IRC log for #asterisk on 20110207

00:00.07DelphiWorldIsUp: http://dpaste.de/As2b/ dialplan
00:02.26IsUpDelphiWorld: go to your outbound, remove exten => 883510009900637,1,Answer() and exten => 883510009900637,1,Echo
00:02.44IsUpgo to your iax peer, change context=default to context=voipms_incoming
00:02.46DelphiWorldIsUp: removed
00:03.06IsUpto your dialplan, add a context [voipms_incoming] and add exten => 883510009900637,1,Answer - exten => 883510009900637,n,Echo
00:03.50IsUpwhen you finish, type 'reload' on your CLI and place a call to your DID
00:04.51DelphiWorldIsUp: same hell :-)
00:05.47IsUpDelphiWorld: paste CLI output to pastebin please
00:07.01DelphiWorldIsUp: http://dpaste.de/8sAg/
00:07.12DelphiWorldIsUp: sory if i missed some lines, i'm blind :-)
00:07.45IsUpDelphiWorld: you probably missed context= on your iax peer
00:08.00IsUpgo to your peer context, set context=voipms_incoming
00:08.10IsUpand add voipms_incoming context to your dialplan
00:08.12DelphiWorldIsUp: but i do the in our out context
00:08.26IsUplets try on different context now
00:08.55IsUpyour current context is little bit confusing
00:10.39DelphiWorldIsUp: now is changed
00:11.04IsUpok do 'reload' on your CLI
00:11.08IsUpand place a call to your DID
00:11.40DelphiWorldIsUp: can you give me how to dial a sip peer registered to me?
00:11.46DelphiWorldIsUp: let say peer 101
00:12.45IsUpDial(SIP/101)
00:12.49DelphiWorldIsUp: call is answered
00:12.54DelphiWorldnow i need to try out echo
00:13.30*** join/#asterisk sourcode (~code@ppp-115-87-234-237.revip4.asianet.co.th)
00:17.17MikeHhrm
00:18.11MikeHHas anyone seen an issue with Dahdi where an analogue card stops working and requires a power off and power on to start again
00:19.13IsUpMikeH: i had this problem. it stops ringing, not even a dialtone
00:19.24MikeHWell
00:19.32MikeHthe card has actually disappeared from Dahdi
00:19.33IsUp/etc/init.d/dahdi stop & start was my solution :p
00:19.42MikeHI had this the other day
00:19.45DelphiWorldMikeH: what card you have?
00:19.51MikeHturn server off and back on again, it's fine
00:20.00MikeHbut a reboot does not bring it back up
00:20.15MikeHstopping and starting dahdi only errors that the device is missing
00:21.18MikeHDelphiWorld, AX4G
00:21.44DelphiWorldMikeH: i use openvox
00:23.02MikeHhrm, lspci shows the card
00:23.08MikeHbut /proc/interupts does not
00:23.43MikeHDelphiWorld, Openvox GSM card?
00:23.57DelphiWorldno, MikeH... FXO
00:24.32MikeHI'm thinking of sending this ax4g card back as faulty given the issues I'm having
00:24.36MikeHand switching to an openvox one
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00:24.47MikeHI'd like to find someone who uses one first though really
00:24.58MikeHFor example the AX4G does not support answer detection
00:25.01MikeHit's ridiculous
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03:19.37shaprGood Morning alla!
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05:02.13solokkhzhello
05:03.04solokkhzcan anyone help identify the problem with asterisk not calling a pstn trunk with the x100p card?
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05:12.51solokkhzsomeone?
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05:58.37rushowrhello all! I'm wondering if anyone might be willing to chat with me about some strange behavior I'm seeing with Local Channels.. I have a SIP channel entering a Queue which uses a Local to lookup info about the agent and then uses SIP to call the agent.... (SIP1 -> Local -> SIP2)...
05:58.59shaprwhat version of Asterisk?
05:59.02rushowrSIP2 is able to hear and speak to SIP1
05:59.06rushowr1.8. (one sec
05:59.32rushowr1.8.2.2
05:59.32shaprI've seen some fixes for Local channels in the 1.8.2 changelog, tried the latest release of 1.8.2?
05:59.55rushowrdoublechecking that 1.8.2.2 is latest
05:59.59rushowrahhh crap
06:00.02rushowr1.8.2.3
06:00.11rushowrlemme go download that and run a quick test call
06:00.21rushowrI'll brb :D
06:00.37rushowrswears he had JUST updated not 4 days ago
06:01.30rushowrmethinks I should just resume using SVN checkouts again
06:03.17rushowr@shapr hey, quick question (it's been a loooonng day)..IIRC, if I want to always have the latest 1.8 vesrsion from SVN I just checkout  the 1.8 branch right?
06:03.50shaprtrunk
06:04.06shaprbut you probably want the latest *release* instead of the latest svn
06:04.10shaprThat's just my opinion, though
06:04.18rushowrer... well, yes and no
06:04.29rushowrbah, let me go doublecheck something
06:04.51rushowrthe releases are tags, which means I would have to constantly go download the latest tarball or switch the source URL from SVN
06:05.47rushowrbut, I could have sworn that using the major release branch would result in you having the latest version within that major version...because they merge trunk back into that branch as well as make a new tag whenever they push a new release within that version
06:05.52rushowrlet me buzz a dev
06:05.54rushowr:)
06:07.38shaprI'm not a dev....
06:09.41rushowris that sarcasm? Sorry mate, I really am very very tired and not sure what you're trying to say...I don't know many of the actual core devs on sight in IRC so if you are, please say so, so that I don't embarrass myself further lol
06:10.00shaprNo sarcasm meant. I just don't know what conventions the asterisk devs use for svn
06:10.20rushowrah ok, good then :D that's what I thought. but I had to check :D
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06:11.54rushowrI've put a question out to the devs
06:13.30rushowrI've been working with a few clients who were on Asterisk version update locks for so long that when this client finally moved up, I found myself unclear about the conventions used as well
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06:13.55rushowrdoing a bit o' googling here too, maybe I'll finally find the reference(s) I had before :D
06:16.17rushowrok here we go
06:16.24rushowrwe were both kinda right shapr :D
06:17.10*** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr)
06:17.43rushowrbranches/1.8 contains pre-release (ALMOST bleeding edge) code for the 1.8.x code
06:17.52rushowrthe stable code is always contained in tags :P
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06:23.39astro__has zaptel.conf and zapada been depreciated for along time?
06:23.46rushowryessir
06:25.49luckman212hi guys, anybody here using a streaming source for MOH?
06:27.16astro__anyone have an idea of what i could develop voip.ca into
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07:21.18rgagnonany ops online or is everyone sleeping?
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07:29.03solokkhzcan anyone help identify the problem with asterisk not calling a pstn trunk with the x100p card?
07:35.37*** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at)
07:35.39schmidtsgood morning
07:50.36wdoekes2morning
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07:53.49solokkhzno one want to help me :(
07:54.16schmidtssolokkhz sorry didnt see whats your problem ;)
07:54.45solokkhzproblem with asterisk not calling a pstn trunk with the x100p card
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07:55.19kaldemar~ask
07:55.20infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
07:56.00shaprsolokkhz: Any particular error?
07:56.29solokkhzwhen i call it looks like its calling but there is no line. when i use tha passthrough phone it works.
07:56.43solokkhzno error
07:56.52solokkhzthats the strange thing about it
07:56.53schmidtssolokkhz what does you see if you start a pri debug on the span?
07:57.44drmessanopri debug?
07:58.08kaldemarx100p is an FXO interface, you won't see anything with pri debug.
07:58.27schmidts:P sorry never used FXO lines with asterisk, only isdn ;)
07:58.47drmessanoand even calling it an "FXO interface" is a stretch :)
07:59.02schmidtsPOTS interface?
07:59.03shaprI've never even seen an X100P
07:59.42solokkhzlol, i am a noob, and want to test it before i upgrade to a more expensive card
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08:00.23solokkhzthe sip extension work great
08:00.34solokkhzbut can go out to pstn
08:00.56shaprIs the X100P supported by DAHDI?
08:01.17solokkhzyeah it sees it
08:01.23solokkhzsee it
08:01.29solokkhzi think
08:01.53shaprAre you using DAHDI 2.4.0.0 ?
08:02.32solokkhz[trixbox ~]# dahdi_hardware
08:02.32solokkhzpci:0000:04:00.0     wcfxo+       1057:5608 Wildcard X100P
08:02.40drmessano*gulp*
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08:03.18schmidtssolokkhz is your hostname only trixbox or are you using trixbox?
08:03.47solokkhzhmmm.. i use trixbox, shouldn't i?
08:04.35schmidtssolokkhz LOL thats like joing #linux and ask if you should use windows :D
08:04.45schmidtsnot that bad but like this
08:05.27solokkhzthey both use the same freepbx and dahdi
08:05.41solokkhzits more like win2k and winxp
08:05.42solokkhzlol
08:06.01solokkhzboth don't work with my card
08:06.03solokkhz:(
08:06.23drmessanoTrixbox doesnt use FreePBX
08:06.53drmessanoThey use a fork of FreePBX 2.4 they call "PBXConfig", which includes crap changes to the code that many cringe at, at best
08:07.29solokkhzwhat sould i use then?
08:08.01drmessanoAsteriskNOW
08:08.34solokkhzi installed it, and couldn't get used to the gui after using trixbox
08:09.13drmessanoBecause it wasnt green?
08:09.22drmessanofor the most part, the GUI should be the same
08:09.50solokkhzlol, i guess there are more crap on the trixbox toclick on
08:09.51solokkhzlol
08:11.51solokkhzbut anyway, the ssh commands are the same
08:12.08drmessanossh is ssh
08:13.09drmessanoAsteriskNOW has a nice module for configuring DAHDI cards and it's also supported on IRC in several places
08:13.51solokkhz<PROTECTED>
08:13.51solokkhz[1]
08:13.51solokkhzactive=yes
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08:37.21solokkhzno dial tone
08:38.10kaldemarwhere do you expect to hear a dial tone?
08:44.52solokkhznot a dial tone, but i excpect it to dial
08:45.12kaldemarcan you receive calls?
08:45.32kaldemarwhat do you see in CLI when you make a call?
08:46.29tzafrir_laptopWhat exactly do you dial?
08:46.37tzafrir_laptopWhen do you expect to get a dial tone?
08:48.00solokkhzcan't receive calls as well through pstn
08:48.38solokkhzi set the outbounf route to call to XXXXXXX and NXXXXX
08:51.33solokkhzlast message say:
08:51.36solokkhz<PROTECTED>
08:51.36solokkhz<PROTECTED>
08:51.55solokkhzand the call still active
08:52.01solokkhzit does not hang off
08:52.05solokkhzjust quiet
08:52.11solokkhzno idaling or something
08:52.17solokkhzdialing*
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08:55.34solokkhzi am starting to think that maybe my card does not work
09:02.06tzafrir_laptopIf you plug a standard phone instead, can you dial?
09:07.14solokkhzyes, standard phone works and the card passthrough plug also work with a regular phone
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09:18.51donatashey. Can I do the same thing excluding Answer(); ? http://p.defau.lt/?0_RLRxStB88EB_MOBidJyg
09:19.18donatasI want, that user won't be charged if I don't answer realy.
09:24.27kaldemardonatas: the Playback app will answer the call anyway if you don't give it the noanswer option. the used channel has to support early media or the caller won't hear your announcement.
09:25.43donatasaha, I see... Is it possible to do that?
09:25.47donataswithout answering
09:26.47kaldemarremove the Answer line and PlayBack(sveiki) -> Playback(sveiki,noanswer)
09:27.13donatasa
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09:27.20donatasthank youi
09:28.01donatasrealy.. noanswer: Play the sound file, but don't answer the channel first (if hasn't been answered already). Not all channels support playing messages while still on hook.  :)
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12:07.40As001Hi is it possible to change music oh hold for logged in agents and that change take effect emidietly without logging out agents ?
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12:08.22As001I tried with moh reload and reload module chan_agents.so
12:08.35As001but it keeps playing old moh until they log out and log in again
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12:12.33JasnejacAs001: can you achieve this with the CHANNEL function by changing the MOH class?
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12:29.56As001ok thanks I will see that
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13:02.18verywisemanhow can i hide caller id and sip's extension number when i make call?
13:04.13kaldemarverywiseman: what version of asterisk are you using?
13:04.29verywisemankaldemar, 1.4
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13:05.19porcheHi all
13:06.02porcheI want to ask which linux distribution you use for asterisk?
13:06.26Tozz_does it matter?
13:06.31Tozz_we're probably all using something else ;)
13:06.40kaldemarverywiseman: app SetCallerPres
13:06.50porcheTozz, I am used to debian
13:07.08Tozz_i'm using ubuntu
13:07.10porchenow I want an upgrade with hardware raid, hosting guys push me for Centos
13:07.13kaldemarporche: then use debian for your install.
13:07.28Tozz_there is no reason not to use debian for hardware raid
13:07.36kaldemarporche: use what suits your needs and you're comfortable with.
13:07.52Tozz_hosting guys probably want to use centos because thats their fav. flavour
13:07.58porcheyeah
13:08.12porchebut issue is, I ordered a debian box from them with software raid
13:08.45porchesomehow, under heavy traffic, its load spikes, when I check the details, it tries to resync swap raid
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13:12.04OlafsenMhi guys
13:12.30OlafsenMis it possible for Sierra W Aircard to work with Asterisk?
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13:26.45verywisemankaldemar, i try SetCallerPres , but also sip extension is appearing
13:28.58superbeefI've got a realtime meetme setup, which works great, minus the fact that it breaks out to transfer mode when somebody hits '#' it breaks out into wanting to transfer
13:29.04Tozz_porche: swraid sucks
13:29.24Tozz_and afaik its not recommended to run  your swap on a sw raid device
13:29.28kaldemarverywiseman: sip extension as in sip user?
13:29.45verywisemankaldemar, yes
13:29.48porcheyeah I figured it out with 400 load on asterisk
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13:29.54porcheserver
13:30.14Tozz_that sounds like IO issues
13:30.14porcheunder high traffic, if swap resync starts asterisk goes out of control
13:30.24kaldemarverywiseman: try username and fromuser settings for the peer in sip.conf.
13:30.27porchea possible problem with disks?
13:30.30Tozz_why would it resync?
13:30.41Tozz_it would only do that if there is something wrong with the array
13:30.45porcheonly clue Tozz, system's ram usage is high
13:30.52porchewas high,
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13:31.15Tozz_i think there is a hardware issue. A load as high as that is almost always lots of IO wait
13:31.17porcheit may be possible that, out of ram, and it started huge swap
13:31.33Tozz_yes but huge swap should not cause the raid array to resync
13:31.42porcheoh is it
13:31.48porcheI am not used to raid at all
13:31.59porchejust they told me that raid option will be good
13:32.05Tozz_hardware raid is always better ;)
13:32.07porchebut I forced them to install debian
13:32.14porchethey wanted centos
13:32.23mzaharievTozz_: you are so wrong about the sw raid :)
13:32.24porchenow I am holding a bomb :)
13:32.42Tozz_mzahariev: I hear that alot, but I'm not wrong ;)
13:32.51blitzragesw raid has burned me more than once
13:33.20Tozz_me too
13:33.26porcheTozz, resync started on swap raid btw, not the main disks
13:33.39Tozz_it should not resync at all if there is nothing wrong with the array
13:33.41porchethey never resync other than auto-check, once a month,
13:33.46Tozz_aaah
13:33.54mzaharievfor example take one sw RAID1 and hw RAID1 and see the diference your self
13:34.30Tozz_mzahariev: have a sw raid disk fail (with IO commands hanging) and your box dies
13:34.38SurrealistHi, can anyone ask me if there is a way to perform some actions through Asterisk CLI? I need manage accounts and meetme conferences for my clients(add, remove, modify). The only thing i need to know is if it's possible on that way, 'cause if not i'm plannig on develop a custom app that use freepbx php functions. Thank you!
13:35.20porcheSurrealist, it can be done from manager API
13:35.27porcheCLI is not the right place
13:35.37blitzrageagreed
13:36.08blitzrageyou can do some basic administrative tasks, but not the type of tasks you've given as an example
13:36.18porchetrue
13:36.38Surrealistporche, blitzrage, and it can be scripted with manager API?
13:37.02blitzrageSurrealist, that's the purpose of the AMI -- to allow external applications to control asterisk
13:37.17porcheSurrealist, yes, check AMI or Manager API Asterisk
13:37.21porcheit's event based
13:37.43Surrealistblitzrage, ok, so i'll put my efforts on this. Many thanks, very appreciated :-)
13:38.10porcheso guys, hardware raid
13:38.19porchethat's the right decision
13:38.25porchecentos or debian ? :)
13:38.43Surrealistporche, thanks to toy too ;-)
13:38.51Tozz_porche: whatever you prefer
13:38.54Tozz_so if u like debian, use debian
13:39.02porcheI like debian a lot, hosting guys like centos, if I push them to debian, I know they will screw up the things
13:39.13Tozz_then use centos ;)
13:39.17porchehehe
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13:39.40Tozz_for asterisk it doesnt matter
13:39.45porchecentos 32 bit or 64 bit?
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13:40.13Tozz_64-bit
13:40.27porcheTozz, I was a centos guy once upon a time, but after I start using debian, what ever box I put, I use debian.
13:40.29Tozz_otherwise you will not be able to use >4G ram
13:40.39porcheyep, time to switch back to centos
13:40.43porchethank you Tozz
13:40.55Tozz_np
13:43.26blitzrageporche, I use both CentOS and Ubuntu (server versions)
13:43.59porcheblitz, so you are happy with centos
13:44.16blitzrageporche, yes I've been using it for years and we documented it here:
13:44.23blitzragegrrr
13:44.28blitzrageI can't type tilde for some reason
13:44.36blitzrageinfobot, tell porche about newbook
13:44.42blitzragethere, a work around :)
13:45.02blitzragethe answer is always, "use whatever you're comfortable with" because Asterisk doesn't care at all what distribution you use
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14:03.08DelphiWorldhi guys
14:03.53porcheblitz, I totally agree, but when I pushed the DC guys, they ruined everything
14:04.51Surrealistmmm I can't find any related commands in AMI to create sip extensions(for example). So i'll have to develop in the hard way :_(
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14:06.38porcheSurrealist, what are you trying to do?
14:06.59blitzrageporche, I don't understand what you mean by that
14:07.39porcheblitz, I meant, I like debian, DC guys like Centos, I pushed the to debian, and I have a bad box in my hands
14:07.41blitzrageSurrealist, where are they being stored? You either need to modify the sip.conf file directly (or the included file, which I usually do), or use something like realtime to load your peers from a database, and then just modify the database
14:07.45porchewhich is a production server
14:07.46porchenow
14:08.02blitzrageporche, then you should have used CentOS because that's what THEY are comfortable with (the people managing the box)
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14:09.14Surrealistblitzrage, so, there is no way to say for example: add exten 6000 bla bla bla, scripted in a php? Or use a function that you can pass some parameters?
14:09.17porchetrue, blitz, I learnt that
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14:10.16Surrealistporche, i want to add users to my asterisk box through drupal. So, when someone registers a username i'll give him a number call(Ex.6000)
14:10.34blitzrageSurrealist, sure but you have to write the function to do that -- asterisk doesn't write that information to the sip.conf file -- you do. So your web application has to be able to store the information about your devices, and write the information to the sip.conf file. It also has to be able to handle the ability to write to the dialplan to enable the routing of calls via extensions.
14:10.43porchecheck realtime
14:10.52porcheSurrealist, this is not AMI
14:10.59porchethis goes to realtime asterisk
14:11.01porchebasically
14:11.04blitzrageit is MUCH easier to use realtime and just write to the database
14:11.06porcheyou add to mysql
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14:11.19Surrealistyes, but i want to use freepbx too!
14:11.24porcheand asterisks loads from there
14:11.26blitzragegood luck to you then
14:11.47porcheI dont know much about freepbx
14:12.48Surrealistmmmmmmmmm, i think i will emulate what freepbx does. In a nutshell, it loads the values in a mysql db and then parse all data to the config files.
14:12.59blitzrageyuck
14:13.03blitzragebut yes, that's what you need to do
14:13.14blitzrageasterisk won't write to the configuration files for you
14:13.37blitzrageSurrealist, I suggest you read the "Relational Database Integration" chapter at http://ofps.oreilly.com
14:14.10Surrealisti have this book, i'll take a look
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14:15.38blitzrageactually you don't have that book because it hasn't been printed yet :)
14:15.44blitzragewhat I pointed you to was the third edition
14:16.02blitzrage(which has an updated chapter on database integration)
14:16.18Surrealistbut i have a module that loads some data in the freepbx db, so i think it will not be much harder to do the rest.
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14:16.35Surrealistouch, yes i have second edition :)
14:17.19ruben23hi guys
14:17.46Surrealistthank you for your time ;-)
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14:21.40*** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk)
14:22.29BlackBishopany ideas why I can't send sms message > 75 chars ( using chan_datacard ) ?
14:23.01wdoekes2that's an odd limit, isn't it 70?
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14:23.49BlackBishophm .. lemme check
14:24.19OlafsenMs it possible for Sierra W Aircard to work with Asterisk?
14:24.26BlackBishop123456789012345678901234567890123456789012345678901234567890 ( 60 chars ) fails ...
14:25.34BlackBishopsame for 12345678901234567890123456789012345678901234567890 ( 50 )
14:25.37BlackBishopweird
14:25.47wdoekes2ok.. 70 is how many utf16 chars you would get in a single sms, vs. 160 of gsm03.38 7bit chars.. but it's something else then
14:26.29BlackBishopwdoekes2: ok, where do I set what type of encoding it should use ?
14:26.35BlackBishopI'd like it to use 7bit PDU
14:26.49wdoekes2I have no idea.. I haven't used asterisk for anything else than sip
14:27.52BlackBishopok, I had smsaspdu=yes .. "SMS message too long, 70 symbols max"
14:27.56*** join/#asterisk awclin (~alinford@g0962184.demon.co.uk)
14:27.59BlackBishopset it to no ... and I got that message
14:28.02BlackBishopwhich isn't good :/
14:28.22angryuserHello, can someone recall if asterisk 1.2 support late SDP negociation ?
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14:31.08wdoekes2ok BlackBishop, so you either get utf16 (max. 70 chars) or you have to send a binary blob (as pdu), right? http://www.dreamfabric.com/sms/
14:31.55BlackBishopyeah, I know the site .. I made a php to convert texts into 7bit pdu messages ..
14:32.06BlackBishopI was thinking about fully using asterisk though ..
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14:52.38Kattydrags in
14:55.07beardylooks at what Katty dragged in
14:56.55Kattybeardy: flu
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14:57.51beardyKatty: Eep
14:58.19Kattybeardy: eep indeed :< i hate people sharing their germs with me
14:59.05beardyKatty: Only time irchugs is better than real ones.
14:59.15Kattyyesr
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14:59.41beardyfeeds Katty soup
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15:07.27*** join/#asterisk odin917 (~Gavin_Sil@64.61.105.226)
15:07.43odin917hey everyone
15:09.01odin917i have an issue where i have configured a sip trunk in my asterisk box, when i send out the registration I get denied by the carrier. I spoke to a tech on the phone and he said that my box was trying to identify as carrierSuppliedUsername@myIPaddress instead of carrierSuppliedusername@myCarrierHost
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15:09.27odin917not sure if its a setting, or if the carrier doesnt know what hes talking about
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15:12.01drmessanoAnyone using 1.8 branch SVN?
15:12.22tobi-what might be the problem if the calls on my asterisk seem to be only half-duplex?
15:16.05*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
15:16.59schmidtstobi- it depends on your bandwith and how many calls you want to have at once
15:17.40schmidtshalf-duplex means you can only send or receive at once, think what this would mean to a stream in both directions
15:17.46tobi-bandwith is 50/10mbit down/upstream and call number is 1 atm
15:18.26schmidtstobi- you are not talking about half duplex, you mean asynchron, right ;)
15:18.31beardytobi-: Sound in only one direction, or what do you mean?
15:18.35tobi-wont be that many concurrent calls
15:18.45tobi-its like walky-talky
15:19.25tobi-moh stops whenever i transmit
15:19.29jayteewe have 50/10 with Comcast and use Flowroute and it works great. We usually never get more than 6 to 8 concurrent calls at peak but we've got bandwidth to spare.
15:19.30tobi-for example
15:20.11schmidtstobi- then you might have another problem which have nothing to do with half-duplex
15:20.39jayteetobi check the phones for silence suppression or "comfort noise" as some manufacturers call it. if it's enabled disable it.
15:21.37tobi-oh that might be it, ill check, thanks for the hint
15:22.06schmidtstobi- or VAD = Voice Activation detection ;)
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15:26.09i_heart_asteriskhi
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15:52.14luckman212does anyone have any idea why a 'moh reload' would cause my MOH to die? (Ast 1.8.2.3 + mpg123 stream)  I am tearing out what little hair I have left on this one
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15:54.22mechbangirchow can i use one SIP peer account for multiple registration?
15:54.41odin917i have an issue where i have configured a sip trunk in my asterisk box, when i send out the registration I get denied by the carrier. I spoke to a tech on the phone and he said that my box was trying to identify as carrierSuppliedUsername@myIPaddress instead of carrierSuppliedusername@myCarrierHost
15:55.23*** join/#asterisk andyoutside (cec00d82@gateway/web/freenode/ip.206.192.13.130)
15:55.25schmidtsodin917 take a look at the sip.conf.sample file i think you can set the peer@host by yourself in the registration string
15:55.30*** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com)
15:56.16logicwrathCan someone recommend a metered carrier that will burst unlimited channels and let me specify CID?
15:56.24luckman212odin917: check the 'fromdomain=' and 'host=' lines in your trunk contexts
15:56.25andyoutsideis there an easy way to upgrade asterisk that I did from source with yum?
15:57.55odin917thanks guys illl take a look now
16:01.56ClintGoudie-Niceandyoutside, you could make uninstall in your source and then yum install the packages you want. I'd make sure to backup all your config first though. It's probably not going to be pretty.
16:04.11*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
16:04.41andyoutsidewould it be easer to use the source code going from 1.8.0 to 1.8.2
16:04.41odin917luckman212: thanks, i think the fromdoamin did the trick
16:05.01luckman212cool
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16:10.52Kattydriveby hugskwishes Qwell
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16:12.03jayteewaves at Katty as she does her driveby
16:12.47Kattyhugs jaytee
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16:13.13DelphiWorldkick out Katty from the sea
16:13.16Tozz_normally u get shot during a driveby
16:13.19andyoutsideClintGoudie-Nice,?
16:13.21Thornhello
16:13.22Tozz_hugs is something new for me
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16:14.15Thornis it possible to place  a call into a queue without answering it and only answer it when an agent answers?
16:14.37ClintGoudie-Niceandyoutside: dunno. I've really only run from RPM, and occasionally slammed source compiles over the top of the rpm (and then later wiped the box)
16:15.21KattyDelphiWorld: :<
16:15.37DelphiWorldKatty: :D
16:15.51Kattykicking isn't nice.
16:16.05i_heart_asteriskanybody using polycoms and AGI as dialplan having issues after transferring calls?
16:16.35leifmadseni_heart_asterisk: what version?
16:16.41i_heart_asterisk1.6
16:16.44leifmadsenoh wait, that was AMI I'm thinking of
16:16.48leifmadsen1.6 is not a version (and it's not even a branch)
16:17.04leifmadsenbranches are 1.6.0, 1.6.1, and 1.6.2. Version examples would be 1.6.2.15
16:17.06i_heart_asterisk1.6.1.20
16:17.27i_heart_asteriskseems to be a problem that asterisk does not notify AGI script of transfer when using polycom's softkeys
16:17.47i_heart_asteriski opened a ticket on mantis but nobody has responded...
16:17.49i_heart_asteriskhttps://issues.asterisk.org/view.php?id=18745
16:17.56leifmadsenthat's because I haven't triaged it yet
16:18.51leifmadsenwhy does your ticket say a different version?
16:19.05leifmadsenyou just told me 1.6.1.20, which isn't a supported branch of Asterisk
16:19.18jayteelittle did Leif know when he was a boy that someday his job would resemble an emergency room attendant.
16:19.27leifmadsenjaytee: indeed :)
16:19.31chazzamheh
16:19.43i_heart_asteriskleifmadsen...i don't think this bug is version specific
16:19.56leifmadsenI don't think you have proven it isn't ;)
16:20.16i_heart_asteriskit happens on version 1.6.2.13
16:20.18i_heart_asteriskalso
16:20.19leifmadseni_heart_asterisk: this is not a bog
16:20.21leifmadsenbug*
16:20.30leifmadsenhow is asterisk supposed to know your call is a transfer or just another call?
16:20.43leifmadsenthe answer is, it can't -- if you need asterisk to know this is a transfer, you have to use the built in transfers
16:20.59i_heart_asteriskwhat about the REFER ?
16:21.16leifmadsenthat is after dialplan has already happened
16:21.17i_heart_asteriskhow can a phone make a transfer and asterisk not know about it?
16:21.30leifmadsenasterisk does know about it, but dialplan functionality has already happened
16:21.39leifmadsenthere is nothing else to do dialplan/AGI wise
16:21.59i_heart_asteriskso, you are saying there is no fix? only built-in transfer work w/ AGI ?
16:22.48leifmadsenthe only way to inform asterisk that a transfer is ABOUT to happen is with a built in transfer
16:23.26leifmadsenruns off for lunch
16:23.29i_heart_asteriskyes but why can't AGI be notified after transfer?
16:23.37i_heart_asteriskor after REFER ?
16:23.48leifmadsenbecause that's how asterisk was built -- there is no dialplan being triggered after the REFER, just briding
16:23.51DelphiWorldhow come iPhone don't have flash? :-)
16:23.52leifmadsenbridging*
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16:24.37jayteewhy do they call them buildings when they're already finished? they should call them builts.
16:24.44i_heart_asteriskcan there be something triggered after REFER ?
16:25.06i_heart_asteriskit seems to me that this should be addressed as many phones don't use asterisk's built-in transfer, and many pple use AGI w/ asterisk
16:25.22jayteea Glock 9mm? maybe an avalanche in Vail?
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16:36.30DelphiWorldwhat i need to install to make my gtalk channel support encription?
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16:41.24Thornit looks like if I Queue() a call without Answer()ing it first it is only answered when a queue member answers it, is this intended behavior that can be relied upon? (1.8.2.3)
16:43.50leifmadsenDelphiWorld: you can't
16:44.10leifmadsenDelphiWorld: I don't know that Google Talk has encryption support at all
16:44.12DelphiWorldleifmadsen: so how do i connect to google talk
16:44.21leifmadsenuse chan_gtalk
16:44.24DelphiWorldleifmadsen: gtalk support tls
16:44.37leifmadsenI don't understand your question then
16:44.47leifmadsenyou use chan_gtalk to connect to Google Talk.
16:45.00DelphiWorldleifmadsen: i'm getting encription  error in the log.
16:45.27leifmadsenshrugs
16:45.30leifmadsenyou haven't shown anything
16:45.37DelphiWorldleifmadsen: while show.
16:45.38leifmadsenvague questions get vague answers
16:46.38DelphiWorldleifmadsen: :)
16:47.10i_heart_asteriskleifmadsen, there is no way to patch asterisk in order to notify AGI after a REFER?
16:47.43leifmadsenthere are probably ways to patch asterisk, sure -- you just have to figure it out. It's probably easier to just monitor AMI for the transfers and then perform an action based on that
16:47.55leifmadseni_heart_asterisk: if there is a way to patch it, that would be a feature request
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16:51.18DelphiWorldohhhh drmessano-lt
16:51.38i_heart_asteriskwhy can't digium accept that this is a bug in asterisk
16:51.58i_heart_asteriskmany phones don't use asterisk's built-in transfer, and many pple use AGI w/ asterisk
16:52.54*** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net)
16:53.51chazzami_heart_asterisk: he said it would be a feature request, not a bug
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16:57.48leifmadseni_heart_asterisk: what you're describing is core asterisk functionality and just how things work. There is no dialplan or other functionality after the REFER and channels are bridged, so there is nothing else that can happen there to notify the AGI that the transfer happened. if you need that information you need to monitor for the transfers via the AMI just like all other applications.
16:57.58leifmadsenyou saying it is a bug doesn't make it so
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17:00.24i_heart_asteriskthis may not be a core problem in asterisk but it is a problem that should be addressed, making it a feature request will just put it on the back burner and probably never be implemented.
17:01.15Thornwhat would be the best way to log things like queue waiting time, talking time etc into a database?
17:01.17leifmadsenthis is not a bug
17:01.41leifmadsenThorn: you could use cdr_adaptive_odbc and write information that way into the DB via CDRs
17:01.51leifmadsenalternatively you could take a look at CEL (included in 1.8)
17:02.10i_heart_asteriskleifmadsen, you believe this is not a problem w/ asterisk's functionality?
17:02.15leifmadsenThorn: also check out queue_log to see if there is enough information being logged there for your use. CHeck out the chapter on ACD in the new book
17:02.16leifmadsen~newbook
17:02.16infobotHelp review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/  Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/
17:02.32leifmadseni_heart_asterisk: I believe this is "just how it works" and is not a bug
17:03.06Thornbtw, queue_log doesn't exist here, does it need to be enabled?
17:03.11i_heart_asteriski understand it is "just how it works" but it is a serious limitation and "how it works" should be fixed
17:05.02Thorn(I have master.csv and messages but not queue_log, 1.8.2.3)
17:05.04leifmadseni_heart_asterisk: that is the definition of feature request that you're saying, which means you're welcome to provide a patch that addresses the deficiency. If you are unable to provide a patch for said functionality you are welcome to hire a consultant. Beyond that, there is nothing else to be done here.
17:05.33leifmadsenThorn: http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id289009
17:08.58i_heart_asteriskliefmadsen, can you at least point me in the direction of where the change in code might occur?
17:09.40leifmadsenI'm not a programmer
17:09.45leifmadsenso I can't help you there
17:10.26i_heart_asteriski see... well thanks for nothing then
17:11.08leifmadsenglad I could help
17:11.13Thornno luck with queue_log :( it was not disabled in logger.conf, I set it queue_log=yes, queue_log_to_file=yes explicitly but still no log file
17:11.34i_heart_asteriskyou mean couldn't help
17:12.19leifmadseni_heart_asterisk: I guess my sarcasm wasn't thick enough
17:12.39jayteeI wonder if his first name is "Richard"?
17:12.47drmessano-ltNot as thick as his "thanks for nothing"
17:12.50drmessano-ltBut close
17:13.13drmessano-ltDamn you for not being a programmer, you unhelpful SOB!  (or something)
17:13.13jayteebecause he sure comes across as a "Dick"
17:13.21nestArlol
17:13.27i_heart_asteriskthis is precisely why asterisk is going to fall by the wayside to freeswitch
17:13.35drmessano-ltlol
17:13.43nestArperhaps it's time for a change of nick
17:13.47drmessano-ltBecause Leif isn't coding?
17:13.47nestArjust sayin'
17:14.01jayteeso are you going to change your nick to i_heart_freeswitch ?
17:14.05i_heart_asteriskwhat does leif do? salesmen for digium?
17:14.22drmessano-ltNo, he wrote the books you should have read
17:14.37jayteeand probably does more in a morning than you do all week
17:14.43i_heart_asteriski don't need to read about asterisk, i already know it's broken
17:14.46leifmadsenI'm the release manager and primary bug marshal for Asterisk, consultant, and author
17:15.19jayteewhere is that highway sign when you need it :-)
17:16.36nestArlotta hate
17:16.56*** join/#asterisk Lensman (~Lensman@169.130.124.38)
17:17.02i_heart_asteriskremember, i heart asterisk...it hurts me to see thses flaws and nothing done bout it
17:17.07drmessano-lti_heart_asterisk, I would say if Asterisk were to fall, it would be due to the lack of community involvement, such as when someone finds a missing/incomplete/poorly implemented feature in Asterisk and rather than attempting to be part of the solution and contribute to fixing it, they just complaint and troll
17:17.31i_heart_asteriskdrmessano-lt, i am more than willing to put in time for this fix
17:17.36drmessano-ltSo do it
17:17.56i_heart_asteriski already asked leif where i could look, but somehow he doesn't code
17:17.57drmessano-ltLeif is not a coder.. don't bash him for not being helpful.. Find someone who can help and work towards a solution
17:18.02drmessano-ltHe doesn't
17:18.11drmessano-ltHe told you he doesn't
17:18.23i_heart_asteriskok, does anybody code in here?
17:18.33drmessano-ltSo find someone who does.. Put up a bounty somewhere.  Hire a consultant
17:18.44leifmadseni_heart_asterisk: I'm sorry I can't tell you it's like 1369 of chan_sip.c that you need to add if (im_awesome == 1) { do_awesome_stuff(1); }
17:19.01drmessano-ltHonestly, I think you owe him an apology
17:19.18leifmadsenI've already moved on and have a new girlfriend, so it's not necessary
17:19.20i_heart_asteriski'm not going to hire a consultant... where is no development community
17:19.24jayteelol
17:19.45drmessano-ltIf you check the bug tracker, the development community is alive and well
17:19.49i_heart_asteriskdigium just wants me to pay for a fix
17:19.55drmessano-ltlol
17:20.04i_heart_asteriskbugs = job security
17:20.15drmessano-ltHave you ever looked at the bug tracker?
17:20.16n3hxsBS
17:20.39leifmadseni_heart_asterisk: no they don't, and you can't hire Digium to program your feature request either. That's what other companies do though.
17:20.49n3hxsthinks heart is misspelled... hate.
17:20.51leifmadsenI'm sorry your issue is not as important to resolve as the other issues.
17:21.04leifmadsens/issue/feature request/
17:21.06jayteethere's always barber college
17:22.10drmessano-ltYou could have already posted a bounty somewhere
17:22.17leifmadseni_heart_asterisk: you're welcome to describe your problem on asterisk-users to get a second opinion on your issue. It's entirely possible I'm wrong, or misunderstand your situation and what you're trying to do, and it really is a bug, in which case your issue could be reopened. But continuing to pout about it in here is not all that productive.
17:23.20i_heart_asteriskliefmadsen you are not wrong, you are correct that the dialplan is not executed after the REFER, the problem is it should
17:24.05leifmadsenthen you're welcome to contribute that particular functionality, or find another way to work around it (such as monitoring for transfers via the AMI as I've suggested previously)
17:24.55i_heart_asteriskindeed, i will see about using AMI, just saddens me nobody agrees this is a seriously flaw
17:25.34p3nguinIf no one else ever had a problem with it, maybe the severity isn't as great as you think it is.
17:25.40jayteenot a serious flaw as most installations don't require that functionality. yours is probably a rare case.
17:25.58leifmadsenI'm not sure what else I can say, so I'm not going to say anything else
17:26.01i_heart_asteriskjaytee, anybody using AGI for dialplan and polycoms will experience this problem
17:26.04leifmadsengoes back to triaging issues
17:26.56jayteei_heart_asterisk, you could disable the softkeys for transfer on the Polycoms and use feature keys but I'm not sure the AGI would accomodate that either.
17:27.16*** join/#asterisk cVsup (~cVsup@189.83.218.188)
17:27.21*** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164)
17:27.28cVsupi have xfo interface as X100P
17:27.35drift-can you have multiple g729 cards in same server?
17:27.46cVsupin my country the caller ID signal as is DTMF
17:28.02cVsupi can use Caller ID?
17:28.15cVsupwith X100P:
17:28.17cVsup?
17:28.29i_heart_asteriskjaytee, i'm considered that but i'm not sure i want to disable the xfer softeys on the polycoms as our customers would not like adjusting to differences
17:29.03leifmadsendrift-: I think so, but that is probably a better question for Digium sales/support
17:30.11JerJeri_heart_asterisk:   hopefully when you mention AGI you really mean FastAGI
17:30.23JerJerbecause, you know, regular AGI does not scale
17:30.27*** join/#asterisk ccesario (~ccesario@187.75.139.188)
17:30.46drmessano-ltdrift-, multiple transcoder cards?  You've managed to impress me.
17:30.54drift-yes
17:30.56drmessano-ltThat just happened
17:31.04drift-lol :)
17:31.23drift-instead of making 4 servers id rather have 2 servers with 2 cards
17:31.32drift-trying to beable to get 300 concurent calls
17:31.38jayteewhy the -lt in the nick now?
17:31.48drmessano-ltI am on my laptop
17:31.50drmessano-ltAt work
17:31.57drmessano-ltMy ILLEGAL LINUX MACHINE
17:32.00jayteeaha! makes perfect sense now that I think about it
17:32.09drmessano-ltstares at the Windows boxes all around
17:32.18leifmadsendrift-: like I said, I'd just ask Digium since they would know for sure if that scenario is supported
17:32.19jayteehahaha, illegal linux machine?
17:32.30drmessano-ltWe don't speak of the Penguin around here
17:32.33*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
17:32.53drmessano-ltLinux = Evil, or so says the Domain Controller in the basement
17:33.05leifmadsenthe comptroller?
17:33.17leifmadsengoes to make another batch of coffee as his right eye stopped twitching
17:33.22drmessano-ltlol
17:33.55jayteehahaha, sounds like here. We have an Asterisk phone system I setup, Polycom phones and I've replaced our POS(not point of sale) Netgear Firewall/Router with Pentium D box running Vyatta 6.1 and yet all my coworkers are Windows developers who still bash linux.
17:34.17fileugh Vyatta
17:34.28fileI was evaluating solutions for my home network router and signed up for documentation access
17:34.28drmessano-lt"Why does your screen look all cool and 3D-ish?"  "I am running Linux"  "Oh, brb, getting the torches and pointy sticks"
17:34.33filethey won't stop emailing and calling me
17:34.45jaytee:-)
17:34.53jayteepersistent buggers
17:34.53fileended up going with pfSense
17:34.54*** join/#asterisk Alborracho (~chatzilla@186.6.152.45)
17:35.12drmessano-ltI can't use anything BSD based
17:35.56JerJeri like to roll my own.  PC Engines ALIX board+Voyage Linux
17:36.25JerJeri have also used routerboards
17:37.46fileI use a dual core Atom 1U server
17:37.49fileit works well.
17:38.54jayteeI built a mini-itx box for under 300 bucks with an Intel Atom D510MO board
17:39.25drmessano-ltI am building a couple systems with SuperMicro 1U boxes.   Right around $300 each
17:39.27jayteebit overkill but I've used it to test both Vyatta and Asterisk on it.
17:40.08jayteeand drmessano-lt, I'm really satisfied with Flowroute and most importantly, so is my boss.
17:41.50drmessano-ltGetting ready to overhaul a $1.25 Million Mobile Command vehicle and install Atom based Ubuntu desktops, an Asterisk system, and moving from POTS/Cell phones to Flowroute
17:42.23paulcMobile Command sounds kinda.. sexy? what's it for?
17:42.33Alborrachohi is there any other way to change the volumen in asterisk, i have a te420b connected through T1s and the audio is really high, ive changed the ss7.conf rxgain and txgain but everything keeps same
17:43.17leifmadsenAlborracho: you could try the VOLUME() function
17:43.21Alborrachohmm
17:43.31Alborrachoin extension.conf right?
17:43.35leifmadsenright
17:43.38leifmadsencore show function VOLUME
17:43.45Alborrachook let me do a test
17:43.57jayteeand changes to gain settings won't take effect until you restart DAHDI
17:44.00drmessano-ltIt's a mobile interoperability device.  Houses a dispatch center, conference room, a dozen or so radios we can interconnect, multiple forms of connectivity, etc
17:44.08leifmadsen(VOLUME() is probably in Asterisk 1.6.2 and above)
17:44.23drmessano-ltpaulc, it's a big radio/telephony wirenut
17:44.47leifmadsenI have an awful router from Rogers which is made by SMC... I should probably do something like what file did and use something embedded to be a router.
17:44.56leifmadsensome day...
17:45.08*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)
17:45.15jayteeI priced out nano-itx and pico-itx systems but the costs are higher for parts than mini-itx systems.
17:45.41paulcdrmessano-lt: Sounds cool.. any pics?
17:45.50Alborrachodo you know what is the maximun and minimum rx and tx gain?
17:46.52drmessano-ltpaulc, not sure what we have online.  Let me see
17:48.01drmessano-ltpaulc: http://www.columbiacountyga.gov/Index.aspx?page=3205
17:48.28drmessano-ltA lot of that info the OLD/Current truck configuration.  90% of the comm equipment is being ripped out thanks to our most recent grant
17:49.36paulcThat's pretty cool..  so you drive it to wherever and then... satellite uplink for data?
17:50.05jayteeAlborracho, with zaptel you could use -100 to 100 but it was always recommended not to go over -11 to 11.
17:50.21jayteenot sure if that's changed in DAHDI
17:51.07Alborrachothx
17:52.26drmessano-ltWell, the current Satellite data system is b0rk3d.  It never really worked.  I implemented EVDO, as the truck is generally outside of the hot zone where it's safe and some chance of a cellular network existing.. but that wont always be the case.  We're getting a new system with this grant.  We also have some Wifi clients to grab Wifi from wherever.. even free Wifi from a McDonalds if that's all that's around lol
17:53.55*** join/#asterisk mechbangirc (~mechbangi@115-186-140-40.nayatel.pk)
17:54.27paulcimagines the truck parked outside Starbucks.. coffees for everyone and a shot of free wifi please :)
17:54.38drmessano-ltlol
17:54.47mechbangircmy AMD() does not work no matter what i do. (2500|1500|300|5000|120|50|2|256). it always consider me machine
17:55.11*** join/#asterisk timahvo1 (~rogue@41.223.57.76)
17:55.27paulcmechbangirc: You're a machine. And you're becoming self aware? ;-)
17:55.51mechbangircpaulc: thanks for leaving me this msg
17:56.10drmessano-ltWe've used Wifi from a restaurant during a test deployment.  It was a nice test of "when all else fails".
17:57.42paulcdrmessano-lt: So are you doing away with satellite totally? or still have it as the ultimate fall-back position?
17:58.48drmessano-ltWith this new grant, satellite will again be the primary method of connectivity.  EVDO follows that, and throwing out the pelican case with the Wifi client will be a distant third.
17:59.18*** join/#asterisk MikeH (~mike@86.63.17.141)
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18:01.26paulcNice. Cool project to work on. And slightly more exciting than my current project, perhaps.. although it has its moments too..
18:02.35drmessano-ltWhich is?
18:07.27p3nguinSecret.  Apparently.
18:12.14Alborrachojaytee: i changed txgain to -10 , then restart dahdi, but it had no effect
18:12.31Alborrachoits really high
18:13.31paulcCurrently working on Asterisk as a mechanism for connecting people who want to pay to talk, with people who get paid to talk.. in a sort of.. "ooh yeah baby, that's right, mmm yes" kind of way
18:13.36Alborrachoi think its a hardware problem with my card...
18:14.37drmessano-ltohhh
18:15.27drmessano-ltThe village people had a song about that
18:18.23Thornleifmadsen: thanks for the new book, it's excellent
18:23.46*** join/#asterisk Ool (~Ool@unaffiliated/ool)
18:24.10*** join/#asterisk wwalker (~wwalker@208.92.232.27)
18:24.56wwalkerstrange voicemail behavior: https://gist.github.com/814886      asterisk says its recorded, then says it's saved, then there is no message....
18:25.16citywokwwalker: make sure asterisk has write permissions to /var/spool/asterisk?
18:25.21fenrusdoes asterisk have permissions to save the files?
18:25.22fenrus:D
18:26.00wwalkerasterisk is running as root (taking over an existing installation, I'll fix perms when they have basic phone functions)
18:27.59*** join/#asterisk OpenSourceWay (~OpenSourc@AToulouse-257-1-54-48.w90-5.abo.wanadoo.fr)
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18:37.09*** join/#asterisk andrei66 (~chatzilla@AMontpellier-151-1-8-211.w92-143.abo.wanadoo.fr)
18:37.26andrei66Hi all
18:40.53*** join/#asterisk andrei66 (~andrei66@AMontpellier-151-1-8-211.w92-143.abo.wanadoo.fr)
18:40.59andrei66hi all
18:41.07*** part/#asterisk OpenSourceWay (~OpenSourc@AToulouse-257-1-54-48.w90-5.abo.wanadoo.fr)
18:42.19fenrushi
18:43.03wwalkerI've run strace on asterisk, and I see asterisk rename() the recording into INBOX and the msg00000.txt file into INBOX.  then they are gone :(
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18:46.45stefmtlHi. I have a question: is it possible to use some Digium Hardware to replace a ADSL modem) ?
18:47.01leifmadsenno sir
18:47.14leifmadsensangoma makes an ADSL modem
18:47.27andrei66i think you can only use a digium card for 56k or fax
18:47.30leifmadsen(it's currently sitting in a box in my drawer if you're interested in buying it)
18:47.34stefmtlok thanks leif
18:50.50*** part/#asterisk Ool (~Ool@unaffiliated/ool)
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19:05.56*** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net)
19:06.00d_preston215Is there a way to use a speed dial to transfer a call to a queue?
19:06.10*** join/#asterisk vinhdizzo (~vinh@dhcp-v011-239.mobile.uci.edu)
19:07.02Thornwhat does queue_log in extconfig.conf do?
19:08.18citywokThorn: I've never done it, but i'm guessing it's so you can write queue_log events to a database instead (or in addition to?) the flat file in /var/log/asterisk
19:08.50andrei66I have a problem : i use asterisk on a ubuntu pc for 3 SIP phones
19:09.15Thornall the other things in extconfig.conf are for reading from databases, aren't they?
19:09.16andrei66and i use a external SIp provider (asterisk registers as client) to dial outside
19:09.50andrei66but i can hear, but who i'm calling can't hear me
19:10.57*** join/#asterisk theHub (~karl@69.177.93.21)
19:11.13paulcd_preston215: You mean like have a single key on the phone that transfers the call to a queue/extension?
19:13.54d_preston215Yes.
19:13.57d_preston215Exactly that.
19:14.31citywokThorn: i know sippeers reads & writes.  lol.
19:15.39Thornyes looks like it, thanks
19:18.23*** join/#asterisk Dovid (~Dovid@213.8.121.90)
19:19.26*** join/#asterisk timahvo1 (~rogue@41.223.57.78)
19:20.04Dovidhello all
19:21.12luckman212is the correct place to submit a bug to Digium https://issues.asterisk.org/main_page.php ?
19:21.29d_preston215Any ideas @paulc
19:22.25rgagnonluckman: yes, but please search to see if it may have already been submitted
19:24.27luckman212rgagnon: the bug I want to file is about mpg123 dying after a moh reload when moh is using res_timing_dahdi .. how would I even search for that?
19:25.08rgagnonvery carefully?
19:25.55luckman212slaps himself on the forehead
19:25.59luckman212of course
19:26.07rgagnonI think they just care that people don't duplicate things without at least checking
19:27.11rgagnonyou can search from here https://issues.asterisk.org/view_all_bug_page.php maybe for "mpg123"
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19:28.51*** join/#asterisk jkroon (~jkroon@dsl-242-3-152.telkomadsl.co.za)
19:31.16leifmadsenluckman212: or look through the list of issues filed against Resources/res_musiconhold etc.
19:31.31*** join/#asterisk ccesario (~ccesario@187.75.139.188)
19:33.14luckman212leifmadsen: thanks, I'm in there
19:34.43luckman212is it really possible that there are only 5 open issues for res_musiconhold ?
19:34.43*** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano)
19:35.18raden_workanyone seen naikorvek ?
19:35.56leifmadsenluckman212: yes that is certainly possible -- it'd be ideal to have zero open issues against res_musiconhold
19:36.05leifmadsen~seen naikorvek
19:36.10infoboti haven't seen 'naikorvek', leifmadsen
19:41.13leifmadsenM18737
19:41.18leifmadsenseanbright: that might be something you can look at if you're bored :)
19:41.22leifmadsenbah!
19:41.24leifmadsenwrong room
19:44.48paulcd_preston215: delayed reply, sorry... I think it depends on the phones you're using as to whether it's possible or not. What phones are you using?
19:45.06*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
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19:46.58d_preston215Cisco 7960
19:47.32d_preston215I'm going to try to just do the transfer feature code and extension to see if that works.
19:50.25paulcd_preston215: Not sure it'll work but see how you go.. I think a guy in here did something similar with Aastra phones, they seem to be a bit more flexible and nice for Asterisk integration. Let me know if you get it working though :)
19:50.33*** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net)
19:50.52MiccWhy does asterisk think another asterisk server has VAD enabled?
19:51.03*** join/#asterisk corygstuart (~corygstua@mysterymachine.dis.anl.gov)
19:51.42corygstuartHello.  I'm going to build an asterisk server and was looking for hardware recommendations.  What's worked for you?
19:54.08MiccI'm guessing the comfort noise message in asterisk is showing the wrong IP or getting it confused with one of the phones. They only phones on the system are SPA942 but I can't find CNG or VAD anywhere in the settings of the phone to disable it.
19:54.09paulccorygstuart: I've done stuff with DL-385s and a few proof of concepts using HP desktop PCs..  I hear Dells are pretty good.. there's a hardware page on the wiki too
19:54.14paulc~hardware
19:54.14infobotextra, extra, read all about it, hardware is http://www.digium.com/index.php?menu=hardware_products. If you don't know what you need, start with an TDM400P and an FXS module.
19:54.36paulcthat wasn't quite the link I was thinking of...
19:54.39MiccI think I remember asking this question about comfort noise before and the answer was its broken and ignore it I think.
19:55.41d_preston215Yeah, feature code + extension on a speed dial didnt work.
19:56.12corygstuartThanks, paulc.  I'll jump on the wiki and take a look.
19:58.36d_preston215Its probably because the speed dial has no input space between the feature code and extension, so asterisk doesn't have enough time to process it.
20:00.41corygstuartOK.  I'm seeing interface cards, has anyone put together any inexpensive servers (e.g. comparable to a SwitchVox)?
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20:19.22devmodwhen was 'same =>' added?
20:27.21leifmadsendevmod: 1.6.2
20:27.35leifmadsen<3 same =>
20:27.51devmodleifmadsen, yup its awesome
20:27.53*** join/#asterisk superbeef (~Lane.Jenn@74-120-209-234-rev.redanvil.net)
20:28.05leifmadsenI use it all the time
20:28.11leifmadsenI think it came from a suggestion I had years ago :)
20:28.21WIMPyalways forgets te n, when using it.
20:28.29WIMPythe
20:28.36leifmadsenI've done that sometimes, but most of the time I'm pretty good
20:28.52drmessano-ltloves same =>
20:29.04leifmadsenloves drmessano-lt
20:29.11drmessano-ltOhhh <3
20:29.35leifmadsenunfortunately it's 'not the truth' monday
20:29.55drmessano-ltYes, and I could love you too because that would create a goto loop
20:29.57devmodI'm testing someone's repo that supposedly fixes an issue and it seems its based on 1.6.0 . Where are patches committed to once approved?
20:29.59drmessano-ltSorry :(
20:30.06drmessano-ltcouldnt
20:30.15drmessano-ltas in, I couldnt mess that joke up and just did
20:30.21_Corey_hmmm, that same thing will save a few keystrokes... :)
20:30.58leifmadsendevmod: just the branches that are being supported
20:31.04*** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn)
20:31.06leifmadsen(1.4, 1.6.2, 1.8, trunk at the present time)
20:31.07drmessano-ltsame is cool because it reminds me of having someone ask you the same question over and over and over again and getting annoyed
20:31.36drmessano-lt"What extension?  SAME"  "what extension?  SAME"  "ok, what about this one?  SAME"
20:31.39leifmadsendevmod: the branch is being done against 1.6.0 just because that developers customer is using 1.6.0, and not because 1.6.0 (or 1.6.1) is supported
20:31.54leifmadsendrmessano-lt: what's even more confusing is the 's' priority :)
20:32.04leifmadsensame => s,NoOp()
20:32.18devmodleifmadsen, ohh ok. That is what I thought but then this ticket is set to ready for testing while the patch proposed is against 1.6.0 so i got confused
20:32.20drmessano-ltassplodes
20:32.26leifmadsen(although 's' is pretty much NEVER used, and would just be for something where you want to do s+101)
20:32.37leifmadsendevmod: ya well oej is confusing sometime s:)
20:32.53leifmadsendevmod: you'd have to request the developer perhaps provide patches for 1.6.2 or whatever branch you're using
20:33.33*** join/#asterisk Mw3 (~mw3@mw3.hu)
20:33.49voxterAnyone know of an ADSL Modem + Router combo that supports handing out DHCP Option 66?
20:34.08devmodleifmadsen, ohh ok haha it's fine I will just test that - Just wondering
20:34.22leifmadsenno worries
20:34.37leifmadsenvoxter: Sangoma ADSL modem on Linux? :)
20:35.01voxterleifmadsen: haha :) Looking for a small all in one unit, like a linksys-sized device
20:35.16leifmadsenvoxter: ADSL modem in a Soekris? :)
20:35.37voxterleifmadsen: oh! also something that costs less than a linksys wrt + thompson adsl modem already does.
20:35.39leifmadsenvoxter: PS: ping me about a beginner DJ set list
20:35.54voxterleifmadsen: sure! i've got a good list for ya.
20:36.20leifmadsenvoxter: learning Ableton Live with an X-Session Pro and Launchpad and need to practice mixing but would love a set list that I know already "goes together"
20:36.22p3nguinWhen a call to my Google Voice phone number never reaches Asterisk, what should I look at to fix it?  Outbound calling via Google Voice works fine.
20:37.10voxterleifmadsen: you know ive been meaning to learn how to use ableton to dj.  I normally use torq.  I'll hit ya up someplace else and we can chat about it some more.
20:37.45adyn~pastbin
20:39.02leifmadsen~pb
20:39.02infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
20:39.05adynthank you
20:42.08*** join/#asterisk Lensman (~Lensman@169.130.124.38)
20:59.25*** join/#asterisk awclin (~alinford@g0962184.demon.co.uk)
21:10.11adynI'm having an issue where I have a several pri's terminate into 2 asterisk boxes which then send the calls to their individual destination pbx's on one pbx randomly as far as I can tell it just stops taking calls I'm thinking its some sort of time out but I can't figure out where, none of the searches I've tried have turned up anything meaningfull, here logs and my iax.conf snippets: http://asterisk.pastey.net/145822
21:11.28adynalso when I try core stop now asterisk just locks up and I have to kill the process manually in order to restart it
21:11.51*** join/#asterisk solokkhz (solokkhz@bzq-79-179-198-191.red.bezeqint.net)
21:11.51adynAsterisk version is 1.6.2.16.1
21:12.42GTXCommIf I want to dail 7 diget dialing and put in 913 without dialing it, how would I go about doing that, if anyone knows off the top of their head, otherwise, I will continue to look IE. _XXXXXX => dail(SIP/provider/${EXTEN});
21:13.18GTXCommArea code
21:14.06GTXCommAEL
21:14.20WIMPyMy most favourite dialplan issue.
21:14.35leifmadsenGTXComm: exten => _NXXXXXX,1,Dial(SIP/provider/913${EXTEN})
21:14.51leifmadsenyou'll have to convert to AEL yourself because I don't use it (but it's obvious what I just did)
21:14.56GTXCommFound it.
21:15.13GTXCommThanks Lief.
21:15.42WIMPyAnd now do it fo a number with unknown length.
21:16.52nestArit's just _X. isn't it?
21:17.27GTXCommworked.
21:17.42WIMPyOk, now the final one: Do it without a timeout.
21:18.33leifmadsenprogressinband=yes I think
21:18.47leifmadsenunless I misunderstand the question :)
21:18.59WIMPyI think you did :-)
21:20.01leifmadsenI'm also trying to figure out why I can't get my VMs to start in KVM
21:20.04WIMPyI'd like to add the area code for interactively dialled numbers of unknown length in order to be able to use enum and to avoid hacks in dundi.
21:20.35WIMPyBut as far as I can see that's not possible without timeout.
21:21.02leifmadsenyou mean pattern matching as you dial the digits? (I'm trying to remember what that is called)
21:21.02WIMPyAnd it makes it impossible to interactively match dundi entries.
21:21.13leifmadsenwhere each DTMF tone is sent as an INFO msg
21:21.13WIMPyoverlap dialling
21:21.17leifmadsenthat's the one
21:21.18WIMPyyes
21:21.27leifmadsenis that what you mean?
21:21.45leifmadsenyou'd have to send a DUNDi request for each overlap dialed number....
21:21.52WIMPyAs far as I can see that would require the possibility to modify EXTEN and re-enter the dialplan.
21:22.06leifmadsenotherwise you need to use something like precaching in dundi
21:22.20WIMPyYes, that works very well. As lond as all your numbers are within the same area code.
21:22.28WIMPylong
21:23.18WIMPyi.e. if I define my extensions both with and without area code.
21:23.46WIMPySo I need to have one dundi context per area code to avoid false matches.
21:24.21WIMPyBut that's not possible for public ENUM.
21:40.05_Sam--anyone looking for a project?   i could use a hand trying to move our 1.4 asterisk from a physical server here at our location, to a rackspace cloud server
21:40.43_Sam--should be like 1-3 hours max.   but not for me, apparently.
21:40.51p3nguin_sam--: Do you mean a VPS?
21:40.58_Sam--p3nguin:  yeap
21:41.18_Sam--both machines (physical and VPS) are debian
21:41.46p3nguin_sam--: How soon do you need to have it completed?
21:42.03_Sam--no huge rush.   just a longer term goal i'd like to accomplish
21:43.10*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
21:43.21p3nguin_sam--: I'll PM you if it's okay.
21:43.30_Sam--pls do
21:45.52*** join/#asterisk Da-Geek (~Da-Geek@90.155.74.11)
21:46.32*** part/#asterisk Da-Geek (~Da-Geek@90.155.74.11)
21:51.52*** join/#asterisk mawhii (~mawhii@99-117-188-231.lightspeed.dybhfl.sbcglobal.net)
21:52.24*** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net)
21:56.26*** join/#asterisk theHub (~karl@69.177.93.21)
21:58.47*** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net)
21:59.04WIMPywonders if WaitExten could be modified and abused...
21:59.20*** join/#asterisk darkskiez_ (~dz@62-50-199-254.client.stsn.net)
21:59.47kukuAny reason why an inbound call would go to fast busy after two rings when connected to a TDM400p, even when asterisk is not running ?
22:00.42*** part/#asterisk joeyjones (~joeyjones@93.186.171.52)
22:01.31KobazSystem uptime: 30 weeks, 7 hours, 19 minutes, 12 seconds
22:01.32Kobazmmm
22:02.54*** join/#asterisk killown (~killown@unaffiliated/killown)
22:03.43killownwhat's the best asterisk gui?
22:04.00kukufreepbx in my opinion
22:04.45killownkuku ok thank you
22:05.06*** join/#asterisk nix8n82 (~nate@63.162.28.112)
22:06.04killownkuku, it has an advanced call recording and true hot desk support?
22:09.26leifmadsendoubt it :)
22:09.36*** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net)
22:10.39*** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
22:10.40killownI'd like to know a complete IP PBX solution based on Asterisk
22:11.08_Sam--killown:  switchvox?
22:13.20p3nguinYour definition of complete is probably different from that of many others here.
22:16.45Kattyhi leif
22:17.45leifmadsenyo
22:17.57*** join/#asterisk powerunits (~as@116.71.187.93)
22:19.08killownI need an asterisk gui prefessional and for end users
22:22.21powerunitshi every one...
22:22.34powerunitsim facing a issue on inbound
22:22.37powerunitshttp://pastebin.com/PCt4GrG7
22:22.48powerunitsthis is my config file and asterisk CLI result
22:22.55p3nguinkillown: Tried FreePBX?
22:23.01p3nguinpowerunits: pastebin
22:23.08p3nguinpowerunits: oops, you already did.
22:23.09killownp3nguin not yet
22:23.17p3nguinpowerunits: sorry  :/
22:23.22powerunits:) no pro
22:24.34*** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164)
22:24.39p3nguinpowerunits: What is the exact problem you're having?
22:25.02p3nguinpowerunits: I see a couple problems, but I doubt they are what you were meaning.
22:25.16powerunitsi am unable to recive inbound calls
22:25.28powerunitswhen i dial my inbound number
22:25.38powerunitson asterisk it give me error msgs
22:26.00p3nguinpowerunits: There is more to the problem than what I see in the pastebin.
22:26.24powerunitshumm
22:26.35powerunitsp3nguin: wht is that?
22:27.09p3nguinpowerunits: Hold on, I'm trying to put the puzzle pieces together on my own.
22:27.18powerunitsok
22:27.40p3nguinpowerunits: What is the "sipaccount" peer?
22:27.53p3nguinThat's your ITSP?
22:28.35powerunitssorry wht is ITSP?
22:29.06powerunitsthis sip account from where i am doing outbound and want to recive inbound calls as well
22:30.17*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net)
22:30.34p3nguinITSP means your provider.
22:30.43powerunitsyes
22:30.46powerunitsit is
22:32.33p3nguinpowerunits: I would prefer to see your configuration more like this:  http://pastebin.com/LFS58tFT
22:32.39p3nguinIt is a much better idea.
22:33.12citywokpowerunits: do you register your peer?
22:33.31p3nguinpowerunits: After you make it like that, make another call to your phone number and show me the error from the CLI.
22:34.00p3nguin"core set verbose 4"  before making the call
22:34.32powerunitslet me see plz
22:35.41powerunitscitywok: yes my peer is getting register.
22:36.14*** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net)
22:36.35adeelhas there been any major modifications in sip.conf from 1.4 to 1.8?
22:36.41Tozz_yes
22:37.16adeelTozz_,  mind giving me a quick overview?
22:37.17*** join/#asterisk Faithful (~Faithful@carame.lnk.telstra.net)
22:37.22Tozz_or, the way SIP peers are identified has changed a little. So u might need fromuser= directives
22:38.29adeelthat's nothing too major
22:38.50Tozz_well, we've had issues with SIP peers not getting registered
22:39.29adeelseems like TCP/TLS peers are a bit touchy...i'm still using good old UDP peers
22:39.47citywoki use TCP but only for integration with OCS/Lync
22:39.51p3nguinpowerunits: There's no indication of you registering to the ITSP based on the sip.conf you showed me.
22:40.04p3nguinAnd of course, there's no reason for the ITSP to register to you.
22:43.16Thornwhere is the queue_log table format documented? I'm looking at https://issues.asterisk.org/view.php?id=17082 but res_odbc complains about column "time" which isn't in that example
22:51.17powerunitsp3nguin: sip is registerd. i can show it to
22:51.28powerunitsi have tried your example... but still same
22:51.31powerunitsresults
22:51.35powerunitson asterisk CLI
22:52.30p3nguinI understand that the results will be the same.  That had very little to do with the changes I made for you.
22:52.46p3nguinI still need to see the CLI output from the failed call.
22:53.01powerunitsok let me show you..
22:53.27*** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net)
22:54.31powerunitshttp://pastebin.com/pPh4iPt7
22:58.42Thornhow unique is uniqueid? can I use it as a database key and as a filename for call recordings?
23:02.24citywokThorn: I do.  It's Epoch time + an incremental call counter i think
23:04.24Thornokay, thanks
23:04.33powerunitsp3nduin: did u get any idea?
23:07.29adeelhow is the support for the directrtpsetup option in sip.conf amongst * 1.4 - 1.8? is it reliable?
23:07.59powerunitshttp://pastebin.com/NHEFhbR4
23:08.40powerunitsi don't understand why on inbound call astierks CLI says username mismatch"??4
23:13.07Thornhow do I get ${DIALSTATUS} into CDR? it seems like no dialplan commands are executed after Dial() (I have a NoOp there too)
23:14.48p3nguinpowerunits: Where does this username exetech-verification come from?
23:15.19powerunitsi have to SIP account register with ITSP
23:15.27powerunitsthis is 2nd SIP account
23:16.28powerunitshttp://pastebin.com/wfMdaSW8
23:16.41powerunitsthis is my sip.conf general part setting
23:16.51powerunitsplz check if any thing wrong here?
23:18.54*** join/#asterisk dr00d (~rtp.aster@b27A5.static.pacific.net.au)
23:19.35dr00dhi - anyone in here able to answer a question about skype for asterisk ?
23:20.00Thornlooks like Fg is the answer
23:20.18dr00dwhats fg ?
23:20.33Thornit's not for you
23:21.16dr00dskype for asterisk isnt for me ?
23:21.25Thornmy message was not for you
23:21.29dr00doh sorry
23:22.27powerunitsp3nguin: any hint or idea?
23:22.32powerunits:(
23:24.14*** join/#asterisk killown (~killown@unaffiliated/killown)
23:25.16killownwhat is a better choice, freebpx or trixbox?
23:25.25citywokneither
23:25.39dr00dlol
23:26.18dr00dcan anyone tell me where i have to put Dial(Skype/[<originator>@]<destination>) ?
23:26.46citywokyour dial plan, where you want it to dial?
23:27.06killowncitywok why neither?
23:27.10mzbkillown: isn't that like asking the difference between Ubuntu and Firefox?
23:27.43citywokb/c you haven't given a list of clear requirements, which is something you need when making a decision :)
23:28.01dr00dyes - i dont understand where the dial plan is
23:28.11mzbone is an 'application', the other is a 'distro' (and I use both those terms loosely;))
23:28.24killownmzb I need a gui asterisk tool for end users, easy to manager to do things like listen recorded calls and another things
23:28.28citywokyea, trixbox is a full OS+Asterisk+freepbx
23:28.55mzbkillown: your preferred distro is???
23:29.00p3nguinkillown: Go with AsteriskNOW if you need a full OS with the tool included.  Do not consider Trixbox if you value support.
23:29.37killownmzb, I have no preferred distro
23:29.37p3nguinpowerunits: Your Asterisk is behind a NAT?
23:29.39killownp3nguin, ok
23:30.06powerunitsno..
23:30.23powerunitsp3nguin: no its not
23:30.23mzbthen AsteriskNOW is definitely for you ... personally I prefer building it all on Debian
23:30.33citywokdebian ftw
23:30.53p3nguinpowerunits: Then reconfigure your system for use without NAT.
23:31.08mzbI've written a script that installs from scratch on a Debian (based) distro
23:31.31*** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com)
23:31.34powerunitshumm let me try
23:31.44powerunitsmeah while can you look this form
23:31.46powerunitshttp://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11169370.html
23:31.57killownmzb p3nguin thank you for the recommendation, I am downloading it right now
23:32.01powerunitsthis user has same issue llike my one
23:32.15p3nguinkillown: AsteriskNOW is a really nice OS.
23:32.21citywokmzb, yep, same here.  fully configures a vanilla debian install with asterisk, freepbx, my custom freepbx scripts, all the tools to autodiscover aastra phones
23:32.36killownp3nguin, this is a linux based os?
23:32.48citywokkillown: no, it's windows.
23:33.11citywokokay... mabye it isn't :P
23:33.36powerunitsp3nguin: i have disable nat from sip.conf genral part
23:33.38killown:(
23:33.42powerunitsbut still :(
23:35.17p3nguinkillown: AsteriskNOW is built on CentOS and includes Asterisk, as well as your choice of no GUI, Asterisk GUI, or FreePBX.
23:36.04p3nguinI realize you don't know me, but if I like it, it can't be crap.  I'm very picky with how things work.
23:36.42dr00doooo extensions.conf ...
23:36.47powerunits:'(
23:36.50dr00dim in love
23:37.10killownp3nguin, thanks man
23:37.30citywokdr00d: wait... you didn't know about extensions.conf before? lol
23:37.35p3nguinhmm
23:38.50dr00dno ive been using freepbx to do everything
23:39.13citywokoh, freepbx rewrites extensions.conf when you hit apply
23:39.14p3nguinTime to wake up and smell the configs.
23:39.30p3nguinFreePBX is not an admin's friend, really.
23:39.33dr00dyes i couldnt see the trees for the forest ...
23:39.37p3nguinIt's more like a manager's friend.
23:39.57citywokyea, it's a pain to work with and around
23:39.57*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
23:41.04dr00dyer im a bit new to asterisk still - its pretty interesting
23:42.52*** join/#asterisk wolvenar (~wolv@sip.wolvenar.com)
23:43.54dr00danother question - extensions.con gets overwritten by freepbx - it says in that file not to make any manual changes there - can i configure dial plans from freepbx ?
23:44.31p3nguinThat's kind of the point of FreePBX.
23:45.10dr00dok - so where do i put Dial(Skype/[<originator>@]<destination>) in freepbx ?
23:45.17dr00din the dial rules ?
23:45.28powerunitsp3nguin: thanks for the help
23:45.36powerunitsi find the solution.
23:45.49powerunitsi simply added insecure=port,invite on sip trunk
23:45.56powerunitsand it start working
23:46.03powerunitshttp://forums.whirlpool.net.au/archive/1133054
23:46.13powerunitshere is that link from wher i get help
23:46.15powerunitsthanks
23:46.33wolvenardr00d, put it in extensions_custom.conf
23:46.46dr00dok ive seen that thanks
23:47.08*** join/#asterisk ChannelZ (channelz@burner.com)
23:48.31*** join/#asterisk jkroon (~jkroon@dsl-241-227-29.telkomadsl.co.za)
23:48.39wolvenarif you look into extensions_additional each has an include for a custom .. say like [macro-user-callerid] will follow with include => macro-user-calleris-custom
23:49.35*** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb)
23:49.35*** mode/#asterisk [+o russellb] by ChanServ
23:50.54dr00dthnx wolvenar - so it would be normal to put asterisk commands for skypeforasterisk in extensions_custom.conf or extensions_additional ?
23:51.05wolvenar<PROTECTED>
23:51.21Thornso is there a way to get DIALSTATUS into CDR? it seems that either execution stops after Dial() or CDR is written immediately after Dial() (if it fails)
23:51.24wolvenardr00d, in the custom
23:51.48wolvenaradditional its over written each change to the webgui
23:53.49*** join/#asterisk IsUp (IsUp@unaffiliated/isup)
23:55.31wolvenarI am having trouble with * attempting to dial out multiple times per call , failing the first one.. continuing later attempts . ( pbx-in-a-flash )
23:55.41wolvenaranyone seen this before ?

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