IRC log for #asterisk on 20121117

00:00.48*** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri)
00:08.53*** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com)
00:09.05dfgas-cr48dijib, hey, are you around?
00:20.22SeRiI AM ON A BOAT!
00:20.50p3nguinYou're on a boat?
00:21.13p3nguinTake a good look at you 'cause you're on a mother trucking boat?
00:21.52SeRiLMAO!!!!
00:22.10SeRiwaz up p3nguin !
00:22.21SeRiWhisky night brother!
00:22.34p3nguinI don't even have any beer.
00:22.40SeRi:(
00:22.47p3nguinI might have to get some.
00:23.00SeRicome over bro. I have plenty modelo and crown/JD
00:24.50p3nguinI'll be right over...
00:24.58p3nguinIt only takes 14 hours.
00:25.09SeRilol
00:26.00SeRibrb going to get me a sniffer and poor me some. brb
00:26.28*** join/#asterisk eit (~scott@173-165-131-2-utah.hfc.comcastbusiness.net)
00:29.48eitWe are using the asterisknow version of freepbx, asterisk, 1.8.11-cert8.  We are having an issue with the asterisk manager where it will work great for a number of hours (5-10 hours) then suddenly asterisk will stop send replies back in AMI interface.  I've checked what I can think to check.  Does anyone have any suggestions on to try?
00:32.07navaismoeit, what error or message are you receiving, what show the debug when it stop
00:33.17eitI have not been able to find any message or error related to it stopping.  It just suddenly is reported as having stopped, by the users.  Then I go look at the logs and at some point the ami seems to have just stopped reporting.
00:33.34*** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806)
00:34.34*** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1003:215:5dff:fe07:4806)
00:38.33navaismoactivate the debug on your asterisk and then activate the full log. When the issue appear check the full log at estimated hour and try to check for errors.
00:39.09dfgas-cr48ok when making a phone call from a sip phone if i hang up it keeps call them
00:39.20navaismovery difficult to diagnose with nothing also you may want a trace wit tcpdump and the consuption of memory and cpu at that moment
00:39.20dfgas-cr48any ideas on how to fix?
00:39.24navaismoeit, ^
00:40.04navaismodfgas-cr48, sip to sip or sip to pstn or pstn to sip?
00:40.23eitnavaismo: I have the full log collecting info now.  I am not sure how to enable debug for logging purposes.  Do I just enable debugging in the asterisk CLI?
00:40.49SeRip3nguin: voip.ms finalize my port today... hells yeeea.
00:41.44dfgas-cr48put going call from sip phone to sip DID line
00:41.50dfgas-cr48what is pstn?
00:42.31p3nguinSo you're calling outbound from a phone on your asterisk system to your ITSP and back inbound to your DID?
00:42.34p3nguin~pstn
00:42.34infobotpstn is, like, Public Switched Telephone Network, or "please stop the nonsense"
00:42.40navaismoeit, yes core set debug 3 from cli
00:42.50dfgas-cr48p3nguin, no
00:42.59dfgas-cr48calling my wifes cell
00:43.07eitOkay done.
00:43.21p3nguinThat has nothing to do with your DID.
00:43.25dfgas-cr48when it starts to ring and i hang up it keeps an open connection
00:43.26p3nguin~did
00:43.26infobotmethinks did is Direct Inward Dialing, or just a phone number
00:44.03dfgas-cr48well i figured that much, sure it has something to do with way asterisk is setup
00:44.08navaismoeit, now just wait for the issue, but if you have a lot of calls your log will be growing then you can gzip it
00:44.42p3nguinIt's probably a problem with your phone.  My Android phone's native SIP UA does the same thing.
00:45.12dfgas-cr48p3nguin, dijib was helping set it up last night and I know its not fully setup but i am trying to get some work done
00:45.22dfgas-cr48p3nguin, that is exactly what i am using
00:45.32dfgas-cr48jellybean
00:45.38dfgas-cr48did you find a way around it?
00:45.42p3nguinYep.
00:45.52p3nguinI stopped using the native SIP phone.
00:46.07dfgas-cr48ahh, so get a different app?
00:46.09p3nguinI use cSipSimple on it, now.
00:46.24dfgas-cr48and it works great?
00:46.51p3nguinIt has pretty good android integration, so I use the native dialer and it pops up and asks if I want to dial out using VoIP or cell phone.
00:47.21p3nguinIt's fairly good with integration of logs and everything.
00:47.23dfgas-cr48well 3 out of the 5 phones are just android phones using wifi
00:47.49p3nguinI don't even have to launch the app to make calls, just use the regular dialer.
00:47.57eitnavaismo, I'll have to wait a few hours to restart things, hopefully we can catch the cause of it.
00:48.14p3nguinOf course you have to make sure it is configured and running in the bg.
00:48.20dfgas-cr48p3nguin, sweet, i will try it quick
00:48.35dfgas-cr48is it a paid app?
00:48.52p3nguinI think I got it for free in the market.
00:48.54navaismoeit, also generate a trace with tcpdump/wireshark in order to see thing in your network
00:49.03dfgas-cr48k
00:49.24*** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-baxlvzwlgvprlowz)
00:49.31dfgas-cr48if it was a paid app it was going tricky on the kids phones
00:49.58*** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell)
00:49.59*** mode/#asterisk [+o Qwell] by ChanServ
00:50.01p3nguinIf you pay for it on one phone, you can always copy the apk to your other phones.
00:50.15dfgas-cr48true
00:50.17*** join/#asterisk ruied (~AndChat66@217.129.155.146)
00:53.53p3nguinI'm about to test out MizuDroid.
00:54.00carrarOK
00:54.06carrarPlease proceed
00:54.10p3nguinThey say, "It's not a SipDroid clone."
00:54.30carraroh wait, please wait 10 more 6 mor emins
00:54.34carrarerr
00:54.37carrar6
00:54.39carrarsoryr
00:54.48carrarOnly start new items on a even hour
00:55.01p3nguinI did test some others, and they either didn't work right or didn't have good android integration.
00:57.35SeRiok back
00:57.59dfgas-cr48p3nguin, if you find any that are top notch please let me kknow
00:58.48SeRip3nguin: uri
00:59.09p3nguinBetter than a uti, I suppose.
00:59.31SeRilol
00:59.32dfgas-cr48p3nguin, still does it :(
00:59.35*** join/#asterisk jblack (~jblack@pool-108-57-61-167.sctnpa.east.verizon.net)
00:59.36dfgas-cr48hmmm
00:59.43SeRiI want to test g722 on the new system
01:00.35WIMPyYou got the G.722 fewer?
01:00.40SeRinew case I am buying http://www.logicsupply.com/products/c159
01:00.42WIMPyGood morning, BTW.
01:00.52SeRiWIMPy: No I have a new system.
01:01.00SeRi:)
01:01.14SeRigm!
01:01.20dijibdfgas-cr48: are you here again?
01:01.21SeRiWIMPy: no whisky yet?
01:01.24dijibhey SeRi
01:01.36SeRidijib: waz up brotha
01:01.52WIMPyNope. Should find something to eat.
01:01.58dijibso i called the isp and the reason the .wav files wernt going through was becuase of file sie
01:02.06dijibcuz ive had a few go through today
01:02.06SeRilol.... sounds like a good idea
01:02.21SeRidijib:  told ya
01:02.38dijibim wondering though how would i adapt it to rar tar or zip the recording prior to emailing it
01:02.38SeRidijib: jump in
01:02.40dfgas-cr48dijib, sup
01:02.56dijibso i think i kindof got it working last night dfgas-cr48
01:03.03dijibSeRi: on my way
01:03.16WIMPyHmm. Are digium asking strange questions on a monthly basis now?
01:04.05dfgas-cr48outgoing wasn't so i fixed the number you had in it and got it
01:04.22SeRiWIMPy: lol... I got an email from them too...
01:04.34p3nguinI'm sure everyone did.
01:05.00dfgas-cr487digit dialing doesn't work
01:05.13WIMPyProbablt. But I usually find the question they ask a little strange.
01:05.16dijibno we didnt set that up... need
01:05.34p3nguinYou get what you pay for.
01:05.58*** join/#asterisk LiuYan (~LiuYan@211.154.128.171)
01:06.03dijiblol
01:06.53dfgas-cr48dijib, also having an issue of when i hang up on my sip phone it just keeps ringing
01:07.07SeRilol
01:07.41*** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26)
01:07.44SeRiNov 16 18:03:47 pbx local0.warn asterisk[1947]: WARNING[2010]: res_musiconhold.c:659 in monmp3thread: poll() failed: Interrupted system call
01:08.15SeRip3nguin: I keep getting that every so often
01:08.22dijibhave to give me a min here talking to a few people
01:09.24dfgas-cr48i got the second DID line added too
01:10.48dijibyeah you just duplicate those two sections in sip.conf
01:11.06dijibyou should add an extension to y sip url in the dailplan for my conference dfgas-cr48
01:11.10p3nguinThat's not right.
01:11.21p3nguinYou don't change sip.conf when you add DIDs.
01:11.44dijibyou do if its in seperate account in voipms like he has
01:11.52p3nguinNope.
01:12.14SeRip3nguin: jump in
01:12.21SeRiis just me right now.
01:12.54p3nguinAnd if you've got him using a new account for every phone number, he needs a new consultant.
01:13.33dfgas-cr48how do i do that?
01:13.41p3nguinHow do you do what?
01:13.48SeRip3nguin: you forgot the number
01:14.01p3nguinOh!  Yeah I did.
01:14.08dfgas-cr48btw inbound from does the same thing when i hang up on my cell it just keeps rining the extention
01:14.54dfgas-cr48extention to extention is the same as well
01:21.01dfgas-cr48dijib, are you there?
01:22.26dijibsry im on the phone
01:22.40dfgas-cr48k i didn't know if inet crapped out
01:24.37dijibya no got busy
01:24.40dijiband now im free
01:24.50*** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com)
01:24.58dfgas-cr48with the 1111 is that just for my ivr?
01:25.12dijibno it just like a memo thing
01:25.26dfgas-cr48ahhh
01:25.55dfgas-cr48is there a way i can set one up to have a pin that when i dial it put in a pin it will record new ivr message?
01:26.03dijibso i used that to make the IVR.wav file
01:26.16dijiblet me just readback what the issue was
01:26.31dijibSeRi: where u at?
01:26.52dijibnot sure what this is: 20:06 < dfgas-cr48> dijib, also having an issue of when i hang up on my sip phone it just keeps ringing
01:27.47SeRidijib: at home.
01:28.04dijibback in 2663
01:28.11p3nguinssh
01:28.12p3nguin^
01:28.17SeRi2600Hz
01:28.19SeRiLOL
01:28.20dijibdfgas-cr48: can i remotely connect again so i can make you an extension to my sip uri
01:28.41dijib2600 is tandem no?
01:28.54dijibhuh p3?
01:29.15dfgas-cr48yah
01:30.17dijibstart teamviewer and tell me the password in a pvt msg
01:30.31p3nguinssh
01:30.41dijibssh you
01:30.47p3nguinWhat kind of Linux admin uses teamviewer to admin a server?
01:30.56dijiblol
01:30.59SeRiwwtf? teamviewr for a linux system?
01:31.03dijibone that runs it on ubuntu
01:31.03p3nguinThat's pretty lame.
01:31.03dijiblol
01:31.12dijibits the application of the box
01:31.13p3nguinlrn2ssh
01:31.19SeRilol
01:31.22dijibits more for fun than anything
01:31.27p3nguinlrn2vim
01:31.38dijibnano is tough
01:31.55SeRiVIM the mark of the BEAST VIVIVI
01:32.07dijiblol
01:32.13dijibi thought it was imporved
01:32.13carrarvim rocks
01:32.19SeRihells yea
01:32.29SeRivi4life putos
01:32.47SeRiLOL
01:33.08navaismoO_o
01:33.54dijibi would rather use vi
01:34.55navaismowondering if the P word if for all of us if so .|.
01:35.56dijibpunda?
01:44.55SeRihttp://www.logicsupply.com/products/c159
01:45.10SeRihttp://www.jetwaycomputer.com/NF9H.html
02:03.43*** join/#asterisk v4x (~v4x@unaffiliated/v4x)
02:14.49dfgas-cr48nano all the way :P
02:20.15dijibhow do i dial a sip uri BTW
02:28.49p3nguinReal men use echo and cat.
02:29.16p3nguinYou can dial a SIP URI from the astcli.
02:29.50p3nguinseri: I got my beerz.
02:33.13SeRip3nguin: uri
02:33.18dijiburi
02:33.35p3nguinUTI
02:33.40SeRiLTI
02:33.40dijibhuh?
02:33.45p3nguinLTS
02:33.51SeRiLSD
02:33.52dijiblong term support
02:33.54dijib:D
02:33.59dijibjalapinios
02:34.00p3nguinNAACP
02:34.04SeRiSTD
02:34.31dijibthis server had some notices and warnings... plus he says there is a dialing issue
02:34.43dijibit just rings and rings even though it does dial out or something
02:34.48dfgas-cr48boo
02:34.48dijibneed dfgas-cr48 to explain better
02:34.50p3nguinI don't know why they stopped selling Silver Thunder at the store here.  For a malt liquor, it was pretty good.
02:34.56dijibhandle_request_subscribe: Received SIP subscribe for peer without mailbox: 13
02:34.59dijibwhats that?
02:35.10p3nguinIt's a malt liquor by Pabst
02:35.12dijibbuy rolling rock
02:35.29dijibhandle_request_subscribe: Received SIP subscribe for peer without mailbox: 13
02:35.30SeRibuy crown to tha mofuking royal
02:35.33dfgas-cr48dijib, p3nguin told me to try different dialer app on my phone
02:35.36dijibhandle_request_subscribe: Received SIP subscribe for peer without mailbox: 13
02:35.40p3nguinI've got some King Cobra tonight.  It's by Anheiser-Busch, but I'll still drink it.
02:35.47dfgas-cr48dijib, it didn't change anything however
02:35.48dijibNov 16 20:33:01] WARNING[5082][C-00000001]: pbx.c:6230 __ast_pbx_run: Timeout, but no rule 't' or 'e' in context 'default'
02:35.50p3nguinNo I didn't.
02:35.58p3nguinI did not say a different dialer app.
02:36.04dijibwhat?
02:36.10dijibfuck cr
02:36.13dijibmakes me black out
02:36.27SeRiyou pussy
02:36.35dijib:(
02:36.41SeRilolz
02:36.42dijibin my mouth
02:36.46p3nguinReceived SIP subscribe for peer without mailbox: 13    <--- the problem is a device named "13".
02:36.58p3nguin13 is not a good name for a phone.
02:37.01p3nguin~devicenames
02:37.02infobotDevices, extensions, and people should be entirely abstracted.  Extension numbers are applied to people, and people are applied to devices.  This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address.
02:37.02dfgas-cr48dijib, from extention to extention, sip phone to world and world to sip phone all does the same
02:37.32dijibhmm
02:37.44dfgas-cr48p3nguin, well you told me that that app you said earlier worked good for you
02:37.48p3nguinseri: So, when you say URI, what do you mean?
02:37.50dijibdfgas-cr48: are you like some asterisk pro or something in hiding?
02:37.59p3nguindfgas-cr48: I said a differnt PHONE app.
02:38.00dfgas-cr48that it fixed your issue
02:38.11dfgas-cr48however for me it didn't work
02:38.15dfgas-cr48yah
02:38.15p3nguinThe dialer has nothing to do with it.
02:38.19dfgas-cr48thats what i mean
02:38.21dfgas-cr48sorry
02:38.36SeRip3nguin: is a sort way to say IP call me.
02:38.36dfgas-cr48the csipsimple
02:38.44SeRiI know is not the right term :)
02:38.56dfgas-cr48dijib, no why? lol
02:39.10dijibdial 12663
02:40.01dfgas-cr48nothing
02:44.40dijibyou were in... ok dial the 2892985605 x2663
02:45.13dfgas-cr48from cell or through asterisk?
02:45.47dijibwhat about from the SIP client
02:47.03dfgas-cr48can u hear me?
02:47.23SeRiif you have a sip client and is using asterisk you cant dial sip uri unless is recognize by asterisk or the client
02:47.53SeRidfgas-cr48: no
02:47.56SeRiwe cant hear you
02:48.50dfgas-cr48hmmm
02:49.18dfgas-cr48i can hear about 5 to 10 seconds worth
02:49.26dfgas-cr48then nothing
02:59.46p3nguinI'll fix it for you.
02:59.56p3nguinI'll even give you a discount.
03:00.35dfgas-cr48lol
03:00.35dijib21:38 < p3nguin> The dialer has nothing to do with it.
03:00.36dijib21:38 < dfgas-cr48> thats what i mean
03:00.36dijib21:38 < dfgas-cr48> sorry
03:00.36dijib21:38 < SeRi> p3nguin: is a sort way to say IP call me.
03:00.36dijib21:38 < dfgas-cr48> the csipsimple
03:00.38dijib21:38 < SeRi> I know is not the right term :)
03:00.41dijib21:38 < dfgas-cr48> dijib, no why? lol
03:00.43dijib21:38 < dijib> dial 12663
03:00.45dijib21:39 < dfgas-cr48> nothing
03:00.48dijib21:44 < dijib> you were in... ok dial the 2892985605 x2663
03:00.50dijib21:44 -!- aidinb [~aidin@unaffiliated/aidinb] has quit [Ping timeout: 260 seconds]
03:00.53dijib21:45 < dfgas-cr48> from cell or through asterisk?
03:00.56dijib21:45 < dijib> what about from the SIP client
03:00.58dijib21:46 < dfgas-cr48> can u hear me?
03:01.01dijib21:47 < SeRi> if you have a sip client and is using asterisk you cant dial sip uri unless is recognize by asterisk or the client
03:01.04dijib21:47 < SeRi> dfgas-cr48: no
03:01.06dijib21:47 < SeRi> we cant hear you
03:01.09dijib21:48 < dfgas-cr48> hmmm
03:01.11dijib21:49 < dfgas-cr48> i can hear about 5 to 10 seconds worth
03:01.14dijib21:49 < dfgas-cr48> then nothing
03:01.16dijib21:59 < p3nguin> I'll fix it for you.
03:01.19dijib21:59 < p3nguin> I'll even give you a discount.
03:01.23p3nguinI'll do it for the low, low price of just $59.95.
03:01.34dfgas-cr48geebus
03:01.34dfgas-cr48lol
03:01.51dfgas-cr48dijib, did you see the voicemails won't play
03:02.08p3nguin(2102.02)  -!- Irssi: Pasting 10 lines to #asterisk. Press Ctrl-K if you wish to do this or Ctrl-C to cancel.
03:02.20p3nguinPRESS CONTROL FUCKING C
03:02.34SeRiLMAO
03:02.38dfgas-cr48lol
03:02.58dijib22:00 < dijib> 21:47 < SeRi> we cant hear you
03:02.58dijib22:00 < dijib> 21:48 < dfgas-cr48> hmmm
03:02.58dijib22:00 < dijib> 21:49 < dfgas-cr48> i can hear about 5 to 10 seconds worth
03:02.58dijib22:00 < dijib> 21:49 < dfgas-cr48> then nothing
03:02.58dijib22:00 < dijib> 21:59 < p3nguin> I'll fix it for you.
03:03.00dijib22:00 < dijib> 21:59 < p3nguin> I'll even give you a discount.
03:03.03dijib22:01 < p3nguin> I'll do it for the low, low price of just $59.95.
03:03.05dijib22:00 < dijib> 21:47 < SeRi> we cant hear you
03:03.07*** kick/#asterisk [dijib!~pabelange@asterisk/contributor-and-bug-marshal/pabelanger] by pabelanger (ass)
03:03.18SeRirofl!
03:03.21SeRiNICE!
03:03.37p3nguinHe's even using irrsi, which has paste flood control.
03:04.33SeRipas to that mofking bin!
03:07.04*** join/#asterisk winery (~dijib@208-96-84-35.eastlink.ca)
03:25.57SeRidfgas-cr48: is your stuf working now??
03:49.03*** join/#asterisk BrettB (Brett@75.92.223.35)
03:49.35BrettBhi
03:59.40BrettBdo any tools exist for decrypting an xml provisioning file?
04:00.02jpsharpdecrypting or decoding?
04:00.40BrettBdecrypt
04:02.30BrettBeveryone says its encrypted
04:03.02jpsharpFor what device?
04:03.36BrettBspa2102
04:07.19jpsharpDunno off the top of my head.
04:09.20BrettBthe file isn't completely unreadable. the first string is "sipura spa+"
04:10.15jpsharpIf it is anything like other Cisco devices, the file isn't really "encrypted" but just squished into a weird binary format.  But I could be wrong.
04:10.31BrettBeven elcomsoft doesn't know what format it is
04:11.36BrettBso if its just in this "weird binary format" how would i go about reading it?
04:12.39jpsharpI hate to sound like Captain Obvious, but find a tool that knows how to read it.  Unless you want to pick it apart yourself with a hex editor.
04:12.59jpsharpReverse engineering, while fun, is tedious.
04:13.49WIMPyOr buy something that comes with documentation next time.
04:14.39BrettBthe documentation doesnt help
04:15.02*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
04:15.47jpsharpCisco likes to take mail text files and force users to wash them through an encoding program before the file can be read by the device.
04:15.55jpsharpplain text, not mail text.
04:16.06jpsharppokes the dead brain cells.
04:16.15infinity1how do you call a macro in the old language from AEL
04:17.15BrettBif i read it from a hex editor the first line reads "SIPURA SPA +&   506D6A7A"
04:17.17infinity1i tried it using the &.. syntax and the console is using GoSub
04:17.20BrettBthe rest is junk
04:19.21jpsharpIt might even be encrypted after that point.  I know the old ATA-186s had the option of encrypting the config file using a shared key stored on the device and server.
04:20.01BrettBhow complex can the key be?
04:22.43*** join/#asterisk hachque (quassel@2600:3c01::f03c:91ff:fe96:5060)
04:23.44infinity1anyone know how i'm getting ~~s~~ in this ?
04:23.44infinity1<PROTECTED>
04:28.14BrettBi hate this thing. i've opened up the box to look for a reset jumper but it doesn't have one
04:46.08jpsharpBrettB: Have you done the factory reset steps?  Dialing **** then 73738# ?
04:47.27slav3_kitten^^
04:52.04BrettByea. asks for password
04:52.13*** join/#asterisk xnixan (~xnixan@unaffiliated/xnixan)
04:53.59slav3_kittenjpsharp, would you recommend any small phones around the size of a cisco 7911 that has caller id and good documentation on how to setup features an such
04:55.54SeRihttp://www.supermicro.com/products/chassis/1u/502/sc502l-200.cfm
04:56.28slav3_kittenSeRi, that's a sexy case right there
04:57.02slav3_kittentoo bad those thigns are so spendy
04:57.39slav3_kittenthat point you may as well just throw a netbook into a rack case
04:58.02SeRislav3_kitten: I am rack mounting my atom server
04:58.20slav3_kittenyou have one of those?
04:58.21SeRip3nguin: iptel.org
04:58.31SeRislav3_kitten: Yes.
04:58.49slav3_kitteni hate you now
04:59.05slav3_kitteni have a rackables 2u dual opteron
04:59.28p3nguindijib: Set your nat value to nat=force_rport,comedia
05:00.03SeRislav3_kitten: to much fucking power and heat. I have a quad core atom server with quat 1gE 16GB RAM.
05:00.15SeRiall running 12v :P
05:00.28slav3_kittenyea my rackables never gets turned on
05:00.42slav3_kittenbut when i really need to mash numbers it's on standby
05:01.04SeRiwhat do you use for computing?
05:01.07SeRiyou code?
05:01.12p3nguina computer.
05:01.31SeRiwrong. You need code to cunpte :P
05:01.36slav3_kittenwhat do you mean SeRi?
05:01.43SeRis/conpute/sompute/
05:01.56slav3_kittenmy main computer is a toughbook cf-29
05:02.04slav3_kittenasterisk is running on a cf-28
05:02.22slav3_kittenrackables is there for heavy computing tasks i have not done in a long time sadly
05:06.17SeRiMine is my asterisk server
05:07.23SeRislav3_kitten: join the conf
05:07.52*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
05:08.01slav3_kittenwhat conf?
05:08.17SeRidijib:
05:08.20SeRidijibs
05:08.43slav3_kittenbut i'm old and tired
05:10.00SeRipuss
05:10.05p3nguiny
05:10.48slav3_kittenwell it's also masturbation night, doubt you guys wanna hear that
05:11.18SeRithats fucked up.
05:11.49SeRilol
05:12.04slav3_kitteneveryone beats it
05:12.04slav3_kitteneven your mom an dad
05:16.07SeRiyour dad is gay
05:16.18SeRilol
05:16.49p3nguinIl y a plus d'une façon de peler un chat.
05:17.14SeRinew orleans?
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05:21.32slav3_kitteni think that's how the french say they surrender
05:23.33WIMPyhttp://chrismoyles.net/soundvault/soundvault.php?fileid=307
05:24.14WIMPyFot thos who want to learn abeout the art of wankee.
05:24.50WIMPyWow that e came several words too late.
05:25.27p3nguinseri: En Inglés, por favor!
05:26.33Nivexslav3_kitten: surrendering involves a cat?
05:26.34SeRino hablo el ingles!
05:27.02SeRiROFL!!!!!!!
05:27.03slav3_kittenNivex, a white cat
05:27.25NivexI just ran what he said through the translator. :)
05:27.28p3nguinTengo un gato en el pantalones.
05:27.52NivexI got most of that. Remind me what "Tengo" is
05:27.58SeRite muerde!
05:28.10NivexUn fiesta en el pantalones!
05:28.29SeRiNivex: *Una*
05:28.43SeRiNivex: *los*
05:28.56Nivex"Lady, I only speak two languages: English and bad English."
05:29.09SeRiNivex: LOL
05:29.20Nivexname the movie
05:29.36slav3_kittenrush hour
05:29.41Nivexbzzt
05:29.58slav3_kittenlethal weapon
05:30.02Nivexbzzt
05:30.16SeRiDebie does Dallas!
05:30.23slav3_kittendie hard
05:30.34slav3_kittenoh fuck it's fifth element
05:30.34Nivexwell at least with Die Hard you've got the right actor.
05:30.37NivexDING!
05:30.52SeRislav3_kitten: dont cheat you fucker
05:30.59slav3_kittenhuh/
05:32.21SeRiLOL
05:33.24slav3_kitteni'm so tired
05:43.05SeRislav3_kitten: go to bed.
05:43.24slav3_kitteni am in bed
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06:18.12SeRiCOMCAST ACCOUNT MGMT0000000000
06:19.11jpsharpNot comcast!
06:19.50jpsharpAnd Fifth Element is an awesome movie.
06:20.38SeRijpsharp: LOL
06:20.44SeRiI hate comcast
06:21.14jpsharpThey're tolerable if you have their business service.
06:21.19jpsharpOtherwise you're boned.
06:22.11jpsharpAnd now I'm going to have to fire up netflix and watch Fifth Element.
06:22.30SeRilol
06:23.49jpsharpI was with my wife somewhere the other day and she got asked for ID. She took it out and said "Leeloo Dallas Multipass".  I cracked up, the clerk looked at her like she had 10 heads.
06:25.08SeRilmao
06:29.32jpsharpReason #3947324932349 i love that woman.
06:29.56Nivexyes, that is made of win
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07:55.37SeRihttp://www.cyborggaming.com/prod/rat9.htm
08:09.55*** join/#asterisk ChrisInSydneyToo (~Chris@60-242-81-231.tpgi.com.au)
08:12.04SeRip3nguin: http://pastebin.com/JkSNDrHk
08:26.42ChrisInSydneyhmm. Just did a core reload on 1.8.14.0 on 1 64bit Centos 5.something. Whole Asterisk process dumped and started again !
08:31.38SeRithats not good
08:32.30ChrisInSydneynope.
08:32.45ChrisInSydneyits on a Xen VM, but still :-/
08:33.17ChrisInSydneyoh well, later. The process has been running for days without a core reload, but plenty of dialplan and sip reloads
08:33.36ChrisInSydneyback to the text editor
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08:47.45SeRi~book
08:47.45infobotAsterisk: 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
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09:01.53ectospasmChrisInSydney: if the segfault occurs again, try upgrading to Asterisk 1.8.18.0.  If it crashes again after that, file a bug.
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09:33.57ChrisInSydneyectospasm: Cheers. I managed to get it to do the same thing once or twice on 1.8.09.something Never though anything of it. But when I get a moment, I'll see what its doing.
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09:34.55ChrisInSydneygot some patches (hacks)  I have applied to ~14.0. Haven't created the .diff files yet
09:35.00ectospasmChrisInSydney: if you do get the segfault again, follow this:  https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
09:35.15ChrisInSydneyk
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09:35.27ChrisInSydneybookmarked :)
09:39.41ectospasmyeah, and beware if you're using nonstandard patches, they're not likely to be supported by the devs
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11:05.07ChrisInSydneyectospasm: The patches are bits of code I have added to the dial, pickup, followme as well as some stuff I have done to app.c and features.c, but I dont suspexct that they are the culprits. Mostly they make available some more channel vars . pretty simple stuff, but I'll test without prior to posting any bugs
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11:05.47ectospasmChrisInSydney: yeah, good way of being not supportable.
11:06.41ChrisInSydneytrue, but the only way to get some behavior from the dial app, and to be able to pick up calls on partial matches
11:07.21ChrisInSydneyectospasm: Main motivation is that I need some customisation to the dial apps privacy options.
11:08.05ChrisInSydneyfix it so that I can specify sound files and the priv-accounce directory using  chan var
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11:11.31ectospasmChrisInSydney: you can't do that with vanilla dialplan or AGI?
11:12.14ChrisInSydneynot without alot more code in the dial plan
11:12.50ChrisInSydneythere is a built in privacy function. dial(something,sometime,p)
11:13.19ChrisInSydneybut all the sound files / recording directory are hard coded
11:13.44ectospasmChrisInSydney: not really, try asterisk.conf to change it.
11:14.30ChrisInSydneyyup. they are hard codec in the C "stuff" in ~/apps/app_dial.c
11:15.08ChrisInSydneyI have simply liberaated those settings and they now read a channel var if set, otherwise its the default behavior
11:15.21ectospasmChrisInSydney: so submit a patch
11:15.43ChrisInSydneyI've been testing and, when I make it all work properly, ill do such a thing
11:17.03ChrisInSydneyits a feature expansion, so I'll see how I go, if they dont want it, I'll still make it available on the interwebs somewhere out in the googles ;-)
11:17.19ChrisInSydneymight need some help in checking and submitting
11:18.45ChrisInSydneyno diffs as yet, and heaps of comments in there so I can see what I have done, but I've used // comments so I can find things I have done and probably I'll need to fix up the code so it is consistent with how the rest is written
11:19.33ectospasmChrisInSydney: remember, it'll have to go into trunk, not a release.
11:19.41ChrisInSydneybut it works and the *chars are properly referenced. I remember that from...when was it ??? 198something
11:20.06ChrisInSydneythats the sort of stuff I'll need help with
11:21.11tompawAve.
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11:22.11tompawIs there a way to use wildcards with confbridge kick? I need to be able to kick certain type of users from the conference.
11:22.12ChrisInSydneyToofell off
11:22.19tompawLike SIP/exttype*
11:25.36ChrisInSydneyActually, one thing, Is there any issues with using MySQL stored procedures to return values from ODBC queries ??
11:26.57ectospasmno issues that I know of, ChrisInSydney...  as long as unixODBC (or whatever ODBC connector you're using) doesn't have a problem with it.
11:27.11ectospasmI don't know exactly though... are you running into a particular issue?
11:29.56ChrisInSydneyNot really, I'm doing a complete overhaul of our systems. I have been using AstDB, but its a pain in the donkey to read / write to from an external app. I'm moving much of the stuff I store there, such as voicemail wait times etc over to MySQL so I can put a web front end on it
11:30.55ChrisInSydneyI figured SPs might be more efficient, however, i have been told I should'nt go there
11:31.03WIMPyPretty easy to use AMI for that.
11:31.26ChrisInSydneyWIMPy, heah, but its still slow, at least I find it that
11:31.54WIMPyHow many changes to the DB do you have from web per second?
11:32.20ChrisInSydneyI did see a patch for MySQL replacing the Berkley internal thing, but it was for 1.4 and I couldnt fight with it
11:32.30WIMPyBut I recently decided that having hints for the CF states would be a good idea. So there's no way around that for me any more.
11:32.54WIMPyWell, it has been replaced anyway.
11:32.54ChrisInSydneybrb, coffee has brewed
11:33.09ChrisInSydneyjust what I need at 10:30pm to get a good nights sleep :-/
11:33.25ChrisInSydneyyou using ODBC too ?
11:33.36ChrisInSydneyCF states ??
11:34.20WIMPyCall Forwarding states.
11:34.41WIMPyI thought I should be able to map them to BLF style buttons.
11:35.24ChrisInSydneycool
11:36.22WIMPySo I have to set custom device states to (de)activate them.
11:36.39WIMPyJust the destinations ant time stay in AstDB.
11:36.57ChrisInSydneysame here. I'm working a presence / profile solution to do call forward, call queue, voicemail and a mini AA, based on what you want to set your "presence" to be
11:37.22WIMPyWell, the device states are just AstDB entries as well, but you need to set them via the device state mechanism in order to get the phone notified.
11:37.32ChrisInSydneyuber flexible, but the tables are a little unwieldy
11:38.12ChrisInSydneydevstate(custom:) -> MySQL table would be a cool patch
11:38.24ChrisInSydneywho's up for it Xp
11:38.44WIMPyWouldn't help.
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11:40.32WIMPyThere shouldn;t be an issue writing to AstBD directly. But you want Asterisk to know about the change so it can send a message out immediately to a phone having subscribed to the state.
11:48.37tompawErm... why would Asterisk automatically kill an AMI telnet session upon placing a call? :/
11:48.37Nuggettelnet is eeeeeeevil!
11:49.03WIMPyYou found a bug?
11:49.55tompawI wish I hadn't. But I'm on 10.8, so I'll try it on 10.10
11:50.32tompawWhich file do I have to copy from 10.8 sources to keep my menuconf selections?
11:50.34WIMPyI know I've seen somethg about a bug closing AMI sessions, but I can't remember on what action.
11:51.02WIMPymenuselect.makeopts
11:51.08tompawOh crap, there's already 11 out :/
11:51.12tompawWIMPy: thx.
11:51.23tompawIs Asterisk turning into Firefox?
11:51.32WIMPyAlready?
11:51.37WIMPyKind of :-)
11:52.12WIMPyInstead of Asterisk 2 we're now waiting for Asterisk 100 I guess :-)
11:52.24WIMPyOr maybe Asterisk2 0.1?
11:52.34WIMPyAsteriskNG 0.1?
11:52.39WIMPyscnr
11:52.41tompawI wonder how many things will break during 10 -> 11 upgrade.
11:53.12WIMPyI don;t know how interesting your config is :-)
11:53.30tompawOnly interesting parts are ConfBridge
11:53.56WIMPy10 already has the new one.
11:54.03WIMPy(or had)
11:54.15tompawAnd ChannelRedirect, nothing else out of ordinary.
11:54.30tompawSo is there a chance it will "just work" (c) Steve Jobs?
11:54.31ectospasmAsterisk 2.0 was going to be built for Windows!
11:54.57ChrisInSydneyectospasm: noooooooooooooooooooooooooooooooooooooooo!!!!!!!!!!!!!!!!!!!!!!
11:55.03ChrisInSydneytell me its April 1
11:55.16ectospasmChrisInSydney: (don't worry, it was an old April Fools' joke)
11:55.24ectospasmfrom like 2005
11:55.26ChrisInSydney:D
11:55.32WIMPySince 1.8 the relevant changes to me have all been internal.
11:55.57tompaw"Macro has been deprecated in favour of Gosub."
11:56.05tompawThis April Fools too?
11:56.34WIMPyNope. But it still works and probably will do so for quite a while.
11:56.48WIMPyBut I think it has beendeprecade for quite a while already.
11:56.56WIMPyurgs
11:57.03WIMPybeen deprecated
11:57.04tompawNice wiki BTW. I wonder when it's gonna beat voip-info.org in Google results :)
11:57.14[gnubie]waves
11:57.36WIMPy[gnubie] that's a nice nick :-)
11:59.00tompawWhy did you change the download file names to 'current'? What was wrong with the full version number?
11:59.07[gnubie]i am having an issue after the callee accepted the incoming call because the caller is still ringing. after the 1 minute timeout, the caller will reach the voicemail while the callee will be disconnected. my call flow is illustrated here => http://imagebin.org/235688
11:59.07tompawstarts to feel outdated.
11:59.13[gnubie]WIMPy: thank you! ;)
12:01.19WIMPy[gnubie]: Which VM (where)?
12:01.24[gnubie]when the caller reached my asterisk box, the caller will hear and automated attendant + ivr. if the caller did not press any extension number until timeout, the caller's call will be forwarded to the callee via dahdi
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12:01.45[gnubie]WIMPy: voicemail is on my asterisk box
12:02.17WIMPyWhat kind of PSTN connection is your dahdi interface?
12:03.03WIMPyBTW: I which everyone would come up with a picture like that so we know what they're talking about.
12:03.41[gnubie]WIMPy: i am using digium's dev kit which uses tdm, an analog fxo/fxs ports
12:04.25WIMPyThat's what I expected :-( Spounds like some sort of call progress detection is going wrong.
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12:05.24WIMPyI guess you might have it set to look for polarity switches your line doesn't have or the like. But that needs to be answered by someone in to the analog stuff.
12:06.11[gnubie]WIMPy: this is my home box by the way.
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12:09.53tompawhttps://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 < this page is missing information about res_config_pgsql, which apparently now wants a callbackextension config in the pgsql table
12:10.34WIMPyHmm?
12:11.03WIMPyI'm pretty sure that's older than 11. I'd even think it's much older.
12:11.17tompawI'm pretty sure I just upgraded from 10.8 and it started bitching about it.
12:11.47tompawSo even if it was present before, it must have been optional. Now it's being used in the queries.
12:11.51WIMPyMaybe it just didn't care about the missing field before?
12:12.03WIMPyOr realtime config gets new features some versions later.
12:12.29tompawWIMPy: it must've ommited that field in the SELECTs. Otherwise, my pgsql would return the same error as it's returning now.
12:12.57tompawGreat, I signed up to this wiki to add this info and now I'm getting "Not Permitted" on all pages.
12:13.15WIMPyThen I have a feeling that using it gives you less features than normal config.
12:13.27WIMPynice
12:14.14tompawWIMPy: Asterisk is using SELECT * FROM ... , which means it can (in theory) dynamically adapt to the amount of information returned.
12:14.31tompawIn 11, it's mentioning callbackextension in WHERE.
12:14.58WIMPyInteresting.
12:15.10ectospasmtompaw: Asterisk wiki?  Yeah, that's only for members of the dev team.  I work for Digium and I can't edit that wiki.
12:15.26tompawectospasm: lol nice, why would they leave a sign up link there?
12:15.28ectospasm...it's the easiest way to cut down on the riffraff
12:15.44WIMPyMust be to schedule (re)registratuons, I suppose.
12:16.06ectospasmtompaw: good question.  Maybe Qwell, leifmadsen, file, or most probably newtonr
12:16.20ectospasm(newtonr is the Digium community liason)
12:17.17tompawI write my own conf files, so I consider myself a dev :P
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12:17.36WIMPy:-)
12:19.21ectospasmtompaw: not that kind of dev
12:19.27ectospasm(-;
12:19.33tompaw:-(
12:20.47tompawI must be at least a little dev'y, because it just started working and I successfully posted a comment ;>
12:21.26ectospasmtompaw: oh, I thought you meant edit a page.
12:21.49tompawOriginally I did, but I am deeply offended by this lack of trust and I shall not mention that again.
12:22.49ectospasmtompaw: well, the problem is it might quickly devolve into something not at all useful.  re: voip-info
12:23.07leifmadsentomaw: that page you're trying to edit is likely generated from code then
12:23.30tompawleifmadsen: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 this is generated from code? :P
12:23.55ectospasmtompaw: that's part of the official Asterisk 11 source distribution
12:24.01leifmadsenmy guess is it is generated from the UPGRADE.txt file
12:24.05ectospasm^
12:24.22leifmadsenso I'd submit a diff to the issue tracker with the change, or at least open a bug for that
12:24.29tompawectospasm: I know mate, I'm just playing up. On a side note, the query is now WHERE name LIKE '%' AND callbackextension LIKE '%' ORDER BY name
12:24.47tompawWhich means that not only the column is now mandatory, but also its content.
12:25.51tompawAlthough it seems to be falling back to the old query when this one fails.
12:27.19tompawleifmadsen: I would also mention on this page that AMI welcome prompt changed to '1.3'
12:27.22tompawFor obvious reasons.
12:27.35leifmadsentompaw: see my previous comment :)
12:28.27tompawleifmadsen: I'll try, but I'm afraid I'm gonna break something :P
12:28.46leifmadsenisn't sure how, but ok
12:29.22tompawleifmadsen: through https://issues.asterisk.org/ ?
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12:33.11mathis_mornin
12:44.06tompawHow do I submit a diff to a particular file via JIRA?
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12:48.55tompawAh, the agreement.
12:55.14tompawAnyway, I upgraded do 11-current, and the calls are still killing the AMI telnet session
12:55.58tompawhttp://pastebin.ca/2252404
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12:57.14tompawAny ide what could be causing this behaviour?
12:57.49ectospasmtompaw: what does the CLI say with full debug turned on?
12:58.09ectospasm...with issues like this, turn on the debug!
12:58.28tompawI shall turn on the debug and come back!
12:58.32ectospasmset "console => notice,warning,error,verbose,debug,dtmf,fax" on logger.conf
12:58.41ectospasm"logger reload" in the CLI
12:58.47ectospasmcore set debug 10
12:58.52ectospasmcore set verbose 10
12:58.57ectospasm...then let'r rip
12:59.42slav3_kittenok so i modified my sip.conf to have the proper extension numbers in caller ID which allowed me to have voicemail go to the proper box when the general extension is dialed. but it broke my wakeup call script which uses caller id information to make a .call file in the future. can i alias something to easily fix that?
13:01.05*** join/#asterisk vlad_starkov (~vlad_star@83.149.8.98)
13:01.48slav3_kittenmaybe i'll just change the name string to the extension number instead of a nice identifier likei have
13:03.55*** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.104)
13:04.22tompawectospasm: I captured a billion billions lines, let me clean it up and will show you in 5 mins
13:04.45*** join/#asterisk LiuYan1 (~LiuYan@222.125.128.148)
13:06.21*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
13:09.13slav3_kittenok fixed it i think
13:12.24tompawectospasm: http://tompaw.xxx/amierror
13:12.37[gnubie]WIMPy: even if i bypass the auto-attendant and ivr which goes straight to the callee, still the same problem - callee accepted the call but the caller still hearing the ring tone until it reaches the timeout (voicemail). kindly check http://pastie.org/5392008
13:21.12slav3_kitten[gnubie], got your extension.conf and dhadi.conf?
13:21.49[gnubie]for those who just came in who may be able to help me on my problem, let me describe it once again. the call process flow is illustrated and can be found at http://imagebin.org/235688
13:21.57[gnubie]when the caller reaches my asterisk box, the call will be forwarded via dahdi (using the digium dev kit) and reaches the callee. when the callee accepted the incoming call, the caller still hears the ringtone while the callee is already in-session but hears nothing. after 60 seconds timeout, the caller will be redirected to my asterisk's voicemail.
13:22.14[gnubie]slav3_kitten: let me get it for you. please hold on.
13:23.07slav3_kittenholds onto himself
13:24.51tompawectospasm: any ideas? it's the first time I'm experiencing something like this :/
13:25.10slav3_kitten[gnubie], you did do this step right : In order to use MeetMe() and DAHDI with Asterisk as non-root, you must change the /etc/udev/rules.d/dahdi.rules so that the OWNER and GROUP fields match the non-root user Asterisk will be running as. In this case, weÂ’re using the asteriskpbx user.
13:27.47[gnubie]slav3_kitten: i don't have the /etc/udev/rules.d/dahdi.rules file. local pots calls works fine.
13:28.54slav3_kitteninteresting. ok so your /etc/dahdi/modules is setup properly as well as /etc/dahdi/system.conf ?
13:30.35slav3_kittenhow's it coming on your dialplan and dahdi configs?
13:31.54ectospasmtompaw: I'm not exactly sure... but it looks like the far end of your SIP channel sends a BYE message, which indicates to Asterisk that it should tear down the call...
13:32.18tompawectospasm: the call flow is just fine, my question is - why is the AMI session being dropped?
13:32.57tompawIt kills the telnet the second I begin the call.
13:33.31slav3_kittenwho still uses telnet?
13:34.20tompawmy software does:P
13:34.21ectospasmtompaw: like I said, I don't know.  You may have found a bug (11 is still on the .0 release [well, 11.0.1...])
13:34.35slav3_kittentomaw, what software
13:34.42tompawectospasm: same thing happened in 10.8. I guess I'll have to find another way to talk to AMI
13:34.46tompawslav3_kitten: mine, the one I'm working on
13:34.50*** join/#asterisk Praise (~Fat@unaffiliated/praise)
13:35.01ectospasmslav3_kitten: it's common to observe AMI with telnet... or would you rather netcat/nc?
13:35.04slav3_kittenhave you ever heard of ssh?
13:35.11ectospasmslav3_kitten: sit down.  Shut up.
13:35.18tompawslav3_kitten: for localhost connection?
13:35.23ectospasmslav3_kitten: ssh would not make sense here.
13:35.42slav3_kittentompaw, thought you were remote monitoring it not localhost, ssh is pointless localhost
13:35.44ectospasm...it is expected that you execute telnet on a trusted network
13:36.02tompawyep, production stage it goes via ipsec vpn
13:36.16slav3_kittenectospasm, i want some smoked salmon ifi'm to sit down an shut up
13:36.43ectospasmslav3_kitten: sorry, totally out of smoked salmon.  I have something the cat dragged in, would that work?
13:36.47slav3_kittensits down, shuts up, and holds his paws out for salmon
13:36.56slav3_kittennah i drag things in myself
13:37.10slav3_kittennormally women with a sexy british accent :D
13:37.28ectospasmsorry, I'm cranky after a long day of work.  I bid y'all adieu.
13:37.56slav3_kittenectospasm, that's cranky?
13:38.05slav3_kitteni thought you were joking around an such
13:39.42*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
13:42.51[gnubie]slav3_kitten: sorry, just noticed your messages. i was checking again my setup.
13:44.29[gnubie]slav3_kitten: my dahdi configs has been there for.. hhhmmmmm.. years?
13:45.03slav3_kitten[gnubie], have you upgraded the dahdi or asterisk version lately?
13:45.15[gnubie]slav3_kitten: but the use case illustrated at http://imagebin.org/235688 is just recently
13:46.42slav3_kittenshrugs
13:46.45[gnubie]slav3_kitten: well, yes since i am currently running v1.8.11 but it's been running for months already.
13:47.02slav3_kittenwell what did you change right before it broke?
13:48.03[gnubie]slav3_kitten: as i've said, the only new that i'm aware is the use case where there are 2 x mobile phones involved as per http://imagebin.org/235688
13:49.38slav3_kitteni'm sadly fully capable of following what's going on in the image
13:50.11slav3_kittenall i'll say is hit page 100 of the book an double check all your settings
13:50.22slav3_kittenshit does not just break for no reason
13:51.19[gnubie]slav3_kitten: have you seen this? => http://pastie.org/5392008
13:53.00slav3_kittenyea i read it
13:53.41slav3_kittenlooks like your system isn't generating the sip event for the user picking up the phone
13:54.10[gnubie]slav3_kitten: isn't it that between the callee and asterisk isn't a sip?
13:54.19WIMPysip? That's a dahdi issue.
13:54.39slav3_kittenWIMPy, it looks like he's got a sip fxs/fxo
13:55.03WIMPyWhat? Where?
13:55.13slav3_kittenwhere everything is prefaced with SIP/
13:56.03WIMPy"Called DAHDI/4/9876543" ?
13:56.06slav3_kittenshit i misread a line
13:56.15WIMPyI don;t see sip there.
13:56.30slav3_kitteni'm a little dyslexic sometimes :|
13:56.51WIMPyYes, too much to read.
13:57.07WIMPyBut the picture surely made it easy.
13:59.03slav3_kitteni'm not an expert on asterisk, so i shouldn't be trying to help anyone
14:03.45[gnubie]slav3_kitten: it doesn't matter if you're not an expert. thanks for your help.
14:05.10slav3_kitteni didn't help shit
14:08.12[gnubie]WIMPy: isn't this a dtmf issue?
14:16.21*** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be)
14:16.43WIMPy[gnubie]: No it's a signalling issue.
14:17.12WIMPy>>Sounds like some sort of call progress detection is going wrong.
14:17.24WIMPyWhat else did I say?
14:17.41slav3_kittenthat he should have a smoke
14:17.53WIMPy>>I guess you might have it set to look for polarity switches your line doesn't have or the like.
14:18.04[gnubie]WIMPy: yes, that's it. but, do you think this is just a configuration issue on my end or even beyond that?
14:18.17WIMPySo it's for someone who knows about FXO stuff.
14:18.35[gnubie]WIMPy: ok, noted. thanks! ;)
14:18.36WIMPyShould be dahdi config.
14:19.00slav3_kittenWIMPy, but if nothing has changed and it suddenly quit working it could be a hardware issue
14:19.22WIMPyOh, must have missed that part.
14:19.29WIMPyOr the telco changed something.
14:19.44slav3_kittenoh that too
14:19.49slav3_kittendidn't even think about that
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14:20.19WIMPyTelcos are evil :-)
14:20.38WIMPyOr why are we doing the stuff ourselves?
14:22.32tompawOK, so AMI is broken - maybe I don't need AMI after all?
14:22.50WIMPyWorks for me.
14:23.11tompawDoesn't work for me and since it's weekend, probably no dev is gonna pick up the subject, so I need a workaround.
14:23.22WIMPyDo you know when it kicks you?
14:23.32tompawIs there a no-AMI way to pause agents in a queue when they're on a call?
14:23.48tompawWIMPy: http://pastebin.ca/2252404
14:23.52[gnubie]WIMPy: you can find more information from http://pastie.org/5392256
14:24.23tompawWIMPy: AMI itself seems to be working fine, I can log in right back in. But this telnet session is closed by Asterisk for some funky reason.
14:25.02WIMPytompaw: That doesn;t really give a clue about the reason.
14:25.24[gnubie]by the way guys, the caller and callee's mobile operators are different and my asterisk dahdi's connection is also a different telco.
14:25.25tompawWIMPy: here's a full debug. But it's not revealing anything, either.
14:25.27tompawhttp://tompaw.xxx/amierror
14:25.27*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
14:25.31WIMPyBut do you need a persistant connection?
14:26.06tompawMaybe not. Can Asterisk pause the agent automatically? Some debug messages suggest it can.
14:26.07slav3_kitten[gnubie], i'm going to as a stupid question here
14:26.32tompawLike: Device 'Local/17862692279@cont-fas' changed to state '2' (In use) but we don't care because they're not a member of any queue.
14:26.33slav3_kittenwhy don't you dial the mobiles via sip, i'd assume you have a DID as well is outbound service
14:27.30tompawWIMPy: basically I'm using the persistent connection to detect agents logging in/out, adding them to queue and pausing/unpausing.
14:28.18slav3_kitten/as/ask
14:28.22WIMPyI don't even spot a message indicating that the connection is closed.
14:28.52tompawWIMPy: I know, there's nothing there.
14:29.10tompawI will try to find a workaround with a static queue config.
14:29.12[gnubie]slav3_kitten: i got a virtual number from flynumber which is from another country. i want to receive calls from that country which goes straight to my mobile phone.
14:30.05WIMPy[gnubie]: You could use a softphone on your mobile.
14:30.22WIMPyUnless data costs you an arm and a leg.
14:30.32tompawWIMPy: also, it only happens with Events: on. I just tried a session with Events off and there's no issue there.
14:30.45slav3_kitten[gnubie], i have a UK DID, and then i use flowroute for my outbound (cheaper) so i have calls coming in route to my cell via flowroute
14:30.55slav3_kittenpenny a minute for domestic US, pay as you go
14:31.22WIMPytompaw: Must be some event I don't get. Like I'm not using queues.
14:31.27tompawprefers 3g/4g carrier @ mobile
14:31.28slav3_kitteni have been pondering a FXS so i can connect it to a cordless phone
14:32.02[gnubie]WIMPy: i have bria on my iphone. does it mean that i have to make this bria registered all the time and waiting for calls via 3g?
14:32.05WIMPyslav3_kitten: There are less ancient ways to do that.
14:32.15tompawWIMPy: probably, yeah. I will moan about it on Monday :)
14:32.22WIMPy[gnubie]: That was the idea.
14:32.36[gnubie]slav3_kitten: let me check about flowroute
14:33.06slav3_kittenWIMPy, yea but it lets me get dahdi experience
14:33.25[gnubie]WIMPy: let me also try it.
14:33.36WIMPyslav3_kitten: Are you sure that's the kind of experience you want?
14:34.00[gnubie]WIMPy: i was expecting dahdi to my phone is more reliable..
14:34.33slav3_kittenWIMPy, if i'm learning asterisk in full... yea, what would you suggest for a better cordless phone though?
14:34.36WIMPy[gnubie]: Analog doesn't provide reliable signalling.
14:35.01*** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be)
14:35.07WIMPyslav3_kitten: A SIP base station? Or if you really want hardware use a BRI base, not an analog one.
14:35.10slav3_kitten^^+1
14:35.30slav3_kittenbri? *googles this*
14:35.32tompawDamn, if I do a ConfbridgeList, it also closes the connection
14:35.51tompawi.e. if I send this command via AMI. it answers and closes the line.
14:35.54tompawwtf????
14:35.54[gnubie];)
14:35.59slav3_kittenbri is isdn?
14:37.13slav3_kittenfuck; bbiaw i forgot i had breakfast plans today
14:37.19WIMPytompaw: Hmm. Sounds fishy.
14:37.24WIMPyslav3_kitten: Yes
14:38.19tompawWIMPy: check this out: http://pastebin.ca/2252421
14:38.57tompawIt says ListItems: 3, but it only lists one of them and sends ConfbridgeListComplete.
14:39.07tompawI'm gonna debug that b*tch so hard.
14:39.41WIMPyAMI and lists is an interesting topic anyway.
14:40.17WIMPyI can;t remember which one it was, but I had somethign where I found it impossible to reliably parse the response.
14:40.22tompawhttp://pastebin.ca/2252423
14:40.24tompawreload please
14:40.29tompawAMI vs CLI
14:41.42tompawWhat the hell is going on here...
14:41.53WIMPySomehow I get a feeling that it doesn;t like certain channels.
14:42.18tompawI'll try to set up the conference, THEN enable debug (to minimize trash), and then send the ConfbridgeList.
14:42.26tompawI wonder if I catch anything this way.
14:42.31WIMPyOr the list of channels in general.
14:46.24tompawWhy is the internet between EU and Panama so slow...
14:47.58*** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com)
14:48.00WIMPyUgh. Why don't I have permission to confbridgelist?
14:49.28tompawOK, that's all I got: [Nov 17 03:43:07] DEBUG[4573] manager.c: Running action 'ConfbridgeList' and then [Nov 17 03:43:07] DEBUG[4573] utils.c: Timed out trying to write
14:49.53WIMPyThat's something.
14:49.56tompawNot sure if that utils.c debug is even related to AMI
14:50.09tompawCLI gives me more: [Nov 17 03:42:14] ERROR[4565]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe
14:50.17WIMPyThere's a chance.
14:50.22tompawApart from that, there's nothing.
14:50.38WIMPyYou get those messages in that order?
14:51.15tompawI just tried the call without AMI telnet session and I also got the same error in CLI. Besides, it appears in CLI in a different moment, way after the telnet is killed.
14:51.23tompawSo back to square one. Nothing :(
14:51.36WIMPyhmm
14:51.45tompawI know
14:51.59tompawI'll try sending the AMI request multiple times to the same call.
14:52.09tompawTo find out if it triggers this error multiple times.
14:52.15tompawThat should answer if they're related.
14:52.20SeRigood morning all
14:53.05WIMPyHow do you find out if it triggers multiple times if you get disconnected the first time?
14:53.57tompawWIMPy: by pressing arrow up followed by enter? :P
14:54.10tompawWIMPy: just tried it, this error has nothing to do with my telnet issue :(
14:54.40*** join/#asterisk wonderworld (~w@dsdf-4db5159e.pool.mediaWays.net)
14:54.44tompawDAMN! That complicates things, I won't even be able to list a conference and kick certain users in a single call :/
14:54.58WIMPyAh, the log.
14:55.26tompawWIMPy: I reconnected 3 times with telnet and did ConfbridgeList
14:55.36tompawThe error only appears once, it's got something to do with call setup, not AMI.
14:56.26WIMPyAh, I guess "reporting" was missing.
14:56.43tompaw?
14:57.26WIMPyTo be able to list conferences.
14:57.56tompawIs there any other way to access AMI other then a telnet? A python API maybe? (FS has one!)
14:58.10WIMPyAMI is not telnet.
14:58.25WIMPyConbridgeList works for me.
14:58.41tompawSo it must be my setup... but why?
14:58.58WIMPyWell, that's the interesting bit...
14:59.13WIMPyAre you using agent channels?
14:59.16tompawNo system errors, lots of ram and hdd...
15:00.00tompawWIMPy: I'm not sure what you mean by that question. My "agents" are simple SIP friends (authenticated using res_conf_pgsql) that are being dynamically added to the queue as they log in.
15:00.24WIMPyok
15:00.37tompawThere used to be a persistend AMI script doing this job, but out of a sudden, it stopped working without me changing anything.
15:00.55WIMPySomethign must have changed.
15:01.18WIMPyHave you tried turning it off and on again?
15:01.29WIMPyAKA core restart ...?
15:01.38tompawI upgraded asterisk from 10.8 to 11.0.1, so I restarted it a multiple times, yeah.
15:01.46tompawI was hoping it's a bug that will go away.
15:01.55WIMPyRight.
15:02.03WIMPyremembers
15:02.26tompawI can try something else. I can set up a confbridge from CLI without that SIP channels that *SEEMS* to be the cause.
15:02.32tompawAnd then try the AMI query.
15:02.55tompawThat should at least narrow it down, don't you think?
15:03.03WIMPyNot sure.
15:03.24WIMPyIt's not an easy one.
15:03.34tompawThis should answer if it's the type of channel (aka "agent" line) or the confbridge itself causing the problem.
15:03.41tompawI'll try it now.
15:04.12WIMPyBut it's not only on the confbridge, right?
15:04.41WIMPyThe initial disconnect was not related to a confbrige list, was it?
15:06.47SeRiwaz up WIMPy
15:06.54*** join/#asterisk deo (~deo@112.198.79.186)
15:07.43WIMPySeRi: A real PBX waiting for me later.
15:07.47tompawWIMPy: correct, it wasn't.
15:08.15tompawWIMPy: I just tried it with only 1 type of channel being in the conference (of both types), and the problem's still there.
15:08.44SeRiWIMPy: Is that right? Which kind/model?
15:09.07WIMPySeRi: Hipath 3550
15:09.53tompawWIMPy: http://pastebin.ca/2252434 << same problem with ListRooms
15:09.57tompawwhat the hell!!
15:11.07WIMPytompaw: I wonder what else fails.
15:11.40tompawWHOA
15:11.43tompawWHOOOOA
15:11.46WIMPyBut apart from AMI you don;t see ant strange things happening?
15:11.53WIMPy?
15:11.57tompawone sec, I think I got it
15:12.43WIMPyEither I need a new keyboard, new fingers or (most probably) a new brain.
15:13.47SeRiWIMPy: Nice!
15:13.53tompawWIMPy: this is super-strange. I tried it locally on the asterisk server and it works perfectly fine!
15:14.15tompawAlthough I telnet'd via the same ipv4 address, as this is the only one that manager bounds to:
15:14.18tompawhttp://pastebin.ca/2252436
15:14.48tompawAm I dreaming? What the hell is going on here?
15:15.07WIMPyHmm. A buffer filling up?
15:15.38tompawWIMPy: notice one thing. In a over-network call, it responds with ConfbridgeListComplete prior to shutting down the line!!
15:15.41WIMPyI have never used it remotely. Only bound to 127.0.0.1
15:16.56tompawBut why is it even different from Asterisk's point of view? If I do "telnet 1.2.3.4" locally, it arrives at Asterisk stack the same way, doesn't it?
15:17.01WIMPyThere seems to be somethign fishy in that socket handling.
15:17.12WIMPyNo
15:18.07WIMPyIt's a Linux soddity. All local traffic goes throuh lo, even if it uses the address of another interface.
15:18.12WIMPy-s
15:18.15tompawWhat do you mean by that. If a deamon listens on a certain ip:port... that should be the same.
15:19.05WIMPyThe bigger and related oddity in Linux is that lo doesn't care about addresses.
15:19.05tompawWIMPy: ok, you're probably right, as I don't know much about linux internals, but yet still, this sounds like something zillion layers away from AMI preparing a ConfbridgeList, don't you thinkg?
15:19.14WIMPyA pretty strange behavioour.
15:19.58WIMPyLooks like a timing thing.
15:20.41WIMPyOn the remote connection there's something missing because there was no tx buffer available.
15:21.19WIMPyBut it's surely interesting that a following (shorter!) message then still gets sent before it barfs.
15:21.28tompawWIMPy: it would all make sense, if not for one fact.
15:21.37tompawLine 22: Event: ConfbridgeListComplete
15:22.09tompawThis thing is being sent prior to killing the session. So something makes AMI think that the Confbridge list has already been fully sent.
15:22.26WIMPyNo, that's the explanation: That message might be short enough to fit without meore space being freed.
15:22.29tompawIf it was a buffer/network problem, the communication would die "unawarely",if you know what I'm trying to say, right?
15:22.32tompawIt would just be cut.
15:22.37tompawAh.
15:22.41tompawI se.
15:22.45tompawHm...
15:23.10WIMPyBut the "error" from the previous (bigger) message is then handled afterwards whicht doesn;t seem to make too much sense.
15:23.55WIMPy(it's not a real error. Just the famous EAGAIN (AKA try again later))
15:24.07tompawThe only question now: is this asterisk's buffer or OS' buffer we're talking here?
15:24.22SeRiWIMPy: Is that going to be your new pbx or you working on it?
15:24.43WIMPyUsually I'd expect that to be cleared by the following successfull write. i.e. just some infomation being silently discarded.
15:25.14WIMPyThe buffer is from the OS. It's up to the application to try again later when it's full.
15:25.15tompawWIMPy: so it looks like the process realizes that one of the messages didn't fit, and after flushing the buffer it simply commits a harakiri?
15:25.34WIMPySomethign like that.
15:25.42tompawOK, how do I make it stop? :P
15:25.58WIMPyAs I said: the whole sequence doen't make too much sense, but parts of it do.
15:26.38WIMPyFind the bug in the logic.
15:26.41tompawWhy did it start happening out of a sudden? IT was an isolated production server with no crazy updates or anything like that.
15:26.58WIMPyI have never looked in to the managers source.
15:27.15tompawoh dog
15:27.17WIMPyMore traffic?
15:27.28tompawNope, it's saturday now, there is no traffic.
15:27.31WIMPyNo, that should be more random then.
15:27.36tompawNo changes in network topology, no equipment changes.
15:28.31WIMPyI don't really have an idea.
15:28.58tompawWhat I hate is the fact that there are no errors thrown anywhere.
15:29.10tompawBOth Asterisk and the OS are quiet about it.
15:29.29WIMPyThere certainly shouldn't be one.
15:29.46WIMPyJust something handling the situation incorrectly.
15:30.25tompawWIMPy: if something makes the suicide decision, this should be followed by an error thrown.
15:30.29tompawAt least according to my logic.
15:31.21tompawI can as well try updating glibc and rebooting the server.
15:32.31WIMPythat's surely not a (planned) suicide. More of an accident.
15:33.14tompawyey, new kernel
15:34.02WIMPyA secret kernel update?
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15:34.43tompawI can as well try it, I'm out of ideas.
15:35.10WIMPyAh, ok.
15:35.45*** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be)
15:35.46tompawOnly a minor update, but maybe they fixed something that triggers something to do something.
15:36.48WIMPyI can;t look into it now. Will be picked up in 15 min to configure a classic beast.
15:37.10tompawgood luck!
15:37.14[gnubie]WIMPy: now the callee is using sip on iphone using bria and it works
15:38.00tompawIt didn't help :<
15:39.46tompawI will try DNAT'ing it via 127.0.0.1
15:40.17tompawwWhat I just said makes no sense. Screw it, I'm gonna have a beer.
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15:48.25WIMPytompaw: If that's not enough, a tcp wrapper / netcat might help.
15:48.57tompawWIMPy: I was just considering proxychains.
15:50.10WIMPycul
15:52.43tompawlaters
15:54.05*** join/#asterisk vlad_starkov (~vlad_star@83.149.8.121)
15:57.11SeRiman I got so drunk last night... LOL
15:59.57*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
16:00.52*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
16:01.46*** join/#asterisk mokmeister (~mokmeiste@93.107.0.242)
16:05.07tompawWIMPy: it works via ncat :/
16:05.13tompawI don't understand WHY!!!
16:12.02jpsharpcats always make it work
16:17.44tompaw;>
16:20.33*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
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16:34.59[gnubie]WIMPy and slav3_kitten: thank you for your help. ;)
16:35.05[gnubie]waves.. gtg now.
16:39.41*** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com)
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17:35.31*** join/#asterisk Silicium (~marco@5.9.215.242)
17:35.41Siliciumhi there :)
17:36.05jpsharpYo
17:36.40Siliciumits quiet longer time, i worked with asterisk (1.2/1.4) but i want to build a small pbx for at home. actually i tried the AsteriskNOW and trixbox distribution
17:37.11Siliciumbut... both of them are somehow more difficult to configure with gui than on the classic way
17:37.26jpsharpThe GUIs always limit you in someway.
17:37.41Siliciumbut i dont want to configure my stuff by hand - is there a distribution which is easyer than direct config editin?
17:37.46slav3_kittenthat and i think trixbox has been dead for a while...
17:37.53[TK]D-FenderOver 2 years
17:37.57Siliciumyep
17:38.14Siliciumi have a trixbux downloaded with an actual asterisk but its not longer based on m0n0
17:38.16[TK]D-FenderSilicium: FreePBX's ISO is pretty much the best option right now
17:38.17slav3_kittenSilicium, imho just install bare bone debian and then compile asterisk from svn
17:38.41Siliciumslav3_kitten: thats what i planned to do next, it seems to be more easy thatn that gui foo
17:39.00Siliciumbut i want just to ask here if there is another way, iam somehow to lazy
17:39.15Siliciumi will look at freepxb
17:41.01Siliciumhmm, freepbx looks some kind of same as the new trixbox
17:41.47[TK]D-FenderThere is no new "trixbox", it is dead.
17:41.58slav3_kittenSilicium, diy is the best way. took me a hour or two to install an get everything running on a old 800mhz pentium m
17:42.00[TK]D-FenderAnd Tribox was originally built around FreePBX.
17:42.10[TK]D-FenderWhich has since progressed forward
17:42.14slav3_kittencurrently using 32 megs of ram
17:43.12Siliciumslav3_kitten: i know :( i spend around a day to install multiple distributions, in this time i would be able to completely install and configure a pbx for 200 persons with IAX IVR etc.
17:43.42Siliciumallright, i will do it form hand then. i was just hoping that there is somethin like trixbox, 2 years ago
17:44.34*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
17:46.04Siliciumthanks :)
17:46.08Silicium(anyway)
17:46.23*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
17:46.24*** mode/#asterisk [+o pabelanger] by ChanServ
17:46.50Siliciumis there something important to know between 1.4 and 1.8?
17:46.58Siliciumor can i just compile and run as 1.4?
17:47.05Siliciummore or less
17:47.26pabelangerSilicium, Read UPGRADE.txt change CHANGES in the root directory of 1.8
17:47.32Siliciumthx :)
17:48.20*** join/#asterisk vlad_starkov (~vlad_star@83.149.8.101)
17:54.02*** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4)
18:12.23SeRibig jump there...
18:27.07SeRiguys I get the following error: /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister
18:27.35SeRisome research told me I had to preload res_fax.so
18:27.42*** join/#asterisk ruied (~AndChat66@po-217-129-155-146.netvisao.pt)
18:28.01SeRiis this true?
18:37.40jpsharpIf you load res_fax.so and then try to load res_fax_digium.so do you still get the message?
18:38.46[sr]WIMPy: you there? my problem i think it be the NT1 termination on that specific place, the remote ISP network gotta have something different than normal
19:10.10dijibholy crap i just woke up
19:10.28dijibSeRi: preload res_fax for what?
19:10.34dijib10+ you dont need it
19:23.00Siliciumallright
19:23.03*** join/#asterisk ruied (~AndChat66@po-217-129-155-146.netvisao.pt)
19:23.07dijibalright what?
19:23.12Siliciumit took 30 minutes to setup
19:23.31dijibthat about right
19:23.39Siliciumit also took about 30 minutes to download the iso image of asteriskNow last week
19:23.42Siliciumhaha
19:23.56dijibtakes me much less
19:24.22Siliciumits sad that there are still no usefull GUIs for asterisk
19:24.30Siliciumexcept own written and expensive ones
19:24.49Siliciumbut yes, its easy to configure. so may just not needed
19:29.43Siliciumvery nice docu about antarctical ice on tv. awfull blue ice seas and rivers
19:29.48Siliciumfuck yeah, analog porn
19:37.35*** join/#asterisk timahvo1 (~rogue@41.212.120.182)
19:43.46*** join/#asterisk CoffeeIV (rgr@rrcs-24-173-111-218.sw.biz.rr.com)
19:52.51[TK]D-FenderSilicium: What is "hard" or "not useful" about FreePBX?
19:54.28Siliciumwebinterface is to complex and diffictult to use
19:54.48Siliciumi was not able to configure a call routing
19:55.10Siliciumand configuring call routing is more easy directly in the extensions.conf so i can do it by hand completely :)
19:55.22[TK]D-FenderCall routing is 2 scrrens.. if you include the tunk config itself...
19:55.25[TK]D-Fenderfar from "hard"
19:55.48[TK]D-FenderAs for your failure ... we haven't seen ywhat you've actually attempted so who's to say ass to why it didn't work.
20:02.17SeRidijib: you arounf?
20:03.08SeRipreloading res_fax.so fixed the warning
20:06.22Siliciumanyway, i do it on the good old way again :)
20:12.10*** join/#asterisk sanon (47c8d864@gateway/web/freenode/ip.71.200.216.100)
20:12.13sanon[Nov 17 15:10:00] NOTICE[1440]: chan_iax2.c:8883 update_registry: Restricting registration for peer 'MSI' to 60 seconds (requested 120)
20:12.20sanonwhere do i change that
20:16.21[TK]D-Fenderin iax.conf
20:21.43sanonwhat do i put in there
20:22.03[TK]D-Fendersanon: Have you looked at the iax.conf sample config?
20:22.50sanonyes and no. I am following along the astriskdocs.org
20:23.16sanonI deleted the sample and replace it with the one provided
20:23.54[TK]D-Fenderyou should still have it in the tarball...
20:25.09sanonnever mind
20:25.13sanonI got it
20:25.23sanonI make a copy of the folder
20:29.21*** join/#asterisk dfgas (~dfgas@71-90-33-37.dhcp.ftbg.wi.charter.com)
20:43.16*** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net)
20:46.49*** part/#asterisk mute (mute@clt.scottn.us)
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21:16.35slav3_kittenslow HDD is SLOW
21:16.43slav3_kitten16MB/s write 17MB/s read
21:19.31dijibpata ftw
21:19.53slav3_kitten1990's PATA :P
21:19.59slav3_kitten20gb drive
21:32.41*** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004)
21:35.58dijibsweet
21:36.15dijibi have so many 120's to 500gb laying around
21:37.13slav3_kittenit's 20 gigs of amazing
21:47.18*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
21:49.05dijibdo you know how many of those drives ive dismantled in the last 6 month for NdFeB
21:49.10dijibmagnets
21:52.21*** join/#asterisk kryptoknight (~markdalby@HSI-KBW-134-3-222-126.hsi14.kabel-badenwuerttemberg.de)
21:53.42slav3_kittenwhy do you want NdFeB mags?
21:53.52*** part/#asterisk kryptoknight (~markdalby@HSI-KBW-134-3-222-126.hsi14.kabel-badenwuerttemberg.de)
21:54.08*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
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22:08.44dijiblook it up
22:12.00*** join/#asterisk fritz09 (~Adium@46.115.33.108)
22:14.01dijiboh i dont know they told my hot saldering iron when im working
22:22.49*** join/#asterisk josefig (~josef@unaffiliated/josefig)
22:43.57*** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se)
22:49.38*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)
23:00.56epaphusHey guys.. is ChanSpy() the only way to listen in on bridged calls?
23:01.11epaphusIs there anyway to listen in on an entire extension
23:03.32ChannelZWhat distinction is it you are making between the two?
23:06.55*** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net)
23:09.30*** join/#asterisk trumee (~parul@93-96-159-40.zone4.bethere.co.uk)
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23:15.12*** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com)
23:16.41dfgas-cr48dijib, yo
23:20.38epaphusChannelZ, well iam not quite familiar how ChanSpy works.. but i want to monitor an extension call after call without interruption
23:22.19*** join/#asterisk P-NuT (~P-NuT_@5ad3493e.bb.sky.com)
23:23.23P-NuThey all.. does anyone know of any modern articles for configuring cisco 7940s?
23:23.42P-NuTfor use withasteriskand sip.
23:23.46P-NuT?
23:28.03*** join/#asterisk timahvo1 (~rogue@41.212.120.182)
23:32.56*** join/#asterisk gusto (~gusto@ppp-93-104-83-64.dynamic.mnet-online.de)
23:34.45dijibdfgas-cr48: hey
23:34.55dijibdid you not hear me inb4
23:36.15dijibdfgas-cr48: i was cleaning my stove. just finished and put some chocolate chip cookies on
23:36.52dfgas-cr48oh no
23:37.14dfgas-cr48um, outbound dialing broke
23:37.18*** join/#asterisk The_Hatta (ident@165-154-121-72.ispnetbilling.com)
23:37.21dfgas-cr48[Nov 17 17:32:58] WARNING[3813][C-00000003]: channel.c:4783 ast_prod: Prodding channel 'SIP/11-00000005' failed
23:37.44dfgas-cr48not sure what
23:37.52epaphusChannelZ, is that how ChanSpy works?
23:37.53dfgas-cr48happened
23:38.02*** join/#asterisk gusto (~gusto@ppp-93-104-78-79.dynamic.mnet-online.de)
23:39.32dijibhmm
23:39.40dijibwell i know the fix for another one of your errors
23:39.43dijibcall in again
23:41.06dfgas-cr48no sound again
23:41.07dijibi can hear you
23:41.10dijibwtf
23:41.15dijibtalk???!
23:41.29dijibi hear it open i guess but nothing from your end
23:41.33dijibwhats up with that
23:41.43dijibdid you save and apply those setting? i would recon....
23:41.58dfgas-cr48?
23:42.06dfgas-cr48settings?
23:42.09dijibwhats the error? or no error just no sound?
23:42.42dfgas-cr48just no sound on that
23:42.44*** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se)
23:43.09dfgas-cr48when calling out to lets say my cell i get
23:43.13dfgas-cr48[Nov 17 17:32:58] WARNING[3813][C-00000003]: channel.c:4783 ast_prod: Prodding channel 'SIP/11-00000005' failed
23:43.30dfgas-cr48i did reboot system today
23:44.38*** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca)
23:45.00dijibhello all again
23:45.04dfgas-cr48lol
23:45.09dfgas-cr48sup
23:45.11dijibdont know what i did there
23:45.17dijibso any errors beyond that 1?
23:45.28dfgas-cr48nope
23:45.44dfgas-cr48i rebooted system today
23:47.20dijibthen something has not started properly
23:47.26dijibcan you get into the asterisk sli?
23:47.30slav3_kittenhmmm where did i see the section about time specific auto attendant branching
23:47.33dijibobiously
23:47.45dijibno clue
23:48.14slav3_kittendijib, you need to give me a better computer
23:49.14*** join/#asterisk gusto (~gusto@ppp-93-104-78-79.dynamic.mnet-online.de)
23:50.30dijibwhy dont you go dumpster dive yourself a C2D like i did
23:50.32dijibbrb
23:54.54infinity1is teliax having issues
23:55.10dfgas-cr48dijib, yo
23:58.34dijibyo
23:58.47dfgas-cr48pm

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