irclog2html for #asterisk on 20061218

00:00.11*** join/#asterisk DavoFrom818 (n=Vito310@cpe-76-173-56-41.socal.res.rr.com)
00:00.15DavoFrom818hi
00:00.20DavoFrom818hey guys is there any software that i can use to compress data being passed back and forth so that the bandwidth usage is minmal?
00:00.32DavoFrom818FYi: im using gsm up its using half a meg for every 2 minutes
00:01.46*** join/#asterisk menace_ (i=menace@12.149.108.200)
00:03.54Strom_CDavoFrom818: the GSM codec is already pretty damned compressed
00:04.18Strom_Cif you want something lower-bandwidth, use g.729
00:04.41DavoFrom818how do i install and activate g.729 for one extension
00:05.13orlokCan somebody look at this http://pastebin.ca/282942 and tell me what i am missing?
00:05.32orlokmust be an issue with my knowledge of how include => 's work
00:05.40*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
00:06.30*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
00:06.46DavoFrom818Strom_C how do i install and activate g.729 for one extension
00:07.12Strom_CDavoFrom818: you need to buy a license from digium
00:07.24DavoFrom818what if i use passthrough its free
00:07.26Strom_CDavoFrom818: how fat is your pipe?
00:07.43DavoFrom818my pipe is fat but the client is charged per MB
00:07.59orlokDavoFrom818: voip may not be the best thing for them then.. :)
00:08.29Strom_CDavoFrom818: yeah...if the client is charged per megabyte, find a different ISP
00:08.37EmleyMoorHmmm... seems it may be the same as the Australia issue
00:08.46DavoFrom818in the country they are at they dont have any other options
00:08.58rudholmDavoFrom818: are they charged per byte transfered, or per bitrate transfered?  that could make a big difference.
00:09.13DavoFrom818per MB
00:09.17orlokEmleyMoor: australia issue?
00:09.44EmleyMoorDistinctive ring detection needs a patch if Caller ID is also enabled
00:10.25EmleyMoor(0,0,0 is detected without the patch)
00:10.43EmleyMoorSeems to affect Australia, Argentina and UK
00:10.52EmleyMoor(see bug 3596)
00:12.13*** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-141.qld.bigpond.net.au)
00:12.50DavoFrom818how can i install and use LPC10??
00:13.07Strom_CDavoFrom818: it's already built in
00:13.15Strom_Cbut god help you if you want to use it - it sounds like shit
00:13.33DavoFrom818what about g.723.1?
00:13.44Strom_CDavoFrom818: asterisk can't support that natively
00:14.00DavoFrom818so 723.1 is not supported by asterisk
00:14.02Strom_CDavoFrom818: when the digium transcoder card comes out, it will include g.723 licenses
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00:14.10rudholmGSM offers pretty good sound quality per byte, IMO
00:15.07DavoFrom818can i use DoD CELP?
00:15.08rudholmthe codec also supports single B-Channel speed (4800bps) but you don't want to go there.
00:15.23EmleyMoordring detection works with callerid detection off
00:15.41EmleyMoor396,384,382 is the standard ring here
00:15.44Strom_CDavoFrom818: g.729 is pretty much your most efficient option - everything else sounds worse
00:15.58DavoFrom818humm and 729 is built in?
00:16.02Strom_Cno
00:16.11Strom_Cyou must buy a license to transcode into and out of 729
00:16.12DavoFrom818is there a free version?
00:16.15Strom_Cif you do passthrough it's fine
00:16.20DavoFrom818ok
00:16.23Strom_Cno, it's a patent-encumbered codec
00:16.27Strom_Cthere is no "free" version
00:16.33DavoFrom818so eyebeam connecting to my server 729 will work?
00:16.47Strom_Cassuming your provider also speaks g.729, yes
00:17.03DavoFrom818no it will speak to my asterisk box
00:17.15Strom_Cwhat codec will your provider be speaking?
00:17.25DavoFrom818umm i think its 711
00:17.37DavoFrom818can not asterisk then take 729 and change to 711
00:17.45Strom_Cthen you will need to purchase one g.729 license for each concurrent call you wish to have up
00:17.47DavoFrom818so that way the server takes the load
00:17.55Strom_CDavoFrom818: yes, that's called transcoding
00:18.00*** join/#asterisk Vegar (i=vegar@51-142-151-213.mtulink.net)
00:18.14DavoFrom818what if client calls is Extension to Extension
00:18.23DavoFrom818and each use eyebeam
00:18.24Strom_Clook, man, it's very simple
00:18.39Strom_Cif asterisk does not need to transcode, then you do not need to purchase licenses
00:18.50EmleyMoorHmmm... trunk revision 27812 has the patch
00:18.53Strom_Cif asterisk does need to transcode, then you do need to purchase licenses
00:18.53DavoFrom818how much is the license?
00:18.58Strom_C$10 per channel
00:19.07DavoFrom818one time fee?
00:19.11Strom_Cyes
00:19.18DavoFrom818where can i buy the license?
00:19.19robl^g729 is icky anyway.  it may reduce bandwidth but it is a PIG for CPU power
00:19.23Strom_Cdigium.com
00:19.56DavoFrom818ok im there
00:20.00DavoFrom818products?
00:20.14Strom_CDavoFrom818: beats me; read the site.
00:20.22DavoFrom818k thnx
00:22.42DavoFrom818ok purchased
00:22.57DavoFrom818is there a readme on how to install the license on asterisk?
00:23.19Strom_CDavoFrom818: no, you have to do it blindfolded while handcuffed to a bear
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00:23.33DavoFrom818Strom_C lol danm
00:23.34DavoFrom818i knew it
00:23.41wunderkinstephen colbert would not like that
00:24.34wunderkinapparantly he has his doctorate.. check the credits :)
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00:34.52DavoFrom818<PROTECTED>
00:35.13DavoFrom818allow=g729
00:35.33DavoFrom818ok thnx
00:36.29QwellDavoFrom818: any time :p
00:38.19fileQwell: .
00:38.54Strom_Cfile: <
00:38.58lenne_dkwhy this in sip.conf: ;    Tip 2: Use separate type=peer and type=user sections for SIP providers
00:38.58lenne_dk;           (instead of type=friend) if you have calls in both directions
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01:21.41_mattis it possiable to run dialup links over voip with asterisk ?
01:23.44tinrshbye all
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01:31.57knarfly8->
01:32.11knarfly8-)
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01:59.09knarfly8-)
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02:14.45bsdfreakanyone here use asterlink?
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02:22.15bsdfreakyes i am.
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03:28.27trelaneanyone mind looking at:http://pastebin.ca/283149  I have an iaxy that refuses to register, asterisk is saying it's invalid, but passwords and such match
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03:33.50irqtrelane: hello
03:33.58irqtrelane: remember me? :)
03:34.05trelaneindeed
03:34.10irqtrelane: anyway, stupid question, but did you power cycle the iaxy? they don't take the new settings until after
03:34.11trelanebeen quite awhile
03:34.36trelaneirq, several times (I've factory reset and re-provisioned it as well)
03:34.53irqhmm
03:34.55trelaneit's got a new account name and as best I can tell it seems to realize that
03:35.07irqwell, i'm happy to give you copies of my relevant files, hang on
03:36.05trelanesure
03:36.12trelanemine should be good but I'm confused as to why it's griping
03:37.16irqhttp://pastebin.ca/283158
03:37.18irqthere you go
03:37.21irqthat's from a live / working config
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03:45.35DaeJeon-Newbiewhat version of mandriva is good to go with an asterisk server?
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03:46.05DaeJeon-NewbieI want to play with an asterisk server on mandriva OS
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03:55.47trelaneDaeJeon-Newbie, the latest?
04:00.03DaeJeon-Newbietrelane: linux 2007?
04:00.27Nuggetasterisk doesn't care.
04:00.46shmaltzis it true that Linux Sivta is being released around the begining of January 2007?
04:00.46Nuggetanything remotely unix-like is enough to play with asterisk.
04:00.52*** join/#asterisk flenders (n=fserto@b03DE.static.pacific.net.au)
04:01.56trelaneDaeJeon-Newbie, if that's the latest, it's best to have the most current os you can
04:02.33Nugget"best" and "current" are blurry subjects.
04:02.41Nuggetuse whatever unix you prefer.  asterisk won't mind.
04:03.34DavoFrom818where can i get royalty free moh music?
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04:05.59flendersHi, I've setup an IVR menu which you get to when dialing our main number. When it says to dial the extension, and you dial any other external number, it dials out. how can I only allow internal extension numbers?
04:06.22Nuggetflenders split up the two types of extension into different contexts.
04:07.15flendersok, I thought about it, but then, would the extensions be allowed to dial out?
04:07.57Nuggetif you set it up that way, sure.
04:08.09icyfire0573flenders: extensions shouldn't be allowed to dial out, but if some extensions need to dial outside numbers they should have the complete dialcode in the extension
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04:08.55flendersicyfire0573: sorry mate, I wasn't clear... I meant the extensions themselves would be able to dial out?
04:08.57Nuggetinternal phones should default to a context which contains or includes the various trunks you want them to be able to access.  the default configuration is set up in this way.
04:09.15Nuggetrevisit the sample configs and develop an understanding of how it works
04:09.41flendersNugget: yeah, I think I screwed up my templates, trying to simplify the whole thing
04:09.42Nuggetthe ivr menu should be in a context which contains or includes the extensions you want to have available to people who are navigating the menu
04:09.47icyfire0573like Nugget says, internal context gets the outside numbers available, and the external context only included the internal extesions
04:10.33flendersthanks for now. I'll give it a go.
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04:39.27puzzledhi
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04:50.04Qwellmitcheloc: :D
04:52.40brookshireqwell!
04:52.49Qwellbrookshire: CANOE!
04:52.56brookshireOMG SPOOKY!
04:53.05QwellWhat happens in the canoe, stays in the canoe?
04:53.07Qwellumm..yeah
04:53.24brookshirehah.. thank god i had a paddle
04:53.29Qwellheh
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04:54.17brookshiremy gui is broken :(
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04:57.27fileeep
04:57.29filethere's people
05:00.46[TK]D-Fenderlies
05:01.19filepeople are scary :(
05:01.19*** join/#asterisk SkramX (n=mark@70.86.176.2)
05:01.48SkramXanyone coded for the Cisco (79x0) IP phone's "web" browser while using SIP (instead of CCM)?
05:03.02JunK-YDay changed to 18 Dec 2006 sounds like a good day!
05:03.23SkramX:)
05:03.34fileJunK-Y, bonjour!
05:03.50JunK-Yfile: salut!
05:03.55fileJunK-Y: ca va?
05:04.10JunK-Youi, toi?
05:04.19filebien
05:04.29SkramXanyone?
05:04.36SkramXxachen: what up? long time no talk
05:05.43fileJunK-Y: how did Julie's birthday go?
05:06.18JunK-Ynot bad.
05:06.24JunK-Ybut mine has just started!
05:06.33fileuh oh - dangerous
05:06.35JunK-Yso she will be my lovely SLAVE mohahahah!
05:06.47filetmi!
05:06.53JunK-Yheheh
05:07.04JunK-Yexit
05:08.08X-Robfeh
05:10.58nvicfhow can I make asterisk register to a voip like a softphone?
05:11.34*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
05:11.38Supaplexto a voip?
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05:11.57Supaplexprovider? like asterisk be the sip client?
05:13.06nvicfI have a voip account, and I want to use asterisk as a pbx with that account, receiving and sending to different interns
05:13.19nvicfit's a receiver account, so I guess it's user instead of friend
05:13.25nvicfbut I'm kind of lost for now
05:14.18Supaplexsip registry
05:14.37nvicfis there an example somewhere?
05:15.08brookshirenvicf: friend call place and receive calls
05:15.14Supaplexwiki
05:15.14brookshires/call/can
05:15.44nvicfbrookshire, yes, that's why I'd use user instead of friend nor peer
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05:16.02brookshirehttp://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
05:16.20brookshireyes.. correct
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05:17.59nochello
05:18.05nukiohey
05:19.05nochow do i pass a unique channel id for everycall on iax2 to a customer
05:19.06noc?
05:21.29nvicfSupaplex, wiki where?there's plenty of things in the wiki
05:21.58Supaplexgoogle for sip registry voip asterisk wiki
05:24.34nvicfif you have the url why do you want me to google for half hour?
05:24.37nvicfI don't understand
05:24.46*** join/#asterisk dorphalsig (i=dorphals@pcsp168-254.supercabletv.net.co)
05:24.48dorphalsigHi
05:25.19dorphalsigI'm trying to get two * servers to speak to each other via SIP
05:25.30dorphalsigso they can "share" part of the dialplan
05:25.51nvicfbye
05:26.00dorphalsigMust I register every single extension I want to use at the remote server?
05:26.15dorphalsigor can I just do one SIP registration at the remote server
05:26.32dorphalsigand then dial like dial(SIP/BLA/extension)
05:26.58shellsharkthat Dial statement is wrong
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05:27.24dorphalsigHow do I do that then?
05:27.27shellsharkhmm maybe not ;)
05:27.31shellsharknevermind, i read it wrong
05:27.54dorphalsigBLA is like the SIP Trunk (the one I have in sip.conf)
05:27.59dorphalsigand the exten is the remote * exten
05:28.36xachenwhoc alled?
05:28.41SkramXhey
05:28.48dorphalsigMe
05:28.50WilliamKheya
05:29.07xachenSkramX: not much, you?
05:29.12SkramXnot much
05:29.25SkramXever used a cisco ip phone with asterisk/SIP and written a web-app for it?
05:29.25dorphalsigDid you guys understand what I asked? I think maybe I wasnt too clear
05:29.26dorphalsig:$
05:29.37xachencan't say I have :P
05:29.39dorphalsigI have this
05:29.43SkramXmy boss may buy me a 7970G if I can write web-apps for it
05:29.56dorphalsig*1 --- SIP --> *2
05:29.56xachen:p
05:30.10xachenI keep begging for a 7970
05:30.11dorphalsigI want to acll exten 888 on *2 from *1
05:30.13xachenbut alas no luck
05:30.31dorphalsigand also I wanna call exten 999 on *2 from *1
05:30.46dorphalsigmust I make a new SIP entry for each?
05:30.51WilliamKhas anyone done anything relating to video w/ asterisk yet?
05:31.27dorphalsigor can I just like make ONE and then dial like dial(SIP/TRUNK/EXTEN) ??
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05:47.37vooduhalHello all.  I have a question.  I've got a sangoma A104 card and I've set up the 4 PRIs, and in the same box I've got a digium modular analog card with 2 FXS and 2 FXO modules.  I'm having a problem figuring out which channels are assigned to which card.  Could someone point me to information that could help me with this?
05:48.50IronHelix\AFKthey go by order
05:48.58*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:49.04vooduhalAnd what would that order be?
05:49.14IronHelix\AFKso if the digium card is the first in the system (highest pci slot i think) it would be 1-4 and the other would be 5-whatever
05:49.29IronHelix\AFKmight go by IRQ
05:49.37IronHelix\AFKbut either wya- they block together
05:49.45vooduhalIs there anyway to determine this on a running system?
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05:50.12IronHelix\AFKas i recall there is, lemme see if i can find it for you
05:50.38vooduhalLmao. Just figured it out.  It goes by module loading order.
05:50.48IronHelix\AFKthere you go :)
05:51.34vooduhalAlso.  really stupid question.  Does the span number need to change with each interface on the quad PRI card?
05:52.33JTyes, they're different spans
05:53.59vooduhalK.  Thank you.
06:01.49vooduhalCan I paste part of a config in this channel or would you rather me post it somewhere else?
06:02.01naftali5? pastebin
06:02.05JT~pb
06:02.07jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
06:02.10vooduhalK.
06:04.02*** part/#asterisk dorphalsig (i=dorphals@pcsp168-254.supercabletv.net.co)
06:06.20vooduhalHmmmm... How can I debug what's going on a PRI?  I've got one channel that isn't coming up.
06:06.34JTpri intense debug span <n>
06:08.12vooduhalLol.  Now how do I turn that off?
06:09.54vooduhalOk, does anyone know what this means? http://pastebin.ca/283273
06:10.48X-Robvooduhal, you should get something back from the NT end after you send a SABME
06:11.11X-Robif you're _receiving_ that, then the other end thinks it's CPE
06:12.02vooduhalAh...  I appended my configs to it now.
06:12.06JTpri no debug span <n>
06:12.15vooduhalDoes anything look out of place?  That's supposed to be span 3.
06:12.57vooduhalWell I thought I did.
06:13.58vooduhalTry this: http://pastebin.ca/283281
06:18.27JTvooduhal: "DMS"
06:18.36JTdoes that mean the lines are connected to a DMS100?
06:18.53vooduhalYes.  Those lines are working.  The Avaya PRI is the one that is screwed.
06:19.19JThrm, they should be switchtype dms100 btw, not national
06:19.49vooduhalWould you change it if it has been working for 7 months? or is it important?
06:20.53JTi dunno, spose you could leave it for now
06:21.00JTyour span definitions are defective
06:21.07JTthey all have timin source set to "1"
06:21.08vooduhalI wouldn't be suprised.
06:21.09JTtiming
06:21.35JTthey should either be 1-n in order of preference, of left at 0
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06:21.41JTs/of/or/
06:22.15vooduhalThat's the second value on the span line correct?
06:22.36JTyes
06:22.43JTthe third is the buildout
06:22.52JTwhich is 0 for most people, since most people have short cables
06:23.35vooduhalBut it is ok if I just set them all to 0?
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06:24.11JTit may work, it may not
06:24.24JTbest practice is to sync off telco lines
06:25.37vooduhalI'm not I sure I understand what you mean, but just to help with information, that specific PRI is connected to a Definity (sp) PBX.  Our old box used pri_net on it's definition.
06:26.19JTwell the 2 connected to telco should have timing priority 1 and 2
06:26.30JTthe rest connected to the pbx can be 0
06:26.55JTso you are replacing a box that already worked doing the exact same thing?
06:27.33vooduhalYep.  The only difference is the old box only had one PRI to the Avaya.  The new box has 2 PRIs to our DMS and one to the Avaya.
06:29.55JTthat's different how?
06:30.17vooduhalOh, the other difference is the new box is using Sangoma hardware and not digium.
06:30.34vooduhalCould there be a setting in the wanpipeX.conf that is causing the problem.
06:30.47JTpossibly
06:30.54JTbut you've described two identical configs
06:30.57JTotherwise
06:33.30vooduhalQuestion.  When using pri_net, does that mean my device will provide clocking to the other end?
06:33.48JTit should do
06:34.04vooduhalI think I may have found part of the problem.
06:34.53vooduhalTE_CLOCK = NORMAL is set in the wanpipe3.conf.  Looks like there is also a TE_CLOCK = MASTER
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06:36.51vooduhalNope, that didn't help.
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06:44.43Mattwj2005Merry Christmas all! :)
06:55.30vooduhalStill having the same problem.  Any other ideas?
07:09.18ManxPowervooduhal: let me check my sangoma configs
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07:11.24ManxPowerfor the spans connected to the telco you want
07:11.24ManxPowerTE_CLOCK        = NORMAL
07:11.24ManxPowerTE_REF_CLOCK    = 0
07:12.00ManxPowerOdd, I have that for all spans.
07:12.14ManxPowerI suspect that for Asterisk the timing is set in /etc/zapata.conf
07:14.34ManxPowervooduhal: T-1/E-1 timing problems generally show up as problems with fax/modem and blips in the audio stream
07:14.45ManxPowerI'm not aware of any other symptoms of that
07:15.33ManxPoweryup, looks like timing is set in /etc/zaptel.conf
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07:19.08warezDUDEhello....is this a nigger-friendly chat channel?
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07:34.16vooduhalManxPower, thanks for looking.
07:34.34vooduhalI'm about to pull my hair out.
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07:34.51vooduhalAfter doing some reading, I've got the the PRI from the telco as the clocking source and I have that synchronized to the PRI to the Avaya and that's reflected in my zaptel.conf now, but still RED. :(
07:34.52vooduhalI've even gone as far as to remove configuration for everything but the two PRIs and still nothing.
07:34.52warezDUDEvooduhal: hmmm...i think you need to microwave your FXO card
07:34.52warezDUDEthat worked for me a few times
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07:34.52vooduhalStill just getting these from span 2 http://pastebin.ca/283333
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08:05.21nharlequi516How do I close a channel (hangup) a call from the asterisk cmdline?
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08:22.22angryusergood morning everybody
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08:24.38Stephniehi
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08:25.21StephnieI have commented out all the content of SIP.conf but SIP RELOAD still take time to reload sip_notify.conf (about 5 to 8 seconds)
08:25.50angryuseri have a tiny problem, zaptel module is not loaded on start on linux(zaptel led's on card are off) when i do modprobe wctdm, they start up w/o problem, but i want to load asterisk automaticly on start, i need something like that : modprobe wctdm then ztcfg then asterisk, hod do i realize that?
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08:35.37merbananplease kick [SuLe] and Cindy[hoRrRr], I got onjoin messages from them
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08:41.07dlynes_laptopWere they sexy messages?
08:41.58hadsI parted and joined and didn't get anything. No one loves me.
08:49.04dlynes_laptopYeah...same here :(
08:49.17dlynes_laptopI love free pr0n just as much as the next guy
08:51.56hadsYeah, I'm all dissapointed now.
08:52.26angryuserwhat new in beta 4?
08:52.58pifdlynes_laptop : typing one-handed?
08:53.35dlynes_laptopnah...two handed
08:53.42dlynes_laptopbut my keyboard's getting kinda sticky :(
08:53.57dlynes_laptopangryuser: did you check the changelog?
08:54.01pifi sympathize :)
08:54.01hadshttp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.0-beta4
08:54.06dlynes_laptopangryuser: the changelog should tell you everything that's new
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08:55.15dlynes_laptopFrag you, kermit!
08:56.39angryuserthx
09:12.33pifdoom -file /home/ldm/plutonia.wad -warp 12
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09:18.37EmleyMoorIs there an easy way to add a patch to a system runb
09:18.55EmleyMoorrunning * from Debian?
09:20.34*** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com)
09:20.54pifapt-get source asterisk; patch; debuild; more; yes.. ahem; apt-get install etc..
09:21.13EmleyMoorI could do with the patch for dring-after-cid
09:21.13EmleyMoor(sometimes called the Australia patch but needed in Argentina, New Zealand and UK as well)
09:21.28hadsIt's included in 1.4 I believe, not sure about 1.2
10:42.30EmleyMoorDebian is likely to be stuck on 1.2 FTTB
10:42.30hadsIndeed, I use Debian but install from source.
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10:42.40tparcinahi channel!
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10:42.41EmleyMoorWhat do you mean by "dialup links"? Data? I doubt it
10:42.41EmleyMoorOoops
10:42.41*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)
10:42.41dlynes_laptopgood morning, tparcina
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10:42.44angryuseri have 1 user behind nat, his fxs has ports routed to recieve and send calls, i need to add another one with send/receive capability, what should i do coz, i cant route same ports to 2 ip's
10:42.44angryuseri need 2 users(same network) <nat>internet<nat>asterisk
10:42.45pifuse a vpn
10:42.45angryuserpif:vpn sounds like an option thx
10:42.45*** join/#asterisk ValDuane (n=valduane@cpc4-basf6-0-0-cust470.nott.cable.ntl.com)
10:42.46DaeJeon-Newbiedo I need to install mpg123 , if I install asterisk 1.4 beta?
10:42.46DaeJeon-Newbiewill get music by defalt?
10:42.46*** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au)
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10:42.47xainhi
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10:42.48tuck3ri'm trying to cross compile 1.4, but make is still /usr/bin/ld not the cross compiler ld. is 1.4 just not coded for cross compiling or am i doing something wrong?
10:42.48tuck3rduring the configure script it detects the proper ld but when i go to compile it doesn't use it
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10:43.06^Sprint^need a sponsor for a big website (40K unique / day). if some host admin can help me. pm please!!
10:43.10holmier^Sprint^: i can help you i think
10:43.11dlynes_laptop[hC]: good morning
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10:43.12*** join/#asterisk zyz (i=zyz@h100n4-vrr-gr1.ias.bredband.telia.com)
10:43.12zyzhello .. is it possible to use asterisk as a replacement for ventrilo? Me and my friends use ventrilo for communications when playing games over the internet and for talking shit. So is it possible to setup asterisk to mimic ventrilo to have separate "rooms" and userlist with online users in each room etc?
10:43.13[hC]dlynes_laptop: hi :)
10:43.13tuck3rzyz: not really
10:43.13You_Asteriskdoes someone use linksys voip phones?
10:43.13tuck3rzyz: asterisk is more meant for phones than vent or TS
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10:43.15santibioticoanybody with a nokia n91 and asterisk?
10:43.15dlynes_laptopYou_Asterisk: lotsa peeps
10:43.26E-bolaare there anything special you need to do in asterisk when you want to use a fax via a pap2 adapter?
10:43.27E-bolai get Unknown RTP codec 100 received
10:43.27pifYou_Asterisk : I have a wip330
10:43.27shellsharkpif: you like it?
10:43.27pifworks fine
10:43.27shellsharkpif: do they have bluetooth capabilities?
10:43.28X-RobE-bola, asterisk doesn't support T38 endpoints. Openpbx does, and asterisk 1.4 only supports passthrough of T38
10:43.28pifminuses: no desktop stand, no speakerphone
10:43.28pifshellshark : no bt
10:43.28shellsharkwhere you could sync your address book, emails, etc from your computer wirelessly?
10:43.28E-bolaX-Rob: whats weird is it works the oposit direction
10:43.28X-RobE-bola, because you're not using T38
10:43.29*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:43.29E-bolaX-Rob: why is it used when i send then?
10:43.29E-bolai can receive but not send
10:43.29shellsharkpif: can you sync with asterisk's directory and/or your desktop by any other means?
10:43.29X-RobE-bola, no idea.
10:43.30X-RobUse OpenPBX.org
10:43.30X-Robor contact sipura and ask them.
10:43.30pifshellshark : dunno, but it's a windows mobile OS
10:43.30shellsharkpif: does it have a USB connection?
10:43.30pifack
10:43.30shellsharkif so, you could probably use ActiveSync
10:43.31pifit's a wi-fone, so you can connect
10:43.31shellsharki know what it is
10:43.31shellsharki was asking if it had a USB port
10:43.31shellsharkso that you could use that to sync contacts
10:43.31*** join/#asterisk angryuser5883 (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr)
10:43.31pifwhy use usb if you can sync over the air
10:43.31shellsharkbecause USB is more secure :)
10:43.32shellsharkdoes it have one or not? not a hard question :)
10:43.32*** join/#asterisk apardo (n=apardo@87.217.144.227)
10:43.32pifi said ack, it you tcp stack broken?
10:43.33pifs/it/is/
10:43.33shellsharkack starts a conversation... what's that have to do with my question?
10:43.33shellsharknevermind, i'll look it up ;)
10:43.33pifSYN starts the convo
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10:43.33Stephnieany good Linux Dedicated Server provider?
10:43.33pifdedibox.fr offers a 160G server with unlimited 100MB ethernet
10:43.34piffor 29 EUR/ month
10:47.56*** join/#asterisk apardo (n=apardo@87.217.144.227)
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10:47.57You_Asteriskcan someone tell me what is the recomanded codec with asterisk
10:47.57You_Asterisk?
10:49.40pifalaw if you have the bandwidth, speex otherwise
10:49.41dlynes_laptopulaw if you're in north america, or japan
10:49.41dlynes_laptopif you don't have the bandwidth, or bandwidth is a high price commodity however, there's g729
10:51.53*** part/#asterisk ^Sprint^ (n=Hanan@84.229.52.104)
10:54.17coppiceI recommend 320kbps MP3. at least it makes the damned RTP overheads look reasonable :-)
10:57.12E-bolaare there no patches that get t.30 working with asterisk?
10:58.10You_Asteriskhehe
10:58.10You_Asteriskthank you
10:58.11You_Asteriskbut i m in africa exactly in morocco
10:58.54*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
10:59.11You_Asteriskthere are many version of g729 aren't?
11:00.12dlynes_laptopYou_Asterisk: check digium's website
11:00.18dlynes_laptopYou_Asterisk: it's $10 USD / channel
11:00.45coppicein the real worls, no. G.729 is always the same 8kpbs codec. there are a number of annexes to the spec for other rates and extra features, though
11:02.20*** join/#asterisk hwt (n=hwt@curb.thorkildssen.com)
11:02.23hwthi.
11:02.47You_Asteriskok
11:03.05hwtwith the app MYSQL(), how can i do an insert query that actually inserts ${EXTEN}?
11:03.26hwtit will always just insert the content of the variable, not the string itself.
11:03.37You_Asteriskso when i m gonna to buy an ip phone i must verify if it supports g729 or not?
11:04.56hwtYou_Asterisk: g729 is optional.
11:05.05Strom_CYou_Asterisk: the majority of IP phones I've run into support g.729 as a low-bandwidth codec
11:05.24You_Asteriskhwt:how?
11:06.24You_AsteriskStrom_C: not all
11:06.54Strom_CYou_Asterisk: well, I haven't checked on all of them
11:07.02You_Asteriski need to by more the 50 ip phones; so i must make a good decision
11:07.19Strom_Cpolycom ip430 is a good choice
11:07.45*** join/#asterisk zotz (n=zotz@24.244.163.157)
11:08.07You_Asteriskhat about the price
11:08.27Strom_Chwt: I know there is a way to escape strings in asterisk, but I can't remember how :)
11:08.48Strom_CYou_Asterisk: price varies; check various resellers
11:09.12You_Asteriskok
11:09.14You_Asteriskthank's
11:09.51Strom_Chwt: try escaping using the backslash
11:10.08Strom_C\$\{EXTEN\}
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11:17.47EmleyMoorI have installed a patched version of asterisk but the patch doesn't appear to have had any effect :-(
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11:18.21dpenevI would like to connect to an asterisk server using nc -u 127.0.0.1 5060 working on my local machine .
11:18.21dpenevunfortunately I don't see anything in the * console as I type in the nc client console. Is the someone who can clarify this to me  
11:18.37dpenevI have set verbose to 10
11:19.11EmleyMoorIs there a way I can tell from the so file whether it contains a particular patch?
11:19.37dlynes_laptopEmleyMoor: strings filename.so | grep symbol_you_expect_it_to_have
11:19.56EmleyMoorHmmm... it's there
11:21.50EmleyMoorIt's the patch for distinctiveringaftercid - it's made no difference - still see 0,0,0
11:22.42dpenevOr is there other simple way I can remotely verify that there is * server up an working responding on given IP without using softphone?
11:24.40monstedi suppose you could telnet to it and see if it responds to SIP requests
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11:25.43monstedor rather use netcat, port 5060 is udp, isn't it?
11:26.08dpenevyes that what I've used nc -u 127.0.0.1 5060
11:26.48dlynes_laptopOr try using sipsak or sipp
11:26.50dpenevbut I don't see any life in the * console as I type in the NC client console?  
11:27.04dpenevIs this normal?
11:27.40dlynes_laptopdpenev: did you do a stop command?
11:28.12dpenevno I just starter * server typing asterisk
11:28.25dpenevthen set verbose 10
11:28.39dpenevit is on fedora C5
11:29.18dpenevbefore that I did a simple server -client test using nc command and it is working fine
11:30.25*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
11:30.29dpenevI mean I tried nc -lu 3333 and then in other console nc -u 127.0.0.1 3333 and I got simple chat application  
11:30.58dpenevAny ideas?
11:31.44Strom_Cdpenev: I think you're overcomplicating the problem :)
11:32.45dpenevyes probably I am new to the asterisk and linux
11:33.09pifyou have a steep learning curve ahead
11:33.18dpenevhow can I verify remotely that I have * server up and running?
11:33.19Stephnieif I include a file in SIP.conf ....and put peers in that file....then do I need to issue sip reload after changing include file?
11:33.49Strom_Cdpenev: ssh into the server, and then "asterisk -r" to connect to the console
11:33.56*** join/#asterisk potsboy (n=chrisg@196.211.16.202)
11:34.33dpenevdoes asterish -r works remotely?
11:34.50dlynes_laptopdpenev: through an ssh session?
11:35.44Stephnieif I include a file in SIP.conf ....and put peers in that file....then do I need to issue sip reload after changing include file?
11:36.38dpenevI well I don't know what ssh is actually, an encrypted channel I guess? well I just look for the simplest way without softphone one to verify if on remote PC theer is asterisk up and running ... can you point m eto the simplest way please
11:37.39Strom_Cdpenev: if you don't even know what ssh is, I strongly suggest you first go learn how to use linux :)
11:38.01dpenev:-) Ok Storm_C thanks anyway!
11:47.51*** join/#asterisk _omer (i=_omer@202.38.55.125)
11:47.52Stephnieany help ?
11:47.52Stephnieif I include a file in SIP.conf ....and put peers in that file....then do I need to issue sip reload after changing include file?
11:47.52Strom_CStephnie: yes
11:47.52_omerany way to avoid sip reload again and again ?
11:47.52_omerthats my problem too :)
11:47.52Stephnieexactly
11:47.53Strom_Cyou can use the realtime interface
11:47.53*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
11:47.53badcfeStephnie: do you experience a problem doing sip reload often?  You make some automatic mechanism around asterisk?
11:47.54Strom_Cyeah, add  'asterisk -rx "sip reload"' to cron :)
11:47.54badcfeStephnie: i do that my self.  and planning to make my fast-AGI server handling it realtime, but for the time being i do repeatedly "asterisk -rx 'sip reload'" and it seems fine..
11:47.54_omeryes. some sip providers dont allow sip registration again and again.
11:48.10Strom_C_omer: so comment out the register line until you're done screwing around with sip.conf, then uncomment it before you do one final sip reliad
11:48.12Strom_Creload
11:49.08_omerhmmmm
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11:52.43FreezeShey guys
11:52.43FreezeSanybody has experience with DUNDi ?
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12:21.52Hermione_!tasks
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12:24.43FreezeScan I set a queue to use context:extension instead of protocol/id ?
12:25.16Strom_CFreezeS: as the queue member?
12:25.29FreezeSyes
12:25.47Strom_Csure, you could do something like member => Local/2368@stations
12:25.53FreezeSbecause I want to use dundi lookup
12:26.11FreezeSaha, thanks, I'll try it
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12:34.12hwtStrom_C: nah, that doesn't work (\$\{EXTEN\}).
12:34.47Strom_Chwt: my question is why you need to pass that into mysql in the first place
12:37.21*** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk)
12:37.33hwtStrom_C: i'm changing macros for a sip friend from the dialplan.
12:38.02hwtStrom_C: stdvmail|SIP/${EXTEN} becomes stdvmail|SIP/s
12:38.02*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
12:38.02Strom_Chwt: are you using realtime?
12:40.35hwtStrom_C: yes.
12:41.20Strom_Cperhaps I'm just tired, but explain why you're doing this from the dialplan?
12:43.36hwtStrom_C: i want to change the extension for a sip friend, by letting a user change the way a call is handled both from a web interface and from an IVR menu.
12:44.06*** join/#asterisk infernix (i=nix@spirit.infernix.net)
12:44.19Strom_Cok - why not use AGI?
12:44.31*** part/#asterisk dpenev (n=dpenev@89.253.156.161)
12:44.54santibioticodoes anyone use nokia n91 with asterisk¿¿
12:45.13hwtStrom_C: that is of course a possibility, but when i found MYSQL() that seemed ideal.
12:45.42Strom_Chwt: alright....I do seem to remember there being a way to do this
12:45.59Strom_Cgive me a minute or six
12:47.00hwtStrom_C: kthanks.
12:49.52*** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk)
12:50.03Dr-Linux|home~dict parse
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12:54.12*** join/#asterisk alesan (n=alesan@adsl-ull-153-85.41-151.net24.it)
12:55.01alesanhi there... I have a question about VoIP using sipura devices... is there a better channel for asking that?
12:55.10Strom_Calesan: ask away
12:55.15*** join/#asterisk saftsack (n=saftsack@pD9E0713C.dip.t-dialin.net)
12:55.30alesanI understand I'm OT :)
12:55.51alesanok
12:56.27alesanI have a sipura 3120 and a 1001
12:57.25alesanI would like to configure them to "trasfer" the pots analog line from a place to another
12:57.26alesanso that if I use an analog phone connected to the 1001 the call is forwarded to the 3120 and then on the FXO port on the pots line
12:57.48alesanand, when a call is made to that analog line, the 3120 automatically forwards the call to the 1001
12:57.59Strom_Cand you want to do this using nothing but the two sipuras?
12:58.10alesanif possible, yes
12:58.28alesanthat's why I asked for a better channel for thi s:)
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12:59.00Strom_CI personally have no clue, but if you hang around here long enough, someone will probably know
13:00.04alesanI have found this
13:00.06alesanhttp://www.provu.co.uk/pdf/sipura/spa_backtoback_1x_spa3000_and_1x_spa1001.pdf
13:00.20alesanit describes how to do that with a spa-3000 and a spa-1001
13:00.36alesanI have already done that but I have problems :)
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13:01.04alesanok Strom_C, I will wait for someone that possibly help me
13:01.17*** join/#asterisk xnon (n=xnon@200.8.5.123)
13:02.25*** join/#asterisk xnon_ (n=xnon@200.8.5.123)
13:03.12alesanis this a correct dialplan: "(S0<:192.168.192.12>)" ??
13:03.25*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
13:04.11dlynes_laptopalesan: have you taken a look at the Sipura User Group forum on Voxilla?
13:04.47monsteddo any of you know of a device that'll do skype-to-FXS with no computer involved?
13:04.50dlynes_laptopalesan: There's a wealth of information there for configuring sipura units
13:05.07dlynes_laptopmonsted: DLink has such an animal
13:05.12dlynes_laptopmonsted: check www.dlink.com
13:05.49dlynes_laptopalesan: There's also the manuals you can download from sipura.com, too
13:06.21*** join/#asterisk Gunnar (n=gunnar@62.97.242.6)
13:06.24dlynes_laptopalesan: however, unless you've got an account with them, you won't be able to download the administrator manuals (you usually need to commit to a minimum buy order to get that)
13:06.51monsteddlynes_laptop: any idea what it's called?
13:06.59dlynes_laptopmonsted: gimme a few
13:08.26monstedi only see the usb phone adapter, which is dependant on a computer being online
13:08.59monsted(oh, btw, somebody get that chan_skype thing done! :))
13:09.16Strom_Cyech
13:09.38alesandlynes_laptop, thanks for the info
13:09.41alesanI'll check that forum
13:10.30dlynes_laptopmonsted: it's already done
13:10.33monstedi've got one of those usb/skype things already - works fine with an FXO gateway and even forwards the CID (skype name) through asterisk to my SIP phone
13:10.39monsteddlynes_laptop: does it work? :)
13:10.47dlynes_laptopmonsted: there's something like three different skype implementations for asterisk
13:11.01dlynes_laptopmonsted: I think they're all commercial implementations, though
13:11.08*** join/#asterisk jeffik (n=Jeff@pool-71-101-197-41.tampfl.dsl-w.verizon.net)
13:11.14monstedah, the PGW thingie and friends?
13:11.38dlynes_laptopno idea what the names were
13:11.44dlynes_laptopI don't use skype for talking on
13:11.52dlynes_laptopI just remember seeing them when I was looking for other stuff
13:11.53monstedthere are plenty of options to pay for using a free service ;)
13:12.13Strom_Cmonsted: why do you want to use skype in the first place?
13:13.12dlynes_laptophrm..I guess it wasn't dlink
13:13.21dlynes_laptoptrying to remember where i saw that skype stuff now
13:13.33monstedStrom_C: because everybody else does
13:13.53monstedStrom_C: SIP is quite useless if noone else can accept a SIP call :)
13:14.10Strom_Cif everybody else hammered white hot railroad spikes six feet into their eyeballs, would you do it too? :)
13:15.40Strom_Ci'd like to see how you've decided who falls into this mythical "everybody else" category
13:15.58cpmpunters
13:16.09cpmfolks who install everything a banner ad tells them to
13:16.15monsted100% of the people i talk with regularly who have any sort of VoIP are using skype
13:16.28monsted0% are using anything else
13:17.17monstednah, sorry - 2% are using some SIP provider who i can't call for free without having an account with them
13:17.53Strom_Csee, my stance on the issue is that skype blows donkeys for quarters, and it's cheap enough to call people on their regular telephones.
13:17.59cpmlike you can't call skpe without having an account with them?
13:18.26monstedcpm: i can't get an account with the others without having a PAID account, typically
13:18.54cpmskpe=proprietary BS, no perceptable business model that won't have to eventually turn it into adware.
13:19.02Strom_Cthe cost of going from "really cheap" to "free" is too much to consider switching to skype
13:19.06cpmsip=open and free standard
13:19.07monstedso far, the ones i've run into all require me to have a small balance on the account to allow me to get a phone number
13:19.46HarryRmonsted, most UK voip services will give you a free 0871 number or (in the case of Tesco Internet Phone) a free local number without paying anything
13:20.10monstedcpm: zealotism is fine, as long as you don't expect to do anything useful - some of us just want what works :)
13:20.10*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
13:20.57EyeCuewhat works is a relative term, as is what works in the best interests of the user
13:21.08EyeCueits the concept not the implementation that is key
13:21.33HarryRI agree entirely in the opposite direction, it's atleast 75% down to the implementation
13:21.39cpmmonsted, that's a tired argument full of complete misapprehensions of reality. Park it.
13:21.59monstedwell, you can go ahead and hold your nose when you see a skype user, but there are after all millions of them and only a few who actually use SIP
13:22.03EyeCueHarryR, i suppose youre right .. iis does 'work'
13:22.23EyeCueas does asp, mssql, *ponders*, access, *ponders* sap, *thinks* vista
13:22.40HarryRyeah and it's really quite an acceptable web server, and mssql can easily hold it'self in the market
13:22.51EyeCuehold itself to what exactly?
13:23.00HarryRagainst competition :)
13:23.07EyeCuereally? thats quite a big call
13:23.26HarryRPeople just have a jaded view of it because it's microsoft
13:23.33monstedand i can't be bothered to get an account with every SIP provider to be able to do what skype hands me on a platter, even if it smell a bit like pee
13:23.42EyeCueif i had a dollar for every isp that would drop iis in a second if their n00b clients didnt have sites that required access/asp, it have a bazillion dollars
13:24.04EyeCuetake ms out of the equation and put another company in there, it makes no difference
13:24.30Strom_Cmonsted: you know, there's another technology that even more people have than skype.  it's called the PSTN :)
13:24.37EyeCue</feeding-the-troll>
13:24.38EyeCueim done
13:24.42HarryRaha
13:25.07HarryRnah, access and asp is really bad combo, and php's only just catching on with the sqlite+php combo
13:25.22EyeCuephp is catching on with itself?
13:25.34monstedStrom_C: but i have to pay for that :)
13:25.42HarryRoh no, it's 'Embracing and extending' ideas from the past
13:25.48monsted(well, my boss does)
13:25.51*** join/#asterisk dasenjo (n=dasenjo@208.195.215.71)
13:25.57HarryRlike drop an application into a directory, and you've got a fully functional app with SQL backend
13:26.18Strom_Cmonsted: you and/or your boss are probably too cheap for your own good
13:26.19HarryRSure it doesn't scale at all... but that's not a concern
13:26.24EyeCuerofl
13:26.32EyeCueno, slcalbility should never be a concern
13:26.43EyeCues/slc/scal
13:26.53*** join/#asterisk Omer^ (i=Omer@203.81.232.55)
13:26.56HarryRoh and just for forward advice... asterisk+sqlite = big no no, don't even think about it
13:26.57EyeCuehang on lemme go buy you a shovel
13:27.15EyeCuebf2 time, bll
13:27.17monstedStrom_C: my "boss" is the largest telco in the country - this is for me to play with
13:27.24*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)
13:27.30cpmEyeCue, Doh! I fell for it again,
13:28.01Strom_C*shrug*
13:28.04Strom_Cto each his own
13:28.04HarryRAre there any really good soft phones which are easy to re-brand?
13:29.03EyeCuemiranda-im+iax+clist_nicer
13:29.06EyeCue+custom skin
13:32.06*** join/#asterisk linlin2 (n=will@71.194.70.13)
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13:34.15santibioticoi'm having a problem with asterisk....i've defined sip users in a certain context
13:34.29HarryRmiranda can do iax?
13:34.36santibioticothen, for special calls, i use the Goto function to jump to another context
13:34.38santibioticobut then
13:34.54santibioticowhenever i want to transfer a call that has been moved to that new context, i cannot
13:34.56HarryRoh nice
13:35.25santibioticoi assume the problem is that asterisk cannot see extensions defined in another context rather than the one specified
13:35.34santibioticobut is there any way to do what i want to?
13:35.39Strom_Csantibiotico: unless you use includes
13:35.43santibioticoi know
13:35.49santibioticobut i don't want to use includes
13:35.53Strom_Cwhy not?
13:35.57santibioticowell
13:36.01santibioticothe problem is the following
13:36.24santibioticowhenever i want to make a call through the ISDN network, i dial 7 + the phone number
13:37.01santibioticoi also implemented a report tool that looks for the cdr and gets information about all calls made
13:37.23santibioticothe problem is that if i use includes or just do not change the context
13:37.47santibioticothe number dialed that appears in the cdr file beggins with 7
13:37.58santibioticoand i want only the phone number to appear
13:38.17santibioticoif i jump to another context with ${EXTEN:1}
13:39.30Strom_Cwhy not modify your report tool?
13:39.30santibioticoi can get a cdr file with the phone number
13:39.30santibioticoi cannot do that
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13:39.31santibioticowell i obviously can
13:39.31santibioticobut i don't want to
13:39.31Strom_Cwhy not?
13:39.31santibioticobecause the report tool is being used by so many people so i don't want to contact everybody to change the report tool version
13:39.43Strom_Chmm.
13:39.43Strom_Cwhat kind of transfer are you doing?
13:39.54santibioticoi know the easiest way is to change the report tool, but i prefer not to do so
13:39.57santibioticoattended one
13:40.17Strom_Cyeah, but are you using the phone's transfer button?  are you doing a switchhook transfer?
13:40.30Dr-Linux|homeanybody is using fax2mail ?
13:40.32santibioticoi'm not using the phone's transfer button
13:40.47santibioticoi just defined '#' as the attended transfer button in features.conf
13:41.19santibioticoi don't want to use the phone transfer button, as in some places i'm using old analogue phones
13:41.21badcfei get transfers even tho i have no phones with transfer or flash-hook or any thing.  seems like analog phones may trigger surch features
13:41.35santibioticoi have no problems with transfers
13:41.46santibioticoi only have problems when transfering after changing the context
13:41.53Strom_Csantibiotico: doing transfers by pressing the # button is not a good idea
13:42.04santibioticoStrom_C why?
13:42.20Strom_Cuse the transfer button on your sip phone and use the switchhook on your analog phones.
13:42.33Strom_Csantibiotico: well, you're running into problems, aren't you? :)
13:43.05santibioticobut do you think the button defined for attended transfer would cause those problems??
13:43.30santibioticowell..anyway...right now i'm trying with the phone's transfer button
13:43.32Strom_Cno, it's not the button definition
13:43.47santibioticoand i'm having the same problems
13:44.03Strom_Cpastebin your CLI output
13:45.37*** join/#asterisk Osochebol (n=Osochebo@58.186.55.201)
13:50.18santibioticoStrom_C
13:50.20santibioticohttp://pastebin.ca/283616
13:51.19*** join/#asterisk Tili (n=tili@253.Red-88-0-147.dynamicIP.rima-tde.net)
13:51.23hwthow do i install Asterisk::AGI for perl?
13:52.20Dr-Linux|homehwt: there is no such perl module
13:52.28Dr-Linux|homehwt: why you wanna install it?
13:52.36Strom_Chwt: read the readme that comes with it
13:52.51hwtDr-Linux|home: i want to create AGI-scripts with perl
13:52.59hwtStrom_C: comes with what?
13:53.06Strom_Chwt: asterisk::agi
13:53.21Dr-Linux|homehwt: then you don't need per module, i guess
13:53.36hwtStrom_C: my question was how do i install it. :)
13:53.38Strom_Csantibiotico: it looks like there's some weirdness in your transfer context settings
13:53.54hwtStrom_C: i guess "where do i get it" is more precise. :)
13:54.01Dr-Linux|homehwt: look at my site, once i wrote about Asterisk Sphinx recognition system
13:54.08hwtDr-Linux|home: url?
13:54.12Dr-Linux|homethat's perl stuff
13:54.21Dr-Linux|homehold on
13:54.24hwtDr-Linux|home: cool.
13:54.28Strom_Chttp://asterisk.gnuinter.net/
13:54.31Dr-Linux|homewww.syednetworks.com
13:54.36Strom_Chwt: there you go
13:55.48Dr-Linux|homehttp://www.syednetworks.com/category/asterisk-integration-with-sphinx/
13:55.55Dr-Linux|homeaww
13:56.16Dr-Linux|homefor some reasons my site has good google PR :P
13:57.28Dr-Linux|homeStrom_C: ever tried mail 2 fax or fax to mail?
13:57.36Strom_Cno
13:58.08Dr-Linux|homehhm..
13:58.26Dr-Linux|homeAsterFax looks very helful for faxing ..
13:58.46Dr-Linux|homebut i've little doubt fax stuff in zapata.conf setting
13:59.00Dr-Linux|homebut i believe that would be easy .. but i never even use fax in real :P
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14:04.39resistancewhat is the function of call return
14:04.58Strom_Ccall the last person who called you
14:11.49*** join/#asterisk af_ (n=af@ip-131-134.sn2.eutelia.it)
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14:16.26l2cach1has anyone used munin with the asterisk_channels plugin?
14:18.47b11d|bblmorning lads
14:19.39*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)
14:20.35EmleyMoorasterisk fails to detect ring cadence on my BT line if I set it up to receive caller ID. I have tried adding a patch that works in Australia to fix this, to no avail.
14:21.01EmleyMoorAny other ideas on this?
14:21.04Omer^WARNING[23526] app_queue.c: Unknown keyword in queue '300': rtone at line 10 of queues.conf
14:21.09Omer^what does this means?
14:21.44EmleyMoorThat it doesn't know what to do with rtone in line 10 of your queues.conf?
14:22.11Omer^rtone=0
14:22.11b11dyep
14:22.30Omer^this is in my queues.conf
14:22.46EmleyMoorOmer: In line 10?
14:22.52Omer^yes
14:23.12Omer^i have made a queue number 300
14:23.13EmleyMoorIt's not understood - what is it supposed to be?
14:23.29Omer^i made it through freepbx GUI
14:23.40Omer^but i didnt add the rtone context my self
14:23.44QwellOmer^: see channel topic
14:23.52EmleyMoorAsked on #freepbx?
14:23.59b11ddoes no one read the topic, ever?
14:24.11Qwellb11d: You'd be surprised
14:24.14*** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)
14:24.21Qwellwell, maybe not
14:24.32Omer^thats why i was just asking about the context
14:24.38Omer^not the gui
14:24.48Omer^does this cotext exsit ?
14:25.05EmleyMoorIf freepbx did it, how would we know?
14:25.40Omer^i mean did you ever used this context in asterisk while making ques
14:25.56Qwellwhat context?
14:26.06Omer^rtone
14:26.07robl^Omer^: In a normal Asterisk install.. only about 4 contexts are part of the default.  In FreePBX there are dozens.. and all specfic to FreePBX.  They are needed to integrate with the GUI
14:26.09Omer^rtone=0
14:26.15Qwellthat isn't a context
14:26.29Omer^oh ok
14:26.41Strom_C~freepbx
14:26.43jbotit has been said that freepbx is the Microsoft BOB of PBXes and NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
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14:27.00EmleyMoorAnyone got any ideas on how to get asterisk to work with dring and cid on BT?
14:28.08robl^I want Asterisk 1.4 released!!!  argh!  the anticipation is killing me
14:28.19Qwellrobl^: send redbull
14:28.29b11dI see
14:28.33b11dI ask.. it gets pushed back a day
14:28.36b11d:P
14:28.49Qwellhe didn't ask.  he demanded :D
14:28.52b11dhahah
14:29.13robl^I'll send red bull, coffee, prostitutes female or male), what ever it takes.  ;_
14:29.15robl^;-)
14:29.40b11ddrunk male prostitutes are the best
14:29.41robl^first born child, maybe?
14:30.30robl^b11d: hrmmm...  *makes mental notes*  I see...
14:30.47b11dhehe
14:31.45blitzrageI look at the bugs on the bug tracker, and don't think 1.4 should be released for at least a couple more weeks
14:32.14robl^blitzrage: that bad, huh?  I just keep salivating over the new feature list
14:32.16b11dwell, i certainly agree that a release shouldnt be dictated by a date on a calendar..
14:32.43blitzrageplus its Christmas time, so I don't imagine there is going to be a feverish amount of patches
14:32.44robl^I agree too.. shouldn't be released until its ready.. but, dang, it.. make it ready!
14:32.44b11dim not even sure what the new features are :P
14:33.02blitzragerobl^: you can just use it RIGHT NOW -- a release means nothing
14:33.10blitzrageI've been using it for a month
14:33.16robl^AEL2, metermades, share line appearances..  *drool*
14:33.28b11dI keep hearing about AEL.. should I be switching to that?
14:33.39blitzrageb11d: depends what you want to do -- I don't use it yet
14:33.44b11dahh
14:33.45blitzrageSLA doesn't work
14:33.48Strom_Cmetermade?  is that like lemonade?
14:33.59blitzrageI think 'metermaid' is the more correct spelling
14:34.22*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
14:34.27robl^blitzrage: don't ask me to spell until after the 3rd coffee.  I am still on the 2nd
14:34.50blitzrageonly those who drink coffee, need coffee
14:35.08b11dno coffee yet here
14:35.09robl^I know.  I am addicted to the stuff.. evil brew.
14:35.10b11dneed it
14:35.30blitzrageI don't drink coffee... and I can get up and be productive within' 5 minutes
14:35.32robl^I can't function in the morning util I have some
14:36.05*** join/#asterisk jmesquita (n=jmesquit@201.7.117.114)
14:37.08badcfecfe is bad, but i need it
14:37.08b11dim not functioning at all
14:37.09b11dtea is better..
14:37.19badcfeuntil i get a cup, abcdef becomes badcfe
14:38.05blitzragecoffee?  beer? cof-fee? be-er? c.o...    b.e.....
14:38.16badcfewtf
14:38.30Strom_Ccombine the two and make boffee
14:38.36Strom_Ccoffeer?
14:38.39blitzrageguess no one watches Simpsons
14:38.41badcfeblitzrage: have you blended tea and coffee.  i have
14:38.52b11di totally do
14:38.53blitzragenope, I don't drink coffee
14:38.56b11dit was the aussie episode
14:39.06badcfecoffee-beer = care
14:39.16*** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it)
14:39.17b11dnothing like depressants & stimulants mixed together
14:40.05*** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it)
14:43.17*** part/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com)
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14:57.24*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:57.24*** mode/#asterisk [+o anthm] by ChanServ
14:57.50*** mode/#asterisk [+o mog] by ChanServ
14:58.05b11dyep
14:58.08b11dthat killed the conversation
15:00.40robl^nah.. I am just at work.. how dare that actually expect to work during WORK..
15:00.59b11dim at work and im on the IRC>.
15:01.01HarryRI still think the sleepers vs caffeen argument is better
15:01.07b11dI do all my real work off-regular hours..
15:01.21b11dwhen im jacked on caffeen pills
15:01.24*** join/#asterisk menace__ (i=menace@66.181.104.31)
15:01.25*** join/#asterisk santiago (n=santiago@190.24.178.30)
15:01.29djfluxI'm at work as well
15:01.40b11dwell, arent we all great members of society
15:01.40HarryRlines of pro-plus racked up on your mousepad?
15:01.43*** join/#asterisk Merlin2006 (n=me@150.2.169.217.in-addr.arpa)
15:01.52b11dtruckers choice over here
15:02.15HarryRjust wait until they discover crack-caffeen :)
15:02.25b11dhehe
15:02.56b11da lot of people around here do meth to stay up for days..
15:04.05HarryRyeah, then don't eat and slip slowly (or fairly quickly) into psycosis...
15:04.22b11dlol
15:04.25*** join/#asterisk nukio (n=deep@cpe-66-65-92-162.nyc.res.rr.com)
15:04.31*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
15:04.33b11di enjoy psycosis sans the meth
15:04.47HarryRsame, what a wonderful bunch of people we are
15:04.53djfluxI just listen to The Crystal Method
15:05.09b11dgreat band
15:05.17djfluxheck yeah! :)
15:05.25b11dthrow on High Roller and chill
15:05.32djfluxyou know it
15:06.06b11di need to go perform my manly "duties"
15:06.08b11dbe back soon
15:10.36*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)
15:10.40*** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net)
15:11.20*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)
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15:11.51b11d.e..e..es
15:11.53b11dAIRWOLF
15:11.59b11dlets go!@
15:13.41aydiosmiogreat theme song
15:13.49b11dyeah, edged out only by Dr. Who
15:14.00aydiosmionah, knightrider is a little better
15:14.05b11di disagree
15:14.11b11di think knightrider is way overhypted
15:14.15b11dhyped
15:14.22aydiosmiothe theme for knight rider is overhyped?!
15:14.29b11dyes, that, and the show itself.
15:14.37aydiosmiooh the show kinda sucked ass
15:14.43aydiosmiothe theme was great though
15:14.48b11donly that one part
15:15.02b11dwe need djflux to mix airwolf, knight rider, and dr. who..
15:15.11b11dinto one ultimate tv theme song intro
15:15.12aydiosmioand macgyver
15:15.12djfluxI can do that :)
15:15.16b11dmake it happen
15:15.36mercestesWoohoo...I wanna copy of that.
15:15.41aydiosmioI actually have all these themes
15:15.55b11dend it out with M*A*S*H
15:16.01*** join/#asterisk toxap (n=toxap@213.227.193.75)
15:16.05mercesteswhat?   fool!  no.
15:16.11aydiosmiob11d: not very dancable
15:16.20b11di just want to hear a drunken hawkeye make a pass at hotlips
15:16.29*** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com)
15:16.55mercestesyea, that was pretty much the themesong.
15:16.55b11dlol
15:17.05mercestesHave you ever heard the words to that themesong?
15:17.18b11dyeah.. i've read them
15:17.48mercestescrazy.....that there was once a day you could play something like that on Tv...and A:  it wasn't a big deal, and B:  it didn't make hoardes of kids go kill themselves.
15:18.03p0g0by johhny mathis, right?
15:18.27mercestesDidn't sound like Johny Mathis in the movie.
15:18.35aydiosmiothere was a time when personal responsibility meant something:)
15:18.39jgoo*creases brow* I have my asterisk setup, all great. I would like to prefix 99 then dial an extension direct to another asterisk PBX... which term should I grok? is it 'trunking' two boxes?
15:18.41*** join/#asterisk TheSkrill (n=pschwab@207.67.85.94)
15:18.47aydiosmioand TV wasn't a replacement for time with your kids
15:18.49p0g0I think he wrote the tune, I think the lyriks were the producres.
15:18.54p0g0*producers
15:19.29aydiosmiojgoo: just make 99 an extension and point it at the other asterisk box
15:19.31*** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net)
15:19.47*** join/#asterisk nays85 (i=nays85@got.root3d.net)
15:20.22b11dheh.. I read that as a "just get a 99 cent extension and point it"
15:20.43mercestesasterisk iax   or 'connecting two asterisk boxes via' and insert sip, iax, pri, etc.
15:21.26jgooaydiosmio...point it... just by ip?
15:21.47b11dphysically align them each to the north
15:21.57b11dgeographic north, not magnetic
15:22.18aydiosmiojgoo: yeah, but I missed your second part about the prefixing to an extension
15:22.31aydiosmiothough it should work just the same
15:22.37b11dim not so sure i like the new digg.com
15:22.50aydiosmiodigg blows, fark forever.
15:23.36aydiosmiorabble rabble rabble
15:23.39*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)
15:25.06b11dfark.. i just hate that name too much to ever go there
15:25.17jeremy_gif i just create an extension only containing Playback("vm-goodbye"). will the user calling this extension hear goodbye
15:25.24jeremy_gwhy cant i hear anything?
15:25.33b11dhmm..
15:25.36b11dwhats the console say
15:26.28jgooI think we can all say slashdot is just redundant now
15:26.34jgooin the light of digg
15:26.38Crescendo_Digg FTW.
15:27.01jgooeven their sad attempt at 'tags beta' was so painfully lame. made my laugh. anyway :-)
15:27.51b11dFTW!
15:28.00b11dyeah.. slashdot is dead..
15:28.09aydiosmioI read fark, gizmodo and make
15:28.12b11dit deserves respect though.. it was the shit for a long time
15:28.16aydiosmioand cnn if I have time
15:29.18jgoo++make
15:29.24jgoo++gizmodo
15:29.29jgoo--engaydget
15:29.33aydiosmiohaha
15:30.04jeremy_gteddy*CLI> load app_playback.so
15:30.04jeremy_gUnable to load module app_playback.so
15:30.04jeremy_g<PROTECTED>
15:30.11mercestesteddy?
15:30.25mercesteshow cute...:D
15:30.30b11dawww
15:30.36mercestesIt looks like your "playback" is fargered up...what ver of * is this?
15:30.37*** part/#asterisk santiago (n=santiago@debian/developer/santiago)
15:30.39jeremy_g:-) my operator has this
15:30.47mercestesis your operator a girl?
15:30.51jeremy_g1.4
15:30.56b11di read that as "my operator has tits"
15:31.01jeremy_g:D
15:31.04mercestes*nods*
15:31.14aydiosmio10:20 < b11d> heh.. I read that as a "just get a 99 cent extension and point it"
15:31.14mercestesmuhahaha...
15:31.17aydiosmioyou need new eyes
15:31.21b11di know :)
15:31.29b11dor that coffee we were talking about
15:31.40mercestesOr that mixed music theme to make the real world worthwhile.
15:31.46b11dhell yes
15:31.51mercestesso...back to Jeremy's operator......is she hot?
15:31.57jeremy_gvery
15:32.04b11dis this her?
15:32.04b11dhttp://zeppelin.stepahead.net/~dan/pz
15:32.09mercestesI would try a nice recompile of asterisk.  Seems like your inherent playback module, namely app_playback.so, failed to compile.
15:32.18jeremy_gawww
15:32.29jeremy_gbut i have not control over that
15:33.15jeremy_gtime to mail my operator
15:33.32b11dim going to go rig some shit up
15:33.32b11dbbl
15:39.58jgooanyone here had any joy with samba shares... I keep getting access denied :(
15:41.20djfluxI have joy with Samba shares everyday ... not related to asterisk, but otherwise, no issues ... in ADS security mode even
15:42.34jgoomy issue is, when it asks for the username, do I do machinename\user or sambaname\user ?
15:43.20djfluxdepends on what security mode you're using
15:43.30djfluxserver, domain, ads, etc
15:43.55jgooaaah. it is set to user
15:44.48*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
15:45.03santibioticoanyone using a nokia n91 with asterisk?
15:49.52*** part/#asterisk TheSkrill (n=pschwab@207.67.85.94)
15:50.31jeremy_gjgoo:smbadduser -a jgoo_samba
15:50.42ValDuaneanyone got any experience with A2Billing? I am trying to get a DID in to the ladies voice...please enter your pin number...and it aint working like trixbox works :s
15:51.48*** join/#asterisk fourcheez1 (n=rich@office.callmaster.co.uk)
15:51.48jeremy_gValDuane:i have it and i sell it.
15:52.02jeremy_gValDuane:its not #a2billing :P
15:52.04*** join/#asterisk stephane (n=stephane@gw.sortilege.net)
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15:52.40*** part/#asterisk brif8 (n=brif8@rrcs-67-78-24-179.se.biz.rr.com)
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15:54.12ValDuane:) had to try didn't I? I guess there is no help channel for A2billing then?
15:54.13tmccraryAnyone in here use any Viking paging inteface boxes?
15:54.24tmccraryI am having a busy signal occur after hang up and it's kind of annoying
15:54.43tmccraryspecifically CPA-7B
15:54.49*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
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15:56.29Rhizome;)
16:00.31*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-124.washdc.fios.verizon.net)
16:00.33SomeOne1hello
16:00.36Rhizomehm
16:00.58SomeOne1how do i get asterisk to wait for a user to dial an extension after it recieves a call
16:01.35*** part/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-124.washdc.fios.verizon.net)
16:01.38*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-124.washdc.fios.verizon.net)
16:01.58SomeOne1how do i get asterisk to wait for a user to dial an extension, and then forward/act accordingly
16:02.15SomeOne1like after it gets a number dialed to it, i want it to say a message and then be like "dial 1 for blah"
16:02.17SomeOne1"dial 2 for blah"
16:03.20[TK]D-FenderSomeOne1: http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
16:03.41[TK]D-FenderSomeOne1: And you should also definately read THE BOOK
16:03.43[TK]D-Fender~book
16:03.47jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
16:04.47*** join/#asterisk lowlevel (n=Stuart@bigbrother.vermeulens.com)
16:04.50SomeOne1heh
16:04.53SomeOne1religiously
16:05.07*** join/#asterisk marv[work] (n=timr@24.214.206.254)
16:05.22*** join/#asterisk qwertz (n=qwertz@20.206.27.217.static.versanetonline.de)
16:06.54jeremy_gUnable to load module app_playback.so
16:06.54jeremy_gDec 18 17:06:23 DEBUG[27756]: config.c:595 config_text_file_load: Parsing /etc/asterisk/modules.conf
16:06.54jeremy_gDec 18 17:06:23 WARNING[27756]: loader.c:305 __load_resource: Module 'app_playback.so' already exists
16:08.39*** join/#asterisk sweeper (i=sweeper@softcheese.net)
16:09.11qwertzHi, installed the asterisk-bristuff 1.2 version on a debian etch rc1 and made the zaptel driver using module-assistant. I've got one hfc-s isdn card in my server. when restarting I always get zaphfc loaded with a vzaphfc module. could anybody please explain what this module is good for?
16:11.43jeremy_gwhats wrong with Playback("vm-goodbye");
16:11.47*** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net)
16:11.52Qwell[]jeremy_g: quotes
16:12.03Qwell[]it would try to play "vm-goodbye".gsm
16:12.10jeremy_goh damn
16:12.12jeremy_gi remember
16:12.54sweeperI don't suppose there are any ~$100 FXS cards out there, eh?
16:13.10Qwell[]sweeper: Digium tdm400p is about that much
16:13.13Qwell[]with 1 fxs module
16:13.28sweepermmmm
16:13.53sweeperI guess a SIP ATA would end up cheaper per port, with less reliability...
16:14.06*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)
16:14.15aydiosmiohow are ATAs less reliable?
16:14.43*** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net)
16:15.17sweeperaydiosmio: consumer-grade stuffs, more points of failure, etc. not really a problem for what I'm doing (home phone stuffs(
16:15.26jeremy_gQwell:its Playback(vm-goodbye) AHHHHHHHH! wasted my 30minutes
16:15.32jeremy_git happens everytime
16:16.47sweeperis there support for stuff like RAD's TDMoIP boxes for *?
16:18.48*** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk)
16:19.14SplasPoodwas attach=yes removed or broken in recent 1.4 betas?
16:19.21SplasPoodor rather, recent 1.4 SVN
16:21.57mercestesOops, sorry Jeremy_g:  didn't catch that..lol
16:25.10*** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net)
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17:01.52doolphhi
17:02.37*** join/#asterisk Menace- (i=menace@66.181.104.31)
17:02.43*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)
17:03.47De_Monhmm
17:06.10*** join/#asterisk humbolto (n=elias@chello062178032026.11.11.vie.surfer.at)
17:06.42*** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
17:06.49doolphuh
17:06.57*** part/#asterisk keith80403 (n=keith804@24-56-189-80.co.warpdriveonline.com)
17:07.07doolphasterisk 1.4 beta4 is out :D
17:07.43*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
17:07.43SheriF_SpacEdoolph: since few days now
17:07.46*** join/#asterisk bkruse (i=bkruse@nat/digium/x-c61864d7350781ed)
17:07.51doolphreally?
17:07.54*** part/#asterisk mithraen_ (n=mithraen@87.228.121.245)
17:08.25De_Monexten => 706,1,Goto(join-1400,1)
17:08.26De_Monexten => _join-.,1,NoOp(It works!)
17:08.41De_Monthat should work right?
17:08.47n3c8is there a changelog for the beta... having difficulty tracking it down
17:09.04De_Mondid you look in docs/ ?
17:09.10De_Monor documentation i dont recall
17:09.23n3c8ill go check again
17:10.22*** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net)
17:11.08De_Monheh, I don't see one either
17:11.46De_Monhttp://svn.digium.com/view/asterisk/tags/1.4.0-beta4/CHANGES?view=markup
17:11.54n3c8kick ass
17:11.56De_Monthats the closest Ive found and it doesnt talk about the beta
17:12.01n3c8doh!
17:12.28n3c8thats anoying... how are they going to convince peopel to try it if it doesn't do anything new n funky, or fix something interesting
17:13.02De_Monhttp://svn.digium.com/view/asterisk/tags/1.4.0-beta4/ChangeLog?sortby=log&sortdir=down&view=markup
17:13.27De_Monthat should be what your looking for
17:14.09n3c8cracking thanks
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17:16.12brodiemcan't wait to try out wideband codecs in beta4
17:16.24Qwell[]brodiem: You have to have a phone that supports it
17:16.36Mouradhello
17:16.39Qwell[]pretty much the only hardphone right now is like the polycom ip650, and only in pass-through
17:16.45Qwell[]afaik anyhow
17:17.03bkrusebrodiem: ouch
17:17.09bkruseQwell: *high five*
17:18.13De_Monexten => _join-.,1,NoOp(It works!)
17:18.28De_Monthat will match join-anutying -- riiight?
17:19.06bkruseanything that dials join-
17:19.47brodiemQwell[] a damn I waas thinking it was g726
17:20.05Qwell[]brodiem: it is
17:20.07Qwell[]iirc
17:20.14De_Monit seems like _join is a special case because its not working.
17:20.22brodiemQwell[] g722-16khz
17:20.30Qwell[]umm...right
17:20.33Qwell[]that's the one
17:20.43Qwell[]726 is...
17:20.43De_Mon_-. matches -anything but _join-.  doesnt match join-anything
17:20.59blitzrage726 is liek 32kbps
17:21.00bkruseDe_Mon: oh i see what you are saying
17:21.02brodiemQwell[] the snom 360's support g726-32khz, I was thinking they wree the same
17:21.30blitzrageit 24, and 16 as well, but I think asterisk only supports the 32kbps version
17:21.34Qwell[]brodiem: a pass-through codec isn't THAT difficult to write.  g722 would be a good place to start, if you want to make a patch
17:21.48Qwell[]wait, there is already
17:22.14Qwell[]but 32kbps != 32khz
17:22.18De_Monhurm _abc-. isnt matching either WTF
17:22.44bkruseQwell: very good point kbps != khz
17:22.50bkrusei see alot of people getting that confused.
17:22.50De_Montypo it was abC
17:23.02Qwell[]if that were the case, ulaw would sound pretty good ;)
17:23.09De_Monalot != a lot either
17:23.17Qwell[]bkruse: ^ pwned
17:23.28brodiemlol I didn't catch that, I assumed g726-32 was 32khz and not kbps
17:24.06brodiemwell no wideband for a while then
17:24.09Qwell[]I'm fairly certain that g722 is the only wide-band codec we support right now.  Obviously, things like speex CAN be wide-band, but we don't support them like that yet
17:24.18brodiemyeah
17:24.37*** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux)
17:24.50bkruseQwell see what i mean :]
17:25.00*** join/#asterisk Osmor (n=osmaroc@adsl196-110-245-217-196.adsl196-16.iam.net.ma)
17:25.01brodiemQwell[], you work for digium?
17:25.05Qwell[]brodiem: yeah
17:25.09brodiemcool
17:25.19bkrusebrodiem: Qwell is digium
17:25.32bkrusethats his full name
17:25.47brodiemoh really
17:25.51De_Monnotuh file is digium qwell is just a muffin boy
17:26.08filemmm?
17:26.23bkruseDe_Mon:  i think you have that backwards :P
17:26.35*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
17:27.07*** join/#asterisk menace_ (i=menace@66.181.104.31)
17:27.11De_Monfile tell bkruse that Qwell[] gives you muffins
17:27.19filenever!
17:27.28bkruseDe_Mon: i give file muffins, and cake!
17:27.52De_Monwe'll call you muffin and cake boy then
17:28.01brodiemI have a question about SIP negotiations... if there are two * boxes -- one is acting as a SIP server w/ an exten defined as type=friend. The other * box is acting as a peer and its SIP exten is defined as type=user. Is there any reason this wouldn't work or do both SIP accounts require the same type= setting?
17:28.15brodiemer *acting as a client
17:28.40De_Monbrodiem type=friend is the same as creating a type=peer and type=user account for them
17:28.55brodiemDe_Mon yeah I know
17:29.05*** join/#asterisk Tili (n=tili@218.Red-88-0-151.dynamicIP.rima-tde.net)
17:29.06De_Monthen why did you ask the question!
17:29.15brodiembecause that wasn't the question :)
17:29.29brodiemhere is what I'm trying to do...
17:29.49blitzragebrodiem: http://bugs.digium.com/view.php?id=8565
17:29.53*** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net)
17:30.45brodiemI'm trying to register multiple SIP registrations to my SIP provider (I have multiple logins, but to the same host). From what I've read type=friend matches the SIP context based on the host name, whereas type=user matches based on the context= for that SIP registration. It seems setting type=user fails with my provider, and I'm wondering if it's because they define type=friend if that causes any problem or not?
17:31.50brodiemI also had the provider put each subacct on a different registration port for each, but didn't help
17:32.12brodiemblitzrage this is 1.2.x btw
17:32.29*** join/#asterisk matt_ (n=matt@2001:4bd0:2056:1:220:edff:feb4:7c9d)
17:32.46Osmorquestion : can i use asterisk in local network without any sip account ?
17:32.51Qwell[]Osmor: sure
17:32.54blitzrageOsmor: of course
17:33.10*** join/#asterisk jeffg (n=jeffg@pdpc/supporter/active/jeffg)
17:33.15*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
17:33.29Osmori use to but it did not work
17:34.29Osmoris there any special configuration that i have to add in sip;conf
17:34.48Osmoror somethinglike that ?
17:36.03Osmorneed help?
17:36.41*** join/#asterisk Pelipe (n=Pelipe@dslc-082-082-077-184.pools.arcor-ip.net)
17:36.56PelipeHello @ all! Any germans here?
17:37.16*** join/#asterisk Merlin2006 (n=me@150.2.169.217.in-addr.arpa)
17:37.32Osmoris there any answer please ??
17:37.53Osmoror any tuto about this?
17:37.56Merlin2006lo, anyone got any idea why asterisk isnt passing dtmf when i call digium support?
17:38.46wunderkinMerlin2006, call not answered yet? i always call directly through misery.digium.com now
17:39.48*** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net)
17:40.06Merlin2006yeah, the call is answered, but none of the buttons that i use on the softphone are being pssed to digium, which means i just get throwen out of the ivr after a bit
17:40.17*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)
17:40.21*** join/#asterisk Teeli (n=tili@61.Red-83-56-252.dynamicIP.rima-tde.net)
17:40.22Pelipedoes anybody know if there is a tutorial in german for asterisk?
17:40.57Merlin2006its prolly somethign iv missed in the config
17:41.04EmleyMoorasterisk does not recognise different ring cadences if I have cid switched on - even with the patch - but it does recognise them with CID off :-(
17:41.25*** part/#asterisk Osmor (n=osmaroc@adsl196-110-245-217-196.adsl196-16.iam.net.ma)
17:41.52hoobastoobaI am having a case of the mondays. I am doing a Playback(hold1) and it tells me my file does not exist. My file is hold1.ulaw and is in my "/var/lib/asterisk/sounds" directory. It was recorded using the exten => 205,3,Record(asterisk-recording%d:ulaw) example.
17:41.56hoobastoobawhat could i be doing wrong?
17:42.38wunderkinhoobastooba, try using .ul
17:44.23EmleyMoorDoes anyone here understand how ring cadence detection works in asterisk?
17:44.28hoobastoobawunderkin: like i said... it was a case of the mondays. ;) I was doing playback(hold1.ulaw) instead of playback(hold1)
17:45.12b11deyy ohhh
17:45.16Merlin2006stuff it, im going home, ill hit me head against this in teh morning
17:45.28b11dftw
17:46.06*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)
17:46.22pifthis bud's for you
17:46.24[TK]D-Fenderhoobastooba: Does exten => 205,3,Record(asterisk-recording%d:ulaw) look anything like "hold1" to you?
17:46.38[TK]D-Fenderhoobastooba: And whats the %d?
17:48.15*** join/#asterisk CunningPike (n=CunningP@204.239.8.149)
17:49.24*** join/#asterisk _matt (n=matt@2001:4bd0:2056:1:202:3fff:fe7e:96b1)
17:49.49b11di'll smoke that pif..
17:50.19hoobastooba[TK]D-Fender: easy... Like i said, it was a case of the mondays. Too many conversations here in my office and trying to work/think at the same time.
17:51.13hoobastooba[TK]D-Fender: %d makes the recodings named asterisk-recording0 asterisk-recording1 asterisk-recording2 etc.
17:51.26De_Monhoobastooba  you acnt work and think at the same time? that must make working hard
17:51.37[TK]D-Fenderhoobastooba: Ok, thats new to me.  I'll go look that up.
17:51.45hoobastoobaDe_Mon: WOW and the flames keep coming.
17:52.03*** join/#asterisk kannan (n=kannan@58.68.25.67)
17:52.59b11dstfu ftw rtfm etc
17:53.12kannanhello all. i have an * box thatregisters with a remote sip proxy for outbound calls when it is behind on a NAT ip, but not when it is on a public IP. Any ideas what the issue may be, Thanx for any help in advance :0
17:53.20linageewtf. why does my ping test on my voip provider always spike to 100ms+ and set off my nagios warnings? :(
17:53.35b11ddoes that only happen from the pbx?
17:53.47De_Monmy flame was completely unrelated! /sulk
17:53.54piflinagee : turn off you porn downloads
17:54.08linageepif: the nagios test box is not coming from my home connection
17:54.10*** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com)
17:54.10linagee:p
17:54.11bkrusepif: +2
17:54.15De_Monpif maybe his voip provider needs to stop downloading porn :P
17:54.37doolphhow can I send sip error message when someone call to the pbx
17:54.57linageeDe_Mon: yes
17:55.18*** join/#asterisk avalone (n=avalone_@dial-272.vl-cen-as3.avtlg.ru)
17:55.22linageeDe_Mon: how do i do that one
17:55.50De_Monlinagee call them and say quit downloading porn my ping is spiking!
17:56.13b11dim going to North Korea to start a phone compnay.. ttyl
17:56.18linageeDe_Mon: they're on the east coast. i wonder if that has anything to do with it. :(
17:56.25linagee(and i'm on the west)
17:56.29De_Mon* I will not be held responsible for the actions of anyone who follows my advice
17:57.57linageeGoogle NASA? is that what it's called now?
17:59.48b11dGASA
17:59.51b11dNOOGLE?
17:59.56wunderkinGoogle NASA Microsoft
18:00.13b11dGOOMSNASA
18:06.40*** join/#asterisk Osmor (n=osmaroc@adsl196-110-245-217-196.adsl196-16.iam.net.ma)
18:07.34*** join/#asterisk mustafa (n=mustafa@202.141.252.73)
18:07.40Osmorquestion : how can i use asterisk on local network ?
18:07.54b11dtcp/ip I'd guess
18:07.56b11dand ethernet
18:07.59brodiemOsmor that is a pretty broad question
18:08.06b11dyeah, dont get so specific :P
18:08.13brodiemlol
18:08.38brodiem"how can I use my car in my local city?"
18:08.40mustafai have 2 pda's connected with my wireless router, everything works fine..inbound, outbound etc, problem is when i try to make a call between both of PDa devices i get nothing but too much noise.
18:08.42b11dhjahahaha
18:08.45mustafai have installede sjphone on both
18:08.45b11dLOL
18:09.09b11dso Osmor..  care to get a bit more detailed?
18:09.39b11di'd guess everything is not fine then eh mustafa..
18:09.45Osmori use an ip phone and i use the local adresse of the server but it did not register ?
18:09.51tzafrirI'm looking at http://www.voip-info.org/wiki/view/UK+Asterisk+Details . Does setting "opermode=UK" have any effect on wctdm's fxs modules?
18:09.54mustafayeah actually
18:10.03b11dyeah you've got to setup your phone to register with asterisk
18:10.14b11dand you've got to setup asterisk to know about the phone which is registering with it
18:10.15mustafai have no idea  
18:10.26tzafrirThe page says it does. I believe that the code says it doesn't
18:10.28mustafaany one can help me out?
18:10.28b11dmustafa.. its just really noisy?
18:10.50b11dbut you CAN hear the person? audio is being exchanged two-way, right?
18:10.54mustafayeah
18:10.59mustafano voice just noise
18:11.04b11djust noise..  no voice..
18:11.10mustafayeah
18:11.22b11dis the phone on the same network as the pbx?
18:11.22Osmorasterisk is already setup and launched but i don t see the phone registerd on asterisk
18:11.29mustafano
18:11.30b11dOsmor "sip show peers" doesnt list it?
18:11.33b11dmustafa.. NAT?
18:11.37mustafano
18:11.39mustafano nat
18:11.41b11dgood :P
18:11.48b11dand you can ping the phone, etc?
18:11.50Osmorping ok on the phone
18:12.00b11dOsmor.. I dont care about ping.. what does "sip show peers" say?
18:12.10Osmorno peers
18:12.12b11dor are you even using sip?
18:12.22b11dhave you added the phone in sip.conf?
18:12.25brodiemOsmor if there are no peers then you haven't added the SIP extension to *
18:12.35l2cach1mustafa: you can call the outside world correct..just inter-phone locally provides noise?
18:12.38mustafayeah
18:12.47Osmoryes a added the phone on sip
18:12.50l2cach1what codecs are you using?
18:12.57b11dwell im not so sure you did
18:13.00mustafano inter phone is also ok.. but only problem is with two of these pda
18:13.03b11dsip show peers would list the entry in sip.conf
18:13.06b11deven if it was registerd or not
18:13.07brodiemOsmor if the ext was in sip.conf correctly you would see the peer listed on a sip show peers
18:13.08l2cach1ah
18:13.34l2cach1can the pda's call out fine?
18:13.38b11di'd bet its the PDA's then mustafa
18:13.38mustafayeah
18:13.46Osmori m going to edit the file right now
18:13.47mustafai can make calls to my cellfone and pstn
18:13.54aydiosmiomustafa: they aren't feeding back are they?
18:13.54mustafaand even other local extensions on my network
18:13.59l2cach1make a call between them and type "sip show channels" in your CLI
18:14.12mustafahmm ok
18:14.14mustafai will try it out
18:14.19b11ddont forget to "reload" when you're done Osmor
18:14.22l2cach1leave the call open..then type that
18:14.39b11ddont you mean "sip show channels" then?
18:14.52b11dpeers will list peers when they are in sip.conf & will show if its reg'd or not..
18:14.55mustafayeah then
18:15.08l2cach1do you see two active channels?
18:15.30mustafai will try it out tomorrow when i ll get back at work
18:15.36l2cach1both using ulaw or whatever codec you setup?
18:15.37l2cach1oh ok
18:16.12mustafai got some hints
18:16.25mustafai will check out codecs thjing
18:16.36b11dits your PDAs..
18:16.43b11ddestroy them
18:16.47b11dpurchase superior ones
18:16.53mustafano its not mine
18:16.58mustafa:P
18:17.01b11dheh
18:17.01aydiosmiothey must not survive
18:18.04robl^they are the spawn of Trapper Keeper 3000
18:18.14b11dtrapper keeper.. hybrid with mustafa
18:18.29b11dlook.. mines got a better photo of dawsons creek on the cover
18:19.18b11di suppose I should go eat lunch or something.. sigh..
18:19.59robl^just find a drunk male prostitute. ;-)
18:20.06b11dthat'd be so sweet
18:20.09Osmorre
18:20.17Osmor:)
18:20.29b11das long as his name is Smokey, i'll be alright.
18:20.36Osmori have the account on sip
18:20.53Osmorbut the command sip show peers don t work
18:20.57b11dwhy not?
18:20.59b11dwhat does it say?
18:21.06b11dyou're doing that command in the asterisk console, right?
18:21.09b11dnot at the command line
18:21.12brodiemlol
18:21.16Osmorno such command
18:21.18brodiemhaha
18:21.21b11d(root@blah) % sip show peers
18:21.21b11d:P
18:21.25brodiemOsmor: asterisk -r
18:21.38Osmori m a beginner ;)
18:21.41b11dthats cool
18:21.44brodiemno kiddin, lol
18:21.45b11dwe all were too once
18:21.57Osmoryes i a m on asterisk -r
18:22.04b11dnow "sip show peers"
18:22.04brodiemOsmor now did sip show peers
18:22.06b11dwithout the quotes
18:22.19robl^and without the: now
18:22.21*** join/#asterisk Miss-tURk[off] (n=_naCiye_@85.106.170.9)
18:22.24b11dmy favorite cli command:  !clear
18:22.35b11dbecause the GD console has no builtin "clear" command
18:22.46Osmori have only the comand show parkedcalls
18:22.53b11dwtf
18:22.55*** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net)
18:23.04b11dwhat version of Asterisk are you running?
18:23.06b11dhow did you install it
18:23.06b11d?
18:23.17docelmois Zoa alive?
18:23.24b11dZoa.. Noa..
18:23.31*** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz)
18:23.36robl^mine favorite CLI command is:  sudo dd if=/dev/random of=/dev/hda
18:23.39brodiemOsmor sip show peers
18:23.49brodiemOsmor don't forget the sip part
18:23.54b11dsudo..  pfft.. I run as root 24/7
18:23.55b11d:)
18:24.02*** join/#asterisk nvicf (n=nvicf@201.250.183.191)
18:24.08brodiemb11d sudo is a pita
18:24.21b11dyep
18:24.36b11d<--   <3  FreeBSD
18:24.53Osmorasterisk 1.2
18:24.53irqhey b11d
18:24.54b11dhey irq!
18:25.03b11dget some more ass shots?
18:25.13b11dOsmor..
18:25.17b11dtype in "sip show peers"
18:25.20b11dwhat is the freaking result???
18:25.48OsmorNo such command 'sip show' (type 'help' for help)
18:25.52b11dweird
18:25.57brodiemOsmor sip show peers
18:26.05b11dyou have one fucked up asterisk installation.
18:26.14bkruseb11d: nah, its probably the modules arent loaded.
18:26.17b11dif you dont have "sip show" commands..
18:26.18pifOsmor : you have to type "set type=gay" first
18:26.21bkruseOsmor: try to type sip (tab tab )
18:26.25b11dhehe
18:26.25bkrusedoes it show you anything?
18:26.33bkruseshow modules like chan_sip
18:26.35b11dhe may not have made the examples..
18:26.46b11dI think without that, it doesnt create the modules.conf shit
18:26.56b11dor, doesnt asterisk auto-load any available mods?
18:26.59Osmorit show nothing
18:27.09Osmorwith tab tab
18:27.09b11dwhat about "show modules"
18:27.35b11dI have 119 modules loaded over here..
18:27.47*** join/#asterisk Bananaskin (n=Bananask@user-514f6735.l4.c3.dsl.pol.co.uk)
18:27.48b11dthat seems excessive
18:27.50Osmori have only show parkedcalls
18:27.55pifdon't unload in my direction
18:28.02b11dok..  your installation is fucked..
18:28.09b11dyou need to check out your modules.conf and shit like that, iirc.
18:28.18brodiemOsmor echo "autoload=yes" >> /etc/asterisk/modules.conf
18:28.29b11dand then "reload"
18:28.36b11ddo that echo statement from the command line, not the Asterisk CLI.
18:28.36bkruseb11d: nice.
18:28.46b11d:P
18:28.52bkrusei bet his /etc/asterisk/asterisk.conf is pointing to the wrong folder......
18:28.59bkruseeg lib instead of lib64 err w/e you want
18:29.00b11dyeah that would cause this..
18:29.16bkruseive def seen it befoer
18:29.19b11ddamn asterisk and its looseness
18:29.20bkrusebefore*
18:29.24Osmorok b11d thanks
18:29.33irqb11d: no unfortunately.. i do have a shot of a girl dancing though, and she was like, doing a finishing move at the end of a song, and her boyfriend lifter her leg up "too high"
18:29.35b11ddont just thank me!
18:29.38bkruseOsmor: is this linux or bsd's(sorry for not reading)
18:29.39nvicfhi, I'm receiving this in an incoming call:  Executing Congestion("SIP/66.114.203.143-08113040", "") in new stack
18:29.39nvicf<PROTECTED>
18:29.40irqbut they're pics from a private party so i can't really share them
18:29.42irqyou'll just have to imagine
18:29.46b11dthats cool irq..
18:29.50nvicfsorry I've flooded a bit
18:30.06irqi hope she was >=18
18:30.08nvicfbut I'm getting ocupied when I try to call my asterisk from outside (voip)
18:30.13bkruse>=18 lol
18:30.13b11dthe FBI will let you know :P
18:30.15bkruseima have to use that
18:30.20*** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net)
18:30.25bkruseFREE AZN-PR0n all >=18
18:30.35b11dlets to to thailand for =<18 :P
18:30.41bkruse:P
18:30.46bkruseor the bash -ge?
18:30.53bkruse:X
18:30.56b11dsigh.. brb
18:32.07b11dahhhh I should go for lunch.. but..  aghh..
18:32.14nvicfbusy everytime
18:32.48b11dI miss CP/M
18:33.19b11dso.. who's coming to the DPRK with me to start "The Glorious Leaders VoIP Company" ?
18:33.22b11dit's a guaranteed hit
18:34.21b11dTGLVOIPC
18:34.36b11dwww.tglvoipc.nk
18:34.39Rawplayerwhy is festival on freebsd so anal?
18:35.25b11dyeah.. it is..
18:35.25robl^CP/M?!?!?!   wow.. that brings back memories.    CP/M 2.2...  no subdirs.. but we had the user command
18:35.25b11dits possible to make it work though.. i used to have it running to read text weather reports..
18:35.25b11dyeah.. those were the REAL days robl^
18:35.40b11draw.. just dont build from the ports..
18:35.47robl^"pip" was a real awesome tool
18:36.10b11dI miss punch.
18:36.27robl^ummmm..  I will say nothing.
18:36.55b11dim going to port asterisk to cp/m
18:37.35*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-52-215.red.bezeqint.net)
18:37.42b11dit makes me sick that everyone wont shut up about some stupid fight at a god damn basketball game..
18:37.46b11dyeah.. there arent any bigger issues..
18:37.54Nuggetb11d: http://macnugget.org/photos/archive-misc/osborne1
18:37.56robl^cp/m is now opensource.  but you may be better off with MP/M
18:38.16b11dahh I had one of those..  
18:38.18bkruseb11d: that was funny though
18:38.50b11dyeah
18:39.13b11dit brought back the nausea typically associated with watching Jerry Springer.
18:41.01b11dI dont care who says it..  watching two midgets fight is not amusing to me.
18:41.17*** join/#asterisk kannan (n=kannan@58.68.25.67)
18:42.00*** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650)
18:42.55kannanhello all. Again , here goes: - my asterisk registers to a sip proxy for outbound calls when on a Natted IP but not if it is on a public IP? Any ideas how i can get it to .
18:42.59b11dI should take off for the day..
18:43.12b11dget it to what?
18:43.26kannanto register when i put it on a public IP
18:43.35b11dno idea
18:43.41bkrusekannan: take out your nat=yes entry, just for kicks
18:43.42kannanme too , lol
18:43.46b11dhehe
18:44.16kannani tried that too , but it should affect the registry anyhow , if i am right?
18:44.19b11dwe're fucked here in the usa.. everyone get out while you can
18:44.20b11dhttp://www.halturnershow.com/ChinaToDumpUSDollars.html
18:44.24kannanshouldn't i meant
18:44.51kannanyou guys can beat Chine
18:44.57bkruseb11d: we are going to die, im moving to japan
18:45.18*** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net)
18:45.19b11dyep..
18:45.22b11dim there man..
18:45.34b11dlets go to Pakistan..  PakiPenguin knows his shit and I like what's going on there..  socially that is.
18:45.39b11dunless Musharraf gets assassinated..
18:45.44[TK]D-FenderWelcome to Canada, aka USA-Lite!
18:46.03b11dif it came down to it,  i'd go back to Canada and live in my off-the-grid cabin..
18:46.13b11dI can "survive" out there..
18:47.10*** join/#asterisk Holos (n=asdf@204.101.26.106)
18:47.20aydiosmiob11d: this does not look like a reputable news source
18:47.24[TK]D-Fenderb11d: We've just finished exporting our winters so life is good!
18:47.28b11di agree.. it doesnt..
18:47.30b11dbut still..
18:47.33nvicfI'm getting a busy signal when I call my asterisk, I'm registering, I'm using my interns, wth?any help?
18:47.49aydiosmiochina's economy depends too much on the US still
18:47.56aydiosmiogive it a few years
18:47.57b11dyep.. it certainly does.
18:48.05b11dthats what I find funny.. Americans are PAYING for their own demise..
18:48.24aydiosmioit's a little more complicated than that, but yes
18:48.27aydiosmioironic
18:48.44HolosI had a server running 15 phones lockup after a month of running great. The owner rebooted it and it started working again. It uses Sangoma A200 cards on IBM hardware. It's running 1.2.13, I don't see a changelog for 1.2.14 so I'm unsure if there is a bug, and we're running queues if that makes a difference, anyone have a suggestion? The error logs are clean on the server, so I'm not sure it was hardware.
18:49.00b11dhaha thanks for telling me its more complicated than that.. up to now, I thought I could account for all the details in a single sentence.
18:49.09b11d:)
18:49.50bkruseHolos: agents/queues have always had a problem, can you TRY 1.2 trunk
18:49.56aydiosmioI said that so that I didn't have to agree with your summation outright
18:50.36Holosbkruse: Is it time thing? or a random thing? Should I be rebooting it weekly?
18:50.45b11dslightly more reputable:
18:50.45b11dhttp://www.washingtonpost.com/wp-dyn/content/article/2006/12/15/AR2006121501404.html
18:51.06aydiosmiob11d: you see that ad at the left? "privatewebhosting.org for whites only"
18:51.15b11dhahahahahHAHAHAH
18:51.24aydiosmioget this
18:51.27aydiosmio"In order to have your web site hosted by us, you need to be a straight, white, non-jew."
18:51.29b11dWOW
18:51.30b11dthats fucked
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18:51.45bkruseHolos: it shouldnt HAVE to be rebooted ever.....i cannot speak for sangoma(which im guessing was the issue, since you didnt have any errors....)
18:51.56b11dactually, yes, it should..
18:52.00b11dreboot when you apply kernel fixes :)
18:52.18[TK]D-FenderSince the Bush administration won't respond effectively to China's currency manipulation, illegal subsidies, intellectual property theft and other transgressions, Congress needs to seize control over China's trade policy," Kearns said.
18:52.30[TK]D-FenderLOL.... Totalitarin BULLSHIT.
18:52.33b11dlo
18:52.34b11dl
18:52.53b11dwe really do want to rule the world..
18:53.49shido6"we"
18:53.57shido6you wont rule anything. They will.
18:54.37shido6hah, that meant the same thing.
18:54.58b11dhehehe
18:55.11b11dI agree though.. its the central bankers and their cronies..
18:55.21b11dlike Mr. Burns said on The Simpsons:  "Money is for the poor!"
18:55.32shido6LOL!
18:56.18b11drealistically though, everyone cannot live at the level of todays millionares or billionares..
18:56.26b11dthe world's resources would be virtually eliminated
18:56.27shido6I love that...
18:56.28b11dwe're too greedy
18:56.31shido6Money is for the Poor
18:56.41b11dthere some truth to that :)
18:56.48shido6it is very true
18:58.36b11dheh, I think it also speaks to our character as humans when we all say we want everyone to have the same things, and then, when opportunity strikes, we're standing on their backs shoving our success down the throats of the proletariat :P
18:59.05b11dI wonder if I can ever NOT be off-topic for more than a few mins?
18:59.23b11dmy apologies
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19:00.49ast_freakHello
19:00.56[TK]D-Fenderb11d: Joining the ever-increasing ranks of the plebians!
19:01.14b11dyeppers
19:01.27ast_freakCan someone help me... I need to know how to see the realtime queues from the CLI... is there a way to do that?
19:01.30[TK]D-Fenderb11d: Time to vote Libertarian!
19:01.35b11dim up for that..
19:01.41*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:01.45b11dor..  lets bring in a new electoral system and new parties
19:02.03b11dalso.. every 3rd election is NO INCUMBENT election..
19:02.04b11d:P
19:03.26b11dwell my tour guides have shown up from the Department of Homeland Security to take me to an all expenses paid trip to Guantanamo Bay, Cuba.
19:03.33b11dSee you in a year :)
19:04.10*** join/#asterisk fiber0pti (n=John@207.114.199.107)
19:04.38*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
19:04.42fiber0ptiDoes anyone have any experience with Snom phones? I have a couple of questions about the line buttons/presense.
19:05.01b11dI have some snoms but no experience with them
19:06.49b11dgreat site for documentaries & stuff:  http://www.jonhs.net/freemovies/
19:07.02l2cach1has anyone setup munin to monitor their asterisk system?
19:07.03b11d'twas on digg awhile back.. check it oot.
19:07.09b11dmunin eh..
19:07.11*** join/#asterisk Assid (i=assid@221.134.1.219)
19:07.11b11dnever heard of it..
19:07.26*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
19:07.38l2cach1http://munin.projects.linpro.no/
19:07.55*** join/#asterisk zotz (n=zotz@24.244.163.157)
19:08.00l2cach1it will provide a web interfase with active channels and other stats...graphed
19:08.22b11dlooks like a handy tool to suppliment vigilance
19:08.26*** join/#asterisk keyhack (n=keyhack@68.236.93.246)
19:09.08l2cach1the script auto logs in via manager.conf (telnet) and gets the info for the graphs every 5 mins.  After logging into the API manager it times out
19:09.17b11dtelnet.. i'd replace that!
19:09.25l2cach1its local..no worries
19:09.29b11dgood
19:09.35l2cach1deny=0.0.0.0 allow 127.0.0.1 so its secure
19:10.04l2cach1anyone have experience with munin? i need to get this working today
19:10.10*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
19:10.10b11dunless I compromise your box some other way.. and then get autologin locally..
19:10.32b11d"I" being the proverbial script kiddle in the middle of nowhere
19:10.37l2cach1its on a secure network...so im not worried...this is a project for my work
19:10.44b11dwhats the problem anyway?
19:10.56l2cach1telnet times out when running the script after logging in
19:11.02CunningPikeAnyone have trouble building 1.4beta4?
19:11.24b11di didnt even know beta4 was ouyt
19:11.33b11dgo here l2:  http://trac.edgewall.org/wiki/IrcChannel
19:11.41b11doh
19:11.43b11dits on freenode
19:11.44b11d#trac
19:11.45b11dgo there
19:12.02keyhackHow is the integration with TTS now?
19:12.08b11dits coming along..
19:12.17b11dok I dont know.. I just wanted to sound like a developer
19:13.09keyhackhaha
19:13.19b11dkey.. you might want to ask that in #asterisk-dev
19:13.26b11dits the developer channel
19:13.38keyhackwell, I played with Asterisk about a year ago, now I want to write a small VoIP calling system, that can play TTS, without slamming my head into the desk trying to make it happen
19:13.56irqkeyhack: you must slam head into wall. not desk.
19:13.58b11dthats cool..  ask in #asterisk-dev
19:14.13b11dand when something doesnt work, I refer you to RFC1925
19:14.24l2cach1whats in #trac ? munin support?
19:14.27b11deyep
19:14.28b11dyep
19:14.31l2cach1thanks
19:14.32*** join/#asterisk dasenjo (n=dasenjo@63.245.86.215)
19:14.33b11dnp
19:14.50b11dread Truth #1 in RFC1925
19:15.02b11dit'll answer ALL questions related to things working or not.
19:16.42b11dim surprised theres no 1.4 feature list up on asterisk.org
19:16.55b11dalthough, since its beta, it doesnt make sense to publish a feature list..
19:16.59b11dsigh
19:17.58CunningPikeb11d: The CHANGES file gives plenty of information
19:18.12b11dyeah but I have to download it :)
19:18.33b11dhahaha  
19:19.02b11dim just saying.. for newcomers.. there isnt a "read the freaking changes file for the feature list" notice on asterisk.org
19:19.17b11dits just like "woah.. 1.4 is coming out soon.. GET READY!!!!!! FEATURES!!! FTW!!!"
19:20.13CunningPikeAnyone else tried building 1.4beta4 yet?
19:20.38b11di'll do it now if you wish..
19:20.46b11dbut im on FreeBSD.. so..
19:20.53*** join/#asterisk obnauticus (i=asd@c-24-21-95-32.hsd1.wa.comcast.net)
19:20.59b11dit'd likely not provide the same results
19:21.19CunningPikeb11d: Very likely, I'd say... :-)
19:21.23b11d:)
19:23.05ast_freakAnyone worked with realtime queues?
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19:38.28[TK]D-Fenderb11d:  : That China Dumping 1 Trillion USD story has just been buried on Digg....
19:38.40Assidhey tkd !
19:38.43Assidlong time
19:39.19robin_szthey dumped 1 trillion USD, what, in landfill?
19:40.04robl^woot!  Desi now sells Aastra label strips for the 9133i phones
19:41.55[TK]D-Fenderrobin_sz: Alledged plan to cash in USE to change for Euro's, etc.
19:42.08[TK]D-Fenderrobin_sz: Basically claiming they havce no faith in its value.
19:42.14*** join/#asterisk adorah (n=admin@87.68.207.125.adsl.012.net.il)
19:42.18[TK]D-FenderAssid: Not that long...
19:42.43robin_sz[TK]D-Fender, I suspect that would please those using Euros ...
19:43.32robin_szbring me my onion bhaji .. im starving
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19:44.49Assidit hasnt ?
19:49.13*** join/#asterisk santiago (n=santiago@190.24.178.30)
19:49.22Assidso what you been upto
19:49.34Assidrobin_sz: you mean pav bhaji
19:50.45*** join/#asterisk osmor (n=osmor@adsl196-110-245-217-196.adsl196-16.iam.net.ma)
19:51.05osmorhi there
19:52.29*** join/#asterisk aadilismail (n=aadilism@202.125.143.66)
19:53.10osmorneed help on asterisk
19:53.13osmor???
19:54.23in-ptHi all
19:54.29xainhi
19:54.29CunningPike~ideas
19:54.35osmorhi
19:54.37in-ptanyone can tell me about queue statistics
19:54.38CunningPike~suggestions
19:54.39jboti heard suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and ...
19:54.57in-pti have an error saying database is empty though it is not
19:56.22*** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au)
19:56.43osmorquestion : i m using asterisk on local network and my ip phone do not register ??
19:57.43nDuffosmor: Use a packet sniffer to look for communication between the phone and the server.
19:58.48osmornDuff i di a test ping and it s OK
19:59.40*** join/#asterisk daneel__ (n=daneel@d213-103-236-241.cust.tele2.fr)
19:59.41nDuffosmor: alternately, look for any logs generated on the server (with a sufficient verbosity level) when the phone is trying to connect; if that doesn't work, fall back to the packet sniffer.
20:00.06nDuffosmor: A ping test is not that useful -- it indicates that the systems *can* communicate, but not whether they *do*.
20:00.27nDuffosmor: Use a packet sniffer to look at any actual communication (ie. SIP registration requests) between the phone and server.
20:02.15osmornDuff: ok
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20:21.26[hC]ANyone know of any sip extension limits that asterisk has before some devices may randomly lose registration?
20:21.34*** join/#asterisk h1 (n=fakhir@ool-44c69453.dyn.optonline.net)
20:22.20[hC]I have 182 sip registrations across 94  polycom phones registered, and some will randomly lose registration once in a while and I cannot figure out why
20:23.13b11dwhat do the logs say?
20:23.18b11dare you using a provisioning server?
20:23.24[hC]not a whole heck of a lot.
20:23.25[hC]yes. ftp.
20:23.25brodiem[hC] i've had similar problems too...I set all of my phones to a 10min re-register and keep qualify turned on
20:23.39[hC]I simply have qualify=yes in sip.conf
20:23.45[hC]and use the default re-registry if 1hr.
20:23.53b11di guess i'd review the logs.. maybe perform a "sip debug" on a peer that frequently loses reg..
20:23.59[hC]it was down at a minute before, and i thought that may have caused it, but it doesnt seem to have changed anything.
20:24.10b11dNAT by chance?
20:24.38[hC]Interesting as well, they have an IP601 with 3 expansion sidecars, and that thing occasionally seems to go into 'cpu overload' (the gui gets all slow and unresponsive) then requires a reboot, and sometimes reboots itself... periodically.
20:24.40[hC]b11d: no./
20:24.45*** join/#asterisk shinux__ (n=shinux@196.220.30.151)
20:25.07[hC]all direct, all PoE.
20:25.16[hC]mainly ip501's
20:26.21b11dwhat version of SIP?
20:26.25b11dis on the polys?
20:26.34b11d2.0.3 is out..
20:26.50[hC]I was running 1.6.7, and i downgraded to 1.6.6 to see if that was it.. it seems to have gotten a bit better, and i thought it had cured it.. until this morning when i got another problem report.
20:27.01[hC]I opted to go backwards first instead of forward
20:27.03[hC]just incase.
20:27.23[hC]I didnt want 2.0.x introducing something else that was worse... :)
20:27.51b11di'd try 2.0.3
20:27.54b11dit's rock solid fo rme
20:27.55b11dfor me
20:28.12[hC]did you have any issues previously, just out of curiosity?
20:28.18b11dnope
20:28.25b11dI used 1.6.7, 2.0.1, and 2.0.3 now
20:28.32[hC]Gotcha
20:28.48[hC]Ive never seen this happen before... Makes me think theres something else about hte site im not thinking about..
20:29.03*** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com)
20:29.18[hC]I have a regular extension and an 'intercom' extension configured on all phones.. the intercom extensions are set to qualify as well. maybe i will disable qualify for them, to take some load off the phones.. but i wouldnt think that would do it.
20:29.27[hC]Maybe ill try going to 2.0.3 first...
20:29.29*** join/#asterisk towelieee (n=phph@do.you.like.my.frippers.com)
20:29.37b11dgive it a shot
20:29.44b11dread the changelog I guess
20:31.03[hC]Did already.. Nothing stood out tremendously.. but you never know, what may have been changed that wouldnt have been identified as a bug, or was small and they didnt mention it..
20:31.13b11dits possible
20:31.25b11di'd run that SIP debug on a peer that freqently becomes unregged..
20:31.34[hC]yeah
20:31.40b11dand let it log the entire event.. from initial registration, to not registering anymore, to registering again
20:31.45b11dand then look at the entire thing..
20:31.47[hC]ill give that a whirl..
20:31.53[hC]it registers again when its rebooted..
20:32.00b11dhow often are they becoming unregistered?
20:32.03b11dor how quickly?>
20:32.06*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)
20:32.10b11dis it always the same phones, or random?
20:32.35[hC]it seems to be random.. there were a couple repeats.. and it happens like... once every few days
20:32.51[hC]the "reception" phone with the operator sidecars does it 1-2 times a day
20:32.55wunderkin[hC], i've had some reboot problems too.... ip430 though... in 2.0.3 there was a fix for the ip601 and using 3 sidecards... power adjustments for poe... it says you have to increase the power when using that many sidecards...
20:33.16b11dI thought they were "sidecars" not "sidecards" -- im an idiot
20:33.16[hC]wunderkin: ahhhhhhh. i had them connect an ac adapter just incase the other day, but that is a good note..
20:33.27[hC]they are sidecars. :)
20:33.32b11d:)
20:33.41*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
20:33.47b11dhave you asked Polycom?
20:33.55b11dif its a known issue
20:33.55[TK]D-Fenderb11d: Not playing with a full deck today ;)
20:34.03[hC]I have not yet no
20:34.06b11dme?  I know..
20:34.28b11dTK you always crush my spirits :)
20:34.35*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
20:34.47in-ptcan anyone please tell me why i am getting "database empty error" with asterisk guru queue stats though database is not empty??
20:35.02[TK]D-Fenderb11d: Then you'd better head to the liquor store pronto!
20:35.07b11dyou're referencing the wrong database?
20:35.18b11ddone and done!  I'm on my way!
20:35.28b11din fact, I thinnk my ex-boss stashed a vodka bottle around here somewhere
20:35.47*** join/#asterisk Tili (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net)
20:35.57*** join/#asterisk converx (n=locid@206-248-129-148.dsl.teksavvy.com)
20:36.17converxhow  can change the default voicemail greeting?
20:36.39b11dreplace vm-greeting.gsm ?
20:37.52[TK]D-Fenderconverx: Have your VM box user actually set one, and tell Voicemail to skip instructions. "show application voicemail"
20:39.11b11dI wish there was an easy way to disable the "advanced options" and "folders" bullshit in the VM.
20:39.20b11dthat confuses my end users EVERY TIME
20:39.39in-ptb11d: the db which i had setup for it ..i had configured in its confguration also
20:39.52*** join/#asterisk menace_ (i=menace@66.181.104.31)
20:39.53b11dweird
20:39.57in-ptb11d: how to debug it any clue will be very helpfull
20:40.14b11dim thinking
20:40.20in-ptok
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20:41.15b11dno idea
20:41.19b11dare you using trixbox?
20:41.24in-ptno
20:41.28b11dok
20:41.32in-pti am using asterisk-1.2.4
20:41.51sweeperweee asterisk
20:41.51b11dasterisk 1.2.13 is out eh
20:42.09in-ptya but i faced some problem with sccp channels so thought to stay with old one
20:42.24b11dis the DB error listed in the changelog between 1.2.4 and 1.2.13 ?
20:42.29b11dmaybe its a bug that was repaired?
20:42.35b11dor.. maybe your DATABASE IS EMPTY
20:42.52in-ptsee i have two instance of queue stats on asterisk-1.2.4
20:42.58b11dtry adding records to it.. maybe your DB wont return "empty" errors..
20:43.14in-ptthe old one is working fine but the new one dont work
20:43.41*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
20:43.42in-pti have values in queue log file rotated by asterisk
20:43.47b11dwell, I dont know then man.. sorry..
20:43.51*** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago)
20:43.53in-ptand q stats also recognising that file
20:43.59in-ptok
20:44.01b11dpermissions are ok?
20:44.14in-ptwhat type of permissions
20:44.20b11ddb permissions, file permissions, etc..
20:44.35in-pti just created default db
20:44.46b11ddont you usually have to allow users access to that stuff?
20:44.55b11dor set a password on the db, etc?
20:45.15in-ptya i have to check that
20:45.34[TK]D-Fenderin-pt: When you say "queue log rotated by *", that implies you're using CSV, no?
20:45.47b11dif you cant read the db, and the db is appropaitely populated, then i'd wager its permissions
20:46.27ltersanyone here do featdmf?
20:46.46in-ptok
20:46.47b11dI miss CP/M
20:46.51in-ptthnx b11d
20:47.02b11dnp
20:47.09b11di hope it helps
20:47.15b11dlet us know what it was :)
20:47.32[hC]yay, being escalated to the voip support group at polycom
20:47.40[hC]its nice being an authorized reseller, they actually help you
20:47.41[hC]:)
20:48.02robl^Polycom is evil unless you are a reseller
20:48.04b11dtell them to release a mod for the 501's that allows for backlit displays and an adjustable seat..
20:48.12*** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
20:48.41*** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago)
20:50.05[TK]D-Fenderb11d: IP 650, and TFB (respectively)
20:50.29b11dyeah im aware of the new ones
20:51.03b11dor.. im going to start a company that sells new seats for poly phones..
20:51.08b11dI hate a non-adjustable seat..
20:51.28[TK]D-Fenderb11d: What you need is... and adustable DESK :)
20:51.29b11dalso, I'd like to eat dinner with David Suzuki ASAP.. hes the man
20:51.35b11dthat is correct TK..
20:52.02djfluxcan anyone suggest to whom I should file a bug?  DTMF while in a call on Ekiga 2.0.2 shipped with FC6 works with * 1.2.13 but not 1.4.0-beta3
20:52.19Strom_Cdjflux: try 1.4.0-beta4
20:52.19b11dbugs.asterisk.org ?
20:52.21b11dI think thats it
20:52.27Strom_Cbugs.digium.com IIRC
20:52.27djfluxStrom_C: k
20:52.31b11dnope..
20:52.33b11dim wrong
20:52.46b11dyou're right Strom_C
20:52.58djfluxI know, but another SIP softphone works so I can see filing with both Digium as well as the Ekiga team
20:53.11b11dhows that mix coming?
20:53.12djfluxjust don't know who the bug should go to
20:53.15djfluxLOL
20:53.30djfluxwell I'm at work so they kinda frown on working on DJ stuff while here
20:53.48b11dI want to be chilling to DrAirRider
20:54.11djfluxI don't think my boss would be too happy seeing me at my desk with my DJ headphones on ;)
20:54.22b11dhe would when he heard what you are working on
20:54.39b11dmaybe you could do some work with Sasha on this?
20:54.40djfluxhaha ... probably ... he's down with 80's music so I'm sure he'd like it
20:54.51djfluxmy boss that is
20:54.56b11dyeqah
20:54.56djfluxdon't know about Sasha
20:55.01b11dSasha is good..
20:55.05b11dI enjoyed Airdrawndagger
20:55.07djfluxyep
20:55.27*** join/#asterisk santiago__ (n=santiago@190.24.179.121)
20:56.12b11dfuck packages..
20:56.16b11dSOURCE ALL THE WAY
20:56.37djfluxI can build daily packages from svn
20:56.39b11dalthough i'll grant you license to do that, because you need your spare time to make that mix..
20:57.01djfluxpackages are just cleaner for me
20:57.14b11dwhatever you can do to rationalize it, go for it..
20:57.15b11d:P
20:57.19djflux:)
20:57.30djfluxpackages are build from source ;)
20:57.37djfluxs/build/built
20:57.41b11dyou might like this:
20:57.42b11dhttp://video.google.com/videoplay?docid=-7085594575841814301
20:57.48b11d"scratch" The Movie
20:58.06b11dA feature-length documentary film about hip-hop DJing, otherwise known as turntablism. From the South Bronx in the 1970s to San Francisco now, the world's best scratchers, beat-diggers, party-rockers, and producers wax poetic on beats, breaks, battles, and the infinite possibilities of vinyl
20:58.33djfluxb11d: already have it on DVD
20:58.41b11dcool.. is it good?  I just downloaded it..
20:58.42djfluxhad it for quite sometime
20:58.49djfluxyeah ... it's awesome
20:58.53b11dexcellent..
20:58.57b11dI look forward to watching it tonight
20:59.03b11dI also didnt mind the "Beef" movies..
20:59.18b11dthe east coast/west coast thing always brought a smile to my face for unknown reasons
21:00.53Strom_CI'm still just getting the hang of beatmatching :)
21:01.30djfluxqbert is CRAZY!  I've been DJing for like 20 years ... Cash Money was the guy who made me want to learn ... scratching with "It's tiiiiiiime" :)
21:01.58*** join/#asterisk Rahail (n=Rahail@209-19-88-240.detroit.mi.D-Conn.net)
21:02.09Rahailany one here to do a small configuration for me
21:02.31b11dman those Freestyle rappers are just amazing..
21:03.01b11dwefunk's DJ Static does some just phenomenal beatmatching..
21:03.06b11dhis cuts are just like..  smooth as silk
21:03.13djfluxyup
21:03.33b11dyou listen to wefunk?
21:03.54*** join/#asterisk CleanerX (n=nix@p54A398B6.dip0.t-ipconnect.de)
21:04.05b11dits a great show..  DJ Static will hook you up with a private download site if you ask nicely :)
21:04.48djfluxI've listened to a little bit ... not much though
21:05.00djfluxthat'd be cool ... I'll have to email :)
21:05.02djfluxthanks
21:05.33[hC]anyone used the dlink dph 540 wifi phone?
21:05.43[hC]I just got one in to replace some wip300's that are god damn terrible.
21:07.05*** join/#asterisk Omer^ (n=omer@202.142.150.12)
21:07.17b11dnp..
21:07.29b11dthey find some pretty (awesome) rare tracks too to play..
21:07.33SkramXanyone played with Cisco 7970(G)'s?
21:07.42[hC]I have a ton of em.
21:07.44*** join/#asterisk obnauticus (i=asd@c-24-21-95-32.hsd1.mn.comcast.net)
21:07.45[hC]7970's
21:07.59b11doh.. I was hoping you had a tone of pretty (awesome) rare tracks for me to download :)
21:08.06b11dtone = ton
21:08.07b11dor tonne
21:08.08b11d:P
21:08.22SkramX[hC]: we are talking Cisco IP phones, right?
21:09.19[hC]SkramX: absolutely.
21:09.19b11dwell that killed the conversation, didnt it?
21:09.21b11doh
21:10.00SkramX[hC]: Cool... so, I assume you have reflashed them or whatever to work w/ SIP (and Asterisk).. have you written apps for them via XML, etc.?
21:10.08[hC]SkramX: I use SCCP w/ them.
21:10.14SkramXOh okay
21:10.15[hC]SkramX: the SIP firmware was horrrrrible...
21:10.17SkramXRight on
21:10.20SkramXjust curious
21:10.23[hC]SkramX: infact, i could not get it to register.
21:10.26SkramXso... what about apps?
21:10.44SkramXthey still don't do HTML or WAP, do they? Just that Cisco XML craziness?
21:10.44[hC]SkramX: ive only used the pre written cisco79xxdir or whatever it was called..
21:11.09[hC]SkramX: its on sourceforge i think.. contact directory system... ive also written a very small app to auto refresh a security camera image
21:11.11SkramXhow does that work? my impression from some reading was that you need CCM
21:11.17b11dCCM = SHIT
21:11.18*** join/#asterisk backblue (n=moo@87-196-12-18.net.novis.pt)
21:11.20SkramXright!
21:11.22b11dok.. im just saying that without any experience..
21:11.25SkramXwell, i've never used it
21:11.26b11dbut I'll bet it sucks
21:11.26SkramXyeah :P
21:11.27[hC]SkramX: you point it to a web server that serves up xml
21:11.42SkramX[hC]: okay.. so how do you tell it the WWW server?
21:11.49[hC]SkramX: in its config file.
21:11.50SkramXsccp configs? actual phone configs?
21:11.56[hC]phone configs.
21:11.59[hC]on the tftp server.
21:12.02SkramXok
21:12.04SkramXcool.
21:12.09b11dtftp > *
21:12.09SkramXoverall impressions of the phone?
21:12.12[hC]there's a ton of info about it on voip-info and chan-sccp.org
21:12.22[hC]SkramX: i use it as my deskphone daily. its only slightly buggy.
21:12.24*** join/#asterisk santiago_ (n=santiago@190.24.179.198)
21:12.31SkramXI work with a WebDev company and we are thinking about getting maybe one or two so I can dev with it
21:12.37[hC]SkramX: overall, its nice. it has its 3 or 4 stupid bugs that id like to get rid of, but...
21:12.39SkramXmost interested in the application standpoint
21:12.43SkramXHrmm
21:12.51SkramXand.. did you all pay the $500 for each or what?
21:13.13b11dthats my issue with the cisco phones..  just not worth the cost when compared to the polycoms
21:13.17[hC]SkramX: i think i pay 4 something for them.
21:13.22SkramXok
21:13.27SkramXremember where you purchased them?
21:13.37[hC]I can get 7940's for 120 bucks though ;)
21:13.39SkramXb11d: polycoms.. do any of them have screens I can write apps for?
21:13.46b11das far as I know you can do it..
21:13.50SkramXright...  but they aren't pretty and lack touch screens, IIRC
21:13.51b11dbut im no polycom expert in that respect
21:14.01SkramXright..
21:14.07b11dask [TK]D-Fender
21:14.10[hC]SkramX: hmm.... i have a cisco reseller dude.
21:14.14[hC]in canada.
21:14.23[hC]i have also bought from voip supply
21:14.30SkramXokay
21:14.32SkramXjust curious
21:14.45[hC]hahahaah this dlink wifi phone plays a song every time you OPEN OR CLOSE the clamshell
21:14.47[hC]every time!!!
21:14.52SkramXwe don't deal with voip at this company yet.. so we'd probably buy our first few from voipseller.com
21:14.56SkramXdamn.
21:14.57SkramXheh
21:15.08SkramX[hC]: which model do you have (dlink wifi)
21:15.44[hC]dph-540
21:15.55SkramXokay
21:16.05SkramXhmm
21:16.22SkramXI have a giftcard to tigerdirect.com and they sell those for $200..
21:16.30SkramXthey sell no other ip phones pretty much.
21:16.38SkramX$3500 to tigerdirect.com and I want a macbook
21:16.50SkramX(which they don't sell)
21:17.10[hC]Im gonna go pick one up tomorrow i think
21:17.12[hC]macbook.
21:17.13[hC]:)
21:17.20SkramXyummy
21:17.29SkramXI'm a student so don't have so much money to blow
21:17.48SkramXnot to be annoying but.. impressions on the dph-540?
21:17.50[hC]I have a mbp already but im getting one for my wife
21:17.51Nuggetyay macbooks.
21:17.55SkramXyeah
21:18.05[hC]SkramX: ive had this dph 540 for about 3 minutes now.
21:18.08[hC]ups guy just dropped it off
21:18.09SkramXI am looking at a refurbed MBP.. or I may hold out for Leopard
21:18.13SkramXoh-- right on
21:18.24[hC]Ive used the linksys wip300, utstarcom f1000, and now this dph 540
21:18.28[hC]oh and the zyxel thing.
21:18.32SkramXright on
21:19.12*** join/#asterisk DaeJeon-Newbie (i=Singh@124.62.150.38)
21:20.05b11dI love you guys
21:20.07*** join/#asterisk VoipMasta (n=fabio@201.139.139.127.cableonline.com.mx)
21:20.22VoipMastaHi there
21:20.32b11dhigh
21:20.41DaeJeon-Newbiei want to configure flite (TTS-ENGINE) with asterisk 1.4 beta. any apt documentation?
21:20.47[hC]the dph 540 and the wip300 have the same firmware... the dph 540 is at 1.0.12 and the linksys is at 1.00.08
21:20.49[hC]interesting.
21:21.06[hC]I wonder who licensed it to who..
21:21.13[hC]or if they both licensed it from someone else.
21:21.13VoipMastaI'm having an issue I can't solve... When I receive a call on a DID it's recorded in the CDR as it's supposed to, but if I dial a new number to redirect that call it's not stored... what can I do?
21:21.58VoipMastaI need to keep track of incoming (through a DID) and outgoing calls
21:22.44VoipMastawould forkcdr() solve it?
21:23.03b11dany similar bug reports posted?
21:24.10luke-jrAnyone tried using google's Click to Call feature for regular dialing? :p
21:24.40*** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk)
21:25.41*** join/#asterisk aao_pwner|oper (i=asd@c-24-21-95-32.hsd1.wa.comcast.net)
21:26.56fiber0ptiIs there a way to seek through audio files in Asterisk. Example: your listening to a message and you push a key to seek 10 seconds forward..
21:26.57*** join/#asterisk bkw__ (n=ASSERTKI@c-68-32-112-142.hsd1.md.comcast.net)
21:27.23b11dive heard of that.. but I cant remember where or when I read about it
21:31.07*** join/#asterisk dacleric (n=dacleric@p548231C9.dip0.t-ipconnect.de)
21:32.25b11dugh..
21:32.27b11di hate this feeling..
21:32.36b11dwhen you dont eat breakfast or lunch and just drank coffee all day
21:32.48b11dim all jittery with a pseudo-headache..
21:32.56b11dand that hungry-but-not-hungry feeling..
21:33.16[hC]im on my way to that at the moment
21:33.29b11denjoy it :)
21:33.57*** join/#asterisk Gankhuu (n=gankhuu@72-166-51-162.dia.static.qwest.net)
21:34.25Gankhuuanybody have any experience with IBM X346 server with Asterisk on it?
21:34.39b11dI do not..  but whats the problem?
21:35.05GankhuuNot a problem... just looking for a stable server platform to implement my asterisk deployment
21:35.09b11dahh
21:35.19b11dso the question is.. what OS do you want to run then
21:35.25GankhuuAlso want opinions on switches to use with snom phone using PoE
21:35.28*** join/#asterisk DaveCanoe (n=Dave@vgateway.libertyrms.info)
21:35.59GankhuuI was thinking a Red Hat OS possibly
21:35.59b11dthey work.. thats about it :)
21:35.59b11dPoE is nice..
21:36.05*** join/#asterisk variable_office (n=variable@cerberus.iswan.net)
21:36.06b11dI wish they could make PoE work across fiber optic cable :)
21:36.30Gankhuuany particular brands people prefer for PoE switches in conjunction with asterisk and VOIP phones
21:36.32GankhuuLOL
21:36.41b11dI like my Cisco.. but thats a standard response..
21:36.56b11di'm using the inline PoE injectors though for my Polycoms..
21:37.11GankhuuPolycom phone?
21:37.19b11dyeah.. I went with Polycom's for my deployment
21:37.31b11di've got one snom 190 at the PBX console.. thats it.
21:37.36l2cach1Polycom all the way
21:37.45b11dand a few Cisco 7914 and 7940's for testing..
21:37.49b11dfor another campus
21:37.59GankhuuI have been looking at phones, but I am still pretty much undecided other than someone suggested snom to me
21:38.00b11dthey bought 147 7940's for like $90k
21:38.03b11ddumbassess
21:38.11GankhuuI don't like the Cisco ones
21:38.19b11dSnoms are OK, but generally, the Polycom's are the favored phones around here
21:38.23Gankhuutoo pricey and heard of firmware problems
21:38.29Gankhuuwhere is here?
21:38.36b11dyou'll hear of firmware issues with ANY vendor :)
21:38.39b11dhere is #asterisk
21:38.55RhizomeAnyone know of a mobile phone that isn't nokia with a sip client?
21:39.07b11di'd like that info too Rhizome..
21:39.32Gankhuuwhat do you like so much about polycoms?
21:39.42b11dlow cost, high reliability, great sound quality, easy to program..
21:39.45b11dand works with asterisk..
21:39.53b11dreasonable support..
21:40.16b11dthe ONLY things that bother me is the lack of a backlit display, and you cant adjust the stand for the phone..
21:40.22b11dwhich at the end of the day, are pretty minor inconveniences..
21:40.24b11dnot show stoppers.
21:40.56b11dthey also release firmware updates pretty regularly, and they are reasonably documented as well
21:41.00GankhuuRight now our system is a tadiran system
21:41.09b11dnever heard of them..
21:41.23sweepermmm
21:41.23GankhuuI think they are an israeli company
21:41.27b11dcool
21:41.51b11dI put on a Napalm Death album as mine..
21:41.59b11dGRAAH!HHHH!H!H!H!H!H!HHAHDHDSHDHHAHHHHHHHHHHHHHHHHHHhhhhhhhhhhhhH
21:42.20b11dit sounds like the klingon rite of ascention..
21:44.29*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
21:44.47b11dI should have bought another vg224..
21:44.53b11dor maybe a vg248 instead..
21:45.34Gankhuuso if I were to look at a comparable model for Polycom, what would be a comparable model?
21:45.42b11dthe 501 is the "standard"
21:45.48b11d301 is the low end, and the 601 is the "big guy"
21:45.53b11dtheres some newer models coming out now..
21:45.57b11dlike the 430 (iirc)
21:46.01b11dits nice, from what I hear..
21:46.14Gankhuulooking at the spec sheet for 501 now
21:46.14b11d501's go for about $145/ea
21:46.25b11dwith PoE injector adapter
21:46.28Gankhuuthey pretty durable?
21:46.35b11di've found mine to be very reliable
21:46.39b11dbut im not throwing them around much
21:46.45GankhuuI
21:46.57GankhuuI am saying from a heavy use criteria
21:47.03b11doh.. yeah they are solid
21:47.03Gankhuunot abuse
21:47.11b11dI kept a single call open between two 501's for over six days straight
21:47.18b11djust to see if it would keep it open..
21:47.19b11dit did..
21:47.25b11dno loss
21:47.28Gankhuudo the Polycoms do SIP only or do they do IAX too?
21:47.35b11dIAX is an asterisk thing
21:47.40b11dits not for the phones individually
21:47.47Gankhuuok
21:47.53b11dno phone runs "IAX"
21:47.57sweepernot yet >.>
21:47.58Gankhuuonly softphones then
21:47.59bkruseb11d: not true
21:48.01b11doh really?
21:48.07bkruseb11d: i know some chinee phones that do
21:48.08bkruseor.....
21:48.09bkruseclaim to
21:48.12b11dthats cool..  i didnt know that..
21:48.19bkruseha, i wouldnt trust it though :X
21:48.24b11dyeah :PP
21:48.26bkrusesaw one called "ETHERNET PHONE"
21:48.28GankhuuI really like the IAX from my little testing
21:48.31bkruseSUPPORTS IAX
21:48.39b11dGankhuu.. you'll see shit like "SIP, MGCP, SKINNY" etc..
21:48.43bkruseGankhuu: iax is amazing, i use it for anything and everything you can think of
21:48.51bkruseb11d: skinny, BLEH :P
21:48.54b11dyep..
21:48.56b11dagreed :)
21:49.08b11dI like SIP.. it is promising..
21:49.13GankhuuI was mostly looking for SIP phones utilizing the ulaw
21:49.27bkrusewasnt designed for phones initially from what i hear, but its deffinitely widely implemented
21:49.31sweeperSIP is ok in theory, it's just that people implement it shittily :/
21:49.31b11dthe poly's do g711 ulaw..
21:49.35GankhuuI also need to start researching PoE switches... I was thinking about Cisco.
21:49.36bkruseGankhuu: then sip is the way to go
21:49.38b11dthat what im talking about sweeper..
21:49.39bkrusepolycom's rock to
21:49.45b11dCisco's PoE works great
21:49.49bkruseand Gankhuu adtran does POE switches and polycom has POE phones
21:49.55b11dI dont trust PoE though.. cant explain why.
21:49.58b11dI went with the PoE injectors..
21:50.03l2cach1we use g729a with all our poly's
21:50.09sweeperjust wire up a patch panel :/
21:50.13b11dl2.. why?
21:50.21l2cach1save bandwidth
21:50.24Gankhuuwell, we have outlet shortage... so seemed like a good answer
21:50.27b11disnt g711 like max 10k/sec ?
21:50.31bkrusel2cach1: sweet, how much is it costing you
21:50.33b11dwhat does g729 run?
21:50.46l2cach1believe its 10bucks/license
21:50.47sweeperbkruse: or wire a 12v psu into your patch panel on the right pins
21:50.52bkruseb11d: kilobites or bytes?
21:50.57b11dkb
21:50.57bkrusesweeper: yep ;]
21:51.01b11dnot kind bud
21:51.02b11d:P
21:51.02sweeperfacking cheap, facking reliable
21:51.04bkrusesweeper: thats exactly what a poe switch does.
21:51.11sweeperbut for like.... $500 more :D
21:51.12bkrusehehe
21:51.25bkruseya, when they are there for my disposal, i use them
21:51.49bkruseb11d: i think g711 ulaw with overhead is more like 80 kbits
21:51.53b11dGankhuu.. i'd seriously consider Polycom phones w/ the inline poe injectors..
21:52.01b11d80kbit??
21:52.07b11dwow..
21:52.08*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
21:52.12bkrusei thought so.......
21:52.14bkruseim almost positive
21:52.17b11dthats fucked
21:52.25b11dkiloBIT or kiloBYTE ?
21:52.29bkruseBIT
21:52.31b11dok
21:52.31djfluxkbit
21:52.33b11dthen that works out
21:52.35bkruseya.
21:52.41djfluxthat's about right
21:52.43b11dyep
21:52.49bkruseGankhuu: maybe iax one day :D    but ya, polycoms rock!
21:52.54b11dwhat kind of b/w util is on the g729?
21:53.06bkrusei have a soundstation here, and it does g729a/b
21:53.09*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)
21:53.20djfluxlike 8kb or something
21:53.27b11dthat SP IP4000 is a nice conf. phone..
21:53.39b11dim happy with them.. $$$$ though
21:54.03bkruse8 kiloBITES, ya, its awesome
21:54.06bkruseand sound quality is quite awesome
21:54.12bkrusevery $$$
21:54.14b11dwell.. of the 300+ endpoints im installing here, i've not seen more than 15 "in use" at any one time..
21:54.15[TK]D-FenderSP 2W + ATA is also a great idea, and more portable as its analog + wireless
21:54.19b11dso im going to stick with g711 for now..
21:54.20b11d:)
21:54.29bkruseb11d: ya, but thats even more of a reason to do g729
21:54.33b11dwhy?
21:54.35bkrusesave even more bandwith, dont need many license
21:54.42b11dbut ive got so much bandwidth it doesnt matter
21:54.43b11d:)
21:54.50bkrusehehe, k :]
21:54.51bkrusegbit?
21:54.54b11dim rocking four oc-3's here
21:54.58bkruse!
21:55.00bkrusenvm
21:55.07bkruseg729 would be a waste :P
21:55.10b11d:)
21:55.14b11dstill..informative.. thanks :)
21:55.23bkrusehehe, np
21:55.34b11dsteal one :)
21:55.38*** join/#asterisk heartones (n=Loveocea@84.36.10.116)
21:55.43b11dno one will notice you hauling your own fiber over rooftops
21:55.44b11d:)
21:56.47b11dso.. the licensing for that codec works on-demand, not per-phone?
21:56.57*** join/#asterisk backblue (n=moo@87-196-12-18.net.novis.pt)
21:57.00b11dso if i only had 15 endpoints using it at any one time, all I would need is 15 licences?
21:57.06b11dnot 300 licences to cover 300 phones?
21:58.06heartonesany one connected NGN gateway (emg 202) ata FXS/FXO/RJ45 to asterisk box
21:58.42heartonesor have an idea how to configure it in zaptel.conf
21:59.43b11dnot I
21:59.54*** join/#asterisk redondos (n=redondos@190.48.17.85)
22:00.09redondosHello. How can I record every outgoing call?
22:00.31b11drecord() ?
22:00.37Supaplexmonitor application
22:00.47VoipMastadoes anyone know how to use Set(CDR(userfield)) in an AGI?
22:01.03*** join/#asterisk resistance (n=dwayne@64-42-247-120.mb.skyweb.ca)
22:01.16b11dno, but this AGI stuff is beginning to intrigue me
22:01.23resistanceasterisk intercom is giving me a huge amunt of feedback
22:01.36resistancei've adjusted rx tx gain
22:01.44resistanceany other suggestions?
22:01.51resistancei can't hear myself talking
22:01.57redondosb11d: Exactly, I want to use it with AGI.
22:02.02redondoserr, ARI I guess
22:02.06redondosSorry got confused
22:02.13*** join/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a)
22:02.15redondosToo meny acronyms for today :/
22:02.21redondosmany*
22:02.22macTijnmany
22:02.25macTijn:)
22:02.27redondos:P
22:02.28b11dI love you guys
22:02.45redondoshow come
22:02.56*** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net)
22:03.06redondosSo, I should use record() on my default context?
22:03.15b11dnah
22:03.15redondosHow to name the files? How to separate them by user?
22:03.23b11dim thinking backwards..
22:03.35b11dyou want to record all outbound calls eh
22:03.39[TK]D-Fenderredondos : Use it anywhere you dial.
22:03.39redondosYeah
22:03.59[TK]D-Fenderredondos : You should know where to put this stuff already.  its YOUR dialplan.
22:04.08redondosok, I understand.
22:04.10redondos[TK]D-Fender: ok, and abuot naming the files and separating them by users, how can that be done?
22:04.15*** join/#asterisk Winkie (n=urmom@87-194-8-125.bethere.co.uk)
22:04.45b11dknow what I love? Local TV station "STORM TEAMS"
22:04.46b11dthose are the best
22:05.03b11dlike come on.. they all have their sleeves rolled up like they've been "figuring" stuff out all night
22:05.04*** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org)
22:05.10b11dtracking the storms..
22:05.28*** join/#asterisk aao_pwner|oper (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net)
22:05.29Strom_Cb11d: so you're from southern california too, eh?
22:05.44b11dnope
22:05.46resistanceis soundblaster live a good choice for intercom sound?
22:05.47b11dnorthern minnesota :)
22:06.04b11dlike they arent just feeding off of NOAA's teat..
22:06.06Strom_Cah, I thought insane news coverage of storms was a southern california thing
22:06.09b11dhopeing they'll throw some scrap to the storm teams..
22:06.18VoipMasta[TK]D-Fender: Do you know how to set the userfield in the CDR from an AGI script?
22:06.43b11dyou better tune in to channel 10 tonight for STORM TEAM COVERAGE!!
22:06.46[TK]D-Fenderredondos : Try using the CALLERID in the finename, as well as the EXTEN dialed, and the TIMESTAMP perhaps. (*HINT*)
22:06.49b11dbecause you NEED the channel 10 storm team..
22:06.55b11dthey are your ONLY HOPE OF SURVIVAL
22:07.00*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:07.07Supaplexwe make way to destruction
22:07.12[TK]D-FenderVoipMasta : Not offhand
22:07.14b11dhehe
22:07.21Strom_CI just added a cool feature to my client's PBX - now anyone can automagically blacklist the last number which called his or her extension
22:07.22redondos[TK]D-Fender: Great! Thanks.
22:07.25*** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com)
22:07.29djfluxStrom_C: k ... Ekiga DTMF still doesn't work with asterisk 1.4.0-beta4  ... bug report suggestions?  I can provide Wireshark traces from Ekiga as well as Twinkle
22:07.36b11dthat'll be dangerous Strom_C
22:07.41harlequin516Anyone have advice on how to get DTMF from a SPA-2102 ?
22:07.43Strom_Cdjflux: bugs.digium.com
22:07.44filedjflux: rtp debug in a bug note on bugs.digium.com
22:07.46b11dI hope you included a way to make them review the list of blacklisted numbers..
22:07.50b11dand gave them a way to whitelist them :)
22:07.58[TK]D-Fenderharlequin516 : use SIP-INFO.
22:08.03djfluxcool ... thanks
22:08.06hmmhesays[TK]D-Fender: have you ever seen this rock discipline dvd john petrucci did?
22:08.16[TK]D-Fenderharlequin516 : And set "dtmfmode=info" for the peer/user
22:08.20Strom_Cb11d: I can manually un-blacklist things, plus only a few people know about the blacklist VSC
22:08.28resistancewhen i dial into my intercom i can hear a crackling on the line, and on the speaker
22:08.37b11dahh
22:08.54[TK]D-Fenderhmmhesays : Nope.... I like DreamTheater & Liquid Tension Experiment, but find him rather "stacatto" personally.
22:09.20hmmhesaysthis instructional video he did is rather good though
22:09.30[TK]D-Fenderhmmhesays : I own no vids and only about 6 books of any kind, most of which I've hardly even looked at :)
22:09.44hmmhesayswell i'm "previewing" this one
22:09.44b11dare you a proficient musician?
22:10.26b11dheh.. perfect pitch is a curse..
22:10.40hmmhesaysi would say [TK]D-Fender is from the mp3's i've heard, as for myself, i'm no slash or stevie ray but I can hold my own
22:10.58b11dthats cool..
22:11.09b11dyeah the music store guy told me to fuck off..
22:11.19hmmhesaysi've been working on advancing my technical proficiency lately
22:11.22*** join/#asterisk FusEion (i=fuse@adsl-69-234-137-101.dsl.irvnca.pacbell.net)
22:11.22hmmhesayswhat?
22:11.24FusEionDCC SEND 39tuio23jgi23gj9i2gj92jgio32g42g
22:11.28*** part/#asterisk FusEion (i=fuse@adsl-69-234-137-101.dsl.irvnca.pacbell.net)
22:11.38*** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net)
22:11.45b11dwhats that some new elite exploit :P
22:11.47*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)
22:11.48*** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net)
22:11.53b11dDCC SEND 39tuio23jgi23gj9i2gj92jgio32g42g
22:11.55b11dHAHAHAHA
22:12.00b11dthats great!
22:12.10*** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net)
22:12.15b11dwtf eh
22:12.15*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)
22:12.16*** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net)
22:12.17*** mode/#asterisk [+b *!*n=noway@*.hcc.mnscu.edu] by Qwell[]
22:12.17*** kick/#asterisk [b11d!i=qwell@unaffiliated/qwell] by Qwell[] (sorry, but...no)
22:12.20*** join/#asterisk rmayorga (i=rmayorga@168.243.89.17)
22:12.33wunderkinhmm
22:12.44*** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net)
22:13.07VoipMastanevermind, got it working
22:13.26*** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net)
22:13.31*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)
22:13.46JTwhy do people have idiotic firewall software that is junk?
22:13.53JTwhat crappy exploits trigger them
22:14.07Supaplexidiotic developers?
22:14.26Supaplexeg, some hardware routers like dlinks are affected
22:14.29JTwhy use the crap, too? :)
22:14.32JTlinksys
22:14.46JTand some windows soft based stuff have been affected by various stuff
22:14.59monstedanything short of a cisco 800 series router is crap :)
22:15.12*** join/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net)
22:15.41JTshrug, *nix does a fine job of routing/nat
22:16.07VoipMastamonsted: I've a cisco 2500 here and also an OpenBSD router/firewall and I achieve better results using the OpenBSD based one
22:16.09Supaplexanything short of FOSS is crap ;)
22:16.22*** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net)
22:16.56harlequin516[TK]D-Fender: Trying ...
22:18.13harlequin516[TK]D-Fender: Still no dice for me...  
22:19.02*** join/#asterisk Manfish (n=themanfi@82-68-173-121.dsl.in-addr.zen.co.uk)
22:19.15*** part/#asterisk santiago (n=santiago@debian/developer/santiago)
22:19.17[TK]D-Fenderharlequin516 : Describe your exact test, including a pastebin of your dialplan and the CLI output that showst he failure
22:19.52fiber0ptidoes anyone know if you can fastforward through voicemail messages? Possibly jumping 10 seconds at a time?
22:20.02harlequin516[TK]D-Fender: Will do..  
22:21.44*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
22:21.53[TK]D-Fenderfiber0pti : Its in the WIKI page for voicemail.conf.  Go Read.
22:22.44*** mode/#asterisk [-b *!*n=noway@*.hcc.mnscu.edu] by Qwell[]
22:23.53*** join/#asterisk Latre (n=IceChat7@dsl-148-233-19-133.prod-empresarial.com.mx)
22:24.27*** join/#asterisk Eliran_Itzhak_ (n=eliran@bzq-82-81-15-57.red.bezeqint.net)
22:24.40Latrehi people, someone has configured recive Fax over E1 an asterisk???
22:26.47l2cach1has anyone used munin to successfully monitor their asterisk system?
22:27.40*** join/#asterisk Tili (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net)
22:28.01bkrusel2cach1: i used cacti, and wrote a plugin for it
22:28.56Latrei need recive fax over E1 in my asterisk.......some link/page ?
22:28.58bkruseit justs uses php -q scriptname.php <hostname(cacti fills this in )> and it returns :call count: :agents logged in: :sip peers registered: w/e you can grab from the manager interface that is a number, you can plot
22:29.08bkruseLatre: one sec.
22:29.09l2cach1cause the plugin for asterisk_channels is bollucks
22:29.24l2cach1times out every time...login is successful for the manager though
22:29.34bkrusel2cach1: what plugin? for cacti?
22:29.35bkruseLatre: http://www.voip-info.org/wiki-Asterisk+fax
22:29.47l2cach1no for munin
22:29.51bkrusenever used munin
22:29.52bkruseuse cacti
22:29.53bkruse;]
22:30.25bkrusei wrote the asterisk plugin, and i can give it to you if you want, i might submit it on the forums, the members are very helpful, and its quite easy to setup (~1 hour)
22:30.38*** join/#asterisk cjlowe (n=cjlowe@CPE-155-143-205-218.vic.bigpond.net.au)
22:30.48l2cach1ok...that would be great
22:30.52l2cach1can i get it plz?
22:31.06bkrusel2cach1: no problem, do you know php at all? its kind of hacked up since only I used it
22:31.16cjlowehowdy everyone
22:31.16bkrusehttp://cacti.net/
22:31.19bkrusecjlowe: hellow
22:31.29Latrebkruse itry that with iaxmodem but not work......try to with spandsp but not works...
22:31.31*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:31.36l2cach1wow...cacti is very nice...munin is lame compared to it
22:31.46bkruseLatre: im not sure what to tell you man
22:32.00bkrusel2cach1: oh ya, and its open souce, so if you know a LITTLE programming, or can hire some addons, its incredible
22:32.03bkruseand the community is great
22:32.13bkrusei have all mine automated now to automatically update graphs and save them etc etc
22:32.20backbluewow, who the hell dont knows cacti? :o
22:32.41bkrusebackblue: agreed, but if you want to customize it, it can get intense quickly
22:32.47bkruseand # you can get from the managers interface, you can plot into an awesome graph
22:32.49bkruseany*
22:33.17l2cach1thanks alot
22:33.20cjloweI have a simple question: I have G.729 licenses and can make calls great if only G.729 is allowed in my sip.conf. For reasons I won't go into, I also need to allow alaw and ulaw, but for some reason when I allow those none of my calls use G729. How do I make it 'prefer' g729 then fall back on alaw?
22:34.00bkrusel2cach1: no problem, hit me up, or email me at bkruse@digium.com when you want the plugin, im going to touch it up a bit
22:34.07[TK]D-Fendercjlowe : make sure the order is set with G729 occuring first on both sides of your call.
22:34.27bkrusecjlowe: yes, and set g729 to prefered on your phone.
22:35.31cjlowebkruse, [TK]D-Fender g729 is set on the phone - when you say occuring first, you mean the allow=g729 should be the first one?
22:35.57cjlowebkruse, [TK]D-Fender never mind, my general context didnt have g729 in it, only my trunks :)
22:36.13l2cach1can cacti also monitor cpu usage and server stats
22:36.15cjlowebkruse, [TK]D-Fender  general section*
22:36.17l2cach1we are using mrtg atm
22:36.40backbluel2cach1: i dont see, what you *cant* monitor with it.
22:37.07l2cach1i know that..but can i easily implement it so it monitors a few server's cpu load and temperature stats to a graph
22:37.13l2cach1munin works for that right now
22:37.19voipmandoes * handle codec negotiation yet? IE: ingress sip entry has allow=g729 first, allow=ulaw second, egress has allow=ulaw (w/o g729 ports), will the call negotiate to ulaw?
22:37.26l2cach1just the asterisk plugin for munin doesnt work
22:37.44resistancedoes anyone use the intercom feature with fxs?
22:38.20fiber0ptiD-Fender: is there anyway to use the voicemailmain application to play other files through an extension and therefore allowing a user to seek through a file?
22:40.33bkrusel2cach1: cacti is pretty far developed, and its AMAZING, i can monitor ANYTHING
22:40.38cjlowevoipman, I've just set up our box to do just that I THINK
22:40.41bkrusedual cpu load average, just take out a day, and see what you think
22:41.34l2cach1awesome..i will try it out
22:41.58[TK]D-Fenderfiber0pti : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ControlPlayback
22:42.21*** join/#asterisk nukio (n=deep@pool-162-84-144-173.ny5030.east.verizon.net)
22:42.26heartoneshas any one tried any ata adaptor from www.tigernetcom.com to get to work with asterisk
22:42.32[TK]D-Fenderfiber0pti : You need to get off your butt and read more...
22:42.44bkrusel2cach1: and email me when your ready, its aweseome
22:42.46*** join/#asterisk Scoundrel (n=scound@host-81-191-169-198.bluecom.no)
22:42.50nukiohey all
22:42.50l2cach1i emailed you...lol
22:42.57nukioi hope everyone is having a great day
22:42.58nukio;)
22:43.29l2cach1i am going to implement it at home, then at work - we have 5 asterisk servers
22:43.34[TK]D-Fenderheartones : Those look like 100% Linksys rip-offs
22:45.54JTbeen done already
22:46.00robin_szno way
22:46.08robin_szway?
22:46.09Strom_Crobin_sz: i have one right here that runs asterisk
22:46.14robin_szcoo.
22:46.24JTrobin_sz: just don't expect to transcode much
22:46.28robin_szheh
22:47.37*** part/#asterisk dasenjo (n=dasenjo@63.245.86.215)
22:48.10[TK]D-Fenders/much/period
22:48.46JTthat doesn't make much sense, wouldn't you s/much/./?
22:48.57JTwhat is it with americans' fascination with the word period
22:49.17[TK]D-FenderJT : Strictly juvenile ;)
22:49.33JTheh
22:49.43resistanceare there alternative ways to setup intercom/paging other than the soundcard?
22:50.34[TK]D-Fenderresistance : Buy and FXS based unit like a Viking, or get phones that support auto-answer + Speakerphone.
22:53.00*** part/#asterisk l2cach1 (n=ghansen@64.128.254.98)
22:53.44resistanced-fender: what is the model number?
22:53.50resistancehttp://www.vikingelectronics.com/products/view_product.php?pid=29
22:53.52resistance?
22:55.01*** join/#asterisk Strom_C_ (n=strom@netblock-66-159-243-60.dslextreme.com)
22:55.42[TK]D-Fenderresistance : Yup, that would do it.
22:56.35resistanced-fender: do u have any idea why i'm having this prob with my sound board?
22:57.09CunningPikeresistance: We use this one: http://www.vikingelectronics.com/products/view_product.php?pid=291. Works great
22:57.41resistancecunningpike: how much does that retail for?
22:58.03CunningPikeresistance: I think it was less than $200
22:58.25[TK]D-Fenderresistance : What problem?  You haven't said anything of value about it.
22:59.13*** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com)
22:59.46[TK]D-FenderCunningPike : Line-level out... nice.  Lets you MUX it up to 10KW power amps to annhilate your pagee ;)
22:59.59monstedCunningPike: you're a queen?
23:00.01resistanceok, i dial into it hear the beep, but one the fxo end it gives a cracklingy sound, i can hear the beep on the speaker, but it's very shitty sound, i can't explain it, and i also can't understand what is being said
23:00.05CunningPike[TK]D-Fender: Hee hee
23:00.17CunningPikemonsted: No, a fish. Pay attention
23:01.06resistanceit sound like farting a beeping at the same time, only the farting is too loud
23:01.24JTis it a full duplex sound card?
23:01.47[TK]D-Fenderresistance : FXO end?  What are you pluggin into what?
23:01.52resistanceit's soundblaster live, not exactly sure
23:02.24resistanced-fender: t1 > channel bank > fone
23:03.58[TK]D-Fenderresistance : And where does FXO appear in that mess?  
23:04.23resistancesorry, fxs
23:04.23[TK]D-Fenderresistance : Your whole description is pretty whacked.  pastebin your dialplan where you do your paging.
23:04.35[TK]D-Fenderresistance : EVERYTHING related to it.
23:04.37resistanced-fender, it's fxs
23:05.01*** join/#asterisk infernix (i=nix@spirit.infernix.net)
23:05.26resistanceexten => *60,10,Dial(CONSOLE/dsp,,A(beep))
23:05.48resistancethat's all i got to test, priorite is 1, sorry
23:06.08mercestesomg...lol
23:06.14mercestesVoIP, ghetto style.
23:06.24JTresistance: PASTE, i think was the point
23:06.31JTwe don't want to debug your typing errors too
23:06.34Strom_M~pb
23:06.36jbotsomebody said pb was a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
23:06.39JTor errors in what you think is relevant
23:07.04JTand you still have no described the electronic setup
23:07.12JTwe do not have powers of telepathy
23:07.28[TK]D-Fenderresistance : Make sure the proper driver is being loaded for your card (TEST IT), and also check your mixer settings.
23:07.52[TK]D-Fenderresistance : and what is it plugged into for you to hear?
23:07.59*** join/#asterisk jm|work (n=jamiem@zen.jamiem.com)
23:08.08resistancestereo speaker :D
23:08.44mercestesCar stereo or home stereo speaker?
23:08.47JTso i assume playing music through the same soundcard and speakers sounds fine?
23:08.50resistancejt: asterisk > tdm400p > adtran channel bank(FXS) > analog fone
23:09.08*** join/#asterisk aao_pwner|oper (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net)
23:09.17resistancejt: haven
23:09.27resistance't tried
23:09.28resistance:D
23:09.32JTwell surely your speaker is connected to the sound card
23:09.38JTnot the analog PHone
23:09.56[TK]D-Fenderresistance : I asked what was plugged into your SOUND CARD.
23:10.32resistanceand i said: a computer stereo speaker, i've also tried earphones
23:10.43*** join/#asterisk pdunkel (n=0x504455@213.235.212.178)
23:10.49JTtry listening to music, not through asterisk
23:10.51JTsee if it works
23:11.09resistanceok, i will, thanks for your help :D
23:11.35JTlike [TK]D-Fender said, may be a mixer issue, or driver
23:11.37[TK]D-FenderJT : Funny, noone listens when I tell them to do that :)
23:11.58pdunkelHi all
23:12.12JTheh
23:12.19resistanced-fender, i'm listening, thanks for u're help
23:12.30pdunkelAnyone know the the status of Snom-Pickup in 1.2.13?
23:12.47*** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net)
23:12.49JT[TK]D-Fender: i would've though it's be obvious troublshooting, if you've never heard good sound out of a sound card setup, to test it with a more basic audio setup
23:13.55resistanced-fender: excuse me for being an ass
23:14.34*** join/#asterisk Op3r (i=Op3r@121.97.192.134)
23:15.02Op3rcan anyone show me a working sample of ivr?
23:15.34*** join/#asterisk waltz (i=walterbr@bas15-toronto12-1168014906.dsl.bell.ca)
23:16.26*** part/#asterisk sb_mx (n=sb_mx@200.78.229.18)
23:17.11*** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net)
23:17.20mercestesgoogle asterisk ivr is a pretty good sample.
23:17.47[TK]D-FenderOp3r : http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
23:17.55[TK]D-Fender~book
23:18.02jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
23:18.02[TK]D-Fender~wikis
23:18.04jbotit has been said that wikis is http://www.voip-info.org
23:18.04[TK]D-Fender~jfgi
23:18.09jbot!google just google it
23:18.17Op3ri found it
23:18.18Op3rbut
23:18.23[TK]D-FenderNO BUTS!
23:18.25Op3rhmm
23:18.29Op3rlet me find it!!
23:18.31Op3r:(
23:18.32[TK]D-FenderYOU READ!
23:18.38Op3ri just need a working one so that I can based it
23:18.39[TK]D-Fender:)
23:19.13[TK]D-FenderThose samples work.  Go read, go try, then show your failed attempt after applying some actual effort!
23:19.49mercestesThank you, come again.
23:20.09[TK]D-FenderSuivant!  Next!
23:20.19Op3rhow do you hack an asterisk server?
23:20.21Op3rhahahaha
23:20.24Op3rjoke joke
23:20.35mercestesdid you google hack asterisk?
23:20.39Op3ryes I did
23:20.44JTwell, if you want an IVR already working exactly how you want it, i believe there are some people here who offer consulting services
23:20.58[TK]D-FenderJT : Indeed I do!
23:21.13Op3rjust need to see how it works
23:21.13Op3r:D
23:21.26JTi'll offer consulting for the right price too ;)
23:21.33mercestesthen hire a consultant...and then go back over what he did..:D
23:21.45Op3rcan he do it for 50 bucks?
23:21.45Op3r:D
23:22.10JTyeah, if it just answers a call, plays a recording, and hangs up :)
23:22.49Op3rhahaha
23:22.56*** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com)
23:22.57Op3rhahaha
23:23.03Op3rI can just direct it to tt-monkeys
23:23.37JTi prefer weasels-eaten-phonesys
23:23.49Strom_Ci thought that file was called tt-weasels
23:23.55vooduhalOdd question all.  We just updated the firmware and drivers for our Sangoma A104d and now when ever you reach the mailbox greeting and press "*" to get VoicemailMain() it just hangs.  Anyone else experienced this?
23:24.24JTStrom_C: i have both... who knows
23:26.12*** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net)
23:28.05vooduhalSo anyone have any ideas what might be causing it?
23:29.08*** join/#asterisk EyeCue (n=eyecue@220-253-130-38.QLD.netspace.net.au)
23:30.23*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
23:31.24*** join/#asterisk Menace- (i=menace@66.181.104.31)
23:34.38*** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net)
23:34.45*** join/#asterisk jerryoc (n=jerryoc@cpe-75-80-102-22.socal.res.rr.com)
23:35.03Op3rdoes any one knows somebody who provide Philippine DID number?
23:35.54*** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com)
23:35.58jerryocgot question...will i need the license version or g729 to be able to use dtmf tones
23:36.33JTyou should be licensed to transcode g.729 in Asterisk
23:36.37JThowever
23:36.46JTg.729 does not transport DTMF tones
23:36.47jerryocwell i'm testing it for now before i buy
23:36.59JTyou must use out of band dtmf transmission
23:36.59jerryocyeah, i'm trying to rfc2833 it
23:37.23JTonly g.711 or ADPCM can send DTMF in-band reliably
23:37.24jerryocstrange though, been trying everything relaxdtmf...etc
23:37.26*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)
23:37.42JTdo you have control over both sides?
23:37.45Qwell[]mitcheloc: !
23:37.57jerryocno...JT i wish :(
23:38.04mitchelocwhats up jas
23:38.09JTthen you will need to talk to your provider
23:38.16Qwell[]no names :D
23:38.17jerryocipsmarx says they support it
23:38.17JTthey need to support out of band dtmf too
23:38.26Qwell[](kidding - I don't care)
23:38.35jerryocipsmarx says they support dtmf rfc2833 over g729
23:38.36mitchelocJason, no problem
23:38.39fileif you know Qwell's real name... you have power over him!
23:38.40mitcheloc=P
23:38.40*** join/#asterisk waverly360 (n=waverly@209.12.249.243)
23:38.44hmmhesaysthats odd
23:38.53JTit's not over g.729, it would be over SIP
23:38.53mitchelocoops, everyone has the power now
23:38.55RoyKhttp://karlsbakk.net/fun/asterisk-installation.wav
23:39.02fileneed his last name too
23:39.03jerryocthey claim that they have other asterisk users using g729 with dtmf
23:39.09mitchelocQwell[]: i suppose it's only fair...go ahead and tell them my name
23:39.12JTthe dtmf has absolutely nothing to do with the codec RTP stream when sending out of band
23:39.19*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)
23:39.26mitchelocand psh on jerry, he stole my naming schema
23:39.42jerryoclol mitch...i'm actually really from the OC
23:39.46jerryocOrange County....lol
23:39.54jerryocu too?
23:39.57mercestesreally?  My wife was originally from the OC.
23:40.05mitchelocjerryoc: who said i'm not?
23:40.09jerryocyup, Huntington Beach ...
23:40.21jerryocwooo...really cool...
23:40.27mitchelocjerryoc: fullerton here, ever go to the scau?
23:40.33mitcheloc* scaug
23:40.37Qwell[]bah
23:40.43jerryocno, i hang out at the disrict next to Chapman University
23:40.44[TK]D-Fenderfile : Say it backwards... and *poof* !
23:41.02mitchelocqwelloc: don't you mean qwelllc or qwellrc?
23:41.09Qwell[]lc?  rc?
23:41.30mitcheloclos angeles county (well i suppose thats LAC) or riverside county (hah!)
23:41.37jerryocfullerton is cool though mitch, that jazz steamers place bangs too..
23:41.43Qwell[]no, because oc > rc :P
23:41.48mitchelocnever been there
23:41.56mitchelocQwell: for sure
23:42.00Qwell[]and fullerton is like the fontana of sb :P
23:42.05Qwell[]erm, of the oc
23:42.07jerryoclol
23:42.27mitchelocfor your safety i'm assuming thats a good thing
23:42.28jerryocfullerton the fontucky of OC
23:42.31*** join/#asterisk docelm0 (i=vircuser@67.110.179.190.ptr.us.xo.net)
23:42.36Qwell[]jerryoc: ;)
23:43.33Qwell[]heh
23:43.34docelm0haha scardy cat..  
23:43.45[TK]D-FenderFor if Qwell were to be exposed to sun-light... he'd COMBUST! 9:4F
23:43.46mitcheloci turned off my lights so they wouldn't know where to shoot!
23:43.58mitchelocdocelm0: i owe you money, ah, pm me your paypal :)
23:44.09docelm0for?
23:44.16mitchelocthat pretty picture you took with me
23:44.21docelm0ohh ya bitch
23:44.26mitchelochaha
23:44.28docelm0you owe me a copy of the picture too
23:44.40jerryoclol, u guys have this much fun in here normally?
23:44.45mitcheloci know, i don't have a scanner, but i'll hook it up, don't worry
23:44.46Qwell[]jerryoc: pretty much
23:44.49docelm0no..  MORE!
23:44.57mitchelocjerryoc: only when i'm here
23:45.02docelm0fucking walmart dude..  there is one in every town
23:45.12Qwell[]Huntsville has TWO
23:45.15mitchelocjerryoc: what was the answer on that scaug? ever been there?
23:45.19docelm0hehe tampa has 8
23:45.24Qwell[]jeebus
23:45.24JTwe have 0 walmarts
23:45.26mitchelocHSV, huh? bah!
23:45.33Qwell[]mitcheloc: you still here?
23:45.36*** part/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net)
23:45.37jerryocnah, haven't been there mictch...location? vibe?
23:45.42mitchelocyes
23:46.22mitchelocjerryoc: not as good of a vibe as my pocket rocket, but still fun
23:46.34mitchelocbrian, where is that e-mail?
23:46.46jerryocscaug, thats a user group, heh?
23:46.48mitcheloc** private message
23:47.25mitchelochttp://www.socalasteriskug.org/
23:47.56waltzwhy the fsuk is everything web-based now-a-days!?
23:48.00waltzwe need some telnet
23:48.06docelm0ohh ya mitcheloc I dont do paypal anymore..  
23:48.07waltzand maybe ssh + ncurses
23:48.24mitchelocdocelm0: fine with me, i'll just e-mail you a copy then and keep the $10
23:48.33waltzget serious people, i'm not going to run X on my server, nor is any sensible person going to do so
23:48.37mitchelocwaltz: why can't we use ssh and xmpp?
23:48.48mitcheloctyping out xml commands sounds fun
23:48.50docelm0I do take credit card..   :P
23:49.04waltz*sigh*
23:49.22waltzmitcheloc: what about snmp?
23:49.26waltzjeez
23:49.39waltzfor administering hardware telnet is wonderful
23:49.39mitchelocwell i prefer pigeon over snmp
23:49.55waltzwhat?
23:50.25waltzwhat the christ is pigeon?
23:50.28mitchelochttp://en.wikipedia.org/wiki/Homing_pigeon
23:50.46waltzdid you just escape from a mental institution or something?
23:51.03waltzdon't waste my time you ass****
23:51.28mercesteswhy did you block out the hole and not the ass?
23:51.35hadsI see it's nice and friendly in here then
23:52.04waltzyes, isn't it
23:52.12docelm0ok Im off to my asterisk users meeting in philly..   cya!
23:52.31*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
23:52.50shmaltzanybody know of a call accounting system for asterisk?
23:53.08[TK]D-Fendershmaltz : res_accpac?
23:53.26shmaltz[TK]D-Fender whats that?
23:53.47mitchelocgot to run, bye bye!
23:53.59waltzhey, shmaltz rhymes with waltz :D
23:54.12[TK]D-Fendershmaltz : Comedy? :)
23:54.20shmaltzyep lol
23:54.29[TK]D-Fenderwaltz : hukt on fonix werkt 4 u!
23:54.42waltz[TK]D-Fender: sorry, i don't speak moron
23:55.19[TK]D-Fenderwaltz : You're evidently an idiot-savant ;)
23:55.46shmaltzanybody know of a call accounting system that works for asterisk?
23:55.53jerryoc...
23:56.04hmmhesaysdefinate call accounting?
23:56.09[TK]D-Fendershmaltz : ast-bill was a name i've seen thrown around here a few times.  Check'em out ont he WIKI
23:56.16JTwaltz: it certainly sounds like you do
23:56.29hmmhesaysasterisk stat v2
23:56.29JTwaltz: or you at least speak pole-shoveus-up-arseus
23:56.50jerryoc....oh boy'
23:56.52hmmhesaysarseus is that a god?
23:57.05[TK]D-Fenderhmmhesays : Defecate call accounting
23:57.16hmmhesayslol
23:57.17*** join/#asterisk shinux__ (n=shinux@196.220.30.151)
23:57.28JThmmhesays: dunno, perhaps some worship bums
23:59.50SkramXhey.. [TK]D-Fender
23:59.55SkramXI heard you were a polycom guru
23:59.59*** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-61bc1d7059bfc6ee)

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