IRC log for #asterisk on 20081207

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00:53.54C4awayis there any analog cards that support the "4-wire E&M" analog lines?
00:54.02C4aways/is/are/
00:54.43C4awayof any of the signalling protocols therein?
00:55.36*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
00:59.11sah-workso i just installed the lastest dl of * and the gui
00:59.21sah-worki log into the gui
00:59.30sah-workand there is nothing to do. there are no pannels.
00:59.33sah-workwhat am i msising
00:59.53[TK]D-Fendersah-work: and then realize the GUI has its own support channel as listed in the channel topic
01:00.19sah-workthanks
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01:18.59sah-workyah. no help there
01:22.38jayteeit's usually like a morgue in the gui support channels, for a patient that refuses to die.
01:25.04jblackisn't the point of a gui to make things so good that assistance is unnecessary? So, by definition, anything not understood should be filed as a bug in the gui's bug tracker.
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01:26.43C4awayinteresting, never noticed that
01:26.51C4awaywho reads topics anyways?
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01:50.07sah-workwith make menuselect, the module embedding is just to compile the module as part of the bin vs loading later correcT?
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02:39.34jksanyone knows where I can download language files for polycom 330 phones?
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02:48.09jayteejks, did you try Polycom's site?
02:48.52jksjaytee, yes, but it didn't have the right language file for the version I'm running
02:49.08jksperhaps I should just update to 3.1.1b and hope that works as well with asterisk as the older ones
02:49.11jayteewhich language and what version?
02:49.30jksI'm running 2.1.1... Danish language
02:55.33jayteejks, their website appears to be weird. it will let me select Dansk as a language but then switches back to English
02:56.43jayteeyou might call your reseller about getting the files  you need or if you can find the language format in a newer version then you can upgrade. I haven't heard of anyone having problems with 3.0 or 3.1 on Asterisk
02:57.38jksguess I better call the reseller... weird that they have to keep the files private
02:59.09jayteejks, they don't keep the english ones private so I don't know why they'd make the dutch ones private. It might be a thing here in the states.
02:59.26jkshehe, I don't want the dutch ones ;-)
03:00.11jayteeI thought you did because you said were running the Danish language
03:01.50jayteejks, what exactly are you looking for?
03:02.09jksDanish like they speak in Denmark
03:02.15jksnot Dutch as they speak in the netherlands
03:03.17jayteeok, I was ignorant of the fact that Danish and Dutch were two languages. I always thought they were the same. My bad. Y'know how us Americans are :-)
03:04.07coppiceA Danish reference means that for the second time today I have pretext to say:
03:04.09coppicejaytee's comments are "a tale told by an idiot, filled with sound and fury, signifying nothing" :-)
03:05.03jayteebet that makes ya feel all smart, smug and superior don't it?
03:09.45coppiceah, but isn't it interesting that I hadn't used that quote for years, yet within just half an hour it fit nicely in two different contexts. there's some higher power at work here, imposing synchronicity upon us.
03:13.37jayteedamn! synchronicity! all these big words are giving me a headache. Think I'll logoff and lie down for a bit.
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03:25.47glazI am trying to establing an IAX trunk between two asterisks, I keep getting Registration Refused, can't figure out what I am doing wrong
03:26.05glazIf I paste my two iax.conf file to pastebin, anyone can take a look at them?
03:28.05glazhttp://rafb.net/p/0kKHN463.html
03:29.12glazSeems OK to me :\ maybe a second look could tell me "hey! you did this mistake on line XX!"
03:29.15glazheh
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03:41.50sah-workanyone else not able to dl the iso for asterisk now
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04:13.50JohnnyBeGood@sah-work try different mirror
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04:37.17simprixDoes the cisco 7921 support sip or skinny ?
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04:49.19glazsimprix: both.
04:51.15SkramXyeah, both
04:51.26SkramXbeen working with skinny on those recently if you have any questions about that
04:56.27glazSCCP on * ? or with cisco call manager?
04:56.34SkramXAsterisk
04:56.45SkramXWork uses CCM however my project is strictly *
04:57.25SkramXIIRC, I havent gotten XMLDefaults.cnf.xml to work the way I want it yet, so I have to use individual SEPXXXX.cnf.xml files (yuck!)
04:57.43SkramXyou?
04:57.51glazI use them with SIP.
04:57.57glazSIPDefault.cnf works fine.
04:58.09SkramXfully compatible with services, etc.?
04:58.13glazyes
04:58.22SkramXim trying to remember why I decided to go with SCCP
04:58.23SkramXheh
04:58.35glazI'm wondering why you when with SCCP :p
04:58.39SkramXpastie SIPDefault?
04:58.41glazs/when/went
04:58.46glazsure, hold on.
04:58.58SkramXwell, at first I thought it'd be easier and go w/o re-loading firmware
04:59.05SkramXbut I ended up upgrading the f/w anyways
04:59.13SkramXk thanks
04:59.32glazI dont have the one I use at the office from here, but I have the one I use at home.
04:59.46SkramXi really want to support 7940, 7980 (Cisco IP Communicator), and 7921s all in one config file
04:59.49glazhttp://rafb.net/p/6h2NhX93.html
04:59.50SkramXk
05:00.06glazI have 7940 and 7960 all over the office
05:00.19SkramXall on SIP and all in one config file?
05:00.47glazwell, will you configure each line on each phone?
05:01.01glazyou can do this and only use SIPDefault.cnf
05:01.05SkramXthats what i've had to be doing
05:01.13SkramXbut i discovered asterisk config templates
05:01.17SkramXneed to play around with those
05:01.17glazor, you create a SIPMA:CA:DD:RE:SS.cnf for each phone
05:01.28SkramXyeah- that's annoying
05:01.38glazcan be yeah.
05:01.53SkramXright now each p hone has a SEPMA:CA...cnf file and then a couple lines in skinny.conf
05:01.56glazor you can use Services to login as an agent
05:02.00glazlike I did for a client
05:02.10SkramXwel
05:02.17SkramXmy whole project is actually about agents
05:02.30SkramXbut i have each phone get a line then the agent logs in on the line
05:02.54glazok
05:03.04glazthere are plenty of ways to do it heh
05:03.24SkramXyeah
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06:33.29carbocalmbbryant: thank you, it works this way
06:33.55bbryantwelcome
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09:04.22mort_gibMorning
09:06.06mort_gibAny recommendations for a nicely priced POE switch??
09:22.39x86mort_gib: Linksys
09:22.58mort_gibLinksys?? -Yeah??
09:23.07x86anyone know of a decent web-based CDR reporting tool?
09:23.16x86preferably with graphs and stuff
09:23.28x86I can't get Areski's to display graphs it seems
09:25.04mort_gibI think that you might have to look at some of the commercial solutions.....
09:27.02SwKcdrtool
09:27.24SwKhttp://www.voip-info.org/wiki/view/CDRTool
09:30.43x86SwK: free?
09:30.55x86w00t! I got call pickup working with BLF....
09:31.11x86works to pick up ringing extensions, wonder about established...
09:33.03SwKyes its free
09:34.36x86looks like it's for OpenSER
09:34.44x86CDRTool is an Open Source solution that provides mediation, accounting and tracing for Call Detail Records enerated by OpenSER by using RADIUS protocol and OpenSER siptrace facility.
09:34.55SwKit does asterisk too
09:34.57x86(from CDRTool's website)
09:35.01SwKkeep reading
09:37.01x86is there a demo of it?
09:37.08x86from the screenshots I don't see any graphs
09:38.56mort_gibx86: How did you get BLF pickups to work?? What handsets
09:41.00x86Linksys 962 with 932 sidecar
09:41.09x86BLF pickups only work with ringing calls it seems
09:41.54x86I can't get the line keys to call regular extensions it seems
09:42.33x86they will show presence on regular extensions, but if I try to call them by hitting the line key, the phone says invalid extension, without even attempting to send the call to Asterisk
09:42.57mort_gibYeah, that's what I get
09:43.12mort_gibSnom 3xx and Asterisk of course
09:43.57mort_gibI can choose to use the buttons as speed dials or as pickup..
09:44.01mort_gibNot both
09:48.39x86it works for both for me
09:48.53x86but only when the line is in ringing state
09:49.11x86also, when I try to page all phones from any linksys phone, I get this: SIP/2.0 487 Request Terminated
09:49.29x86but when I page TO the linksys phones, from say X-Lite, it works fine
09:50.44x86like the Linksys phones are cancelling the inbound call for some reason
09:50.47x86any ideas why?
09:55.56x86also, I have a working call parking line key, but I can not transfer a call to it
09:56.11x86when I hit transfer, it will only let me input an extension it seems, I can't hit a line key?
09:59.25mort_gibHmm, I use Snoms transfer, not call parking
10:00.04x86well I can't get the line keys that have local extensions on them to call that extension
10:00.24x86line keys with local extensions are only working for presence and call pickup (while ringing)
10:00.35mort_gibThere ARE things that some of the old PBX's do better...
10:02.58x86well Cisco Call Manager does a great job of traditional key system emulation
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10:17.23mort_gibNever had a look at Cisco's Call manager... Don't like Cisco much
10:17.48mort_gibHad this discussion with a MS/Cisco administrator about command line
10:18.23mort_gibLame clown claims that (then) was 2007 and we should not have to use the command line anymore...
10:19.28coppicethey are usually the kind who believe the same about brains
10:19.37mort_gib-Saying this while he is using telnet to configure our border gateway! -So what are you doing there?? That looks like you are using a commandline over an unprotected connection!
10:19.37Nuggettelnet is eeeeeeevil!
10:19.47mort_gibYep...
10:23.40mort_gibAnyways, Sunday blues see you all later...
10:31.19x86coppice: hey, ever use Linksys phones? specifically a 962?
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10:59.30respectingplease what' i want an asterisk version for windows ?
11:00.05respectingi have a company equiped with windows Pc's and i want to use asterisk but it works only on linux? is there any version for windows
11:02.01orly_owlhttp://www.asteriskwin32.com/
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11:03.03respectingwawwwwwwwwwwwwwwwww
11:03.08respectingthat's wonderfull
11:03.15respectingthank you
11:03.15respectingMr orly_owl
11:03.25respectingyou have just saved my ass
11:13.12tzafrir_laptoprespecting, hmm... I'm not really sure how well that works
11:14.14tzafrir_laptopIIRC it's a dated version
11:15.58C4awaywow, for some odd reason the concept of running asterisk on a windows server NEVER crossed my mind
11:16.50C4awaykinda like running a porsche engine in a gremlin
11:17.31C4awaytechnically, with enough work I suppose you could get it functional, but it's just an odd concept
11:22.12C4awayin the case of asterisk on windows, you get to pay more for licensed operating sytem to run an app that was not designed to run in that environment, probably get buggy and questionable results ... when there is a free operating system the software was designed to run on
11:27.05respectingwait a second i think that's like asterisk for linux
11:27.11respectingthey did the same job
11:27.17respectingi was on their website
11:27.26respectingand they did the same as in linux
11:27.47tzafrir_laptoprespecting, give it a shot. Tell us what it was like
11:28.05tzafrir_laptopThough I'm not sure how well supported it actually is
11:28.38tzafrir_laptopGenerally I would recommend you to set up a separate Linux system
11:28.50respectingthe * for windows works as a gatekeeper(PABX) and it has the ability to work as a gateway
11:29.12respectingbetween RTC(ISDN in english) and voip network
11:29.45tzafrir_laptopWhat ISDN adapter do you have?
11:31.00respectingzapatel(i'm gona make my intership about * :the company manager tell me the whole idea that he want to deploy * on windows)
11:31.44tzafrir_laptopI have no idea if they actually ported zaptel to windows. It takes a huge ammount of work. And I have heard of no one who did that
11:32.20tzafrir_laptopMaybe they have drivers for some specific cards
11:32.39tzafrir_laptopGenerally test it yourself before even mentoining it to anyone else
11:32.48respectingyeah the problem that suppose a bank that have only windows laptops and pc's how can we make a voip server without a windows version of *?
11:33.39tzafrir_laptoppeople's laptops don't need to have asterisk servers. Do you want to put osftware voip phones on them?
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11:34.09respectingno here's the problem.in a bank there's only windows
11:34.19respectinghow can u make voip server their?
11:34.39respectingyou can not say to them please buy a new pc and install linux then install *
11:34.49respectingyou must have a windows version of *
11:35.08tzafrir_laptoprespecting, the PBX should be on a dedicated server in most setups
11:35.29respectingyes and that server must be linux is that correct?
11:35.44tzafrir_laptopFor Asterisk: yes (if you value your time)
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11:56.23dfasWhat companies provide managed asterisk hosting?
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13:02.52GNUtoohello, I had bad luck with siproxd and ekiga...so could asterisk be used as sip proxy? if yes is there any infos on that: http://it.slashdot.org/it/08/12/06/1914242.shtml ?
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13:10.58GNUtooby the way does I  still need dahdi  in order to make meetme work?
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13:22.19Ko_deZHi. I have a problem with my incoming DID (SIP). The people calling me are not getting a ring tone, just silence. I have been trying to figure this out, and the only thing I find is that I should add a 'r' option to the dial command for the incoming connection, but it does not help me much.
13:22.44Ko_deZThere is still silence at the calling end.
13:22.52Ko_deZIf I pick up, everything is OK though.
13:24.23Ko_deZmy extensions.conf is here: http://pastebin.com/d55ebe9e1
13:24.27tzafrir_laptopGNUtoo, yes
13:24.45GNUtootzafrir_laptop, yes for dahdi?
13:25.01tzafrir_laptopYes
13:25.08tzafrir_laptopWhat version of Asterisk do you use?
13:25.25GNUtooi'm actually compiling the 1.6.0.2
13:25.31GNUtoos/i/I/
13:29.03Ko_deZhaha, cool bot.
13:29.14Ko_deZs/cool/awsome/
13:29.26Ko_deZI like it!
13:34.37orly_owls/like/don't like/
13:38.09Ko_deZfail!
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13:38.34orly_owlWorth a try.
13:38.37Ko_deZmy favorite: http://failblog.org/2008/10/07/cow-curiosity-fail/
13:38.43Ko_deZsure.
13:38.54Ko_deZI would have done that too =)
13:39.26Ko_deZNeed to get some of those fail/win stamps they are selling.
13:40.49orly_owlI want a kb with no OS specific keys. >_>
13:41.05Ko_deZI figured out my "silence" problem when people call me btw. I removed the Answer before using Dial to redirect to the internal numbers.
13:41.56orly_owlI have to figure out * first.
13:42.41Ko_deZorly_owl: http://www.doobybrain.com/2007/10/29/computer-keyboard-calculator/ <-- probably not what you where looking for though...
13:43.27orly_owlNo, but that's a really nice looking calculator.
13:44.03Ko_deZhttp://www.kingmedia.com.au/product_info.php?manufacturers_id=16&products_id=79&osCsid=ae1227a8b4fff8aa1185dfcd86a7118f <-- even smaller
13:44.13orly_owlNow show me the flipped version for use with the left hand.
13:44.14Ko_deZit sure is
13:44.23Ko_deZhaha.
13:45.22orly_owlI saw an apple alu kb recently. didnt like the OS keys and some other key labels, but DAMN is it a nice looking keyboard. felt well built too
13:46.14orly_owla happy hacking kb in that style would be nice
13:46.25Ko_deZHumm. Can you find a link?
13:47.01orly_owlhttp://en.wikipedia.org/wiki/Image:Apple_wired_thin_keyboard-2007-08-11.jpg
13:47.13orly_owldodgy photo though
13:47.28orly_owlhttp://www.apple.com/keyboard/
13:48.33orly_owlhttp://ln-s.net/2Y+x
13:49.18Ko_deZoh...
13:49.25Ko_deZthat _is_ nice
13:49.50orly_owl=D
13:50.07Ko_deZhow are the buttons?
13:50.13orly_owlfelt ok
13:50.29orly_owli didnt use it though.
13:50.36orly_owlwasnt plugged in
13:50.41Ko_deZright.
13:50.50orly_owlmy current kb http://notzzap.zzzzz.ws/model-a-kb.jpg
13:51.02Ko_deZkeytronic ftw!
13:51.09orly_owlkeytronic?
13:51.29Ko_deZhttp://www.testipenkki.net/atk/artikkelit/keytr/kuvat/kta2001-022.jpg
13:51.41Ko_deZbest there ever was...
13:51.59orly_owlbleh. OS keys
13:52.16Ko_deZah, yes, the newer ones had those. Mine did not.
13:52.22Ko_deZfrom the early 90's
13:52.30orly_owlah ok
13:53.14orly_owli like the gap betwen alt+ctrl
13:54.33orly_owlwhy are ip phones so expensive? POTS phones can be had for $10
13:57.56coppiceA simple POTS phone costs about $1 to make. A fancy caller ID phone with a nice LCD display costs about $8. An entry level IP phone with, and simple LCD display, costs in the high 20s.
13:57.59Ko_deZorly_owl: http://communication.howstuffworks.com/telephone.htm. POTS phones a _very_ basic.
13:58.12orly_owlthat's true
13:58.54Ko_deZI mean, a speaker, a mic, and a hook switch, and you are good to go. Does not get any cheaper than that.
13:59.03orly_owlso if an entry level ip phone costs high 20s, why are they selling for $100 minimum?
13:59.40coppiceyou can get entry level IP phones for about $40 in reasonable numbers
14:00.11orly_owlah ok
14:00.23orly_owlreasonable would be 50 at least?
14:00.45coppiceyeah. I'm not talking about the 1M price break :-)
14:01.58Ko_deZorly_owl: you can get a voip adapter for far less than 100$.
14:17.40orly_owlbut if a voip adapter is over $40, might as well get an ip phone
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16:34.50invalidrecordhi guys is AMI solid enough to use for billing info or should i do that from a custom application for stability?
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16:45.25x86good morning
16:46.00invalidrecordmorning
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16:52.55jayteeinvalidrecord, I'd recommend using CDR data from either mysql or postGRE sql. you can store the data in a local sql database or use the unixODBC connector for writing to a sql database on another system.
16:54.32x86jaytee: cdr_addon_mysql.so doesn't have to use a local database, wtf are you talking about
16:54.42x86jaytee: niether postgres
16:55.09invalidrecordok ill do that may be simpler
16:55.14invalidrecordthanks
16:55.27jayteex86, so you don't need the unixODBC connector? I store locally so I haven't tried setting an external database. good to know, thanks
16:55.47invalidrecordyou still need odbc but db dont hve to be local
16:56.16invalidrecordyou guys used realtime?
16:56.22jayteeno
16:56.31invalidrecorddamn
17:03.41[TK]D-FenderWhy are we comparion a DATABASE with a LIVE PBX MONITORING PROTOCOL?
17:04.06jayteewe weren't or at least I wasn't
17:04.45jayteeand I'm assuming comparion is Quebecian for comparing :-)
17:05.51[TK]D-Fenderjaytee: No its "OMG I only got up a few minutes ago, have a headache and am not entirely there, lets just hybrid-conjugate everything!"
17:06.06jayteelol
17:06.12jayteego get some coffee
17:07.14jayteex86, so you're sayin if I want to write out my CDR data to an external mysql database I don't need the unixODBC connector? all of that is built into the cdr_addon_msyql.so?
17:08.03x86jaytee: right
17:08.32x86jaytee: hence the "hostname" configuration option in cdr_mysql.conf (for example)
17:09.13x86the ODBC stuff is an abstraction layer to allow you to connect to any database that your ODBC driver supports
17:09.15jayteex86, must have missed that in the docs. I have it running local on my * box but eventually I want to move the data to another system because I'll have 2 * boxes, one for primary and one for failover and I'll want the CDR collection to be "seamless"
17:10.14x86[TK]D-Fender: ODBC is not a "live pbx monitoring protocol", it's Open DataBase Connctivity
17:10.21jayteex86, aha! so I'd only need the unixODBC connector if I was going to write out the data to another database like MS SQL Server 2005 or something else non-mysql.
17:10.58jayteex86, I think he knows that. I think he was referring to AMI vs sql storage of CDR data
17:11.22x86jaytee: Asterisk comes with a driver for pgsql, and the addons package has a driver for mysql... anything else you'll need the ODBC driver for (but ODBC also works with MySQL and Postgres if you'd want that)
17:11.32x86jaytee: ah
17:11.51[TK]D-Fenderx86: I was referring to invalidrecord's mention of AMI <-
17:11.56x86[TK]D-Fender: gotcha
17:12.13[TK]D-Fenderx86: ..... wanna join me for coffee? ;)
17:12.37jayteex86, I knew that Postgres was included but I used the mysql addon. I just didn't know I could do it direct from the addon without using ODBC. Glad you pointed that out to me, it'll save alot of work.
17:12.38x86I wish i had some... used the last of it last night (this morning actually, around 2am)
17:12.57x86told the wife to know her role and go get some, but she snottishly declined :(
17:13.30jayteepours x86 and [TK]D-Fender each a cup of his freshly roasted Kenya AA Gjanika roasted last night. "Cream and sugar?"
17:14.02x86jaytee: the only reason mysql's not included in the default distro is something about licensing... i think Digium want's everything distributed in the core of Asterisk to be truely free, while the mysql code is not truely free, or something like that
17:14.28x86skip the cream and sugar, I use hot cocoa mix instead :)
17:14.36x86yum yum dim sum
17:15.12jayteex86, cool! kinda makes it a mocha
17:15.35jayteeI sometimes add a dollop of Hersheys syrup for that
17:15.57x86ok, is there a way I can use PickupChan with a dynamic channel name? such as SIP/7002-08faa7a0 (where 7002 is my extension)?
17:16.29x86jaytee: the cocoa makes it chocolatey, the marshmallows make it sweet :)
17:16.32x86works great
17:18.29jayteex86, I sometimes mix beans from different crops when I roast but usually I roast and drink just the single varietal bean so as to pick up all the nuances. I'm a coffee snob. Won't drink Starbucks because they overroast all their stuff on the dark side.
17:19.02x86hmm, perhaps I can write an AGI to do get the dynamic part of the channel nam
17:19.04x86name*
17:19.31invalidrecordok i have a pbx which supports multiple companies i am driving extensions from a realtime config each company has one  context does this sound right
17:20.01[TK]D-FenderinvEach company should probably have multiple contexts each
17:20.38[TK]D-Fenderinvalidrecord: and that's just for minimal separation of class
17:20.54invalidrecordwhat dialout long distance dial out local etc etc?
17:21.29[TK]D-Fenderinvalidrecord: etc * 50
17:21.47[TK]D-FenderinvIVR's, etc
17:21.52invalidrecordi was going to have some global contexts for that which i would include the companies in so dialout-local Xcomp ycomp zcomp
17:22.21invalidrecordno ivr's needed at this time
17:22.25[TK]D-Fenderinvalidrecord: If you even have to ask about that well... you might want to reconsider your role in the project ;)
17:22.52jayteeand if you want to split out the billing you probably want separate outbound contexts for each company for CDR data to segregate the billing and make it easier to run queries against the DB for a single customer.
17:22.54invalidrecordleartning on my feet im a rails dev
17:23.10invalidrecordonly really palyed with avaya b4
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17:23.39jayteenever had a chance to work on Avaya systems. I'm an old Nortel retread
17:23.59invalidrecordipoffice was ok but fking licence nightmare
17:24.20invalidrecordsplicecom was good
17:24.25jayteeNortel's licensing scheme is just as bad or probably worse
17:24.46invalidrecordwe replaced the avayas with that in the end as bos had no confidence in asterisk
17:25.10invalidrecordhe wanted to pay thousands gave him a warm cosey feeling
17:25.42jayteewith Nortel systems you need licenses for the TN's (terminal numbers=physical analog or digital ports) and also licenses for the DN's (directory numbers or "extensions")
17:28.12[TK]D-Fenderinvalidrecord: Bosses do things like that
17:28.56invalidrecordyeah true he also hadf played with slackware so thought he was a guru, I used to have to rebuild ldap or other systems almost every monday tosser!
17:29.24invalidrecordinsisted he had root even when i wouldnt trust him with windows but he signed my cheques what do you do
17:29.52jayteelook for another job
17:30.00invalidrecordheheh exactly what i did
17:30.21invalidrecorddoubled my money halfed my stress overnight
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17:30.53jayteeif only the economy wasn't in the shitter I'd be looking more in earnest than I am.
17:32.10beekHi jaytee -- have a moment for a PRI question?
17:32.10invalidrecordi was so lucky i just switched job and it took me 24hrs and i got a nice slice of equity and a good package, can believe i was that lucky!
17:32.25invalidrecordit amazes me isdn is still about
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17:32.32andreafhi *
17:33.27beekjaytee -- I took your advice and configured a test loopback for my PRI problem -- diagrammed here:  http://www.pastebin.ca/1278904
17:34.37beekI used port 4 instead of the telco and have the dialplan simply answering, then repetively playing the "congratulations" message.
17:35.27beekI have twenty-three phones dialed in, thus am using all 23 channels on spans 1,2 & 4.   It has been happily running since last night at 7:00pm.  With the office phones all playing this, you ought to hear the cacophony.
17:35.35invalidrecordis it me or is the orilley asterisk book a lot more superficial than mpst orilley books therte seems verry little in in rerally
17:36.18beekMy question is:   given that I have had zero errors since this began, can I assume that the problem is definitely the Telco, despite their protestations to the contrary?
17:36.23[TK]D-Fenderinvalidrecord: I find it a little to "open theory" without enough practical explanation and implementation
17:36.31[TK]D-Fenderinvalidrecord: Fails to drive home some important points.
17:37.09invalidrecord[TK]D-Fender: where should i be looking this has fallen on me and I have to work it out but this ook dosent have what i need really
17:37.33[TK]D-FenderinvaWhat, just for billing?
17:37.33jayteebeek, yes barring the one point that it still could be problems with the CSU and the only way to determine that is to replace it with a known good one.
17:38.13beekjaytee: I used the CSU that is currently hooked up.  I just ran a cross-over cable from span4 to the "net" connection on the CSU I was connected to the telco with.
17:38.26jayteeinvalidrecord, check some of the items on this page, it may be of help: http://www.voip-info.org/wiki/view/Asterisk+billing
17:38.35beekI configured span 4 to use _net signalling.
17:39.03invalidrecordwell i work for a virtual office company and we provide an voip network handeling call answering for multiple companies and the associated billing no outgoing calls except the odd forwarding of an extension to a pre difined number
17:39.58beekjaytee: Thanks for the tip.   Now I go back to do battle with the Telco tomorrow.
17:40.04jayteeso you're currently using the same CSU as part of the loopback? that's the part of the diagram that threw me off but now it makes sense. If you're not getting errors now I'd document it thoroughly and then call your telco and somewhere in the conversation mention the word "lawyers". :-)
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17:41.47beekjaytee: That's what I'm going to do.   I had a gal from the telco here on Thursday and she showed me on her T-Berd that the errors were coming from "my side."  I showed her the errors on the CSU that pointed to the "Net" side.   Hooking things up this way at least allows me to test all of the cabling and equipment that's my property.
17:43.39beekOh, and Allison just counted the number of times she's repeated that prompt on my phone:   1,828.
17:43.42jayteebeek, if she says her T-Berd was showing the errors coming from your side, which point did she say the errors were introduced? if they're sending timing and you're getting frame slips but that goes away when you loopback how could it be your end? especially if you changed nothing in the configuration of the cards.
17:44.41beekjaytee: All she showed me were the error counts on the equipment in their office.   The "outgoing" line showed a couple but the "incoming" side showed a metric shitload.
17:45.05andreafhi, i'm looking for information about to routing all outcoming call to a SIP provider with asterisk. I'm looking for tips and link on google (i'm googling also with not so good success)
17:45.52andreafgoogling on voip-info.org
17:46.05jayteeand yet if you eliminate the span to them, it all works perfectly? what does that tell you?
17:47.17beekjaytee: exactly.     I wanted to ensure that my test harness was valid, which you confirmed, so now I have the amunition to go back into battle.
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17:48.05jayteebeek, good luck. getting most telcos to cooperate takes patience and a certain amount of luck. They always want to blame the CPE side for faults.
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17:48.40beekI assumed that configuring span4 as a master, using the internal clock, then inserting that into where the telco plugs in should give me a conclusive and convincing test.
17:49.08jayteebeek, it should
17:49.55beekjaytee: well, thanks again for all of your help.   I'm a T1/PRI newbie and the telco can easily blow smoke up my ass.  It's nice to have a resource such as yourself who has done this before to help debunk what they tell me.
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17:51.44jayteebeek, you can also enable debug info in the logs and turn on pri debug or pri intense debug at the CLI to capture the info to the log files but the log files will get BIG fast so  you don't want to leave it like that for too long.
17:52.22beekjaytee: I'll do that.  I have a terabyte available for logging and if that's what it takes to capture the problem, I'm willing to wallow through it.
17:52.25etfonhomey_jaytee, that should be a capital, bolded, underlined ALWAYS.  It's SOP for them.
17:52.58jayteebeek, I'm still something of an * newbie myself. I've had over ten years working with T1's but not constant enough to consider myself an "expert" but enough to get by on.
17:53.01[TK]D-Fenderandreaf: "routing all outcoming call to a SIP provider with asterisk" ?  Where are the calls coming from?  And then going to?  What do you have currently?
17:53.42jayteeetfonhomey_, yeah, I'll have to remember that :-)
17:53.45etfonhomey_jaytee, although I actually got some halfway decent service from the telco the other day with a telecommuter's DSL connection
17:54.13beekjaytee: Thanks again and I'll let you know how I make out.
17:54.54jayteeetfonhomey_, to be honest I haven't really run into major headaches with my current provider, Time Warner Telecom. Their people at their NOC have actually been very helpful and didn't automatically assume it was a problem at my end when I had issues.
17:55.01etfonhomey_jaytee, the connection was:  DSLmodem --> ASA --> PC and the user kept losing their Internet connection.  The telco guy after coming out and telling us it was my Cisco ASA, left us an extra DSL modem.
17:55.15jayteebeek, your welcome. good luck wrestling with them.
17:55.29jayteeetfonhomey_, lol
17:55.56etfonhomey_jaytee, I configured the second DSL modem for the default (no bridge mode) so that when the user lost Internet they could undock their computer and move the phone cable to the 2nd DSL modem and plug directly in to test the connection.
17:56.13etfonhomey_jaytee, Doing so proved it wasn't the ASA.
17:56.15andreafTKD-Fender are originate inside the asterisk (with the sample.call) and have to go to a "real" number tought a SIP provider
17:56.34andreafsorry outgoing not outcoming, my fault
17:56.39jayteeetfonhomey_, don't ya love it when they actually hand you the ammo to shoot their legs out?
17:57.17etfonhomey_jaytee, I hate PPPoE, but for this user's location, DSL is faster than cable Internet.  But anyway, back on topic...
17:57.22[TK]D-FenderandreSo you want * ot Originate a call like an automated message system?
17:57.31andreaftkd-fender: something like a "default route" in ip network. All call generate inside the asterisk are routed to a SIP provider.
17:58.00andreafyes: i can originate the call (with originate or the sample.call script) but I have to route it to a SIP proxy (external) where I have an accoutn
17:58.21[TK]D-Fenderandreaf: What would these calls be doing?
17:59.01andreafTK-DFender: polls, adv, automatic outbound callcenter something like that.
17:59.39andreaftk-dfender: but the problem is that i can't configure my asterisk to "peer" to another sip proxy, apparently failed the authentication
18:00.19[TK]D-FenderandreOk, well you should go install *, install a soft-phone and sit down with the BOOK and start learning how the dialplan works.  This is the hard part.
18:00.55[TK]D-Fenderandreaf: Ok, failing to authenticate is another matter.
18:01.22andreafyep, i'm full of docs, i can originate the dial (more or less) with "originate" cli command. but apparently the authentication with the SIP proxy fail (and the call is not routed)
18:01.30[TK]D-Fenderandreaf: pastebin your SIP.conf masking only passwords and removing all comments, and include the CLI output of your failed attempt at verbose 10, SIP DEBUG enabled.
18:01.32[TK]D-Fender~pb
18:01.32jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/
18:01.34[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^
18:01.56andreafok,
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18:03.50andreaftk-d fender: it works, sorry I use the "magic"  parameter in sip.conf ";auth=mark:topsecret@digium.com"
18:03.55andreafyeppa!
18:04.41andreafwith my username and password on the sip proxy esternal server. quite a woodo I think, but my mobile handset rings :)
18:05.46andreaftkd-fender with in the CLI originate SIP/+MYNUMBER@PROVIDER application Dial(SIP/+MYNUMBER)
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18:16.22x86I'm trying to do 'core show channels concise' from an AGI, but i'm not having any luck... keeps failing saying "unable to find application 'core'"
18:16.39x86any ideas what I can do to get the current channel stats?
18:17.10Mark17hello, has someone time to help me with configuring asterisk to use a bluetooth device and and incoming calls from the bluetooth device should go to a certain sip account OR trunk and incoming calls from the SIP account OR trunk should go to the bluetooth device
18:17.24Mark17if needed i am willing to pay for the help with paypal
18:18.34[TK]D-Fenderx86: And you're failing to show us exactly HOW you're calling that.
18:19.14seanbrightAND cross-posting to -dev
18:19.28[TK]D-Fenderto DEV?   Good greif
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18:19.33jayteetsk tsk
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18:19.53invalidrecord[TK]D-Fender: can i ask you a q bassed on the orilley book do you have a copy? trying to get my head round contexts
18:20.01seanbright~ask
18:20.01jbotask is probably Questions 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.
18:20.04[TK]D-Fender^^
18:21.47invalidrecordok well on page 133 they have a dialplan with 2 contexts defined the employees one defines 101 and 102 extensions, my question is couldnt the same ability to dial 101 and 102 in the emplyes context be acvhived with just the internal context and dialing 1 & 2 i dont see how they are that diffrent
18:22.34invalidrecordmust be missing something
18:23.30[TK]D-Fenderinvalidrecord: this is showing how to make an IVR.
18:24.04[TK]D-Fenderinvalidrecord: When you dial a company, do you expect to dial somehting like "1" to reach the sales dept?  Or expect to have to puch 10 digits?
18:24.21seanbrightx86: get_variable
18:24.27x86seanbright: ok
18:24.39[TK]D-Fenderinvalidrecord: Yes they lead to the same person, but they may have a more direct & privately known number so you can JOE who just happens to be in the SALES dept
18:24.57x86seanbright: [Dec  7 12:24:56] ERROR[27187]: pbx.c:2772 ast_func_read: Function CHANNELS not registered
18:25.14seanbrightx86: pastebin the actual script
18:25.15invalidrecordcool ok i was just checking i wasnt midssing some nuance
18:25.18[TK]D-FenderinvSo somebody calling your company only has to press "1" to talk to joe in sales and not have to know his longer "direct" like number
18:25.47invalidrecordso they are practicaly aliases which also define routing
18:26.27x86seanbright: http://pastebin.ca/1278932
18:26.35[TK]D-Fenderinvalidrecord: Not really... the don't do the exact same things... just SOME things in common.
18:26.35Corydon76-digx86: are you using 1.6 or 1.4?
18:26.36seanbrightx86: and actually, the error is pretty clear.  you don't have the CHANNELS function loaded.
18:27.10invalidrecordbut calling 101 would be effectivley the same as just dilaing 1 in this example
18:27.35x86Corydon76-dig: 1.6.0.1
18:27.47[TK]D-Fenderinvalidrecord: No.... 101 dials them INDEFINITELY.  the other will only ring for 10 SECONDS.  And then even play a MESSAGE.  Is this the SAME?  I don't think so.
18:28.06[TK]D-Fenderx86: Where did you invent that parameter-less function from?
18:28.07x86seanbright: does it ship with astrisk 1.6.0.1?
18:28.10invalidrecordok i wasnt specific enough, i have it thznks@!
18:28.20x86[TK]D-Fender: from Corydon76-dig
18:28.24[TK]D-Fenderx86: this is not something that AGI does.. this is for AMI <-
18:28.34x86[TK]D-Fender: not according to Corydon76-dig
18:28.51[TK]D-Fenderx86: Continuing on your "how do I dump my channels
18:29.10[TK]D-Fenderx86: AGI is not a means of dumping channels.  AGI controls dialplan apps.
18:29.14invalidrecordgood news is for now i only need one context per company :-) no ivr or anything clever
18:29.30[TK]D-Fenderx86: AGI is a way of ening up in an outside script that you can go make AMI calls through
18:29.36x86[TK]D-Fender: cory said in 1.6.0.1 I can use GET VARIABLE CHANNELS() to get a list of active channels
18:29.41Corydon76-digx86: it's only in 1.6.1
18:29.50x86oh, weak!
18:30.09x861.6.1 is stable?
18:30.15invalidrecordhow far from stable is 1.6 if im developing a new app should i target 1.6?
18:30.15seanbrightnot released
18:30.16Corydon76-digYes
18:30.18[TK]D-Fenderinvalidrecord: Every device can dial the exact same things?  No special inbound processing?
18:30.41x86Corydon76-dig: no way I can compile just that module to handle CHANNELS() and plug it into 1.6.0.1?
18:30.54invalidrecordno no special processing we have multiple companies that have one main extension that incomming is routed to and the others can only call eachother
18:31.11Corydon76-digx86: It might or might not work
18:31.18Corydon76-digI have not tried it.
18:31.20x86:(
18:31.37[TK]D-Fenderx86: Just use AMI and get it DONE already...
18:31.49x86well I would use AMI, but Asterisk::AMI seems to have trouble getting full responses back from 1.6
18:32.04[TK]D-Fenderx86: You're wasting time on stuff they will get released "eventiually" and stable... who know how much firther past that...
18:32.08Corydon76-digI don't see anything in the module that depends upon 1.6.1 features
18:32.13invalidrecordno outgoing or anything all we want is to have one incomming number that goes to their first extension and the ability for that to transfer to other extenbsiuons in their context
18:32.26invalidrecordextensions
18:32.36[TK]D-Fenderinvalidrecord: Guess you could do that cleanly enough in 1 context.
18:32.40x86Corydon76-dig: not sure, but it doesn't work...
18:32.43[TK]D-Fenderinvalidrecord: Still kind of crappy
18:32.54x86Corydon76-dig: Asterisk::AMI is a bit crappy anyway
18:33.04invalidrecord[TK]D-Fender: how would you attack it then
18:33.15x86Corydon76-dig: also, it's event-based...
18:33.18Corydon76-digx86: you asked for a way to do it
18:33.22x86Corydon76-dig: I'd like to use this from an AGI
18:33.40x86Asterisk::AMI has to run an eventloop, which sucks
18:33.40Corydon76-digCHANNELS() is the only way from AGI
18:33.45x86hmm
18:33.58x86is there a tarball of 1.6.1?
18:34.03[TK]D-Fenderinvalidrecord: Well if every device is just as functional as the rest, no IVR's, or anything at all more that what you've described then sure.
18:34.11Corydon76-digSure, there's a beta tarball
18:34.17x86w00t
18:34.18x86url me?
18:34.27Corydon76-digdownloads.digium.com
18:34.29[TK]D-Fenderx86: Eventloop?  No need to sit for the response.
18:34.30x86hehe
18:34.31invalidrecordok cool all we do is provide an answering service for time being
18:34.41Corydon76-digpub/telephony/asterisk/releases/
18:34.50x86[TK]D-Fender: I'm relying on the response, why wouldn't I want to wait for it?
18:35.12[TK]D-Fenderx86: not as an event.  What are you doing exactly to try this currentky?
18:36.36x86[TK]D-Fender: Asterisk::Manager's sendcommand()
18:37.24[TK]D-Fenderx86: AMI "COMMAND"?
18:37.28*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
18:37.38x86[TK]D-Fender: looks like, yes
18:37.47[TK]D-Fenderx86: Works really straightforward for my use of dumping queues.
18:37.57[TK]D-Fenderx86: Don't come back as an "event"
18:38.24x86[TK]D-Fender: from the source code of the module, looks like I can capture the output to a hash by calling it like: %myhash = $ami->sendcommand('foo')
18:38.42x86my %myhash only ends up with some status line and "channels will follow"
18:38.58x86think it's a bug with Asterisk::Manager, seems to be trauncating output
18:39.31[TK]D-Fenderx86: Don't forget EOL stuff in there
18:39.42[TK]D-Fenderx86: I parse the string out the hard way
18:40.31*** join/#asterisk oej (n=olle@ns.webway.se)
18:43.05x86[TK]D-Fender: opened the output in a text editor, there is nothing extra in the file :(
18:43.20x86i'll upgrade to 1.6.1-beta3, no biggie
18:44.48[TK]D-Fenderx86: Congratulations on jumping from 1.2 right though "bleeding edge"
18:47.20x86well I've been using 1.4 in an outbound call center for some time
18:47.50x86ugh, 1.6.1-beta3 doesn't compile
18:48.49x86http://pastebin.ca/1278949
18:52.06x86gah, all this is wasting my time, it's not possible to do what I want anyway
18:52.18x86I want to pickup a channel that's already answered, using PickupChan
18:52.57x86I pinned a call up, got the channel name, put an extension in my dialplan to pickup that channel name, and it still fails with "No target channel found for SIP/7001-08f5ff60."
18:53.06*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
18:54.22x86I want to barge in on a bridged call, with full two-way audio (which rules out ExtenSpy / ChanSpy)
19:03.50[TK]D-Fenderx86: BRIDGE <-
19:07.27x86[TK]D-Fender: does that require me to know the full channel name? like SIP/7000-abcdef12? or can I use it like SIP/7000?
19:08.40[TK]D-FenderSIP/7000 is not a channel
19:09.33*** join/#asterisk j-tek (n=NTech@86.105.21.220)
19:10.22x86so I run into the same problem that I must somehow extract the channel name
19:12.42[TK]D-Fenderx86: Why is it we never see exactly what's going on?
19:14.05invalidrecordhow do i specify a prefix for a context so say a global operator can dial 123 exten 1000 or 124 exten 1000
19:15.21[TK]D-Fenderinvalidrecord: that makes no sense
19:15.56[TK]D-Fenderinvalidrecord: I think I see what you're getting at....
19:16.18invalidrecordim prob using wrong terms :-S
19:16.30[TK]D-Fenderinvalidrecord: You are using numbered contexts?
19:16.50invalidrecordwas thinking it would be best as we have deskphones
19:16.51[TK]D-Fenderinvalidrecord: like [123] for company "123"?
19:16.56invalidrecordyeah
19:17.18invalidrecordso our main operator can dial each company by prefix then exten number
19:17.48[TK]D-Fenderinvalidrecord: then something like "exten => _XXXXXX,1,Goto(${EXTEN:0:3},${EXTEN:3},1)
19:18.29invalidrecordok ill go work out how that matches and implement thanks
19:20.35hardwirepokes Carlos_PHX
19:21.22*** part/#asterisk j-tek (n=NTech@86.105.21.220)
19:21.35invalidrecord[TK]D-Fender: is numbered contexts the wasy to go?
19:21.38invalidrecordway
19:21.38*** join/#asterisk ShaunWing (n=chatzill@dsl-243-95-10.telkomadsl.co.za)
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19:22.48[TK]D-Fenderinvalidrecord: You need to sit down and think out your deployment.  You should alrady be loking at the ups & downs of each.  Go try stuff.
19:23.16farkusI'm having a problem where my call audio sounds fine, but the monitor recordings have short periods of silence ( afew milliseconds) occasionally that makes the recording sound jumpy
19:23.19ShaunWingsay, can't get Asterisk 1.4 to redirect the rtp to my vsp. redirect = yes in sip.conf doesn't do the trick. Any ideas?
19:23.43invalidrecordyeah im just asking as i think it through should prob test more ask less. Thanks man!
19:26.21*** part/#asterisk ngeloa (n=ang@81-208-36-86.ip.fastwebnet.it)
19:30.43x86[TK]D-Fender: because you suck at asking for what you want to see heh
19:32.04x86I'm having a terrible time with a Linksys SPA962 and a Linksys SPA942 with paging... I can page FROM X-lite to both Linksys phones and it works like a champ, but if I page FROM either Linksys phone, the call is cancelled almost immediately
19:32.36x86any idea why that might be?
19:33.04*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
19:33.44jayteex86, because you suck at configuring Linksys phones, heh
19:33.59x86jaytee: tis why I'm asking for help? :)
19:34.11jayteex86, was just kiddin ya man!
19:34.16x86hehe
19:34.30x86jaytee: have you ever done paging with linksys phones?
19:34.55jayteex86, if ya want a real hemorrhoid try the SPA-3102 ATA. Works like a champ when you get it working but configuring it is a pain in the ass
19:35.05jayteex86, no I use Polycom phones
19:35.44x86wish I could say the same... boss bought these POS linksys phones without first asking me
19:36.12x86of course, the salesman always knows more than the technical guy implementing the system...
19:36.15[TK]D-Fenderx86: Can't see your code, can't see the resulting variables to see if you extra crap chars.  Don't see channel dumps to back up the one's you're trying to grab
19:36.21[TK]D-Fender.x8you suck at showing ANYTHING.
19:36.43x86sure do, because I have no idea what you need until you ask
19:37.00[TK]D-Fenderx86: Your code doesn't work?  SHOW US THE CODE
19:37.10x86here, I'll do a sip debug on the peer (linksys 962) trying to page
19:37.12[TK]D-Fenderx86: is this the right channel?  DUMP THEM
19:37.19jayteedon't know what the equivalent setting is on Linksys but on a Polycom it's ring-answer. I modified that in the Polycom configs to use a particular ring type and a duration of 1200 ms and then use SipAddHeader to send Alert-Info: Ring Answer to the phone when dialing, then I've got it setup in the dialplan so any extension you dial you just add an 8 in front of it for paging, intercom.
19:38.23*** join/#asterisk viliar (n=viliar@viliar.dialup.corbina.ru)
19:40.38etfonhomey_jaytee, like this:  http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
19:40.40etfonhomey_?
19:40.41x86[TK]D-Fender: http://pastebin.ca/1278984
19:41.45viliarhello all
19:41.50jayteeetfonhomey_, yep in fact I used that as a rough guide to doing it
19:42.21[TK]D-Fenderx86: This isn't the channel dumping issue...
19:42.24jayteeviliar, hello
19:42.29x86[TK]D-Fender: 611 is my paging extension, 7000 is my Linksys 962, 7001 is my Linksys 942, 7002 is X-Lite
19:42.38viliari have a little q.
19:42.45x86[TK]D-Fender: it's a SIP debug of 7000 calling the paging extension
19:42.46[TK]D-Fenderx86: and WAY too much excess debug crap
19:42.50viliarprobably someone can point me to right sie
19:42.59[TK]D-Fenderx86: broken EOL chars, etc
19:43.10x86[TK]D-Fender: doesn't look like broken EOL to me
19:43.12viliarquistion is,as i think, is noob.
19:43.22x86[TK]D-Fender: actually, I think like 11 says the problem
19:43.24jaytee~ask
19:43.24jbotmethinks ask is Questions 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.
19:43.29[TK]D-Fenderx86: #334 begs to differ.  Normal CLI doesn't do shit like that
19:43.40[TK]D-Fenderx86: 1 line = 12 line.  no stupid ^M's visdsible
19:43.58x86[TK]D-Fender: the ^M is the EOL sent from the phone
19:44.06viliarthere is some of astersik users. Everyone have sip acc ount and mobile phone
19:44.09x86[TK]D-Fender: inside the SIP header
19:44.13[TK]D-Fenderx86: You saying it looks like that from * CLI?
19:44.22x86*nod*
19:44.26[TK]D-Fenderx86: W-T_F
19:44.37viliarso I want to route call to sip, if it's unreachable - route to mobile
19:44.51viliarhere is no problem
19:45.07Kobazhow would i reset the queue information in 'queue show', without having to unload/load the module
19:45.26[TK]D-Fenderx86: But yes, a 401 sure looks like something to fix
19:45.48x86not sure why it'd be 401'ing though... lemme show you my dialplan
19:46.05viliarbut scheme is more complicated if I want to move some calls to more than one account.
19:46.21[TK]D-Fenderx86: no need
19:46.30[TK]D-Fenderx86: 401 has nothing to do with dialplan
19:46.31Kobazwalnut*CLI> queue show
19:46.31Kobazgroup3       has 0 calls (max unlimited) in 'roundrobin' strategy (0s holdtime), W:4, C:5, A:123, SL:4.0% within 60s
19:46.40Kobazhow would i reset those numbers?
19:46.46*** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net)
19:46.52viliarso it's good to be, if was a they to describe abstract user and send call to him, not directly to sip or mobile.
19:47.02viliar*if was a way
19:47.27jayteeviliar, try reading up on local channels in the WIKI or in the book
19:47.33[TK]D-Fenderviliar: Makes no sense.  "abstract" what?  for who?  What do you want to have happen exactly?
19:47.57viliarok
19:47.57[TK]D-Fenderviliar: And there is no "direct".  Everything goes through the dialplan.  It does what you tell it to
19:48.19etfonhomey_Kobaz, I don't see any way to do that from the CLI.
19:48.20x86[TK]D-Fender: hmm, ok... what am I missing then?
19:48.27[TK]D-Fenderx86: DEVICE AUTH.
19:48.38x86[TK]D-Fender: 7000 can call any other extension just fine (and subscribe to all hints)
19:48.39[TK]D-Fenderx86: 401 = fix your damn sip.conf
19:48.40Kobazetfonhomey_: me neither... other than 'module unload app_queue' 'module load app_queue'
19:48.51Kobazetfonhomey_: which seems kinda bad
19:48.55x86[TK]D-Fender: what's wrong with it if it's only breaking on a single extension?
19:49.07[TK]D-Fenderx86:  turn down all that extra psycho-shit debug and show 2 SMALL samples for working vs non-working
19:49.13etfonhomey_Kobax, do a "module reload app_queue"  that'll do it quickly.
19:49.20x86[TK]D-Fender: also, it DOES ring the other devices, but as soon as other devices start ringing, the call is cancelled BY the originating Linksys phone
19:49.24Kobazetfonhomey_: it doesn;t
19:49.30viliarexample. As if was context [user1]. where is 1,1,Dial(SIP/User1). 1,2,Dial(SIP/Out/User1Mobile)
19:49.32Kobazetfonhomey_: a reload doesn't clear the stats
19:49.32[TK]D-Fenderx86: And you'd better stop calling DEVICES as EXTENSIONS...
19:49.43viliarso I can route call to this context
19:49.53x86[TK]D-Fender: show me example of how I did that?
19:49.57etfonhomey_Kobaz, hmm, always thought a reload just did an unload/load at the same time.  Guess not.
19:49.59[TK]D-Fenderviliar: What call?  From where?  What did they dial?
19:50.11x86[TK]D-Fender: http://pastebin.ca/1278995
19:50.33Kobazetfonhomey_: reload will call the module's internal reload function, which can do whatever it wants... it depends on how the module was written
19:50.38*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
19:51.00[TK]D-Fenderx86: <sip:10.42.69.252>\ <-- having a fixed value like this looks bad
19:51.17[TK]D-Fenderx86: and go show me the working call, and the failing call.
19:51.19jayteeshudders
19:51.23viliar[TK]D-Fender: call from menu is going to some user. It can be done via context user1. Main target - to move call SIP or Mobile.
19:51.30x86[TK]D-Fender: the IP will never change... guess I could put it in a global var, but that's just vanity at this point, not the problem
19:51.39[TK]D-Fenderviliar: "move call = huh?
19:51.53x86[TK]D-Fender: ok, you have failing already... let me make a working call...
19:51.58viliar[TK]D-Fender: sorry for a bad english
19:52.04[TK]D-Fenderx86: I said a SMALL sample
19:52.16[TK]D-Fenderviliar: and this : 1,1,Dial(SIP/User1) will ring forever
19:53.12viliar[TK]D-Fender: no. after timeout it's going to next hop. It's simplified example
19:53.29viliar[TK]D-Fender: next hop - is mobile.
19:53.41x86[TK]D-Fender: not sure how to get what you want!
19:54.00x86[TK]D-Fender: sip set debug peer FOO is giving you all that crap, not sure how to make it less chatty...
19:54.08x86[TK]D-Fender: tell me what you want
19:54.11*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:54.58x86here's the big one of the working call
19:54.59x86http://pastebin.ca/1279001
19:55.46viliarI try to re-describe.
19:55.59x86[TK]D-Fender: this is with core set verbose 0 and sip set debug peer 7002 (x-lite phone)
19:56.18[TK]D-FenderOk, this crap is too much.  I'm stepping out for a while.
19:57.05viliarIncoming call - it's going to main menu. From it it going to user1. if it unreachable - call going to user2,user3,
19:57.39viliaruser1,user2,user3 can be registered sip account or their mobile phones as next step
19:57.42x86gah, why does he have to be such an asshole?!
19:57.51*** join/#asterisk SiberAIR (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com)
19:58.14x86how hard is it to just freaking tell me what you need?!
19:58.20viliarso i just search for apropriate way to create such scheme
20:01.23viliarthe more simple way, as i think here is to describe in some way users and all their phones in macros or context, and than point to it from dial plan.
20:01.33beekviliar: would the FollowMe application do it?
20:02.05viliarthx. wait a second, i try to look.
20:05.48viliar<beek>: thx. looks good. do you think it can be used with Dial(user1,user2)?
20:08.01beekviliar: I'm not sure.  I've never used it before.  It would be easy to try it, though.
20:08.08ShaunWingsay, can't get Asterisk 1.4 to redirect the rtp to my vsp. redirect = yes in sip.conf doesn't do the trick. Any ideas?
20:08.52viliar<beek>:  thanks again. will thiks around this :-)
20:14.17ShaunWing>say, can't get Asterisk 1.4 to redirect the rtp to my vsp. redirect = yes in sip.conf doesn't do the trick. Any ideas?
20:22.12x86ShaunWing: looks like you don't know what redirect=yes actually does
20:23.07ShaunWingI understand it tries to send rtp from ednpoint (Quintum) to VSP
20:23.31ShaunWingHowever c=IN shows me that redirection not taking palce
20:23.34ShaunWingplace
20:24.24ShaunWingThe Quintum is on piblic ip - nonat. The asterisk 1.4 server is also on public ip with nonat. The vSP is also on public ip.
20:24.44ShaunWingI'm chewing up bandwidth on my hosted Asterisk server.......
20:29.56ShaunWingAny ideas?
20:30.31x86ok so you still don't understand redirect=yes
20:30.36x86as it doesn't touch RTP
20:30.59x86redirect allows two endpoints to re-invite and bypass the proxy
20:31.57x86you might get further telling us what the end goal is
20:34.12etfonhomey_ShaunWing, read about canreinvite=yes (or no), maybe that's what you're looking for.
20:35.56ShaunWingSorry, Its as you say, just expressing it incorrectly as new to Asterisk. What I require is that the Asterisk in teh middle (the proxy) does nto carry the rtp through it
20:37.36ShaunWingThe Quintum is on public ip - nonat. The asterisk 1.4 server is also on public ip with nonat. The VSP is also on public ip. I understand that adding in sip.conf reinvite = yes with allow the end points to redirect rtp directly to each other and not need the proxy.....
20:37.46ShaunWingbut its not working in my case....
20:39.27*** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com)
20:40.01Mark17hello, has someone time to help me with configuring asterisk to use a bluetooth device and and incoming calls from the bluetooth device should go to a certain sip account OR trunk and incoming calls from the SIP account OR trunk should go to the bluetooth device. if needed i am willing to pay with paypal. i use debian and asterisk (only configuring the part so i can create a new trunk for the bluetooth device and everything else is done is also an
20:40.01Mark17option)
20:40.13x86ShaunWing: canreinvite=yes is the option
20:42.36ShaunWingTx, but its not working.
20:43.35etfonhomey_ShaunWing, pastbin your sip.conf
20:44.54*** join/#asterisk LND (n=LND@89.193.180.58)
20:47.04ShaunWingTx. http://pastebin.com/m54934d39
20:52.14etfonhomey_ShaunWing, I've never seen trun=yes in a sip.conf before.
20:52.21etfonhomey_trunk=yes*
20:53.16ShaunWingI understand that allows multiple channels
20:53.40etfonhomey_ShaunWing, you don't need it.
20:53.55ShaunWingok
20:54.18ShaunWingWill this fox the redirect issue?
20:54.59ShaunWingfix sahll I say
20:55.05etfonhomey_Is this a production system?
20:55.10*** join/#asterisk LND (n=LND@89.193.214.255)
20:55.12etfonhomey_Or something you're playing around with?
20:55.55ShaunWingIts my server that is live but I can play aorund with it
20:57.38etfonhomey_ShaunWing, then I would simply your config and add features slowly.  I'm assuming RoB5768 is a local extension?
20:58.09ShaunWingno
20:58.22ShaunWingIts the registration for the Quintum
20:58.36ShaunWingthat is sending the traffic to the Asterisk
20:59.19ShaunWingthe reinveite=yes is supposed to tell it to redirect its rtp to the VSP (voipsw1)
20:59.29ShaunWingok will make changes slowly
20:59.34ShaunWingtx for the advice
21:00.01ShaunWingbut at the moment I have tried all I can think of the get the redirect to work...
21:00.05ShaunWingAny ideas please?
21:00.06x86reinvite has nothing to do with RTP, again
21:02.37etfonhomey_ShaunWing, here's my simple SIP line for my ITSP:  http://pastebin.com/d310edd33  Also, you should include your [general] section and any register => xxx  lines in your pastebin.
21:03.37etfonhomey_ShaunWing, and don't forget to mask your usernames and passwords.
21:05.27ShaunWingtx. My general section is default.
21:05.43ShaunWingCan you please tell me what insecure=port,invite
21:05.44ShaunWingdoes?
21:06.32etfonhomey_http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure
21:09.23ShaunWingtx
21:09.37ShaunWingI see you don't use reinvite=yes
21:12.05etfonhomey_That's because I want my * box between my SIP peers.
21:12.31etfonhomey_RTP peers to be more accurate (and it is canreinvite=yes NOT reinvite=yes).
21:13.14ShaunWingok I understand. Sorry - tx. Can you make any suggestions to get my system working the way I need , please?
21:15.52etfonhomey_ShaunWing, you need to start from scratch and get a basic call working with canreinvite=no and then change to canreinvite=yes, then debug.
21:16.12ShaunWingBasic calls are working perfectly
21:16.29ShaunWingI just want the reinvite to work
21:16.45etfonhomey_You have tons of other crap in your sip.conf.
21:17.13andrewnanyone know of a provider that can complete calls to toll free numbers that are restricted to Canada?
21:21.41*** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com)
21:25.31rhousandThis is alittle out side of the realm of this channel but I am not sure where to ask. I have set up my managed switch to mirror the traffic on the port where my Voip(MGCP) traffic goes out to a prot where i am running tcpdump to grab and play it with wireshark. If i grab the traffice from my router i can play it but if i grab it from the mirrored port it will not play. any ideas why?
21:26.03rhousandPlease feel fre to recomend another channel for me to try.
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21:27.55ShaunWingok, happy to remove
21:27.56ShaunWingwhat should I remove please?
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23:18.55C4colodoes any know if it is possible to have more than one audio device available to asterisk, and if so how would you 'address' them?
23:19.34C4colowould they be console/1 console/2 etc?
23:20.28C4coloI have a customer that wants up to 5 line-out channels
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23:36.38C4coloso is there no way to address multiple console audio ports?
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23:51.19Mark17hello, has someone time to help me with configuring asterisk to use a bluetooth device and and incoming calls from the bluetooth device should go to a certain sip account OR trunk and incoming calls from the SIP account OR trunk should go to the bluetooth device. if needed i am willing to pay with paypal. i use debian and asterisk (only configuring the part so i can create a new trunk for the bluetooth device and everything else is done is also an
23:51.19Mark17option)
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