IRC log for #asterisk on 20111217

00:00.38raden_workUnable to connect to remote asterisk (does /run/asterisk/asterisk.ctl exist?)
00:01.09raden_workAsteris-Server:/run/asterisk # dir
00:01.10raden_worktotal 4
00:01.10raden_worksrwxr-xr-x 1 asterisk asterisk 0 Dec 16 18:07 asterisk.ctl
00:01.35leifmadsendo you have permissions to connect?
00:01.55raden_workim root
00:02.05raden_workon local machine
00:02.55tompawWIMPy: http://pastecode.com/i5 :-)
00:03.32raden_workleifmadsen, always a new issue :)
00:04.04WIMPytompaw: What's that?
00:04.20tompawWIMPy: AMI in action loggin agents in automatically.
00:04.42WIMPyAh, so yu are already using it.
00:04.47tompawyep
00:05.30SeRiguys voip.ms premium international routes is having issues
00:05.44tompawNow to capture the logout/timeout... this will be slightly more complex.
00:05.50WIMPySo if the qualify thing works it should be relatively easy to add to what you already have.
00:05.51tompawSeRi: and?
00:06.08SeRiif does not concern you do not read it.
00:07.40tompawWIMPy: haven't tested it yet. Will try now, but I'm concerned about the number of possible ways a user(agent) can log out. He can simply sign out, or be signed out by qualify...
00:13.42raden_workleifmadsen, here we go again :(
00:15.29WIMPytompaw: Easy so far. But you may want to log them in again if they have only become temporarily unreachable.
00:18.03tompawWIMPy: If they're kicked out by qualify, they will have to REGISTER again, correct?
00:19.59WIMPyYou decide what you put in to your script.
00:34.53raden_workRemote UNIX connection disconnected
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00:35.09SeRi`pb
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00:35.13SeRi~pb
00:35.13infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
00:35.13raden_work<PROTECTED>
00:35.15raden_work<PROTECTED>
00:35.17raden_work<PROTECTED>
00:35.19raden_workAsteris-Server*CLI>
00:35.21raden_workcrap sorry
00:35.23raden_workleifmadsen, ^^^ keeps coming up on console now :(
00:35.26raden_workSeRi, i know bro
00:35.52tompawWIMPy: it's working both ways now :-)
00:36.07tompawWIMPy: I simply changed minexpiry and maxexpiry to 60
00:36.08WIMPyraden_work: So you've got some funny GUI running?
00:36.09leifmadsenraden_work: ok, so you have something (likely a script) connecting and disconnecting
00:36.33raden_workleifmadsen, i just installed
00:36.45leifmadsenshrugs
00:36.45tompawWIMPy: not AMI captures those expires and keeps the queue clean, yey!
00:37.02tompaws/not/now/
00:37.17carrar-- Remote IRC connection
00:37.42carrarWelcome to IRC
00:38.16carrarbeen here long raden?
00:38.19carrar;)
00:38.56carrarqueues up a list of SMRT4SS comments
00:39.47tompawLuckily, QueueAdd doesn't unpause an agent if he's already in the queue :-)
00:40.35SeRidijib: You are in?
00:47.27*** join/#asterisk zerohalo (~zerohalo@74.60.136.128)
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01:13.04philippel_macwondering if anyone knows what specific version the CLI format changed from "core set global ..." to "dialplan set global ..." ?
01:28.21*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
01:35.42leifmadsenphilippel_mac: like... 1.6.1 or something
01:35.59leifmadsenit was... 2 astridevcons ago
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01:40.46philippel_macleifmadsen:  thanks, that gets it closer ...
01:41.00leifmadsenwhy does it matter?
01:41.27leifmadsenthere was a huge cleanup of the CLI at... 2 astridevcons ago by mvanbaak, JunK-Y and myself
01:42.00leifmadsenif it does, just use cli_aliases.conf to make it act like it used to
01:42.03leifmadsenruns away
01:42.14philippel_macbecause we send that via the AMI from FreePBX/ARI  to set DEVICE_STATE() for various features like DND, Follow-Me, etc
01:42.42philippel_macso I need to send the right syntax depending on the * version
01:48.38leifmadsenor just need to set cli_aliases.conf
01:48.51leifmadsenthen your code doesn't need to change
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02:23.50philippel_macwe don't supply that file, would rather just detect what they are running and send the correct command, that is how we do a lot of other stuff
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03:20.34SeRiWIMPy: 1.8.8.0 is now been build on astlinux. should be able to get this copy instead of the old one. I do not have access to the webui source so I am waiting to see when is done building what version the webui will display
03:21.32SeRiI trust 1.8.8.0 more than 10 right now :) well I feel safer to build. I will go down that rout later on.
03:21.49SeRis/rout/route/
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04:19.42pyite_macanyone here have experience with yealinks + intercom?
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04:58.07dijibyour not here are you Seri?
04:58.19dijibbrb logging into another irssi client out back.
04:58.49*** join/#asterisk sixohquad (~sixoh@184.65.142.249)
05:02.04sixohquadGood evening all. Wondering if somebody could tell me approx how many concurrent calls a Celeron g530 (2x2.6) w/ 2gb ram could handle?
05:02.33sixohquadI'm not sure if there even is such a measure lol
05:03.22sixohquadI'm thinking in a business setting w/ clarity as a must
05:04.22[TK]D-Fenderwhat do you need to handle, and what will it be doing exactly?
05:05.48*** join/#asterisk nobodyshome (~root@bas10-kitchener06-1176001642.dsl.bell.ca)
05:06.01nobodyshomeso anyone here?
05:06.30[TK]D-Fendernobodyshome: Your channel listing may be a plce to start guessing
05:06.55nobodyshomei hadn't seen anybody speak since Seri at 22:something
05:07.27nobodyshomethis is dijib btw
05:07.59[TK]D-Fender[23:19]pyite_macanyone here have experience with yealinks + intercom?
05:08.17pyite_machi
05:08.32nobodyshomewhats yealinks
05:08.37pyite_macYealink phones :)
05:08.46nobodyshomedont know of em
05:08.49pyite_macwondering if you know anything about the behavior of the Intercom Barge functionality + Asterisk...
05:08.57pyite_macnobodyshome: really? wow ! heh i thought everyone knew them
05:08.59pyite_macwww.yealink.com
05:09.05pyite_maccheap Cisco knock-offs hehe
05:09.10nobodyshomecant ive only got console
05:09.49nobodyshomeyou did mean intercom bridge function right?
05:10.15p3nguinseri: I'm back.
05:10.24SeRip3nguin: !!!!!!
05:10.26SeRi:)
05:10.35SeRiwaz up buddie
05:10.58nobodyshomep3nguin: just as a failed login attempt apparently
05:11.09SeRimy internet issues are back :(
05:11.11nobodyshomei just woke up from a 3hour nap
05:11.19nobodyshomewhats the issue?
05:11.36SeRiwaz up dijib :)
05:11.40nobodyshomeim up
05:11.42SeRioh major internet issues
05:11.51nobodyshomewhats dah issue?
05:11.59nobodyshomenick dijib
05:12.10SeRimy upload channel and downlad channels have bad snr
05:12.53nobodyshomebad enough they cannot garentee the service their selling you
05:13.02nobodyshomeguarentee
05:13.13SeRibad enough to come and replace a whole addressable tap
05:13.19nobodyshomethats not even how to spell it is it
05:13.37SeRiand several ones around
05:13.48nobodyshomehuh?
05:14.33p3nguinaddressable?  What's wrong with addressable?
05:15.05nobodyshomewhats the hellissa this addressable docsis term
05:15.09SeRiI am not sure :/ may be a dijib special?
05:15.25nobodyshomelooks like someything i would pull offf
05:16.40nobodyshomeback in 20, at least
05:16.59SeRinobodyshome: http://www.electroline.com/en/products/addressable/index.html
05:17.22SeRihow the hell do you mix an addressable tap with docis?
05:17.38SeRis/docis/docsis/
05:18.01SeRip3nguin: I was connected for like an 1hr
05:19.12SeRiwill be testing again soon. building 1.8.8.0
05:21.48SeRithe 560 it's on the way. should be here by monday
05:22.32SeRis/it's/its/
05:22.42SeRis/its/is/
05:22.46SeRilol
05:22.48SeRio well
05:23.22[TK]D-FenderSeRi: How much did you win it for?
05:23.41SeRi[TK]D-Fender: 124.99
05:23.47[TK]D-Fenderhell of a deal
05:23.52SeRiyes :)
05:24.04SeRi*used but in good conditions*
05:24.14SeRibought the 321 from the same guys
05:32.22p3nguinWere you just listening to music until your internet went out?
05:32.44p3nguin(on my system)
05:33.16SeRiyes
05:33.36SeRiit just droped
05:33.50SeRimy modem snr are on the -
05:33.53p3nguinYou probably got tired of the same five sonds.
05:34.02p3nguinsongs
05:34.52SeRilol
05:34.57SeRinot really.....
05:35.04SeRiI left it there till it droped
05:35.07SeRiI went to eat
05:35.14SeRidown stairs
05:35.26WIMPySeRi: Do you have a summary / documentation on what you build, BTW?
05:35.49SeRiWIMPy: Yes
05:35.58SeRiIt will be have to be polish
05:36.05SeRipolished*
05:36.18p3nguinPolish... as opposed to Russian
05:36.22SeRibut astlinux in general has its own site
05:36.58SeRiI am keeoing nores and licenses on what I am adding and building
05:37.05SeRisnores/notes/
05:37.09SeRifuck man
05:37.12SeRitired
05:37.14SeRilol
05:37.23SeRione sec
05:37.24WIMPyYes, but there are amy things to choose and may patches to apply.
05:37.36SeRiindeed
05:37.57WIMPy(or remove)
05:38.47dijibback
05:42.15dijibso equipment faliure
05:45.39SeRi?
05:57.36SeRiok back
06:01.32SeRiso hungry :( so late...
06:02.55SeRip3nguin: I have an open ticket with voipms regarding routing.
06:03.10SeRitheir premium international routing is not working
06:03.27SeRiI was getting 503
06:11.36dijiboh henry
06:14.19SeRiwhats going on dijib ?
06:17.07dijibim pooped u?
06:17.18SeRioh. eeeewwwww
06:17.21dijibim horrible for having this window in focus
06:17.35dijibi think ive been having a heart attack today
06:17.44dijibsore left arm
06:18.42SeRicool
06:19.44dijibyou?
06:22.04SeRiastlinux-1.0-5310 - Asterisk 1.8.8.0
06:22.07SeRi:D
06:23.53dijibim curious about asterisk 10
06:24.11dijibwhat bennefits does it have?
06:25.08dijiband what am i missing frmo my asterisk install?
06:36.38SeRimhhhhhhh no moh.....
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06:41.04nobodyshomeits got moh
06:42.29nobodyshomei mean 3rd party apps to add to my fullblown asterisk install
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06:48.31SeRiI am not talking about that
06:48.49SeRiI am talking on my own build... loks like persistant storage is acting up again
06:49.02dijibmoh is eh
06:49.10dijiblocks the persistent storage>?
06:49.22SeRiignore me :P
06:49.27dijibmpg123 streamplayer?
06:49.37dijibmoving from terminal to terminal
06:49.41dijibwent outside for a smoke
06:49.45WIMPyoh no, no moh no mo'
06:49.54SeRilol
06:50.01dijibjust smoked one
06:50.15SeRigot it working now
06:50.29WIMPyGimme moh
06:50.38SeRiyou got it!
06:50.43SeRi:P
06:50.50dijibhorrible rap music
06:50.56SeRiworking out some of the issues I am finding
06:51.00WIMPyRight, you didn't break mine :-)
06:51.16SeRidijib: pass me your mpg123 moh context
06:51.19WIMPyThat's not rap. It's just too much jitter.
06:51.20SeRiWIMPy: lol!
06:51.39dijibi have lie 10,000 different mpg123 contexts
06:51.54SeRithe one you use for your music streaming moh
06:52.42dijibthis was my lat working one application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://icecastsource2.amri.ca/chtz-mp3
06:52.56dijibunder set context with set options
06:53.01dijibmode=custom
06:53.11dijibdijit=4
06:53.14dijibwhatever
06:53.15p3nguindoesn't even know what an moh context is.
06:53.34dijib[station]
06:53.39dijibchannel?
06:53.41dijibwhat?
06:53.43p3nguinclass
06:53.49dijibk there you go
06:54.04dijibits that a transversable term
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06:55.14*** mode/#asterisk [+o leifmadsen] by ChanServ
06:55.18dijibtraversable
06:55.41SeRip3nguin: sorry.
06:55.42dijibs/transversable/traversable
06:55.47p3nguinsed: -e expression #1, Unterminated `s' command
06:55.49dijibwhat did i do wrong?
06:56.29SeRip3nguin: LOL
06:56.51p3nguinYou must terminate your expression.
06:56.58p3nguins/must/have to/
06:57.11dijibs/transversable/traversable/
06:57.29WIMPyAs if infobot would really be sed compatible.
06:57.30p3nguinAnd it only corrects the last thing you said.
06:57.51p3nguinWell, it's a perl bot, and perl's s replacement works exactly the same as sed's.
06:58.08dijibmy back hurts
06:58.13p3nguinmine too
06:58.20p3nguinvery much
06:58.22dijibok ok ok, h264 conference, howto:
06:58.28WIMPyI don't believe that.
06:58.36dijibmy back and left side
06:58.41dijibdamn smoking too much
06:58.43WIMPys#believe#see#
06:59.05dijib;you;have;me;lost
06:59.05WIMPyAnd it certainly doesn't like options like g.
06:59.07dijib;
07:00.46p3nguinYou mean You can't get it to work?
07:00.55p3nguins/You/I/g
07:01.05p3nguinLooks like g works just fine.
07:01.30dijibhttp://images.4chan.org/k/src/1324077565147.jpg
07:01.35*** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com)
07:01.38WIMPyIt never did for me. foo one foo two.
07:01.43WIMPys/foo/bar/g
07:01.52dijibneuclear cake processing facility iran.
07:02.00WIMPyIs that new?
07:02.06p3nguinI don't think so.
07:02.29p3nguins^think so^know^
07:03.15p3nguinThe bot must not replace on all chars like that, but I'm quite certain that perl itself does.
07:04.10WIMPyI'm pretty sure using /g didn't come up as expected, before.
07:04.24SeRip3nguin: I ahve not play moh mpg123.... does this sound about right? http://pastebin.com/U07Qe9Uh
07:04.50SeRiI have a small fealing is backwards :/
07:06.55p3nguinYou only need mode and application.  http://pastebin.com/4Gx73WE5
07:08.14p3nguin$ echo "foo food foot"|perl -pe 's^foo^bar^g'
07:08.14p3nguinbar bard bart
07:08.31p3nguinperl handles those chars just fine.  It's apparently the bot that doesn't do replacement with them.
07:09.03p3nguineven s#foo#bar#g works
07:09.17p3nguinBot must be picky.
07:12.23SeRibroken....
07:12.26SeRires_musiconhold.c:659 in monmp3thread: poll() failed: Interrupted system call
07:12.38SeRilooks like another damn astlinux issue
07:14.59SeRirofl
07:15.10SeRiit's broken.... well I guess I am back at the table
07:15.33*** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net)
07:15.39WIMPyWell, it needs somethig to search for. I guess it wouldn't be too smart trying to invoke that function for all lines starting with an s.
07:16.37SeRimpg123 is missing libs
07:17.03p3nguinI'm not sure why / is the most common char to use, but that's the one I always see.
07:17.32p3nguinIn my pkgbuilds, I use | for my sed replacements.
07:24.19p3nguinAnyone know off hand what the max file size on vfat is?
07:24.36WIMPy2GiB
07:24.58p3nguinYou sure it's not 4?
07:25.10WIMPyThe evil thing being that Windows will tell you the disk is full if you try to write somethig bigger.
07:25.24SeRiWIMPy: he is right
07:25.31SeRi32Bit 2GB
07:25.35p3nguinI was trying to write a 4.4G file to vfat on my thumb drive.  It was taking a long time... then exited.
07:25.41p3nguinIt said File too large.
07:25.43WIMPyIt should be, but you know C programmers...
07:25.49p3nguinThe part that did copy is 4.0G
07:26.01p3nguin4294967295
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07:26.14p3nguinSo that's an fs limit, then.
07:26.15SeRiFat32
07:26.19p3nguinWonderful.
07:26.58WIMPyYes. Took me some time to realize that because of the misleading message on Windows.
07:27.23p3nguinNow I don't know what I'm going to do.
07:27.58WIMPyUse another FS.
07:28.09SeRintfs
07:28.17WIMPyIf you want to use it on windows that would have to be ntfs.
07:28.31SeRiyou could use ntfs-3g on linux
07:28.37SeRiread and writes to ntfs
07:29.00WIMPyYes, don't use the old one.
07:29.06SeRi+1
07:30.56p3nguinI'm using neither Windows nor Linux, and I need it to be FAT 32.
07:31.08SeRimac
07:31.12WIMPyNo go
07:31.28p3nguinmayyyyybe I can use ntfs.  I'll try it tomorrow.
07:31.31WIMPyUse some archiver that can split.
07:31.40p3nguinCan't.
07:31.48p3nguinNeed one file.
07:32.15WIMPyNot on FAT.
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07:35.21p3nguinI found another alternate file that is 4.00 GB.  I don't know if I can squeeze it on or not.  I have to try it.
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07:48.20SeRiok guys I am off to bed. gots to work tomorrow. ill continue at it tomorrow.... Thanks for help guys
07:48.44p3nguinI already went to sleep like an hour ago.
07:48.50SeRilol
07:48.53SeRihahaha
07:48.55SeRicya!
07:48.58SeRig/n
07:50.18dijiblater
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08:31.17phix:)
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08:33.53Kattyohai
08:35.31Kattyi think everyone must have gone to sleep :<
08:36.58*** join/#asterisk ChannelZ (channelz@burner.com)
08:50.13singlerKatty: or did not get up yet :)
08:51.01Kattyanother possibility!
08:52.08WIMPyHi Katty! You've been away for a long time.
08:52.20WIMPyDid you have better plans?
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08:58.42Kattyrk
08:58.45Kattyerm
08:58.50Kattyi'm never away, i just always lurk!
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10:45.46kresp0hi all
10:46.12WIMPylo you
10:47.19kresp0anyone want to receive a test call from spain?
10:47.27kresp0warning: my english is bad
10:48.02kresp0if so, pm me
11:04.29tompawkresp0: would you like to buy spain proper for .005? ;)
11:05.26WIMPytompaw: You sell ghosttowns?
11:05.38tompawWhat are ghosttowns?
11:05.45kresp0tompaw: i really dont think that yours will be proper
11:06.02tompawI got 2 E1s of Telefonica and BT going through AS5350.
11:06.02WIMPyFamous spanish property markets.
11:06.33tompawAh, no, but I currently live in one, and the weather here is amazing, too bad I have to go back on Monday :/
11:06.36WIMPyAS5x00 make acceptable seats.
11:06.42kresp0tompaw: but if you, or anyone else on the internet, want to call me to sell me something... do it: +34 911163221
11:07.43tompawit's only 2 E1s of postpay half a cent traffic mate, if you don't want to, why call ;)
11:07.59tompawWIMPy: btw - the idea you gave me last night works perfectly
11:08.17WIMPyGood
11:08.28WIMPyAlways nice when something works.
11:08.40WIMPyDoesn't hapeen too often these days.
11:09.35kresp0thank you tompaw, thats a proper line for sure!
11:09.38kresp0i only have a residential pstn line
11:09.51*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
11:10.26tompawNah, it's not bad. Since I dropped ConfBridge in favor of MeetMe, things work great :-)
11:13.24kresp0tompaw: do you sell traffic to make calls to spanish mobile? how much would be?
11:18.02tompawkresp0: sorry, we do not do mobiles.
11:20.10kresp0ok, thank you tompaw :)
11:23.35WIMPyYes, that holding bug in ConfBridge is extremely annying.
11:24.02WIMPyBut I didn't want to have dahdi just for MeetMe. And after my latest experiences even less.
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11:26.24tompawWIMPy: it was simply hanging my asterisk killing it every 2-3 hours. Dahdi was a pain, since I needed to recompile the whole kernel (ovh - custom kernels!), but once it's set up, it's rock solid.
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12:00.49CaneToadAny tricks to allowing a SIP invite to be received by asterisk [no registration is involved]?   What I'm seeing is that net snooping shows the invite coming in, but it presently doesn't even show if I "sip set debug on".  If I register against a server then that works, but with this unregistered request coming in, it seems a little different.
12:02.29CaneToadmaybe I have to tell asterisk to allow requests from this address without registration ???
12:04.19WIMPyThere is not relation between sending registrations and sending calls to Asterisk.
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12:07.15CaneToadmy problem is that I can see the request coming in on the correct port using snooping software, but asterisk ignores it completely....it doesn't even show it when using "sip set debug on"... I don't understand why asterisk is ignoring it
12:21.58kaldemarCaneToad: did you just upgrade to 1.8?
12:23.55CaneToadno
12:27.29kaldemarasterisk doesn't like headers sent by some clients when pedantic=yes in sip.conf. the default value was changed from no to yes in 1.8. try setting it to no.
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14:14.27leifmadsenso I have a question... what *is* it that asterisk does?
14:14.31leifmadsenlike... what is it used for?
14:14.36leifmadsenand why is it free?!
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14:17.18singlerit does that it is programmed for, and it is free because developers decided so :)
14:17.27IsUphello
14:17.38singlerhi
14:22.56leifmadsenbut like... what are some of the things it can do?
14:23.03leifmadsenlike, can I DJ with it?
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14:25.22francisvgarciadoes anyone know why I am having this error when using the mp3player application app_mp3.c:133 timed_read: Poll timed out/errored out with 0
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16:37.10Kobazanyone have any recomendations for a bluetooth speakerphone
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16:43.11kresp0hi all, i'm trying to use the PrivacyManager(), but when a call without caller-id information comes from the pstn, my spa3000 sends to * the pstn user name ("pstn") instead of nothing. Any suggestion?
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16:49.25[TK]D-Fenderkresp0: Compare the value yourself
16:51.18kresp0thankyou [TK]D-Fender, i'll do that
16:58.38[TK]D-FenderThat app and LookupBlacklist(), etc are largely a waste of space & time.  You can do the same yourself with under half a dozen lines of dialplan yourself and often better.
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17:09.30kresp0yesterday we replaced LookupBlacklist() with one simple dialplan line
17:11.20*** join/#asterisk cerberus_za (~coert@8ta-151-22-106.telkomadsl.co.za)
17:11.43[TK]D-FenderSelf-evidently redundant....
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17:33.17pkzsweHi! I would be grateful for some pointers on how to set up a modem pool over VOIP. Is it possible with Asterisk?
17:35.32kresp0pkzswe, do you have a real modem hardware?
17:36.30kresp0it is possible to use a modem over voip, but you have to use g711u and lower the bps
17:36.40pkzswekresp0, no, but will get if necessary.
17:37.15pkzswekresp0, use case is there are ppl who right now are shut off from the net but have access to plain old phone lines.
17:38.27pkzsweAccording to http://bit.ly/tXJiaG it looks like it is possible
17:38.38kresp0it is possible, sure
17:39.07kresp0you will need:
17:39.26kresp0* DID number provider
17:39.50kresp0* Real hardware old-school external modem
17:40.27kresp0* asterisk server with an fxs port OR a linkys pap2t (2 fxs ports)
17:41.59kresp0* a computer connected to the modem(s) and configured to run a ppp server
17:42.23kresp0i.e: http://werebuild.eu/wiki/Egypt/Howto_PPP/Telecomix
17:43.18kresp0and btw pkzswe: who are those people?
17:44.41pkzswekresp0, right now they are in central asia and time is important. Next time they are somewhere else.
17:46.06kresp0pkzswe: send them some of this numbers: http://pad.telecomix.org/dialup-arch
17:47.07pkzswekresp0: thank you.
17:48.15pkzswekresp0: are there service providers who package and sell this type of setup?
17:48.39kresp0pkzswe: sorry i dont know
17:49.11kresp0pkzswe: join #blackout.monitor chameleon.irc.telecomix.org
17:58.13pkzswekresp0: http://www.reuters.com/article/2011/12/17/us-kazakhstan-clashes-idUSTRE7BG08D20111217
17:58.41kresp0thank you pkzswe
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18:14.49WIMPyleifmadsen: A very effective way to get old without achieving much.
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18:17.13[TK]D-Fender[12:33]pkzsweHi! I would be grateful for some pointers on how to set up a modem pool over VOIP. Is it possible with Asterisk? <- realistically?  No
18:17.35[TK]D-Fenderpkzswe: Modem data can't survive jitter, packet-loss, echo cancellation, etc.
18:17.37[TK]D-FenderALL DOA
18:17.57[TK]D-FenderYou might survive 2400 baud for a few moments for CC processing ... maybe
18:18.07[TK]D-Fenderbut any real data?  Quite dead...
18:18.28WIMPy[TK]D-Fender: What do you think will be faster? A modem over VoIP (if it connects at all) or calling someone to read out something for you?
18:18.39[TK]D-FenderDepends :0
18:18.40pkzsweHow about using a different codec like G.711?
18:18.58WIMPypkzswe: It can only get worse.
18:19.00[TK]D-Fenderpkzswe: companding will still have loss.  Connecrtions will still have jitter and packet loss
18:19.11[TK]D-Fenderpkzswe: Echo cancellation still happens.
18:19.17[TK]D-FenderAll death-sentences
18:19.51pkzswe[TK]D-Fender, WIMPy, Ok. So providing even a slow internet connection will be difficult.
18:19.54WIMPyI'm not sure about Asterisk 10, but as far as I remember, Asterisk isn't even bit-transparent.
18:20.00[TK]D-Fenderpkzswe: try near-impossible
18:20.36WIMPyI'd never try internet of such a setup. Mabe a terminal could be made to work.
18:20.36pkzsweSo, transfering pictures would be better off using a fax?
18:21.25WIMPyNo, you need PCM to terminate the modem sessions instead of VoIP.
18:24.46pkzsweWIMPy, ok, thank you. I guess a
18:24.48kresp0pkzswe, WIMPy, [TK]D-Fender: it is possible. i've done it before and it works
18:25.00kresp0You have to use g711u AND use lower bps
18:25.19pkzswekresp0, yay!
18:25.19kresp0it works ok, the modem can take care of all that jitter and data loss
18:26.17[TK]D-FenderAs I said, you might keep 2400 for a bit..
18:26.20[TK]D-Fenderon a prayer
18:27.02[TK]D-FenderIf this is a business plan then please make sure both barrels are loaded when you put the shotgun to your head :)
18:27.35[TK]D-FenderAnd step over here onto this plastic sheet .... we'd hate to have to clean up too much of a mess
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18:28.32kresp0[TK]D-Fender: <pkzswe> kresp0, use case is there are ppl who right now are shut off from the net but have access to plain old phone lines.
18:29.37WIMPyWell, get some phone lines yourself.
18:30.21kresp0that would be the best option, if you can pay it
18:31.25WIMPyif you cannot even pay for a phone line, who is going to pay for complicated version?
18:32.00[TK]D-FenderWe've long since entered the realm of the psychotic...
18:32.32florz.o()
18:32.43WIMPyOk, so this is at least the right place.
18:32.49WIMPyHi florz
18:33.28florzhi
18:33.53florzno, a modem will not deal with jitter
18:34.42florzat least not the kind of packet latency jitter we are speaking about here
18:35.09florzhowever, you can just put a jitter buffer in front of it in order to get rid of the jitter
18:35.38WIMPyA static jitterbuffer, I may add.
18:36.18florzwell, in the optimal case one that doesn't ever try to reduce latency
18:36.34florzand one that introduces latency from the beginning
18:36.36WIMPyyes
18:37.32florzso the amount of data in it can vary widely without ever getting empty or full
18:38.21florzpacket loss doesn't really happen on a reliable internet connection as long as there is no congestion
18:39.21florzand whether there is any audio processing happening at the gateway and whether that gets disabled when guard tones are transmitted depends on the provider
18:39.38pkzswe<PROTECTED>
18:40.57WIMPyYes, it should be possible in theory, but as I said: That doesn;t mean it's practically possible with Asterisk.
18:41.13WIMPyDoes someone know if *10 is bit-transparent?
18:42.13WIMPyreally should try, but I don't dare. There's a limit to bad results I can tolerate.
18:45.50florzyou totally can make modem connections via an ata through asterisk to the pstn
18:46.12p3nguinSounds pretty useless.
18:46.36p3nguinIf you can do VoIP, why not use the IP part of it for your data that you'd be putting over the modem?
18:46.53florzp3nguin: I think it's called "legacy"
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18:47.37florzpotentially, you don't want to swap out, say, your production facilities because you are switching to a VoIP PBX
18:47.37WIMPypkzswe gave an example. Fax is another.
18:47.37[TK]D-FenderI also love how we haven't pinned down HOW you get to the "PSTN"
18:47.38p3nguinAs in, the device only has a modem and no NIC... and cannot be changed?
18:48.08WIMPyAnd remote access that doesn;t require internet can be very handy as well.
18:48.09p3nguinI didn't scroll up, so I didn't notice the example.
18:48.11florzp3nguin: and potentially it doesn't even speak any IP, yes
18:48.54florzpkzswe: well, conceptually it's as simple as kresp0 said
18:49.14p3nguinI fax through an ATA and asterisk all the time.
18:49.14pkzswep3nguin: use case is ppl who only have plain old telephony access because internet is shut down.
18:49.50florzpkzswe: the telecomix guys don't have any dialin available atm?
18:50.11kresp0florz: you're right, the modem will not deal with jitter. that work is done by the ata or the asterisk
18:50.48florzwell, it needs to, if it isn't, the modem most likely will get totally lost
18:51.04florzit may be able to recover from some occasional hickup
18:52.20[TK]D-Fenderpkzswe: Internet is "shut down"?  Is this in a gov't crack-down zone?
18:53.14p3nguinJust a crack zone.
18:53.22kresp0florz: i hear a modem, so i assume that yes, the telecomix dial-up link is up
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18:53.51pkzsweflorz, there are, but there will be a constant need for modem access in the future. Investigating options for more setups.
18:54.30kresp0florz: if you want to know for sure, i was looking for an excuse to use my modem over voip again :D
18:55.04florz=:-)
18:55.10kresp0ok then
18:56.26[TK]D-Fender[13:52][TK]D-Fenderpkzswe: Internet is "shut down"? Is this in a gov't crack-down zone?
18:56.41p3nguinCrack.  Just crack.
18:57.07p3nguinWill trade crack for internet.
18:57.07[TK]D-Fenderp3nguin: I'm asking for his situation, because it may be crazy AND desperate.
18:57.15[TK]D-FenderThe two are not mutually exclusive
18:57.38pkzswe[TK]D-Fender, yes.
19:01.22[TK]D-FenderSorry to hear.  If you have to support real people in such desperate situations, get ready to pay for something better.
19:01.51[TK]D-Fenderpkzswe: Because you don't go through that kind of trouble to fail
19:02.29WIMPyDon't make them suffer even more.
19:04.51kresp0let's try those dial-up-over-voip links
19:05.00kresp0brb (i hope)
19:05.16florzkresp0: but please don't block them unnecessarily
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19:10.56WIMPyThinking about that topic: How do you disable EC on a per call basis on dahdi?
19:14.12pkzswe[TK]D-Fender, I know, but it is a matter of trying vs nothing at all. Info on regular dial-in providers (who seem to be fewer for each year) are most welcome.
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19:18.00WIMPyI am sure T-online still do modem.
19:19.32WIMPyBut if those ppl are to use dialup abroad, I fail to see how getting a phone line could be a cost issue.
19:25.49florznow, no clue what the rates are in kazakhstan
19:26.11florzbut international calls are not necessarily expensive anymore?
19:26.48WIMPyUsually depends on quality.
19:27.25florzyeah, sure
19:27.32WIMPySome el cheapo carrier using VOIP with a lossy codec will be of no use.
19:28.02florzyeah
19:28.38florzbut even g.711 is low-bandwidth really by today's standards
19:29.05WIMPytrue
19:29.52WIMPyBut then we have RTP to increase bandwidth. [cough]
19:30.49florz=:-)
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19:46.24*** join/#asterisk Wellard (~Wellard@2001:470:1f11:cf8:60b7:88c:9498:8fa7)
19:46.38Wellardhi there, anyone able to help me out with the asterisk-gui?
19:46.54WIMPyWellard: Join #asterisk-gui
19:47.01Wellardalready have, nobody is responding.
19:47.09Wellardwhich is why i came here
19:47.10Kobazwe don't know asterisk-gui here
19:48.38Wellard179 people here any nobody know?
19:48.56p3nguinThis is an asterisk channel, not a gui channel.
19:49.09WIMPyIf we were in to Asterisk-GTU, we would be in #asterisk-gui
19:49.10Kobazcompletely different projects
19:49.27carrarDump the gui
19:49.35WIMPys/GTU/GUI/
19:49.47WIMPyYes, we can help you free yourself :-)
19:50.32carrarYou don't want to use training wheels all your life
19:51.03KobazWellard: it's like asking an engine specialist mechanic about how to fix your stereo in your car
19:51.27KobazWellard: asterisk is the engine, asterisk-gui is the stereo
19:51.42carrarnaw, I wouldn't give it that much credit
19:51.54carrarStereo's are a must have!
19:52.24Kobazheh
19:52.25Wellardapart from #asterisk-gui (where nobody is responding), can anyone at least suggest where else I can go? the documentation i have from the asterisk wiki isn't helping.
19:52.38Kobazgoogle might know
19:52.49Wellardyeah, googled too. but no dice.
19:52.59carrargoogle more
19:53.11Kobazyou could start by announcing your problem, but most likly we can't be of much help
19:53.16carrarreinvent your google criteria
19:53.48carrargoogle also isn;t the only search engine
19:53.52Wellardgui asks for a parameter in manager.conf, enable=yes. i have this set. yet it complains.
19:54.13Kobazwhat complains? the gui complains or the asterisk manager?
19:54.18Wellardthe gui
19:54.24Kobazyeah... no idea
19:54.25Wellardasterisk is up and running
19:54.32carrarThen you are good
19:54.37Wellardi just cant get the two to talk
19:54.47carraredit the config files
19:54.53Kobazthe gui has thousands of lines of code, none of which I've seen
19:55.17carrarmore reasons to dump the gui
19:55.18Kobazso the gui is probably not setting up the config properly, or you're missing something in the config
19:55.53Wellardthat's what i'm thinking, just can't work out what. each how-to says 'put enabled=yes in the [general] section of the config'
19:56.02WIMPypicks up his screwdriver...
19:56.04Kobazyou need users also
19:56.11Wellardso, yeah. done. restart. nope. still dont work.
19:56.26Kobazhaving the manager enabled will have it enabled.. but you need to be able to authenticate
19:56.45carrarinstall asterisk now ;)
19:56.55Wellardi have another section in my manager.conf called [admin], with a secret and read/write perms set
19:57.03Kobaz~asterisk-gui
19:57.03infobot[~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0.  For support go to  #asterisk-gui
19:57.32Kobazmm
19:57.59Wellardso until somebody wakes up in #asterisk-gui, i'm momentararly stuck.
19:58.06Kobazit seems so, sadly
19:58.21carrarWhy depend on software you can't figure out?
19:58.43Wellardcrap, think i know why now.
19:58.47carrarWhat is it about the gui that you can't do with a none gui version of asterisk?
19:58.59p3nguinFuck it up.
19:59.02Wellardlooks like i need gui 2.1
19:59.07Wellardfor asterisk 1.8
19:59.18Wellardso yeah, i think i fucked up somewhere :P
19:59.19Wellardlol
19:59.28p3nguinI can't imagine that you *need* the gui.
19:59.48IsUphello
19:59.59carrarWait till gui start changing all your files without you knowing it
20:00.08carrarthat should be fun
20:00.11Wellardno, not need. but for an asterisk newbie, i felt it'll come in handy for now until i'm more comfortable with the config files
20:00.27p3nguinAt that point, it is too late.
20:00.29carrarI would disagree
20:00.41p3nguinOnce you go GUI, you can't go non-GUI easily.
20:00.44carrarI would disagree that the gui will make learning easier that is
20:01.12p3nguinUsing the GUI doesn't help you learn anything but the GUI.  It certainly does keep you from learning how to use Asterisk, though.
20:01.52carrargui is also ok if you don't want to learn asterisk and just need something simple working
20:02.10Wellardexactly, only need simple right now
20:02.13carrarif you can get it working :)
20:02.22Wellardi'm not doing anything complicated with dial plans, trunking, etc.
20:03.01carrarbut somethign simple can easly be done without a gui
20:03.07Wellardokay, to verify manager access is working. how can I do that via the cli?
20:03.26carrartype:  /join #asterisk-gui
20:03.34IsUpi am running asterisk-1.8.2.4 at the moment, compiled from tarball. i want to upgrade to asterisk-1.8.8.0, what should i do?
20:03.35carrarthat should work
20:03.58Wellardcarrar, thanks, i'm already there. everyone is afk.
20:04.19carrarbe patience then
20:04.38carraror pay your support team more
20:05.05carrarIsUp, reading the ChangeLog would be a good start
20:05.18Wellardhmm, think i found a problem.
20:05.40carrarobviously backup everything
20:05.46Wellardasterisk is complaining it cant open the asterisk database, that might be an issue, huh?
20:05.47carrarhave a play to revert back
20:05.50carrarplan
20:05.57IsUpcarrar: ive read all
20:06.18IsUpcarrar: should i run './configure' as usual? i mean should i follow usual process?
20:06.29carraryes
20:06.49IsUpcarrar: does it overwrites all my conf?
20:06.51carrarinstall just like you did before
20:07.01carrarerr compile that is
20:07.02Wellardokay, management interface doesn't appear to be up.
20:07.06carrarsince thats what you are doing
20:07.30carrarWellard, please type all that in the appropriate channel
20:07.52Wellardso, you can't help with the AMI either then?
20:08.06carrarYou want to create yourself a personal AMI login?
20:08.39Wellardi've created an account, but even though i have enabled=yes in the manager.conf, the AMI is not accessable
20:09.00carrarHow are you testing the AMI login?
20:09.16Wellardtrying telnet localhost 5038
20:09.17Nuggettelnet is eeeeeeevil!
20:09.37Wellardroot@asterisk:~# telnet localhost 5038
20:09.38WellardTrying 127.0.0.1...
20:09.38Wellardtelnet: Unable to connect to remote host: Connection refused
20:09.43IsUpargh
20:10.21carrarIs your config even active?
20:10.56carrarIsUp: ./configure does not over write your config files
20:11.06*** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my)
20:11.07IsUpcarrar: okay, thanks
20:11.08Wellardhmm, okay.. think i'm onto something now.
20:11.24IsUpWellard: did you do 'reload'?
20:11.27carrarYour asterisk .conf files that is
20:11.35carrarin the /etc/asterisk directory
20:11.53Wellardyup
20:12.35carrar~book
20:12.35infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
20:13.10Wellardcarrar, if you dont want to help, just say so.
20:13.52Wellardnow it looks like it's a problem with my asterisk configuration. afaik, the manager.conf is correct. yet, the AMI is not available.
20:14.04carrarWellard, I want to help you help yourself
20:14.08carrarso you can learn
20:14.39Wellardokay, figured this bit out.. the 'manager show settings' says AMI: No .. so no, it's not reading the config
20:15.30Wellardthe asterisk account has the correct file permissions to /etc/asterisk
20:15.41Wellardsince i can su to that user and read the config
20:16.57carrarSure is a lot of documentation about AMI in the book
20:18.03Wellardim sure there is, but just a little hint or something to point me in the right direction maybe?
20:18.32carrarindex is already a great place to start
20:18.33Wellardis there part of the configuration that calls the manager.conf? such as include "manager.conf"?
20:18.35carraralways
20:19.08*** part/#asterisk pkzswe (~pkr@178.73.220.38)
20:19.22carrarMaybe even the AMI Quick Start section
20:19.28carrarBut I am only guessing
20:19.52*** join/#asterisk oej (~olle@87.96.134.129)
20:20.27Wellardthats very nice, but i don't actually have the book to hand.
20:20.46carrarWell I can't help you if you can not read
20:20.52carrarbedst of luck
20:20.54carrarbest
20:21.43Wellardyou're not really helping
20:22.03*** join/#asterisk kresp0 (~kresp0@212.59.203.101)
20:22.05carrarYou're not reading
20:23.25carrarA link to a URL of the book online and you are not even willing to bother to look at it
20:23.29carrargood luck
20:24.40kresp0hi florz,all, after some work now im now over a modem-over-voip
20:25.14Wellardshow me the download link. oh wait, gotta get my credit card out.
20:25.22carrarI already did
20:25.25carrarscroll back
20:25.58carrarPay once you find the book usfull
20:26.18kresp0but i didnt manage to use the telecomix nodes, for some reason my ppp logs says that the modem should be dialing but it isnt
20:28.52Wellardhmm, think i found the problem
20:29.25Wellardcrap. after all that.
20:30.21Wellardit's working now. thank for not helping.
20:30.36Wellardissue with permissions on a directory i overlooked.
20:31.30*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
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20:35.53IsUpi am getting "codec_dahdi.c:578 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory" on Asterisk 1.8 startup, any ideas?
20:40.36Wellardi spotted that earlier. have you got the dahdi kernel module installed?
20:41.34IsUpyes, it's installed. anyways, its not important atm
20:47.45*** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au)
20:54.43*** join/#asterisk kresp0 (~kresp0@dialin-212-144-107-150.pools.arcor-ip.net)
21:06.13*** join/#asterisk engrxyz (~qeqweqwe@cpc3-basl7-0-0-cust788.basl.cable.virginmedia.com)
21:14.51*** join/#asterisk kresp0 (~kresp0@212.59.203.13)
21:38.20*** join/#asterisk kresp0 (~kresp0@98.200.217.87.dynamic.jazztel.es)
21:40.01kresp0anyone knows how to do a call forward using an analog phone connected to an spa3000?
21:40.14kresp0(and the spa3k is connected to an *, of course)
21:41.36kresp0i need to be able to redirect any telemarketer to their special extension
21:41.49kresp0from any phone on the house
21:47.14kresp0ok, found it: just a hook flash do it
21:53.54kresp0anyone knows what is the difference between CALLERID(num-valid) CALLERID(num-plan) CALLERID(num-pres) ?
21:53.56*** join/#asterisk singler (~singler@84.15.129.49)
21:54.28*** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de)
21:57.47kresp0about "CALLERID(num-pres)", found this on func. callerid source:
21:57.49kresp000043  * Do not document the CALLERID(pres) datatype.
21:57.49kresp000044  * The name and number now have their own presentation value.  The pres
21:57.49kresp000045  * option will simply live on as a historical relic with as best
21:57.49kresp000046  * as can be managed backward compatible meaning.
21:58.34kresp0ow, it was another option sorry
21:59.21WIMPyYes it has been changed from over simplified to over complicated in 1.8.
22:01.11*** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file)
22:01.11*** mode/#asterisk [+o file] by ChanServ
22:08.05kresp0WIMPy, all: do you know if there is a datatype on the callerid function (* 1.8) to see the caller-id of the original caller?
22:08.44kresp0something that survives forwards
22:11.30kresp0i've got it, just have to make a Set(DB(lastcaller/${CALLERID(num)})=1) when the call arrives the *
22:17.41WIMPyThe caller id should always be that of the caller.
22:19.09WIMPyThe forwarder should appear in (RDNIS).
22:22.25kresp0strange: if i do the forward using a voip phone, ${CALLERID(num)} gives me the original caller, but
22:23.16kresp0if i do the forward using a "hook flash" on the analog phone connected to the spa3k, it gives me the forwarder number
22:23.45kresp0forwarder number = original call recipient, the one who forwards the call
22:24.09WIMPyThey can be different as well.
22:24.57p3nguinLots of packet loss on the ISP today.  Voice calls suck.
22:25.19WIMPyStrictly speaking it's not forwarding, BTW.
22:25.35kresp0WIMPy: yes, they are different: the voip phone do the forward and finish the call on his side
22:27.26kresp0and with the analog phone i have to do one hook flash, dial the redirect extension, wait until the call is stablished,  then hookflash again
22:28.06kresp0and after all of this, i may hung up the phone
22:29.07WIMPySo that's not forwarding at all. That's an attended transfer.
22:29.19kresp0aha, thank you WIMPy
22:30.06WIMPyThat's a new call that has not relation to the first one until you actually complete the transfer.
22:30.20WIMPys/not/no/
22:30.27kresp0ok. understood.
22:30.32p3nguinYou don't necessarily have to use the database for retaining caller id info.  You can use a new variable and make it inheritable.
22:30.58kresp0oh, nice to know
22:31.05kresp0and do you know if there's a way to do direct transfers instead of attended transfers using hook flash?
22:31.22WIMPyThat depends on the device.
22:31.25kresp0i'll try the inheritable variable way, thank you p3nguin
22:31.29p3nguinI use Set(_externalCID=${CALLERID(num)}) and then later use Set(CALLERID(num)=${externalCID}) when dialing out.
22:31.31WIMPySo you have to check its manual.
22:32.04p3nguinSo the incoming cid is captured and passed along to a new channel, then that value is set back to the caller id number when dialing.
22:32.34p3nguinIf you need it to survive more than one level of new channel, use __externalCID instead of _externalCID.
22:33.10WIMPyShouldn;t the callerid change automatically after completing the transfer?
22:33.41p3nguinIf you do your hook flash transfer method, if you just flash again while the new phone is ringing, does the call end or does it transfer?
22:34.26WIMPyI'd try to flash, dial the destination and then hang up.
22:34.33p3nguinIf it ends the new call and you go back to the original call, it sounds like you'll be stuck with an attended transfer.
22:35.03p3nguinIf it connects the original call to the ringing phone before it gets answered, that's a blind transfer.
22:35.17WIMPyThere's probably a configuration option transfer by hangup or something.
22:35.29p3nguinAlternatively, there is asterisk's feature DTMF transfers.
22:35.46kresp0the call ends, because the spa3000 is waiting for more numbers. and when the asterisk pick the call it answers without rings
22:35.46p3nguinNo hook flash required for that.
22:35.54WIMPyDoes chan_sip update the callerid during transfer?
22:36.29p3nguinIt depends on the phones and other channels being used for the call.
22:37.01WIMPyOk, so at least it could work if you're lucky.
22:37.09p3nguinIf you call me and I am using an ATA and a regular phone with caller id on it, it will probably not update when you do a transfer.
22:37.48p3nguinIf you call me on my SIP phone, there is a chance it might update.
22:38.17kresp0WIMPy, it seems that in the case of direct transfers (blind?) chan_sip dont update the callerid. when making an attended transfer, it does
22:38.48p3nguinRemember that caller id should display the information of the person who should be on the other end when I answer my phone.
22:38.49kresp0damit, i'm so sorry about my english
22:38.52WIMPyWith a blind transfer there shouldn't be anythign to update.
22:39.04p3nguinIf you are calling me to transfer someone else to me (attended transfer), I should see that you are calling me.
22:39.15kresp0correct p3nguin, i want that
22:39.21p3nguinThat's default operation.
22:39.37p3nguinIf you call me, I see your cid info.
22:40.26WIMPyAnd as soon as the transfer is completed, it should update on both ends.
22:40.45p3nguinIt depends on the channels and devices in use.
22:40.49kresp0i want the callerid updated on the * !
22:41.08kresp0ok, i'll tell my case: i have an extension to send the telemarketers when they call me.
22:41.28kresp0when someone reaches that extension, their caller id is added to the black list
22:41.35kresp0among other thing
22:41.39kresp0things
22:42.03kresp0if a telemarketer calls me and i'm on the sip phone, there're no problem:
22:42.25p3nguinYou should blind transfer the caller to the blacklist extension.
22:42.27kresp0just press forward, press the magic numbers and forward again. i may hung up and forget
22:42.36p3nguinForward?
22:42.41p3nguintransfer
22:42.52p3nguinForward is something else.
22:42.56p3nguinYou're doing a transfer.
22:43.02kresp0the phone says "forward"
22:43.06p3nguinWeird.
22:43.18p3nguinMust be made by someone who does not speak English.
22:43.57WIMPyWell, what's called forwarding on the SIP phones isn't forwarding, either.
22:44.12p3nguinWhat would you call that?
22:44.33WIMPyIt's called Call Deflection.
22:44.48WIMPyForwarding is a swtich/server based feature.
22:44.55p3nguinI see.
22:45.17WIMPyAnd both are called Call Diversion.
22:45.40p3nguinSo if the dial to my phone is preempted by a dial to another phone, that's call forwarding?  But if I use my forward key to redirect the call somewhere else, that isn't call forwarding?
22:46.32WIMPyYes
22:46.54WIMPyCF could be somethig you set up via feature codes in AstBD or something.
22:46.57p3nguinI actually have both methods available to me.  I have *72 for forwarding via dial plan, and I also have forward keys on my phones.
22:47.42p3nguinI have a conditional Dial before the Dial for my phone.  If the forwarding is enabled, the conditional Dial() runs, calling the number set.
22:48.12p3nguinIf forwarding it not enabled, the conditional dial never runs, so my phone is called.
22:48.14WIMPyCFU
22:48.30p3nguinU?
22:48.37WIMPyUnconditional
22:48.38p3nguinoh
22:48.51p3nguinYeah, unconditional call forwarding.
22:49.14p3nguinI never got around to implementing the other types of forwarding.
22:49.32WIMPyhas got them all.
22:49.33p3nguinI've never really needed it since the dial plan after Dial(myphone) takes care of most of it.
22:49.46WIMPyIt's a long repetitive dialplan.
22:50.07p3nguinIf my phone rings and I do not answer it, it'll go to unavailable voice mail.  If my phone is busy, it'll go to busy voice mail.
22:50.18WIMPySure, but you might want to make it user configurable that way.
22:50.43p3nguinI use a single routine for that, and then a subroutine to actually dial the devices, which returns back to the main part after the Dial() exits.
22:51.03p3nguinOther wise I would have a lot of duplicated dial plan.
22:51.13p3nguins/Other wise/Otherwise,/
22:51.30kresp0p3nguin, WIMPy: sorry, sorry i was wrong (again)
22:51.31kresp0on the sip phone says "transfer" not "forward"
22:51.34WIMPyThat't why I use (user changable) forwarding stored in AstDB.
22:51.56kresp0i suppose that doesnt matter now, but just in case
22:52.16p3nguinBack to your issue... configure blind transfer by DTMF in features.conf.
22:52.35p3nguinThen you can blind transfer your caller to the blacklist extension by dialing some numbers.
22:52.50kresp0thank you p3nguin, that should work too
22:52.52p3nguinno hook flash, no attended transfer.
22:53.01kresp0much better
22:53.04*** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net)
22:53.47p3nguinYour transfer key on your IP phone should do both attended and blind.  If you press transfer a second time immediately after you hear the ringing begin, it should blind transfer.
22:54.11p3nguinIf you wait for an answer, it will complete the attended transfer.
22:54.21p3nguin(the same as your hook flash method)
22:54.26WIMPyI guess that's what some ppl call a half blind transfer.
22:55.00p3nguinProbably.  I seem to remember seeing both types of transfer buttons on my phone in the past.
22:55.19p3nguinI just don't remember for sure, and I don't remember how they were labeled.
22:55.52WIMPyI'm not sure I've ever seen a real blind transfer.
22:56.04luckman212anybody know why I'm getting a "SIP/2.0  489 Bad Event" error back from Asterisk (1.8.8.8) when my Polycom 650 tries to subscribe to MWI ?   I made a pb of the sip debug .... http://pastebin.com/6Tdd3Npw
22:56.24luckman212the exten is in the "[default]" vm context
22:56.40WIMPyDon't use the default context.
22:57.00WIMPyDid you allow subscriptions?
22:57.22WIMPyIt says "Looking for 703 in from-internal".
22:57.33luckman212hmm...   not even sure how to not allow subscriptions
22:57.54luckman212WIMPy: is that wrong?
22:58.05p3nguinIt seems like I had to put the call on hold, enter a new number to dial, then hit the blind transfer key.  I never heard any ringing during the process.
22:58.06WIMPyallowsubscribe
22:59.17luckman212I'm sure I allow subscriptions,  I have the phone subscribed to certain DEVSTATE hints (e.g. DND  and I get BLF for the other extens, so that's working
22:59.28p3nguinI don't see any problems in that pastebin.
22:59.40luckman212should it be Looking for 703 in 'default' ?
22:59.49p3nguinIt isn't.
23:00.14WIMPyNo, you should configure it in from-internal.
23:00.28WIMPyDon't use default unless you know exactely, you have to.
23:00.31*** part/#asterisk Wellard (~Wellard@2001:470:1f11:cf8:60b7:88c:9498:8fa7)
23:00.33p3nguinI'm guessing the from-internal name is your vm context.
23:00.33luckman212the problem is, when I set "subscribecontext = default" in my sip.conf  then my DND, FollowMe, NightMode etc (all the hints) stop working
23:02.38p3nguinYou can't do one thing one way and another thing another way -- it is a system, which must be configured as such.
23:02.38p3nguinsubscribecontext is for hints, right?
23:02.38luckman212p3nguin: yes, when I change that, the hints stop working
23:02.38WIMPyyes
23:02.38p3nguinThat has absolutely nothing at all to do with mailbox subscription.
23:02.38p3nguinYou set your subscribecontext to whatever context in your dial plan that has your hints in it.
23:02.38luckman212I think the reason for that is because this phone does not have it's 'context=' line set to from-internal in sip.conf
23:02.38p3nguinNo, the reason is that you don't understand what you're doing.  I want to help you understand it.
23:02.48p3nguin<p3nguin> You set your subscribecontext to whatever context in your dial plan that has your hints in it.
23:02.51luckman212it's set to something else, so I can dial sip URIs (it hits a context called 'enable-sipuri-dialing' first, then if its not a SIP URI, it passes the call to from-internal
23:02.53p3nguinStart with that part.
23:03.20p3nguinMine is 'devices' context.
23:03.37p3nguinOnce you have that done, we'll move on to the next step.
23:03.44*** join/#asterisk fisted_ (~fisted@unaffiliated/fisted)
23:04.14[TK]D-Fenderfreepbx = from-internal
23:04.38p3nguinSome people use it as an arbitrary dial plan context.
23:04.46luckman212when I run a 'core show hints'  I get a bunch of different contexts I think...  they aren't all the same
23:05.02p3nguinYou can have hints in various places if you want.
23:05.04[TK]D-FenderAlso phones do not subscribe to MWI.  It doesn't work on a subscription basis .
23:05.10[TK]D-Fender* sends to phones regardless based on setting the mailbox
23:05.13p3nguinSince when?
23:05.22[TK]D-FenderAnd the most comman FUBAR on that is the VM context
23:06.19p3nguinMWI has always been subscription based.  If you do not configure a mailbox, and the phone tries to access MWI events, asterisk will spew out a message that "peer whatever is trying to subscribe to mailbox something..." if I remember right.
23:06.50p3nguinI could be losing my mind, but I'm sure that's what it said.
23:06.52WIMPy"Received SIP subscribe for peer without mailbox: $peer"
23:06.59p3nguinThat's it!
23:07.17p3nguinI knew it was something with the word "subscribe" in it.
23:07.37luckman212shit our guests just arrived
23:07.51p3nguinBetter hurry.
23:08.00WIMPyTell them to come back later.
23:08.11p3nguinor give them a keyboard and tell them to help.
23:08.49WIMPyDepends on how trustworthy they are.
23:09.07luckman212dont trust anyone
23:09.37WIMPyThen you shouldn;t really let them in in the first place.
23:09.59p3nguinUgh.  My packet loss is so bad that my ssh session keeps freezing up while running mtr to check the packet loss.
23:11.40p3nguinTo one destination, 6-10% on every hop but the first one (my own router).  To another destination, 3-6% on every hop but the first one.
23:12.02p3nguinEven the hop from my router to the ISP's first router has way too much packet loss.
23:12.13p3nguinI wonder if someone cut the fiber again.
23:16.01filedon't look into the fiber to check
23:16.47Kobazanyone want a sangoma a101d pci
23:17.20WIMPyI never tried Sangoma.
23:17.26Maliutawant vs. need vs. can use
23:17.37Kobazwant/want to buy
23:17.42MaliutaI always want toys
23:17.55WIMPyis also collecting toys.
23:18.22Maliutawhether I need them or will use them beyond initial conf/testing/playing is another thing
23:18.24WIMPyAnd compare them.
23:18.40Kobaz$497 on voipsupply, i can do $400, it hasn't been used much
23:19.12WIMPyHe, I get 4-port cards for one hundred.
23:19.14Kobazthey are really nice cards, the debug data you get is fantastic, we've been moving to move voip
23:19.18WIMPyor got
23:19.29MaliutaKobaz: by "much" you mean you only ran the 1000v 10amp circuit through it for less than 60 seconds? ;P
23:19.42Kobazmoving to move voip rather
23:19.44WIMPyLOL
23:19.50Kobazdamnit
23:19.53Kobazmoving to *more*
23:20.00KobazMaliuta: heh no, the card works
23:20.49WIMPySangoma seem to be something ppl don't want to get rid of. Rarely see them on ebay.
23:21.10Kobazyeah there's two
23:22.12WIMPyHFC cards are also rare.
23:22.23WIMPyBut lots of Digium stuff.
23:24.04MaliutaI'm happy with my Digium tmd400
23:24.48WIMPyI don't collect antiques.
23:27.06*** join/#asterisk kotis_ (~kotis@udp019436uds.hawaiiantel.net)
23:27.41WIMPyActually that's a lie. I do collect antiques. I've got soem E1 NAS.
23:27.45*** join/#asterisk kresp0 (~kresp0@98.200.217.87.dynamic.jazztel.es)
23:37.32p3nguinWhat kind of $100 card did you get?
23:38.12WIMPyTE407P, but 101EUR
23:43.00p3nguin~change 101 EUR to USD
23:43.16p3nguin:(
23:43.24WIMPyxe.com
23:43.40*** join/#asterisk kresp0 (~kresp0@177.169.16.95.dynamic.jazztel.es)
23:43.59p3nguin~excuse
23:43.59infobotmethinks excuse is http://www.randomexcuse.com/
23:44.30p3nguin101.00 EUR = 131.790 USD
23:44.43p3nguinNot bad.
23:44.48kresp0how should i put an * console on tty12 ? i've tried adding "asterisk -rvvv" to /etc/inittab and the process is launched on boot, but i only see a blinking cursor when doing ctrl+alt+f12
23:45.15WIMPykresp0: openvt
23:45.36p3nguinI would probably run a getty on that tty and run asterisk -r there myself.
23:45.43p3nguinasterisk -R even.
23:45.55kresp0thank you again WIMPy, thats just what i need :D
23:45.57WIMPyprefers rasterisk -R
23:46.30WIMPy... so I don't kill the remote console when I have to killall -9 asterisk.
23:47.16kresp0wow killall -9, poor *
23:47.36WIMPypoor me!
23:47.40kresp0xD
23:47.43p3nguinIt didn't stand a chance.
23:47.54WIMPyIt should behave.
23:48.08p3nguinSometimes you have to beat it with a big stick.
23:49.33kresp0and WIMPy, what is the difference between "rasterisk -R" and "asterisk -R" ?
23:49.41WIMPyNone
23:49.45p3nguinmore typing
23:49.57WIMPyIt's jst that it isn't hit by the killall.
23:50.35WIMPyYes, more, but less more than having to restart it.
23:50.35p3nguinIf you run asterisk -R, doesn't it launch rasterisk -R?
23:51.06WIMPyIt's only one binary.
23:51.17p3nguinI'd like to know how I can make -R wait for more than 30 seconds.
23:51.36WIMPyPatch it.
23:51.50p3nguinI guess I should look for the code and do that.
23:52.07WIMPyHowever there's a cath again. You can;t kill it while it tries to reconnect.
23:56.46p3nguinAny idea what file that would be in?
23:59.39*** join/#asterisk pc500 (~kvirc@96.18.214.187)
23:59.42WIMPyasterisk.c

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