IRC log for #asterisk on 20070704

`00:05.02`	`*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)`
`00:05.02`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:05.36`	`Aces1Up`	`i'm running my asterisk box on a vmware image of fedora 7, and getting the following error, is this a bad compile type error or connection / NAT issue? Sip_Xmit of 0xaddress f8 (len 492) to ip:5060 returned -2 Bad file descriptor`
`00:12.19`	`Aces1Up`	`If i need to start from scratch and recompile zapata drivers and asterisk drivers, can i just recompile with the same tarball? or should i perform certain steps for a reinstall/`
`00:12.25`	`Aces1Up`	`i'm running fedora 7`
`00:12.43`	`*** part/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)`
`00:13.12`	`JT`	`what's the sip address?`
`00:13.29`	`Aces1Up`	`you talking to me JT?`
`00:13.47`	`JT`	`yes, doesn't seem to be anyone else talking`
`00:15.13`	`Aces1Up`	`i get that error on all my peers... 81.201.84.28`
`00:15.23`	`Aces1Up`	`says destination unreachable.`
`00:15.49`	`JT`	`btw you really shouldn't run asterisk in vmware`
`00:15.57`	`JT`	`i'd check the networking setup in vmware`
`00:16.48`	`Aces1Up`	`will double check, so you don't think it is a compile error?`
`00:17.29`	`Aces1Up`	`i compiled both zaptel and asterisk with no errors during compile, only thing i notice afterwards is i did not create symbolic link to sources for zaptel drivers.`
`00:17.34`	`JT`	`don't think so`
`00:17.44`	`JT`	`sip has nothing to do with zaptel`
`00:17.50`	`Aces1Up`	`ok doke.`
`00:20.03`	`Aces1Up`	`hrmmm got a segmentation fault when reloading sip.conf`
`00:20.27`	`Aces1Up`	`guess recompile eh?`
`00:20.40`	`JT`	`stop using vmware would be an even better idea`
`00:21.30`	`Aces1Up`	`well, i rebooted asterisk after seg fault, and is working now with new sip settings.`
`00:22.11`	`*** join/#asterisk troy- (n=tabmeist@toroon12-1177845255.sdsl.bell.ca)`
`00:22.15`	`JT`	`ok, just expect it to be dodgy though`
`00:22.21`	`troy-`	`zaptel is telling me i dont have the kernel sources installed, how can i fix this?`
`00:24.28`	`Strom_M`	`install the kernel sources maybe?`
`00:25.41`	`*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) [NETSPLIT VICTIM]`
`00:25.41`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM]`
`00:25.41`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM]`
`00:25.41`	`*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]`
`00:25.43`	`troy-`	`Strom_M, i did yum install kernel-devel-* but that didnt solve it`
`00:26.42`	`*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)`
`00:31.29`	`troy-`	`can anyone help me install kernel sources? did yum install kernel-devel-*`
`00:32.31`	`*** join/#asterisk djs_2_6 (n=DJS@cpe-075-182-081-167.nc.res.rr.com)`
`00:34.01`	`snuff-work`	`sounds good enough.. but`
`00:34.33`	`snuff-work`	`your probably using kernel-smp.. so yum install kernel-smp-devel*`
`00:35.24`	`troy-`	`snuff-work, yup i am using smp but after doing that it still cant find it`
`00:35.41`	`troy-`	`You do not appear to have the sources for the 2.6.9-55.ELsmp kernel installed.`
`00:35.49`	`troy-`	`where is it looking?`
`00:39.23`	`*** join/#asterisk kolian123 (n=kolian@124.107.63.223)`
`00:39.29`	`kolian123`	`Hello`
`00:39.52`	`kolian123`	`Anybody played with TE405P in HDLC mode?`
`00:40.08`	`kolian123`	`Does it work with newer kernels 2.6.20?`
`00:40.40`	`troy-`	`do i need zaptel if i dont have an interface card?`
`00:40.53`	`kolian123`	`you need ztdummy from zaptel`
`00:42.04`	`troy-`	`even if im just using sip and iax?`
`00:42.28`	`JT`	`you don't always need ztdummy/zaptel`
`00:42.40`	`JT`	`only if using meetme conferences, iax2 trunking or MoH`
`00:43.08`	`JT`	`troy-: /usr/src/linux or /usr/src/linux-2.6 probably`
`00:43.32`	`troy-`	`JT for some reason its telling me it cant find kernel sources`
`00:43.41`	`troy-`	`even though i have installed them using yum`
`00:44.27`	`troy-`	`any thoughts?`
`00:45.41`	`JT`	`troy-: did you even bother to look to see that those locations were pointing to the correct source tree?`
`00:46.22`	`troy-`	`im not sure where zaptel is looking for kernel sources`
`00:47.09`	`JT`	`i just suggested some locations to check are setup correctly`
`00:47.12`	`JT`	`please CHECK them`
`00:47.53`	`troy-`	`<PROTECTED>`
`00:48.01`	`troy-`	`i have kernel folders in /usr/src/kernels/linux* though`
`00:48.12`	`troy-`	`(they actually start with 2.6.9)`
`00:48.32`	`*** join/#asterisk fujin (n=fujin@unaffiliated/fujin)`
`00:48.55`	`troy-`	`JT would that be incorrect?`
`00:49.23`	`fujin`	`anyone got any experience with linux-ha and asterisk? (heartbeat)`
`00:49.36`	`JT`	`yes it would be incorrect`
`00:49.50`	`JT`	`add symlinks, if you can find the correct kernel source somewhere`
`00:51.12`	`troy-`	`JT the output from uname -r is: 2.6.9-55.ELsmp`
`00:51.12`	`troy-`	`<PROTECTED>`
`00:51.30`	`JT`	`guess that will do`
`00:52.00`	`tzafrir_laptop`	troy-, zaptel looks in /lib/modules/`uname -r`/build
`00:52.39`	`tzafrir_laptop`	`basicaly: install the matching kernel-devel for your kernel package and you should be done (kernel-smp-devel, actually)`
`00:53.05`	`tzafrir_laptop`	`start with: uname -r; rpm -qa \| grep kernel`
`00:54.42`	`troy-`	`tzafrir_laptop, i already did yum install kernel-devel-smp`
`00:55.05`	`tzafrir_laptop`	`kernel-smp-devel`
`00:55.32`	`troy-`	`tzafrir_laptop, right my bad`
`00:55.35`	`troy-`	`and it still cant find it`
`00:55.48`	`tzafrir_laptop`	`Another pitfal: your installed kernel-smp-devel may be more up-to-date than your current kernel version. That is the reason for my command above`
`00:56.03`	`JT`	`strace the zaptel compilation, you can see what files it is looking for`
`00:56.16`	`tzafrir_laptop`	`that's too big a gun`
`00:56.22`	`troy-`	`tzafrir_laptop, i just downloaded the CD now`
`00:56.23`	`JT`	`lies ;)`
`00:56.32`	`troy-`	`tis the truth!!!`
`00:56.45`	`JT`	`i always strace when make can't find something`
`00:56.50`	`troy-`	`JT would you mind showing me what the proper ln -s command would be, think im messing it up?`
`00:57.18`	`JT`	`in /usr/src`
`00:57.19`	`troy-`	`JT the output from uname -r is: 2.6.9-55.ELsmp and my folder is called 2.6.9-55.0.2.EL-i686`
`00:57.27`	`JT`	`ln -s /path/to/sources/ linux`
`00:57.29`	`JT`	`and`
`00:57.31`	`JT`	`ln -s /path/to/sources/ linux-2.6`
`00:57.48`	`tzafrir_laptop`	`kolian123, it seems that certain modules have some problems with 2.6.22 . But I believe earlier versions would be OK`
`00:58.02`	`tzafrir_laptop`	`(some modules == only ztdynamic?)`
`00:58.17`	`*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`00:58.34`	`troy-`	`JT i could a make error 2 at the end, did it work?`
`00:58.39`	`tzafrir_laptop`	`(and anyway, the fix is trivial and already in the SVN)`
`00:58.47`	`troy-`	`*got :P`
`00:59.26`	`troy-`	`does the make error 2 at the end signify that zaptel built just fine?`
`00:59.50`	`JT`	`error means it failed and aborted`
`01:01.11`	`troy-`	`omg i think its working`
`01:01.39`	`*** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net)`
`01:02.16`	`troy-`	`ITS ALIVE!!!!!`
`01:02.27`	`kolian123`	`hi tzafrir`
`01:02.41`	`kolian123`	`i'm on 2.6.20-16`
`01:02.50`	`JT`	`troy-: nice`
`01:02.57`	`*** join/#asterisk Braxus (n=bhsieh@66.147.214.164)`
`01:03.03`	`kolian123`	`getting this on HDLC`
`01:03.07`	`kolian123`	`hdlc0: Unable to set HDLC protocol information: Invalid argument`
`01:03.08`	`troy-`	`JT after my make install i can leave it since i dont have any telephony cards?`
`01:03.18`	`JT`	`kolian123: what's the setup?`
`01:03.20`	`troy-`	`(move on to compiling asterisk)`
`01:03.26`	`JT`	`troy-: yes`
`01:03.29`	`kolian123`	`TE405P`
`01:03.54`	`kolian123`	`span=3,1,1,esf,b8zs`
`01:03.54`	`kolian123`	`# termtype: te`
`01:03.54`	`kolian123`	`nethdlc=49-72`
`01:04.04`	`JT`	`kolian123: pastebin.a zaptel.conf and zapata.conf`
`01:04.05`	`kolian123`	`hdlc0 shows up ok`
`01:04.10`	`JT`	`eh`
`01:04.11`	`kolian123`	`one sec`
`01:06.18`	`troy-`	`JT you deserve a medal`
`01:06.47`	`JT`	`;)`
`01:06.56`	`yonahw`	`tzafrir: when do you sleep?`
`01:07.18`	`JT`	`kolian123: are you trying to run data or something weird?`
`01:07.29`	`troy-`	`JT (company is going to be pissed if the phones dont work tomorrow morning)`
`01:08.24`	`JT`	`you planned to setup the whole pbx overnight?`
`01:08.25`	`fujin`	`owned!`
`01:08.31`	`troy-`	`JT yeah :)`
`01:08.31`	`fujin`	`good luck on that matey`
`01:08.41`	`JT`	`crazy`
`01:08.54`	`troy-`	`yep`
`01:08.57`	`fujin`	`haha`
`01:09.00`	`fujin`	`mind you, I've done it before`
`01:09.02`	`fujin`	`2 day setup`
`01:09.35`	`troy-`	`bell is coming tomorrow to do the POTS terminations`
`01:10.11`	`Hymie`	`anyone here using unidens?`
`01:10.22`	`troy-`	`oh btw done compiling and installing`
`01:10.41`	`troy-`	`now just have to generate config files for all the phones, extensions and voicemail`
`01:10.44`	`JT`	`fujin: it depends if you have much past experience or not really`
`01:10.48`	`fujin`	`ls`
`01:10.51`	`fujin`	`woops`
`01:10.55`	`troy-`	`wrong window :P`
`01:10.56`	`fujin`	`JT: I must have got lucky`
`01:11.04`	`fujin`	`somehow my alt-tab got back to here`
`01:11.17`	`JT`	`you can do a simple system in an hour or two if you know wtf you're doing`
`01:12.18`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`01:12.19`	`*** join/#asterisk zuesman (n=zuesman@66.39.201.241)`
`01:12.33`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`01:12.38`	`troy-`	`JT considering all the phones are the identical model, i just recorded the MAC IDs, reset the phones and boom, configured`
`01:12.46`	`troy-`	`TFTP baby`
`01:13.07`	`JT`	`evil tftp ;)`
`01:13.54`	`*** join/#asterisk alrs (n=lars@pozug.com)`
`01:14.05`	`[TK]D-Fender`	`FTP = better`
`01:14.45`	`rob0`	`SMTP = horrible mess`
`01:16.32`	`JT`	`telnet`
`01:16.41`	`JT`	`gar`
`01:18.14`	`[TK]D-Fender`	`JT : Nuget-bot's been triggered too recently`
`01:18.33`	`JT`	`how recent?`
`01:22.03`	`*** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca)`
`01:25.53`	`Hymie`	`anyone with unidens?`
`01:25.56`	`Hymie`	`fender dude!`
`01:26.10`	`Hymie`	`[TK]D-Fender: there, this will highlight for your name, Fender person!`
`01:29.12`	`*** join/#asterisk DJ_Kit (n=lamass@83.149.52.8)`
`01:29.17`	`DJ_Kit`	`hi guys ;)`
`01:30.09`	`JT`	`Hymie: "defender"`
`01:38.02`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`01:38.42`	`[TK]D-Fender`	`Hymie, UIP-200 = flaming piece of shit.`
`01:38.42`	`aptura`	`Our major cable network here is having some major issues. Big time packet loss and high ttls. Affecting the voip service, Cable tv.`
`01:39.05`	`[TK]D-Fender`	`Hymie, wothy only of being put into rape-risk areas for their expendability.`
`01:39.31`	`[TK]D-Fender`	`Hymie, TFTP = ugly and their intuitiveness blows`
`01:40.29`	`fujin`	`providing you have a reasonably smart tftp client implementation`
`01:40.34`	`*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)`
`01:40.55`	`crimethinker`	`a national holiday where children are given explosives. what could possibly go wrong?`
`01:42.11`	`DJ_Kit`	`hi`
`01:42.27`	`crimethinker`	`Hello, DJ_Kit.`
`01:42.38`	`[TK]D-Fender`	`crimethinker, 2 words : Natural ^#@$ing Selection <------------------`
`01:42.39`	`DJ_Kit`	`i need to tune my f*** server.... it's couldn't run... ANd i don't know how to fix it`
`01:42.41`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`01:42.53`	`Hymie`	`[TK]D-Fender: yes, uniden sucks. I have a client that I'm actually thinking of just buying their old phones (and then throwing them away) and replacing them with plycom 301s`
`01:42.59`	`Hymie`	`[TK]D-Fender: damned uniden`
`01:42.59`	`aptura`	`Shaws network is going to crap tonight. Never been this bad before.`
`01:43.08`	`[TK]D-Fender`	`Hymie, 301 = WASTE`
`01:43.19`	`[TK]D-Fender`	`Hymie, Why on earth would you do that?`
`01:43.22`	`Hymie`	`[TK]D-Fender: why? The price point means _everything_`
`01:43.31`	`Hymie`	`[TK]D-Fender: because the unidens are utter and completely useless`
`01:43.34`	`[TK]D-Fender`	`Hymie, Because the IP 320 DESTROYS the 301`
`01:43.39`	`Hymie`	`they lcok up, they drop calls, they don't work, etc`
`01:43.40`	`DJ_Kit`	`it runs, but my client is using only peer-to-peer mode`
`01:43.40`	`aptura`	`and how in the fricken world is it affecting my ast box on my local network. I think the registries are looking there connection and its freezing my box.`
`01:43.54`	`*** join/#asterisk fx0 (n=fx0@cypher.punk.net)`
`01:44.06`	`Hymie`	`[TK]D-Fender: well, I'll look at the feature set diference, but things like PoE are meaningless to me`
`01:44.17`	`[TK]D-Fender`	`Hymie, 301 = $115. 320 = $95, has PoE built-n, pixel display, SPEAKERPHONE, MicroBrowser, and supports EVERYTHING`
`01:44.31`	`Hymie`	`speakerphone, hmm`
`01:44.32`	`[TK]D-Fender`	`Hymie, Get with the "now"!`
`01:44.33`	`aptura`	`I will probebly disapear again.`
`01:44.50`	`Hymie`	`[TK]D-Fender: poe built in, but... hopefully I can use normal power`
`01:45.39`	`Hymie`	`[TK]D-Fender: anyhow, asterisk 1.4 works even less well with the crappy unidens.. now they won't even remote reboot`
`01:45.50`	`Hymie`	`which was the only thing that kept them even partially usable`
`01:45.53`	`Hymie`	`a nightly 3am reboot`
`01:46.38`	`Vorondil`	`Phones that require a nightly reboot?`
`01:47.26`	`Hymie`	`hmm, need thr 330`
`01:47.42`	`Hymie`	`these guys won't require additional jacks for the phones`
`01:47.47`	`Hymie`	`er, rewire`
`01:47.50`	`Hymie`	`or require I guess ;)`
`01:48.48`	`*** join/#asterisk _shad_ (n=shad@mail.topan.ca)`
`01:49.25`	`_shad_`	`After upgrading to 1.4.5, when I do a hangup of an incoming sip call from a provider, instead of hanging up, it just gives a busy signal. Anyone else seeing this issue?`
`01:49.48`	`[TK]D-Fender`	`_shad_, Don't stop thre, move right on to 1.4.6 :)`
`01:50.06`	`_shad_`	`[TK]D-Fender: Using debian unstable, doesn't have it yet :)`
`01:52.10`	`aptura`	`still here`
`01:52.37`	`[TK]D-Fender`	`_shad_, Packaged ... shudder*`
`01:52.37`	`aptura`	`TK seen a case where a gateway providers network issues can some how adversly effect the asterisk box when using local features?`
`01:52.55`	`[TK]D-Fender`	`aptura, ..... huh?`
`01:53.10`	`OloBola`	`has anyone been able to install lumenvox via yum? It's been timing out all day even though I can download directly with no problems.`
`01:53.15`	`JT`	`err what`
`01:53.19`	`JT`	`320/330 has speaker phone?`
`01:53.27`	`aptura`	`I am getting interitant freezes on cli and no responce. Shaw cable services is having some major issues. I get intermitant ping replys from there dns servers.`
`01:53.57`	`OloBola`	`by timing out I mean it just sits there at 35% etc forever`
`01:54.05`	`aptura`	`Never seen anything like this before.`
`01:54.58`	`[TK]D-Fender`	`JT : Yes.`
`01:54.59`	`aptura`	`Cable programming still cutting in and out :)`
`01:55.11`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`01:55.26`	`JT`	`[TK]D-Fender: nice`
`01:58.34`	`OloBola`	`is anyone interested in ssh'ing into my machine to finish up a lumenvox installation (I can paypal you for the help)? The license server is installed, just can't get SRE to install. Yum keeps stopping and I don't know how to work around it.`
`02:04.55`	`*** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au)`
`02:09.17`	`[hC]`	`anyone set up sip hints that watch an IAX2 trunk? somehow? :)`
`02:13.02`	`[TK]D-Fender`	`[hC], setup is obvious enough.`
`02:13.23`	`*** join/#asterisk tako-san (n=Tako-san@216.232.147.102)`
`02:13.32`	`[hC]`	`[TK]D-Fender: Only problem is there's no way to identify each iax channel being used, so how would you set hints on individual channels?`
`02:14.09`	`Hymie`	`[TK]D-Fender: ok, so I ordered 12 330s to replace these damned Unidens`
`02:14.20`	`[hC]`	`IAX channel names always append a -## which seems to be random`
`02:14.22`	`Hymie`	`[TK]D-Fender: you said you loved them, and wanted to buy my old Unidens? I can give you a good deal!`
`02:14.41`	`[TK]D-Fender`	`[hC], No different that SIP's suffixes.`
`02:15.00`	`Hymie`	`if I had the time, I'd sue uniden`
`02:15.00`	`[TK]D-Fender`	`Hymie, in bizarro-world where love=hate maybe!`
`02:15.09`	`[hC]`	`[TK]D-Fender: I havent done it with SIP either, just Zap. Ala ,hint,Zap/1`
`02:15.14`	`Hymie`	`those bastards just cost me $1500 or so`
`02:15.23`	`Hymie`	`hmm, maybe I'll sue in small claim's court!`
`02:15.40`	`litage_`	`is it normal behaviour for asterisk to email voicemail notifications that have already been sent when asterisk is restarted?`
`02:16.14`	`[TK]D-Fender`	`[hC], then get off your ass and try!`
`02:16.34`	`[hC]`	`[TK]D-Fender: haha. I just dont know how/if it would even be possible.`
`02:16.41`	`[TK]D-Fender`	`[hC], YES.`
`02:17.32`	`[hC]`	`[TK]D-Fender: any thoughts on where to begin? you obviously know about something that i dont!`
`02:18.06`	`[TK]D-Fender`	`[hC], Exten => teliax,hint,IAX2/myteliaxentry`
`02:18.27`	`[TK]D-Fender`	`[hC], Exten => polycom,hint,SIP/z0mgApolycom`
`02:18.44`	`[hC]`	`[TK]D-Fender: that is obvious of course, I mean, how would you have hints for 4 different calls over teliax?`
`02:19.07`	`[hC]`	`[TK]D-Fender: lets say you can make 4 calls via teliax, and you want to indicate when each of those 'call paths' are in use?`
`02:19.11`	`Eminence`	`FollowMe() seems to be too new to be found in AsteriskTFOT. whereelse should i look for documentation and examples?`
`02:19.20`	`[TK]D-Fender`	`[hC], you won't know about INDIVIDUAL channels unless you call limit 4 different peers.`
`02:19.37`	`[hC]`	`[TK]D-Fender: thats what I figured.`
`02:19.46`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
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`02:22.27`	`[hC]`	`Im going to presume that by setting incominglimit=1 and outgoinglimit=1, i can technically one call in and one out over a "friend" iax link.`
`02:23.22`	`[TK]D-Fender`	`[hC], Yes, something like that.`
`02:24.01`	`[hC]`	`[TK]D-Fender: Well that screws up the whole idea. Even if i set dedicated peers/users for in/out, that makes hints a nightmare.`
`02:24.14`	`[hC]`	`would be nice if there were just a 'calllimit'`
`02:24.28`	`[TK]D-Fender`	`[hC], For multi-connection things like that, yes`
`02:25.24`	`*** part/#asterisk dillydally (n=email@58.69.243.203)`
`02:29.43`	`*** join/#asterisk kolian123 (n=kvirc@124.107.63.223)`
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`02:30.06`	`kolian123`	`Anyone managed to get zaptel HDLC up?`
`02:30.25`	`kolian123`	`What version of kernel is the most stable?`
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`02:33.54`	`OloBola`	`is anyone interested in ssh'ing into my machine to finish up a lumenvox installation (I can paypal you for the help)? The license server is installed, just can't get SRE to install. Yum keeps stopping and I don't know how to work around it.`
`02:36.40`	`JT`	`sre?`
`02:37.03`	`kolian123`	`ser?`
`02:37.08`	`kolian123`	`JT`
`02:37.18`	`JT`	`ser doesn't sound right in this context`
`02:38.04`	`kolian123`	`sre`
`02:38.58`	`kolian123`	`need some help with data mode on a te405p`
`02:39.12`	`kolian123`	`JT, would you know what's working?`
`02:39.44`	`JT`	`i've never used data mode in zap before`
`02:40.58`	`kolian123`	`i see:)`
`02:41.04`	`*** join/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net)`
`02:44.19`	`[TK]D-Fender`	`kolian123, its rare tos ee, but documented on the WIKI`
`02:45.05`	`JT`	`when this pri gets installed, i will technically have a 9 span system ;)`
`02:45.08`	`kolian123`	`Hi Tk, do you have a link?`
`02:45.14`	`JT`	`~thewiki`
`02:45.15`	`jbot`	`from memory, thewiki is at http://www.voip-info.org/wiki-Asterisk`
`02:45.17`	`[TK]D-Fender`	`~wikis`
`02:45.18`	`jbot`	`well, wikis is http://www.voip-info.org`
`02:45.29`	`kolian123`	`Yes all configured`
`02:45.40`	`kolian123`	`nethdlc`
`02:45.44`	`kolian123`	`zaptel compiled`
`02:45.54`	`kolian123`	`with the _NET option`
`02:46.05`	`kolian123`	`hdlc module is loaded`
`02:46.37`	`kolian123`	`get an error Invalid argument when i run sethdlc-new hdlc0 hdlc`
`02:46.47`	`kolian123`	`There is something broken`
`02:47.25`	`OloBola`	`can anyone suggest a way to resolve this:`
`02:47.25`	`OloBola`	`Is this ok [y/N]: y`
`02:47.25`	`OloBola`	`Downloading Packages:`
`02:47.25`	`OloBola`	`LumenVoxSRE-7.5-303.fc5.i 12% \|=== \| 11 MB 02:37 ETA`
`02:47.36`	`OloBola`	`it stops download everyime`
`02:47.37`	`kolian123`	`Digium claims to have HDLC PPP but everything is broken with new kernels`
`02:47.53`	`kolian123`	`latest patch for pppd is two years old and won't even compile`
`02:48.05`	`kolian123`	`with new libc`
`02:48.15`	`kolian123`	`terrible support!!!`
`02:48.19`	`JT`	`i think the sangoma hdlc stuff works fine on the other hand ;)`
`02:48.32`	`kolian123`	`yes sangoma is well documented`
`02:49.27`	`kolian123`	`They need to update user manual that it's all broken with new kernels...nobody running 2.4 anymore`
`02:52.32`	`[TK]D-Fender`	`kolian123, Slackware > YOU :)`
`02:52.52`	`kolian123`	`hehe:)`
`02:53.02`	`kolian123`	`uname -a`
`02:53.31`	`OloBola`	`should I just be patient and let yum/lumenvox sit while it's (supposed to be) downloading?`
`02:54.15`	`JT`	`is it actually moving?`
`02:55.05`	`OloBola`	`I've tried all day, it stops at 15%, 33% etc and just sits. I let it sit for a half hour before ctrl-c'ing it`
`02:55.27`	`JT`	`is it using http or ftp?`
`02:55.32`	`OloBola`	`http`
`02:55.37`	`JT`	`i see`
`02:55.49`	`JT`	`something wrong with your network maybe`
`02:56.10`	`OloBola`	`I tried rebooting and was able to ping google forever.`
`02:56.20`	`OloBola`	`no dropped pachets`
`02:56.25`	`JT`	`sure`
`02:56.33`	`JT`	`different type of network problem`
`02:57.36`	`OloBola`	`I can download the file directly`
`02:57.49`	`OloBola`	`maybe it's yum?`
`02:57.56`	`kolian123`	`yum!`
`02:58.03`	`OloBola`	`uh oh`
`02:58.19`	`JT`	`maybe`
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`02:59.30`	`OloBola`	`when I try and install the rpm directly I get dependency errors`
`02:59.41`	`Krooks`	`how do I play those .gsm files recorded by idefisk ?`
`02:59.55`	`JT`	`Playback`
`03:01.14`	`Krooks`	`Playback is an app ?`
`03:02.02`	`JT`	`yes`
`03:02.12`	`Krooks`	`ok thanks`
`03:02.49`	`kolian123`	`Alright, its a kernel`
`03:03.07`	`kolian123`	`doesn't work with 2.6.20 on kubuntu server`
`03:03.13`	`kolian123`	`but works on Edgy`
`03:03.16`	`kolian123`	`2.6.17`
`03:03.19`	`kolian123`	`great!!!`
`03:04.34`	`*** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net)`
`03:05.17`	`JT`	`hmm`
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`03:11.07`	`*** part/#asterisk ovnicraft (n=logan@201.218.4.214)`
`03:17.36`	`OloBola`	`can I just move a file yum is trying to download into the "yum dir" (wherever that is)?`
`03:17.55`	`OloBola`	`or tell yum to pull it from my disk?`
`03:18.36`	`OloBola`	`I can download it, yum can't for some reason.`
`03:18.47`	`OloBola`	`it downloads all the other files fine though.`
`03:20.21`	`[TK]D-Fender`	`So DL it manually and install it manually`
`03:21.17`	`OloBola`	`I get dependency errors when installing manually`
`03:21.20`	`OloBola`	`tons`
`03:22.12`	`OloBola`	`the only time I'm at the CL is when I'm trying to setup *`
`03:22.18`	`OloBola`	`so I be lost`
`03:22.38`	`OloBola`	`a few days a year`
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`03:28.45`	`kolian123`	`i'm apt-get guy`
`03:28.47`	`OloBola`	`ok it wasn't that bad, the only thing I'm missing at this point is: libnspr4.so`
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`03:31.00`	`[TK]D-Fender`	`OloBola, that appears to be something you can nab another way. Try "yum install gaim" and the retry`
`03:32.13`	`OloBola`	`oh great, thanks`
`03:32.55`	`HaMYaI`	`is there any type of T1/E1 wiring config apart from "straight through" and "cross over"?`
`03:33.22`	`[TK]D-Fender`	`HaMYaI, that covers it.`
`03:33.54`	`HaMYaI`	`the one from telco device to my E1 seems to be 1245 and 5421 and it works`
`03:35.16`	`HaMYaI`	`[TK]D-Fender: it's a HDSL modem or something`
`03:35.26`	`crimethinker`	`sleepy time`
`03:36.11`	`[TK]D-Fender`	`HaMYaI, thats a straight cable`
`03:37.30`	`HaMYaI`	`[TK]D-Fender: hmm, thought it's gotta be 1245 and 1245`
`03:37.42`	`JT`	`that's crossover`
`03:37.47`	`JT`	`err`
`03:37.50`	`JT`	`wait`
`03:37.54`	`JT`	`that's straight :)`
`03:37.58`	`[TK]D-Fender`	`JT : Last I recall its 1/4 & 2/5`
`03:38.09`	`JT`	`12345678 > 12345678 = straight cable`
`03:38.25`	`JT`	`sangomas come with straight cables, but they're rubbish`
`03:38.32`	`JT`	`i refuse to use those supplied cables`
`03:38.36`	`[TK]D-Fender`	`JT : 12345678 = 12345678= straight cable, get your operators right too!`
`03:38.40`	`[hC]`	`they seemed alright to me, they were nice and flat`
`03:38.44`	`JT`	`flat cables`
`03:38.50`	`JT`	`[TK]D-Fender: that's what WRONG with them.`
`03:38.55`	`JT`	`[hC]: even`
`03:39.16`	`[TK]D-Fender`	`Screw it, I just used Cat5e :)`
`03:39.16`	`HaMYaI`	`JT: hi, yesterday I tried to link my Quad Tor2 to my Dialogic D/300 but it was unsucessfull`
`03:39.16`	`JT`	`flat cables in differential serial comms is a big NO`
`03:39.23`	`JT`	`[TK]D-Fender: they're better cables, cat5e :)`
`03:39.54`	`[TK]D-Fender`	`HaMYaI Thats because you need chan_smokesignal.so as the translator! You know how that legacy tech works!`
`03:40.08`	`[hC]`	`JT: in that, they are too easily damaged? or they too easily allow for interference?`
`03:40.50`	`HaMYaI`	`[TK]D-Fender: is it included in asterisk 1.4.6?`
`03:40.56`	`JT`	`[hC]: too easy for interference to be induced`
`03:41.02`	`JT`	`and it weakens the differential signals`
`03:41.18`	`[hC]`	`huh. ill have to replace the flat one i put in then. its been okay so far, but..`
`03:41.23`	`JT`	`you wouldn't use a flat cable for an ethernet LAN, why would you use it for T1/E1?`
`03:41.32`	`[TK]D-Fender`	`HaMYaI, No, smoked, Mesquite, teriyaki, and honey-garlic are ALL extra.`
`03:41.39`	`JT`	`they only get away with it because it's so short, i'd never do it`
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`03:59.28`	`Krooks`	`JT: there is no app called app`
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`03:59.34`	`Krooks`	`JT: there is no app called playback`
`03:59.34`	`troy-`	`JT all done :)`
`03:59.50`	`troy-`	`pbx installation = 3.5 hours`
`03:59.52`	`JT`	`Krooks: there is`
`04:00.04`	`Krooks`	`I looked for it everywhere`
`04:00.13`	`Krooks`	`do you have an url`
`04:00.13`	`Krooks`	`?`
`04:04.34`	`*** part/#asterisk HaMYaI (n=LAMER@ppp-58.8.12.130.revip2.asianet.co.th)`
`04:04.38`	`lowlevel`	`uhm... guys, I have a question for once.`
`04:05.23`	`lowlevel`	`I've been using the 'flash()' program for a while on a zap channel (pots from local telco) ... to do three way calling...`
`04:05.26`	`Krooks`	`give up. anyone got url for playback ? I wasted much time looking for it`
`04:05.41`	`lowlevel`	`recently I ditched that... and started using an IAX2 voip # as my main channel now...`
`04:05.56`	`lowlevel`	`and the flash() program only seems to work with Zap channels....`
`04:06.36`	`Krooks`	`I can't play the .gsm files created by idefisk. Any tip any one ?`
`04:06.56`	`lowlevel`	`... I guess my question is... can I do a threeway call with this type of channel, and how would I go about it...`
`04:06.56`	`JT`	`Krooks: i take it you're joking.`
`04:07.00`	`JT`	`Krooks: http://www.google.com.au/search?hl=en&q=asterisk+cmd+playback&btnG=Google+Search&meta=`
`04:07.06`	`Krooks`	`mplayer, xine and vplayer can't playit`
`04:07.09`	`JT`	`Playback is a very commonly used app`
`04:07.36`	`Krooks`	`Its a commadn`
`04:07.43`	`Krooks`	`Its a command`
`04:07.48`	`Krooks`	`on asterisk`
`04:08.02`	`Krooks`	`I'm on a client machine using idefisl`
`04:08.04`	`Krooks`	`I'm on a client machine using idefisk`
`04:08.41`	`Krooks`	`anyways, thanks. Just a misunderstanding`
`04:09.16`	`JT`	`well you were asking about asterisk`
`04:09.24`	`JT`	`you can use sox to convert .gsm files`
`04:10.26`	`Krooks`	`JT: thanks`
`04:12.10`	`[TK]D-Fender`	`Krooks, use Idefisk, dial an exten on *. Have it PLAYBACK your gsm file. End of story. This isn't Raw Cat Science!`
`04:13.49`	`Krooks`	`[TK]D-Fender: thanks.`
`04:14.40`	`Krooks`	`Sorry. I'm can be annoying sometimes. Sorry JT`
`04:16.14`	`*** join/#asterisk tako-san (n=Tako-san@216.232.147.102)`
`04:17.10`	`Krooks`	`Btw, idefisk save the conversations into date_input.gsm and date_output.gsm. I don't see anywhere is there an option to save it as one file. Any tip anyone ?>`
`04:17.41`	`JT`	`record with asterisk instead? ;)`
`04:17.46`	`JT`	`or convert them later`
`04:18.53`	`Krooks`	`So recording with idefisk, there is no other option, right`
`04:19.25`	`JT`	`don't know. check idefisk documentation`
`04:19.36`	`Krooks`	`I need like a audio app that can 'superimpose' the output on the input. :)`
`04:19.44`	`Krooks`	`or something like that`
`04:19.52`	`Krooks`	`ok thanks`
`04:20.18`	`JT`	`sox probably`
`04:20.21`	`[TK]D-Fender`	`yup`
`04:21.34`	`Krooks`	`ok I got to hear the .gsm files with soxplay`
`04:21.56`	`Krooks`	`I'll check sox on the joining part`
`04:26.20`	`*** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal)`
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`04:34.52`	`lowlevel`	`hmm, perhaps I need a different approach`
`04:36.20`	`[TK]D-Fender`	`lowlevel, you aren't dealing witha single channel any more. What you do is use your PHONE's 3-way calling feature and place another call out your peer.`
`04:37.16`	`Qwell`	`Krooks: I bet if you were to email the guys at asteriskguru, and tell them your idea, they might add it`
`04:38.05`	`OloBola`	`where can I download libnspr4.so?`
`04:38.14`	`[TK]D-Fender`	`OloBola, did you try what I suggested?`
`04:41.09`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`04:41.51`	`OloBola`	`yes I did. I tried but the install failed, along with a few others I tried.`
`04:42.18`	`OloBola`	`EOFError`
`04:42.32`	`lowlevel`	`tk: that might help if I had a phone ;)`
`04:42.52`	`*** join/#asterisk tuxd00d (n=tuxinato@ip70-190-51-106.ph.ph.cox.net)`
`04:43.51`	`[TK]D-Fender`	`lowlevel, What are you using?`
`04:44.05`	`[TK]D-Fender`	`lowlevel, * CLI to dial?`
`04:44.26`	`[TK]D-Fender`	`lowlevel, Because soft-phones have 3-way calling too...`
`04:44.56`	`lowlevel`	`tk: I'm calling into asterisk from a cellphone...`
`04:45.14`	`lowlevel`	`tk: its setup to ignore my call.. call me back`
`04:45.19`	`lowlevel`	`tk: then let me dial out.`
`04:45.32`	`lowlevel`	`basically using it to avoid outgoing minutes on my cellphone.`
`04:46.33`	`lowlevel`	`I could have it dial out on another DID I guess...`
`04:46.50`	`lowlevel`	`but... I liked the three way method it used to do before I had the voip did's`
`04:50.55`	`lowlevel`	`brb, gonna plug a phone it to see if I can even do it with a phone`
`04:51.09`	`[TK]D-Fender`	`lowlevel, You should be able to do multiple calls through your provider.`
`04:51.55`	`lowlevel`	`yeah..`
`04:51.56`	`lowlevel`	`I can`
`04:51.57`	`JT`	`lowlevel: i've done callback before, it's not that hard`
`04:52.09`	`lowlevel`	`asterisk is handling it fine`
`04:52.14`	`lowlevel`	`from an extension`
`04:52.54`	`[TK]D-Fender`	`lowlevel, So you call into . It calls you back. You then want to have do a 3-way conference between you and 2 other parties?`
`04:53.36`	`lowlevel`	`tk: no, I call into * ... it calls me back.. asks me # 2 call.. and threeways thatnumber with ME`
`04:53.49`	`[TK]D-Fender`	`lowlevel, well.. its jsut you and THEM, no?`
`04:53.55`	`lowlevel`	`tk: correct`
`04:53.57`	`lowlevel`	`well`
`04:54.03`	`[TK]D-Fender`	`lowlevel, then it isn't a 3 way call!`
`04:54.03`	`lowlevel`	`and asterisk`
`04:54.17`	`JT`	`...`
`04:54.24`	`[TK]D-Fender`	`lowlevel, 2 phones calling each other through * isn't a 3-way call!`
`04:54.25`	`JT`	`asterisk is not a party to the call`
`04:54.36`	`JT`	`it just happens to facilitate it`
`04:54.41`	`[TK]D-Fender`	`lowlevel, And yes, you can just dial out your same provider.`
`04:54.43`	`lowlevel`	`... ok. so it can hand off the call?`
`04:54.50`	`JT`	`err`
`04:54.53`	`JT`	`make call file`
`04:54.56`	`[TK]D-Fender`	`lowlevel, a call is a call is a call`
`04:54.56`	`JT`	`call file calls you back`
`04:55.04`	`lowlevel`	`jt: yeah I haev that done`
`04:55.04`	`JT`	`and drops you into DISA context or an IVR`
`04:55.05`	`[TK]D-Fender`	`JT : he's already got that part.`
`04:55.07`	`JT`	`problem solved`
`04:55.21`	`[TK]D-Fender`	`lowlevel, You've already done everything then. Whats the problem!?`
`04:55.40`	`lowlevel`	`;) perhaps I'm just thinking of it totally ass backwards`
`04:55.47`	`lowlevel`	`heres what it USED to do.`
`04:55.51`	`JT`	`a definite possibility ;)`
`04:55.53`	`lowlevel`	`I had an analog telephone line in`
`04:55.57`	`lowlevel`	`Zap/4`
`04:56.03`	`lowlevel`	`<PROTECTED>`
`04:56.06`	`lowlevel`	`asterisk ignores my call`
`04:56.10`	`lowlevel`	`then calls me back on this line.`
`04:56.17`	`lowlevel`	`asks me for whatever # I want to call`
`04:56.20`	`lowlevel`	`then three way calls that number.`
`04:56.34`	`JT`	`even better on digital, you can make it hang up without answering :)`
`04:56.35`	`lowlevel`	`so.. asterisk is making the call to me, and the third party`
`04:56.37`	`lowlevel`	`and its a three way call.`
`04:56.38`	`[TK]D-Fender`	`lowlevel, how is this ANY different with your ITSP!?`
`04:56.58`	`lowlevel`	`now.. I've ditched that POTS line... and i've got this VOIP DID`
`04:57.01`	`JT`	`lowlevel: no, it's making a call to you, and giving you a DISA dialtone`
`04:57.08`	`[TK]D-Fender`	`lowlevel, And stop calling it a 3 way jut because * RANG YOU and not the other way around. Its YOU in the DIALPLAN doing whatever`
`04:57.19`	`JT`	`lowlevel: then it makes another leg of the call when you give it a number, and they are bridged`
`04:59.22`	`lowlevel`	`yeah, using the three way calling feature from my telephoen service provider.`
`05:00.02`	`JT`	`yeah well you need 2 lines`
`05:00.04`	`JT`	`really`
`05:00.06`	`lowlevel`	`now... I guess the issue now, is that I'm still trying to do it htat way`
`05:00.15`	`lowlevel`	`and it can't really do that`
`05:00.48`	`[TK]D-Fender`	`lowlevel, then only a 3-way call because you were using 1 stupid telephone line for 2 calls and * shouldn't count. Either way you just dial out your ITSP. End of story.`
`05:01.14`	`[TK]D-Fender`	`lowlevel, You don't "flash" your call. That idea is NON-EXISTANT. Just dialout it again!`
`05:01.49`	`lowlevel`	`tk: ok.`
`05:01.58`	`lowlevel`	`(yep, ass backwards ;)`
`05:02.14`	`lowlevel`	`thanks , hoepfully that answers my question`
`05:02.15`	`[TK]D-Fender`	`lowlevel, Virtually every ITSP lets you use at least 2 channels at a time.`
`05:02.20`	`lowlevel`	`tk: awesome`
`05:02.30`	`[TK]D-Fender`	`lowlevel, many up to 5, or more depending.`
`05:02.33`	`lowlevel`	`wow`
`05:02.34`	`lowlevel`	`:)`
`05:02.43`	`JT`	`some unlimited :)`
`05:02.59`	`lowlevel`	`one reason I liked the flash method or three way method.. was that I could have asterisk 'drop off'`
`05:03.04`	`lowlevel`	`and the line was then available`
`05:03.06`	`[TK]D-Fender`	`lowlevel, Thats because when you get down to it a 3-way call is just 2 independant calls that the telco bridges for you.`
`05:03.08`	`lowlevel`	`(centrix)`
`05:03.21`	`lowlevel`	`er centrex`
`05:03.30`	`[TK]D-Fender`	`lowlevel, Ah yes, wll thats a rare telco service`
`05:03.57`	`[TK]D-Fender`	`lowlevel, I've seen it before. Thats technically what happens on a Polycom SIP phone when the host hangs up a 3-way call.`
`05:03.57`	`lowlevel`	`tk: yeah, no problem... I'm just trying to clear it up/figure it out withoutgetting too technical`
`05:04.12`	`[TK]D-Fender`	`the other 2 parties stay connected.`
`05:04.20`	`lowlevel`	`yeah.. if you have centrex lines, they do that`
`05:04.28`	`lowlevel`	`you drop off.. and the two parties can talk.. on your bill`
`05:05.55`	`[TK]D-Fender`	`lowlevel, Ok, THAT is all but certinly NOT happing in your ITSP scenario.`
`05:06.10`	`lowlevel`	`not anymore.`
`05:07.36`	`[TK]D-Fender`	`lowlevel, it is remotely possible that the TRANSFER app MIGHT work, but the likelyhood is dreadfully slim.`
`05:07.49`	`OloBola`	`hey TK, where do I need to put libnspr4.so? I got it finally.`
`05:08.10`	`[TK]D-Fender`	`OloBola, no idea. did you try the method I told you to?`
`05:08.46`	`OloBola`	`I tried but I kept getting errors`
`05:08.56`	`[TK]D-Fender`	`OloBola, even on just GAIM?`
`05:09.20`	`OloBola`	`yep, tried a few others too`
`05:09.29`	`[TK]D-Fender`	`OloBola, eek.... ok well GL...`
`05:09.35`	`[TK]D-Fender`	`bed time...`
`05:10.13`	`OloBola`	`good night, thanks`
`05:13.01`	`litage_`	`is it normal behaviour for asterisk to email voicemail notifications that have already been sent when asterisk is restarted?`
`05:13.15`	`lowlevel`	`tk: nod, yeah doubt it too`
`05:20.53`	`OloBola`	`Does anyone know where I need to place libnspr4.so to get lumenvox running?`
`05:21.23`	`OloBola`	`it's dependency that's not on my machine, lumenvox needs it.`
`05:22.15`	`*** join/#asterisk InHisName (n=Administ@c-71-225-221-149.hsd1.pa.comcast.net)`
`05:24.28`	`Aces1Up`	`is there a way to tell if ztdummy is loaded or not in asterisk?`
`05:25.19`	`JT`	`not in asterisk, but: lsmod\|grep ztdummy`
`05:26.41`	`Aces1Up`	`jt thanks.`
`05:29.06`	`Aces1Up`	`jt daaang, hrmm i can load the zaptel and that shows in lsmod, but not ztdummy, does that mean i didn't compile it correctly?`
`05:29.38`	`JT`	`maybe, or it hasn't been loaded`
`05:29.53`	`Aces1Up`	`do you load ztdummy as a module like zaptel?`
`05:30.00`	`JT`	`yes`
`05:30.18`	`JT`	`you don't need to load zaptel, anything that depends on it will load it`
`05:30.31`	`Aces1Up`	`hrmm ok, well ztdummy, is not in the modeprobe.d directory so i assume it has not been compiled correctly.`
`05:31.18`	`Aces1Up`	`in the future ot telephone book it says you have to uncomment a line in the makefile to compile ztdummy, but i did not see that line in the latest version of the zaptel drivers.`
`05:32.19`	`JT`	`have you tried just manually loading it?`
`05:32.57`	`Aces1Up`	`the only way i know how to manaully load it is via the modprobe command.. and there is no module for ztdummy in the modprobe.d directory.`
`05:33.05`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`05:34.36`	`Aces1Up`	`well i found this guide.. so will try it and see what happens http://www.aussievoip.com/wiki/ztdummy`
`05:35.18`	`JT`	`that doesn't sound like the normal location for modules`
`05:35.27`	`Aces1Up`	`hrmm, using fedora 7`
`05:35.54`	`Aces1Up`	`that is the directory the install put the zaptel module in.`
`05:37.04`	`Aces1Up`	`is it a bad idea to recombile and build the zaptel and ztdummy module if the asterisk has already been compiled?`
`05:37.28`	`JT`	`might be alright`
`05:37.33`	`JT`	`if it's the same version`
`05:37.33`	`Aces1Up`	`lol`
`05:37.36`	`JT`	`anyway`
`05:37.37`	`Aces1Up`	`yes it is.`
`05:37.42`	`JT`	`updatedb;locate ztdummy`
`05:42.31`	`Aces1Up`	`USR/SRC/ASTERISK-1.4.6/ZAPTEL-1.4.3/.TMP/VERSIONS/ZTDUMMY.MOD`
`05:42.35`	`Aces1Up`	`is that it?`
`05:42.59`	`JT`	`nup`
`05:43.32`	`Aces1Up`	`hrmm i think i have to recompile.`
`05:44.31`	`OloBola`	`Does anyone know where I need to place libnspr4.so to get lumenvox running?`
`05:44.43`	`Aces1Up`	`well what the heck.`
`05:44.51`	`Aces1Up`	`i just did a modprobe ztdummy and it loaded it!`
`05:44.57`	`Aces1Up`	`where the heck is the file?`
`05:45.51`	`*** join/#asterisk troy- (n=tabmeist@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com)`
`05:46.06`	`troy-`	`how can cordy be at home and work at the same time :P j/k`
`05:47.09`	`JT`	`Aces1Up: did you updatedb first?`
`05:47.12`	`JT`	`as root`
`05:47.51`	`troy-`	`JT i thought i was going to be at work till tomorrow morning setting this up, in and out in 3.5 hours`
`05:48.20`	`JT`	`cool :)`
`05:49.04`	`Aces1Up`	`yes`
`05:49.08`	`Aces1Up`	`jt yes`
`05:49.22`	`OloBola`	`local variable 'confname' referenced before assignment`
`05:50.37`	`JT`	`OloBola: warning or error?`
`05:51.10`	`troy-`	`JT whats the best way for asterisk to recognize SIP-based faxing`
`05:51.24`	`JT`	`magic`
`05:51.33`	`troy-`	`i have the fax machine hooked up to a Cisco 186I1 ATA`
`05:51.34`	`Aces1Up`	`has anyone used both app_conference and meetme? is there one that utilizes bandwidth better?`
`05:51.46`	`troy-`	`but how can i route fax calls to that extension?`
`05:52.22`	`JT`	`Aces1Up: why would it change bandwidth usage at all?`
`05:52.28`	`JT`	`troy-: how do faxes come in?`
`05:52.56`	`troy-`	`over a SIP trunk`
`05:53.23`	`Aces1Up`	`jt, i was reading that one can handle more users.`
`05:53.30`	`Aces1Up`	`app_conference.`
`05:53.42`	`Aces1Up`	`nevermind.`
`05:54.04`	`JT`	`it might`
`05:54.12`	`JT`	`troy-: no such thing as a sip "trunk" :)`
`05:54.15`	`JT`	`troy-: from where`
`05:54.18`	`JT`	`an itsp?`
`05:54.23`	`troy-`	`yes`
`05:55.40`	`troy-`	`we have 4 channels from the provider`
`05:56.44`	`JT`	`troy-: forget about faxing over sip`
`05:56.49`	`JT`	`especially over the Internet`
`05:56.51`	`JT`	`not going to happen`
`05:56.54`	`troy-`	`that bad?`
`05:57.18`	`JT`	`yes, modem signals over laggy/lossy voip connections do not work`
`05:57.45`	`troy-`	`there is roughly 80ms of latency between our PBXs`
`05:58.10`	`JT`	`not good enough, it needs to be constant very low latency, no jitter, no packet loss, or it has no chance of working`
`05:59.49`	`troy-`	`JT damned`
`06:00.04`	`troy-`	`JT bell is coming to install a POTS line tomorrow at the expense of $44 per month =P`
`06:00.21`	`JT`	`wow you guys really do get ripped off in the US`
`06:00.33`	`troy-`	`canada`
`06:00.38`	`JT`	`ah`
`06:00.58`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`06:01.13`	`*** join/#asterisk sergee (n=sergee@195.94.224.197)`
`06:01.25`	`JT`	`here in .au, standard ripoff price for a POTS line is AUD$35 which is about USD$30, but you can get them for AUD$20`
`06:02.05`	`troy-`	`well keep in mind these are business lines, so bell feels they may as well charge extra`
`06:02.24`	`JT`	`heh`
`06:02.28`	`JT`	`same for business here`
`06:02.31`	`JT`	`same price`
`06:02.33`	`JT`	`as home`
`06:03.30`	`troy-`	`what i'd like to do ideally, is get an integrated T1 (split data and voice channels)`
`06:03.46`	`troy-`	`unfortunately here T1s start at like $600 for data-only, making it too costly`
`06:04.06`	`JT`	`cool in theory, but i think you can only get the voice sections in those with evil CAS T1 signalling`
`06:04.28`	`JT`	`you know there are fax to email services for faxing? much cheaper than installing a landline`
`06:05.09`	`troy-`	`you are correct :> unfortunately we use big multi-function printers at the office, and everyone wants to be able to receive/send faxes through them`
`06:05.23`	`JT`	`right`
`06:05.37`	`JT`	`you need to use T.38 to do realtime faxing over lossy networks`
`06:05.49`	`troy-`	`yeah, sounds about right`
`06:06.17`	`troy-`	`i actually reinstalled the os + asterisk because we were having quality issues where cals would either drop or cut in and out, couldnt figure out what was wrong`
`06:06.48`	`*** part/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV)`
`06:07.03`	`troy-`	`ITSP was blaming me and i was blaming them :P`
`06:07.36`	`JT`	`with faxing?`
`06:07.49`	`troy-`	`no voice`
`06:08.13`	`JT`	`ah ok`
`06:08.53`	`troy-`	`problem was that there were no errors being written to logs, so it was nearly impossible to figure out what was wrong`
`06:09.25`	`JT`	`there usually isn't`
`06:09.30`	`JT`	`usually the symptoms tell you`
`06:10.01`	`troy-`	`but how do you tie the symptoms to a solution?`
`06:10.36`	`JT`	`by diagnosing the problem`
`06:11.03`	`troy-`	`(what would you do if someone said they were experiencing dropped calls and quality degredation)`
`06:11.21`	`lowlevel`	`go with gut? heh`
`06:11.48`	`JT`	`troy-: i'd listen to the calls, and say what was wrong with them :)`
`06:12.39`	`troy-`	`heh, usually they start getting choppy then one party cant hear the other for 15 seconds then quality is restoerd`
`06:12.46`	`lowlevel`	`somehow.. there alwasy seems to be 45 coke cans on my desk`
`06:13.02`	`bcnl`	`can the IMAP voicemail storage of asterisk 1.4.x put the messages on a exchange server allongside the users normal email?`
`06:14.29`	`Aces1Up`	`lowlevel always exatly 45?`
`06:15.11`	`*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)`
`06:15.40`	`JT`	`troy-: what sort of connection is this?`
`06:16.02`	`troy-`	`we used to have metro ethernet and now business ADSL, same problem on both`
`06:16.36`	`JT`	`what phones?`
`06:17.40`	`troy-`	`cisco 7941 running SIP`
`06:17.51`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`06:17.53`	`JT`	`hmm ok`
`06:18.09`	`JT`	`behind nat?`
`06:18.25`	`troy-`	`correct the server and phones are on the same subnet`
`06:19.14`	`litage_`	`is it normal behaviour for asterisk to email voicemail notifications that have already been sent when asterisk is restarted?`
`06:21.37`	`troy-`	`JT one thought was to add QoS to the network to see if that was it, but chances are that wont change much`
`06:21.46`	`JT`	`err`
`06:21.53`	`JT`	`is there nat anywhere?`
`06:24.01`	`troy-`	`all the phones and server are on an internal subnet`
`06:24.33`	`troy-`	`since the server doesnt have an external IP technically there would be NAT there`
`06:31.08`	`troy-`	`JT ? :(`
`06:32.00`	`JT`	`well`
`06:32.03`	`JT`	`do calls go over that nat`
`06:32.35`	`troy-`	`they have to`
`06:33.13`	`JT`	`what is doing the nat?`
`06:33.59`	`troy-`	`Linksys RVS4000`
`06:34.45`	`lowlevel`	`aces: well... no.. not always.`
`06:34.53`	`lowlevel`	`aces: sometimes.. theres 43`
`06:34.56`	`lowlevel`	`or.. 21`
`06:35.00`	`lowlevel`	`or 185`
`06:35.20`	`lowlevel`	`sometimes theres other cans`
`06:35.26`	`lowlevel`	`like.. root beer`
`06:35.31`	`lowlevel`	`or gingerale`
`06:35.37`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`06:36.08`	`lowlevel`	`night guys`
`06:36.10`	`lowlevel`	`crashing hard.`
`06:37.26`	`troy-`	`JT think it could be a feature on the router?`
`06:38.13`	`*** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it)`
`06:40.02`	`JT`	`never heard of that model`
`06:40.09`	`JT`	`but you never know, it could be causing an issue`
`06:41.17`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`06:42.08`	`*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)`
`06:42.34`	*** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
`06:43.41`	`troy-`	`JT can you think of anyother possibilities?`
`06:44.11`	`JT`	`troy-: can't temporarily change the router?`
`06:44.37`	`troy-`	`i dont have anything to swap it with, problem is that the issues are infrequent and dont follow a pattern`
`06:45.28`	`flenders`	`troy-: swap it for a linux box`
`06:45.57`	`flenders`	`any box with 2 network cards will do`
`06:46.07`	`troy-`	`flenders, we dont even know if thats the issue :)`
`06:46.28`	`JT`	`your alternative was reinstall asterisk....`
`06:46.37`	`JT`	`which will probably do nothing`
`06:46.46`	`troy-`	`JT its made a noticable improvement in sound quality`
`06:47.14`	`troy-`	`calls are more crisp and have less jitter, wont know if the problems are a thing of the past till call volume picks up tomorrow`
`06:48.46`	`JT`	`heh`
`06:48.55`	`JT`	`you should get proper phone lines`
`06:49.07`	`JT`	`VoIPoI is not usually the same quality`
`06:51.41`	`troy-`	`JT yeah, imagine how much 6 channels would cost on a PRI + LD compared to VoIP`
`06:51.45`	`troy-`	`probably 10x more`
`06:52.03`	`JT`	`depends on how expensive it is in your area`
`06:52.12`	`JT`	`but it's silly to run businesses off VoIPoI`
`06:52.14`	`troy-`	`well if a single POTS line is 44 bucks :P`
`06:54.28`	`creativx`	`JT whats the last oI for`
`06:54.35`	`JT`	`over Internet`
`06:54.39`	`troy-`	`over intarweb`
`06:54.49`	`creativx`	`yeah that was my assumption`
`06:54.59`	`creativx`	`i can see the need to be blear`
`06:55.03`	`creativx`	`*clear`
`06:55.09`	`troy-`	`heh`
`06:55.16`	`JT`	`clear as a pri call`
`06:55.20`	`creativx`	`we run voipoi`
`06:55.26`	`JT`	`only?`
`06:55.28`	`creativx`	`yes`
`06:55.34`	`JT`	`not so smart :P`
`06:55.44`	`creativx`	`I dont see why not`
`06:55.48`	`JT`	`...`
`06:55.58`	`creativx`	`If our net goes down, everything else we do goes down too`
`06:56.04`	`troy-`	`heh`
`06:56.04`	`creativx`	`so why should we let people call us and nag about it ;)`
`06:56.07`	`troy-`	`thats interesting :)`
`06:56.09`	`JT`	`so many variables that can impact upon your phones`
`06:56.24`	`JT`	`creativx: do you have any landlines?`
`06:56.30`	`creativx`	`yes i have aswell`
`06:56.40`	`Swat2`	`Anyone able to refer me to some doco on how to get SLA (Shared Line Appearances) to work w/ asterisk 1.4.x ?`
`06:56.41`	`creativx`	`havent gotten rid of them quite yet :)`
`06:56.51`	`creativx`	`they are connected to our old ericsson pbx`
`06:56.54`	`JT`	`don't get rid off them you need them for emergencies`
`06:56.59`	`creativx`	`i need em for fax`
`06:57.00`	`creativx`	`aswell`
`06:57.10`	`JT`	`it's best to have 1 not connected to a pbx at all but just a basic phone`
`06:57.37`	`creativx`	`well voop lets us set up numbers that our inbound calls will be redirected to if we are unavailable`
`06:57.48`	`JT`	`i mean for outbound`
`06:57.51`	`creativx`	`so having that + a basic line would be enough`
`06:57.52`	`creativx`	`we have cellphones`
`06:57.53`	`JT`	`emergency calling`
`06:58.01`	`JT`	`cellphones don't work in big emergencies`
`06:58.09`	`creativx`	`its a small country`
`06:58.10`	`creativx`	`;)`
`06:58.14`	`creativx`	`we only have small emergencies`
`06:58.18`	`flenders`	`haha`
`06:58.20`	`creativx`	`but yes your point is taken`
`06:58.26`	`JT`	`everyone assumes cellphones will always work, they're one of the least reliable forms of comms on the pstn`
`06:58.30`	`JT`	`what country?`
`06:58.32`	`creativx`	`norway`
`06:58.36`	`JT`	`ah`
`06:58.40`	`creativx`	`our gsm network is fairly decent`
`06:58.45`	`creativx`	`atleast in this geographical area`
`06:59.02`	`*** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net)`
`06:59.07`	`JT`	`trust me, it won't last for very long during a major power outage or similar problem`
`06:59.27`	`*** part/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net)`
`06:59.29`	`JT`	`most cell networks won't survive a loss of power for over 30-120mins`
`07:00.06`	`creativx`	`thats why i was planning on keeping 3 analog lines.. 2 for fax and 1 for backup phone. it should be a red phone`
`07:00.29`	`JT`	`yeah, i've seen ex us-military/fed phones on ebay us that are red ;)`
`07:01.55`	`*** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)`
`07:02.06`	`creativx`	`heheh, that would be classy`
`07:02.10`	`*** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)`
`07:02.18`	`*** join/#asterisk angryuser (n=Miranda@df01t2-212-194-207-236.d4.club-internet.fr)`
`07:02.39`	`angryuser`	`misdn_l2l1_check application is implemented only in 1.4.5?`
`07:04.52`	`*** join/#asterisk matsk (n=mk@194.68.102.173)`
`07:04.58`	`troy-`	`JT yeah most have DC battery packs, but no genset on site`
`07:05.16`	`JT`	`right`
`07:05.40`	`JT`	`where was that massive blackout a couple years back, new york?`
`07:05.43`	`troy-`	`CO will almost always have a generator with racks of batteries and a transfer switch`
`07:05.49`	`troy-`	`yeah`
`07:05.50`	`JT`	`their cellphones were useless`
`07:05.55`	`JT`	`did not work at all`
`07:06.06`	`JT`	`after a short amount of time`
`07:06.33`	`troy-`	`JT problem is now that a lot of PSTN access is switched over fiber ATM with muxes, there often isnt power to supply those during a failure`
`07:06.35`	`*** join/#asterisk jmls (n=jmls@62.49.235.130)`
`07:06.36`	`JT`	`yeah the transfer switch is only for the input to the rectifiers`
`07:06.45`	`troy-`	`very common now in office buildings`
`07:06.54`	`JT`	`the equipment is always linked to the batteries via bus bar`
`07:07.01`	`troy-`	`yup :)`
`07:07.05`	`JT`	`yeah that's pretty nasty`
`07:07.26`	`troy-`	`well its much cleaner then having a few hundred pairs demarced into the building`
`07:07.52`	`JT`	`not if they aren't going to power it properly`
`07:08.02`	`JT`	`anyway you need that copper for dsl`
`07:08.02`	`troy-`	`heh heh :>`
`07:08.22`	`troy-`	`copper pairs can be muxed off a DS3`
`07:08.33`	`Krooks`	`anyone wrote a script to combine a out.gsm file and a input.gsm file.`
`07:08.48`	`troy-`	`DS3 is muxed off an OC3 which provides capacity for 3 DS3s`
`07:09.10`	`JT`	`sure`
`07:09.19`	`JT`	`you need a mini DSLAM to provide dsl though`
`07:09.23`	`JT`	`at the mdf`
`07:09.38`	`JT`	`you can't just mux a few MHz into 64000bit/s`
`07:09.41`	`troy-`	`often worth it if you have lots of subs in a high concentration area`
`07:10.59`	`troy-`	`JT not to mention usually they will just do ethernet instead of wasting their time with DSLAMs`
`07:12.15`	`JT`	`sure, that assumes that the customer wants to get Internet off whoever is providing the mux`
`07:12.35`	`troy-`	`when you have fiber coming in, technically there are plenty of possibilities`
`07:13.47`	`JT`	`i realise`
`07:13.58`	`troy-`	`our last office was in a cogent powered-building, we had diverse fiber coming in (dual GigE), 100Mbps to our floor and 10Mbps service bidrectional`
`07:14.21`	`JT`	`heh`
`07:14.51`	`troy-`	`the building was probably 20 stories high, so it was worth it for the provider to do that`
`07:15.20`	`JT`	`i don't doubt it`
`07:17.30`	`*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)`
`07:17.31`	`troy-`	`the best i have seen is with housing developments, the developers sell off sole telecom operating rights to a subsidiary`
`07:17.45`	`troy-`	`guaranteed revenue for a predefined period of time`
`07:18.02`	`JT`	`not that good for the consumer`
`07:18.09`	`JT`	`people don't want to be forced into a provider`
`07:18.18`	`troy-`	`you want a POTS line? that will be $1000 =P`
`07:19.27`	`JT`	`choice in everything`
`07:19.30`	`JT`	`not just pots`
`07:20.21`	`troy-`	`yeah TV / Internet / Phone`
`07:21.52`	`JT`	`yeah, i wouldn't want to be forced them`
`07:24.17`	`snuff-work`	`yer there has been a few in australia now like that troy`
`07:29.30`	`troy-`	`snuff-work, yeah not surprised`
`07:29.52`	`troy-`	`if you dont put the extra - on my name it doesnt alert me :)`
`07:38.31`	*** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net)
`07:38.36`	`*** join/#asterisk yassaccan (n=yassacca@admin197.hgo.se)`
`07:41.45`	`*** join/#asterisk ikaro (n=ikaro@pdpc/supporter/active/ikaro)`
`07:41.59`	`ikaro`	`hi`
`07:43.54`	*** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net)
`07:44.34`	`Swat2`	`any ideas why h264 video calls wont work like so: <videophone> <-sip-> <asterisk1.4> <-IAX2 Trunk-> <asterisk1.4> <-sip-> <videophone> but will work like: <videophone> <-sip-> <asterisk1.4> <-SIP Trunk-> <asterisk1.4> <-sip-> <videophone>`
`07:47.48`	`kaldemar`	`Swat2: the IAX2 trunk is probably dropping the video stream.`
`07:49.23`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.10)`
`07:50.29`	`tengulre`	`hi,all`
`07:51.58`	`JT`	`snuff-work: onenetwork eh?`
`07:52.33`	`snuff-work`	`yep`
`07:53.11`	`creativx`	`triple play`
`07:53.16`	`creativx`	`innit`
`07:53.18`	`JT`	`snuff-work: hosted pbx?`
`07:53.21`	`JT`	`no...`
`07:53.30`	`snuff-work`	`yes jt`
`07:53.35`	`Swat2`	`kaldemar: any way to fix this?`
`07:53.40`	`JT`	`mostly asterisk?`
`07:53.49`	`snuff-work`	`yer`
`07:53.57`	`JT`	`snuff-work: you guys run your own datacentres, or just have space in some?`
`07:54.12`	`snuff-work`	`heh i wish own dc`
`07:54.37`	`JT`	`is there any decent half carrier independant datacentres in melbourne?`
`07:55.35`	`snuff-work`	`don't know of any.. we just use our carriers dc's`
`07:55.40`	`*** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md)`
`07:55.45`	`JT`	`ah ok`
`07:55.48`	`JT`	`is it decent?`
`07:56.46`	`snuff-work`	`haven't had any issues`
`07:56.52`	`KpoH`	`hi all, I can't find in asterisk 1.4 app_pgsql, what do I do wrong? :)`
`07:58.32`	`snuff-work`	`this would be a cool dc to be in..`
`07:58.34`	`snuff-work`	`http://searchdatacenter.techtarget.com/qna/0,289202,sid80_gci1262497,00.html?track=NL-455&ad=595792&asrc=EM_NLT_1719830&uid=6261664`
`07:59.28`	`*** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it)`
`08:00.29`	`JT`	`snuff-work: yeah, i'm in the sydney one`
`08:00.32`	`JT`	`pretty good`
`08:00.38`	`JT`	`but globalswitch sydney is even better`
`08:00.43`	`JT`	`in terms of redundancy`
`08:00.50`	`JT`	`but security is iffier`
`08:01.07`	`troy-`	`JT my redundancy has always simply meant multiple sites`
`08:01.25`	`JT`	`troy-: it'd be nice for the sites to be reliable too`
`08:01.36`	`ikaro`	`i've gotten the exciting job of setting up a asterisk box to use in our company ( 30 people ) the only problem is im very green in this area, i've got * installed in a gentoo box and i got an account from voxbone, is this all that I will need ?`
`08:01.37`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`08:01.38`	`troy-`	`lots of things would be nice :)`
`08:02.14`	`troy-`	`JT i wouldnt mind being in the voip carrier business, but i've only got a few months till i have to quit work for university :)`
`08:02.46`	`JT`	`ikaro: you should look at getting a PRI service from a telco ideally, or failing that, some POTS lines`
`08:03.09`	`JT`	`troy-: there's no money in the bargain end of town`
`08:03.44`	`KpoH`	`hm, i found app_sql_pgsql in asterisk 1.2 but not in asterisk 1.4, what happens with them?`
`08:04.00`	`JT`	`odbc probably, KpoH`
`08:04.09`	`ikaro`	`jt, so you have some reference to what a pri service is, telco and pots ?`
`08:04.16`	`JT`	`ikaro: yes`
`08:04.18`	`JT`	`~thebook`
`08:04.19`	`jbot`	`i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`08:04.21`	`ikaro`	`i guess I'll have to do some googlingh`
`08:04.40`	`KpoH`	`JT: but i need app_ to do custom SQL query in AEL for example`
`08:04.47`	`ikaro`	`but you say, i need a pri service from telco or else som pots lines`
`08:04.52`	`JT`	`KpoH: odbc can do the same thing`
`08:04.54`	`troy-`	`JT you mean volume is required in order to make cash?`
`08:05.06`	`JT`	`ikaro: it would be best, yes`
`08:05.22`	`JT`	`troy-: what cash? those people run on tight margins`
`08:05.28`	`ikaro`	`jt, so i dont need voxbone at all ?`
`08:05.57`	`snuff-work`	`going in for the retial user there is nothing in it for a company`
`08:06.21`	`troy-`	`i wanted to target corporations who have existing PRI capacity from a LEC in cogent-powered buildings and convert them over`
`08:06.30`	`JT`	`ikaro: depends, you can use a hybrid system if it seems more economical, but VoIP over the Internet will not be as reliable and high quality as a PRI`
`08:07.28`	`KpoH`	`is obdc performance good enough?`
`08:07.37`	`JT`	`i don't see why not`
`08:07.48`	`ikaro`	`jt, the decision makers are always in favour of enconomical solutions .. even when it ends up costing more in the end .. short sighted.`
`08:08.01`	`JT`	`true`
`08:08.18`	`JT`	`ikaro: a lot of factors can degrade performance over the Internet`
`08:08.39`	`ikaro`	`jt, we bought a 2mbit/768 line dedicated for the voip`
`08:09.04`	`ikaro`	`but yes.. other factors can play in and degrade performance.`
`08:09.15`	`JT`	`ikaro: upload isn't enough to do more than 9 ulaw/alaw codec calls`
`08:09.44`	`JT`	`ikaro: you've got to have emergency services always available in case you ever need it`
`08:09.49`	`JT`	`or the company could be sued`
`08:09.52`	`angryuser`	`JT: the manufacture of * hardware cost nothing as i heard, like 150-200$ of 2000$ card, echo cansellation included`
`08:09.56`	`JT`	`and more importantly someone could die`
`08:10.03`	`JT`	`angryuser: i don't doubt it`
`08:10.27`	`troy-`	`thats like saying the manufacturing of a car costs like nothing :)`
`08:10.49`	`angryuser`	`troy-: do you got any FACTS ?`
`08:10.52`	`JT`	`echocancellation is all octastic firmware licensing costs`
`08:10.56`	`JT`	`and profit`
`08:11.17`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`08:11.20`	`ikaro`	`its so overwelming the amount of information needed to be digested and understood before even getting started .. this factor will delay the whole project alot :(`
`08:11.32`	`JT`	`angryuser: obvious to anyone who knows anything about electronics manufacture, the item often costs 1/10th to 1/5th what it sells for`
`08:12.31`	`troy-`	`angryuser, both products are sold by for-profit corporations, of course there is a significant markup involved`
`08:12.41`	`snuff-work`	`very true.. cause some poor bastid had to design it..`
`08:12.51`	`snuff-work`	`or even worse some of the money is licensing fees`
`08:13.07`	`JT`	`it's mostly supply & demanding`
`08:13.09`	`JT`	`demand`
`08:13.12`	`angryuser`	`JT: i am talking about digium hw, design dont cost that much, the just want all that profit,`
`08:13.34`	`JT`	`yeah design isn't as much as people make out`
`08:13.40`	`JT`	`this stuff isn't that groundbreaking`
`08:13.45`	`troy-`	`angryuser, if the card costs $150 and they sold it for $180 how are they going to pay salary, rent, development etc?`
`08:14.34`	`angryuser`	`troy-: not even funny, lover price greater turnover`
`08:14.54`	`angryuser`	`troy-: better * advertising and integration`
`08:14.58`	`JT`	`err`
`08:15.02`	`JT`	`this is asterisk`
`08:15.10`	`troy-`	`the volume isnt high enough to make up for an 80% loss of profit`
`08:15.13`	`JT`	`you need something to be in HIGH DEMAND to get HIGH TURNOVER`
`08:15.32`	`troy-`	`JT case in point :)`
`08:16.33`	`angryuser`	`JT: you will get hight demand if price go down, when you buy a cheapest tdm400, you rethink twice before buy`
`08:16.49`	`JT`	`angryuser: this is still a specialist field`
`08:16.52`	`angryuser`	`JT: it is like 400� in europe`
`08:16.55`	`JT`	`it's not exactly RAM chips`
`08:17.00`	`JT`	`or bread`
`08:17.13`	`troy-`	`angryuser, i disagree but a cisco call manager solution costs 10x more then a similar asterisk-pbx implementation`
`08:17.49`	`JT`	`angryuser: the cheapest tdm400p is 400eur for you? i suggest you find a new supplier`
`08:18.34`	`angryuser`	`troy-: wemm the asterisk could win more field, only rich companies can buy all cisco stuff, * is meant to be integrated in middle business also`
`08:18.59`	`troy-`	`angryuser, asterisk is cheap enough for its market segment, i'd say they could probably even raise prices if anything`
`08:20.03`	`JT`	`angryuser: middle businesses can afford asterisk generally`
`08:20.16`	`JT`	`if they can afford a key system, they can afford asterisk based solutions`
`08:20.17`	`troy-`	`our 4 user + fax asterisk implementation, cost $2,500 using mid-range gear`
`08:20.26`	`angryuser`	`troy-: i never said that it is costly, * casts hale or 1/3 of proprietary system, but i heard some info from one person about profits, they can bring prices down`
`08:20.41`	`angryuser`	`*costs half`
`08:21.09`	`troy-`	`who cares? i am more then willing to pay a few hundred for a card :)`
`08:21.26`	`troy-`	`the application software is open source, what more could you want..`
`08:21.41`	`*** join/#asterisk uwe (n=uwe@dogbert.palnet.com)`
`08:23.20`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`08:23.23`	`angryuser`	`troy-: i care, a lot of countrys under developpement unable to acces to hw because of it's price`
`08:23.49`	`troy-`	`so use voip without physical telco service`
`08:23.50`	`JT`	`boo hoo`
`08:23.57`	`JT`	`seriously`
`08:24.09`	`JT`	`a pentium IV is also expensive in these countries`
`08:24.13`	`JT`	`such is life`
`08:24.25`	`troy-`	`Jt wow we get along so well :)`
`08:24.53`	`JT`	`they can save money by reusing old stuff, and coming up with clever solutions`
`08:24.57`	`JT`	`like i bet they already have`
`08:25.34`	`JT`	`bottom line is companies are here to make money, and i'd rather they be here than not here if they provide something.`
`08:25.44`	`JT`	`something usefu.`
`08:25.48`	`JT`	`useful :)`
`08:26.24`	`angryuser`	`off 30 min`
`08:28.21`	`OloBola`	`well I finally got lumenvox to install`
`08:28.30`	`OloBola`	`can't get the example to build though`
`08:28.40`	`OloBola`	`here is the result: http://www.pastebin.ca/602718`
`08:28.46`	`angryuser`	`http://www.rowetel.com/ucasterisk/hardware.html troy-: JT: just look what it is, and i am sure you will understand what i am talking about`
`08:29.23`	`troy-`	`any idea what a pair of nike shoes cost?`
`08:30.12`	`JT`	`angryuser: sorry, what's the relevance of that site to the discussion?`
`08:30.29`	`*** join/#asterisk rcy (n=rcyeske@DSL-209.90.181.71.Vancouver.primus.ca)`
`08:31.41`	`angryuser`	`JT: open hardware`
`08:31.50`	`JT`	`yeah open hardware is cool`
`08:32.42`	`JT`	`some stuff is hard to make even if you had the schematics`
`08:34.08`	`*** part/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md)`
`08:38.42`	`angryuser`	`JT: the best choice would be to buy already printed board, without components`
`08:38.54`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`08:38.59`	`JT`	`it's not a good choice for a lot of people`
`08:39.08`	`OloBola`	`example.cpp:1:26: error: LVSpeechPort.h: No such file or directory`
`08:39.10`	`JT`	`and soldering SMDs is not easy anyway`
`08:39.29`	`JT`	`businesses don't want to construct their own telephony boards anyway, that's madness`
`08:42.40`	`Swat2`	`hmmm`
`08:42.48`	`Swat2`	`nope h264 not working at all`
`08:42.57`	`troy-`	`JT not to mention having a card thats uncertified`
`08:43.03`	`JT`	`yeah`
`08:43.13`	`JT`	`i think angryuser has been smoking a bit too much of the crackpipe`
`08:43.23`	`troy-`	`especially that could kill a machine that costs a thousand +`
`08:44.40`	`snuff-work`	`called SMDs for a reason.. let the automation take care of it.. sucks soldering SMDs too fiddly`
`08:46.34`	`angryuser`	`JT: if you need a certifying sticker , i can sent it to you by mail`
`08:47.25`	`JT`	`angryuser: no, as in BUILT PROPERLY using a quality controlled process, and tested for correct operation`
`08:47.28`	`JT`	`not a stupid sticker`
`08:48.14`	`JT`	`i think a lot of these certifications are pretty useless, but what is useful is know your card has had a good chance of being built properly`
`08:48.17`	`troy-`	`night JT`
`08:48.27`	`JT`	`so stop being a damn miser and pay up :)`
`08:48.29`	`JT`	`night troy-`
`08:48.39`	`angryuser`	`JT: it is not maddnes, it opens just the possibility to anyone manufacture their own product under theit brand, witch will bring prices down, which is a good thing ;)`
`08:49.00`	`JT`	`angryuser: anyone is free to design and manufacture their own card`
`08:49.33`	`*** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk)`
`08:49.56`	`angryuser`	`JT: yes ...`
`08:50.19`	`OloBola`	`while trying to make/build the lumenvox example, I get a "no such file or directory" error for "LVSpeechPort.h" which I found in the include dir just below where I'm building from. I googled and found someone else with the same issue, he wrote something about incorrect environment vars, but that was it. Any suggestions?`
`08:50.29`	`JT`	`angryuser: there are some cheaper alternatives, perhaps you should try them`
`08:51.07`	`angryuser`	`JT: i have no problem with money, i have bought allready the needed cards`
`08:51.34`	`*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)`
`08:51.58`	`angryuser`	`does anyone know if misdn_check_l2l1(application) option is implemented from * ver 1.4.5?`
`08:52.04`	`Zeeek`	`guten Morgen`
`08:52.32`	`Zeeek`	`Got Flash? http://asterisktv.com/`
`08:53.54`	`Zeeek`	`what'dya think?`
`08:56.00`	`snuff-work`	`angryuser, i dont see that command in 1.4.6`
`08:56.35`	`Uatec`	`hey, has anybody noticed that linksys online chat support is not human, it's just a very very clever AI?`
`08:56.51`	`Zeeek`	`like this channel?`
`08:56.59`	`Uatec`	`lol, not so clever here`
`08:57.47`	`Zeeek`	`the AI is less clever or the live humans?`
`08:57.50`	`angryuser`	`snuff-work: and the misdn_l2l1_check ?`
`08:59.13`	`snuff-work`	`i see no misdn apps`
`08:59.16`	`*** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au)`
`08:59.56`	`*** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za)`
`09:00.01`	`Uatec`	`Zeeek, in here? take your pick.`
`09:00.09`	`Nobbie`	`heya =)`
`09:01.51`	`alrs`	`Uatec: are the linksys phones any good? I had to provision one today remotely, and I wasn't thrilled with the web interface.`
`09:02.03`	`Zeeek`	`I like it`
`09:02.33`	`Nobbie`	`alrs: which model ?`
`09:02.34`	`snuff-work`	`mm i like the ease of snom config via web :)`
`09:02.40`	`Zeeek`	`who has asterisk video for the Asterisk TV channel? http://asterisktv.com`
`09:02.44`	`Nobbie`	`we have 950 SPA942 which work well.`
`09:03.04`	`Zeeek`	`conference video, humor, stuff about what you do with asterisk (keep it clean)`
`09:03.22`	`Zeeek`	`and don't forget Mark live this Friday for the opening`
`09:04.55`	`Nobbie`	`i have a problem with * 1.4.x (even in latest .6) whereby SNOM360 is an agent in a Queue, and sometimes when a ringing call is answered, there's no voice. any ideas ? or how to troubleshoot that ?`
`09:05.16`	`alrs`	`Nobbie: SPA942`
`09:05.52`	`Nobbie`	`alrs: like a i siad, we have 950 of them, they work well. we provision them by TFTP which we maintain using a TFTP module we developed for FreePBX`
`09:06.27`	`Nobbie`	`unfortunately there no auto linking of TFTP file to Device/User _yet_ but it works well.`
`09:06.44`	`*** join/#asterisk casix (n=casix@edifici-pub.adam.es)`
`09:07.33`	`casix`	`hello`
`09:07.37`	`Nobbie`	`the SPA942 support CDP since version 5 (even later versions of 4), which means that it gets the Voice VLAN ID from the Switch`
`09:08.06`	`Nobbie`	`if the switch is Cisco, or supports CDP and voice vlan id sending upon request`
`09:09.35`	`alrs`	`Nobbie: Yeah, I was trying to get it to do TFTP, which isn't very clear in the web interface`
`09:09.42`	`alrs`	`Nobbie: but I got it working`
`09:10.04`	`Nobbie`	`i can give you a sample TFTP File if you want`
`09:10.18`	`JT`	`~phones`
`09:10.19`	`jbot`	`[phones] http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever.`
`09:10.29`	`alrs`	`Nobbie: I have to use the Endpoint Configuration in Trixbox`
`09:10.57`	`alrs`	`Nobbie: Which appears to work OK for that phone, once you figure out the cryptic name for the TFTP field in the web interface`
`09:11.23`	`Nobbie`	`mmm, haven't seen much of trixbox yet, will give it try sometime.`
`09:11.35`	`casix`	`I have a problem with asterisk in realtime, when it try to match an extension it matches first this _346XXX. than _346[1234]X.`
`09:11.56`	`JT`	`argh, too much freepbx/trix talk :P`
`09:12.02`	`casix`	`when i'm calling 3461111`
`09:12.33`	`Strom_M`	`casix: ....so instead just have two non-conflicting pattern matches?`
`09:12.35`	`alrs`	`JT: That doesn't count as a round of Trixbox faboydom`
`09:12.47`	`Nobbie`	`casix: try: _346[^1234]X. and _346[1234]X. ?`
`09:12.53`	`JT`	`heh`
`09:13.34`	`Nobbie`	`JT: and there was a chance for someone to take up a asterisk issue based on my question of the ghost callers ...`
`09:13.51`	`JT`	`Nobbie: sure....`
`09:14.10`	`casix`	`Strom_M: yes, but it repliques a lot of things and make it hard to administrate`
`09:14.22`	`casix`	`Nobbie: i'll try`
`09:15.02`	`casix`	`thx`
`09:15.05`	`Nobbie`	`casix: not sure if it would work, but you could also try shortning the first pattern to be of same length as the 2nd. it could be matching on the 1st becuase it's matching more characters, instead of specific ones in a shorter pattern`
`09:15.29`	`Nobbie`	`ie: _346XX. and _346[1234]X.`
`09:17.52`	`*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)`
`09:21.12`	`casix`	`no no it matches the _346XX. because the query that asterisk make is order by exten" and`
`09:21.53`	`casix`	`the X goes first than [, and i supose that asterisk search this order`
`09:22.08`	`snuff-work`	`heh.. one day someone will get around to redoing the whole asterisk extension ordering`
`09:22.34`	`*** join/#asterisk Dovid (n=Dovid@79.178.24.155)`
`09:26.06`	`*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)`
`09:27.59`	`*** join/#asterisk version5 (i=version5@nat/ibm/x-4e23c89682f43d9c)`
`09:29.06`	`version5`	`hey guys, i'm trying to design a dialplan that will answer, wait an unlimited amount of time for a user to enter an extension, pass the extension off to an AGI script and then go back to waiting for the next extension to be entered`
`09:29.19`	`*** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it)`
`09:29.26`	`version5`	`at the moment i've something like this`
`09:29.29`	`version5`	`exten => 999,n,WaitExten(0)`
`09:29.29`	`version5`	`exten => 1,1,AGI(getexten.agi)`
`09:29.32`	`mosty`	`is there a way to start/stop recording calls from AMI?`
`09:30.10`	`version5`	`i had hoped the WaitExten(0) part would allow an unlimited amount of time for a keypress but it doesnt`
`09:30.27`	`version5`	`also Set(TIMEOUT(absolute)=0) doesnt seem to work either`
`09:31.07`	`mosty`	`set it to 9999 or something`
`09:31.13`	`version5`	`the other issue i have is after the agi script has ran i wanted to jump back to the WaitExten again. Something like this`
`09:31.14`	`JT`	`where do they land to hit the ivr?`
`09:31.18`	`version5`	`exten => 9,1,AGI(getexten.agi)`
`09:31.19`	`version5`	`exten => n,1,Goto(main-menu)`
`09:31.29`	`version5`	`they land in this [main-menu] context`
`09:31.47`	`JT`	`with what extension?`
`09:32.16`	`version5`	`999`
`09:32.21`	`JT`	`err`
`09:32.29`	`JT`	`IVRs must start in s`
`09:33.19`	`version5`	`eh? what effect would that have? the agi script appears to get called at the moment`
`09:33.40`	`JT`	`search the wiki for ivr`
`09:33.42`	`JT`	`~thewiki`
`09:33.43`	`jbot`	`somebody said thewiki was at http://www.voip-info.org/wiki-Asterisk`
`09:33.50`	`version5`	`thanks`
`09:34.52`	`mosty`	`what protocol does ichat use? is it possible to talk with asterisk? (voice, maybe video)`
`09:35.46`	`version5`	`http://rafb.net/p/Q6hQwB30.html <--- thats what my dial plan looks like at the moment, in case you can spot any other stupid errors besides the 's' extension issue`
`09:39.17`	`Dovid`	`version5: This will throw u erorrs: exten => _.,`
`09:39.31`	`Dovid`	`you should use exten => _X., instead`
`09:40.14`	`version5`	`hrm, whats the difference?`
`09:40.19`	`mosty`	`is is possible to suppress those _. warnings?`
`09:40.53`	`Strom_M`	`mosty: no, they're there for a reason`
`09:41.03`	`Dovid`	`u can supress them but not smart too`
`09:41.07`	`Dovid`	`they are there to help u`
`09:41.27`	`Strom_M`	`u is not a word`
`09:41.58`	`Strom_M`	`is it REALLY that horrendously difficult to type the "y" and the "o" as well?`
`09:42.19`	`mosty`	`i've never had issues with patterns, i use _. where i want everything to match. i've never accidentally matched everything`
`09:42.36`	`Strom_M`	`mosty: you'll match the special extensions too`
`09:42.40`	`Strom_M`	`h, t, i, etc`
`09:43.10`	`Strom_M`	`bad idea, especially with h`
`09:43.31`	`version5`	`it appears after the agi script is called the call hangs up instead of jumping back to the start of main`
`09:43.43`	`version5`	`does the agi script have to do something to keep the call alive?`
`09:43.45`	`mosty`	`Strom_M, yes, that's the behaviour i want`
`09:44.04`	`Strom_M`	`weirdo :)`
`09:44.49`	`mosty`	`basically, when i work with dialplans written by others, i sometimes want to insert a context that does something and then passes control to the same extension in another context`
`09:50.13`	`ikaro`	`im reading asterisk book and i got one question, do I really need an analog (digium? Zaptel ? )card in the box that runs asterisk so to able to use asterisk at all ?`
`09:50.31`	`ikaro`	`or using a nic is enought ?`
`09:50.39`	`Gh0sty`	`ikaro: no`
`09:50.46`	`Gh0sty`	`there is zapteldummy driver`
`09:50.52`	`Gh0sty`	`so you dont need a card`
`09:51.26`	`ikaro`	`i c ... so even it is a voip setup i will need some kind of analog virtualization`
`09:51.58`	`ikaro`	`sorry if the question its too silly.`
`09:52.15`	`mosty`	`ikaro, you only need one of those cards if you want to call to or from the PSTN`
`09:53.09`	`kaldemar`	`you don't need the dummy timer driver either if you only do VoIP.`
`09:53.22`	`creativx`	`if you are iax trunking you do`
`09:53.23`	`Gh0sty`	`you needed it for something`
`09:53.31`	`Gh0sty`	`was it voicemail or what was it ... :/`
`09:53.43`	`Gh0sty`	`yes probably trunks`
`09:53.46`	`kaldemar`	`meetme conferences also need a timer.`
`09:54.05`	`Gh0sty`	`but anyways its not that you emulate analog its just for some timers i thought`
`09:54.09`	`Gh0sty`	`lol`
`09:54.13`	`Gh0sty`	`great minds etc ;p`
`09:54.27`	`Gh0sty`	`its been 4 years since i used asterisk`
`09:54.30`	`ikaro`	`so if someone calls from their home phone - plain analog line, and they call our voip number, will it work or I need the analog card in my box ?`
`09:54.34`	`Gh0sty`	`and see i still remember parts ;p`
`09:54.59`	`Gh0sty`	`i now want to try and setup one on debian but it doesnt seem to like debian much :(`
`09:55.04`	`mosty`	`ikaro, you only need an analogue card if you want to plug asterisk directly into an analogue line (or phone)`
`09:55.04`	`OloBola`	`I finally got lumenvox up and running, now asterisk won't start! wahoo.`
`09:55.22`	`ikaro`	`mosty, ok, thanks.`
`09:55.23`	`Gh0sty`	`ikaro: or your sip provider has analog connection`
`09:55.23`	`Zeeek`	`did you setup the pizza demo?`
`09:55.26`	`Gh0sty`	`or you need a card`
`09:55.42`	`OloBola`	`sure did.`
`09:55.49`	`*** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it)`
`09:55.58`	`Zeeek`	`and what did you order? The humungous?`
`09:56.32`	`Zeeek`	`what is wahoo?`
`09:57.02`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`09:57.42`	`*** join/#asterisk Zaggynl^Laptop (i=az@145.89.181.59)`
`09:57.52`	`Zeeek`	`ikaro if your voip number comes in via SIP or IAX, you don't need any hardwxre`
`09:58.11`	`Gh0sty`	`the sarcastic way of saying yay? ;p`
`09:59.33`	`ai-a`	`whats the difference between "exten = " and "exten => " ?`
`10:00.22`	`Zeeek`	`exten => will work ?`
`10:00.49`	`OloBola`	`Zeeek: wahoo is the sound of my enthusiasm for spending 12+ hours killing * box :)`
`10:00.50`	`Zeeek`	`it's.... lunchtime`
`10:01.06`	`Zaggynl^Laptop`	`How do you properly set up call transfering?`
`10:01.14`	`mosty`	`ikaro, if you want to do analogue trunking or meetme conferencing (and probably a few other things), it helps to have one of those cards though`
`10:01.26`	`Zeeek`	`oh, "asterisk won't start! wahoo" I missed the punctuation*`
`10:01.29`	`ai-a`	`well,, i have standard install of asterisk now. and it has the sample extention.conf and so on.. and has all these "exten = " at the bottom. I cant follow the ext...conf file. to find the flow.. ... can i delete all the .conf files , and will the gui build them again ?`
`10:01.34`	`Zaggynl^Laptop`	`I've changed features.conf, extensions.conf and made a new extensions with t) in it`
`10:02.09`	`Zeeek`	`OloBola what's happening? It dies right away?`
`10:02.12`	`mosty`	`Zaggynl^Laptop, what about the T option? you know the difference between t and T right?`
`10:02.27`	`Zaggynl^Laptop`	`T also allows the caller to transfer`
`10:02.44`	`*** join/#asterisk yugowas (n=y@dh107-242.xnet.hr)`
`10:02.53`	`Zeeek`	`show application Dial`
`10:03.04`	`Zeeek`	`god I miss BKW`
`10:03.21`	`mosty`	`Zaggynl^Laptop, just checking that you had the correct one in your dial command options`
`10:03.31`	`Zaggynl^Laptop`	`I'll add both`
`10:03.40`	`yugowas`	`hi all, does anyone know why the monitor application does not send an event such as monitor started/stopped event ?`
`10:03.43`	`Zaggynl^Laptop`	`"break it first, then fix it" :)`
`10:04.15`	`mosty`	`Zaggynl^Laptop, then "show features" to make sure you're using the correct keypresses`
`10:04.30`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`10:04.30`	`Zaggynl^Laptop`	`show features?`
`10:04.40`	`Zeeek`	`show: http://asterisktv.com`
`10:04.44`	`mosty`	`show features.`
`10:05.15`	`Zaggynl^Laptop`	`I appended 'include => featuremap' to extensions.conf`
`10:05.49`	`mosty`	`i've never seen that before, features.conf works without it`
`10:05.58`	`Zaggynl^Laptop`	`hmm okay`
`10:07.09`	`mosty`	`do "show features", you'll see if it's enabled or not`
`10:08.46`	`ai-a`	`Why do thoses guys all have bushy eye brows ?`
`10:08.49`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`10:10.07`	`Zeeek`	`who?`
`10:10.40`	`Zeeek`	`quick see the asterisk HOW TO (not)`
`10:10.47`	`Zeeek`	`<PROTECTED>`
`10:13.16`	`version5`	`is there some way to avoid a hangup after a call to an agi script?`
`10:13.32`	`ai-a`	`why should agi scripts hangup in the first place version5 ?`
`10:13.44`	`version5`	`i dont know`
`10:13.46`	`version5`	`im getting this`
`10:13.48`	`version5`	`http://rafb.net/p/8NfC8H41.html`
`10:13.52`	`version5`	`after i call the script`
`10:14.16`	`ai-a`	`well, need to see your conf files.. whats happening after the agi call ?`
`10:14.22`	`version5`	`one sec`
`10:14.38`	`ai-a`	`and inside the agi script itself... your wrote it... where's your flow ?`
`10:15.19`	`version5`	`http://rafb.net/p/N14EyE67.html`
`10:15.27`	`version5`	`thats the ivr menu`
`10:19.03`	`Siya`	`Small question,`
`10:19.18`	`Siya`	`How can I prove that calls come into the wrong trunk from my SP?`
`10:19.48`	`mosty`	`unplug every other trunk`
`10:19.51`	`kaldemar`	`version5: you don't have priority 2 for your extensions.`
`10:20.02`	`Siya`	`I have two trunks with same provider but all calls come into one trunk`
`10:20.12`	`Siya`	`SIP trunks not analog`
`10:20.29`	`version5`	`http://rafb.net/p/qQBsTU16.html <-- thats the agi script, which doesnt actually output anything for some reason`
`10:20.36`	`OloBola`	`ZeeeK: http://www.pastebin.ca/602831`
`10:21.03`	`mosty`	`Siya, disable the other trunk`
`10:21.17`	`ai-a`	`version5: where do you expect your flow to go after ivr,1,1 ?`
`10:21.17`	`Siya`	`can I unregister it from cli?`
`10:21.28`	`mosty`	`actually you mean "account", SIP does not support trunking to my knowledge`
`10:21.38`	`Siya`	`mosty: you're right`
`10:21.44`	`mosty`	`disable the account in sip.conf and do a sip reload`
`10:21.47`	`Zeeek`	`OloBola looks like a version mismatch of the lum module`
`10:22.08`	`version5`	`i expected it would fall through to exten => _X,1,GoTo(s,playback) ; jump back to the start`
`10:22.12`	`Siya`	`Another thing, anyone seen this before: "chan_sip.c:12035 handle_response_register: Got 200 OK on REGISTER that isn't a register"`
`10:22.23`	`version5`	`its getting caught by the hangup extension though`
`10:22.25`	`ai-a`	`version5: 1 doesnt follow 2 .`
`10:22.27`	`ai-a`	`eh`
`10:22.28`	`Zeeek`	`OloBola what versions of asterisk and your OS?`
`10:22.28`	`ai-a`	`version5: 1 doesnt follow 1 .`
`10:22.34`	`kaldemar`	`errr:`
`10:22.48`	`Zeeek`	`SVN... yeah that might do it`
`10:23.17`	`version5`	`ah, sorry i misunderstood. I thought once it was a different extension pattern the priority would be reset to 1`
`10:23.18`	`kaldemar`	`version5: _X,2 will fix your behaviour.`
`10:23.23`	`Zeeek`	`I don't think you can expect to run stuff like Lumenvox in SVN`
`10:23.40`	`version5`	`wont the hangup extension still catch it though`
`10:23.48`	`*** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu)`
`10:23.59`	`OloBola`	`Zeek: trunk and FC5`
`10:24.15`	`version5`	`no, apparently that does work.`
`10:24.19`	`version5`	`cheers guys`
`10:24.48`	`version5`	`i assume what was happening was there was no matching extension/rule so a hangup was generated instead?`
`10:24.53`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`10:25.46`	`kaldemar`	`auto fallthrough was happening like the output said.`
`10:29.01`	`DarKnesS_WolF`	`i can't use dtmfmode=inband with g729 !?`
`10:30.06`	`Siya`	`how do I fix this: chan_sip.c:16637 reload_config: Can't add wildcard IP address to domain list, please add IP address to domain manually.`
`10:30.09`	`mosty`	`DarKnesS_WolF, you can if you have g729 licences`
`10:30.14`	`mosty`	`i think`
`10:31.23`	`DarKnesS_WolF`	`mosty: i'm testing using the opensource one .. but what the licences has to do with dtmfmode ? i can already use g729 and it works but i got error i can't use inband with g729`
`10:31.38`	`kaldemar`	`DarKnesS_WolF: you need a 64k codec for inband dtmf.`
`10:32.26`	`*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)`
`10:32.39`	`Siya`	`Grrrr call comes into the right account when I bring down the other account`
`10:32.53`	`Siya`	`bringing it back up restores the situation again :(`
`10:32.54`	`DarKnesS_WolF`	`kaldemar: it works with ulaw but this strange phone don't work with rfc or the info the phone don't work .`
`10:33.04`	`Siya`	`How can I figure out if it's my failt or my SP?`
`10:35.07`	`mosty`	`Siya, setup one of the accounts on a different machine`
`10:35.29`	`Siya`	`mosty: I have only one * server...`
`10:36.40`	`kaldemar`	`Siya: posting some configs and cli output might get you a bit more help.`
`10:37.19`	`Siya`	`kaldemar: well it's *now so configs are slightly differently layed out :)`
`10:37.32`	`ai-a`	`what is users.conf ? is that just for the gui ? im following extensions.conf to follow the flow of *gui's implementation,, where does it register sip accounts, and skip voicemail depending on whats in users.conf ? im wanting to add some small flow for recording all calls for specifc extension ranges.`
`10:37.51`	`Siya`	`and as dialing (ingress/egress) and caller id's etc all work fine...`
`10:38.18`	`Siya`	`ai-a: users.conf is used for extensions and sip accounts`
`10:38.32`	`Siya`	`I donn't use IAX so no idea bout that`
`10:38.44`	`ai-a`	`why have they changed it ?`
`10:38.56`	`ai-a`	`would have been best to keep sip inside sip.conf`
`10:39.01`	`Siya`	`no idea you can ask in #asterisk-gui though :)`
`10:40.00`	`Siya`	`kaldemar: screen output shows no errors, it shows:`
`10:40.00`	`Siya`	`<PROTECTED>`
`10:40.05`	`Siya`	`instead of:`
`10:40.18`	`Siya`	`<PROTECTED>`
`10:40.58`	`Siya`	`so I have no errors on my side to go by. To me this seems that my SP just picks the wrong session to initiate the call to my server`
`10:41.24`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
`10:41.29`	`Siya`	`I'll try to bring down the other account after lunch but I think it will not show anything else`
`10:41.37`	`version5`	`right.... and one last question, why would my agi script not be printing anything to the screen. Using something like sys.stderr.write() should print to the console, right?`
`10:42.38`	`version5`	`assuming i've connected with -c or -r that is`
`10:42.45`	`mosty`	`version5, i don't think so. use the noop agi command`
`10:44.26`	`*** join/#asterisk currach (n=currach@213-202-141-138.bas502.dsl.esat.net)`
`10:47.53`	`OloBola`	`will reinstalling asterisk overwrite my conf files etc?`
`10:48.24`	`mosty`	`depends entirely on how you reinstall`
`10:49.49`	`OloBola`	`I spent all day trying to lumenvox up and running only to find out that "the latest version of asterisk" actually means 1.4.3.`
`10:50.04`	`OloBola`	`so I guess I need to install 1.4.3`
`10:50.40`	`*** join/#asterisk nfi\|ermes (n=ErMeS@217.220.121.62)`
`10:52.50`	`Zeeek`	`I have it running on the lastest version`
`10:52.57`	`Zeeek`	`.6, right?`
`10:52.58`	`ai-a`	`have -gui installed, how can i add me own custom flow to all calls within , and custom flow on specific extension ranges ? Im wanting to add another context for these ext. to follow (ie. prepend some initial agi / call recording flow)`
`11:01.04`	`*** join/#asterisk lwh (n=lwh192@66.212.165.24.tor.pathcom.com)`
`11:05.55`	`Swat2`	`"zap show" isnt showing anyhting.. what am i missing?`
`11:06.56`	`Swat2`	`No such command 'zap show'`
`11:08.13`	`*** join/#asterisk saftsack (n=saftsack@p57A77B5C.dip.t-dialin.net)`
`11:09.21`	`*** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il)`
`11:10.04`	`Swat2`	`never mind, im a retard, it didnt compile zaptel in astersk`
`11:17.36`	`*** part/#asterisk _E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com)`
`11:27.23`	`Dovid`	`Swat2: we all have dog days :)`
`11:27.46`	`Gh0sty`	`hm thats an expression im not familiar with :p`
`11:27.54`	`Gh0sty`	`funny`
`11:28.02`	`Ch0Hag`	`What can I liken Skype to to persuade somebody to hold back?`
`11:28.26`	`mosty`	`doody`
`11:29.38`	`nicox`	`hi JT?`
`11:32.43`	`JT`	`hi`
`11:38.23`	`nicox`	`now its Best: 100.000000 -- Worst: 99.938965 -- Average: 99.991119`
`11:38.38`	`nicox`	`but the last hour, the error does not happen`
`11:39.49`	`*** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it)`
`11:39.57`	`*** part/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it)`
`11:40.42`	`*** join/#asterisk marc\cba (n=marc@cpc3-whit2-0-0-cust629.cdif.cable.ntl.com)`
`11:44.02`	`torch`	`morning guys ..`
`11:45.50`	`torch`	`quick question ... I'm running asterisk 1.2.18 and everytime I restart the asterisk (or zapata module) ...calls from my pstn to my asterisk extensions are just mute ...`
`11:46.17`	`torch`	`does anybody know why?`
`11:46.23`	`torch`	`am I doing something wrong?`
`11:47.43`	`*** join/#asterisk guillote_GNU (n=guillote@host252.200-82-63.telecom.net.ar)`
`11:57.28`	`*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)`
`11:58.11`	`*** join/#asterisk friedrich\| (n=friedric@e177242110.adsl.alicedsl.de)`
`12:04.50`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`12:07.10`	`nicox`	`is this worst thing okay? 99.951172`
`12:10.42`	`*** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de)`
`12:11.27`	`Fl1p`	`hi anyone knows how to get detailled info about all active channels ? "show channels" only shows an incomplete ChannelID`
`12:11.37`	`Fl1p`	`but i need the whole one`
`12:12.41`	`mosty`	`what other info do you need?`
`12:13.34`	`krdian_`	`show channels concise or show channels verbose (asterisk 1.4)`
`12:14.51`	`Fl1p`	`need only the full channelid but concise and verbose works! Thx a lot!`
`12:15.20`	`Fl1p`	`other question : i'm currently playing with the features.conf and want this special scenario:`
`12:15.50`	`*** join/#asterisk Krooks (n=blah@124.82.105.208)`
`12:17.49`	`Fl1p`	`two peoples talking to each other the caller puts him on hold calls someone else (like attended but:) talks to him and only push the '9' button and will be retransferred to the first, the other will hangup`
`12:18.22`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:19.04`	`Krooks`	`anyone got some time to chat. I'm demoing to some people here. host=60.52.11.214, user=user5 password=1234 extension=5555. Protocol must be iax2 , NOT sip. Can use idefisk.`
`12:19.54`	`Fl1p`	`i've tried a lot using feature Application map Goto which doesn't work for me, also the atxfer at *2 doesnt show any reaction`
`12:20.06`	`Krooks`	`Can call me at 1901.`
`12:22.10`	`RypPn`	`Krooks: gimme a mo`
`12:22.24`	`Zeeek`	`No answer`
`12:22.34`	`RypPn`	`Krooks: dial me again`
`12:22.52`	`*** join/#asterisk eAi2k (n=cow@81-86-205-45.dsl.pipex.com)`
`12:24.29`	`eAi2k`	`hi - anyone used the linksys SPA-3102 (or similar) and know why the web interface doesn't work?`
`12:24.33`	`Zeeek`	`get anything Krooks?`
`12:24.43`	`Krooks`	`nope`
`12:24.45`	`Zeeek`	`I dialed you it answered and I heard no audio`
`12:25.12`	`Krooks`	`I heard someone speaking but u or that someone could hear me.`
`12:25.13`	`Zeeek`	`whoo answered at that ip then?`
`12:25.18`	`Krooks`	`me`
`12:25.31`	`Zeeek`	`I heard nothing`
`12:25.33`	`Zeeek`	`great demo`
`12:25.44`	`Fl1p`	`lol`
`12:25.53`	`Krooks`	`But I saw RynPn's name appear`
`12:26.03`	`Krooks`	`was that you ?`
`12:26.07`	`Zeeek`	`no`
`12:26.08`	`mosty`	`eAi2k, you have to turn the web interface on`
`12:26.11`	`RypPn`	`yeah, I've logged out, assuming a clash with zeedo`
`12:26.15`	`RypPn`	`oops Zeeek`
`12:26.36`	`Krooks`	`ok I'm going to call 5555 again`
`12:27.27`	`Krooks`	`oh great, now both of you unregistered.`
`12:27.47`	`RypPn`	`make up another extension for me to try Krooks`
`12:28.07`	`Krooks`	`yeah good idea. we can make a 3 way call`
`12:28.26`	`Krooks`	`RypPn : user4 1234 4444`
`12:28.27`	`RypPn`	`3way 1way, cant wait`
`12:28.40`	`*** join/#asterisk javar (n=javar@69.79.134.24)`
`12:29.41`	`RypPn`	`rings the nu tone`
`12:29.44`	`RypPn`	`then*`
`12:32.51`	`Krooks`	`user4 please call again`
`12:33.27`	`Krooks`	`user4 we got a codec error`
`12:33.33`	`RypPn`	`still rings a few times then busy tone`
`12:33.36`	`Krooks`	`thats why you keep disconnecting`
`12:33.49`	`Krooks`	`use gsm`
`12:33.54`	`RypPn`	`what codec do you want the phone on?`
`12:33.56`	`RypPn`	`k`
`12:34.25`	`Krooks`	`oh man. We could have done a 3 way call bu Zeeek got to go`
`12:34.37`	`Krooks`	`still`
`12:34.45`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`12:34.49`	`Krooks`	`codec error. it says unknown codec`
`12:34.55`	`*** join/#asterisk msetim (n=msetim@200.195.161.164)`
`12:35.21`	`Krooks`	`I don't know.`
`12:35.27`	`Krooks`	`I call u`
`12:35.43`	`*** join/#asterisk juuva (i=juuva@peili.org)`
`12:35.55`	`Krooks`	`You using idefisk RynPn`
`12:36.28`	`*** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br)`
`12:36.30`	`shazaum`	`hi all`
`12:36.36`	`shazaum`	`somebody?`
`12:36.51`	`RypPn`	`Krooks: looking for gsm on this phone`
`12:36.58`	`*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)`
`12:37.11`	`Krooks`	`shazaum: hi`
`12:37.54`	`RypPn`	`Krooks: I dont have gsm on my desk phone`
`12:38.07`	`Krooks`	`is it a IP phone ?`
`12:38.15`	`Krooks`	`or a softphone`
`12:38.16`	`*** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)`
`12:38.22`	`*** join/#asterisk guillote_GNU (n=guillote@190.7.27.17)`
`12:38.30`	`Krooks`	`use idefisk`
`12:39.18`	`Krooks`	`RynPn: you around ?`
`12:39.20`	`[TK]D-Fender`	`unless you're going to register your PHONE direct with Krooks, it doesn't matter WHAT phone you use.`
`12:39.28`	`Krooks`	`iax`
`12:39.32`	`RypPn`	`direct`
`12:39.36`	`Krooks`	`It must be iax`
`12:40.46`	`RypPn`	`I have no windows to install idefisk on sry Krooks`
`12:41.00`	`Krooks`	`It runs on Linux`
`12:41.09`	`Krooks`	`There is linux version`
`12:41.10`	`RypPn`	`cba`
`12:41.29`	`Krooks`	`someone please cba?`
`12:41.38`	`Krooks`	`stand for ?`
`12:42.00`	`*** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it)`
`12:42.19`	`[TK]D-Fender`	`RypPn: You. Don't. Need. IDEFISK.`
`12:42.41`	`Krooks`	`you need iax`
`12:42.59`	`Krooks`	`if its iax, it'll work`
`12:43.10`	`[TK]D-Fender`	`Krooks: Correct. And noone here should need an IAX soft-phone OR hardphone.`
`12:43.29`	`[TK]D-Fender`	`Krooks: Also you should have set up SIP. IAX is NOT a tech to sell * on.`
`12:43.36`	`Krooks`	`ooo k`
`12:43.50`	`Krooks`	`I had problem with SIP behind firewall`
`12:43.53`	`[TK]D-Fender`	`Krooks: not that this should be a problem for your test,.`
`12:44.26`	`[TK]D-Fender`	`Krooks: Unless its a Cisco PIX or one of a bunch of shitty D-Link models, SIP works just fine`
`12:44.50`	`Krooks`	`RynPn: are you still trying ? or give up already ?`
`12:45.12`	`JT`	`can't be arsed is my guess`
`12:45.25`	`RypPn`	`Krooks: I. got. fed. up.`
`12:45.31`	`JT`	`ask for too difficult and demanding free help, funnily enough it dries up ;)`
`12:45.56`	`Krooks`	`ok`
`12:46.05`	`Krooks`	`I guess JT is right.`
`12:46.20`	`JT`	`gsm only eh? not very compatible`
`12:46.22`	`Krooks`	`sorry`
`12:46.30`	`Krooks`	`and speex`
`12:46.42`	`Krooks`	`and iLBC`
`12:46.54`	`RypPn`	`all the oddballs`
`12:46.58`	`JT`	`wow :P pretty much all the codecs that people are LEAST likely to have`
`12:46.59`	`Krooks`	`hehe`
`12:47.13`	`[TK]D-Fender`	`JT : this should NOT pose a problem for ANYONE here and should already have been COMPLETED successfully.`
`12:47.17`	`Krooks`	`Thats what comes with idefisk`
`12:47.22`	`[TK]D-Fender`	`JT : its really kind of SAD`
`12:47.33`	`[TK]D-Fender`	`Krooks: Get OFF this idefisk kick of yours!`
`12:47.39`	`[TK]D-Fender`	`Krooks: Its not needed!`
`12:48.12`	`Krooks`	`alright`
`12:49.34`	`JT`	`[TK]D-Fender: this exercise you mean?`
`12:50.19`	`[TK]D-Fender`	`JT : Exactly`
`12:50.22`	`JT`	`yeah`
`12:50.28`	`Krooks`	`This channel is quiet anyway. I did not point a gun to anyone's head to help me.`
`12:51.23`	`RypPn`	`Krooks: I'm more than happy to help you test/demo, just you'll need g711, g723 or g729 for my phone to connect`
`12:52.46`	`Krooks`	`RynPn: thanks. appreciate it. but on this windows version of idefisk, those codecs are not installed`
`12:52.48`	`JT`	`g.723, now that's HARD :P`
`12:52.52`	`JT`	`Krooks: ...`
`12:52.57`	`JT`	`g.711 is ALWAYS THERE`
`12:53.03`	`JT`	`it's the main codec for the pstn`
`12:53.45`	`ReDNeQ-`	`jeez this guy is pissin me off and I havent even helped`
`12:53.52`	`RypPn`	`lol`
`12:54.01`	`Krooks`	`:) lol`
`12:55.14`	`Krooks`	`whats a-law and u-law. I just added those codecs`
`12:55.20`	`JT`	`...`
`12:55.21`	`JT`	`g.711`
`12:55.39`	`JT`	`why not just enable everything`
`12:56.22`	`Krooks`	`I just did. I enabled every codecs. By default not all codecs was enabled`
`12:56.25`	`RypPn`	`heh, this is just internal extensions, wait till we get to routing...`
`12:56.40`	`Krooks`	`RynPn : Are you still regostered`
`12:56.44`	`RypPn`	`nope`
`12:56.48`	`Krooks`	`I want to try one last time`
`12:57.42`	`*** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net)`
`12:58.15`	`RypPn`	`go for it`
`12:58.36`	`RypPn`	`still dying`
`12:58.41`	`Krooks`	`:( not work`
`12:58.45`	`Nobbie`	`ouch, * 1.4.6 is doing strange stuff on our system =( gets into a bad state where no calls are made. the logs show no errors, and a Dial() command gets logged but doesn't go any further`
`12:58.49`	`Zeeek`	`just add this line to finish the install: PlayBack(demo-congrats)`
`12:58.52`	`Krooks`	`Thanks man. Its ok.`
`13:00.09`	`Nobbie`	`and no core gets dumped`
`13:00.13`	`[TK]D-Fender`	`RypPn: If you are configuring your PHONE then you are in the wrong channel.`
`13:00.46`	`RypPn`	`[TK]D-Fender: why are you having a go at me?`
`13:01.03`	`Zeeek`	`[TK]D-Fender d'oh`
`13:01.21`	`[TK]D-Fender`	`RypPn: the two of you are going about this the hard way and I'm kind of hoping you'd see what tool should be playing which job.`
`13:02.08`	`Zeeek`	`no one should be playing at all`
`13:02.14`	`Krooks`	`Zeeek : I found a Mic Boost checkbox on idefisk and enabled it. I hope it sound load now.`
`13:02.20`	`Zeeek`	`except if you want to watch http://asterisktv.com`
`13:02.21`	`Krooks`	`loud`
`13:02.31`	`JT`	`[TK]D-Fender: my i love softphones ;)`
`13:02.53`	`Zeeek`	`like everything, they have their place`
`13:03.17`	`JT`	`testing and overseas airports only ;)`
`13:03.33`	`Zeeek`	`good examples`
`13:03.58`	`JT`	`Zeeek: i voiped it up from kansai international airport in osaka, japan, that was pretty cool`
`13:04.15`	`Zeeek`	`I'm gonna veture to guess that 90% of the people with laptops and headsets at the lats Astricon I attended were using one of two IAX softphones`
`13:04.20`	`JT`	`since their payphones eat about $2/min on international calls`
`13:04.36`	`Zeeek`	`I always havez both SIP and IAX softphones on my laptops`
`13:04.43`	`Zeeek`	`for that reason among others`
`13:04.44`	`JT`	`i have both as well`
`13:04.58`	`Krooks`	`Zeeek: you use what iax softphones ?`
`13:05.01`	`Zeeek`	`although idefisk does SIP I don't think I've ever tried it`
`13:05.14`	`JT`	`must be a new idefisk version`
`13:05.18`	`[TK]D-Fender`	`JT : Softphones aren't "bad", they have their place and for mobile suers & testing, sure, why not. For far away family memebers to call you? SURE! But for anyone to have to mention ANY specific phone for the need he requested is ludicrous :)`
`13:05.24`	`Zeeek`	`idefisk and every other one I find, but idefisk is the best IMO so far... for me`
`13:05.27`	`JT`	`still no decent open source iax softphones`
`13:05.37`	`JT`	`idefisk is closed source`
`13:05.47`	`Zeeek`	`I don't mind if something works`
`13:06.05`	`Zeeek`	`I even pay for software ocassionally`
`13:06.16`	`JT`	`it's not worth paying for`
`13:06.17`	`Zeeek`	`up to about $29,99`
`13:06.29`	`JT`	`i wouldn't even pay for eyebeam`
`13:06.38`	`Zeeek`	`I might if I needed it`
`13:06.45`	`JT`	`Zeeek: you in europe?`
`13:06.49`	`Zeeek`	`yes`
`13:07.31`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`13:07.32`	`JT`	`commas instead of full stops/dots, not cool :P`
`13:07.45`	`Zeeek`	`where?`
`13:07.55`	`Zeeek`	`oh that was an accident.`
`13:07.55`	`JT`	`$29,99`
`13:08.07`	`Zeeek`	`I hate those little differences`
`13:08.15`	`JT`	`hehe`
`13:08.27`	`Zeeek`	`PAL, SECAM, NTSC`
`13:08.38`	`Zeeek`	`2007-07-04`
`13:08.51`	`JT`	`imperial, metric`
`13:08.52`	`Zeeek`	`4/7/07`
`13:08.59`	`JT`	`only crackheads use imperial either`
`13:09.06`	`JT`	`american date format is the worst`
`13:09.09`	`JT`	`it's stupid`
`13:09.17`	`Zeeek`	`Slowly I turned, step by step... 2.54 centimeter by 2.54 centimeter`
`13:09.21`	`*** join/#asterisk guillote_GNU (n=guillote@190.7.27.17)`
`13:09.28`	`JT`	`month before day, follower by year, dum dum dum dum dum`
`13:09.34`	`Zeeek`	`I like Japanese = mysql`
`13:09.45`	`Zeeek`	`MSB LSB rulez`
`13:10.43`	`Zeeek`	`so Krooks should I call you one last time?`
`13:10.51`	`Zeeek`	`going once, going twice...`
`13:11.03`	`Krooks`	`yes please`
`13:16.08`	`*** join/#asterisk CVirus (n=GoD@196.218.187.30)`
`13:17.22`	`Zeeek`	`shit, raining, no fun to walk home :(`
`13:17.51`	`*** join/#asterisk romano2k (i=freenode@shtak.fr)`
`13:19.04`	`Zeeek`	`I was waiting for it to get better, but it got worse. SOunds like software releases!`
`13:21.42`	`*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`13:21.57`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`13:24.14`	`[TK]D-Fender`	`Zeeek: So... call successful?`
`13:25.39`	`Zeeek`	`ya it worked`
`13:30.29`	`*** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net)`
`13:30.42`	`*** part/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net)`
`13:32.57`	`*** part/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de)`
`13:38.05`	`*** join/#asterisk Inkubot (n=inkubot@200.75.4.10)`
`13:38.05`	`Siya`	`can I change source ports for accounts?`
`13:38.29`	`Inkubot`	`hi *`
`13:38.40`	`[TK]D-Fender`	`Siya: Try being specific.`
`13:39.43`	`romano2k`	`hi! is there an irc channel for french asterisk users?`
`13:39.52`	`*** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk)`
`13:40.06`	`Siya`	`[TK]D-Fender: erm I have two accounts to the same SP and I'm affraid that I'm having problems as * will use the same port number for all ourgoing sessions`
`13:40.30`	`Siya`	`source:5060 - destination:5060 is fine when the destinations are unique`
`13:40.43`	`Siya`	`this case two accounts have the same destination`
`13:40.55`	`[TK]D-Fender`	`Siya: Thats perfectly fine.`
`13:41.04`	`*** join/#asterisk [GuS] (n=gdnet@unaffiliated/gus/x-663402)`
`13:41.09`	`[TK]D-Fender`	`Siya: thats what SESSION ID's are for`
`13:41.40`	`[GuS]`	`Hello guys!! i have a new question :P... the latest version of asterisk does not include the init.d script?`
`13:41.50`	`Siya`	`[TK]D-Fender: ?! how can * determine which session is being addressed when receiving packets on port 5060 from this source...?`
`13:41.53`	`[GuS]`	`cause i've reinstalled asterisk, and i have not the script inside there`
`13:42.03`	`*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)`
`13:42.06`	`tzafrir`	`[GuS], on which distro?`
`13:42.15`	`[TK]D-Fender`	`Siya: Because each one comes in with seperate AUTH credentials perhaps?`
`13:42.43`	`[GuS]`	`tzafrir: kubuntu feisty`
`13:42.46`	`Siya`	`[TK]D-Fender: so each sip packet includes credentials?`
`13:42.57`	`[GuS]`	`i've installed the version before that, and the script was there`
`13:43.04`	`[TK]D-Fender`	`Siya: What do you think the username & secret are?!`
`13:43.07`	`[GuS]`	`now is not`
`13:44.10`	`Siya`	`[TK]D-Fender: for session auth, starting another session on the same port is usually ok as the source ports should be randomised, but if they're not then the receiving end will assume the packets are meant for the established session (which is already authenticated)`
`13:44.17`	`[TK]D-Fender`	`[GuS]: And what did you do in this reinstall to set up those scripts? Also didn't Ubuntu get RID of the normal init process?`
`13:44.21`	`*** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca)`
`13:44.33`	`[TK]D-Fender`	`Siya: .....No.`
`13:44.35`	`[GuS]`	`nothing weird, i've installed asterisk as always [TK]D-Fender`
`13:44.55`	`Siya`	`[TK]D-Fender: man in the middle attacks work this way, hence encryption and repeated handshakes are used in cases where this is a problem`
`13:44.55`	`[TK]D-Fender`	`Siya: You completely misunderstand call flow.`
`13:45.02`	`[GuS]`	`and yes, i've reinstalled just in case to test`
`13:45.22`	`[GuS]`	`and nothing... i just started asterisk from command line... but that script inside init.d is not there :s`
`13:45.22`	`Siya`	`[TK]D-Fender: I do understand networking and data transport very well though`
`13:45.26`	`[TK]D-Fender`	`Siya: SIP is always authed and is only there to setup RTP. RTP will come in on random ports.`
`13:45.54`	`Siya`	`[TK]D-Fender: yeah it's the * part I don;t fully grasp`
`13:46.07`	`Siya`	`* will register with my SP fine (both sessions)`
`13:46.23`	`[TK]D-Fender`	`Siya: Each call is independant on its own port. Call seqno's are tracked, auth is tracked. No, you do NOT need a specific port per account, that is LUDICROUS.`
`13:46.40`	`tzafrir`	`[GuS], you need to copy the init.d script manually`
`13:46.48`	`tzafrir`	`or better get the one from the package`
`13:46.49`	`[GuS]`	`ok,`
`13:47.09`	`[GuS]`	`i know how to setup the script... but was weird that was not copied inside there`
`13:47.11`	`*** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net)`
`13:47.14`	`*** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com)`
`13:47.29`	`[GuS]`	`in which dir of the source dir is tzafrir exactly?`
`13:47.38`	`Voicemeup`	`http://www.grandcentral.com/howitworks/callswitch`
`13:47.49`	`Voicemeup`	`is this simply a conference style sewitching ?`
`13:47.53`	`Siya`	`I'm trying to figure out why only one account picks up both accounts calls`
`13:48.04`	`Voicemeup`	`likehe service calls on another leg.. then bridges 2 new ones and drops first ?`
`13:48.30`	`*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)`
`13:48.48`	`[TK]D-Fender`	`Voicemeup: Looks like`
`13:49.01`	`Voicemeup`	`hmm`
`13:49.08`	`[TK]D-Fender`	`Voicemeup: Maybe more like parking.`
`13:49.12`	`Voicemeup`	`To switch from one phone to another without hanging up, just press the () button while you're talking. Your other phones will ring and you can pick up the one you want and hang up the other.`
`13:49.22`	`Voicemeup`	`yeah press the * and it dials other phones. hmm`
`13:49.30`	`Voicemeup`	`wonder if this is * based`
`13:49.34`	`[TK]D-Fender`	`Voicemeup: Or 1-touch trasnfer to all.`
`13:49.40`	`Voicemeup`	`yeah`
`13:49.46`	`Voicemeup`	`Best of all, your caller won't even hear the switch.`
`13:49.48`	`[TK]D-Fender`	`Dead easy actually.`
`13:49.53`	`Voicemeup`	`that the thing im trying to figure`
`13:49.57`	`Voicemeup`	`yeah ?`
`13:50.18`	`[TK]D-Fender`	`Voicemeup: AMI redirect + callfile`
`13:50.32`	`Voicemeup`	`the ivr parts is good .. ivr per calleird .. nxxxxxxxxx/callerid , 1, ivr-1.php`
`13:50.39`	`Voicemeup`	`hmm`
`13:50.42`	`[TK]D-Fender`	`Voicemeup: In fact... you don't even have to hang up ;)`
`13:50.43`	`Voicemeup`	`ill try`
`13:50.48`	`Voicemeup`	`but you will hear the ring ?`
`13:51.16`	`Siya`	`[TK]D-Fender: Any hint on how I can debug how my SP is adressing my * server? I need proof that either my server is broken or theirs is`
`13:51.59`	`[TK]D-Fender`	`Voicemeup: Nope.... by pressing "*" you'd do a dynamic feature that will pull the CHANNEL of your call and launch the dial-out callfile based on it. Upon answer it'd do an AMI redirect to streal the call transparently ;)`
`13:52.13`	`[TK]D-Fender`	`Siya: Pastebin your configs.`
`13:52.14`	`[TK]D-Fender`	`~pb`
`13:52.15`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org`
`13:52.20`	`Inkubot`	`i want to make a cluster of asterisks... for load balancing, someone try that ? what distro i should use or could be any distro (i want slackware) ? ultramonkey is my friend ??`
`13:52.57`	`[TK]D-Fender`	`Inkubot: No, that is a WEB HA solution`
`13:53.15`	`[TK]D-Fender`	`Inkubot: * would require all sorts of stuff. SIP proxy, backend DB servers, the works.`
`13:53.26`	`Voicemeup`	`and.. a switch that works`
`13:53.31`	`[TK]D-Fender`	`Inkubot: Definately rsync for the flat stuff`
`13:53.42`	`Voicemeup`	`tried hte big ip , and server iron and no luck on udp session sticking`
`13:53.45`	`[TK]D-Fender`	`Inkubot: If you have to ask..... you're not qualified :)`
`13:53.52`	`Voicemeup`	`i think the biggest server iron can do it.. bu`
`13:53.53`	`Voicemeup`	`but`
`13:54.24`	`Voicemeup`	`else i always had the UDP part not stickign to the session , but hey , im always in trouble`
`13:54.45`	`Voicemeup`	`[TK]D-Fender , you know if asucsh a thing exists ? a pc scope ?without hardware ?`
`13:55.01`	`[TK]D-Fender`	`Voicemeup: Not a clue, its out of my scope too ;)`
`13:55.16`	`Voicemeup`	`getting AC noise from 9p to 6am and trying to see why. if i unplug ups its ok.. but i still hear hummmm on radio.. so its global`
`13:55.19`	`[TK]D-Fender`	`Voicemeup: And sure as hell not likely for . is not made for this....`
`13:55.21`	`Siya`	`[TK]D-Fender: http://pastebin.ca/603040 (users.conf only, no idea where *now builds or store the register string but it does this fine)`
`13:55.30`	`Inkubot`	`[TK]D-Fender: what do you mean ? i'm not qualified ?`
`13:55.38`	`Voicemeup`	`yeah no link to * .. just an out of subject question`
`13:55.42`	`[TK]D-Fender`	`Siya: Oh God.... GUI's`
`13:56.07`	`[TK]D-Fender`	`fromuser = 3110yyyyyyy <-------- hmmmmmmmmm`
`13:56.22`	`*** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de)`
`13:56.36`	`lilalinux`	`is anybody here using mISDN with debian?`
`13:56.42`	`[TK]D-Fender`	`secret = yyyyyyy <---------------- hmmmmmmmmmmmmm`
`13:56.44`	`*** join/#asterisk s0ck (n=m@unaffiliated/s0ck)`
`13:56.49`	`Siya`	`[TK]D-Fender: yeah SP requires it`
`13:56.50`	`[TK]D-Fender`	`Siya: LOOKS like auth to me.`
`13:57.14`	`Siya`	`[TK]D-Fender: hehe and the auth works fine when registering`
`13:57.23`	`[TK]D-Fender`	`Siya: users.conf = ASS`
`13:57.42`	`Siya`	`once a call comes in it doesn't need to register again right? My SP knows where and how to reach my server`
`13:57.57`	`[TK]D-Fender`	`Siya: You also completely misunderstand REGISTERING`
`13:58.00`	`Siya`	`[TK]D-Fender: tell digium that (I had nothing to do with that) :)`
`13:58.11`	`Siya`	`[TK]D-Fender: possible`
`13:58.30`	`[TK]D-Fender`	`Siya: Register only tells the server where to send calls. it doesn't mean they aren't AUTH'd on arrival`
`13:58.40`	`Siya`	`right ok`
`13:58.44`	`[TK]D-Fender`	`Siya: It only updates them on your IP`
`13:58.51`	`Siya`	`ic`
`13:59.20`	`[TK]D-Fender`	`Siya: Several ITSPs I've seen don't even SUPPORT registering and you need a fixed IP or hostname.`
`13:59.27`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`14:00.00`	`Siya`	`[TK]D-Fender: both I have, these are ITSP's which offer their own softclients and support * as well`
`14:00.17`	`[TK]D-Fender`	`Siya: Yes, I've seen them before.`
`14:00.21`	`ai-a`	`is there a way i can perform a perl, agi, script for all calls, to do some sql functions, and begin call recording based on the ext called, and called from ?`
`14:00.38`	`[TK]D-Fender`	`ai-a: yes.`
`14:01.02`	`ai-a`	`i added some exten = _X!,1,... but it wont continue the flow as its looking for pri 2 next.`
`14:01.06`	`Siya`	`so my server registers which tells my ITSP that for both number I have the IP is xyz`
`14:01.19`	`[TK]D-Fender`	`ai-a: And why the hell don't you HAVE a priority 2?`
`14:01.29`	`[TK]D-Fender`	`Siya: Correct.`
`14:01.39`	`[TK]D-Fender`	`Siya: And whent he call comes it... AUTH <-----------`
`14:01.42`	`ai-a`	`because im inserting it above previously made ext conf... created by the gui.`
`14:01.45`	`[TK]D-Fender`	`in*`
`14:01.57`	`[TK]D-Fender`	`ai-a: And who's falt is that? :)`
`14:02.00`	`[TK]D-Fender`	`fault*`
`14:02.04`	`Siya`	`Then when a call comes in my ITSP connects to the IP address and authenticates, how does * know which account/trunk to use`
`14:02.23`	`ai-a`	`in what context we blaming fault ? in using a gui or trying to modify the ext file ?`
`14:02.25`	`[TK]D-Fender`	`Siya: Because they sund AUTH with the call`
`14:02.52`	`[TK]D-Fender`	`ai-a: You tell me you don't have a priorty 2 for that exten. This is YOUR JOB. Go make it.`
`14:03.12`	`Siya`	`[TK]D-Fender: so either my ITSP is using the wrong auth for the call or * authenticates and then still picks the wrong account/trunk`
`14:03.22`	`[TK]D-Fender`	`ai-a: And the gui isn't there so you can invent whatever psycho dialplan you want. Its a completely stupid cookie-cutter system.`
`14:03.43`	`[TK]D-Fender`	`Siya: Correct. perhaps its the "insecure" options in there...`
`14:03.53`	`JT`	`lilalinux: run away!`
`14:03.55`	`ai-a`	`ok, the gui makes its own [default] exten 850,1,VoiceMail [foo] include default when i do exten _X!,1... alove that include,, 850 wont work now.`
`14:04.04`	`[TK]D-Fender`	`Siya: SOME places don't always auth calls once the IP is registered.`
`14:04.52`	`Siya`	`[TK]D-Fender: hmmm, I need to google me some sip call setup debugging manuals`
`14:04.55`	`[TK]D-Fender`	`ai-a: _X! is clearly NOT 850. there is no such thing as "above". It will look for the next priority for that EXACT pattern.`
`14:04.56`	`[GuS]`	`no luck [TK]D-Fender, the init.d scritp is not being copied inside the init.d dir.. :S`
`14:05.12`	`[TK]D-Fender`	`Siya: disable the "insecure" line and relaod.`
`14:05.14`	`tzafrir`	`[GuS], the current script is in contrib/init.d`
`14:05.41`	`[GuS]`	`ok, thanks tzafrir! Maybe this is a bug? cause i installed 2 versions before this and was woking great`
`14:05.50`	`JT`	`Siya: a few RFCs would go down a treat ;)`
`14:06.13`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`14:06.24`	`Voicemeup`	`Siya : also please note, if oyu have 2 devices (trunks) connected to same itsp (from same ip) it will use last one`
`14:06.41`	`tzafrir`	`[GuS], a known bug, IIRC. Until 1.4.5 (including) it broke the "config" target. Now that target only notifies you that it has failed to copy the init.d script`
`14:06.43`	`Voicemeup`	`this is a last matched used policy.. i think`
`14:06.47`	`[TK]D-Fender`	`Voicemeup: ummm.. wtf?`
`14:06.54`	`Voicemeup`	`unless it matched something else`
`14:06.56`	`Voicemeup`	`yeah`
`14:07.07`	`[TK]D-Fender`	`Voicemeup: We are not talking about fighting over the same ACCOUNT`
`14:07.09`	`Voicemeup`	`well.. ;) depends.. check it out`
`14:07.11`	`Voicemeup`	`ok`
`14:07.12`	`[GuS]`	`ahhh ok tzafrir, so is know...`
`14:07.13`	`Voicemeup`	`sorry then`
`14:07.32`	`Voicemeup`	`anyhow a call should come to your asterisk box from an ip..`
`14:07.38`	`JT`	`Voicemeup: what sort of scope were you after?`
`14:07.44`	`Voicemeup`	`and come to EXTEN@yourip`
`14:07.51`	`Voicemeup`	`ah john`
`14:07.58`	`Voicemeup`	`well i hear noise on the line..`
`14:08.01`	`Voicemeup`	`testing theory`
`14:08.02`	`Siya`	`Voicemeup: I tried to verify that but I only use the accounts for DID, I noticed no such behaviour though I had to reload sip to turn accounts off/on`
`14:08.18`	`Siya`	`JT: The implementation is what I'm worried about`
`14:08.38`	`[TK]D-Fender`	`Siya: You should really ditch the GUI and do this from scratch. You have much better control and an idea of whats going on.`
`14:08.50`	`JT`	`Voicemeup: an oscilloscope perhaps`
`14:08.55`	`Voicemeup`	`JT_ also.. if i unplug the ups ( and headphones behind ups) i hear no more humms...`
`14:09.16`	`Siya`	`[TK]D-Fender: if only I had the time`
`14:09.18`	`Voicemeup`	`but still can hear on local sound system here..`
`14:09.55`	`Voicemeup`	`so since intel etc.. has nice little app to monitor the power etc..was wondering why the hell cant they make a nice app to see the wave of the ac coming in the power supply or MB..`
`14:09.59`	`Voicemeup`	`MB actually`
`14:10.08`	`[GuS]`	`ok, now i have the script, thanks tzafrir :)`
`14:10.12`	`JT`	`that's dreaming`
`14:10.18`	`JT`	`you need to monitor it before that point`
`14:10.18`	`Voicemeup`	`yeah i know`
`14:10.24`	`*** join/#asterisk ramindia_ (n=ramindia@202.63.96.9)`
`14:10.27`	`Voicemeup`	`coze the power supply regulates it`
`14:10.32`	`JT`	`you can use the sound card as an oscilloscope`
`14:10.37`	`Voicemeup`	`but its still being heard.. in headfones..`
`14:10.39`	`Voicemeup`	`ah`
`14:10.43`	`Voicemeup`	`TRUE..`
`14:10.44`	`JT`	`and analogue tv tuner cards i think`
`14:10.50`	`JT`	`but not mains :)`
`14:10.58`	`Voicemeup`	`got a ati 9800 aiw here ;)`
`14:11.01`	`lilalinux`	`JT: pardon?`
`14:11.12`	`Voicemeup`	`sounffoundry audacity etc et c?`
`14:11.22`	`JT`	`lilalinux: misdn`
`14:11.40`	`Voicemeup`	`http://www.electronics-lab.com/downloads/pc/index.html`
`14:11.42`	`lilalinux`	`JT: well, everybody tells me something else`
`14:11.43`	`Voicemeup`	`not dreaming that much`
`14:11.50`	`Voicemeup`	`<PROTECTED>`
`14:11.50`	`Voicemeup`	`<PROTECTED>`
`14:11.50`	`Voicemeup`	`Oscilloscope for Windows is a Windows application that converts your PC into a powerful dual-trace oscilloscope and spectrum analyzer.`
`14:11.59`	`lilalinux`	`I was trying to get 2 hfc cards working together since 2 weeks`
`14:12.03`	`Voicemeup`	`http://www.zelscope.com/`
`14:12.10`	`JT`	`Voicemeup: you still need to monitor mains without blowing up your pc`
`14:12.15`	`JT`	`thx for the flood`
`14:12.15`	`lilalinux`	`with bristuff and zaphfc -> no chance`
`14:12.23`	`JT`	`lilalinux: err`
`14:12.30`	`JT`	`lilalinux: trying to link the two?`
`14:12.40`	`lilalinux`	`JT: no, one NT and one TE`
`14:12.56`	`JT`	`so you are trying to link them?`
`14:12.58`	`lilalinux`	`it tells me my CPU would be throttled`
`14:13.07`	`lilalinux`	`it tells me that about 10000000 times in a second`
`14:13.12`	`JT`	`what tells you that?`
`14:13.13`	`JT`	`err ok`
`14:14.09`	`lilalinux`	`I've read tons of sites, and most say I should use mISDN`
`14:14.31`	`lilalinux`	`now I wanted to use mISDN just to see that the maintainer of mISDN pissed off 1 year ago`
`14:14.31`	`JT`	`most people are idiots too ;) NT mode in misdn is not worth contemplating`
`14:15.06`	`JT`	`lilalinux: ok, so what have you actually done with bristuff so far?`
`14:15.45`	`lilalinux`	`bristuff works fine with 1 hfc card, but the kernel crashes when using 2 of them`
`14:15.56`	`Voicemeup`	`JT`
`14:16.00`	`Voicemeup`	`you wont believe this`
`14:16.02`	`JT`	`tried upgrading the kernel, or bristuff?`
`14:16.13`	`JT`	`Voicemeup: ?`
`14:16.14`	`Voicemeup`	`this scope uses the sounds card like you said.. it looks like AC on crack..`
`14:16.38`	`JT`	`Voicemeup: what's the sound card hooked up to?`
`14:16.46`	`Voicemeup`	`computer`
`14:16.50`	`Voicemeup`	`hooked to ups`
`14:16.52`	`Voicemeup`	`to ac`
`14:16.59`	`Voicemeup`	`ifi i unpug ups its clean`
`14:17.12`	`Voicemeup`	`BUT.. i also hear the huim on a non ups related sound system`
`14:17.12`	`JT`	`you have the mic in connected to the ups?`
`14:17.15`	`Voicemeup`	`so its in the lines`
`14:17.17`	`JT`	`i see`
`14:17.19`	`lilalinux`	`JT: problem is, we wanted to get rid of compiling stuff manually and solely use packages`
`14:17.22`	`JT`	`so your ups may be stuffed`
`14:17.32`	`JT`	`lilalinux: not a good strategy with asterisk`
`14:17.36`	`ramindia_`	`iam working on meetme, when iam making call i dont hear ring tone, until i connect and hear the voice,what is need to change to hear ring tones`
`14:19.05`	`lilalinux`	`JT: so with every kernel update, our PBX breaks and somebody has to recompile it? gr8`
`14:19.12`	`Voicemeup`	`http://img514.imageshack.us/my.php?image=test1hj0.gif`
`14:19.24`	`JT`	`lilalinux: how often do you randomly upgrade your kernel?`
`14:19.43`	`lilalinux`	`JT: whenever debian decides to`
`14:19.44`	`pj_`	`twice a day`
`14:19.46`	`Voicemeup`	`and this is the clean one http://img444.imageshack.us/my.php?image=test2vf2.gif`
`14:20.02`	`JT`	`lilalinux: maybe you should make it not do that`
`14:20.50`	`JT`	`randomly changing software on PBXes isn't advisable ;)`
`14:20.53`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`14:21.06`	`Voicemeup`	`ok found the fault.. ups is doign this.. for some reason..`
`14:21.21`	`*** join/#asterisk zuesman (n=pb0036@66.39.201.241)`
`14:21.22`	`JT`	`Voicemeup: this your whole military base ELF problem?`
`14:21.27`	`Voicemeup`	`nope`
`14:21.39`	`*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)`
`14:21.40`	`Voicemeup`	`doesnt explain the SN on the copper lines..`
`14:21.44`	`Voicemeup`	`coax i mean`
`14:21.51`	`Voicemeup`	`and its not me its the whole town`
`14:21.54`	`Siya`	`[TK]D-Fender: no auth as far as I can see... I just picks trunk_3 (still don;t know why) debug of the peer: sip*CLI>`
`14:21.58`	`Siya`	`<--- SIP read from 81.23.228.150:5060 --->`
`14:22.00`	`Siya`	`INVITE sip:s@217.195.248.252 SIP/2.0`
`14:22.03`	`Siya`	`Record-Route: <sip:81.23.228.150;lr=on;ftag=as4bf5f515;fcd=yes>`
`14:22.04`	`Voicemeup`	`i doubt my ups can generate enough crap for this.`
`14:22.22`	`Siya`	`rats... http://pastebin.ca/603060`
`14:22.36`	`lilalinux`	`JT: what's wrong with mISDN?`
`14:22.46`	`JT`	`alpha software`
`14:22.51`	`JT`	`hardly works`
`14:22.56`	`JT`	`has horrible debugging`
`14:23.02`	`JT`	`can't use chan_zap on it`
`14:23.25`	`lilalinux`	`IC`
`14:23.59`	`[TK]D-Fender`	`Siya: Also your register is not specifying the exten so it makes them hard to seperate. thats why its targeting "s"`
`14:24.10`	`Siya`	`hmmm`
`14:24.23`	`[TK]D-Fender`	`Siya: Ditch that GUI, and while you're at it, pastebin the ENTIRE call attempt`
`14:24.24`	`lilalinux`	`JT: Can you recommend a card that supports NT mode and is included in the kernel?`
`14:24.27`	`[TK]D-Fender`	`~pb`
`14:24.27`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org`
`14:24.29`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^`
`14:25.25`	`JT`	`lilalinux: nothing`
`14:25.32`	`*** part/#asterisk ramindia_ (n=ramindia@202.63.96.9)`
`14:25.48`	`[TK]D-Fender`	`Siya: Oh... you already did... nvm`
`14:25.50`	`JT`	`lilalinux: there is no telephony cards for asterisk that are made in the last half century that have drivers built into the kernel`
`14:25.51`	`*** join/#asterisk call (n=gabriel@c9069b1c.static.spo.virtua.com.br)`
`14:25.58`	`JT`	`s/is/are/`
`14:26.20`	`lilalinux`	`lol`
`14:26.29`	`[TK]D-Fender`	`Siya: "Looking for s in DID_trunk_3 (domain 217.195.248.252)" See it IS picking the right CONTEXT for the call apparently`
`14:26.56`	`Siya`	`[TK]D-Fender: pastebin updated but I doubt it will give more info as to why * is picking that 'trunk'`
`14:27.09`	`[TK]D-Fender`	`Siya: And HERE "Peer audio RTP is at port 83.149.75.105:62650" is where the ports are selected`
`14:27.54`	`Siya`	`the right context would be trunk_2 not trunk_3`
`14:28.00`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`14:28.00`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`14:28.01`	`Siya`	`so it does pick the wrong context`
`14:28.21`	`[TK]D-Fender`	`Siya: To: <sip:31107142242@budgetphone.nl> This could be stripped to find out where it should have gone. Now if your REGISTER was set up properly this would be easier.`
`14:28.34`	`[TK]D-Fender`	`Siya: You are going to suffer a LOT with that setup.`
`14:28.50`	`Siya`	`That's the audio port which has little to do with the sip session which build the call (the audio path)`
`14:29.19`	`*** join/#asterisk tako-san (n=Tako-san@154.5.212.245)`
`14:35.49`	`[TK]D-Fender`	`Siya: No, that has : "Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl" to identify it`
`14:36.53`	`[TK]D-Fender`	`Siya: If you fixed your register you'd be able able to easily direct your calls.`
`14:37.19`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`14:38.19`	`lilalinux`	`JT: do you recommend compiling asterisk manually, too?`
`14:38.30`	`pj_`	`oh yeah`
`14:38.38`	`Siya`	`[TK]D-Fender: I'll ask on #asterisk-gui how the dev's go about constructing the register string`
`14:39.07`	`af_`	`the flash operator panel works with 1.4?`
`14:39.35`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`14:39.53`	`*** join/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com)`
`14:39.58`	`*** join/#asterisk ManxPower (n=manxpowe@41.sub-75-202-64.myvzw.com)`
`14:41.29`	`E-bola`	`Do anybody know a webpage that has info/tips on setting up linksys SPA922's with asterisk?`
`14:42.05`	`*** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)`
`14:43.20`	`[TK]D-Fender`	`E-bola: You only have to fill in like 3 silly boxes...`
`14:43.58`	`[TK]D-Fender`	`E-bola: And Linksys is a waste in North America......`
`14:44.14`	`E-bola`	`im in europe`
`14:44.25`	`E-bola`	`And im looking for settings to improove the quality`
`14:44.33`	`E-bola`	`im currently upgrading the firmware to see if it improoves`
`14:44.39`	`E-bola`	`the speakerphone particularly is quite bad`
`14:44.41`	`[TK]D-Fender`	`E-bola: Didn't think so based on your connect, but ok... you're fogiven :)`
`14:44.54`	`[TK]D-Fender`	`E-bola: www.voxilla.com`
`14:45.00`	`E-bola`	`Always good to hide your ip :)`
`14:52.17`	`E-bola`	`what woudl i do on www.voxilla.com ?`
`14:53.21`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`14:53.27`	`InHisName`	`I have lotsa echos with SPA-3000 when PSTN call comes into my * and I use the FXS extension on the SPA-3000. What can I adjust to imrove it?`
`14:55.05`	`Voicemeup`	`f receiving a call and sounds MR roboto, but outound is perfect, Try this. In advanced mode and as an admin log into your device and go into the SIP tab. Find the section called RTP Packet Size: and change from 0.030 to 0.020`
`14:55.16`	`[TK]D-Fender`	`InHisName: WWW.VOXILL.COM <- GO CHECK THE FORUMS`
`14:55.30`	`[TK]D-Fender`	`InHisName: WWW.VOXILLA.COM <- GO CHECK THE FORUMS`
`14:55.34`	`[TK]D-Fender`	`silly caps.`
`14:56.18`	`InHisName`	`[TK]D-Fender, any specific topic to search down to fix my echos ?`
`14:56.46`	`Voicemeup`	`InHisName tried my thing ?`
`14:57.03`	`[TK]D-Fender`	`InHisName: Go into the linksys forum and LOOK`
`14:57.53`	`*** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de)`
`14:58.00`	`InHisName`	`Voicemeup, I'll check in a sec`
`14:58.07`	`Fl1p`	`hi, how can i enable Features Globally ?`
`14:58.51`	`Voicemeup`	`hmm wont fix echo..`
`14:59.04`	`Fl1p`	`so i dont have to SET(DYNAMIC_FEATURES... every time`
`15:01.58`	`ManxPower`	`I'm sorry, but Asterisk does not support lazy admins`
`15:02.03`	`ManxPower`	`use a macro if you want`
`15:04.19`	`InHisName`	`Voicemeup, what does the RTP packet size do ? Shorten the time from .3 to .2, makeing echos shorter ? Probalby a wrong guess.`
`15:04.35`	`*** join/#asterisk skyphyr (n=alanj@135.196.58.222)`
`15:04.40`	`Voicemeup`	`wrong nothing to do with echo`
`15:04.52`	`Voicemeup`	`its the packet sampling size`
`15:05.00`	`Voicemeup`	`my bad`
`15:05.10`	`ManxPower`	`Asterisk REQUIRES 0.20 RTP packet size. If the device is using any other packet size you will have massive audio problems.`
`15:05.15`	`skyphyr`	`hi all - is there some additional formating required for the [1234] pattern matching? I'm not getting expected results (i.e. a match) with exten => _N[78]XXXXXX`
`15:05.23`	`Voicemeup`	`are you using an iax trunk ?`
`15:05.28`	`InHisName`	`what benefit is there going from .3 to .2 ?`
`15:05.28`	`ManxPower`	`You will need to fix that issue before you have any chance of being able to fix echo`
`15:05.28`	`skyphyr`	`yes`
`15:05.29`	`Voicemeup`	`echo is always the endpoint device..`
`15:05.35`	`skyphyr`	`oh... oops`
`15:05.44`	`ManxPower`	`Voicemeup: ?WRONG! Echo is always on the far end analog loop`
`15:05.54`	`Voicemeup`	`yeah`
`15:05.56`	`Voicemeup`	`true`
`15:06.03`	`Voicemeup`	`hehe jumping steps here.. sorry`
`15:06.10`	`*** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com)`
`15:06.13`	`ManxPower`	`InHisName: The benefit is that your audio will sound better`
`15:06.28`	`Voicemeup`	`but if you had0.30 in rtp size make it 0.20 and retest`
`15:06.29`	`ManxPower`	`InHisName: now, go make the change and then come back here.`
`15:06.32`	`InHisName`	`Voicemeup, I have PSTN incoming calls on my SPA-3000 and pick up extn on other line of SPA-3000 thru the * machine.`
`15:06.53`	`Voicemeup`	`do the change then pish the pstn call to ECHO test`
`15:06.57`	`InHisName`	`Change made, just need to stop typing and test it out.`
`15:06.58`	`Voicemeup`	`on * and see if all good`
`15:07.03`	`[TK]D-Fender`	`InHisName: Go back to READING THE FORUM.`
`15:07.06`	`Voicemeup`	`start with basics`
`15:07.14`	`[TK]D-Fender`	`skyphyr: Looks fine`
`15:07.58`	`skyphyr`	`thanks [TK]D-Fender... wonder what I missed`
`15:08.03`	`skyphyr`	`oh wait`
`15:08.03`	`ManxPower`	`InHisName: The echo cannot be solved by Asterisk, because the SPA is the device doing the IP/PSTN conversion, the echo must be reduced on the SPA. So basically you are wasting yuur time here.`
`15:08.21`	`skyphyr`	`it's me - I wanted _[78]XXXXXXX`
`15:08.25`	`skyphyr`	`no _N`
`15:08.27`	`skyphyr`	`:_)`
`15:08.29`	`InHisName`	`Back to reading and I see four new messages I have not 'read' yet. Probably I need to read one from long ago...`
`15:08.45`	`[TK]D-Fender`	`skyphyr: that would be "bad" wouldn't it? ;)`
`15:09.03`	`[TK]D-Fender`	`InHisName: SEARCH. there are all sorts of articles on it`
`15:09.42`	`*** join/#asterisk joetester (n=joeteste@216.191.34.13)`
`15:10.44`	`skyphyr`	`[TK]D-Fender hehe - yes - I might trying thinking before editing extensions.conf :-)`
`15:11.06`	`*** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com)`
`15:11.29`	`joetester`	`Hey, quick sip question, wtf is that: WARNING[2641]: chan_sip.c:12440 handle_response: Remote host can't match request NOTIFY to call '044469cf16cb15ef7f6739e71fbd31be@myipaddress'. Giving up.`
`15:11.31`	`joetester`	`<PROTECTED>`
`15:12.08`	`lee_is_me`	`Can anyone point me to definitions for QueueMemberStatus? Looking at the source, I can see that it is an Integer Member of the "member" struct, but I have been unable to track down the CONSTANT definitions that represent QueueMemeberStatus`
`15:14.18`	`*** part/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com)`
`15:18.10`	`lee_is_me`	`I can see that there is an enum called queue_member_status, but that has only 3 values (0-2), but it seems to me that QueueMemberStatus returns values up to 6 which seems to match more the queue_result enum`
`15:18.36`	`lee_is_me`	`or would this question be more appropriate for dev list/channel?`
`15:22.04`	`lee_is_me`	`Anyone know the channel for the developers, maybe I could ask in there`
`15:22.44`	`lee_is_me`	`this reminds me of my first sexual experience. I was nervous, it was dark and I was all alone.`
`15:22.53`	`lee_is_me`	`;)`
`15:23.28`	`*** join/#asterisk tld (n=terje@elde.net)`
`15:24.19`	`blitzrage`	`lee_is_me: #asterisk-dev`
`15:24.22`	`ManxPower`	`lee_is_me: ask on the -dev mailing list`
`15:25.36`	`joetester`	`Hey blitzrage!`
`15:25.42`	`*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)`
`15:25.49`	`blitzrage`	`hey`
`15:26.06`	`joetester`	`How's it going, aren't you off work?`
`15:27.20`	`blitzrage`	`I live in Canada`
`15:28.46`	`joetester`	`Oh... I'm confused then. Drainbamage you know.`
`15:30.25`	`blitzrage`	`yep`
`15:30.30`	`blitzrage`	`how is it confusing?`
`15:30.31`	`[TK]D-Fender`	`blitzrage: And you get American Holidays with your job, no? :)`
`15:30.33`	`blitzrage`	`I live in Toronto...`
`15:30.36`	`lee_is_me`	`Thanks all, I'll try on the dev list or dev channel.`
`15:30.46`	`blitzrage`	`[TK]D-Fender: Canadian & American :)`
`15:30.57`	`[TK]D-Fender`	`blitzrage: BASTARD`
`15:31.06`	`*** join/#asterisk eAi2k (n=cow@81-86-205-45.dsl.pipex.com)`
`15:31.11`	`blitzrage`	`that's not really true`
`15:31.16`	`blitzrage`	`I don't get "holidays"`
`15:31.19`	`blitzrage`	`if I don't work, I don't get paid`
`15:31.42`	`joetester`	`And when DO you work exactly?`
`15:32.06`	`eAi2k`	`hey - I'm using an SPA-3102 with asterisk, and I've managed to get outbound calls (from a softphone) to work, but I can't get inbound calls to work. Does anyone know what I might have done wrong? I don't think asterisk is even getting the calls...`
`15:32.07`	`blitzrage`	`depends`
`15:33.30`	`[TK]D-Fender`	`eAi2k: Go follow the guides at www.voxilla.com`
`15:33.57`	`[TK]D-Fender`	`eAi2k: And know that that question you asked won't get you ANYWHERE since you provided no details.`
`15:34.05`	`[TK]D-Fender`	`eAi2k: You could have done EVERYTHING wrong.`
`15:34.52`	`eAi2k`	`I'm aware of that, thank you`
`15:36.03`	`mosty`	`eAi2k, inbound calls from where?`
`15:36.42`	`eAi2k`	`well, anywhere`
`15:36.49`	`eAi2k`	`PSTN`
`15:37.55`	`mosty`	`how is asterisk connected to the PSTN?`
`15:38.29`	`eAi2k`	`using an SPA-3102`
`15:38.31`	`eAi2k`	`(linksys)`
`15:38.36`	`eAi2k`	`I think thats the source of the problems`
`15:38.44`	`ManxPower`	`eAi2k: I had the same problem at one time.`
`15:39.04`	`ManxPower`	`It took a while to solve, but it was not all that difficult`
`15:39.14`	`mosty`	`that's an ATA right? so you're having problem with an ATA and not asterisk?`
`15:39.22`	`skyphyr`	`can I change the extension a macro believes it's being called from befor the macro is called?`
`15:39.34`	`eAi2k`	`I'm having problems with the connection between the two`
`15:39.40`	`ManxPower`	`skyphyr: Yes.`
`15:39.52`	`skyphyr`	`do I change MACRO_EXTEN or something else?`
`15:39.55`	`ManxPower`	`eAi2k: nod It was pretty easy for me to solve.`
`15:40.12`	`ManxPower`	`skyphyr: Use a goto to goto the extension you want, then run the macro`
`15:40.27`	`skyphyr`	`ahhh ok - thanks`
`15:40.33`	`ManxPower`	`Anytime you say "can I change the EXTEN value", the answer is "use a goto"`
`15:40.36`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`15:40.44`	`eAi2k`	`ManxPower: any idea how?`
`15:40.45`	`*** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net)`
`15:41.06`	`Voicemeup`	`Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (2) ]`
`15:41.17`	`ManxPower`	`eAi2k: Yes. I went to www.voxzilla.com and read the forums. Just like people have been telling you to do for the past 30 mins`
`15:41.32`	`Voicemeup`	`trying to figure out... that a busy or seomthing else ?`
`15:41.51`	`eAi2k`	`past 8 minutes actually`
`15:41.54`	`eAi2k`	`and once`
`15:41.57`	`skyphyr`	`lovely :-) btw I want a user's extension to either go to a physical phone or to a sip account (which each user has) - I'd just been setting variables for each user and using those in the dialplan - so I can change their physical phone later if I need - anything wrong with this plan?`
`15:41.58`	`eAi2k`	`and I've already done that`
`15:42.04`	`rbd`	`hey guys, I want to use the playback command to play an audio file not in the asterisk sounds directory. I can use symlinks to do this right?`
`15:42.04`	`Voicemeup`	`- Zap/1-1 is circuit-busy, disconnect(69) and a relase 77`
`15:42.10`	`ManxPower`	`Voicemeup: http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf`
`15:42.18`	`Voicemeup`	`rbd no use /fullpath`
`15:42.21`	`Voicemeup`	`no quotes`
`15:42.28`	`rbd`	`ok, sounds good`
`15:42.31`	`skyphyr`	`the phones are sip too, but I've given each hardphone it's own account`
`15:43.11`	`ManxPower`	`eAi2k: then you did not look hard enough.`
`15:43.28`	`ManxPower`	`eAi2k: But you are wasting your time asking here.`
`15:43.36`	`ManxPower`	`and wasting our time as well.`
`15:43.42`	`eAi2k`	`thank you for your help`
`15:44.14`	`Voicemeup`	`as in this code ? Cause: Circuit/channel congestion (34)`
`15:44.20`	`Voicemeup`	`or class = Network Congestion (2) ]`
`15:44.25`	`ManxPower`	`Voicemeup: the cause`
`15:44.50`	`Voicemeup`	`hmm that says no B channel avail and its BS`
`15:45.05`	`Voicemeup`	`since theres @ least 10 freeones.. and calls going trough ok`
`15:45.14`	`Voicemeup`	`except that one`
`15:45.30`	`ManxPower`	`Voicemeup: no, it means there is no B-Channel available SOMEWHERE between Asterisk and the destination number.`
`15:45.31`	`romano2k`	`Hi! Does anyone uses Asterisk with a Kiwak account?`
`15:45.48`	`ManxPower`	`You might get that if you tried using 0 as a leading number for a toll call in USA instead of the 1 that is required.`
`15:45.59`	`ManxPower`	`IT could also happen if your carrier did not have enough trunks`
`15:47.03`	`Voicemeup`	`ah`
`15:47.11`	`ManxPower`	`Voicemeup: carriers commonly return bad congestion codes, if you retry the call it then goes thru.`
`15:47.17`	`ManxPower`	`it could be caused by MANY issues.`
`15:47.20`	`Voicemeup`	`same`
`15:47.30`	`ManxPower`	`what is the CLI output of the Dial line`
`15:47.34`	`Voicemeup`	`no sounds.. nothing same code.. hmm calling GT ,,..`
`15:47.49`	`rbd`	`I sometimes will want to use the playback command to play a file I'm still streaming in to the filesystem. does the playback command stop playing back once it hits the end of the file stream, or will it wait until it hits an EOF marking?`
`15:47.52`	`lilalinux`	`has anybody experience with the http://pkg-voip.buildserver.net repository?`
`15:48.08`	`Voicemeup`	`<PROTECTED>`
`15:48.20`	`Voicemeup`	`<PROTECTED>`
`15:48.28`	`ManxPower`	`Voicemeup: stop wasting my time. Either paste the actual number or go away`
`15:48.40`	`ManxPower`	`you can /msg it to me if you want.`
`15:48.55`	`Voicemeup`	`hehe`
`15:49.03`	`ManxPower`	`The problem is likley the ACTUAL NUMBER or number format`
`15:49.17`	`Voicemeup`	`nah pm me`
`15:49.17`	`Voicemeup`	`ill paste the pri debug span 1`
`15:49.21`	`Voicemeup`	`i cant pm no idea why trillian crap`
`15:49.41`	`ManxPower`	`I don't want the debug span right now, I want the CLI output`
`15:49.56`	`ManxPower`	`IT is a lot of work to parse a PRI debug and usually you can solve the problem with out it.`
`15:50.45`	`*** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca)`
`15:50.55`	`Voicemeup`	`waoh.. didndt know pm where throttled too`
`15:51.33`	`ManxPower`	`that is why pastebin is such a great thing`
`15:51.37`	`*** join/#asterisk tako-san (n=Tako-san@154.5.212.245)`
`15:51.44`	`Voicemeup`	`pb'ed you`
`15:53.07`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`15:53.40`	`polerin`	`PB is awesomeness indeed`
`15:55.33`	`[TK]D-Fender`	`Voicemeup: Spamming people in PRIVATE too huh? PB > YOU`
`15:55.58`	`lilalinux`	`JT: should I use 1.2 or 1.4?`
`15:55.59`	`[TK]D-Fender`	`Voicemeup: best thing with PB is that with a decent one you can cut&paste direct to your own file.`
`15:56.31`	`[TK]D-Fender`	`lilalinux: 1.4 makes certain things easier, but 1.2 is considered more stable.`
`15:57.04`	`*** part/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de)`
`15:58.21`	`lilalinux`	`[TK]D-Fender: ok, but basically I can do everything with both of them?`
`15:58.34`	`lilalinux`	`e.g. I don't need the new config format`
`15:58.45`	`lilalinux`	`if that's the only difference`
`15:59.30`	`mosty`	`1.4 has new commands, more features`
`15:59.59`	`__DAW`	`and many deprecated commands in 1.2 are gone in 1.4`
`16:00.47`	`[TK]D-Fender`	`lilalinux: Pretty much anything of any importance`
`16:00.56`	`lilalinux`	`thx`
`16:05.28`	`ManxPower`	`and UPGRADE.txt should tell you everything you need to know.`
`16:05.54`	`*** join/#asterisk tako-san (n=Tako-san@24.68.129.29)`
`16:06.44`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`16:10.35`	`*** join/#asterisk Phuntom (n=Phuntom@80.233.159.254)`
`16:10.40`	`Phuntom`	`hi ya!`
`16:10.51`	`*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)`
`16:11.16`	`Phuntom`	`hi ya! have anyone installed asterisk+openserv?`
`16:13.02`	`*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)`
`16:13.03`	`Hmmhesays`	`openserv or openser`
`16:13.48`	`Phuntom`	`openser`
`16:13.51`	`Phuntom`	`sorry`
`16:13.52`	`Hmmhesays`	`aye`
`16:14.21`	`Phuntom`	`what manual have you used`
`16:14.27`	`Phuntom`	`?`
`16:14.45`	`Hmmhesays`	`the default routing script in openser has a lot of notes`
`16:16.35`	`Phuntom`	`openser.cfg?`
`16:17.25`	`*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)`
`16:19.25`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`16:20.21`	`*** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net)`
`16:32.58`	`*** join/#asterisk railsaddict (n=RailsAdd@ppp-70-246-49-173.dsl.rcsntx.swbell.net)`
`16:34.16`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`16:37.37`	`*** join/#asterisk tako-san (n=Tako-san@24.68.129.29)`
`16:47.42`	`*** join/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com)`
`16:48.58`	`Andri[DK]`	`Whined about this yesterday but... does anyone know why on earth playing wave files through my Asterisk (1.4) just doesn't work. Congestion, Busy, MusicOnHold and talking to other extensions works perfectly, just playing sound files is completely silent, including VoiceMail`
`16:51.24`	`*** join/#asterisk nasls_lsa (n=chatzill@85.72.164.18)`
`16:51.39`	`*** part/#asterisk nasls_lsa (n=chatzill@85.72.164.18)`
`16:52.28`	`mosty`	`Andri[DK], maybe you should submit a bug report. or use a different format for now`
`16:53.07`	`Andri[DK]`	`hmm, what other formats can i try ?`
`16:53.59`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`16:54.20`	`Andri[DK]`	`This just feels like a very silly error since mp3 playing works and even .wav files through moh`
`16:54.22`	`mosty`	`any of the other formats asterisk supports. use whatever your devices use most often, or alaw`
`16:54.32`	`*** join/#asterisk tako-san (n=Tako-san@24.68.129.29)`
`16:54.40`	`torch`	`quick question ... I'm running asterisk 1.2.18 and everytime I restart the asterisk (or zapata module) ...calls from my pstn to my asterisk extensions are just mute ...`
`16:54.43`	`torch`	`am I doing something wrong?`
`16:54.54`	`torch`	`does anybody know why?`
`16:55.04`	`*** join/#asterisk mrdigital (n=mrdigita@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com)`
`16:55.26`	`mosty`	`torch, what does the full log say?`
`16:56.02`	`torch`	`got check ... do you want to see the full log just after I restart de asterisk ?`
`16:56.21`	`skyphyr`	`am I being excessively hopeful here? http://www.pastecode.org/35`
`16:56.31`	`mrdigital`	`how do i get a fxo pci card working on asterisk X101p`
`16:56.34`	`skyphyr`	`not sure how I can go about what I'm after with this`
`16:57.24`	`skyphyr`	`it seems that you can't use variables as extensions in extensions.conf?`
`16:57.59`	`Zeeek`	`you can`
`16:58.29`	`Strom_M`	`skyphyr: no, that's not valid syntax`
`16:58.35`	`*** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)`
`16:59.22`	`Strom_M`	`pattern matches are acceptable, and you can use conditional branching to change behavior based on the contents of variables`
`16:59.25`	`[TK]D-Fender`	`skyphyr: exten => ${EXT_${EXTEN}},1,Macro(inbound,${${EXT_${EXTEN}}}) <_ you cannot do variables or expressions as the exten itself`
`17:00.21`	`[TK]D-Fender`	`skyphyr: exten => _3XX,1,GoTo(${EXT_${EXTEN}}) <- that only fills in a PRIORITY. Go read the parameter list again`
`17:00.24`	`torch`	`mosty, just got from my CLI "!! Got a UA, but i'm in state 1"`
`17:00.38`	`[TK]D-Fender`	`sky and priority jumping like you have in there is so 1.0`
`17:00.50`	`torch`	`mosty, and then ..."B-channel 0/1 successfully restarted on span 1" (for all the channels)`
`17:01.08`	`mosty`	`torch, look in the full log, not in the console log`
`17:01.40`	`*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)`
`17:02.02`	`skyphyr`	`thanks [TK]D-Fender - think that's given me enough to figure a decent solution :-)`
`17:02.30`	`*** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com)`
`17:03.49`	`torch`	`mosty, where should I post the log? it's huge!!`
`17:05.11`	`[TK]D-Fender`	`skyphyr: http://www.pastecode.org/36`
`17:05.20`	`[TK]D-Fender`	`torch: www.pastebin.ca`
`17:05.31`	`skyphyr`	`hmmm odd error here - app_voicemail.c: Unable to read password`
`17:06.05`	`[TK]D-Fender`	`skyphyr: Nows a good time to set your DTMFMODE as well....`
`17:06.18`	`skyphyr`	`[TK]D-Fender - thanks - sorry I meant an error with what I was trying, not yours`
`17:06.29`	`skyphyr`	`DTMFMODE is for the SIP connection?`
`17:06.41`	`[TK]D-Fender`	`skyphyr: indeed`
`17:06.46`	`skyphyr`	`thanks :-)`
`17:10.27`	`torch`	`mosty: http://www.pastebin.ca/603235`
`17:10.28`	`skyphyr`	`lovely - had to set it on the phone as well, but now it's all working`
`17:12.45`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`17:14.41`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`17:14.44`	`skyphyr`	`thanks again for the help - time to head home now`
`17:15.20`	`[TK]D-Fender`	`torch: yay, a completely BROKEN FreePBX setup.`
`17:15.22`	`[TK]D-Fender`	`~freepbx`
`17:15.23`	`jbot`	`hmm... freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there`
`17:15.25`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^^^^^`
`17:15.43`	`torch`	`TK: really??! whad did I do wrong?`
`17:15.54`	`torch`	`I mean ... I did follow all the docs`
`17:16.27`	`[TK]D-Fender`	`torch: So many misincluded contexts, app errors, etc...... its a flaming mess.`
`17:16.36`	`[TK]D-Fender`	`torch: and this is NOT a FReePBX support channel`
`17:17.16`	`torch`	`TK: sorry about the freepbx question .. my bad :-)`
`17:17.24`	`torch`	`well .. me fix that first then .. :-)`
`17:17.29`	`mosty`	`torch, you should try #freepbx for that, they will know better than us`
`17:17.44`	`torch`	`k guy ..thx a lot`
`17:17.48`	`torch`	`guys`
`17:30.57`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`17:30.57`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`17:36.41`	`*** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net)`
`17:43.15`	`blitzrage`	`anyone here use pfsense?`
`17:43.39`	`blitzrage`	`...the reason I ask is...`
`17:43.40`	`blitzrage`	`<blitzrage> hey all... I'm running 1.2-BETA1 on my soekris box, and running into an issue where I'm trying to get multiple SIP phones to register, which didn't seem to have an issue with 1.0.1 (or my linksys router... or iptables on my other linux router), and curious if anyone has seen this happen before? I even tried resetting the state table and rebooting all the phones... I can get one phone to register to server_`
`17:43.40`	`blitzrage`	`1, another p`
`17:43.41`	`blitzrage`	`<blitzrage> hoen to register to server_2, but my third phone won't register to server_1 or server_2 if it boots up after the first 2 phones (but does register if it is the first phone to boot)`
`17:50.38`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`17:51.34`	`mosty`	`run a packet logger`
`17:57.46`	`*** join/#asterisk tako-san (n=Tako-san@24.108.162.254)`
`17:59.06`	`*** join/#asterisk joetester (n=joeteste@216.191.34.13)`
`18:03.33`	*** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com)
`18:04.39`	`*** join/#asterisk ovnicraft (n=logan@190.10.180.87)`
`18:04.50`	`ovnicraft`	`i have an error 502`
`18:05.24`	`ovnicraft`	`with x-lite`
`18:05.25`	`[TK]D-Fender`	`ovnicraft: Cool, thats like TWICE as good as an error 251`
`18:05.36`	`ovnicraft`	`calling`
`18:05.47`	`ovnicraft`	`ups..`
`18:06.16`	`ovnicraft`	`with my x-lite calling 393612 result is error 502 bad gateway`
`18:07.04`	`[TK]D-Fender`	`ovnicraft: well go point it to your * server then`
`18:07.21`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`18:11.17`	`*** join/#asterisk tsurko (n=tsurko@77.70.15.52)`
`18:12.12`	`ovnicraft`	`the nwaht`
`18:12.14`	`ovnicraft`	`waht`
`18:12.16`	`ovnicraft`	`what`
`18:14.04`	`[TK]D-Fender`	`ovnicraft: PASTEBIN is your friend.`
`18:14.06`	`[TK]D-Fender`	`~pb`
`18:14.06`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org`
`18:14.08`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^^^^^^`
`18:14.38`	`*** join/#asterisk tako-san (n=Tako-san@24.108.162.254)`
`18:18.09`	`tsurko`	`hello, i have a question about AEL2`
`18:19.02`	`tsurko`	`let's say I have a context with some vars in it - are there individual "copy" of this vars (if there are several paralel calls in this context) for each call?`
`18:19.31`	`Strom_M`	`if they're channel variables, then yes; each channel has its own instance of those variables`
`18:20.02`	`*** join/#asterisk gardo (n=gardo@121.97.211.162)`
`18:20.04`	`tsurko`	`how are channel variables declared?`
`18:20.10`	`[TK]D-Fender`	`tsurko: They aren't`
`18:20.22`	`[TK]D-Fender`	`tsurko: Read(jsutavarimadeupnow)`
`18:20.37`	`*** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk)`
`18:22.57`	`tsurko`	`i see`
`18:23.57`	`tsurko`	`and what about if they are declared in a macro - if the macro is called by several times can the vars be different?`
`18:24.33`	`tsurko`	`more precisely - if a value is set in "instance a" of the macro will it affect "instance b" ?`
`18:24.46`	`Strom_M`	`tsurko: it`
`18:24.54`	`Strom_M`	`it's associated on a per-channel basis`
`18:25.37`	`*** join/#asterisk guillote_GNU (n=guillote@host252.200-82-63.telecom.net.ar)`
`18:26.40`	`tsurko`	`okay, and last question - is there something like C pointers in AEL?`
`18:28.01`	`*** join/#asterisk ManxPower (n=manxpowe@015-828-410.area5.spcsdns.net)`
`18:29.10`	`*** join/#asterisk irule (n=irule@189.164.43.194)`
`18:48.31`	`[TK]D-Fender`	`tsurko: No`
`18:48.49`	`[TK]D-Fender`	`tsurko: AEL is NOT a programming langaue. it IS extensions.conf parsed a little differently.`
`18:49.55`	`*** join/#asterisk Yomer (i=Y0mer@207.193.204.50)`
`18:49.56`	`FuriousGeorge`	`[TK]D-Fender: it is something of a "logical" language, wouldnt you say?`
`18:50.06`	`Yomer`	`hi there`
`18:50.17`	`ManxPower`	`in fact AEL is translated into standard dialplan stuff at startup`
`18:50.26`	`[TK]D-Fender`	`FuriousGeorge: it is exactly 0% smarter than extensions.conf, because thats what it gets parsed back to on load.`
`18:51.01`	`FuriousGeorge`	`[TK]D-Fender: even extensions.conf though, has conditional branching, you can mangle a loop and arrays out of it`
`18:51.09`	`Yomer`	`has anyone worked with liknsys SPA400's? im havinf trouble with caller id`
`18:51.17`	`[TK]D-Fender`	`FuriousGeorge: Yes, indeed you can.`
`18:51.18`	`ManxPower`	`It may, however, be easier for someone with programming experience to write AEL because they are more familiar with programming style syntax than the original Asterisk dialplan syntax`
`18:51.43`	`[TK]D-Fender`	`ManxPower: Yeah, but the similarity ends REAL fast, doesn't it? :)`
`18:52.39`	`ManxPower`	`[TK]D-Fender: I don't know. I'm waiting for 1.4/AEL2 to be stable before I start translating my macros, subroutines, etc from extensions.conf format to AEL`
`18:53.17`	`[TK]D-Fender`	`ManxPower: I'm surprised you'd bother. AEL1/2 is just a rehash of what you already know giving nothing new. Its just 1 more thing to break.`
`18:53.28`	`FuriousGeorge`	`ManxPower: is AEL2 supposed to be "smarter", as [TK]D-Fender said, than extension.conf and by extension ael.conf`
`18:53.35`	`ManxPower`	`[TK]D-Fender: you have apparently not seen my macros.`
`18:53.51`	`[TK]D-Fender`	`ManxPower: not THAT bad really, and yes I have ;)`
`18:53.58`	`*** join/#asterisk tako-san (n=Tako-san@24.108.162.254)`
`18:54.14`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net)`
`18:54.22`	`ManxPower`	`FuriousGeorge: no, AEL2 is supposed to work better because a token parser was used (flex?) rather than trying to do all that tokenizing internally`
`18:54.48`	`FuriousGeorge`	`learn something new everyday in here`
`18:54.55`	`ManxPower`	`i.e. it should have vastly fewer bugs`
`18:55.03`	`ManxPower`	`I don't believe much about the "language" has changed.`
`18:55.24`	`irule`	`I have a 2wire "HomePortal 1800HW" adsl router, my boss asked me to tell him what pots to fw to asterisk for SIP to work, I repplied with 5060 TCP and UDP, plus 10000-20000 range UDP, that is what I investigated, and confirmed that those values are in rtp.conf and sip.conf, my boss comes back telling me that he can place calls but the 3rd callee can not hear what the caller is saying. sipphone - internet - router - * - zap - POTS ...`
`18:55.48`	`irule`	`am I missing other ports?`
`18:56.01`	`FuriousGeorge`	`you need to foreward 10000-20000 rdp from router to your boss' phone`
`18:56.12`	`ManxPower`	`irule: DO YOU DO NOT!!!!!!!!!!!`
`18:56.26`	`FuriousGeorge`	`ManxPower: no?`
`18:56.29`	`irule`	`to the phone? lol`
`18:56.36`	`ManxPower`	`Port forwarding is normally only required if ASTERISK is behind NAT.`
`18:56.42`	`ManxPower`	`you don't need to portforward to the phone.`
`18:56.45`	`irule`	`the phone is NOT in my ooffice FuriousGeorge`
`18:56.57`	`Juggie`	`irule, did you enable nat=yes for the phone`
`18:57.01`	`FuriousGeorge`	`i meant to say to asterisk, not to your boss' phone`
`18:57.02`	`irule`	`it is at my bosses home`
`18:57.09`	`irule`	`let me see`
`18:57.18`	`irule`	`is that oonly in sip.conf?`
`18:57.20`	`ManxPower`	`you set nat=yes in the [device] section for that phone, then make sure Asterisk is on a PUBLIC IP address.`
`18:57.31`	`Strom_M`	`irule: is the phone itself behind a separate NAT?`
`18:57.40`	`Juggie`	`irule, its in the peer definition for the phone`
`18:57.44`	`FuriousGeorge`	`or if its not get a dyndns service and use externhost=yes`
`18:57.59`	`ManxPower`	`Now if Asterisk is behind a NAT router then you need to do portforwarding on the asterisk NAT router as well as localnet and externhost/externup`
`18:58.02`	`irule`	`Strom_M good question`
`18:58.13`	`irule`	`Ill hace to check with my boss`
`18:58.16`	`ManxPower`	`FuriousGeorge: please be quiet or read more carefully`
`18:58.19`	`FuriousGeorge`	`Strom_M: that has never made a different`
`18:58.24`	`Juggie`	`i would assume the phone is behind nat`
`18:58.25`	`FuriousGeorge`	`difference to me`
`18:58.28`	`FuriousGeorge`	`ManxPower: relax dude`
`18:58.44`	`ManxPower`	`FuriousGeorge: When you stop giving out wrong information`
`18:58.45`	`irule`	`yes we are all friends here! :D`
`18:58.47`	`FuriousGeorge`	`99% chance he is having nat issues, just my take`
`18:58.53`	`irule`	`lol`
`18:58.57`	`irule`	`naaaaaaaaaaaa`
`18:59.13`	`Juggie`	`FuriousGeorge, and using a dynamic ip service will fix that how?`
`18:59.20`	`ManxPower`	`Asterisk behind NAT: port forward UDP 5060 and the ports in rtp.conf, use externip=/externhost= and localnet.`
`18:59.50`	`ManxPower`	`Phone behind NAT: nat=yes and canreinvite=no in the phone's sip.conf section and that is all`
`19:00.08`	`Juggie`	`oooo`
`19:00.09`	`FuriousGeorge`	`Juggie: if asterisk is behind nat you want to use either externip or externhost to fix the SIP headers`
`19:00.21`	`irule`	`nat=no in all my phones, oops! lol`
`19:00.31`	`ManxPower`	`irule: are your phones behind nat?`
`19:00.36`	`Juggie`	`irule, also like manx said, make sure canreinvite=no`
`19:00.51`	`irule`	`ok`
`19:01.07`	`ManxPower`	`irule: port forwarding on the phone's NAT router can cause problems.`
`19:01.22`	`FuriousGeorge`	`plus, you dont need to do it`
`19:01.33`	`irule`	`it?`
`19:01.51`	`Juggie`	`there should be no custom settings on the router at the other end, except if they are running a hardware firewall and need to let the traffic in/out`
`19:02.03`	`irule`	`what are externip or externhost?`
`19:02.22`	`FuriousGeorge`	`i have a bunch of servers all behind nat. i interface them with a sip phone also behind nat at my house. i need to do exactly 0 configuration on the phones end to make that work`
`19:02.22`	`Juggie`	`settings in sip.conf only used if your asterisk server is not on a public ip.`
`19:03.18`	`Juggie`	`irule, does your * box have a public ip?`
`19:04.17`	`tsurko`	`irule, I have similar setup, and I'm using openvpn, to connect the client with Asterisk. But you'll have to have another PC acting like router.`
`19:04.17`	`irule`	`juggie dynamically asigned by adsl provider. in changes on every reconnect`
`19:04.46`	`FuriousGeorge`	`get a dyndns service of some sort`
`19:04.56`	`Juggie`	`irule, i'm talking about your * server now, not the client end.`
`19:04.57`	`irule`	`tsurko I told that to my boss, he said he does not want openvpn :s`
`19:05.19`	`Juggie`	`it is directally connected to an adsl modem or at least has a internet ip from the adsl modem.. which changes?`
`19:05.36`	`irule`	`jugguie adsl lan router internal ip is 192.168.5.254 and * is 192.168.5.3`
`19:06.00`	`FuriousGeorge`	`irule: and the external ip changes?`
`19:06.01`	`irule`	`my boss uses cute little GUI from router to forward calls`
`19:06.14`	`Juggie`	`irule, ok then, you need to set externhost=yourdynamichost.com in sip.conf`
`19:06.28`	`Juggie`	`and then sign up for some dynamic ip service and setup your * box to keep its ip updated`
`19:06.29`	`FuriousGeorge`	`is there an echo in here?`
`19:06.29`	`irule`	`192.168.5.254 and 192.168.5.3 are static`
`19:06.37`	`Juggie`	`this i would like to say is a less then ideal solution.`
`19:06.48`	`Juggie`	`and no one in their right mind would recomend it`
`19:06.56`	`FuriousGeorge`	`like you and i just did`
`19:07.20`	`FuriousGeorge`	`i definitely recommend getting a static ip over that, yes`
`19:08.11`	`irule`	`bosses bosses bosses! their only goal in life is to spend less money lol`
`19:08.28`	`FuriousGeorge`	`exactly`
`19:08.32`	`Juggie`	`a static ip should not cost very much on top of your existing adsl plan`
`19:08.38`	`Juggie`	`presuming you are using a business adsl plan`
`19:09.03`	`irule`	`+100 USD for my country :s`
`19:09.13`	`FuriousGeorge`	`externhost will work`
`19:09.24`	`FuriousGeorge`	`try that and decide if you need something better`
`19:09.58`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net)`
`19:10.28`	`FuriousGeorge`	`so like i was starting to say, since this is 99% chance a nat issue, you need to set externhost and localnet in sip.conf, foreward the correct udp prots to asterisk (10000-20000), and set nat=yes on that phone`
`19:10.31`	`irule`	`how about the current ip address in externhost?`
`19:10.40`	`FuriousGeorge`	`that will work till it changes`
`19:10.54`	`FuriousGeorge`	`www.dyndns.com or something`
`19:11.36`	`FuriousGeorge`	`your "router" probably has a built in feature for an account with some service, check that first`
`19:11.51`	`FuriousGeorge`	`also need to pass port 5060 for sip signalling to asterisk`
`19:12.30`	`FuriousGeorge`	`its not the most obvious setup in the world, but it works`
`19:12.35`	`FuriousGeorge`	`ymmv`
`19:13.33`	`FuriousGeorge`	`i have great results in one place using "business" cable ISP (no static ip possible), and same residential service at home`
`19:13.55`	`irule`	`oh cools`
`19:14.13`	`FuriousGeorge`	`but in the end if you think that VoIP is gonna be as reliable as PSTN or PRI you are mistaken`
`19:14.19`	`FuriousGeorge`	`make sure your users know that`
`19:14.57`	`FuriousGeorge`	`and if you call out over ADSL ISP, make sure your users have at least one POTS line incase that chan is UNAVAIL`
`19:19.10`	`FuriousGeorge`	`i cant emphasize that last point enough really, where im from, if people can't call emergencey services when they need to, you can/will get sued`
`19:19.13`	`*** join/#asterisk crimethinker (i=ircuser@legacy.diamond.org)`
`19:24.22`	`FuriousGeorge`	`im off`
`19:24.32`	`irule`	`thanks`
`19:24.35`	`irule`	`cu`
`19:28.02`	`*** join/#asterisk FuzzyB (n=FuzzyB@c-24-20-152-73.hsd1.mn.comcast.net)`
`19:28.38`	`FuzzyB`	`what does one dial to get into his or her voicemail on an asterisk server?`
`19:29.35`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`19:29.35`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`19:30.00`	`[TK]D-Fender`	`FuzzyB: Whatever you told it to use.`
`19:30.20`	`FuzzyB`	`i didn't set it up, and the guy that did isn't around right now and i've forgotten what he told me`
`19:30.36`	`[TK]D-Fender`	`FuzzyB: YOU tell * what to do with an call arriving at your server. So put it wherever you want, however you want.`
`19:30.40`	`FuzzyB`	`my call log shows that I have voicemails, i'm also getting email annoucements about it`
`19:31.00`	`[TK]D-Fender`	`FuzzyB: Well we aren't psychic. If you don't know your own setup, we know even LESS`
`19:31.11`	`FuzzyB`	`hrm`
`19:31.24`	`[TK]D-Fender`	`FuzzyB: I could say dial 28374687236478263478 for VM and that COULD be right, but I wouldn't bet on it.`
`19:31.24`	`FuzzyB`	`well at last now I know that much, thank you`
`19:32.05`	`[TK]D-Fender`	`FuzzyB: I'm figuring right now you can't show us your configs either.... would that be the case?`
`19:32.10`	`FuzzyB`	`yea`
`19:32.25`	`FuzzyB`	`so # brings me to a directory`
`19:32.26`	`[TK]D-Fender`	`FuzzyB: Ok, You've got yourself painted right into a corner then.`
`19:32.29`	`FuzzyB`	`yea`
`19:32.38`	`FuzzyB`	`but at least thanks for telling me all this`
`19:32.44`	`[TK]D-Fender`	`FuzzyB: There's no telling where or how any of that was set up for you.`
`19:32.45`	`FuzzyB`	`is there a common way it's setup?`
`19:32.59`	`[TK]D-Fender`	`FuzzyB: Nope. * is whatever you make it to be.`
`19:33.26`	`FuzzyB`	`ok`
`19:33.48`	`[TK]D-Fender`	`FuzzyB: I could say that "123" will dial my cell unless its tuesday and raining in New York.`
`19:34.06`	`FuzzyB`	`sweet`
`19:34.18`	`FuzzyB`	`if i could make my phone not bother me when it was raining`
`19:34.21`	`FuzzyB`	`that would be the shit`
`19:34.34`	`blitzrage`	`[TK]D-Fender: the funny thing is you really COULD only send it unless it was raining if you hooked asterisk into a weather report :)`
`19:34.35`	`[TK]D-Fender`	`FuzzyB: Entirely do-able.`
`19:34.50`	`FuzzyB`	`hehe`
`19:35.06`	`[TK]D-Fender`	`blitzrage: Yeah... and like is that Raw Cat Science?!?! NOT!`
`19:35.48`	`irule`	`FuzzyB exten => 8500,1,VoicemailMain in your dialplan, restart * and dial 8500 with your phone`
`19:35.55`	`[TK]D-Fender`	`blitzrage: Oh, I finally setup WordPress on my server. Time to start writing those Tutorials I've been thinking about.`
`19:36.19`	`[TK]D-Fender`	`irule: He has no access to his setup, nor do I suspect he knows the slightest thing about * at all`
`19:36.26`	`FuzzyB`	`8500 sounds like a test circuit`
`19:36.40`	`irule`	`:)`
`19:36.42`	`[TK]D-Fender`	`irule: I'd tell you what else I suspect, but well... I'll just wait for the incriminating evidence to present itself ;)`
`19:36.49`	`blitzrage`	`8500 is the default VM extension on Cisco phones`
`19:37.01`	`FuzzyB`	`ah`
`19:37.05`	`FuzzyB`	`ty`
`19:37.37`	`FuzzyB`	`w00t`
`19:37.39`	`FuzzyB`	`i figured it out`
`19:38.21`	`irule`	`if you have no access, Id start the server with trinity os, run mountallfs, start sshd and confortably edit files from my desktop lol`
`19:39.02`	`*** join/#asterisk javb (n=javb@190.80.224.21)`
`19:39.29`	`javb`	`i have a TDM400P, since this morning, just ONE CHANNEL is getting TOTALLY distotion when im trying to use it.`
`19:39.36`	`javb`	`Could it be the FXO module? or what?`
`19:39.38`	`FuzzyB`	`dial my extension and then hit *`
`19:39.43`	`javb`	`It was working great yesterday`
`19:39.50`	`FuriousGeorge`	`javb: did you try that line on a different channel`
`19:40.02`	`FuriousGeorge`	`sounds like it could be a problem on the pole`
`19:40.42`	`javb`	`FurioysGeoge, yes, an the line is ok.`
`19:40.46`	`FuzzyB`	`funny that didn't work the 2nd time`
`19:40.59`	`javb`	`I have 4 lines, put another line, in that port, and STILL GETTING THE HORRIBLE SOUND`
`19:41.21`	`FuriousGeorge`	`javb: im afraid the only way to be sure is to pop open the box and try swapping modules`
`19:41.33`	`FuriousGeorge`	`see what the problem follows`
`19:41.39`	`javb`	`:(`
`19:41.46`	`javb`	`Wow... i will..`
`19:42.00`	`[TK]D-Fender`	`javb: see if its config is different than the others. If it isn't, then things have just gone flakey. Could be Zaptel, could be the module/card.`
`19:42.07`	`javb`	`just came here before to see if someone else has had this issue before.`
`19:42.13`	`[TK]D-Fender`	`javb: Do a full reboot and if it clears well.....`
`19:42.23`	`FuriousGeorge`	`nah, mine would just stop working when i used em`
`19:42.27`	`FuriousGeorge`	`and it was usually fxs`
`19:42.33`	`javb`	`I undestand`
`19:43.04`	`javb`	`let me try a new reboot`
`19:44.40`	`javb`	`we havent had storm o anything that would damage the card`
`19:45.06`	`javb`	`and just THAT port is no working, i dont think rebooting will do anything, but "fingers crossed"`
`19:48.13`	`javb`	`Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)`
`19:48.16`	`javb`	`any ideas now?`
`19:48.38`	`[TK]D-Fender`	`javb: * hasn't been started`
`19:48.49`	`[TK]D-Fender`	`javb: First guess : zaptel error`
`19:48.55`	`FuzzyB`	`well thank you`
`19:48.59`	`*** part/#asterisk FuzzyB (n=FuzzyB@c-24-20-152-73.hsd1.mn.comcast.net)`
`19:49.11`	`javb`	`[TK]D-Fender, can u elaborate?`
`19:49.13`	`JT`	`lilalinux: yes compile, yes 1.2, yes bristuff`
`19:49.14`	`blitzrage`	`javb: run asterisk in the forground to see what the issue is`
`19:49.17`	`[TK]D-Fender`	`javb: make sure the modules are loaded & run "ztcfg -vvvv" before starting *`
`19:50.01`	`javb`	`Channel map:`
`19:50.01`	`javb`	`Channel 01: FXS Kewlstart (Default) (Slaves: 01)`
`19:50.01`	`javb`	`Channel 02: FXS Kewlstart (Default) (Slaves: 02)`
`19:50.01`	`javb`	`Channel 03: FXS Kewlstart (Default) (Slaves: 03)`
`19:50.01`	`javb`	`Channel 04: FXS Kewlstart (Default) (Slaves: 04)`
`19:50.02`	`javb`	`4 channels configured.`
`19:50.04`	`javb`	`Changing signalling on channel 1 from Unused to FXS Kewlstart`
`19:50.06`	`javb`	`Changing signalling on channel 2 from Unused to FXS Kewlstart`
`19:50.10`	`javb`	`Changing signalling on channel 3 from Unused to FXS Kewlstart`
`19:50.12`	`javb`	`Changing signalling on channel 4 from Unused to FXS Kewlstart`
`19:50.14`	`javb`	`i have this..`
`19:51.21`	`[TK]D-Fender`	`PASTBIN`
`19:51.23`	`[TK]D-Fender`	`~pb`
`19:51.24`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org`
`19:51.35`	`[TK]D-Fender`	`javb : do NOT spam in here like that`
`19:51.56`	`[TK]D-Fender`	`javb: Fine, no go try and start *`
`19:51.57`	`javb`	`oh, sorry.`
`19:52.02`	`javb`	`It started.`
`19:52.09`	`*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)`
`19:53.00`	`javb`	`[TK]D-Fender, now, the channel 2 works...`
`19:53.15`	`javb`	`I would like to undestand what JUST happened... can u help?`
`19:54.47`	`[TK]D-Fender`	`javb: I already gave you the guess-loist.`
`19:54.53`	`[TK]D-Fender`	`list*`
`19:56.26`	`*** join/#asterisk Holos (n=asdf@38.99.155.222)`
`19:57.00`	`javb`	`I undestand. But why AFTER REBOOTING EVERYTHING goes to normal... and it WAS working`
`19:57.11`	`Holos`	`Hi, Did something change for Queues? I'm unable to specify the time to ring the queue. I have Queue(queuename\|\|\|5) that should ring for 5 seconds right?`
`19:57.24`	`javb`	`i mean, is like a weird problem?`
`19:57.33`	`*** join/#asterisk fbffff (n=fbffff@c-24-12-65-14.hsd1.il.comcast.net)`
`19:57.54`	`[TK]D-Fender`	`Random OT HTML question : can someone point me to the tag or header info I need to do an immediate browser redirect to another URL?`
`19:58.05`	`Holos`	`[TK]D-Fender: Meta refresh`
`19:58.24`	`[TK]D-Fender`	`Holos: that limits the overall time in the queue, and I believe app_queue's parameters many have changed order.`
`19:58.43`	`Holos`	`[TK]D-Fender: <meta http-equiv="refresh" content="10;url=http://wikipedia.org">`
`19:58.56`	`Holos`	`[TK]D-Fender: How do I find the app_queue paramaters?`
`19:59.25`	`[TK]D-Fender`	`Holos: "show application queue"`
`19:59.34`	`[TK]D-Fender`	`Holos: in * CLI`
`20:00.17`	`Holos`	`[TK]D-Fender: Thanks...`
`20:02.44`	`[TK]D-Fender`	`Holos: np, and ditto.`
`20:05.27`	`*** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk)`
`20:08.33`	`*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)`
`20:12.11`	`Voicemeup`	`: anyrecomendation for bangladesh white..`
`20:12.21`	`*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)`
`20:12.38`	`*** join/#asterisk LeddyHM (n=NONE@polar.artica.net)`
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`20:15.54`	`*** join/#asterisk pfn (n=pfnguyen@hanhuy.com)`
`20:22.52`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`20:24.14`	`crimethinker`	`Las Vegas Kwik-E-Mart: +1 702 270 2964`
`20:24.57`	`crimethinker`	`call them and ask for Apu`
`20:44.45`	`Holos`	`Anyone have any experience with Red Fone Devices?`
`20:46.39`	`*** join/#asterisk kirberich (n=robert@i538705AF.versanet.de)`
`20:46.59`	`kirberich`	`good evening`
`20:47.49`	`Voicemeup`	`crimethinker ?`
`20:47.50`	`Voicemeup`	`http://www.cfnews13.com/News/Local/2007/7/2/orlando_7eleven_becomes_kwikemart.html`
`20:48.15`	`*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`20:50.03`	`kirberich`	`a question regarding isdn: i have a isdn line with two b channels. Is it possible to connect a call coming in on one channel to an external number using the other channel of the same card?`
`20:53.12`	`*** join/#asterisk madcap (i=madcap@unaffiliated/madcap)`
`20:56.50`	`Strom_M`	`kirberich: i don't see why it wouldn't be possible`
`20:56.59`	`crimethinker`	`Voicemeup, yes, there's something like 12 of them in north america`
`20:57.48`	`__DAW`	`kirberich: yes`
`20:57.58`	`kirberich`	`in that case, how is it possible ;)`
`20:58.15`	`kirberich`	`when i just try a Dial(CAPI/somenumber) it says the line is busy`
`20:58.24`	`kirberich`	`i guess i have to tell asterisk somehow to use the other channel, but how?`
`20:58.28`	`*** join/#asterisk DEac- (n=deac@Platin.DenKn.de)`
`21:00.08`	`Strom_M`	`kirberich: do regular outbound calls work using that method?`
`21:02.53`	`*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)`
`21:06.31`	`*** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca)`
`21:09.03`	`*** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net)`
`21:10.09`	*** join/#asterisk Tee_` (i=tee@stop.rooting.us)
`21:14.43`	`kirberich`	`Strom_M, woops ;) capi/contr1/number did the trick ;)`
`21:15.30`	`Strom_M`	`there ya go`
`21:17.20`	`*** join/#asterisk Holos (n=asdf@38.99.155.222)`
`21:17.49`	`Holos`	`Does anyone have the US/**** Location files for a 7961 phone?`
`21:18.08`	`*** join/#asterisk aao_pwner (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net)`
`21:21.33`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`21:22.15`	`Holos`	`The file I need is: US/g3-tones.xml`
`21:26.21`	`*** join/#asterisk ManxPower (n=manxpowe@015-833-972.area5.spcsdns.net)`
`21:32.59`	`*** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com)`
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`21:45.49`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`21:52.26`	`Hymie`	`anyone currently using la crappy unidens?`
`21:52.38`	`Hymie`	`with asterisk 1.4`
`21:54.38`	`shido6`	`sip phones?`
`21:55.45`	`*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca)`
`21:55.58`	`*** join/#asterisk jm\|laptop (n=jm\|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com)`
`21:56.56`	`*** join/#asterisk ManxPower (n=manxpowe@015-833-972.area5.spcsdns.net)`
`21:59.46`	`*** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net)`
`22:07.58`	`JT`	`kirberich: wow, one of the few people on the planet still using capi :P`
`22:08.48`	`kirberich`	`jt, in my situation, it's the most sensible option by far ;)`
`22:09.27`	`JT`	`what situation would that be?`
`22:10.18`	`kirberich`	`well, i have a router, i bought an isdn card for 20 euros, and now i have the biggest and baddest "answering machine" i can think of ;)`
`22:10.57`	`kirberich`	`it's just awesome what one can do with shitty hardware, a telephone line, asterisk and a little time ;)`
`22:13.05`	`JT`	`and you're saying capi is the only software under asterisk that supports your card?`
`22:15.21`	`*** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`22:16.56`	`*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)`
`22:17.19`	`JT`	`kirberich: ?`
`22:17.38`	`kirberich`	`sorry, i was gone`
`22:17.41`	`kirberich`	`well`
`22:17.54`	`kirberich`	`i didn't really figure there could be anything else that would support it`
`22:18.04`	`JT`	`what sort of card is it/`
`22:18.14`	`kirberich`	`it's an avm fritz!usb isdn card`
`22:18.35`	`kirberich`	`fritz!usb v2.0 that is`
`22:18.40`	`JT`	`i think that's hfc chipset`
`22:18.46`	`JT`	`bristuff probably supports it`
`22:19.04`	`k31th`	`Hum, i have an ITX system here I want to build a solid state pbx`
`22:19.22`	`JT`	`k31th: i bet it's something horrible like VIA chipset`
`22:19.30`	`k31th`	`JT: yep`
`22:19.30`	`kirberich`	`JT, what advantage would it offer to use that instead of capi?`
`22:19.47`	`JT`	`kirberich: being able to use all zaptel features`
`22:19.55`	`k31th`	`its a via board.`
`22:20.03`	`JT`	`accessing your channel as if it was any other zap channel`
`22:20.05`	`k31th`	`tbh this is only a test`
`22:20.27`	`JT`	`k31th: any hardware boards going on it?`
`22:20.31`	`k31th`	`im riht in thinking putting a r/w FS on a CF card based HD is bad news`
`22:20.41`	`k31th`	`JT: not for now.`
`22:20.50`	`k31th`	`just IAX trunks for now.`
`22:21.07`	`JT`	`i hope you mean normal iax connections, not trunks`
`22:22.34`	`k31th`	`http://www.icp-epia.co.uk/index.php?act=viewProd&productId=66`
`22:22.43`	`k31th`	`thats the ide disk i have`
`22:39.49`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`22:50.31`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`22:52.29`	`Aces1Up`	`anyone knoe where i download the asterisk sounds package?`
`22:52.42`	`JT`	`asterisk.org`
`22:52.43`	`[TK]D-Fender`	`Aces1Up, www.asterisk.org`
`22:53.50`	`romano2k`	`good evening everyone! i just launched asterisk on my debian with the initscript`
`22:54.23`	`romano2k`	`now i have an infinite loop of asterisk crashing and telling : "Asterisk ended with exit status 1, Asterisk died with code 1."`
`22:54.27`	`romano2k`	`what should I do ?`
`22:54.46`	`[TK]D-Fender`	`romano2k, stop running it as a daemon, and run it manually to see where it crashes`
`22:55.09`	`[TK]D-Fender`	`romano2k, Most common reason is Zaptel failure`
`22:55.31`	`romano2k`	`[TK]D-Fender, when i do "/etc/init.d/asterisk stop", it answers "failed to kill 10227: No such process"`
`22:55.55`	`[TK]D-Fender`	`romano2k, Do it the HARD way and just nuke the processes`
`22:55.55`	`JT`	`romano2k: check the process list for any script like safe_asterisk`
`22:56.14`	`romano2k`	`JT, [TK]D-Fender, right, i killed everyone :D`
`22:56.29`	`romano2k`	`should i try something like "asterisk -cvvv" then?`
`22:57.04`	`[TK]D-Fender`	`romano2k, "asterisk -gvvvvvvvvvc"`
`22:58.02`	`romano2k`	`Asterisk Ready.`
`22:58.02`	`romano2k`	`*CLI>`
`22:58.33`	`romano2k`	`and my X-Lite client logs and works properly`
`22:59.04`	`romano2k`	`I have a warning about sqlite3`
`23:00.56`	`[TK]D-Fender`	`romano2k, Probably ok to stop * and restart the service normally.`
`23:01.24`	`romano2k`	`[TK]D-Fender: now it works.`
`23:01.31`	`romano2k`	`i didn't change nothing o_O`
`23:04.49`	`romano2k`	`i also get this message on launch of asterisk : "WARNING[15102]: res_musiconhold.c:852 moh_register: Unable to open pseudo channel for timing... Sound may be choppy."`
`23:04.58`	`romano2k`	`and I wish I could use musiconhold`
`23:05.17`	`JT`	`romano2k: no zaptel?`
`23:06.43`	`romano2k`	`JT: i don't use it`
`23:06.56`	`romano2k`	`JT: I mean, I don't even really know what it is`
`23:07.12`	`romano2k`	`JT: i'm using asterisk with sip only`
`23:07.29`	`*** join/#asterisk saftsack (n=saftsack@pD9E07BB0.dip.t-dialin.net)`
`23:11.53`	`romano2k`	`any solution for my problem?`
`23:12.21`	`[TK]D-Fender`	`romano2k, install Zaptel and use Ztdummy for your timing source`
`23:12.37`	`romano2k`	`wow :)`
`23:12.40`	`JT`	`romano2k: what [TK]D-Fender said... you need zaptel for MoH`
`23:12.46`	`romano2k`	`okay!`
`23:12.52`	`romano2k`	`i'll try, thank you`
`23:15.06`	`*** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com)`
`23:15.14`	`DrukenLPY`	`afternoon everyone`
`23:15.49`	`[TK]D-Fender`	`DrukenLPY, considerably so.`
`23:16.02`	`DrukenLPY`	`almost evening..`
`23:16.05`	`blitzrage`	`Al Gore III got pulled over for doing 100mph in a Prius...`
`23:16.08`	`JT`	`lies, it's 0915!`
`23:16.09`	`blitzrage`	`is that even possible? :)`
`23:16.37`	`JT`	`al gore III?`
`23:16.39`	`DrukenLPY`	`even if possible... how safe would that be?`
`23:16.51`	`JT`	`it's only 100mph, how bad could it be`
`23:16.56`	`blitzrage`	`depends how well the car handles at that speed I suppose`
`23:17.00`	`DrukenLPY`	`it's a prius...`
`23:17.04`	`JT`	`so`
`23:17.12`	`blitzrage`	`I've done that speed in a 1990 Chevy Lumina`
`23:17.13`	`JT`	`they seem well built`
`23:17.31`	`JT`	`toyotas are generally of good built quality also`
`23:17.32`	`blitzrage`	`I'm surprised it has that high of a top end speed`
`23:17.39`	`blitzrage`	`but good to know`
`23:17.49`	`JT`	`160km/h is NOT fast`
`23:17.55`	`JT`	`any modern car can do that`
`23:17.58`	`blitzrage`	`Toyota should use that as a selling point. Damn yanks thinking hybrid cars are slow`
`23:18.14`	`blitzrage`	`JT: right.. but it's a hybrid. I'm not THAT shocked, but still`
`23:18.19`	`JT`	`why would a hybrid be slow, electric is the best for off the line torque`
`23:18.31`	`blitzrage`	`torque != top end speed`
`23:18.34`	`JT`	`if you want a torque monster, get an electric car.`
`23:18.42`	`blitzrage`	`at that speed, it'd probably be mostly the gas engine anyways`
`23:18.43`	`JT`	`the problem is storing the energy`
`23:18.47`	`JT`	`sure`
`23:19.07`	`rob0`	`How bad is his ticket going to be?`
`23:19.19`	`JT`	`the engine is 1.5L`
`23:19.27`	`DrukenLPY`	`anyone want to share their bell contact for a pri... or as they call it... "megalink"`
`23:19.37`	`JT`	`plenty of power to get to 160km/h in something that light`
`23:19.45`	`[TK]D-Fender`	`blitzrage, 160 kph isn't that bad... I hit 180 in my old T-bird.... at least... thats where the needle stopped ;)`
`23:19.47`	`JT`	`probably can do 200km/h`
`23:19.53`	`blitzrage`	`JT: seen this car? http://www.forbes.com/resourceful/2003/10/21/cx_dl_1021vow.html`
`23:20.20`	`JT`	`DrukenLPY: haha, telstra here used to call some services a "megalink"`
`23:20.24`	`blitzrage`	`rob0: probably bad, since he had marijuana, zantax, valium, and a couple other prescription drugs with no prescription`
`23:20.31`	`rob0`	`ouch!`
`23:20.39`	`blitzrage`	`[TK]D-Fender: ya, I've done that too`
`23:20.44`	`[TK]D-Fender`	`blitzrage, Neato... just fast ehough to generate the g-forces needed to hold back the vomit from acceleration ;)`
`23:21.17`	`Juggie`	`blitzrage, my old civic was caped @ 160`
`23:21.20`	`Juggie`	`i hit that on the 401 :)`
`23:21.26`	`JT`	`blitzrage: not bad`
`23:21.53`	`JT`	`is this the presidential al gore, or someone else`
`23:22.10`	`blitzrage`	`JT: vice-pres`
`23:22.10`	`rob0`	`Gore III would be the son of the ex-VP`
`23:22.13`	`blitzrage`	`his son`
`23:22.13`	`madcap`	`his son`
`23:22.17`	`JT`	`ah`
`23:22.41`	`madcap`	`At 2:15 am on July 4, 2007, Gore was arrested in Laguna Hills in Orange County, California for speeding over 100 MPH in a Toyota Prius.[10] Gore admitted to recently smoking marijuana and was found to be in possession of a small amount of marijuana along with Xanax, Valium, Vicodin, Adderall, and Soma. Police reported that Gore had no prescriptions for the pharmaceuticals`
`23:22.45`	`madcap`	`haha.`
`23:22.59`	`DrukenLPY`	`Juggie: my minivan can hit 160....`
`23:24.00`	`blitzrage`	`aiight, I'm out for a bit -- upgrading to FC7 on my laptop`
`23:24.13`	`*** part/#asterisk macli (n=macli@nmc.brc.ubc.ca)`
`23:25.03`	`Yomer`	`has anyone worked with linksys SPA400 ? im having problems with Caller ID`
`23:29.00`	`k31th`	`Is there a decent front end to asterisk? i need a front end for a user that wants one... is there any thing half decent around`
`23:30.11`	`JT`	`COPS: Gore Edition`
`23:31.02`	`*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)`
`23:31.27`	`JT`	`k31th: not really`
`23:31.40`	`k31th`	`JT: is there nothing that edits the configs directly ?`
`23:32.55`	`fujin`	`I'd hope not`
`23:32.58`	`fujin`	`a frontend would be silly`
`23:33.17`	`fujin`	`generally you don't need to edit your configuration much after crafting the initial dialplan, with well-written macros anyway`
`23:33.23`	`JT`	`there is, but they suck`
`23:34.25`	`fujin`	`how's the asterisk-gui project coming along?`
`23:34.33`	`fujin`	`to be completely honest I don't think i'd use it even if it was stable`
`23:40.05`	`k31th`	`asterisknow you mean fujin ?`
`23:40.12`	`JT`	`no`
`23:40.15`	`JT`	`that's the distro`
`23:40.24`	`JT`	`that contains asterisk-gui`
`23:44.41`	`*** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net)`
`23:45.29`	`*** join/#asterisk HaMYaI (i=HaMYaI@125-25-206-179.adsl.totbb.net)`
`23:45.35`	`*** join/#asterisk mightnare (n=mightnar@s230165.ppp.asahi-net.or.jp)`
`23:49.16`	`mightnare`	`in zapata.conf, if transfer=yes requires threewaycalling to be also enabled, how do i totally disable threewaycalling for all my users... explicitly specify in each of their entries?`
`23:52.19`	`[TK]D-Fender`	`mightnare, disable BOTH in zapata and force them to use DTMF transfer.`
`23:55.47`	`mightnare`	`i was hoping for a better option, still enable flash-hook transfer though disable threewaycalling totally`
`23:57.00`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`23:58.33`	`[TK]D-Fender`	`mightnare, another great reason to NEVER use Zaptel FXS.`

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