IRC log for #asterisk on 20110717

00:02.03*** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net)
00:08.28*** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net)
00:10.18*** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net)
00:16.44ChannelZdang something is missing here... half my 'core' commands are gone
00:18.25WIMPythought the idea of the 'core' commands was that they will allways be available.
00:23.01dymsame here.
00:25.11ChannelZseems my codec_g729 is hanging the load process which was the problem.  hmm.
00:26.34ChannelZstill can't get gtalk to work
00:26.34dymDoes anyone know of a decent softphone (os x) that can have multiple sip accounts active?
00:26.44dymor do i actually have to install an asterisk locally? :/
00:30.01*** join/#asterisk digilink (~digilink@unaffiliated/digilink)
00:35.16p3nguindym: Is zoiper or twinkle available for OS X?
00:35.34dymyou tell me, cutiepie
00:35.36dym:)
00:35.56WIMPyzoiper is
00:36.16dymi really like sjphone
00:36.22dymso slim and decent
00:37.32p3nguinI'm not sure why it's my responsibility to find software for you.
00:37.59dymits not :|
00:38.18p3nguinBut the two suggestions I gave you allow multiple active accounts.
00:38.42WIMPyBut zoiper is limited to two in the free version.
00:38.46dymp3nguin: thanks. im trying one out
00:46.34dymp3nguin: thanks for the suggestions. clients are too colorful.
00:47.35WIMPySet your deskto to greyscale.
00:47.37WIMPyp
00:50.42dym:D
00:50.43dymdoubt it
00:51.29p3nguinYour requirement was to have multiple accounts active.  You didn't say anything about the interface must not have colors.
00:52.26dymp3nguin: why the tenseness? (:
00:52.32dymall peachy
01:05.06*** join/#asterisk techknowlust (~patrick@li257-211.members.linode.com)
01:05.36techknowlustI'm having trouble with 486 Busy Here errors when internal SIP calls are answered
01:05.50techknowlustrunning asterisk 1.8.5.0 with very basic dialplan
01:06.07*** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net)
01:09.41*** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45)
01:30.07p3nguinno supper = room for beer
01:31.25johnnyasteriskroom for beer + lots of beer - supper = hangover
01:35.07p3nguinI think I saw some pretzels, so I can snack on those if I get hungry later.
01:49.55*** join/#asterisk joker2u (~root@li345-191.members.linode.com)
01:51.47joker2uwhat directory are the asterisk modules located?
01:53.47p3nguinprobably /usr/lib/asterisk/modules
02:02.34joker2up3nguin thank you
02:06.03techknowlustI'm getting 486 busy here errors with internal sip calls between softphones. Any ideas as to how I can solve this ?
02:06.17techknowlustreally simple setup. not sure what's causing this
02:06.32p3nguinYou'll need to check the sip debug.
02:06.44techknowlustok.
02:08.10techknowlusthttp://pastie.org/2224962 that's the sip debug log for the call
02:11.09techknowlustsome 404's and 401's in there which confuse me
02:13.32p3nguinI don't see any "486 busy here" in there at all.
02:13.54techknowlustthe 486 was shown in the core verbose rather than sip debug
02:16.28techknowlustbasically the call dies as soon as the other end picks up
02:21.24joker2uwhich asterisk module effects call forwarding?
02:22.04p3nguinforwarding of what?
02:22.10p3nguina SIP call?
02:23.50joker2up3nguin just like an edit in extension.conf. forward to a particular DID.
02:24.28p3nguinYou aren't using the terminology in a manner that makes any sense to me.
02:24.36joker2up3nguin app_exec.so???
02:24.54p3nguinYou don't forward calls to DIDs.  Calls are received and extensions run commands.
02:25.09WIMPyDoesn't make any sense to me, either. What do yu want to do?
02:25.19WIMPy+o
02:26.16p3nguinI'm guessing he's just trying to receive a call, but without a sensical expression, it's nothing more than a mere guess.
02:27.08WIMPydoes not feel like being able to make a guess (yet).
02:27.49joker2uI am working with py-asterisk and the documentation is very limited. Very simple. I have and extension that via the api I want to be able to 'send calls for the extension' to voicemail via agi
02:28.19p3nguinThe extension would execute VoiceMail() to send a call to voice mail.
02:28.34ChannelZarf
02:28.56joker2uI found the voicemail module but forwarding it back to a DiD I can't see what module would be used?
02:29.17dym04:24:53 < p3nguin> You don't forward calls to DIDs.  Calls are received and extensions run commands.
02:29.39WIMPyWTF do you mean by "back to a did"???
02:30.06joker2uWIMPy because choice is an outside line. not the box.
02:30.16joker2uchoice #2
02:30.34WIMPyYou want to send it to an external voicemail?
02:30.42p3nguinIt's almost as if he's just regurgitating terms he's heard or read somewhere, without any context for their usages.
02:31.02joker2uWIMPy voice mail or a pstn number (DiD) those are the options.
02:31.32WIMPyWhere?
02:31.38joker2up3nguin DiD = dial in direct.
02:31.45p3nguinwrong
02:31.54p3nguindirect inward dialing
02:31.57p3nguinclose, though.
02:32.18p3nguinI'll give you half a point for a good try.
02:32.29radenwhy is it soooo freaking hard to get away from windows
02:32.30joker2up3nguin it's a phone number, lordy.
02:32.36radenI wish this freaking world would embrace linux !!!
02:32.43WIMPyraden: Is it?
02:32.45p3nguinraden: I haven't used Windows in 10 years.
02:32.58WIMPyI have never used Windows.
02:33.05p3nguinSeems easy enough, then.
02:33.11joker2uwhat module would edit the dial plan?
02:33.15dymp3nguin: lovely :D
02:33.23ChannelZBarf.  Gtalk just does not want to work.  I see the jabber messages on the console with debug on but * just seems to ignore them or something.. no errors, but nothing is happening
02:33.23p3nguinpbx_config.so
02:33.29WIMPyBut I know people who did, but haven't done so for years.
02:33.49radenp3nguin, I have a company that insists all cad work be done in alibre cad
02:33.53WIMPyjoker2u: None. Use vi or whatever.
02:33.58radenp3nguin, try running quickbooks in wine :(
02:34.06radenp3nguin, I hate windows
02:34.09radengrrrrr
02:34.15p3nguinWhy would I want to use quickbooks?
02:34.29radenjust the whole business world very hard to get away from windows :(
02:34.30radensucks :(
02:34.31joker2uWIMPy vi? how in the world do you script vi?
02:34.48p3nguininsert mode
02:34.51ChannelZEvery time you use vi, another unicorn dies
02:35.27radenLMA O
02:35.35WIMPyjoker2u: Write a script to generate (part of) your dialplan.
02:35.51WIMPyI guess most of us do something like that.
02:35.51radenhow stable is 11.4 ?
02:35.55p3nguinIt's not that hard.
02:35.55radenon 11.1 at moment
02:35.57p3nguinor just type the entire thing yourself.
02:36.06WIMPyOr use realtime.
02:36.07p3nguinWindows 11.4 should be fine.
02:36.16joker2uI'll stick with python.
02:37.07joker2uvi isn't 'class' compatible. it's flat, won't work.
02:37.28p3nguinAsterisk 11.4 should be fine.
02:37.42p3nguinQuickBooks 11.4 should be fine.
02:37.55p3nguinVIM 11.4 should be fine.
02:37.58radenp3nguin, opensuse 11.4 dork
02:38.10dym(:
02:38.21p3nguinYou expected me to know you were talking about openSUSE, and you call me a dork?
02:38.21dymraden: why in gods name would you use suse?
02:38.43WIMPyLuckily I haven't seen suse for many years. But mabe it has become usable in that time.
02:38.57dymor increasingly homosexual
02:39.21p3nguinI've installed openSUSE for a few people, and I didn't find much wrong with it.
02:39.51dymdont like yast
02:40.09p3nguinuse zypper
02:40.10WIMPynoone like yast
02:40.14dymaptitude.
02:40.30radendym, its alot better than it was in 10.x
02:40.32p3nguinyast install aptitude?
02:40.41WIMPyBut whe I had contact, they always used very custom kernel, often rather unstable.
02:40.51dym:D
02:40.54dymgood one p3nguin
02:41.07p3nguinaptitude install archlinux
02:41.14dymall within suse, right?
02:41.15dym:P
02:41.52dymgawd, just picked up someone from downtime. reminded me how much i hate drunk kids
02:41.58*** join/#asterisk fulcan (~root@li345-191.members.linode.com)
02:42.02dymdowntown* crap - sysadmin kicking in
02:42.03WIMPyis planning to try lubuntu on his netbook.
02:42.12dymWIMPy: good choice
02:42.21dymwait lubuntu?
02:42.27dymWhat Window Manager is that?
02:42.37dymah, light
02:42.37WIMPyNFI
02:42.40p3nguinlxde?
02:42.44WIMPyI'd expect to get a choice.
02:42.47dymmhh
02:42.48dymdoubt it
02:42.56radenI cant believe mac is actually making a comeback in the OS wars
02:42.57p3nguinYou don't get a choice.
02:42.59dymxfce prolly
02:43.01p3nguinYou get what you get, but then you can change later.
02:43.03WIMPyBut it's likely defaulting to lxde.
02:43.20dymraden: im a os x user
02:43.22WIMPyThat's choice enough.
02:43.26dympretty sarisfies one too
02:43.32dymlooks like gnome
02:43.41dym(lubuntu)
02:43.49radendym, how u like it ?
02:43.50p3nguinIt's still beyond me why anyone would want to use any buntu.
02:44.00dymraden: i fell in love with it a few years ago
02:44.10dymi mean come on - it has a native bash
02:44.14dymwhats not to love?
02:44.16radenp3nguin, why you say that p3nguin
02:44.23radendym, that is true
02:44.33dymp3nguin: why is that?
02:44.36radendym, id use it if it wasnt so freaking hardware specific
02:44.46dymraden: its all about the kernel
02:44.48dym;)
02:45.01WIMPyA standard desktop is absolutely inappropriate for portable devices.
02:45.07p3nguinAlmost everyone I know who has used other distros as well as buntu distros makes fun of buntu distros.
02:45.20dymWith what reason?
02:45.30ChannelZHmm.  Jingle is no longer used for gtalk in 1.8 is it?  The wiki seems to only talk about the chan_gtalk and jabber.
02:45.52WIMPyIt accesses the harddrive every few seconds, even when doing nothing.
02:45.59radenp3nguin, there is a lot of stupidity in buntu
02:46.11WIMPyfinds that completely inacceptable.
02:46.20dymbuntu is what it is - linux for monkeys
02:46.32dymbased on a first class distro tho
02:46.36WIMPyBut I don't care to trace the offenders down.
02:46.58dymWIMPy: rocking the iostat
02:47.13WIMPyDoesn't show anything.
02:47.49WIMPyI guess that meas something is contantly forking processes or threads that die immediately again.
02:48.23dymps aux / top ?
02:48.27dyml2debug
02:48.47*** join/#asterisk niekie_ (quasselcor@CAcert/Assurer/niekie)
02:49.07*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
02:49.08WIMPytop isn't informative, either.
02:49.16p3nguinHow about iotop?
02:49.18dymWIMPy: sorted right it probably is
02:49.41WIMPyNo, I've been the way up to iotop.
02:49.58dymThen, dear friend, you shall suffer.
02:50.32WIMPyThat's why I want to try some stripped down distro.
02:50.34p3nguiniotop -aoP
02:50.42*** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk)
02:50.47dymThats a selfdestroy command
02:50.49dymobv.
02:52.22dymGod im Bored. Time for some prank calls
02:52.35johnnyasterisklol
02:53.03johnnyasterisksure prank calls.... the ones that cost $2 per min ;-)
02:53.10dymnah, local
02:53.15dym<3 call files
02:53.56johnnyasterisknow thats not really a great use fort asterisk is it
02:53.58johnnyasterisklol
02:54.13dymcourse it is.
02:54.42johnnyasteriskwe have our own auto dialer which uses asterisk
02:54.50johnnyasteriski prefer using originate ;-)
02:55.07dymthats deprecated since 1.8
02:55.14dymisnt it?
02:55.29WIMPyrenamed
02:55.30dym"core dial"
02:55.44WIMPychannel originate
02:55.47dymorly
02:55.51johnnyasteriski am still using 1.2.27
02:55.54johnnyasterisk;-)
02:56.03WIMPyurgs
02:56.11johnnyasterisklol
02:56.13dymoh dear, johnny*
02:56.47johnnyasteriskhave made it to 1.4.40 on another server but took too long to change some of the customizations
02:56.59johnnyasteriskalso found huge problems with reload on 1.4
02:57.09johnnyasteriskwhich prevented moving to it sooner
02:57.37dymJABBER: Keep alive packet <-- really annoying
02:57.45dymon what debug level are those msgs supressed?
03:00.36ChannelZI think with jabber debug on, you just get everything
03:00.39ChannelZthat's the point
03:01.01*** part/#asterisk joker2u (~root@li345-191.members.linode.com)
03:01.16ChannelZThis is driving me crazy.  I can call out with gtalk and it works, but if I call in with gtalk nothing happens.
03:01.33ChannelZno make-a sense-a
03:01.54dymOH
03:02.05dymthanks ChannelZ - forgot about "jabber debug"
03:02.31ChannelZInterestingly I don't get these keepalive packets you speak of
03:04.02dymi get that in debug
03:04.09dymbut im connected to my own jabber server
03:04.15dymnot to gtalk
03:04.16ChannelZhow often?
03:04.22ChannelZoh
03:04.26dymquite frequently
03:05.32ChannelZI must have something set wrong.  I get a bunch of jabber messages when I call in but asterisk seems to just do nothing.
03:05.55dymNo idea what you are trying to do. i use jabber to notify on calls.
03:08.52ChannelZI'm trying to use Google Talk to call in my Asterisk
03:09.12ChannelZnot GV via a phone number, but like voice chatting a contact from gmail
03:10.09dymSo Google Voice -> Local Asterisk Number?
03:12.08ChannelZmostly
03:12.17ChannelZNOT
03:12.26ChannelZGoogle Talk, not Voice
03:12.36ChannelZIE not a regular phone number involved anywhere
03:14.58ChannelZMaybe it can't work with Talk.  It used to.
03:16.28ChannelZWait a minute.  It works with the standalone Google Talk client but not the web browser one.
03:26.31ChannelZthis is annoying
03:28.36*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
03:42.59*** join/#asterisk OldMonk (~raju@59.178.175.121)
03:43.03OldMonkhi
03:44.02ChannelZOH HAI
03:44.52OldMonki have some 12 PRIs connected to an asterisk server using redfones.  all the channels are in a single group.  when dialling, i use something like: Dial(DAHDI/g1/${EXTEN}), which sends the call to the first free channel in the lowest-numbered PRI channel
03:45.07OldMonkis there any way to load balance the calls across the PRIs?
03:45.08*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
03:45.53OldMonkso that the first 12 calls, e.g., go to the first channel of each PRI in turn, the next 12 to the second channel of each PRI, etc?
03:51.34ChannelZnot without manually writing that logic into the dialplan
03:51.36ChannelZbut who cares?
03:51.56OldMonk?
03:52.19ChannelZIE what are you really 'balancing'?
03:52.43ChannelZIt's not like the PRI gets clogged if you use a bunch of channels on one
03:53.18OldMonkwe're facing problems with congestion, and the service provider is asking if we can do this
03:54.09OldMonkok, where would you get a situation where the sum total of in-progress + dialling calls exceeds 30 on an E1?
03:54.14OldMonkwe're seeing that regularly
03:54.33ChannelZGuess I don't understand the finer points.  They're basically guaranteed bandwidth, what good is a PRI where you can only use a few channels
03:55.16OldMonkto repeat, how the F**K can you have more than 30 calls on an E1?
03:55.51p3nguinuse another E1
03:55.52*** part/#asterisk fulcan (~root@li345-191.members.linode.com)
03:56.11ChannelZI was under the impression you had multiples
03:56.16OldMonkwe do
03:56.29p3nguinProblem solved.  NEXT!
03:56.38OldMonkbut at times we see >30 calls on a single E1
03:57.01ChannelZscratches his head
03:57.10OldMonkp3nguin: it's asterisk that is assigning those calls to the E1
03:57.32OldMonkIOW, problem not solved, next problem may kindly await its turn in the queue
03:57.55ChannelZBut again, so what?  If they're in one group and you're dialing the group, it should go into the next available channel on the next E1...
03:58.38OldMonkto re-repeat, asterisk is assigning >30 calls to a single E1.  under what circumstances would this happen?
03:59.16ChannelZit's not possible (sans bug).  It sounds like maybe your group is bogus
03:59.31ChannelZwhat do you mean "assigning > 30 calls"
04:00.02OldMonksum(in-progress calls + dialling calls) > 30
04:00.14OldMonki don't know how else to state it
04:01.44ChannelZThe question doesn't make sense.  I am getting the feeling this has something to do with these "redfone" devices
04:02.04*** join/#asterisk joker2u (~root@li345-191.members.linode.com)
04:02.20ChannelZwhich I know nothing about what they do or how they work so I'm out
04:02.32p3nguinI don't know that dahdi can actually put more calls on a "line" than there are channels.
04:02.42OldMonkp3nguin: precisely
04:02.56p3nguinSo then it's a non-issue and the problem doesn't exist.
04:03.18OldMonk...except that it's happening in front of my eyes
04:03.27ChannelZIt would help to know what your actual problem is.
04:03.43ChannelZYou get back 'CONGESTION' when trying to place a call?
04:03.49OldMonkbut sure, if you feel that o5strich is a better nick, go right ahead ;)
04:04.42OldMonkChannelZ: i don't know what the precise return code is, but here're the symptoms:
04:04.48joker2ucan anyone help me understand astobj2.h? this is the class object for the api for ALL of the asterisk internal objects? this is what py-asterisk connects too?
04:05.11OldMonk1. when my callers try to place a call they get a congestion tone (very fast beep for a few seconds, then disconnect)
04:05.42OldMonk2. at that time, when i count calls on a per-E1, i see some E1s that have more than 30 calls on them
04:06.13OldMonk3. number of in-progress calls is always <30, but sum(in-progress + being dialled) calls >30
04:06.16OldMonk.
04:06.18ChannelZHow do these redfone devices actually connect to your Asterisk such that you're using DAHDI?
04:06.26OldMonkChannelZ: TDMoE
04:06.52ChannelZit sounds like their driver is broken and is somehow screwing up channel grouping or something
04:07.59OldMonki don't think there's a separate driver: it uses whatever passes for the stock DAHDO TDMoE driver
04:08.03ChannelZIf the entire system is idle, if you dial DAHDI/36 does it go out the second E1?  (IE do your DAHDI channels truly correspond 1:1 with your E1 channels)
04:08.07OldMonks/DAHDO/DAHDI/
04:08.56OldMonkChannelZ: i didn't know you could specify a channel in a dahdi dial command... i specify channelgroup
04:09.20ChannelZEvery channel is addressable
04:09.48OldMonkok, haven't tried that then
04:10.09ChannelZand a channel group is simple a range/collection of channel numbers, which is why we keep saying you can't have "more than 30 calls on an E1"... Asterisk just uses as many channels as you tell it to
04:10.35OldMonkChannelZ: then HTF can sum(....) be >30 ?
04:10.43ChannelZWhich is why I think this virtual DAHDI driver is at fault, not handling all of the actual PRI channels correctly
04:10.59OldMonkincidentally, the problem is intermittent
04:11.01*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
04:11.21OldMonkmost of the time it just works beautifully
04:12.29*** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net)
04:12.35ChannelZI'd start rattling redfone's cage.  Either your DAHDI config is not quite right and your channels aren't grouped properly, or something is going on beyond Asterisk's control
04:12.53OldMonkholdon, let me paste chan_dahdi.conf
04:14.07ChannelZWhen you dial the group, it should come back and say something like "-- Called DAHDI/g1/5551212" and then " -- DAHDI/X-X making progress" or whatever -- the X-X being the actual channel number it's using
04:14.20OldMonkhttp://pastebin.com/uxBzE0Nu
04:14.24ChannelZ(for me it says 'answered' because I'm using POTS)
04:15.07*** join/#asterisk Mango (~Mango1234@S010620cf30c62cb6.vc.shawcable.net)
04:16.03ChannelZWell that looks normal enough.. channels 125 through 619 (mostly) in group 1
04:18.48ChannelZwonders what the maximum number of channels possible even is or where it's defined...
04:18.55OldMonk1024
04:19.27ChannelZhmm
04:19.54OldMonkfound out the hard way, by trying to assign more than that to a single dial server
04:20.10ChannelZAre you using mixmonitor or chanspy or conferences or anything?
04:20.21OldMonkchanspy, but on a different server
04:20.51ChannelZDoing some random googling I see some references to 512 psuedo-channels
04:24.41*** join/#asterisk nix8n82 (~nate@24.143.28.16)
04:26.22*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
04:27.01OldMonkguess i have to look into the detailed channel allocation to figure out what's really happening.
04:27.05The_REVPlease Please Please does anyone know a way to export extensions.conf from a box running freepbx so I can rebuild this shitty ox and run is with out freePBX
04:27.42ChannelZfreePBX uses realtime which means it's in the SQL database
04:28.04The_REVright... and I want to covert it to a static so I can rebuild...
04:28.13The_REVstupid SQL fucking things up
04:28.29OldMonkobviously, if there're >30 calls on an E1 then some of the channels are being reused; i'd presume that asterisk is trying to dial out on a channel that already has a call in progress.  the question is, why?
04:28.56ChannelZjust note that even if you get it exported you're going to have tons of crazy FreePBXisms (macros and stuff)
04:29.45ChannelZthat said I'm not sure if there's any tools floating around that'll do it
04:30.06The_REVi can clean up crazies...
04:30.08The_REVdamn...
04:30.16ChannelZOldMonk: incorrect status by the TDMoE driver?
04:30.37OldMonkChannelZ: c'est possible
04:30.48OldMonkor maybe incorrect status from the provider
04:31.11ChannelZAll asterisk knows about is a big pool of channels.
04:31.46OldMonkChannelZ: it seems to be either getting the wrong channel status, or misinterpreting
04:32.07ChannelZIt tries the next available one when you use a group.  Maybe one of the channels somewhere in the middle is dead, or returning an error for some reason. You'll really need to look at PRI debug to figure out what precisely is going on
04:32.09OldMonkif the first, then i need to figure out where that incorrect status is originating
04:32.34OldMonkif the latter then it's probably a bug
04:32.58OldMonkand fscked if i can understand pri debug
04:33.29OldMonkanyhow, the people over at redfone can handle that part
04:34.39OldMonktomorrow's a calling day, will investigate
04:34.41OldMonkthanks folks
04:35.27ChannelZThat was interesting
04:37.33MangoI have a phone behind NAT connecting to an Asterisk server outside NAT.  I want direct media between the phone and the carrier, so I've set up Handle VIA Received on the phone.  This works, but it breaks local calling between phones behind the same NAT.
04:37.45MangoIs there any way to fix that without port forwarding?
04:38.34ChannelZDo you have control over the Asterisk server?
04:38.40MangoYes.
04:39.21ChannelZOh.. wait nevermind
04:39.45MangoI could pretend it's someone else's Asterisk server if that would help :P
04:40.47ChannelZno I was just thinking wrong outloud
04:41.07Mangohehe
04:42.25ChannelZI guess if the device doesn't have the concept of an internal and external network in order control when it lies about its IP and when it shouldn't, I'm not sure.
04:43.58MangoThe weird thing is the phones only seem to respect the received= header when it comes from the Asterisk server, not when it comes from each other.
04:45.07MangoSPA921
04:46.29ChannelZhmm
04:51.28*** join/#asterisk godmachine-x6 (~godmachin@h214.179.90.75.dynamic.ip.windstream.net)
04:59.36ChannelZhmmmm.
05:00.20ChannelZSoooo the Google Talk in-the-browser is still referencing jingle
05:01.53DrDigitali need to find these riser cards for this server http://www.weirdstuff.com/cgi-bin/item/62043  mines just a tad slower
05:02.20DrDigitalahh and i dont have an intel motherboard either
05:02.30DrDigitalbut the case is almost identical i think
05:06.37ChannelZhmm this is confusing
05:34.30*** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593)
05:56.30*** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net)
05:57.53sawgoodIs Skype for Asterisk worth the $66 dollar per channel license?  I was thinking of buying 5 channels before the experiation date ...
05:58.24p3nguinIf you're doing it for the novelty value, you should probably save your money.
05:58.52sawgoodyeah ... that was the case ... just to have it working before they cut it off
05:59.24sawgoodThen after the cut-off allow other Asterisk boxes to 'proxy' through the SFA box with the licenses ...
05:59.27sawgoodIf that was possible
06:04.58*** join/#asterisk deltaflyer4747 (5ef24b32@gateway/web/freenode/ip.94.242.75.50)
06:05.03deltaflyer4747morning
06:23.44*** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.109)
06:23.45*** join/#asterisk boazb (~b@bzq-82-80-219-90.red.bezeqint.net)
06:30.38*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
07:00.52deltaflyer4747now i am screwed, i just upgraded to Asterisk 1.8.5.0-1digium1~squeeze and it crashes on startup :(
07:02.51deltaflyer4747lets try fresh install :)
07:03.38deltaflyer4747and doesn't work :)
07:11.23deltaflyer4747http://pastebin.com/AvFhmrwd - if anyone knows whats wrong, i'd be glad.
07:18.40ChannelZwell it seems like your database is horked, unavailable, or * is misconfigured to talk to it
07:19.25*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
07:20.19deltaflyer4747ChannelZ: this is default installation - just apted the * and there isn't a word about any database prerequisites
07:20.45deltaflyer4747i followed this time step by step the https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
07:21.35ChannelZwell it seems to be configured for realtime
07:22.04deltaflyer4747meaning?
07:22.24ChannelZit's trying to use a MySQL database for some of its configuration
07:22.38ChannelZThat may or may not be related to your problems, or only a small part of them
07:22.46deltaflyer4747okay, i will give it one.
07:22.52deltaflyer4747no problem so far
07:23.03ChannelZwas this a previously working installation?
07:25.44deltaflyer4747ChannelZ: i used 1.6, today i upgraded to 1.8, it's not working. I purged old installation, installed new from scratch, same story.
07:26.20ChannelZso you're just setting this up for the first time to play?
07:27.40deltaflyer4747you could say that.
07:27.54deltaflyer4747althrough i ofcourse saved my previous config files :)
07:28.09ChannelZwell screw these packages, just build the source
07:28.24ChannelZit takes 10 minutes and you'll know how it's configured
07:29.08deltaflyer4747i thought that digium knows how to compile packages of their own product :)
07:30.11ChannelZI'm sure they do but you can too
07:30.45deltaflyer4747:) okay, downloading etc :)
07:30.47ChannelZIf it's just bailing because whatever configs you have are bogus, that's one thing.. but if it's coreing, that's another.
07:31.51deltaflyer4747yea, i kept default configs and its the same story :-/
07:32.02deltaflyer4747now... reading through ./configure options...
07:32.20*** join/#asterisk wonderworld (~ww@port-92-201-164-64.dynamic.qsc.de)
07:32.54ChannelZjust do it
07:32.56ChannelZ./configure
07:32.57ChannelZmake
07:33.00ChannelZbe happy
07:33.10deltaflyer4747straight like that? okay.
07:33.34ChannelZyou can 'make menuconfig' in between configure and make and tweak some things if you really feel you must
07:33.49ChannelZActually I do just to turn on/off the sound packages as desired
07:34.06deltaflyer4747okay
07:34.09deltaflyer4747will check
07:34.41ChannelZjust purge your old package off completely before you install, so you're not tripping over leftover bits possibly in other directory trees getting even more confused
07:34.44deltaflyer4747just fixing some prerequsities...
07:34.51deltaflyer4747yep, of course :)
07:35.49deltaflyer4747well, i hope this will be fix for the chanspy problem i'm facing :)
07:36.02ChannelZwhich was what
07:36.53deltaflyer4747yesterday evening we (me + p3nguin )found that chanspy either won't produce neccessary sound route through W option OR kills actual call between spied parties
07:38.18deltaflyer4747okay, i have to install dahdi first
07:38.35ChannelZhmm.  I've never actually used whisper
07:38.48deltaflyer4747ChannelZ: http://deltaflyer.cz/ast.txt
07:38.53deltaflyer4747thats what i need to do
07:39.54ChannelZso you're trying to listen to a security camera
07:40.04deltaflyer4747yes
07:40.10deltaflyer4747while calling the security camera
07:40.20deltaflyer4747that way the security camera will not hear itself
07:40.50deltaflyer4747so ... calling 11- 81 and making chanspy for 1001 on 11 with W - putting audio from 1001 to 11 only :)
07:42.51deltaflyer4747+ i think that WIMPy suggested that long time before we found for ourselves but then i haven't understood him
07:43.54ChannelZI'm not sure I understand what extension 81 has to do with it
07:43.59deltaflyer4747see
07:44.04deltaflyer4747that IPcam has 2way audio
07:44.14deltaflyer474781 is SIP exten on that camera
07:44.27ChannelZit speaks sip?
07:44.31deltaflyer4747yep
07:44.41deltaflyer4747but only as for incoming audio - ie sound FROM the camera.
07:44.50ChannelZSo if you dial it, it answers, and sends you audio from it's mic/sends your audio to a speaker?
07:45.15deltaflyer4747to get audio out of the camera (ie its MIC) i had to use MOH because its not transferred to SIP channel but only to its video stream
07:45.32deltaflyer4747and there goes the 1002 extension :)
07:45.41deltaflyer47471002 = MOH with audio stream from camera
07:45.55deltaflyer474781 = sip extension for audio TO camera
07:46.57deltaflyer4747its the only way how i can get 2way audio with that camera (btw manufacturer said its impossible)
07:48.18deltaflyer4747so i spent whole yesterday trying to get that chanspy working :)
07:50.36deltaflyer4747with great help of guys here
07:54.52deltaflyer4747ok, compiling.
07:55.51*** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com)
08:02.06deltaflyer4747needs faster cpu
08:02.24ChannelZdon't we all
08:03.24deltaflyer4747well, that old singlecore P4 is really slow :)
08:03.36deltaflyer4747hooray, compiled
08:03.45deltaflyer4747downloading sounds etc :)
08:04.11deltaflyer4747luckily (atleast that) i have good network connection :)
08:08.17deltaflyer4747okay, lets see.
08:08.30deltaflyer4747darn, initscript :)
08:10.19*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
08:12.57deltaflyer4747ChannelZ: and guess what :)) Same thing :D
08:13.27*** join/#asterisk Syrex (~syrex@dsl-146-17-03.telkomadsl.co.za)
08:13.36ChannelZwhat exactly is happening?
08:13.39deltaflyer4747http://pastebin.com/9zfRm5pE
08:13.45deltaflyer4747exactly the same thing :D
08:13.50deltaflyer4747asterisk won't start
08:14.08deltaflyer4747things it has something to do with druides...
08:14.14deltaflyer4747*thinks
08:14.47deltaflyer4747so the problem lies elsewhere
08:15.01ChannelZyes but is it crashing afterwards or what is going on?
08:15.02*** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593)
08:15.12deltaflyer4747it immediatelly crashes
08:15.23ChannelZAnd what configs are you using?  it seems to be loading a lot of crap you probably aren't even using
08:15.24deltaflyer4747thats everything whats in the logfile
08:15.31deltaflyer4747default configs
08:15.44deltaflyer4747ie from sources
08:15.57ChannelZyou have every single one of them installed?
08:16.13ChannelZI'm not sure why that would make it crash but it's probably not a good idea regardless
08:16.19deltaflyer4747i just made "make samples"
08:16.29deltaflyer4747and tried to run the *
08:16.48deltaflyer4747ie configure &&make&&make install&&make samples
08:17.30deltaflyer4747i can delete all files and put there only asterisk, extensions, sip and musiconhold conf files
08:20.13*** join/#asterisk esperanto (~rusty@pool-71-114-139-216.hrbgpa.dsl-w.verizon.net)
08:22.50ChannelZyes
08:23.25ChannelZthe samples are meant as guides, not a completely functional configuration.  There is a ton of crap in there that I'm sure will blow up
08:25.12deltaflyer4747ofc
08:30.24ChannelZI gotta go to bed
08:35.37deltaflyer4747i just copied all neccessary files and ... no go.
08:35.41deltaflyer4747still the same story.
08:36.21deltaflyer47471.8.5.0 won't start for me
08:39.29*** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45)
08:39.45deltaflyer4747so i hope someone will be able to help me. . .
08:45.26deltaflyer4747hempf, it does work now :)
08:45.32deltaflyer4747from packages
08:46.02deltaflyer4747that means that the aptitude purge haven't cleaned all previous bits and pieces
08:46.28esperantohey fellas, I am trying to generate a sip packet that would take 1.6.2.16-1 down
08:46.29esperantovulnerability http://web.nvd.nist.gov/view/vuln/detail?vulnId=CVE-2011-2529
08:46.29esperantodoes anybody know how exactly sip packet should look like?
08:46.49deltaflyer4747lol :D he's back :D
08:47.02deltaflyer4747esperanto: okay, i will explain that for loop from yesterday for you.
08:47.05deltaflyer4747It was countdown.
08:47.11deltaflyer4747meaning - you can guess.
08:48.13deltaflyer4747this is not an hacking chan
08:48.28deltaflyer4747this is asterisk SUPPORT not DESTRUCTION
08:48.56esperantowell, I didn't get it, just need to elaborate on this a little bit
08:49.18esperantoyeah, I understand that, but I need to do poc test, otherwise admins won't change anything
08:49.28*** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk)
08:50.19esperantoI use sipp to generate packets, tried pasting /0 almost everywhere and no luck
08:52.55Bipulwell i have gone through Asterisk architecture i Found IN COMMUNICATING with core PBX we need some protocol like DAHDI Which is DIgum asterisk hardware/ second is SIP which is protocol in communicating peer to peer connection only used for transporting voice data 3 one is IAX2 protocol Inter-Asterisk Exchange Protocol, IAX  h.323 old protocol used for  gnome and windows one = so my question is can we use RTP protocol ?
08:53.45Bipulin communicating with PBX core?
08:55.21tzafrirBipul, RTP for what, exactly?
08:56.27deltaflyer4747bipul: you want to listen to RTP stream, right? :)
08:56.30Bipullike RTP protocol is used for transporting videos and real time measurement data
08:56.38deltaflyer4747oh, ok
09:02.38tzangerhm, vodafone's got a decent deal
09:02.48tzanger4 euro for a microsim with 250MB of data on it
09:03.45tzangerif I'm not mistaken the recharge is more expensive than just spending another 4 euro for another 250MB
09:04.29batfastadHi everyone. I've just checked on our asterisk/freepbx VPS and noticed that /var/log/asterisk/full is getting pretty large... 20MB. Just read this info on how to rotate the Asterisk logs... http://www.voip-info.org/wiki/view/logrotate and just wondering if that info is still current?
09:10.20*** join/#asterisk old_monk (~nightrid3@91.176.160.33)
09:13.30*** join/#asterisk Micc_ (~Micc@c-98-232-41-66.hsd1.wa.comcast.net)
09:13.47Micc_why does my 1.8.5.0 keep reloading the dialplan every minute or two?
09:17.37Micc_seems like every 60 seconds its reloading the whole dialplan.
09:24.43deltaflyer4747check the config
09:24.48deltaflyer4747asterisk.conf
09:26.32Micc_is there a new setting in there that reloads the dialplan by default?
09:27.26Micc_I'm using my 1.6 configs, just changing things that need to be changed along the way.
09:27.49deltaflyer4747what does the logs say?
09:28.00Micc_nevermind. I think I figured it out.
09:28.03Micc_runaway cron job
09:28.06deltaflyer4747:)
09:28.08deltaflyer4747nice
09:28.25deltaflyer4747anyone have experience with chanspy / extenspy ?
09:29.28Micc_I've used chansply on 1.6, haven't tested it on 1.8 yet
09:29.59deltaflyer4747have you tried "whisper" ?
09:30.09Micc_yup, thats what we use it for
09:30.49deltaflyer4747so ... the spy can talk to spied person ?
09:32.35deltaflyer4747because i can hear the spied extension, but it cannot hear me (spy)
09:33.44deltaflyer4747do you have some example i could test please?
09:34.12Micc_yeah, the spy should be able to hear the device your spying on plus the call bridged to that device, plus in whisper mode be able to talk to the device without the person bridged to that device hearing anything you say.
09:34.33Micc_what version of asterisk are you using?
09:34.51deltaflyer47471850
09:34.54deltaflyer4747latest
09:35.02deltaflyer4747wasn't working in 1.6 either
09:35.03deltaflyer4747(for me)
09:35.16Micc_have you found 1.8.5.0 to be stable so far?
09:35.25deltaflyer4747i have it for ... 1 hour :)
09:35.34deltaflyer4747so... so far so good :D
09:35.49Micc_ok, I've been testing all day. so far so good. A few minore tweaks and had to fix parking.
09:36.11deltaflyer4747well, i need that chanspy and its not working for me :-(
09:36.16Micc_I'm about to put it into production I think. I was getting dead locks on 1.6.2.19
09:36.25Micc_let me look up how I'm doing that.
09:37.12deltaflyer4747thanks
09:37.27Micc_exten => *7701,1,ChanSpy(SIP/sandler1,wq)
09:37.32deltaflyer4747as i said - i can hear the spied extension, but that extension cannot hear me.
09:38.17Micc_what kind of device are you spying on?
09:38.42deltaflyer4747deskphone
09:38.43deltaflyer4747sip
09:39.17Micc_so you're talking to someone else with the deskphone first, then spying in from another sip device?
09:39.35deltaflyer4747yes.
09:39.38Micc_what parameters are you calling ChanSpy with?
09:39.57deltaflyer4747exten => 3131,n,Chanspy(SIP/11,w)
09:41.02Micc_that should work fine.
09:42.02Micc_it says don't use it with monitor/mixmonitor/record
09:42.08deltaflyer4747doesn't
09:42.17deltaflyer4747i'm not using any of those
09:42.37Micc_not sure what to tell you. It works for me.
09:42.49Micc_although I should probably test it in 1.8.5.0
09:42.53deltaflyer4747i know it SHOULD work.
09:43.06deltaflyer4747but it doesn't
09:43.23Micc_we do it with aastra phones.
09:44.23deltaflyer4747i have linksys deskphone (11), ipcam (81) and siemens wifi sip phone (211). I call 11 -> 81 and then dial from 211 to 3131
09:45.07Micc_try it in a different combination
09:45.30deltaflyer4747ok, i can dial 211->81 and 11->3131 :)
09:45.31deltaflyer4747sec
09:45.31Micc_maybe 81-211 then 11 spy
09:45.40Micc_yeah, try that.
09:47.01deltaflyer4747same story.
09:48.56deltaflyer4747i can hear 211 but 211 cannot hear me
09:49.34*** join/#asterisk bmg505 (~leon@196-209-7-14.dynamic.isadsl.co.za)
09:49.59deltaflyer4747like no whisper option was in place at the first time
09:50.54deltaflyer4747wouldn't some SIP.conf parameter break that?
09:51.15Micc_I don't know.
09:51.26deltaflyer4747yea, me neither
09:51.30Micc_maybe like reinvite or something.
09:51.43deltaflyer4747my thoughts exactly
09:51.47deltaflyer4747will enable it
09:51.51deltaflyer4747in a few, lunch time :)
09:51.58Micc_seems like maybe something could have an affect on it, but I can't think of anything off the top of my head.
09:52.28Micc_I don't think that will matter, but you can try it.
09:52.35Micc_we have canreinvite=no
09:53.28deltaflyer4747okay, that wasn't it.
09:53.36deltaflyer4747i had it too... :(
09:54.30deltaflyer4747darn
09:54.50deltaflyer4747maybe... do you have any zaptel devices?
09:55.06deltaflyer4747like isdn cards etc
09:57.13Micc_no, I haven't used one in a couple years.
09:57.17deltaflyer4747okay
09:57.21Micc_used to do a bit with some PRI cards.
09:58.35deltaflyer4747okay, but now you don't have any
09:59.57Micc_they fixed that cisco phone registration problem in 1.8.5.0 didn't they?
10:00.57deltaflyer4747i don't know of any
10:01.07deltaflyer4747never had cisco phone
10:01.30Bipuldude whear is sip.conf in /etc/asterisk ?
10:01.51deltaflyer4747exactly there
10:02.01Bipulbut there is nuthing
10:02.10deltaflyer4747really?
10:02.21Bipulbipul@bipul-desktop:/etc/asterisk$ ls
10:02.21Bipulasterisk.conf  manager.d
10:02.21Bipulbipul@bipul-desktop:/etc/asterisk$
10:02.22deltaflyer4747ls -al /etc/asterisk|grep sip
10:02.31deltaflyer4747lol :)
10:02.39deltaflyer4747then you are missing a ton of config files :)
10:02.42deltaflyer4747what did you do?
10:03.05Bipulnuthing
10:03.16Bipulcan i reinstall it
10:03.29deltaflyer4747then thats exactly what you get : nothing :)
10:03.42deltaflyer4747you had that sip.conf yesterday, right?
10:03.48Bipulyes
10:03.56Bipulbut now there is nuthing
10:04.36Micc_*cross fingers* 1.8.5.0 going into production
10:04.51deltaflyer4747so you had to delete them.
10:04.57Bipuldeltaflyer4747,  now what shud i do now ?
10:05.00deltaflyer4747Micc_: GL & hf
10:05.20deltaflyer4747Bipul: try to remember what happened.
10:05.45Bipul<PROTECTED>
10:05.56deltaflyer4747those files
10:06.18deltaflyer4747there are like 100 files in that directory
10:06.34Bipuli have reinstalled the asterisk that day
10:06.45deltaflyer4747what distro do you use
10:06.52Bipulubuntu
10:06.58deltaflyer4747great!
10:07.15deltaflyer4747then follow this https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages guide
10:07.45deltaflyer4747but ubuntu for server... tsk tsk tsk...
10:08.01Bipulno it's just a desktop
10:08.13deltaflyer4747okay then
10:08.14Bipulwill it work on desktop 10.10 ubuntu
10:08.29deltaflyer4747look on the webpage i posted
10:08.50deltaflyer4747its written there ;)
10:09.30deltaflyer4747how did you get previous asterisk?
10:09.45Bipulwell sudo apt-get install asterisk
10:09.52deltaflyer4747okay
10:10.11Bipuli know this link i have follow all the instruction of this page in past
10:10.19deltaflyer4747are you 100% sure you want to fully reinstall it and begin from scratch? [y/N]
10:10.37Bipulbut yah
10:11.04deltaflyer4747sudo aptitude purge asterisk asterisk-dev asterisk-config
10:13.10deltaflyer4747Micc_: will continue in a few minutes
10:14.03Bipuldeltaflyer4747,  then after apt-get clean ?
10:14.08deltaflyer4747no
10:14.27deltaflyer4747aptitude purge erases everything those packages are creating... or SHOULD erase :))
10:14.29Bipulsudo aptitude purge asterisk asterisk-dev asterisk-config i have used this command.
10:14.45deltaflyer4747did it gave you any errror?
10:14.57Bipulnops
10:15.04deltaflyer4747okay
10:15.08Bipullet me pastbin you the details
10:15.13deltaflyer4747ok.
10:16.05Bipulhttp://pastebin.com/CudqAiLz
10:16.17*** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com)
10:16.32deltaflyer4747good.
10:16.42deltaflyer4747now check that you have digium repo in place
10:16.47deltaflyer4747cat /etc/apt/sources.list
10:16.51Bipulso now shud i follow again those instruction
10:17.09deltaflyer4747should contain "deb http://packages.asterisk.org/deb "
10:17.13deltaflyer4747and deb-src
10:17.46Bipulyes  >>>deb http://packages.asterisk.org/deb maverick-proposed main deb-src http://packages.asterisk.org/deb maverick-proposed main
10:17.55deltaflyer4747good.
10:18.27Bipulnow shud i installe it.
10:18.59deltaflyer4747aptitude install asterisk-1.8 asterisk-dahdi
10:19.08deltaflyer4747sorry, sudo before that
10:19.28Bipulyes but dahdi is for hardware perpose right ?
10:19.38deltaflyer4747then ommit it :)
10:19.45Bipulbut i dont have digium card so it's necessary? to install
10:20.05Bipulalright !
10:20.07deltaflyer4747if you want to use meetme, then yes
10:20.26deltaflyer4747meetme needs zaptel-dummy for timing
10:20.45Bipulsure why not one day i will come to meet you :p
10:20.50deltaflyer4747:D
10:21.03deltaflyer4747meetme = conference rooms :)
10:21.25Bipuloh ic :p
10:21.34deltaflyer4747yep
10:21.59BipulHow much it fetch to perchase a digium card?
10:22.09deltaflyer4747what for?
10:22.13deltaflyer4747btw...
10:22.22deltaflyer4747did you installed those python packages, right?
10:22.27Bipulconnecting betwen PSTN
10:22.41deltaflyer4747sudo apt-get install python-software-properties
10:22.53deltaflyer4747what pstn
10:23.38Bipulpublic switched telephone network
10:23.39deltaflyer4747get some SIP provider
10:23.50deltaflyer4747thats way cheaper and easier
10:23.53deltaflyer4747and more reliable
10:23.57deltaflyer4747etc
10:24.09Bipulyes i am reading the books and i have decided to work on sip
10:24.33deltaflyer4747analog nor isdn have no future
10:24.41Bipulok
10:25.40Bipulpython is already installed
10:25.47deltaflyer4747okay.
10:25.52deltaflyer4747just recheck - to be sure
10:25.59nicola_pavhello. anyone familiar with this kind of error: Channel 0/13, span x got hangup, cause 81?
10:27.06*** join/#asterisk radic (~radic@tmo-096-134.customers.d1-online.com)
10:30.22Micc_oh crap, got a major problem. its not parsing sip correctly.
10:30.33Micc_my extensions are coming in wrong like this
10:31.09Micc_Executing [4252505555;phone-context=+1;npdi=yes@inbound-userfield:1]
10:32.12Micc_why would it be doing that?
10:32.54deltaflyer4747err... don't get the full story
10:33.32nicola_pavhello. anyone familiar with this error: !! pri_hangup() line:1431 Called with invalid call ptr (0x9c26158)
10:36.49Micc_wtf, its rejecting almost all my incoming calls saying the extension isn't found in the context. but it is there.
10:37.25deltaflyer4747include?
10:38.31Micc_I have a shitload of includes.
10:38.41Micc_but this works fine on 1.6.2.19
10:38.56Micc_I'm gonna have to roll back if I can't figure this out pretty quick.
10:39.21Micc_it seems asterisk isn't parsing sip headers the same.
10:39.58Micc_is there a way I can remove everything after the ; with some dialplan commands?
10:43.38Micc_ok, that works for now, a bit of a hack though.
10:44.46*** join/#asterisk Jasnejac (kvirc@81.91.107.236)
10:45.31Bipuldeltaflyer4747,  there is problem in configuring sip.conf for creating SIP account
10:45.46deltaflyer4747Micc_: what what ?
10:45.49deltaflyer4747Bipul:  ?
10:45.57Bipulhttps://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts   << i cant find such matches in my sip.conf
10:46.42deltaflyer4747matches to what? those are pure example
10:47.06deltaflyer4747http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
10:47.12deltaflyer4747read this
10:47.20Bipuli mean i want to fix my own password and sip and ip
10:48.47Micc_deltaflyer4747, for some reason with 1.8.5.0 my dialplan was matching extra crap in the sip header with the number make it so I could not receive any calls from 360
10:49.02deltaflyer4747weird
10:49.10Micc_I almost had a heart attack until I fixed it with a couple lines of dialplan code.
10:49.21deltaflyer4747hehe
10:49.45Micc_But why its doing that is a big mystery. Have you tested receiving calls from your provider on 1.8.5.0 yet?
10:49.54dymHey all. Im having trouble registering my softphone, which is located behind a nat. Here is the logs: http://paste.debian.net/123140/ - nat=yes is enabled, but still things seem to be going wrong. any idea?
10:49.59deltaflyer4747will try right away
10:50.09deltaflyer4747dym: host=dynamic
10:50.32dymis
10:51.00deltaflyer4747qualify
10:51.10esperantoenable sip debug and toubleshoot
10:51.21deltaflyer4747Micc_: works
10:51.39Micc_who is your provider?
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10:51.45Micc_I have a couple others I can try.
10:51.48Micc_maybe its just 360
10:51.53deltaflyer4747Micc_: some local :)
10:52.07deltaflyer4747from my country
10:53.31Micc_deltaflyer4747, so you don't have a sip provider? You're using a land line?
10:53.41deltaflyer4747its sip
10:54.34Micc_vitelity seems to be fine, so maybe its just 360, but it was fine in 1.6.2.19. I guess asterisk is probably trying to be more copmliant with the sip rfc.
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10:56.17deltaflyer4747DAMN it
10:56.28deltaflyer4747why ccannot i get that whisper working
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10:57.38Micc_This doesn't look like proper sip header.
10:57.50deltaflyer4747:))
10:57.55deltaflyer4747theres your problem :)
10:58.12Micc_I gotta pastebin this so you can see how screwed up this is. 360 is on crack.
10:59.02Micc_http://pastebin.com/MJTADYHa
10:59.32_zoom_fellas do you have any ideas what is really going on why microsoft killed skype-asterisk license?
10:59.39deltaflyer4747nice
10:59.56deltaflyer4747zoom: asterisk is free, right?
11:00.09_zoom_but skype is not
11:00.23deltaflyer4747and did you get ANYTHING for free from m$ ?
11:00.39deltaflyer4747except tons of troubles
11:00.49_zoom_:)
11:00.52Micc_deltaflyer4747, am I right? That isn't correct sip format is it?
11:01.25deltaflyer4747Micc_: i'd have to study correcct sip first
11:01.43deltaflyer4747never needed to know that
11:01.49_zoom_so they are going to bury skype a live
11:01.51_zoom_:(
11:02.07*** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk)
11:02.20deltaflyer4747DAMN I HATE CHANSPY :-(
11:04.29*** part/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it)
11:04.46Micc_deltaflyer4747, I wish I could be of more help with that.
11:05.05deltaflyer4747i know
11:05.22Micc_maybe I should test chanspy on my 1.8.5.0
11:06.05deltaflyer4747http://pastebin.com/KGzESjS5
11:06.12deltaflyer4747this is whole transaction...
11:08.15Micc_I thought you had to have the other call going before doing chanspy? Looks like ChanSpy is the first call here.
11:08.26deltaflyer4747doesn't matter
11:08.33deltaflyer4747i can hear spied party
11:08.39deltaflyer4747but not vice versa
11:09.48deltaflyer4747http://pastebin.com/XsizAMHU
11:09.53deltaflyer4747same story
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11:13.13deltaflyer4747i know this SHOULD work, but it doesnt :(
11:13.41Micc_deltaflyer4747, works great for me on 1.8.5.0
11:13.50deltaflyer4747:( DAMN
11:13.58deltaflyer4747whats wrong with this setup :(
11:14.11deltaflyer4747what codec do you use?
11:14.53deltaflyer4747ie sip show channels
11:15.35Micc_g722
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11:16.14deltaflyer4747i c
11:16.18deltaflyer4747could be the difference
11:16.21deltaflyer4747i use alaw
11:16.34Micc_maybe.
11:16.38Micc_its worth a shot
11:16.43Micc_or try ulaw
11:16.49deltaflyer4747or that
11:16.52Micc_shouldn't make a difference, but who knows.
11:17.08deltaflyer4747but ... i am affraid that the camera knows only alaw ...
11:17.35deltaflyer4747lets see :)
11:21.09deltaflyer4747my deskphone doesn't support 722
11:21.11deltaflyer4747:-/
11:21.13deltaflyer4747darn
11:21.24Micc_deltaflyer4747, were the others all using alaw too? it probably helps if they are all using the same codec.
11:21.36Micc_they should all support ulaw I would thing.
11:21.38deltaflyer4747they were
11:21.58Micc_getting tired. 4:21am here.
11:22.35Micc_try a soft phone like zoiper.
11:22.47deltaflyer4747will try...
11:23.00deltaflyer4747but now... go take a rest, i gotta go shopping anyways
11:23.03deltaflyer4747thanks pal
11:23.39Micc_good luck and good night.
11:23.43deltaflyer4747i hope i will catch someone else in the evening :)
11:23.51deltaflyer4747good night to you, 1:23 pm here :)
11:24.01deltaflyer4747(gmt + 1 DST)
11:24.07deltaflyer4747CET
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11:27.57joker2u?
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11:31.30deltaflyer4747*
11:32.19deltaflyer4747afk, shopping
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11:41.48Bipulpermit=192.168.5.0/255.255.255.0 ; replace with your network settings < shud we use my Dynamic IP instead of 192.168.5.0 , as i have define host=dynamic
11:44.50nightrid3rBipul u use the address of hosts that are allowed to connect
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11:45.16joker2uI am trying uderstand the asterisk ao2 iterator. Is the object I am passing data too? it is my understanding that py-asterisk is the interface to the asterisk api that passes/translates to the ao2 object? astobj2.h to be specific? Am I understanding this correctly?
11:46.19Bipulnightrid3r,  i have used  permit=my dynamic IP/255.255.255.0 <-- is it correct formate
11:48.52Bipulthis setting shud be for my other computer
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12:07.16lcbhi. could you please lead me to directions on how to run a basic interactive answering system for 2 analog phone lines, on a debian based server. Is asterisk too much for it?
12:08.05WIMPylcb: That's exactely what Asteris is good at. Try the
12:08.07WIMPy~book
12:08.07infobotAsterisk: 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook
12:08.53lcbWIMPy: great. i were trying to overpass the whole manual due to language concerns.
12:10.04lcbWIMPy:  btw, is it possible to have a remote interface /frontend to control it?
12:10.21WIMPyControl what?
12:10.47lcbor do i need to run graphically on the server, meaning installing a graphical desktop environment
12:10.54WIMPyIt has several options for remote control.
12:11.01lcbohh ok.
12:11.19lcbWIMPy: thanks so much
12:11.40WIMPyNo. There are web frontends, but they have some limitations and are only recommended for basic installs.
12:12.15lcbi see. in your opinion a basiv desktop environment would be great, don't you think?
12:12.24lcb*basic*
12:12.30WIMPyAsterisk has a shell for control as well as the management interface and the option to call external scripts.
12:12.49WIMPyDesktop envirronment?
12:13.26lcbgraphical environment (gnome, kde, etc)
12:13.47WIMPyYou don't need that.
12:14.13WIMPyAnd it is said to be able to cause timing issues on slow machines.
12:14.51lcbi see. so a remote solution would be better, i guess. or locally trough CLI
12:15.06lcbWIMPy: tks. have a good Sunday.
12:15.40WIMPyI've been using it for years on my desktop without issues. on a Core2Duo 2.4GHz.
12:16.27lcbWIMPy: that's the same one i'm trying to install asterisk. but is a server, with 4 G ram
12:17.07lcban old desktop with no monitor or keyboard (accident)
12:17.32lcbi mean, laptop, sorry
12:18.40lcbreading the book. i'll be back in one month ;)
12:19.20WIMPythinks it will be earlier :-)
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12:20.18lcb:)
12:20.42lcbpaying and downloading the book was fast, though
12:21.08lcb:)
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12:37.53deltaflyer4747Bipul: i haven't seen any "permit" line in the example i've sent to you
12:38.03deltaflyer4747WIMPy: Hi, have spare minutes?
12:39.03WIMPydeltaflyer4747: Never. But try to ask anyway :-)
12:39.09deltaflyer4747:))
12:39.16deltaflyer4747remember my yesterdays problem?
12:39.21WIMPyyes
12:39.26deltaflyer4747i think i get your idea now
12:40.08deltaflyer4747but when i try to chanspy on 11 for 1002, i get no sound through whisper to 11
12:40.19deltaflyer474711 is spied-on extension, 1001 is the spy
12:40.32deltaflyer4747tested with pure sip devices, but no luck
12:41.35WIMPyYou're saying chanspy doesn't work for you, not even without that camera thing?
12:41.58deltaflyer4747yes
12:42.07deltaflyer4747http://pastebin.com/XsizAMHU
12:42.36deltaflyer4747exten => 3131,1,Chanspy(SIP/11,qw)
12:42.58deltaflyer4747i dial from 11 to 81 and then from another sip phone i dial 3131. I can hear 11 but 11 cannot hear me.
12:44.11deltaflyer4747when i do W, i ofc. cannot hear 11 yet 11 still cannot hear me.
12:44.21deltaflyer4747all extensions use same codec
12:44.31deltaflyer4747(alaw)
12:44.54WIMPyWhat Version are you using? Have you checkt your version has option w?
12:44.55Bipuldeltaflyer4747,  can you past me a link whear i can learn  how to setup a trunk to my voip account
12:45.07deltaflyer4747it doesn't matter if i create chanspy before or after dialing 11-81 http://pastebin.com/KGzESjS5
12:45.11deltaflyer4747Bipul: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
12:45.15deltaflyer4747already pasted today ;)
12:45.28deltaflyer4747WIMPy: latest available - 1.8.5.0
12:45.40WIMPyHmm.
12:45.52deltaflyer4747upgraded today from 1.6
12:46.02Bipulhave not sleep for 2 days my mind ohh..
12:46.19deltaflyer4747mov bipup,sleep
12:46.44deltaflyer4747WIMPy: same setup works for Micc_
12:46.48deltaflyer4747in 1.8.5.0
12:46.50Bipulthe link which you have given to me is for sip trunking?
12:47.17deltaflyer4747Bipul: for sip general config.
12:47.29WIMPydeltaflyer4747: I haven't actually tried to call ChanSpy with w, only activated it via DTMF.
12:47.45deltaflyer4747extension and trunk are not so much different
12:47.47WIMPyBut if it works for others, it starts to become strange.
12:47.57deltaflyer4747WIMPy: i tried that as well, wasn't working (option d)
12:48.31deltaflyer4747# works for changing volume, 456 does nothing.
12:50.53WIMPyOk, just checked. It works for me with qw.
12:51.22WIMPySo we need to find out why it doesn't work for you.
12:51.46WIMPyUnfortunatly I don't have an idea what might cause that.
12:51.58deltaflyer4747i can give you every log you ask for
12:52.15deltaflyer4747but ... i don't have that idea as well.
12:52.28deltaflyer4747sec, i will call external line if the problem is in 81...
12:52.53deltaflyer4747it is.
12:52.56deltaflyer4747DAMN !
12:53.01WIMPyHuh?
12:53.04deltaflyer4747yes.
12:53.12deltaflyer4747i dialed external line and it works!
12:53.20WIMPyWhat's happening exactely?
12:53.30deltaflyer4747if i dial 81, it doesn't work.
12:53.41deltaflyer4747if i dial external line, it works.
12:53.52deltaflyer4747gotta go, be back in 0.5h
12:53.56WIMPy81 is a phone?
12:53.58deltaflyer4747(moving furniture)
12:54.01deltaflyer474781 is that camera.
12:54.02deltaflyer4747afk
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12:57.46deltaflyer4747back
12:57.51deltaflyer4747so thats strange.
12:57.54*** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex)
12:58.41deltaflyer4747let me test something
12:59.33WIMPyAsterisk might get confused because the cam is not sending RTP.
13:01.33deltaflyer4747i don't know really...
13:01.59deltaflyer4747i might know the answer
13:02.06deltaflyer4747<PROTECTED>
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13:08.21WIMPyLooks perfectly ok to me.
13:09.17WIMPyBut someone more in to RTP and how ChanSpy works internally might be able to tell you more.
13:10.17deltaflyer4747i'm trying to recall how is that core debugging command in cli...
13:10.56WIMPycore set vebose|debug #
13:11.08deltaflyer4747thanks
13:11.13WIMPyI think there's an RTP debug as well.
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13:12.23deltaflyer4747yes
13:12.40deltaflyer4747now only how could i see it ...
13:13.07deltaflyer4747ok
13:13.08deltaflyer4747:)
13:15.34batfastadHi everyone. We currently have a hosted Asterisk/FreePBX system running in an OpenVZ container, we're only 15 extensions and it works great. But does anyone have any experience in running under Xen?
13:16.19deltaflyer4747okay... i see lot of << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/11-0000000d]
13:17.46batfastadI'd like to move it under our control on our colocated Xen server. We don't use any special telephony hardware as we have a SIP trunk provider for PSTN conversion
13:18.55deltaflyer4747thats apparently the output from the camera itself.
13:18.56WIMPydeltaflyer4747: I'm off for an hour or so. But that's an area where I won't be able to provide much help anyway.
13:19.14deltaflyer4747WIMPy: i c
13:20.10Jasnejacbatfastad: I have several installations under Xen - no problems at all
13:22.36deltaflyer4747okay, filtered those null frames with dtmftype=inband
13:24.51batfastadJasnejac: That's great news. I figured it should be similar if not better than OpenVZ because they're both paravirt
13:27.05deltaflyer4747darn with that routing
13:35.04deltaflyer4747i might see an problem here :)
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14:01.38Bipuldtmfmode= ? what is that is it related to sounds ,tones
14:01.56Bipulis it codec?
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14:14.14JasnejacBipul: in call key press sounds
14:14.36deltaflyer4747Bipul: have you ever TRIED to read the http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf i've sent you?
14:14.40Bipulok thanks Jasnejac
14:14.48deltaflyer4747its described there ;)
14:14.56Bipulyes  deltaflyer4747
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14:17.36Bipulok after editing the things at sip.conf we have to reload the sip.conf ar CLI>
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14:24.49Bipulp3nguin,  ping
14:24.56p3nguinyes?
14:25.11Bipulcan  i pm you
14:25.15p3nguinyes.
14:25.22Bipulthank you.
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14:44.40dymgreetings!
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14:49.49*** mode/#asterisk [+o russellb] by ChanServ
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14:50.21[sr]hi
14:53.25WIMPyIsn't it great if you can start the day with a steak and a cake fom yesterdays BBQ?
14:54.08[sr]BBQ = ?
14:54.35p3nguinYou don't know what bbq is?
14:54.37WIMPyBarbecue
14:54.51[sr]WIMPy: in that case i agree
14:54.52[sr]:p
14:55.13[sr]p3nguin: no... i'm portuguese.. :p
14:55.28[sr]but barbecue I know of course, but now i know what BBQ is :)
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15:14.31deltaflyer4747so... anyone able to help me with chanspy problem? Its perhaps something with RTP data
15:15.58p3nguinI only happens when using the moh through a local channel.  When I spy on a phone, it works correctly.
15:17.04p3nguins/I/It/
15:17.52p3nguinI didn't try spying from my phone using a local channel.  I should test that later.
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15:19.48deltaflyer4747p3nguin: it happens only when i spy on the 81, if i call regular extension, it works. . .
15:20.09p3nguinYour incorrect usage of terminology confuses me.
15:20.18deltaflyer4747so there is something terribly wrong with that extension
15:20.36deltaflyer4747?
15:20.39p3nguinExtensions are how you write them.  If it's wrong, you wrote it wrong.
15:20.53deltaflyer4747ok, to that channell...
15:22.40deltaflyer4747afk, brb, going to test motion detection of that camera
15:22.56WIMPyI suspect the cam might not send any RTP, which is something that caused massive problems in the past. Might still upset ChanSpy.
15:23.29p3nguinI tested both chanspy and extenspy using two phones and a local channel that calls moh.
15:23.55p3nguinIt connects the local channel, moh starts, but the sound never makes it to the phone.
15:24.13WIMPyNo need to go that far
15:24.25p3nguinI had to replicate his setup.
15:24.46WIMPyThere's no MOH involved yet. It works on a call between two phones, but not on a call between a phone and the cam.
15:25.15p3nguinWhen someone presses the button at him cam, it puts the cam's audio into an moh stream.
15:25.51WIMPyUnless i misunderstood something.
15:25.56p3nguinI think you did.
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15:26.44WIMPyYes, but that is not the issue. The connection between the phone and the cam seems to have issues already.
15:27.02p3nguinThe cam has a speaker and a mic.  When the button is pressed, it connects a call between the cam and the deskphone.  The speaker on the cam can hear the deskphone, but the deskphone cannot hear the mic from the cam.  Because of that, he rigged the cam's mic audio into an moh stream.
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15:27.27WIMPyJepp. Been though that setup yesterday.
15:27.49p3nguinSo the deskphone could listen to that moh and hear the cam's audio.
15:27.50pqwild68Hello
15:28.19pqwild68Is anyone using the alarmreceiver module?
15:28.59WIMPyI guess it wouldn't exist if noone had a need for it.
15:28.59p3nguinBut the deskphone can only have one call at a time: either connected to the cam so the cam can hear the deskphone, or connected to moh to listen to the cam's audio
15:29.50WIMPyI know. I suggested to use ChanSpy in whisper mode to inject MOH to only one leg of the call.
15:30.12p3nguinThat's what I tested.  It doesn't work.
15:30.23WIMPyBut whisper doesn't work on that call. Not from a 3rd (2nd) phone, either.
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15:30.47WIMPyThat would make two issues then.
15:30.53p3nguinAs I said, the spy starts, the local channel is connected, the moh starts, but the moh audio is not being put into the call.
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15:32.05p3nguinSince I use ChanSpy and ExtenSpy regularly and they work correctly when spying from my phone directly, my next test is to spy a local channel which dials my phone.  If it fails, then the problem is spying via local channel.
15:32.46WIMPyIt might also be because MOH is alway on hold.
15:33.20WIMPyhas just tested that. There are indeed two issues as it seems.
15:33.21p3nguinIf I call the extension where MusicOnHold() runs, the music plays.
15:33.25WIMPyBad.
15:33.57p3nguinI also need to test calling the moh via local channel, similar to the way I tried to spy from it.
15:34.08p3nguinSo I have two tests to perform.
15:34.34WIMPyThe MOH from an empty confbridge doesn't arrive, either.
15:34.48deltaflyer4747guys guys ...
15:35.03deltaflyer4747you are both right.
15:35.16deltaflyer4747chanspy cannot spy for MOH
15:35.32deltaflyer4747and my setup is somewhat faulty as camera doesn't send any RTP (aka "silence detection")
15:35.58p3nguinI don't understand why moh can't spy via local channel.
15:36.23p3nguin~book
15:36.23infobotAsterisk: 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook
15:36.26p3nguinbipul: ^^^
15:38.15WIMPyPlayback instead of MOH doesn't work, either.
15:39.34WIMPyUgh.
15:39.37deltaflyer4747so first thing i need to do is somehow produce some RTP data or force asterisk to ignore it
15:39.58WIMPyAfter I called ChanSpy from a phone, I got a fragemnt of what I used in the Playback.
15:41.38deltaflyer4747is there a way how to force * to ignore "silent detection" on some phones
15:44.46p3nguinI think Asterisk ignores silence detection from everything, since it doesn't support silence detection.
15:45.55deltaflyer4747well...
15:45.55dym:D
15:46.31deltaflyer4747on http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf is important sentence: Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppr
15:47.22dymSip Phones
15:47.25dymNot the Asterisk
15:49.14deltaflyer4747dym: you don't get the story ;)
15:49.58dymI guess p3nguin will answer anyways
15:50.25deltaflyer4747camera is not sending any RTP data
15:50.43deltaflyer4747ie there is "silence detection turned on" on that camera
15:52.33p3nguinAsterisk does not care.
15:52.52p3nguinAsterisk does not support silence suppression.
15:52.58WIMPyIt uesed to.
15:53.02WIMPy-e
15:53.07p3nguinIt never has in my lifetime.
15:53.33deltaflyer4747https://issues.asterisk.org/view.php?id=5374
15:53.37WIMPyOn a phone with silence suppression I couldn't hear anything unless I talked myself.
15:53.48deltaflyer4747thats it WIMPy
15:53.53p3nguinThat's the phone's problem and has nothing to do with Asterisk.
15:53.58deltaflyer4747Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream.
15:54.08deltaflyer4747ie
15:54.13WIMPyI just tried to put the chanspy on a local channel and use MOH directly. Doesn't work, either.
15:54.26deltaflyer4747no incoming RTP stream = no data sent
15:54.27WIMPyNo. That was an Asterisk Problem.
15:54.54WIMPyAsterisk didn't send RTP unless also receuving.
15:54.54WIMPyFrom the same devide, tah is.
15:54.58WIMPythat
15:55.39WIMPyWould be logical if it didn't send, what it didn't receive from the other end.
15:55.47pai have a local asterisk server with a dahdi channel to connect to the telephone network. how can i configure asterisk to be able to send out faxes through this dahdi channel? (i dont care about receiving faxes, for the moment)
15:56.49dympa: http://ofps.oreilly.com/titles/9780596517342/asterisk-Fax.html#Fax_id265396
15:57.36pathanks a lot, i check it out right now!
15:57.43dymnp
15:59.08deltaflyer4747so guys... this seems hopeless, right... :(
15:59.56*** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com)
16:01.37*** part/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com)
16:03.01WIMPyClose
16:03.20deltaflyer4747?
16:03.38WIMPyYou might be able to overcome the RTP issue by putting a conference in between.
16:03.56WIMPyBut the injection Issue is also present.
16:04.36WIMPyBut at least we learned something new.
16:05.20deltaflyer4747well, as i asked, can i setup such conference, where everybody listens to the manager and manager can choose to whom he listens to?
16:05.32WIMPyNo
16:05.36deltaflyer4747:(
16:06.06deltaflyer4747or...
16:06.13deltaflyer4747might as well do the trick ...
16:06.26deltaflyer4747no, won't work
16:06.34WIMPyI just thought about an additional conference to make the phone actually receive (silent) audio so that ChanSpy *might* work on that channel.
16:06.43deltaflyer4747yea i know
16:06.58deltaflyer4747but there is still chanspy vs moh
16:07.15WIMPyYes. Two issues.
16:07.18deltaflyer4747:(
16:07.21deltaflyer4747crap
16:09.22WIMPyHas anyone here tryed using Imcomplete()?
16:13.34WIMPyIs there any way to manipulate the dialed extension, other thatn Goto()?
16:15.28*** join/#asterisk irroot (~irroot@197.169.159.67)
16:18.40deltaflyer4747OMG :)
16:18.47deltaflyer4747that camera uses wrt54g hardware :D
16:19.07deltaflyer4747line from config: "/etc/wrt54g.large.ico"
16:20.28WIMPyYou might be able to replace the client then.
16:22.05deltaflyer4747exactly!
16:22.11deltaflyer4747just trying to SSH into it
16:23.49deltaflyer4747afk, lift is again stuck.
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16:41.55[sr]I'm sad :(
16:46.04deltaflyer4747me2
16:48.20*** join/#asterisk timahvo1 (~rogue@41.223.57.77)
16:48.43p3nguinmetwo!
16:49.41deltaflyer4747?
16:52.07p3nguin2 = two
16:52.12p3nguinme2 = metwo
16:53.15[sr]p3nguin:  :p
16:55.59[sr]WIMPy: when you invite me to DE to present some friends (girls) to me? :p
16:56.13dymWIMPy: has friends/girls?
16:56.15dym:D
16:56.44*** join/#asterisk fabiobik (5d660b57@gateway/web/freenode/ip.93.102.11.87)
16:56.52fabiobikHi guys
16:57.00WIMPy:-)
16:57.32deltaflyer4747p3nguin: well then read it aloud...
16:57.34fabiobikI have an Huawei E220 usb internet modem and i know i can convert it to SMS gateway
16:57.43fabiobikand i can unlock voice feature too
16:57.46p3nguinI DID!
16:58.05fabiobiki want to know if i can make an gsm gateway with asterisk
16:58.26WIMPySee chan_datacard
16:58.27fabiobikand this usb modem
16:58.27*** join/#asterisk radic (~radic@tmo-097-61.customers.d1-online.com)
16:58.36WIMPyOr google for that.
16:58.49fabiobikp3nguin what you did?
16:59.01dymfap
16:59.10*** join/#asterisk Nasga (~Nasga@AAmiens-157-1-106-45.w86-208.abo.wanadoo.fr)
16:59.20fabiobikWIMPy but you know if this possible?
16:59.28deltaflyer4747:))
16:59.50WIMPyLike I said: Look at chan_datacard.
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17:00.30*** part/#asterisk DennisG_ (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl)
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17:01.16dymAnyways - gsm gateway?  What exactly do you mean by that
17:01.18dym?
17:01.40*** join/#asterisk DennisG_ (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl)
17:03.08fabiobikdym gsm gateway to make calls
17:03.20dymwell
17:03.25dymsince its a data device: doubt it.
17:03.50fabiobikdym my data device can be unlocked to voice too :)
17:04.08nightrid3rfabiobik there are cards with up to 4 gsm devices on it
17:04.46fabiobiknightrid3r not understand
17:04.49fabiobiknightrid3r what u mean
17:05.10WIMPyThe E1550 (which I have) seems to exist with voice support, but that's all I could find out. No idea if is's a matter of unlocking or reflashing or whatever.
17:05.40fabiobikWIMPy is that Huawei?
17:05.50WIMPyyes
17:06.05fabiobikhumm
17:06.11nightrid3rfabiobik http://www.openvox.cn/products/show.php?itemid=150&lang=2
17:06.39WIMPychan_datacard detects it ok, but no calls.
17:07.58fabiobiknightrid3r yes but that is 300 dollares
17:08.37*** join/#asterisk godmachine-x6 (~godmachin@h18.171.140.67.dynamic.ip.windstream.net)
17:09.33BipulTHanks alot ! to this asterisk community >:D<
17:09.53fabiobikwhat im trying to do is this conect my data modem unlocked with asterisk and then make an an record with my voice saying "Dial the number"  and then write the number and connect to that number with sip
17:10.01fabiobikIts possible or im crasy
17:10.02fabiobiklool
17:10.30fabiobik?
17:10.36dymWhat does the gsm gw have to do with all that?
17:11.08fabiobikdym establish the connection between my phone and pc?
17:12.08dymi dont quite understand the szenario
17:12.28dymmake a voice record and dial the number therefore you need a gsm connection from your mobile to asterisk
17:12.36dymdoesnt get much more odd
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17:13.21WIMPyI guess it's a call through thing. But it wasn't clear to me, either.
17:13.55deltaflyer4747YAY, i got telnet into camera ! :)
17:13.56Nuggettelnet is eeeeeeevil!
17:14.08dymtelnet into camera?
17:14.15dymare you all on drugs?
17:14.29nightrid3ryay ASCI pron :)
17:14.41deltaflyer4747dym: ?
17:14.42fabiobikya its calling trught that
17:14.52deltaflyer4747Nugget: i am glad atleast for that :)
17:14.57fabiobikmy english is not the best
17:14.57fabiobiklol
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17:16.15fabiobikdym what i want to doo is make calls trught that ...   and for example on connection  ear "dial the number"
17:16.32fabiobik*when the connection was stabilshed
17:17.13*** join/#asterisk Leddy (leddy@89.238.176.88)
17:18.17nightrid3rfabiobik DISA ?
17:18.37fabiobikwhat is that?
17:18.53nightrid3ri guess thats what your looking for
17:19.02deltaflyer4747:D
17:19.04WIMPyA dialplan application you may want to call for that scenario.
17:19.33fabiobikphone ----> pc ----> SIP
17:21.04fabiobikis that?
17:25.13deltaflyer4747O M G
17:25.33deltaflyer4747do you know what processes that SIP ?
17:25.55WIMPyI guess you will tell us in a moment.
17:26.06deltaflyer4747rtsps...
17:26.16deltaflyer4747Streaming Server App Version: 1.1.0.0 RTSPSTREAMING Server Module Version: 1.7.1.4
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18:51.08tonsofpcsI'm looking at running an asterisk system as a temporary hold-over between PBXs (replacing an old relay-driven PBX with a modern system -- will eventually use the asterisk setup as a failsafe amongst other things).  Is there a nice simple frontend and/or easy to configure pre-made distribution of some kind?
18:53.29deltaflyer4747there is
18:53.42deltaflyer4747but i don't know if i can mention it there
18:54.45tonsofpcscan you mention it in a PM?
18:54.53mickecarlssonis shameless
18:55.07mickecarlssonFreePBX Distro or AsteriskNOW
18:55.19mickecarlssonBoth run FreePBX
18:55.26deltaflyer4747or trixbox for that matter...
18:55.35mickecarlssonDONT got trixbox, it is dead
18:55.45mickecarlssonand use a very old freepbx
18:55.49tonsofpcsthanks  :)
18:56.06*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
18:57.04tonsofpcsasterisknow looks cool.... I wonder if anyone has made a bootCD version
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18:57.37tonsofpcs(live CD)
18:57.43mickecarlssonNope
18:57.58mickecarlssonAsteriskNOW is made by Digium
18:58.11mickecarlssonUse FreePBX as the GUI, or their own, chosen at install
18:58.19deltaflyer4747mickecarlsson: really? thanks for the info
18:58.41mickecarlssonYes
18:58.54deltaflyer4747good to know
18:59.08mickecarlssonFreePBX Distro is a Netinstall, download all setup from Internet
18:59.27mickecarlssonhttp://www.asterisk.org/asterisknow/
18:59.40tonsofpcsnow I just need to find some hardware :)
18:59.58mickecarlssonYou can use VMware or VirtaulBOX to test it out
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19:00.12deltaflyer4747:))
19:00.15tonsofpcsoh, I mean interface hardware :)
19:00.24deltaflyer4747thats expensive.
19:00.33tonsofpcs(and if someone makes a card that can talk to nitsuko handsets.....
19:00.33mickecarlssonWell, go by SIP and you wont need it
19:01.13tonsofpcsmickecarlsson: sure you will, it's just contained in the handset base instead of the PBX 'host'.
19:01.51mickecarlssonI use DECT phones that talk SIP
19:02.11mickecarlssonNever used any hardware except for the server
19:02.23mickecarlssonBy hardware, I mean any FXS/FXO card
19:03.04mickecarlssongtg, soccer on tv
19:04.01deltaflyer4747yea, sip DECT phones are really cheap
19:05.00tonsofpcsmickecarlsson: it's football and they've been hitting the ball out so much that there's plenty of time to chat
19:05.05tonsofpcs(go USA!)
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20:00.31AlexForsterBest: 99.967 -- Worst: 96.487 -- Average: 99.186181, Difference: 99.985145
20:00.48AlexForsterdefinitely the cause of my choppy audio (asterisk on centos under hyperv), correct?
20:01.06WIMPyProbably
20:01.57AlexForsteranyone know a solution off the top of their heads, before i go googling?
20:02.20WIMPyShared IRQs?
20:03.02AlexForsterthat sounds like "install integration services"
20:03.08AlexForster?
20:03.16WIMPyDrivers of other hardware locking up?
20:03.18WIMPyWhat?
20:04.12AlexForsternevermind, misunderstood
20:09.56deltaflyer4747is p!$$3d off :-/
20:11.55p3nguinSomeone once told me that it is better to be pissed off than to be pissed on.
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20:42.19okeip3nguin: hello, can u help me ? :)
20:45.17okeiguys what is TARGET0 funciton in this string? exten => _20[0-3],1,Set(TARGETNO=${EXTEN})
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20:46.04p3nguinTARGETNO is a variable in that line.
20:46.31okeip3nguin: yes but whats function in reality?
20:46.44okeiit's only formal?
20:47.18p3nguinShow me the rest of your dial plan and I'll try to figure out its purpose.
20:47.24WIMPyIt's just some variable.
20:47.47p3nguinI already said that, but apparently that answer wasn't good enough.
20:47.58DrDigitalanyone able to help me figure out what riser cards i need and what kind of ram to buy
20:48.00DrDigitalhttp://www.alvio.com/xABK_PID56007_S2891G2NR_tyan-computer_tyan-thunder-k8sre-s2891-s2891g2nr-socket-940-eatx-motherboard-oem.html&referrer=froogle
20:48.10DrDigitali have that in a 1U rackable systems case
20:48.36okeip3nguin: http://pastebin.com/DBqQMNAA
20:48.41okeiexample from tutorial
20:49.25p3nguinThey are using the variable to retain the original number that was called.
20:51.09okeip3nguin: retain? or determine
20:51.17p3nguinretain
20:51.48p3nguinIt is being set when the call comes in.  It is then being used later when the call goes to voicemail.
20:52.07*** join/#asterisk Mango (~Mango1234@S010620cf30c62cb6.vc.shawcable.net)
20:52.36WIMPy... assuming that extension and VM box are identical.
20:53.29p3nguinWell, they are doing it regardless of the end result being a failure or not.
20:56.31lcbsomeone, pls, could help me resolve this 'make: *** [all] Error 2' while installing on a ubuntu 11.04 server  ->
20:56.45lcbhttp://pastebin.com/B1S0nGvL
21:01.10lcbwait... i missed LibPRI install... probably due to that
21:01.55WIMPyDid you use the latest Asterisk?
21:02.16lcbWIMPy: i'm following "by the book" :)
21:02.26jmwpcIn my contest for inbound google voice calls, I can't seem to get my extension to ring more than 3 times. On the other end of the call, I can hear 2 more rings before being sent to google voice mail. I'm dialing 2 extesions (it did the same thing with a single extension): Dial(SIP/101&SIP/102,20,D(:1)) ... I have increased the timeout setting up and down with no change, what am I missing?
21:02.40*** join/#asterisk g0rdorin (~g0rdorin@46-116-34-84.bb.netvision.net.il)
21:02.44jmwpc*context*
21:03.33p3nguinThat's still just one extension, not two.
21:04.20jmwpcp3nguin: both are ringing :) I even tried a dialgroup... same thing happened. I can't seem to get the timeout to change.
21:04.37p3nguinShow me your dial plan.
21:04.40lcbWIMPy: this one 'svn co http://svn.asterisk.org/svn/asterisk/branches/1.8'
21:05.01lcbWIMPy: isn't that correct?
21:05.06jmwpcp3nguin: k... let me get it pasted
21:05.50p3nguinlcb: Don't use the svn unless you know what you're doing.  Use the current RELEASEd version.
21:05.50WIMPylcb: That's fine.
21:06.09p3nguinAnd if you have to ask if it's right, you obviously don't know what you're doing.
21:06.22lcbp3nguin: is not the first time i'm using it... but if say so.
21:07.02WIMPyIt's 1.8, not trunk.
21:07.04lcbp3nguin: i asked because as you know might be several sources of it, i don't mean on this software.
21:07.45lcbbtw, i'm doing it all as the book says
21:07.59p3nguinwimpy: That svn will check out the current releast of 1.8 rather than the current day's build?
21:08.26WIMPyNo, but it's supposed to be stable.
21:08.57WIMPyOnly trunk might not be. In theory that it, off course.
21:09.05WIMPyis
21:09.06lcbnow i forgot the release * to add it on svn co http://svn.asterisk.org/svn/libpri/tags/1.4.
21:09.16lcb:(
21:09.26*** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593)
21:09.34WIMPy???
21:10.00lcbWIMPy: to pull it from SVN is necessary that
21:10.46WIMPyNot sure what you mean.
21:10.58lcbcd ~/src/asterisk-complete/ | mkdir libpri | cd libpri/ | svn co http://svn.asterisk.org/svn/libpri/tags/1.4.<your version number>
21:11.04jmwpcp3nguin: http://pastebin.com/EfmvVeWT
21:11.41jmwpcp3nguin: (2 separate google voice accounts are being handled here)
21:11.45p3nguinThat'll never work with all those spaces in it.
21:12.36jmwpcp3nguin: the ones by the separator (comma)?
21:12.53WIMPyuses /branches/1.4
21:13.45p3nguinjmwpc: Correct.  You can't have those spaces in extensions.conf.
21:14.08jmwpcp3nguin: ahh... thanks... let me give that a try
21:14.27p3nguinexten => 123,1,Stuff()
21:14.46p3nguinprovided you have app_stuff.so, of course.  :)
21:15.01jmwpcp3nguin: Coding habit :)
21:15.41p3nguinlines 26 and 34
21:15.52p3nguinDelete those Answer() lines.
21:15.58Kaminekohello, im looking for someone with draytec ippbox 2820 experience. i have a problem to get sip trunk to my asterisk/frepbx box working. i always get 401/403 eeror messages
21:16.00p3nguinDon't answer before the dial.
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21:18.19jmwpcp3nguin: Remove it entirely? or put it somewhere else? (Just for reference... I'm working from this tutorial: http://pcprob.blogspot.com/2011/03/howto-use-google-and-asterisk-for-free.html)
21:18.37p3nguinRemove the Answer() lines.
21:19.52p3nguinEither use the SendDTMF() method, or use the Dial(D()) method.  There's no need to mix the two.
21:20.07p3nguinIf you use the SendDTMF method, you'll Answer(), then Wait(), then SendDTMF().
21:20.27p3nguinIf you use the Dial(D()) method, you'd do nothing else before the Dial.
21:20.52p3nguinnothing that changes the actual call, that is.  You can obviously change things like CID and whatnot.
21:21.09jmwpcp3nguin: so it rings longer now, but never answers. It keeps ringing to VM on the other end.
21:21.52p3nguinYou'll have to pick up the phone, of course.
21:22.36jmwpcp3nguin: :) Of course... I did, it was dead air, and kept ringing to VM on the calling phone (my cell)
21:23.08p3nguinI don't know what's going on with that.  It's as if the gtalk/jabber stuff isn't actually working right.
21:23.08jmwpcSo the other method would be Answer -> Wait -> SendDTMF -> Dial... correct?
21:23.33p3nguinright.  It's on the blog page you linked me to.
21:24.10jmwpcp3nguin: I'll give that a try again
21:24.27p3nguinAnd when you use the SendDTMF method, you don't need the D(:1) in your Dial().
21:24.58jmwpcp3nguin: got it.. .
21:25.21jmwpcp3nguin: I can just exclude that parameter then?
21:25.40p3nguinYou'll need to take it out of the Dial command.
21:26.15p3nguinI mean, you could leave it, but then people will hear "1" in their ears when you pick up your phone.
21:26.26p3nguinIt won't technically hurt anything.
21:26.47jmwpcp3nguin: I'll leave it in there for people I dont' like :)
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21:28.07lcbsuccessfully installed libpri-1.4.12 . again 'make' of dahdi is not finishing . same pastebin -> http://pastebin.com/B1S0nGvL
21:28.52lcb(long ago. i was wating for previous issue being done)
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21:31.13WIMPylcb: That dahdi version is most probably too old for your kernel version.
21:32.27lcbso WIMPy, http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ ?
21:33.22WIMPyEither that or take the trunk from svn if you feel comfortable with the latest.
21:33.46WIMPyShould I have said "Trying" the latest?
21:33.54lcbWIMPy: :)
21:33.58lcbdon't worry
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21:36.14lcbit seems to be 2.4.1.2+2.4.1/
21:39.30lcbno Errors & Warnings, Inc., on make
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21:40.12okeip3nguin:  penguin, when i trying to leave voicemail debug is http://pastebin.com/L075A9he here, when i have writed this string in voicemail.conf 1001 => 123456,Rati Jokhadze, why this message is right?
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21:41.28okeiand what is voicemail number?
21:41.31okei1001 this?
21:41.40lcbWIMPy: no errors at all. however, no hardware found. not worried by now because the modem i have in there for now is a data/fax and i'll buy another one.
21:42.18WIMPydahdi (or zaptel for that matter) has never supported modems.
21:42.22lcbwondering though why that data/fax was not found
21:42.29WIMPyThere used to be another channel for that.
21:42.29lcbahh ok
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21:43.18lcbWIMPy: thanks for the ""Trying" the latest"
21:44.19MangoI'm using Asterisk 1.8.4.4 and trying to subscribe to MWI from my DID provider.  I've added mwi => mango@inbound1/12345 to my sip.conf.  It sends the subscribe packet, then inbound1 replies with 401 Unauthorized, but my Asterisk never responds with the challenge.  It ignores the 401 Unauthorized and retransmits the original SUBSCRIBE packet a bunch of times.  Any ideas?
21:47.17jmwpcp3nguin: I think I got it. The main issue was probably the spaces. I went back to the D(:1) method since I lose google voice mail if I answer before dialing. I get 4 rings now, just about the same time google kicks to VM.
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21:57.28lcbshould i install samples? 'make samples'?
21:58.14WIMPyIf you want an asterisk that allows a lot more than you want it to: yes.
21:58.15lcbi still have 'sudo make config' but after make install i got that suggestion
21:58.28lcbyes. tks
21:58.40WIMPyDon't run it on the internet with the sample configuration.
21:58.53lcbok
21:59.10lcbWIMPy: keep sudoing, isn't?
21:59.16lcbsudo make samples
21:59.16WIMPyThe sample configs are good for reference, but you can look at them in the source dir.
21:59.24WIMPyyes
21:59.38lcbahh ok, so might be better to leave the defaults.
22:00.17WIMPyDefault means not supported in many cases.
22:01.16lcb'sudo make config' by the book... result: System start/stop links for /etc/init.d/asterisk already exist.
22:01.19WIMPyOnly configure those parts you actually need. And make sure they don't do things you don't want.
22:01.27lcbso just forget it. is configured
22:01.40WIMPySo where did they come from?
22:01.48lcbWIMPy: ok. i'll keep reading the book
22:02.02WIMPythought that was a fresh install?
22:02.23lcbWIMPy: i don't know... probably through any apt-get i did before... :( i don't recall it
22:03.16lcbWIMPy: :( probably i did it... because i am looking for a good program like this for two weeks
22:03.28WIMPyDid you remove the packages?
22:03.36lcblooks like i didn't
22:03.45lcbi'll go over again
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22:04.01WIMPyMight be a good idea to do so and the do the 'make install's again.
22:04.07WIMPythen
22:04.20lcbi thought i did an install in this machine i'm using now... :(
22:04.29lcbwith apt-get
22:04.55lcbyes, i will
22:05.49lcbfor now it looks great, smooth installation (besides that dahdi outdated)
22:06.56WIMPythat's why I usually try to avoid hardware, not supported by the standard kernel. But that isn't always possible here.
22:10.51lcbi'll install again tomorrow. by now apt-get --purge remove asterisk* cleaned it
22:11.23WIMPywill try todays version.
22:11.32WIMPyThe one from a few days back seems foul.
22:11.39lcbWIMPy: thank you very much for your time. i have to go. need to prepare some work for tomorrow. bye
22:13.05WIMPydefinitely needs somethign to detect the deth of an asterisk.
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22:21.20AviMarcusmunin?
22:21.28xbphi
22:22.30WIMPyMaybe it can be done in a channel as well. That wouldn't be too bad.
22:23.59WIMPymunin doesn't look capable of detecting a dead Asterisk.
22:24.18xbpi think you can detect by 5060
22:24.23xbptelnet to it see if sip is answering
22:24.32p3nguinSIP is UDP.
22:24.49xbphrm?
22:25.20xbpflorida bites today
22:25.42p3nguinSIP is UDP.  The telnet client uses TCP.  Ergo, telnet is useless to testing SIP.
22:25.57WIMPyI have a feeling it locks up when a peer dies during a call.
22:26.16WIMPyBut the last time it was IAX, now it was SIP.
22:31.30p3nguinSet(__myvar=myvalue)  <-- this is correct for indefinite inheritance of the variable, right?
22:31.58WIMPyyes
22:32.58p3nguinIs there any reason to only use _ instead of __ for that purpose?  I can't see any reason it would hurt to cause it to be indefinite rather than just once.
22:33.59WIMPyNFI. But I wouldn't see why you couldn't do it to all variables.
22:34.15p3nguinThe only reason I could come up with would be if you might need to test the variable for a null value later, and you don't want to explicitly unset it.
22:34.32WIMPyyes
22:39.19MangoSo what could cause Asterisk to ignore a 401 Unauthorized from another peer?
22:41.35p3nguinfailure to auth, perhaps?
22:42.10MangoPlease explain more :)
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22:43.14p3nguinLike, your device has not been preauthorized... so the 401 is the challenge to get your device to authenticate.  If it fails authentication, it'll generate a 404.
22:43.34MangoWell...my Asterisk doesn't exactly fail to authenticate, it doesn't even try.
22:43.53p3nguinIf it failed, it would have received a 404, so I know it didn't fail to auth.
22:44.02MangoRight.
22:44.15p3nguinI think it's a 404, anyway.  Maybe it's a 403.
22:44.19p3nguinI'd have to go look up the codes.
22:45.42p3nguinI don't know what would keep your system from trying to authenticate after getting the 401 from your peer, though.  That's the challenge.
22:45.51Mangoscratches his head.
22:47.13MangoI set up IP authentication.  Now I get a different error.
22:48.05MangoMe: SUBSCRIBE!  Remote: 200 OK!  Remote: NOTIFY! Messages-Waiting: yes!  Me: SIP/2.0 481 What the heck was that!
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22:48.33MangoActually I said "Call/Transaction Does Not Exist", not "What the heck was that!"
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