic of cards, has anyone used zapmicro cards? ZMX-100 in particular
14:24.07*** join/#asterisk mocker (n=user@198.247.173.227)
14:25.20[TK]D-Fendertsuk: ...
14:25.21[TK]D-Fender~cheap
14:25.22jboti guess cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
14:25.25flujan[TK]D-Fender http://pastie.caboo.se/88607 how the calls go... muted with eyebeam with two codecs enabled....
14:25.49tsuk:D
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14:28.29mockerCan someone take a look at the example at http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext and let me know if that's actually the way regcontext is supposed to work?
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14:29.23mockerJust adding a priority 2 for each extension that you want to dial to your sip registration context?
14:30.25mockerBecause when I do a 'show dialplan' I see two 'sipregistration' context, one created by pbx_config and one created by SIP
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14:35.56[TK]D-Fendermocker: Pastebin it.
14:36.11coolhpGood morning all.
14:36.22mocker[TK]D-Fender: My 'show dialplan' ?
14:36.50[TK]D-Fendermocker: Clearly.
14:37.53coolhpI was wondering : Has anyone ever attempted to write a SIP Proxy that would also relay REGISTER requests to another host ?
14:38.07coolhpI mean it as a NAT traversal solution...
14:38.36mocker[TK]D-Fender: http://pastebin.ca/661084
14:39.38flujancoolhp: for what I know, the openser DONT do it, and it looks a good open source sip proxy.
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14:39.43[TK]D-Fendermocker: Well not sure if its normal to see ti split... I can believe it and imagine that it WORKS.  the problem is that your PATTERN match just isn't going to happen....
14:39.43mocker[TK]D-Fender: I guess I'm questioning that example on voip-info
14:40.20[TK]D-Fendermocker: 5100,1 will NOT continue on _51XX,2 <---
14:40.27[TK]D-Fendermocker: that is dialplan 101
14:40.35mockerYeah, I see that.
14:40.41coolhpflujan : That was my thought too... but I'm also wondering why... it doesnt make much sense...
14:40.47mockerI'm wondering wtf the point of having a sip registration context is then.
14:40.51[TK]D-Fendermocker: chacnge your test and confirm it
14:41.06[TK]D-Fendermocker: It changes wheter an exten will 404 or not.
14:41.07mockerIf I have to manually write in every extension *anyway*
14:41.09coolhpWhy not create a proxy that will act as a relay for REDIRECT requests but change the From IP address in the headers...
14:41.22[TK]D-Fendermocker: You do it so you don't even get a MATCH if it isn't registered.
14:41.30coolhpThat would allow SIP registrations to be proxied too....
14:41.43coolhpAnd therefore NAT issues for multiple users behind NAT would be resolved.
14:41.46[TK]D-Fendercoolhp: Just install Open SER and move on :p
14:42.02coolhpAll you'd have to do is forward your SIP ports to a single machine or device.
14:42.05mocker[TK]D-Fender: I'll test w/ 5100 as the match and see.
14:42.15mocker[TK]D-Fender: I still think I'm missing something because this feels hacky.
14:42.34[TK]D-Fendermocker: If it feel hacky then you DO perfectly understand it ;)
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14:42.46Sweeper[TK]D-Fender: it'd be cool if you could progress from expressions to exact matches tho
14:42.53Sweeperyou could do branching code :D
14:43.06[TK]D-FenderSweeper: Add that to the long list of what I'd think would be "cool"
14:44.16flujan[TK]D-Fender: did you check my pastie? It is a problem with my sip.conf configuration for the sip trunk and the g729 stuff?
14:44.18BlackthornWhen I boot up my box, the zztools reports the pri as unconfigured. I run sudo ztcfg and then zztools again it goes yellow then green. Is this normal? Do I need to run ztcfg as a startup config somewhere?
14:44.56MavvieBlackthorn: should be run as part of modprobe
14:45.08Mavviesee /etc/modprobe.conf:
14:45.15Blackthornok thanks
14:45.21Mavvieinstall wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg
14:47.35datachomperSo I set up to receptionists in simple queue system. One is weighted more than the other. The only way the queuing seems to work, where a call will spill over to the next receptionist, is if I limit their lines on the polycom to 1.
14:47.42datachomperHowever, this now causes them to be unable to transfer.
14:47.47datachomperAm I missing something?
14:49.11Blackthorni am running ubentu would the /etc/modprobe.conf be in a differnt place? because there is no file at the moment
14:49.34Mavviethat's always an interesting question. This is on FC6
14:51.45Mercestes<PROTECTED>
14:51.48Mercestes....
14:51.49Mercestesdamnit
14:52.06MercestesThat wasn't an advertisement. =/  Just me being retarded.
14:52.15tsukcould've been worse :P
14:52.15Blackthornhehe
14:52.20Mercestesyea...
14:52.23Mercestesvery true
14:52.30Mercestes<PROTECTED>
14:52.32Mercestesor smoething.
14:52.54Blackthornoh.. you know about that one too eh? :P
14:53.08MercestesKnow about it?  I'm the founder!
14:54.40Corydon76-digWe suspected as much
14:55.16Sweeper<PROTECTED>
14:55.39MercestesNah, that'd be the other channel.
14:55.40Corydon76-digSweeper: no, that's #asstricks
14:55.46MercestesYea, that one.
14:55.56Sweeper:D
14:56.02coppiceSweeper: why not #lambsex ? isn't that more concise?
14:56.16Sweepercoppice: ask Mercestes :P
14:56.30MercestesHow about #baaadewes
14:56.32Sweeperalso, that #asstricks actually has people in it XD
14:56.44MercestesYea, the userlist is about the same as #asterisk.
14:56.48Mercestesconversation tends to be a little different.
14:56.59tsukwell, everyone needs a hobby
14:57.00Corydon76-digA little
14:57.05MercestesJust a little.
14:57.05Blackthornwell.. being pretty new to *nix distro i'm still having a difficult time figuring these thigns out. I look on my fc4 * box (that I am replacing) I see the modprobe.conf file just like yous say.
14:57.14MercestesIn here, you go "Asterisk sucks!" and you get kicked.
14:57.25MercestesIn there you go "Asstricks sucks!" and they go "yes, we do.  *winks*"
14:57.43Blackthornbut i don't see any such file on the ubuntu box. And your responce up there confused me. do  need to run the ztcfg as a startup.. or do i need to run the command to modprobe the coard upon startup?
14:58.15MercestesBlackthorn, I think in ubuntu it might be modules.autoload.kernel.something.something.something.
14:58.59Blackthornok i'l jump over to the ubuntu cahnnel and ask there. but in that file do i need to run the modprobe command or do i just put the ztcfg?
14:58.59Mercestesor something like that.
14:59.10Mercestesoh, I remember.
14:59.37Mercestesyou put it in a <tab> <tab> <backspace> m <tab> <tab> <something with the word modules in it>
15:00.31*** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136)
15:01.49[TK]D-Fenderflujan: thx :)
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15:02.55flujan[TK]D-Fender: np. :)
15:03.04flujan[TK]D-Fender: enjoy the beer. :D
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15:04.09mocker[TK]D-Fender: Slowly starting to make sense of this.
15:04.20mockerDocumentation == mailing list
15:04.21mocker;)
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15:08.30[TK]D-Fendermocker: "if you're not running around screaming like the rest of us.... you clearly haven't understood the depth of the problem" ;)
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15:15.11mocker[TK]D-Fender: It *does* seem to work w/ _5XXX
15:15.27[TK]D-Fendermocker: That so doesn't make sense....
15:15.37mockerI just had to not use the same context name as I was using for the sip registration in sip.conf
15:15.45mocker[TK]D-Fender: Welcome to Asterisk!
15:15.45mockerer..
15:15.48mocker:)
15:15.55[TK]D-Fender:)
15:16.12mockerhttp://readlist.com/lists/lists.digium.com/asterisk-users/8/40824.html
15:16.20mockerThat explains it better than anything else I've read.
15:17.16[TK]D-Fendermvanbaak: <------------------
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15:41.48luisgrinhi, im lookin for some very cheap pbx hard and soft mainly to save phone conversations
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15:44.07Mercestes~cheap
15:44.07jbotmethinks cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
15:44.32luisgrinOK! :)
15:44.34Mercestes:)
15:44.36[TK]D-Fenderluisgrin: http://www.radioshack.com/family/index.jsp?categoryId=2032094&cp=2032052.2032075.2032077
15:44.45luisgrinthanks a lot
15:44.54MercestesWhat are you trying to do?
15:45.17[TK]D-FenderMercestes: Like he said, record calls.
15:45.36Mercestes[TK]D-Fender, technically, he said "save phone conversations."
15:45.56coppiceis that like saving fallen women?
15:45.58MercestesThat could be a recording, a transcript, CDRs, or cheaper service.
15:46.04[TK]D-FenderMercestes: Oh yes, the semantit divide!
15:46.11[TK]D-Fendersemantic*
15:46.12MercestesOr, rescuing them from ruin.
15:46.43*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
15:47.03[TK]D-FenderMercestes: Yes, hand transcriptions by nude barely-legal virgins... now if he said THAT, then he'd have your full attention ;)
15:47.45Mercestesbarely legal for sex or child labor laws??
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15:48.10[TK]D-FenderMercestes: .... whichever floats your boat
15:48.16MercestesI don't know what the big deal about virgins is.  Statistically, they never put out.
15:48.21blitzragelol
15:48.44tzangerhahha
15:48.55[TK]D-FenderMercestes: Its because they'll nothing to gauge just how much you suck at sex by ;)
15:48.59tzangerhow's that line from patch adams go?  Let's go down to the maternity ward... you know those chicks put out
15:49.07mvanbaak[TK]D-Fender: ????
15:49.10Mercestestzanger, that's it.  :D
15:49.13[TK]D-FenderMercestes: You're the best they've ever had :p
15:49.21Mercestes[TK]D-Fender, that is also statistically true.
15:49.41Mercestes[TK]D-Fender, I prefer asians tho.  I can suck all I want and they just accept it and move on.
15:49.41Sweeper[TK]D-Fender: on the plus side, with Mercestes it's not likely to be very painful ;)
15:49.43[TK]D-Fendermvanbaak: Was just referencing you here for someone thanking a ML post by you
15:49.51MercestesSweeper, .....you've never had sex with me then.
15:50.02MercestesSweeper, foreplay is painful...
15:50.04SweeperMercestes: statiscally try
15:50.08Sweeper*true
15:50.11Sweeperthe former, anyways
15:50.12mvanbaakah
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16:13.15datachomperwtf, did I accidently join #asstrix
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16:15.59tzangernobody EVER accidentally joins #asstrix
16:16.00[TK]D-Fenderdatachomper: Forgot the Vaseline, didn't you?
16:16.02tzangernice try though
16:26.31mockerGetting DUNDi to automatically know where my phones are w/o hardcoding them is becoming a chore.
16:26.46mockerBut the end result should be worth it.
16:26.57mocker(at least that's what I keep telling myself)
16:32.25mockerCurrently what's vexing me is that I have _77XX patterns on server one, and 7730 on server two.  DUNDi does a search and sees the pattern match of _77XX on the first server and never looks to see the more precise search on the second server.
16:32.52mockerWhich is how it's supposed to work, I just have to figure out how to rewrite my dialplan now.
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16:50.15rsilvaD-Fender are you there?
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17:15.35olrraianyone knows one good softphone multiplataform?
17:15.46Qwell[]olrrai: zoiper, or whatever it's called now
17:16.10olrraiok
17:17.12VoicemeupWTH is megacz ?
17:17.17Voicemeupplease msg me
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17:17.59caio1982olrrai:  Zoiper is buggy on my Mac so I tend to use SJphone as much as possible on linux as well (works beatifully, but it's SIP-only)
17:19.01Maliutasoftphones are crap
17:22.47*** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au)
17:22.49Pengguhi all
17:23.08Pengguanyone know how to change the timing of digium tdm hardware to follow the telco's ?
17:23.20Penggu(coming from a faxing perspective)
17:23.26Pengguthere's mention on online docs about this
17:23.32Penggubut not mention how* to do it
17:23.38Penggunot=no
17:23.43*** join/#asterisk Yarq (i=Nwm@88.250.219.95)
17:23.45Yarqhello
17:24.33[TK]D-FenderPenggu: span =[port],[0=our clock,1 = telco clock],[lbo = almost always 0],..........
17:25.11[TK]D-FenderPenggu: This is only documented in about a DOZEN easy to find places, and has nothing to do with fax, but rater clocking stability period.
17:25.44coppiceits secretly hidden in the example config file, too
17:25.53Dan0maN_Workheh
17:26.34Pengguhmm
17:26.45Penggumy zaptel.conf has only 3 configured lines
17:27.10[TK]D-FenderPenggu: that says a lot (of NOTHING)
17:29.14Pengguive only got fxsls=17-20, loadzone=au and someother line =au
17:29.24Penggujust reading the .conf file atm..
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17:31.13[TK]D-FenderPenggu: Doesn't sound like a DIGITAL line to me, and thats the only kind of device that has a "timing" parameter
17:31.24Penggudoes span config apply to td...
17:31.25Pengguoh
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17:32.00Pengguso on tdm i dont need to worry about that stuff (in relation to the zaptel side of thigns anyway) ?
17:32.21flujanwell, I am having problems with the codecs preferences and asterisk... here goes a example...
17:32.42flujanI have a sip trunk that support ulaw, alaw and g729.
17:33.00flujanI want asterisk to try to use g729 before trying ulaw
17:33.23flujanso disallow=all, allow=g729,allow=ulaw,allow=alaw on sip.conf
17:33.38[TK]D-Fenderflujan: If you are accepting G.729 you can remove the rest because it will not necessarily negociate  beyond the first if ti agrees.
17:34.26flujanyeap, but if I have both codecs enabled on the softphone, it dials but do not transmit audio.
17:35.12flujan[TK]D-Fender take a look at it: http://pastie.caboo.se/88607
17:35.58[TK]D-Fenderflujan: [Aug 17 11:23:14] WARNING[24659]: channel.c:2991 set_format: Unable to find a codec translation path from ulaw to g729 <- looks like you're out of licenses to me...
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17:36.50flujan[TK]D-Fender: I still don't install the licenses... I am using a softphone with g729 enabled...
17:37.14[TK]D-Fenderflujan: Problem is that * can't play back audio to the channel because of it.
17:37.17flujanit should be a pass-thru right?
17:37.44[TK]D-Fenderflujan: And you are dialing Zap in there too... that is NOT pass-through
17:37.49flujan[TK]D-Fender: strange... If i just leave the softphone with the 9729 it works....
17:37.51flujanops
17:37.54flujang729
17:38.07flujanwhen I configure the softphone with both if gives this problem... :(
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17:38.32[TK]D-Fenderflujan: a G729 softphone using your Zap will need to transcode as well.
17:38.47pifiuhey everyone
17:39.00flujan[TK]D-Fender: it is not using my Zap, it is using my trunk that supports g729
17:39.09flujanthis is what is driving me crazy. :(
17:39.10[TK]D-Fenderflujan: its only pass-through WHILE your soft-phone & ITSP are talking in the same codec.  EVERYTHING ELSE is transcoded
17:39.20pifiuHas anyone used tmobile @home hotspot service with asterisk?
17:39.41[TK]D-Fenderflujan: if * is playing a sound, then it is transcoding.  So forget IVR, etc in that case.
17:39.55flujan[TK]D-Fender: ok no ivr... just a plain call
17:40.05[TK]D-Fenderflujan: unless you pre-encode your audio to G.729
17:40.20[TK]D-Fenderflujan: Pastebin a better isolated case to examine.
17:40.22flujanwhen I remove the g711 codes from the softphone it works...
17:40.30flujandial to the sip trunk both using g729
17:41.08Yarqhow do you set the callerid of the outbound sip call ?
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17:42.42flujanI am also having this problems... http://pastie.caboo.se/88667
17:42.56SweeperYarq: http://www.voip-info.org/wiki-Asterisk+cmd+SetCallerID
17:43.02Sweeperamazing stuff, that :D
17:43.19notoriousrab1982hi all, i have a question about 1.4/1.2 - exiting from the console, with 1.2, you can type exit and get back to the command line, 1.4 does not have this and doing Ctrl+C sometimes shuts down asterisk and sometimes doesn't, how are you guys exiting from a 1.4 console?
17:43.34flujan[TK]D-Fender: I can't paste a isolated case now... the machine is on production. :(
17:43.48[TK]D-Fenderflujan: Your setup is clearly trying to transcode and can't.  There isn't anything more to be said for it...
17:44.11[TK]D-Fendernotoriousrab1982: "exit" <- thats it.
17:44.34[TK]D-Fendernotoriousrab1982: If it won't let you then you are likely not CONNECTING to a running *, but rather running it LIVE like "asterisk -gvvvvc"
17:44.36flujan[TK]D-Fender: yes, My question is these... WHY the machine is trying to transcode? BOTH, the softphone and the sip trunk supports g729.
17:44.56[TK]D-Fenderflujan: Because ASTERISK is trying to send audio.
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17:45.22flujanif I leave g729 on the softphone it works... BUT if I set up two codecs on the softphone it stop working...
17:45.25[TK]D-Fenderflujan: 100 things happen BEFORE the call is BRIDGED.
17:45.46BlackthornHow do I get zaptel to compile on ubuntu? I do a make clean and that seems ok. i then try to make isntall and it errors out
17:46.10flujan[TK]D-Fender: I see... we cant see what is happening right?
17:46.11[TK]D-Fenderflujan: perhaps your softphones prefernce is tkaing over.  Be consisitant with your devices as to what codecs they should be using.
17:46.17notoriousrab1982D-Fender - type exit and it says 'no such command' app_meetme.so => (MeetMe conference bridge)
17:46.18notoriousrab1982Asterisk Ready.
17:46.20notoriousrab1982*CLI> exit
17:46.21notoriousrab1982No such command 'exit' (type 'help' for help)
17:46.23notoriousrab1982*CLI>
17:46.35[TK]D-Fenderflujan: No, you aren't SHOWING a specific call in its entirety.  There is nothing tracable there.
17:46.38notoriousrab1982is there something missing from my 1.4 build, would this be possible
17:46.43Yarqhttp://www.pastebin.ca/661267 like that ?
17:46.51flujannotoriousrab1982: how do you started asterisk?
17:46.54flujanok
17:46.56[TK]D-Fendernotoriousrab1982: How do you get to CLI?
17:47.01notoriousrab1982asterisk -cvvvv
17:47.04MrMister2I want to have apcupsd make a phone call to my cell to warn me in case the power goes out. As anyone done something similar? I have the php script working but can't get apcupsd to call it :(
17:47.08[TK]D-Fendernotoriousrab1982: Thats what I jsut told you!
17:47.25[TK]D-Fendernotoriousrab1982: You can't exit * that way when you are not connecting to an ALREADY RUNNING INSTANCE
17:47.30flujannotoriousrab1982: start it using the -gvvvvv and them asterisk -r for the CLI. There will be the exit command there. :D
17:47.36[TK]D-Fendernotoriousrab1982: You need to run * as a DAEMON and THEN connect to it.
17:47.59flujan[TK]D-Fender: I will get a entire call and will pastebin it. :D
17:48.25notoriousrab1982so we need to set it up to do service asterisk start then do asterisk -r then you can exit using 'exit'?
17:48.35[TK]D-Fendernotoriousrab1982: Correct
17:48.48[TK]D-Fendernotoriousrab1982: Otherwise you're stuck in there until you take down *
17:49.01[TK]D-Fendernotoriousrab1982: Which you would do by "stop now"
17:49.10notoriousrab1982cheers guys, i see
17:49.20[TK]D-Fendernotoriousrab1982: which I'm sure is NOT how you want to run/monitor *
17:50.09Pengguahh
17:50.11notoriousrab1982the setup i had did not allow service asterisk start, ie setup as a daemon, so will have to do that first, i thought when you build asterisk it automatically set it up as a service to automatically run at boot
17:50.20Penggumy 1st fax recvd via iaxmodem
17:50.25Penggusent, rather
17:51.54datachomperls
17:52.40[TK]D-Fendernotoriousrab1982: it offers to install the SCRIPTS to do so but IIRC you still have to ACTIVATE them yourself.
17:54.30BlackthornHow do I get zaptel to compile on ubuntu? I do a make clean and that seems ok. i then try to make isntall and it errors out
17:54.31notoriousrab1982how do I activate them, is this a step after make install
17:54.56Blackthorninstructions says about "make menuconfig" but that does do anything
17:55.51jkiffnotoriousrab1982: Yes, and how you do so depends on your distribution.
17:55.59Dan0maN_WorkBlackthorn:  is ubuntu an RPM based distro?  i haven't used it
17:56.18jkiffDan0maN_Work: Thankfully, it is not.
17:56.20datachomperubuntu is debian based
17:56.22notoriousrab1982thanks jkiff :)
17:56.24olrraiDan0maN_Work: no, is a .deb based
17:57.16[TK]D-Fendernotoriousrab1982: depends on your distro
17:57.19Dan0maN_Workheh.  k.  if it were an RPM based, i would say you would need the kernel-devel package, which supplied the kernel header files.  does ubuntu have the equivilant?
17:57.27notoriousrab1982i have centos
17:57.40Blackthorni have asterisk up and working transfering calls, placing calls to voicepulse etc etc. I've installed a pri card, and i'm trying to get zaptel loaded.. thers no zap in the asterisk menu
17:58.30[TK]D-FenderBlackthorn: After compiling and installing Zaptel you should recompile *
17:58.42jkiffDan0maN_Work: It does.  Ubuntu's development packages are blah-dev.
17:58.54jkiffIn any case, that's a question for #ubuntu, not #asterisk.
17:58.59[TK]D-Fendernotoriousrab1982: "chkconfig asterisk on", "chkconfig zaptel on"
17:59.10*** join/#asterisk troy- (n=troy@remote.firstcoverage.com)
17:59.22notoriousrab1982cheers
17:59.37tzafrir_laptoplinux-headers-`uname -r`
17:59.43troy-My asterisk server uses a 1U chassis, do I have enough room to install a 4-port FXO card (it has a slot for a pci card)
17:59.57[TK]D-Fendernotoriousrab1982: np.  And of course for that "get me running NOW" value = "service zaptel start" , "service asterisk start"
18:00.08*** join/#asterisk Tako-san (n=Tako-san@24.68.129.29)
18:00.35[TK]D-Fendertroy-: Very obviously depends on your chassis & MB
18:00.53[TK]D-Fendertroy-: And the CARD.
18:00.58tzafrir_laptopBlackthorn, ./build_tools/install_prereq test
18:01.03notoriousrab1982cheers, no another note, is anyone doing network monitoring with asterisk, using a server like cacti - is it fair to say that cacti would be a good one to try, or has anyone got any success with other monitoring server programs
18:01.07tzafrir_laptopwhat does it tell you?
18:01.08[TK]D-Fendertroy-: Lets see you try and squeeze a TDM2400 in there ;)
18:01.15CCFL_Man2anyone want to trade a MFT-T1 card for a DVM-T1 card? :P
18:01.45troy-[TK]D-Fender, haha nooh
18:01.56troy-[TK]D-Fender, but the question remans, will i have enough room?
18:02.12*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
18:02.27*** join/#asterisk hyphen (n=user@dsl081-022-037.phl1.dsl.speakeasy.net)
18:02.55*** join/#asterisk ManxPower (n=manxpowe@032-385-751.area5.spcsdns.net)
18:06.36JerJerCCFL_Man2:   I'll trade you a X100P for a MFT-T1 card  :P
18:07.29Blackthornwhat i'm looking at is that there is no chan_zap.so in the /usr/lib/asterisk/modules directory... So Im guessing i need to go through tha make clean;make insall
18:08.00MercestesBlackthorn, haven't you done that yet?
18:08.06JerJerand make sure chan_zap has the necessary deps to be compiled
18:08.16CCFL_Man2JerJer: thats an unfair trade :P
18:08.21styelzcheck make menuconfig
18:08.40CCFL_Man2JerJer: you have incomming T1 trunking?
18:10.46JerJeri haven't been retrofitted with T-1 signaling yet
18:10.58JerJerstill using biological synapses
18:11.09ManxPowerCCFL_Man2: nobody uses T-1 trunking if they have any choice in the matter.
18:11.12ManxPowerThey use PRIs
18:12.08*** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2)
18:12.14CCFL_Man2ahh
18:12.16*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
18:12.30CCFL_Man2this card claims optional PRI
18:13.08CCFL_Man2i got an mc3810 i want to connect to a channel bank
18:13.25CCFL_Man2so i need the DVM-T1 card
18:13.37JerJerebay
18:13.46Blackthornwhen i tried the make menuconfig i get http://pastebin.com/d623cfef2
18:14.02ManxPowerMost channel banks don't support PRI.
18:14.34CCFL_Man2ebay
18:14.47CCFL_Man2this one supports CAS
18:14.50tzafrir_laptopBlackthorn, for asterisk, try the following: wget http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/prereq.sh
18:15.00tzafrir_laptopsh prereq.sh test
18:15.10tzafrir_laptopsee what it recommends you to install...
18:15.23CCFL_Man2which is why i wanted the DVM-T1 card
18:15.25ManxPowerBlackthorn: try su -l instead of sudo
18:15.47tzafrir_laptopif you changed you configuration in any way, you may need to re-run ./configure
18:16.12CCFL_Man2dammit
18:16.22CCFL_Man2$40
18:16.55*** join/#asterisk notoriousrab1982 (n=chatzill@207.47.34.74.static.nextweb.net)
18:17.59styelzBlackthorn, if yu want chan_zap you need to recompile asterisk, not zaptel-module
18:18.20scooby2What would make asterisk randomly not goto voicemail? Working on a new server with 1.4.x but the old one 1.2.x in production randomly just stopped answering this week.
18:20.18[TK]D-Fenderscooby2: Nothing.  its your dialplan and you haven't shown us anything.
18:21.24scooby2i was looking for confirmation on the nothing. Dialplan has not changed in over a year.
18:22.51[TK]D-Fenderscooby2: Show us something and we'll be able to comment.
18:23.54Blackthornin 1.4.4 asterisk would you get a "zap" option on the menu if asterisk regonized that the digium t1 card was in green status?
18:24.51[TK]D-FenderBlackthorn: Makes sense.... what are you doing to see this status?
18:26.16Blackthornzttools shows the t1 up and running green status.. But Asterisk dosn't see it. When I check the modules directly there is no chan_zap.so (sp?) file. So theres no zap channel moduel to load
18:26.37Blackthornso i felt that zaptel needed to be complied.
18:26.39Blackthornand installed
18:26.54[TK]D-FenderBlackthorn:  you need to recompile * after installing zaptel
18:27.15[TK]D-FenderBlackthorn: So taht chan_zap gets built
18:27.20styelzZapata Telephony: Depends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E), tonezone(E)
18:27.40Blackthornumm..
18:27.47styelzyer
18:27.48Blackthornok i understand what your saying.
18:28.16BlackthornBUT, zap isn't installed as far as I can tell and when I try to make install it dosn't
18:28.46*** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net)
18:28.46[TK]D-FenderBlackthorn: failing on "make install" is clearly a BAD sign.
18:29.58Blackthornwhen i do a make install it basicly goes down to "autoconf: no input file"
18:30.04Blackthornthen errors out
18:31.25[TK]D-FenderBlackthorn: try it as "sudo su -"
18:32.21Qwell[]Blackthorn: What distro are you installing on?
18:32.33[TK]D-FenderQwell[]: Ubuntu I believe.
18:32.36styelzif your kernel header is old.. you may need to symlink config.h to autoconf.h
18:32.45styelzor te other way around
18:33.42*** part/#asterisk lib (i=private@mongoose.hypa.net)
18:34.10Blackthornit's ubentuo 6.x and the headers been updated.. i remember doing that
18:34.12Mercestes<PROTECTED>
18:34.19styelzok
18:34.35Qwell[]ubentuo?
18:34.41Blackthornasterisk works fine :) i't up and running this moment with calls
18:34.42Qwell[]6. what?
18:35.37Blackthorn6.06 lst
18:35.44styelztyr ./build_tools/install_prereq install
18:35.45BlackthornLTS
18:35.53putnopvutAKA Dapper Drake
18:35.56styelzform zaptel dir
18:36.06tzafrir_laptopBlackthorn, what's the output for: ls -l /lib/modules/`uname -r`/.config
18:37.26tzafrir_laptopsorry:
18:37.37tzafrir_laptopls -l /lib/modules/`uname -r`/build/.config
18:38.27Blackthorni keep getting an eeror that there is no such directory
18:38.36Blackthornhowever if i go to /lib/modules
18:38.48tzafrir_laptopBlackthorn, where do you get it from?
18:39.36tzafrir_laptopthat command returns "no such file or directory"?
18:39.49*** join/#asterisk masterisk (n=mascool@70.88.122.206)
18:40.13Blackthornls: /lib/modules/2.6.15-26-amd64-server/.config: No such file or directory
18:40.27styelzyou missed the mistake
18:40.42styelz6 lines up
18:40.50tzafrir_laptopright. So you have no linux-headers for your linux-image installed
18:40.58styelzin build
18:41.21styelzls -l /lib/modules/2.6.15-26-amd64-server/build/.config
18:41.28tzafrir_laptopapt-get install linux-headers-`uname -r`
18:41.34Blackthorn-rw-r--r-- 1 root root 62884 2006-08-02 23:29 /lib/modules/2.6.15-26-amd64-server/build/.config
18:41.35styelz....
18:41.38styelzbetter
18:41.48tzafrir_laptopor the script styelz told you about: build_tools/install_prereq
18:42.14styelzthat script worked for me.. im on Ubuntu 7.10 though
18:42.15tzafrir_laptopah, ok
18:42.16styelzheh
18:43.12tzafrir_laptoplet me know how this was solved, I'm off to bed right now
18:43.34styelzgee its only 4:30 am
18:43.42styelzshit
18:43.47styelzer  excuse me
18:44.21pifiuHas anyone used tmobile @home hotspot service with asterisk?
18:44.33anonymouz666tzafrir: that issue I told to you about chan_zap and wctdm was when the fleed patch was applied.
18:45.03anonymouz666tzafrir: when it fakes the polarity reversal
18:45.24anonymouz666I can't understand why chan_zap can't take it and it is generating that core I paste to you
18:45.29*** join/#asterisk Op3r (n=Op3r@121.97.177.105)
18:45.53CCFL_Man2pifiu: no, but i hear you need to do nat
18:46.07CCFL_Man2well, maybe not
18:46.25pifiuive never used it
18:46.39pifiubut wanted to pickup one of the phones since tehy are cheap and i do have tmobile
18:46.42pifiuit would be great
18:46.44*** join/#asterisk jmesquita (n=jmesquit@201.7.117.114)
18:52.05Blackthornthanks for all your help, didn't get it resolved but i'm done for the day. gota get some other work done
18:52.45*** join/#asterisk GESO (n=ssssssss@acfg65.neoplus.adsl.tpnet.pl)
18:57.29DavieyHmm.. anybody else notice that setting outbound callerid doesn't seem to work in asterisk 1.4?
18:58.00Daviey-- but does work for extensions set in users.conf?
18:58.13[TK]D-FenderDaviey: Sure it doesn, show us your failed attempt
18:58.33Davieywell i can show you my failing dialplan if that helps?
18:58.39Davieythe console won't show much..
19:00.51[TK]D-FenderDaviey: It would is you were at verbose 10 with channel debug and doing NoOps....
19:01.38Davieyhttp://pastebin.com/m7be91891
19:02.00DavieyI'm doing noops, but not chan debug.. let me do that
19:03.45[TK]D-FenderDaviey: And that pastebin is worthless.
19:05.28[TK]D-FenderDavI'll want to see the CLI output of the failed call attempt.
19:05.37*** part/#asterisk olrrai (n=olrrai@host119.190-137-116.telecom.net.ar)
19:06.19[TK]D-FenderDaviey: You also have CLASHING EXTENS between included contexts./
19:07.31[TK]D-FenderDaviey: And it'd be nice to see your SIP peer setup as well.
19:08.37Davieyshould that matter?  I'm litterally bouncing a call in and out on a provider
19:10.01[TK]D-FenderDaviey: Yes, you can screw up your peer so that IT overrides everything
19:10.29Davieyah
19:12.00Davieyhttp://pastebin.com/d1c2b07e3
19:14.40[TK]D-FenderDaviey: this looks like a SIP account, where you showed me an IAX one.
19:14.58Davieysorry, call comes in via sip - and goes out via iax
19:15.37[TK]D-FenderDaviey:  I obviously need to see what you are dialing OUT ON <----------
19:15.58hmmhesayswell I have another ep2500 on the way [TK]D-Fender
19:16.04hmmhesaysmy PA system is going to rock
19:16.33[TK]D-Fenderhmmhesays: Behringer power amps?
19:16.46hmmhesays[TK]D-Fender yeah
19:16.50hmmhesayscheap and they get great reviews
19:17.01Davieyhttp://pastebin.com/m3c97e40b
19:17.07[TK]D-Fenderhmmhesays: Yeah, but are YOU any good? :)
19:17.15hmmhesayshaha I can hold my own
19:17.25[TK]D-Fenderhmmhesays: Link me :)
19:17.34hmmhesaysI don't have any mp3's online at the moment
19:17.39hmmhesaysthe band is still looking for a vocalist
19:17.43[TK]D-Fender:O
19:17.48Daviey[TK]D-Fender: I can't see that being the problem as callerid is set properly when called from an extension, just not on my dialplan i initially pastebinned
19:18.45[TK]D-FenderDaviey: You see it being set in the CLI output.  If that is right then either thr PROVIDER is rejecting it (they say they are wide open) or your peer is wrong!
19:18.53[TK]D-FenderDaviey: Stop thinking, start SHOWING.
19:19.32*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
19:20.00Davieywhat do you want to see?
19:20.54[TK]D-FenderDaviey: *** Your peer setup ***
19:21.40Davieywhere is that?
19:22.04*** join/#asterisk zcionn_ (n=a@58.69.243.203)
19:22.12[TK]D-FenderDaviey: ...................
19:22.15[TK]D-FenderDaviey: ...................
19:22.36[TK]D-FenderDaviey: Do you have a CLUE where you setup Voipjet in there at all?
19:22.47Davieyyeah.. users.conf
19:23.18[TK]D-FenderDaviey: Well show me his entry!
19:23.42Davieyi did!
19:24.27Daviey20:17:00 < Daviey> http://pastebin.com/m3c97e40b
19:24.29Daviey20:17:00 < Daviey> http://pastebin.com/m3c97e40b
19:24.37*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
19:29.50*** join/#asterisk mog (i=mog@nat/digium/x-fb2d7ef63e002dc5)
19:29.50*** mode/#asterisk [+o mog] by ChanServ
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19:30.38Daviey[TK]D-Fender: ?
19:31.03*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
19:31.58[TK]D-FenderDaviey: not sure (I HATE users.conf)
19:32.30Daviey:) Ok, thanks anyway
19:33.12timtimi must be thick or something
19:33.23timtimno matter what I try, all my calls end up as 404
19:33.39wackerDoes anyone know what's up with ftp.digium.com?  There doesn't seem to be an FTP server running on that machine.
19:33.39Corydon76-digtimtim: sounds like wrong context setting
19:33.46[TK]D-Fendertimtim: pastebin your call attempt at verbose 10 & SIP debug (assuming SIP) enabled.
19:33.57[TK]D-Fender~pb
19:33.58jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:34.21Corydon76-digI don't know why we bother posting pastebin.com.  It is dog slow
19:36.09*** part/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
19:36.59[TK]D-FenderCorydon76-dig: No, its VERY fast and has been for several noths.
19:37.03[TK]D-Fendermonths*
19:37.24Corydon76-digNot compared to pastebin.ca
19:37.24*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
19:37.33[TK]D-FenderCorydon76-dig: True before it was garbage, but after they renovated all is good.
19:37.38timtimhttp://pastebin.com/d5b22c95e
19:37.40[TK]D-FenderCorydon76-dig: Is now...
19:37.59[TK]D-Fendertimtim: SIP/2.0 401 Unauthorized <- bas SIP auth
19:38.04[TK]D-Fenderbad*
19:38.14timtimis that cisco1?
19:38.27[TK]D-Fendertimtim: Looking for 1005 in everything (domain 192.168.1.200) <-   go check this context.
19:38.41[TK]D-Fendertimtim: and if you still can't see it, PASTEBIN it.
19:39.09timtimexten => 1005,1,Dial(SIP/cisco2)
19:39.09timtimexten => 1005,2,Hangup
19:39.26timtimcisco1 I know about, its I haven't got around to tftping it the new config, cisco2 should work though
19:39.38[TK]D-FenderCorydon76-dig: Even faster than .ca here.
19:39.46[TK]D-Fendertimtim: pastebin the whole context.
19:40.17timtimhttp://pastebin.com/d5b009ebd
19:41.13[TK]D-Fendertimtim: pastebin "show dialplan"
19:41.44*** join/#asterisk sysreq` (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
19:41.47timtim-= 0 extensions (0 priorities) in 0 contexts. =-
19:42.05timtimi'm no expert, but that looks a litte wrong to me
19:42.20[TK]D-Fendertimtim: indeed it was.  go verify what FILE you're looking at.
19:42.57[TK]D-Fendertimtim: Feel free to paste the "ls" of it.
19:43.24timtimits in the right place
19:43.54[TK]D-Fendertimtim: please prove its condition.
19:44.10[TK]D-Fendertimtim: "ls -l /path/to/file/
19:45.31timtim-rwxr--r-- 1 root root 315 Aug 17 21:41 /opt/etc/asterisk/extensions.conf
19:45.52[TK]D-Fenderopt?!
19:46.00timtimits a package for my nslu2
19:46.22[TK]D-Fendertimtim: do a "reload" and see if it can find it.
19:46.41[TK]D-Fendertimtim: and verify asterisk.conf's for the path expected.
19:47.11timtimthere is nothing in the console about not finding it
19:47.31timtimand I'll double check asterisk.conf
19:47.38[TK]D-Fendertimtim: PASTEBIN, ALSO YOUR ASTERISK.CONF PLEASE
19:48.34[TK]D-Fendersevard: I work in all-caps, was actually yelling...
19:48.45timtimhttp://pastebin.com/d1675b06b
19:49.00sevardAll day I just watch #asterisk, I can actually _feel_ the vein pulsating in your forehead over IRC.
19:49.19[TK]D-Fendersevard: You shoulda seen the LAST guy....
19:49.20*** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu)
19:49.33b11dhey all
19:49.52[TK]D-Fendertimtim: and running * as root?
19:50.00[TK]D-Fendertimtim: do a reload and PB it.
19:51.52*** join/#asterisk Ebola (n=Ebola@host86-138-7-229.range86-138.btcentralplus.com)
19:52.06timtimPB?
19:52.30timtimyeah, i'm running it as * until I can get everything set up nicely
19:52.36timtimroot even
19:52.50timtimah makes sense
19:53.16b11dpb = pastebin ?
19:53.34timtimrealised that heh
19:54.03sevard=hey b11d
19:54.18timtimhttp://pastebin.com/d5bab64d4
19:55.50*** join/#asterisk ToyMan (n=Stuart@pool-71-169-18-129.pghk.east.verizon.net)
19:56.10[TK]D-Fendertimtim: not just "module" , "dialplan reload" or "reload"
19:56.29b11dhey sevard
19:56.32[TK]D-Fendertimtim: res_features.c:2220 load_config: Could not load features.conf <- not a great sign either
19:56.49[TK]D-Fendertimtim: I'd doubt the permissions on your folder...
19:56.58[TK]D-Fendertimtim: thats a lot of bad news there
19:57.44Trevor_banyone else having issues with Teliax today?
19:57.47timtimthe etc folder is pretty bare
19:57.58timtimsip iax extensions and asterisk are the only files in there
19:59.07timtimthis intergrated box only has about 32M of ram so I cut everything to a bare minimum
20:05.18[TK]D-Fendertimtim: can you please pastebin the full output of "reload"
20:09.04timtimits the same as the last pb
20:09.56*** join/#asterisk Tako-san (n=Tako-san@24.68.129.29)
20:11.39kkn088ET merc avi
20:11.44[TK]D-FenderIT SHOWS NO PARSING OF any CONFIGS... NOT GOOD
20:11.47kkn088oops sorry
20:12.05timtimhmm
20:12.12[TK]D-Fendertimtim: and you did "module reload" which is not what I asked and I can't confirm the nature of.
20:12.33kkn088normaly I use my cmd qmsg for all quakenet chan
20:12.36timtimit says reload is going to be depreicated and you should use module reload instead
20:12.48DavieyIdeally mythtv backend aswell <grin>
20:12.51Daviey1.4 vs 1.2 eh
20:13.06Daviey(ignore the mythtv line, dang irssi)
20:13.13timtimi've turned up verbosity to 99 and i see it parsing sip and iax.conf
20:13.53[TK]D-Fendertimtim: anything about extensions.conf?
20:14.23timtimno
20:14.38timtimwhich makes me think, is there any specific module to be loaded that parses extensions.conf?
20:15.14[TK]D-Fendertimtim: yes
20:15.14[TK]D-Fender1 sec
20:15.14timtimi've got bitten in the ass before by missing modules.
20:15.14[TK]D-Fenderpbx_config.so
20:15.15timtimthat could explain a lot
20:15.19[TK]D-Fendertimtim: specifically try to reload it
20:16.23timtimyeah it worked that time
20:16.27*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
20:16.38timtimi can see the dialplan in the cli now
20:18.09*** join/#asterisk unspin (n=unspin@24.82.161.85)
20:19.23unspinusing res_mysql_realtime, is it possible to set a "language=es" value in the database for a specific user account and have asterisk reflect that change during a voicemail session?
20:19.27[TK]D-Fendertimtim: make sure you are loading your modules on start...
20:20.02[TK]D-Fenderunspin: If you're referring to a devices config in a table, sure.
20:20.03timtimwill do, thanks for everything fender
20:20.03RyushinWhat voip providers do people recommend in the US?  This is for business use.
20:20.10unspinthats what i thought also
20:20.30unspini've got that exact string set in the database but when i enter voicemail main
20:20.40unspinthe language is set to en
20:21.37Op3rRyushin: commpartners
20:21.38[TK]D-Fenderunspin: Go pastebin some backup.
20:22.15[TK]D-FenderRyushin: teliax seems to be relatively rell respected around here.
20:22.37RyushinThanks [TK]D-Fender.  I'll check them out.
20:25.50*** join/#asterisk fakhir (i=Fakhir@unaffiliated/fakhir)
20:27.57*** part/#asterisk lirakis (n=etamme@65.200.191.253)
20:29.14flujanrussellb: are you online?
20:29.19russellbno
20:29.37flujan:P
20:29.38*** join/#asterisk hijacked (i=RcR2@66.255.220.17)
20:29.43flujanhere http://pastie.caboo.se/88726
20:29.55Yarqdoes anyone here use Betamax ?
20:30.17unspinFender, http://pastebin.org/717
20:30.26russellbflujan: i'm not digium support :)
20:30.26flujanthis problems could be caused by a bad board?
20:30.32russellbit's possible, yes
20:30.39flujanrussellb: I now but...
20:30.40flujanok
20:30.42flujanthanks
20:33.25CCFL_Man2anyone here have the latest cisco mc3810 firmware? :P
20:33.36*** join/#asterisk ToTo (n=toto@host110-162-dynamic.0-87-r.retail.telecomitalia.it)
20:33.39ToTohi all
20:34.35unspinFender, i'm not explicitly setting language anywhere else
20:35.30blitzrageCCFL_Man2: you shouldn't really ask for stuff that is illegal to distribute...
20:36.02ToToi'm buying asterisk cookbook, is this a good book?
20:36.12Sweepermeh
20:36.17blitzrageToTo: it's not done yet
20:36.42ToToblitzrage: i'm buying it on www.play.com
20:36.50YarqI need a voip service which would let me set different CID's for every extension
20:36.55Qwell[]~thebook
20:36.56jboti guess thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
20:36.56Yarqany suggestions ?
20:36.57blitzrageToTo: ya... I'm one of the authors -- it's not out
20:36.57Qwell[]ToTo: buy that
20:37.05blitzrageactually... buy the 2nd edition :)
20:37.30Qwell[]blitzrage: when does it hit shelves?
20:37.38CCFL_Man2blitzrage: well, true, though i doubt cisco will give me discount on cco on something thats EOL and for home use :P
20:37.42Sweeperdon't buy TFoT, you'll read it once, and then it'll sit on your shelf :P
20:37.48blitzrageCCFL_Man2: still... :)
20:37.54ToToblitzrage: so, what book i have buy?
20:37.55Qwell[]Sweeper: You should reference it every time you have a problem...
20:38.09ToToblitzrage: the first ed?
20:38.12SweeperQwell[]: it sucks horribly as reference material :P
20:38.19Qwell[]no, it doesn't
20:38.29Sweepergoogle is about 100 times better
20:38.51blitzrageQwell[]: it's ok -- you can't please everyone. We're talking about telephony here.
20:38.54Sweeperand TFoT is largely evangelism :P
20:39.03blitzrageSweeper: thanks :)
20:39.14CCFL_Man2blitzrage: not sure how to use it yet though, this will be my first ever voice configuration
20:39.53blitzrageToTo: Asterisk: The Future of Telephony, 2nd Edition is the book you can look at. It went to press on Wednesday.
20:40.00blitzragedepends how much of a rush you're in
20:40.11blitzrageif you need something now -- you can read the online (free) version of the 1st edition.
20:40.12rudholmSweeper: when I was new to Asterisk, I found the "rah rah!" aspect of TFoT to be off-putting.
20:40.38blitzrageinteresting how enthusiasm can be off-putting :)
20:41.02blitzrageI guess the book should have been much more boring to read
20:41.15rudholmit was a bit much
20:41.16Sweeperrudholm: it was all good and well, after a couple weeks of having a nortel shit all over the place, and I read the whole thing
20:41.42Sweeperbut the binder has stayed on my shelf ever since :P
20:41.55rudholmif the authors demonstrate such a strong emotional bias, it makes their conclusions as to suitability for a given purpose suspect.
20:42.07Sweeperit's not as bad, say, as the O'Reilly "Hacks" series
20:42.30rudholmanother problem, now, though no fault of the authors is that it's out of date.
20:42.44*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)
20:42.44blitzragerudholm: 2nd edition went to print on Wednesday, which covers 1.4
20:42.46rudholmbut I don't think it's a *bad* primer
20:42.50rudholmcool
20:42.55rudholmI might pick up a copy
20:43.03blitzragedisclaimer: I'm one of the authors :)
20:43.03Sweeperblitzrage: is it also CC?
20:43.07blitzrageSweeper: of course
20:43.09rudholmI figured
20:43.18rudholmdid you fix chapter 7?
20:43.21Qwell[]blitzrage: Did I make it in the book? :p
20:43.23Sweepercool
20:43.33blitzrageQwell[]: yep :)  shoot... I forgot to change newb to nub
20:43.37rudholmsome of the digital stuff could use a bit of polish, as I think I've mentioned here before
20:43.38Qwell[]w00t :D
20:44.13blitzragerudholm: I don't deal with traditional telephony -- I've not seen any email discussing it
20:44.13jcolpQwell[] is an Asterisk allstar
20:44.17blitzragehe sure is!
20:44.26blitzragejcolp is mentioned...
20:44.27jcolpQwell is not
20:44.28Qwell[]and not a total nub
20:44.31Qwell[]or, newb, as it were
20:44.37rudholmthat business with digital audio output consisting of curves with flat facets wasn't quite right
20:45.05Sweeperwas it TFoT or the cookbook that was gonna have an adhearsion section?
20:45.29blitzragerudholm: there is always http://www.oreilly.com/catalog/asterisk/errata/
20:45.37blitzragerudholm: we can fix what we're not told is broken :)
20:45.41blitzrages/can/can't
20:46.14rudholmyeah, I just mentioned it here, didn't formally mention it
20:46.30blitzrageya.. believe it or not I don't read this channel 24hrs a day
20:46.39rudholmme neither :)
20:47.14rudholmsomeone, I think it might have been TK-Dfender got a bit defensive about it, so I dropped it
20:47.26rudholmwasn't in the mood to argue pointlessly
20:47.47rudholmI find I'm less often in that mood as I get older :)
20:48.13blitzragethat's ok... I don't make enough money on it to care that much :)
20:48.19Sweeperhttp://youtube.com/watch?v=Tx1XIm6q4r4 <-- win
20:48.28rudholmyeah, I know one doesn't get rich writing such books
20:48.31blitzragebut if people have an issue with something, then we would "fix" it... but if we don't hear about it, then we can only assume everything is perfect
20:48.40blitzrageI think I made $5k over the last 2 years
20:48.47rudholmwas talking to Randal Schwartz about that just yesterday (he wrote "Learning PERL")
20:48.59blitzrageand it sells well
20:49.03rudholmyes, it does
20:49.18blitzragebasically I would have made more money working at McDonalds for the same amount of hours I put into it :)
20:49.25rudholmbut given that it took two years to write, I think he said he'd make more money per hour flipping burgers :)
20:49.48blitzrage;)
20:49.52rudholmhaha, I guess you guys came to the same conclusion :)
20:49.57blitzrageindeed
20:50.12Sweepernow those Agile book writers, they probably make a lot of money. see, since they're so agile they can write books faster! :P
20:50.14blitzrageif you were doing it full time, you could make good money, because you'd have several books out at once
20:50.24rudholmyeah
20:50.31blitzragebut that's a LOT of initial work
20:50.34*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:50.42rudholmhe's got Learning PERL and Intermediate PERL, etc
20:50.43blitzrageI make a lot more money doing Asterisk consulting
20:50.57rudholminstalls and stuff?
20:51.23rudholmwhere abouts in the world are you?
20:51.41blitzragerudholm: I built a clustered, virtual PBX system for an ITSP
20:51.45blitzrageI live in downtown Toronto
20:52.01blitzrageI write mad dialplan :)
20:52.07rudholmcool
20:52.17rudholmmy company is planning to move to *
20:52.20jcolphe has the mad dialplan skillz
20:52.23rudholmit's going to be a huge migration
20:52.37blitzrageand I tend to find a bunch of bugs that I make jcolp and russellb fix :)
20:52.40blitzrageoh, and putnopvut now :)
20:52.43rudholmheh
20:52.44blitzrageand of course Corydon76-dig
20:52.46russellbo.O
20:52.53blitzragethey're my base :D
20:53.01putnopvutAlways a pleasure.
20:53.02putnopvut:)
20:53.07blitzrageyou guys are my hero's :)
20:53.09russellbchan_iax2 is my bi0tch
20:53.22blitzrageand Qwell[] should be scared if I ever start to use skinny :)
20:53.31rudholmI'm trying to convince our telecom guys to go with Polycom 550s
20:53.38blitzrageI love the polycom phones
20:53.41rudholmme too!
20:53.47blitzragewouldn't deploy anything else
20:53.48rudholmI just got one for my house and I love it
20:53.56jcolpblitzrage: <3
20:54.10rudholmthe sound quality is excellent, the config is easy...
20:54.18rudholmjust a great phone
20:54.32blitzrageI have a 301, 501, Cisco 7912, 7960, and 7970, Linksys SPA-942, and Mitel 5220
20:54.34Mercestesrudholm, the config is not particularly easy.
20:54.43Mercestesrudholm, it is a great phone though.
20:54.45blitzragethe Mitel works awesome, but the handset is too small
20:54.49rudholmI'm trying to get Strom to get one so we can do G.722 calls between each other.
20:55.06blitzragethe Polycom config is simple if you've done it before
20:55.12rudholmmaybe my config was easy because it was simple and I haven't gotten into the advanced features yet
20:55.37blitzragerudholm: ditto -- I just wanted the phone to work. The most complex thing I really changed was turning off the MWI audio tone, and the dialplan of the phone
20:56.11*** join/#asterisk high-rez (n=gus@carrera.bourg.net)
20:56.29high-rezCan someone recommend a good IAX provider for calling to europe?
20:56.36blitzragefor anyone who wants to propose interesting projects that I should write about in the cookbook, feel free to email me at leif.madsen@gmail.com with your ideas.
20:56.38rudholmI like teliax.com
20:56.52Mercestesblitzrage, exactly
20:56.54blitzrage</announcement> :)
20:56.57*** join/#asterisk nirz (n=nirz@bzq-79-179-88-51.red.bezeqint.net)
20:57.02rudholmoh, you're Leif Madsen :)
20:57.03high-rezrudholm: their call quality to some of .eu is flaky :|
20:57.14[TK]D-Fenderrudholm, IP 550 = overprices waste
20:57.19blitzragerudholm: no.. I just wanted to barrage him with emails :D
20:57.23rudholmI just called Sweden and it was fine, but that's not a lot of data points I guess.
20:58.01high-rezrudholm: yeah.  they're good for germany too, it seems.  but if you go into eastern europe, i think it all goes over some other provider not the PSTN that is really poor quality. (lots of drops, bad jitter etc)
20:58.09*** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil)
20:58.21high-rez(i have several accounts with them though for various customers for US, which they're great at)
20:58.36rudholmhigh-rez: are you sure it's not just eastern europe?  they don't seem to have a very good PSTN in general
20:58.56high-rezrudholm: Yeah, if I call to the same #'s from my cell phone, or even from voipgate, quality is perfect.
20:59.05rudholmhigh-rez: you could try voipjet, but they're likely to be even worse.
20:59.45rudholmhigh-rez: you might try asking them about it, they seem pretty interested in making customers happy.  they have a web forum.
21:00.28*** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net)
21:02.48*** join/#asterisk asterisknerds (n=asterisk@66.7.122.93)
21:03.18asterisknerds<PROTECTED>
21:05.37*** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net)
21:06.55*** join/#asterisk mihinomenest (i=czHt@66.255.220.17)
21:06.56unspin[TK]D-Fender, did you get a chance to have a look at my pastebin details?
21:07.15[TK]D-Fenderunspin, I missed it when I left for home
21:07.21[TK]D-Fenderunspin, pastbin it again
21:07.45unspin[TK]D-Fender, http://pastebin.org/717
21:07.53CoffeeKidanyone have any luck using AGI Manager's PlayDTMF function to execute something from features.conf?
21:08.31unspinI get the impression that the voicemail module expects the language to be set beforehand
21:08.39unspinsetting the timezone through the database works as expecetd
21:08.46[TK]D-Fenderunspin, I don't recall language being an optiong in voicemail.conf.  It takes that from the current CHANNEL
21:09.27unspini was hoping it would slurp up the config option during authentication
21:09.43unspinsince the call is coming in from a different (implicitly trusted) source
21:10.21unspinits available during authentication, maybe i'll make a patch
21:10.32[TK]D-Fenderunspin, well it sure doesn't come from voicemail.conf
21:10.50[TK]D-Fenderunspin, you need to set it in either the inbound device, or in the dialplan.
21:11.12unspinI'm using Asterisk as a central voicemail server, so the client isn't uniquely identified until they are authenticated by voicemailmain
21:11.48unspinbut i guess voicemailmain doesn't parse the language option and change the local value during authencation
21:11.58Davieyunspin: maybe add authenticate() in the dialplan then call voicemailmain(s,exten) ?
21:12.38Davieymaybe you can add a labguage setting in the middle?
21:14.26unspinin my current scenario the identity of the user has been verified by a separate system, so i dont think i can use authenticate()
21:15.06[TK]D-Fenderunspin, thats not its job.  What you should do is use VMAuthenticate, then look them up in a language table, the go into voicemailmain w/o asking for a pass.  it'll look transparent to the user
21:15.42[TK]D-FenderVmauthenticate <--------------
21:16.18unspinok, i'll check that out
21:16.22unspinthanks
21:17.40flujanguys, based on your knowledge of asterisk
21:17.41flujanhttp://pastie.caboo.se/88746
21:17.53*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
21:18.19unspinthe only issue is that the initial password prompt wont have the language set
21:18.35flujanit show a hardware problem on my Digium card? I try to change the cables to span 1 to 3 3 to 2 2 to 4 an the problem still ocurr... any ideas?
21:19.10flujanso I think that it is not a problem with my telco.
21:19.26unspinperhaps i'll need to forward a language flag along with the RURI from the remote system
21:19.28Deeewayneflujan: was your configuration ever working ?
21:19.36[TK]D-Fenderunspin, Sorry... * can't be PSYCHIC :)  So just set a default and live with it ;)
21:19.44flujanDeeewayne: yeap
21:19.52unspinsorry fender, i'm just thinking outloud
21:19.56*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
21:20.00unspini'm very thankful for you assistance =)
21:20.13[TK]D-Fenderunspin, np
21:20.16Deeewayneand it just stopped working without any changes on your side ?
21:20.24Mercestesunspin, I would say it would be up to your remote system to pass the language information to asterisk since it is identifying them.
21:20.43*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
21:20.45unspinMercestes, thats what i'm thinking now as well
21:21.00unspineven if i patched voicemail main (when passed as extension) to set the language variable
21:21.35unspini would have a problem for people dialing the voicemail number without a specified extension
21:21.48*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
21:21.53unspinthanks for your time everyone
21:22.20[TK]D-Fenderunspin, You can always mod the number that * receieve to include it encoded
21:22.25Mercestesunspin, just setup a prompt menu for it.  Press 1 for english, 2 for spainish, 3 for klingon, 4 for raging lebian teen cheerleader, 4 for german.
21:22.38Mercestess/4 for german/5 for german/
21:22.41Qwell[]Mercestes: Did you intentionally reuse 4? :P
21:22.58MercestesQwell[]:  no, I in no way suggest that germans are the best candidates for raging lesbian teen cheerleaders. =/
21:23.10Mercestesmostly because a:  they dont' shave their legs and B:  they're all bigger than me.
21:23.11*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
21:23.12[TK]D-FenderMercestes, "raging teen lesbian" is very getteral and may too easily be interpreted AS German ;)
21:23.25[TK]D-Fendergutteral*
21:23.30Mercestes[TK]D-Fender, very true
21:26.04flujanDeeewayne: yeap I changed nothing...
21:26.21unspinMercestes, lol!
21:26.31flujanDeeewayne: to be honest, i update asterisk to 1.4.10.1
21:26.40flujanwas using the 1.4.5
21:26.58flujando you believe that this could be the cause of such problems?
21:27.36*** join/#asterisk Dreamless (n=dreamles@port63.ds1-arsy.adsl.cybercity.dk)
21:28.33Deeewayneyou never know...bugs do slip into releases :-)
21:28.59Deeewayneafter updating to 1.4.10.1, how long did it work until you starting seeing these problems ?
21:30.22Qwell[]Deeewayne: what?  never
21:30.23flujandunno maybe two or three hours... or I just detected the issue after the upgrade...
21:30.26Qwell[]Asterisk has no bugs :p
21:30.54flujanI read the changelog and see no problem in chan_zap.c
21:30.57flujananyway
21:31.07flujando you think I should downgrade?
21:31.12Qwell[]flujan: worth a shot
21:32.11Deeewaynewe did scrub 1.4.10.1 with the old bugBeGone, but maybe we missed something
21:32.41flujanok
21:32.43Deeewayneflujan: I would suspect software more than a hardware issue
21:32.43flujanwill try it
21:32.56Deeewayneits a great data point to go back and verify that the issue goes away
21:33.56CCFL_Man2i'm not sure if i can use this MFT T1 card to connect to my channel bank
21:34.24CCFL_Man2is a TDM cross connect basically like connecting to a channel bank?
21:35.21*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
21:36.46*** join/#asterisk _omer (i=Stephnie@u15157627.onlinehome-server.com)
21:37.46_omerhello
21:38.10_omerIs there any website to buy routes instantly......???
21:38.32b11doh hell yeah
21:38.36nDuffCCFL_Man2: TDMV is pretty much the standard way of connecting to a channel bank, yes.
21:38.49b11dwww.instaRoute.com is my favorite
21:38.51_omerb11d??
21:38.53hmmhesaysbuy routes?
21:38.55b11d:)
21:39.00b11dyeah im pretty much kidding..
21:39.07b11di mean, maybe instaroute.com works.. dunno
21:39.08b11d:)
21:39.09_omerreally :)
21:39.11b11dhaha
21:39.12hmmhesaysb11d liar!
21:39.15_omerhehe
21:39.17b11das usual :)
21:39.47CCFL_Man2nDuff: i can't figure out if i should just use with MFT-T1 card or get a DVM-T1 card for this mc3810
21:40.25Mercestesb11d!!!!
21:40.32b11dMercestes!!
21:40.33_omersomething like www.buysellminutes.com    (second Priority)
21:40.37b11dhow are you?
21:40.51Mercestesmeh
21:40.53Mercesteshow are you?
21:40.54CCFL_Man2nDuff: so i can connect the mc3810 to this channel bank
21:40.56b11dbtw, you still have the sweetest name in the world..
21:40.58Mercestesgot ya a girl yet?
21:41.06MercestesAwww..thanks.  :D
21:41.07hmmhesaysI had one
21:41.08b11dno not yet :)
21:41.09hmmhesaysshe left me
21:41.12hmmhesaysthen I got her back
21:41.12Mercestesany boys then?
21:41.15b11dnothing permanent anyways
21:41.18hmmhesaysand she let me do naughty things to her
21:41.20hmmhesaysand that was fun
21:41.24b11dno, no boys either ;)
21:41.29Mercestesaw.
21:41.30MercestesI'm sorry
21:41.36b11dim not :)
21:41.38_omerhey you guys are really doing V-O-I-P with her ;)
21:41.40Mercesteshmmhesays, define naughty
21:41.47hmmhesaysI subscribed to guitar world today
21:41.47b11dman.. school is starting up here next week.. NICE crop this year :)
21:41.59Mercestesb11d, ...Way to be enterprising!  :D
21:42.04b11dso..  I need all hotties to show up in my office for
21:42.06b11d"email training"
21:42.08b11dor some thing
21:42.09Mercestesb11d, isn't that a grade school near you tho?
21:42.14b11dHAHA
21:42.14b11dno!
21:42.16Mercesteslol
21:42.17b11dIt's a college damnit :)
21:42.20Mercesteshehehe
21:42.22MercestesUh huh.
21:42.22*** join/#asterisk Tako-san (n=Tako-san@24.68.129.29)
21:42.28b11dmy old boss would have been at the elementary school though
21:42.42b11dhe just got out of a federal penn for that very thing
21:42.45b11ddid 7 years
21:43.01Mercestesb11d, heh, must not've been very good then.
21:43.26MercestesNever done illegal...
21:43.28b11di guess not... i dont want to find out, personally. :)
21:43.40b11di've doen things in other countries that is legal, that isnt here. :)
21:43.46Mercestes....made out with my current wife on her 18th birthday tho....
21:43.46hmmhesaysMercestes: well I got it explore her throroughly
21:43.48b11dspeaking of wich, im heading to Amsterdam here next week
21:43.50Mercestesthat's close neough
21:43.53MercestesI guess.
21:44.01MercestesAmsterdam??  Aww..I wanna go. =/
21:44.05Tako-sanAre there major differences between ATA devices?  Any recommendation on makes/models?
21:44.06b11dcome with :P
21:44.10MercestesI want me some legal 14 year old action
21:44.18b11dTako-san.. do NOT get a Cisco VG-224 or anyhthing.. they lick.
21:44.22CoffeeKidAnyone ever had any luck using Manager API's PlayDTMF function?
21:44.27CCFL_Man21 Multiflex T1(slot 3) RJ45 interface(v01.K0)
21:44.30Tako-sannoted
21:44.31Tako-santhanks
21:44.38b11dnp.. i've got two of the bastards right now :)
21:44.54b11dMercestes.. dunno where you could get that..
21:44.55Tako-sanAny other comments about ATA devices?
21:44.58b11dmaybe Thailand or some shit :)
21:44.59CCFL_Man2anyone use the cisco mc3810?
21:45.04Mercestesb11d, yea, I was just thinking, thailand.
21:45.06nDuffCCFL_Man2: No clue; I don't generally mess with Cisco.
21:45.15b11dhey CCFL_Man2..  the #cisco guys are really good with that stuff..
21:45.15Mercestes...of course, they all look 14 on up to age 30 anyways so I would never know the difference
21:45.15CCFL_Man2ahh, ok
21:45.19b11dif you have questions about it
21:45.24b11dthey totally helped me with my vg-224
21:45.32CCFL_Man2ahh, thanks
21:45.34b11dnp
21:45.51b11dMercestes.. i think thats the wisest thing i've heard all month
21:46.04b11d<PROTECTED>
21:46.05b11danyways
21:46.22b11dsick fuck :)
21:46.38Mercesteslmao
21:46.45Mercestesyou know it's true. :P
21:46.48b11di do know!
21:46.50hmmhesaysi'm loving this new puddle of mudd song
21:46.55b11dhavent heard it..
21:47.01b11dfuck that reminds me, I want to get that new Rush album..
21:47.02hmmhesays"famous"
21:47.03b11dhavent heard it yet
21:47.09hmmhesaysit rocks, download it
21:47.12Mercestess/get/download/
21:47.13b11dok
21:47.21hmmhesaysdefinately better than anything they have released to date
21:47.28hmmhesaysits a straight out rocker
21:47.37b11dthe single song, or the album in it's entirety?
21:47.44hmmhesaysthe song
21:47.49hmmhesaysI haven't got the whole album yet
21:47.58hmmhesaysspeaking of which, I have tickets to velvet revolver
21:48.00hmmhesays!!
21:48.18b11dsweet!
21:48.21b11di love that name
21:48.37b11dkind of ironic, like Jello Biafra
21:48.39MercestesI like to get online....set updates with 14 year olds that I know are stings, and then call the police department and give them the stings address and tell them, "yes, there is this 14 year old trying to hook up with older men."
21:48.40Mercestesit's funny
21:49.02b11dlol Mercestes.. that would rock
21:49.10*** join/#asterisk h0 (i=Fakhir@unaffiliated/fakhir)
21:49.11b11dturn things around on old Chris Hansen from Dateline
21:49.16Mercestesbwahahaha
21:49.28hmmhesaysb11d: i've seen them before, they play some gnr and stp tunes they also did a cover of "wish you were here" by pink floyd
21:49.42b11dhow was the cover?
21:50.05hmmhesaysphenomenal, slash was playing a double neck guitar, just him and scott weiland on stage
21:50.09b11dhey Mercestes.. ever see that "to catch a predator" where they bust the SAME GUY twice, in two days?
21:50.21b11dI'd love to have seen that hmmhesays..
21:50.31Mercestesb11d, nah, didn't see that.  I *did* see the one where they had this model posing as an underaged teen.
21:50.33Mercestes...she was hot.
21:50.37b11dlol
21:50.38hmmhesaysthey're played the excel energy center in two weeks
21:50.54hmmhesays*playing
21:50.57b11dnice.. im going to be gone by then though L(
21:50.57b11d:)(
21:51.00MercestesI don't know why they just don't taser their eyes immediately...
21:51.01b11dnext time I guess
21:51.24Mercesteswhy put them on TV and go "Aren't you embarassed?"  ....just kill them.  =/
21:51.30b11dMercestes.. everyone really knows "to catch a predator" is really "how NOT to get caught hooking up with that hot 15 year old from across the street on yahoo or myspace"
21:52.09b11dlike Chris Hansen himself isnt some sick twisted pedo.. he's got the perfect cover now..
21:52.11Mercestesb11d, I'm going to start a news cast.....set up dates with 14 year olds, and send a news crew instead.  "Why are you trying to hook up with older men from the Internet??"  Big camera and bright lights in their face...air it on national television...send a copy to their parents.
21:52.16Mercestesnow *THAT* would solve the problem in short order.
21:52.26b11dman I'd LOVE to see that show!
21:52.27hmmhesayshe can pay for 18 year olds to pretend they are 15
21:53.00b11dnot bad :)
21:53.09b11dim hoping to do that very thing in Amsterdamn :P
21:53.12b11d-n
21:53.16Mercesteslol
21:53.29Mercestesnice...now I definately want to go. =/
21:53.45b11dI'll think of you guys when im hooting on a 2 foot hash joint and enjoying five girls..
21:53.49[TK]D-Fender"If a woman tells you she's twenty and looks sixteen, she's twelve. If she tells you she's twenty-six and looks twenty-six, she's damn near forty." - Chris Rock
21:53.56b11dhe had that right on!
21:54.06*** join/#asterisk Gadger (n=stuart@host81-137-229-241.in-addr.btopenworld.com)
21:54.26MercestesAin't that the truth?
21:55.35MercestesI was hitting on this girl at a bar...at 11p at night, she has a tatoo on her ankle, hanging out with an older crowd, talking about how she can't wait to get off probation so she can drink again...
21:55.39Mercestesturned out to be 16.
21:55.49b11dnice :)
21:55.56MercestesIn this day and age...I can see how one can get a little confused.
21:56.16b11dyeah.. my ex-boss was hitting this 15 year old.. when I met her, I would have guessed she was 24..
21:56.25MercestesI miss the good old days....
21:56.40Mercesteswomen weren't allowed to leave the house until they were 18, and then they were sold to the highest bidder.
21:56.44b11dlol
21:56.47b11dok Borat :)
21:57.08*** join/#asterisk ShakataGaNai (n=davis@adsl-66-122-88-147.dsl.renocs.nvbell.net)
21:57.31ShakataGaNaiQuestion - How do I force Asterisk to record _all_ incomming calls (automagically)?
21:57.40MercestesShakataGaNai, monitor
21:57.53MercestesShakataGaNai, show application monitor, google monitor, etc.
21:58.03b11dgoogle "asterisk cmd monitor" :)
21:58.13ShakataGaNaiGot it -Thanks
21:58.14Mercestesexacticaly
21:58.20b11dexactimagically
21:58.31ShakataGaNaigooglemagical?
21:58.37Mercestesexactilicious
21:58.44b11dnice :)
21:59.08b11dso.. get anyone to step on your nuts lately Mercestes?
21:59.16Mercestesnot recently
21:59.18b11dweak
21:59.21Mercestesyea
21:59.29*** part/#asterisk ShakataGaNai (n=davis@adsl-66-122-88-147.dsl.renocs.nvbell.net)
21:59.30MercestesI'm the dom now a days.. =/
21:59.38Mercesteswhich means.....
21:59.41MercestesI have the biggest push over ever.
22:00.00b11dwell then "correct" it ;P
22:01.12b11di gotta go lads..
22:01.15b11di'll be back :)
22:08.15CoffeeKidis there anyone to get rid of "  == Parsing '/etc/asterisk/manager.conf': Found" messages in the cli?
22:08.20CoffeeKid*anyway
22:11.29MercestesCoffeeKid, decrease your verbosity
22:11.36GadgerAnyone care to advise on an 2.6.18 & mISDN issue ?
22:13.51GadgerI am using a Digium B410P on a UK ISDN in PTMP mode and I am getting poor one way speech quality
22:14.47GadgerDoes anyone know if kernel 2.6.18 has inbuilt SMP support that I need to disable ?
22:17.46*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
22:35.16[hC]oh noes
22:35.18[hC]they moved astricon
22:35.24[hC]good thing i hadnt purchased my plane tix yet
22:35.30Qwell[][hC]: same airport
22:35.34[hC]oh, okay then. :)
22:35.53[hC]i was worried when the hotel reso people said the hotel was still under construction and may not open in time
22:44.35*** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil)
22:46.48*** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net)
22:54.01knarflyI just upgrade my * server from 1.2.13 to 1.4.10 on a FreeBSD machine. I copied over the conf files but I'm seeing this message: Prefixing the mailbox with an option is deprecated. Can anyone tell me what that means?
22:54.42Qwell[]knarfly: means you need to read the UPGRADE file
22:55.27knarflyQwell, where can I find a copy of it...?
22:55.47Qwell[]in the top of the source dir
22:56.20knarflyQwell, thanks...I'll check it out.
22:59.01*** join/#asterisk Mavvie (n=edwin@ppp121-44-39-32.lns3.syd7.internode.on.net)
22:59.18Yourname`Hmmm, is there such a thing like answering machine detection "classes". Like different forms of AMD? For example, first 3 seconds of greeting, then wait for 4 words.. if that's qualified, move on the AMD class 2, or something like that?
23:03.58knarflyQwell, was there something specific you were referencing. I just checked the file and the README and didn't find any references to mailbox prefixing
23:04.35blitzrageknarfly: Voicemail(u100@default) is deprecated.  Use Voicemail(100@default,u)
23:04.45blitzrageand UPGRADE.txt is handy
23:06.53knarflyblitzrage, I was just in UPGRADE.txt...where did you find that reference...that's the one I'm after?
23:07.19blitzrageknarfly: sorry, I just knew what the message meant that you pasted
23:07.34blitzrageI didn't read the file for it... but I'd assume it's in there somewhere
23:07.40blitzragebut maybe it's not :)
23:07.57*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
23:08.08knarflyblitzrage, thanks...so I move the the u to become a suffix in lieu of a prefix?
23:08.29blitzrageknarfly: ya -- the syntax for applications is much more consistant in 1.4 than 1.2
23:09.33YarqUnable to create channel of type 'IAX2' (cause 3 - No route to destination)
23:10.16Yarq(the remote host is pingable )
23:13.58*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
23:14.51*** join/#asterisk luke-jr|work (n=luke-jr@ip72-206-114-236.om.om.cox.net)
23:15.02luke-jr|workDo SIP phones generally obey SRV records?
23:26.40*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
23:27.50*** join/#asterisk fakhir (i=Fakhir@unaffiliated/fakhir)
23:28.10*** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk)
23:30.34*** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net)
23:30.49*** join/#asterisk nirz (n=nirz@79.179.88.51)
23:34.23*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
23:36.20knarflyanyone ever setup * to use an FM radio card to do moh?
23:36.25YarqAny nufone users?
23:37.56luke-jr|workknarfly: interesting idea
23:38.39Yarqknarfly: Recently one of my friends found a USB Fm radio card
23:39.04Yarqhe used to connect it at his house and from the office he used it for listening to radio (he had a script where he was able to change the channel)
23:39.11knarflyI think it can be done...I just dug up an old ISA FM card and installed it in one of my servers...I loaded xtuner and the darned thing worked pretty good.
23:40.14knarflyI'm getting bored listening to my mp3 files all the time on hold and want some fresh music and prehaps some news once in a while
23:40.50[TK]D-FenderYarq, that error can mean jsut about anything.  Pastebin the full CLI output of the call with IAX2 debug and your IAX peer entry maskeing only passwords.
23:41.06knarflyI read somewhere about someone doing it with a real radio wired into the sound card..I'll have to find that article again.
23:41.38knarflyYarq, did they get it working?
23:45.52Yarqyes
23:46.53knarflyFound something on the FM card + sound card combination that might work...http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
23:50.45Yarqhttp://pastebin.ca/661708
23:51.47[TK]D-Fender<Yarq> Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
23:52.06Yarqyes
23:52.16[TK]D-FenderYarq, Your first error was IAX2 now you're showing me SIP, and you have channel debug information for NEITHER
23:52.29Yarq[TK]D-Fender
23:52.32Yarqwhat I'm trying to do is
23:52.46YarqI am calling my asterisk box with a sip softphone
23:53.04[TK]D-FenderYarq, Do the call again with SIP debug ENABLED.
23:53.14Yarqand dial the extension 6001 (which needs to ring my cell over Nufone using IAX2)
23:53.56*** join/#asterisk SgtDitt (n=sgtditt@adsl-074-236-020-203.sip.mia.bellsouth.net)
23:54.32Yarqhttp://pastebin.ca/661710
23:57.50[TK]D-FenderYarq, Ok, and now again your are dialing IAX2 to nufone!
23:57.52[TK]D-FenderWTF!
23:58.01SgtDitthello all. happy friday
23:58.08[TK]D-FenderYarq, You are flip-floping all over the place.
23:58.31SgtDittcan anyone help me with getting queues to work with the AsteriskNow Distro???
23:58.45YarqTK
23:58.48[TK]D-FenderSgtDitt, You're in the WRONG channel.
23:58.51Yarq[TK]D-Fender
23:59.05YarqAt office I am using softphone (which connects over sip)
23:59.48[TK]D-Fender<PROTECTED>

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