IRC log for #asterisk on 20080907

00:03.31[TK]D-Fenderjplank>I could just tell my * that **2003 and 2003 is the same thing, can't I <- no, because they aren't the same.  What you DO with ti however is up to you
00:03.46idohttp://blogtech.oc9.com/index.php?option=com_content&view=article&catid=4:asterisk&id=77:20071121ast&Itemid=6
00:03.59*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
00:05.46jplankbut I have to be able to let * know that calls that are refered to **2003 should goto 2003, right?
00:07.08jplankis this correct
00:07.10jplank"When * gets a sip REFER message, it looks through the context and dial  plan."
00:08.02ManxPowerjplank: maybe it does the same thing on a TRANSFER, as a refer could be a transfer
00:08.25jplankthe aastra uses refer for transfers
00:08.42ManxPowerbut you have REFER so plastered into your brain you can't even conceive of a different way.
00:09.14ManxPowerso, how exactly would a REFER happen from a Zap channel or a H323 channel transfer?
00:09.37ManxPowerAsterisk does not, for the most part, give you access to the lower level protocols, with the exception of being able to get and set SIP HEADERS.
00:09.42jplankI have refer plastered in my brain because thats what the aastra is sending after it creates a new call to transfer calls
00:10.08ManxPowerjplank: Maybe you should look in channelvariables.txt
00:10.26ManxPowerpay SPECIAL attention to the TRANSFER variables
00:12.14ManxPowerblindtransfer, bridgepeer, maybe even dnid variables may contain the info you need.
00:12.33jplankI'm going to check them all
00:12.40jplankwell
00:14.20ManxPowerjplank: you will have more luck if you look on voip-info.org
00:14.28jplankI am actually
00:14.31ManxPowerfor things like asterisk, blf, call parking
00:16.29ManxPowerexten => 701,hint,park:701@parkedcalls
00:16.42ManxPowerThat's only for 1.4
00:17.00*** join/#asterisk dacs (n=haiger@unaffiliated/dacs)
00:20.15jplankerrr transfer() is only for issuing a SIP REFER not readying one
00:21.17ManxPowerI don't believe I ever mentioned transfer() (app_transfer)
00:22.13ManxPowerthere are at least three kinds of transfers in Asterisk.  I was referring to DTMF transfer and phone key transfers, not IVR transfers (like transfer())
00:24.32[TK]D-Fenderyou can detyermine a SIP blind transfer, but not an ATTENDED transfer
00:24.48jplankI don't think we are on the same page with the problem
00:25.03[TK]D-Fenderonly half a job, and then you have to put the detection code INTO all of your extens
00:25.49ManxPowerjplank: on polycoms you just press the BLF and it's just like it dialed whatever was configured in that BLF (in our case extensions)
00:25.51*** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net)
00:26.15ManxPowerpress the BLF labeled "Marty" and it's just like you dialed 3120 (marty's extension)
00:26.20jplankwith the aastra phone, you can setup a BLF/Xfer key. I have to set the value of the key to **+extension to be able to pick up ringing calls, but the problem is when I want to use the button to transfer, it sends a SIP REFER to **2003, not 2003
00:27.18[TK]D-Fenderjplank: you miss the point.  that button regers to ONE value.  I will not send "A" in one case, and "B" in another
00:27.25[TK]D-Fenderrefes
00:27.27jplankI understand that
00:27.28[TK]D-Fenderrefers
00:27.37jplankI'm trying to get the asterisk to know the difference
00:27.52ManxPowerjplank: why don't you just have it send 2003 and handle it inside the extension as what to do.
00:28.03[TK]D-Fenderjplank: there isn't a difference.  It has one value, and that is what it sends
00:28.10[TK]D-FenderManxPower: FreePBX <-
00:28.25jplankif it sends 2003, then how can I get call pickup to work?
00:28.43[TK]D-FenderManxPower: He's hoping to get away with a BLF key performing both pickupt AND normal dialing
00:28.44ManxPower[TK]D-Fender: I do not speak of FreePBX.  It is up to the user to translate from real Asterisk to that painted GUI fop of a PBX.
00:29.04jplankfender: yes
00:29.10ManxPower[TK]D-Fender: it's funny to watch him do it.  Did you already tell him that can't be done?
00:29.18jplankno
00:29.26jplankwell, it works as a speed dial
00:29.29jplankit works for call pickup
00:29.33ManxPowerof course it does.
00:29.36[TK]D-Fenderjplank: it will dial *1* exten regardless of your intent.  IN that exten it is YOUR job to check to see if it should be doing a "pickup" or not.
00:29.52jplankI totally understand that
00:30.03[TK]D-Fenderjplank: fGood, so get to it.
00:30.15ManxPowerI'd configure one button the "**" and then configure the individual BLFs to send the extenstion
00:30.15jplankright
00:30.27jplankManxPower: thats how I originally had it
00:30.31jplankjust trying to get fancy
00:30.36*** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk)
00:30.47ManxPowerthen I'll just put you on /ignore for wasting my time.
00:30.55jplankno
00:31.15jplankif your telling me theres no way to compensate for **2003 inside the SIP REFER thats one thing
00:31.34ManxPowerjplank: there isn't.  You might be able to do it in the dialplan, as [TK]D-Fender has already said.
00:31.50ManxPowerif you really want to work with the low level SIP protocol then you want something like OpenSER
00:31.54jplankit doesn't seem like the call makes it to the dialplan
00:32.01jplankyea, your right about that
00:32.01[TK]D-Fenderjplank: how does * know you magically mean to send them somewhere else?
00:32.10ManxPowerjplank: it does, you just may not see it.
00:32.25jplankit doesn't, I'm trying to figure out how to tell asterisk it should go somewhere else
00:32.27ManxPowerturn of sip debug and you'll see the call make it to Asterisk
00:32.39[TK]D-Fenderjplank: you don't get to have it both ways.  it goes to 1 exten period.  that is its job.  go fix your extens to try to be smarter
00:32.42ManxPowerit will be rejected with a 404 if the destination does not exist.
00:33.58ManxPowerjplank: in the 7 years I've been using Asterisk I've never heard of someone wanting direct access to the REFER packet when using BLF.  That should tell you something.  It's the wrong approach.
00:34.05jplankand I assume that telling the asterisk **2003 destination exists as 2003 would screw up the call pickup
00:34.22ManxPowerjplank: try it and SEE.
00:34.34jplankManxPower: now that you tell me that, I know
00:34.45[TK]D-Fenderjplank: it will hit whatever matches **2003 in the dialplan and then proceed to do whatever you tell it.
00:35.34jplankbut if that was the case, wouldn't it match exten => _**.,n,Set(REALEXT=${EXTEN:2})
00:36.06[TK]D-Fenderjplank: that ppattern matches (assuming your have a priority 1 for it, and its in the proper cont4ext, etc)
00:36.44jplankbut thats the issue isn't it, there is no proper context for **2003
00:36.46jplankI get it now
00:37.04jplankI think I do at least
00:37.45jplankif I created a new extension in sip.conf **2003 and put it in xyz context and then matched in there, it *should* work
00:38.10[TK]D-Fenderjplank: a SIP DEVICE is not an EXTENSION
00:38.23jplanki'm sorry thats what I meant
00:39.02[TK]D-Fenderjplank: and this has nothing to do with your SIP devices.
00:39.15jplankerr
00:39.19jplankI'm off then
00:39.32jplankisn't that what * matches the SIP REFER to?
00:39.39[TK]D-Fenderjplank: No, it isn't
00:39.49[TK]D-Fenderjplank: SIP refer targets an EXTENSION.
00:40.01[TK]D-Fenderjplank: just like absolutely every other call you make
00:40.45[TK]D-Fenderjplank: * does not magically match up what you dial as referring from one device directly to the other   Everything dial = DIALPLAN.
00:43.49jplankthen shouldnt a simple exten => **2003,1,dial(SIP/2003) work?
00:44.09jblackjplank: Yes. I told you something much like that long ago. :)
00:44.14jplankI tried it
00:44.15jplankit doesn't
00:44.20jplankI still get a 404
00:44.26[TK]D-Fenderjplank: And you didn't show us e the failure
00:44.31jblackYou may not be in the extension that you think you are.
00:44.44jblackPardon, in the context that you think you are
00:44.48jplankhmmm
00:44.51[TK]D-Fenderjplank: if it failed then its because its not in the context being used by the device calling for it
00:45.08jblackEcho!
00:45.11jplankI think I'm back at the freepbx problem again, huh?
00:45.27jblackI wouldn't know anything about freepbx
00:45.40jayteelol
00:45.50*** join/#asterisk luca`gervasi (n=Ashutto@host76-170-dynamic.21-87-r.retail.telecomitalia.it)
00:45.52jblackI honestly don't.
00:45.53luca`gervasiHello
00:45.54[TK]D-Fenderjplank: No, if its not in the right context, you should know where it belongs and should have put it there
00:45.57jayteeso..... who's on first?
00:46.33jblackjaytee: Start up asterisk -r, get debug and verbose running at a high level, and trace a call. That should tell you what context you're going to, and tell you how to adjust things. Ok?
00:47.23jayteeum, jblack, if I had the link to the rabbit with the pancake on it's head I'd send it to you! :-)
00:47.33luca`gervasii have throubles, i'm a newbie...i tryed a simple configuration with one friend phone, but it says registration error. I tryed the console whith core set verbose 999999999999999 and core set debug 99999999999999 and sip debug ip <phone_ip> but it says nothing about my attempt to connect... is there away to enable mooore debug?
00:47.46[TK]D-Fenderjplank: jblack thats jplank you should be reffering to :)
00:48.15[TK]D-Fenderluca`gervasi: blarg!
00:48.22[TK]D-Fenderlucstrike that..
00:48.28[TK]D-Fendercan't type... missing thumb...
00:48.30luca`gervasi[TK]D-Fender ??? whay is it?
00:48.39luca`gervasiok :D
00:48.54[TK]D-Fenderluca`gervasi: started answering you and wrote a message for someone else by accident
00:48.57jplankthanks guys for your help (and i'll try that jblack ) got to go though, dinner just came and girlfriend is freaking I've been on my computer all day
00:49.20luca`gervasi[TK]D-Fender, got it :D anyway....stop to me if you can :D
00:49.23jayteewhat a trooper, he injures himself and yet he's still in here slugging it out trying to help everyone.
00:49.44[TK]D-Fenderluca`gervasi: pastebin the error you get when you try to register.
00:49.45[TK]D-Fender~pb
00:49.46jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:49.47[TK]D-Fender^^^^^^^^^^
00:50.12[TK]D-Fenderjaytee: I'd be off playing pool already were it not for this.  My week is &^%ed now.
00:50.20luca`gervasi[TK]D-Fender, simply a registration error on my local Ekiga and Ip301... no messages at all in asterisk...
00:50.33*** join/#asterisk geoff2010 (i=geoff201@181.sub-75-250-58.myvzw.com)
00:50.35jaytee[TK]D-Fender, yeah, that's gonna suck for a while.
00:50.43jayteeI haven't played pool in ages.
00:50.50jayteeused to be quite good at it.
00:51.16jblackwelcome
00:51.24[TK]D-Fenderluca`gervasi: do "sip debug" and look for the packerts.  if you see NOTHING then you're either firewalled, or your phones are configured wrong, or * filed to bind, etc.
00:51.30jayteemy buddy had a nice full size regulation table in his rec room, we used to play most every nite.
00:51.51[TK]D-Fenderjaytee: I'm told I'd be a strong AA in BSA ratings
00:52.12jayteeyou ought to compete then and pick up some extra cabbage :-)
00:52.15[TK]D-Fenderjaytee: Whatever that is supposed to tell me :)
00:52.30*** part/#asterisk geoff2010 (i=geoff201@181.sub-75-250-58.myvzw.com)
00:52.34luca`gervasi[TK]D-Fender, trying right now
00:53.25luca`gervasiuhm... nothing...
00:53.32jaytee[TK]D-Fender, hey I saw a Minolta digital SLR today for 25 bucks in a pawn shop with a 35 to 70 autofocus zoom on it but I can't remember the body model now.
00:54.17[TK]D-Fenderjaytee: sounds like an HTSi
00:54.30[TK]D-Fenderluca`gervasi: disable your firewall and test again.
00:54.31luca`gervasi...i have no firewall at all, and the phones connects to asterisk with only a switch between... netstat -lnp shows asterisk listening on all interfaces 5060
00:54.54luca`gervasiis there some kind of acl somewere in the config files?
00:55.11[TK]D-Fenderluca`gervasi: if you see not packets with "sip debug" then your phones aren't configured properly
00:55.32[TK]D-Fenderluca`gervasi: ACL would only apply if you were even getting traffic, which you aren't
00:55.34luca`gervasihell...i can't be wrong on two devices :(
00:57.26[TK]D-Fenderluca`gervasi: something is very wrong.
00:57.48[TK]D-Fenderluca`gervasi: go prove your firewall, and veryify that * is the one that bound 5060 UDP
00:57.48luca`gervasiyou can say it :D...sigh sigh... but... what? :D
01:03.47*** join/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com)
01:05.12drdrainDoes Dundi need a hole through the firewall to work?  If so what port number?
01:07.29[TK]D-Fenderdrdrain: http://www.google.ca/search?hl=en&q=dundi+port+firewall&btnG=Google+Search&meta=
01:07.53*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
01:16.11jeevfender
01:16.13jeevit's google.com
01:18.58*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
01:19.14[TK]D-Fenderjeev: yes, and the USA is the center of the universe, and you all speak American, and enjoy American cheese on your hamburgers.
01:19.26jblack100% american beef.
01:19.29jeevfender, i'm not american by choice!
01:20.09[TK]D-Fenderjeev: Have you chosen to move out?
01:20.12jblackThe US does get some things right. Like double bacon cheeseburgers.
01:20.19jeevno, i have too many family and friends here!
01:20.30[TK]D-Fenderjeev: then you've made your choice :)
01:20.33jeevjblack, that's why asterisk developers are overweight ;)
01:20.40jeevhar fender.
01:20.54jeevi'm not american, i can't leave my friends and family, especially my parents behind
01:20.59jeevi know white people do that ALL the time
01:21.03jblackFat people are insurance against societal collapse.
01:21.04jeev"i've grown up, time to say fuck you and bounce"
01:21.09jeevlol jblack
01:21.09jeevhow so?
01:21.48jblackWhen shipping fails, it'll be up to people like me to outlive the skinny, thereby having access to the meager remaining resources.
01:21.56jeevhahahaha
01:22.02jeevnice
01:22.06jeevhow fat are you
01:22.11jblackJust 240.
01:22.21jeev4'9 ?
01:22.36jblack5'10. And the bulk of it is actually bone, because of a weird disease I have.
01:22.38jeevi can't believe my ex was 4'11
01:22.40jblackOhh! Neighbor fight!
01:22.41jeevwtf was wrong with me
01:22.50jeevahh, ok, just cause you said disease, i'll ok it.
01:23.02[TK]D-Fenderjeev: midget pr0n
01:23.14*** part/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com)
01:23.23jeevi was like what
01:23.31jeev14 inches taller than her
01:23.31jeevheh
01:23.40jeevi was 5 asian penises taller than her
01:26.07jeevi'm sl33py
01:26.18jblackGee. Thanks for the approval. ;)
01:26.26jeevhaha
01:26.35jeevdood, im getting a lot of udp shit to an office box
01:26.36jeevwtf
01:26.49jeevis there some new exploit or worm or what
01:28.46YournameHi guys, sometimes an agent is logged into a queue via SIP, and his internet connection drops. The agent is logged in for several minutes afterwards. And so when the agent comes back online in 2-3 mins, he cant log back on cuz Asterisk thinks he's still logged on
01:29.27*** join/#asterisk geoff2010 (i=geoff201@181.sub-75-250-58.myvzw.com)
01:48.17*** part/#asterisk geoff2010 (i=geoff201@181.sub-75-250-58.myvzw.com)
01:50.00*** join/#asterisk Putzz (i=Putzz@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com)
01:50.48Putzzhey guys im trying to install a tdm400 never had problems before, but having problems now lspci outputs: Ethernet controller: Digium, Inc.: Unknown device 8005 (rev 11)  but not channels are showing
01:51.03Putzzit has 4 fxo modules
01:51.10*** join/#asterisk hfb (n=hfb@cpe-76-87-161-213.socal.res.rr.com)
01:52.22[TK]D-FenderPutzz: You've installed zaptel, modprobe's the drive do your card?
01:52.30Putzzyes sir
01:52.35[TK]D-FenderPutzz: If so pastebing "cat /proc/interrupts"
01:52.39[TK]D-Fender~pb
01:52.39jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
01:53.33Putzzhttp://www.pastebin.ca/1196086
01:57.41Putzzdoes it look good or bad?
01:59.53*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
02:03.43Putzz?
02:08.10*** part/#asterisk Sweeper (n=sweeper@74.51.109.60)
02:10.23*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
02:13.35[TK]D-FenderPutzz: "modprobe wctdm" and try again
02:15.59Putzzsame output
02:16.48*** join/#asterisk purplev45 (n=archway@71-91-227-114.static.stls.mo.charter.com)
02:17.06[TK]D-FenderPutzz: try another slot.  If it persists, call Digium support
02:17.44Putzzok thanks appreciated
02:18.19purplev45having an IAX2 problem I'm hoping somebody can point me in the right direction on.  Got a 4 port FXS card which (zaptel) which configures fine;  I get a dial tone, but when I try to place a call it automatically hangs up.
02:18.42purplev45Using VOIPJet with my test account configured as they recommended.
02:19.11purplev45Debug shows:
02:19.28purplev45    -- Starting simple switch on 'Zap/1-1'
02:19.28purplev45<PROTECTED>
02:22.26[TK]D-Fenderpurplev45: Zap has nothing to do with IAX2.
02:22.50[TK]D-Fenderpurplev45: pastebin your extensions.conf & zapata.conf
02:24.19purplev45extensions is default save the following:
02:24.38purplev45exten => _1NXXNXXXXXX,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ
02:24.38purplev45exten => _1NXXNXXXXXX,2,Dial,IAX2/<userid>@voipjet/${EXTEN} ; VoipJet.com NANPA
02:24.38purplev45exten => _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ.
02:24.39purplev45exten => _011.,2,Dial,IAX2/<userid>@voipjet/${EXTEN} ; VoipJet.com WORLD
02:24.58purplev45<userid> is set to my actual user id.
02:25.44purplev45zapata.conf is:
02:26.00purplev45threewaycalling=yes
02:26.00purplev45transfer=yes
02:26.00purplev45canpark=yes
02:26.00purplev45cancallforward=yes
02:26.00purplev45callreturn=yes
02:26.01purplev45echocancel=yes
02:26.03purplev45echocancelwhenbridged=yes
02:26.05purplev45rxgain=0.0
02:26.07purplev45txgain=0.0
02:26.09purplev45group=1
02:26.11purplev45callgroup=1
02:26.13purplev45pickupgroup=1
02:26.15*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
02:26.15purplev45immediate=no
02:26.30Qwellblinks
02:26.54Qwellthere goes the last 45 minutes of chat off my screen
02:28.22purplev45I believe my zapata.conf is default as well.
02:29.56[TK]D-Fenderpurplev45: PASTEBIN
02:29.58[TK]D-Fender~pb
02:29.59jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
02:30.31[TK]D-Fenderpurplev45: and your zapata.conf is incomplete.  There is not channel declaration line
02:30.58[TK]D-Fenderpurplev45: now pastebin them BOTH COMPLETELY please
02:32.30purplev45sorry, been 15 years since I've IRC'd; I'm a bit rusty, I apologize to the group.
02:32.35purplev45looking up pastbin.
02:34.31*** join/#asterisk BugKhaM (n=BugKhaM@125.25.137.191.adsl.dynamic.totbb.net)
02:34.35Wayhighsweet.. finally got enum working really well
02:34.59WayhighMy wife had been pissed that our 800#'s were going out enum and one of them was a bad network connection
02:35.06BugKhaMhow to uninstall zaptel+asterisk 1.4 and install 1.2.x instead?
02:35.18tristanbobwhat are some good free Sip to Sip services?
02:35.21BugKhaMis there a "make uninstall"?
02:35.30Wayhightristanbob: voxalot and sipbroker
02:35.35Wayhighthey link most of them together
02:35.36tristanbobWayhigh, thanks
02:35.51Wayhighno problem.. gizmo5's awesome too
02:35.56WayhighI'm setup to use all of them :)
02:36.01purplev45TK: http://pastebin.com/m17250c7c (zapata.conf)
02:36.18*** join/#asterisk kamanashisroy (n=kamanash@119.30.34.1)
02:36.30Wayhighhey.. anyone have any ideas why I'd hear crosstalk on my tdm400?
02:36.39WayhighIt's picking up a TV somewhere in the house
02:36.57WayhighI noticed it while doing a zapbarge with the line onhook
02:37.14QwellWayhigh: sure it's your house?
02:37.18purplev45TK: http://pastebin.com/m1bb769f7 extentions.conf
02:38.19Wayhighqwell: not positive.. but it sounds like a TV.. it's hard to hear ya know
02:38.31Wayhighqwell: any other ideas?
02:38.37BugKhaMFedora 9 seems to have problems with sip configuration. For example, "autodomain" doesn't work in F9+asterisk 1.4 anymore
02:38.45QwellWayhigh: if it's your house...better wiring
02:39.10Qwellif it's not...  call the telco
02:39.21Wayhighqwell: it could be the other zap/4 wiring I guess..
02:39.23Qwellare you in an apt?
02:39.28Wayhighnaw it's a house
02:39.40Wayhighthat same line is the one that runs all over the house
02:39.45Qwellsure it's a TV and now just somebody elses conversation?
02:39.51Qwellnot*
02:40.00Wayhighqwell: it's either a tv or aradio
02:40.03Qwelleasy test - turn them off
02:40.12WayhighI've had other things pick up the cross talk before too.. like sonic headphones
02:40.30jeevcat /dev/urandom > /dev/irc/asterisk
02:40.31jeevwoops, sorry!
02:40.49Qwelljeev: /exec cat /dev/urandom
02:40.51Qwelltry it
02:41.03jeevuh huh
02:41.06Wayhighe/xec pwnage jeev
02:41.11QwellI won't kick you
02:41.15jeevmirc doesn't have exec!
02:41.18Qwellnewb
02:41.23jeevnewb?
02:41.24jeevpfft
02:41.34jeevrip your head off and mv it to urandom
02:42.42Wayhighso what you're saying is that moving to urandom is roughly equal to deficating down someone's throat?
02:44.36purplev45TK: Figured it out
02:44.56purplev45you remark pointed me in the right dir.
02:45.22purplev45I had the context set to internal for my Card.
02:50.35[TK]D-Fenderyup
02:52.39Wayhighsup fender?
02:53.10*** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25)
02:53.24tristanbobWayhigh, favorite free softphone for call-centers?  zoiper?  3cx?  xlite?
02:54.52Wayhighhmm.. for callcenters? I'd say probably xlite as that's the most popular everywhere.. zoiper is good too but I've always found xlite easier to use
02:55.39Wayhighfor a call center I'd probably just get a bunch of PAP2T's or some other ATA.. I'm not a huge fan of softphones
02:56.04Wayhighhell.. go on freecycle and ask for vonage adapters
02:56.07Alton2Yes, I was wondering if any old cheap phone would be better than a osftphone.
02:56.11WayhighYou should get a few free ones you can unlock
02:56.43WayhighAlton: that $8 silver phone with the callerid display on the handset that walmart sells works REALLY well
02:57.07Alton2I guess I meant IP phones.
02:57.28[TK]D-FenderAlton2: And hard-phone would be better than a soft-phone
02:57.30[TK]D-Fenderany*
02:57.32Alton2I have Budge-Tone 100s here, don't laugh, just for home use.  They need rebooting from time to time but are very cheap.
02:57.43[TK]D-Fender~gs
02:57.44jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
02:58.04Wayhighfender: aren't the higher end GS phones better?
02:58.17Wayhighbtw.. if anyone here wants a bunch of Snom 300's.. I know where to get some way cheap
02:59.12Wayhighin my opinion, whatever you do, you should stay away from x100p.com's products
02:59.54Wayhightheir ata is crackly and tinny.. and doesn't have a real fxo.. just a passthrough
03:00.15[TK]D-FenderWayhigh: High end crap is better than lowend crap, yet still crap
03:01.11Wayhigh~cheap
03:01.11jbotwell, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
03:01.19[TK]D-FenderAlton2: For where you are Polycom IP320/330 depending on your wiring
03:01.40Alton2Price?
03:02.19*** join/#asterisk sacitec (n=tobi@201.166.16.254)
03:02.47sacitechello, anyone working with SIP client on an iphone to work with asterisk ?
03:06.21tristanbobWayhigh, these people currently use avaya phones with headsets
03:06.44tristanbobso they are already used to the headset thing, and they will be using the Switchvox switchboard
03:06.47*** join/#asterisk nicoAMG (n=superunk@201.203.50.42)
03:07.43[TK]D-FenderAlton2: www.telephonydepot.com
03:09.23*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
03:10.11jeevfender lovesssssssssssssssssss grandstream
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03:25.02*** part/#asterisk purplev45 (n=archway@71-91-227-114.static.stls.mo.charter.com)
03:25.16jblackSomebody should break into the grandstream office and take photos of their phones. I bet they'd be polycoms
03:34.22drmessanoPAP2s
03:46.27*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
03:50.17jeevgreat
03:50.19jeevjaytee is here
03:50.30jayteewhoop-dee-doo
03:50.36mchouWayhigh: what's this $8 Walmart phone you're referring to?  Is it only good for callerID or?
03:50.43jayteebeen coding my IVR, god what PITA
03:58.46sacitechello, anyone using AsteriskC2D as VoIP client for Iphone ?
04:00.12Wayhighmchou: it's a land line..
04:00.51Wayhighwhat's the general thought here on aastra's?
04:01.22[TK]D-FenderWayhigh: Only if you need teh CT DECT or massive amounts of presence on 1 phone
04:04.12YournameWill someone be kind enough to help me get call forwarding operational? Like *22<phonenumber> will enable call forwarding, and *23 will disable it? Asterisk 1.4.21. Thanks.
04:05.00[TK]D-FenderYourname: "core show function DB" <-
04:06.10*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
04:06.28Yourname[TK]D-Fender: That's helpful enough, but the code can't be different for the most part. Is there anywhere online that you could point me to please?
04:06.58*** join/#asterisk Thorn (n=thorn@unaffiliated/thorn)
04:07.07[TK]D-FenderYourname: www.freepbx.org
04:12.51*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552046.dsl.bell.ca)
04:13.42Yourname[TK]D-Fender: You mean dig in their confs? lol
04:13.57[TK]D-FenderYourname: why not.
04:14.07Yournameheh
04:14.10YournameI guess so.
04:14.22YournameHowever, they probably have it mangled in their macros/gui code, etc.  :S
04:14.48Yourname[TK]D-Fender: So that might not work out. If it was two contexts kind of a thing, then maybe.. but I'm sure it's intertwined.
04:16.13[TK]D-FenderYourname: For someone who is "thinking" your dedication to learning how to use a single silly dialplan function to do the job is quite telling
04:16.57*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
04:17.30Yournamelol
04:17.43YournameI'd be a bad cop. I'd shoot first, talk later.
04:25.24*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
04:28.32*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
04:43.06jblackWhere's drmessano. I want to see what he thinks about that microsoft commercial
04:43.46*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
05:00.02obnauticusanyone here know what form factor motherboard the IBM eServer x series run?
05:06.32[TK]D-Fenderobnauticus: I'm sure it varies by model, and are each proprietary
05:14.52jayteeI think they're all PowerPC architecture too
05:16.20JTerr what?
05:16.28JTiSeries is PPc
05:16.42JTxSeries is x86
05:16.57jayteereally? hmmm, didn't know that.
05:17.25JTand yeah i think the "form factor" would probably be proprietary
05:17.54jayteemost likely
05:18.09jayteeat least it isn't MicroChannel
05:30.17*** join/#asterisk nr4q (i=Ritalin@c-76-123-225-55.hsd1.tn.comcast.net)
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05:34.03outsider12hello - does anyone have the sip firmware for cisco 7975g?
05:45.12*** join/#asterisk mahlon (i=mahlon@martini.nu)
05:52.33jblacksurely cisco does?
05:53.07mchouIs the "local time" encoded in the incoming callerID?  It seems some phones adjust their local clocks in this fashion.
05:53.34mchoufor any skew, that is
05:53.43*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
05:56.35outsider12yea, but their sip images are on lock-down - i've been at it for days
05:57.21[TK]D-Fenderok, checkout time, later all
06:27.26*** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt)
06:27.27*** part/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt)
06:27.33*** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt)
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06:33.27*** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl)
06:34.40*** join/#asterisk Ast-M (n=chatzill@156.162.187.81.in-addr.arpa)
06:37.32Ast-MAnyone interested in a interesting problem?  I have two asteisk servers in two sites, I need to answer reception calls in both sites and transfer, I have dialplan in place and both systems including hints seem transparent. Problem is Trombone effect if calls are answered and then transfered on remote switch.
06:38.06Ast-MThinks everyone is probably asleep, maybe I should be.
06:39.21drmessanowhich microsoft commercial?
06:48.09*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
07:14.35rue_mohrAst-M,
07:14.37rue_mohrhu?
07:14.42rue_mohrecho problem?
07:15.02Ast-Mrue_mohr: Not echo, problem with too many calls
07:15.35rue_mohrhm still dont get you
07:15.44Ast-Meg:   Call comes in to switch 1 via iax, the queue rings 2 sip extensions and also an IAX trunk to switch 2
07:16.10rue_mohrk
07:16.16Ast-Muse on switch 2 answers call then transferes call back to switch 1.
07:16.40rue_mohrhmm I see, the molten cheese problem
07:17.03Ast-MSo I end up with 3 calls on ADSL link,   the incomming IAX trunk, the connection to the other switch at the other site and back again.
07:17.22rue_mohrthinks
07:17.54Ast-MI think asterisk could know and maybe already copes with this.
07:17.59Ast-MNot sure though.
07:18.12Ast-MIts a bit like a reinvite in SIP I think.
07:20.43rue_mohrI'm thinking it should
07:21.04rue_mohrI dont see why it would transfer a call back to the origionating source
07:21.14rue_mohrwhat you need here is automagic
07:21.24rue_mohroh did I mention I'm not a pro at this?
07:21.54Ast-Mnore me, I thought I knew some stuff but then there is always something ;)
07:22.21Ast-MI guess I could use the hope its not too many calls at that same time and not worry option, but I would like to do it as well as I can.
07:22.22rue_mohrmy only suggestion might be to send the call diectly to the other phone via a tunnel instead of via the second asterisk machine
07:22.38Ast-MHere is a little more detail...
07:23.52Ast-MCall arrives at switch 1 via IAX from ITSP,   rings a queue that contains some local sip extensions but also some extensions on a seperate asterisk systems at a different site bia IAX
07:25.19Ast-MIf a receptionist at the other site answers ( almost half the calls )  then this is fine if the call is for someone at that site ( 250-299)  but if its for a user at original site (200-249) this will get transfered back via iax to switch one.
07:26.49Ast-MThe basic idea is that there are two seperate switches but there is one "virtual" reception of 4 people these are spread across two sites.
07:27.30Ast-MI thought of registering the receptions phones on both servers
07:29.55rue_mohrI'm not too up on native bridges
07:30.07rue_mohrI know asterisk likes them
07:30.17Ast-Mthanks for thinking about it :)
07:32.16*** join/#asterisk HellBound (i=hellboun@should.have.tried.shellium.org)
07:32.26HellBoundHello
07:32.49HellBoundI wan to setup private voip server is that possible if yes then how :D
07:33.28Ast-MHelloBound,  maybe you need to give us a little more detail?
07:33.35HellBoundok
07:34.08rue_mohrget a computer and install asterisk
07:34.36HellBoundAst-M Im running windows XP so i wan to setup a voip server for private use to make long disrance calls for free :))
07:34.55rue_mohrget a computer and install linux, and asterisk
07:34.56HellBoundrue_mohr ok im chcking asterisk :)
07:35.07Ast-MHave to say, Linux
07:35.24HellBoundasterisk need linux :(
07:35.27rue_mohrdont try to run asterisk on windows
07:35.52HellBoundok
07:36.13HellBoundAsterisk will run on Ubuntu 7.10 gusty gibbon
07:36.14Ast-Mmaybe try a virtual machine, vmware?
07:36.27rue_mohrwindows cannot handle the interrupt rates or the data rates
07:36.39HellBoundok Ast-M :)
07:36.49rue_mohrno, not under vmware
07:36.56rue_mohrthat would be worse than windows
07:37.06HellBoundthen
07:37.14rue_mohra 1Ghz machine is just fine
07:37.18Ast-Mmaybe .
07:37.24HellBoundmy one in 3 GHZ
07:37.26rue_mohrsay even 512M of ram
07:37.37HellBound1 GB Ram my one
07:37.46rue_mohrbut not running windows or vmware
07:37.49Ast-MI would always go for a nice clean linux install and then build asterisk from source,  do you need any cards?
07:37.59Ast-Mor is it all sip/iax?
07:38.15*** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk)
07:38.49Ast-Mchan_ooh323.c:3390: error: expected â)â before string constant   Hmmph
07:38.50HellBoundbro tell me i can run Asterisk on Ubuntu or not pls
07:39.07Ast-Myep,  apt-get install asterisk
07:39.16HellBoundok
07:39.18HellBound:)
07:39.27Ast-Mbut its is worth building the source and its is easier than you thing.
07:39.58rue_mohrasterisk on ubuntu would be fine, but make it a good machine so the gui load dosn't scew it up
07:41.11HellBoundok
07:41.34HellBoundcan i make free pc-phone calls using Asterisk
07:41.35HellBound:D
07:42.46Ast-Mummm, asterisk is the switch you really need a client like a soft phone, but there is a console mode probably overkill if thats all you need.
07:43.48*** join/#asterisk slider750 (n=Slider@ip68-96-75-158.oc.oc.cox.net)
07:43.54HellBoundis there anyway to set up a server where users can call anywhere for free
07:44.18HellBoundi wan to run the server and wan to open it for public for free of cost :D
07:44.23*** join/#asterisk slider750 (n=Slider@ip68-96-75-158.oc.oc.cox.net)
07:44.24rue_mohryou cant make calls to the normal phone net work without a bridge to it, companies charge for those bridges
07:44.35rue_mohrbut you can make sip to sip calls free
07:44.44HellBound:(
07:44.46HellBoundoh
07:44.56rue_mohrbut skype is free too
07:45.06HellBoundyup only pc to pc
07:45.09HellBoundnot pc to phone
07:45.18rue_mohrpc to phone need a bridge
07:45.26rue_mohryours or via a service provider
07:45.45HellBoundis there any free bridge lol
07:45.55rue_mohrnot that I know of
07:45.59Ast-MHellBound: I think you think that SIP is magic and can make free phone calls? without someone paying?
07:46.32HellBoundok thas why i wan to run a server where every one can make free calls anywhere
07:46.33rue_mohrAst-M, well actaully, my system has an aix link that does that
07:46.39HellBoundis that possible
07:46.56rue_mohrbit I have all my phones at the hosue on a channelbank, with my landline... so
07:47.05Maliutaif it goes onto the PSTN _someone_ has to pay
07:47.20rue_mohralot too
07:47.41rue_mohrless if you have more than 13 lines, cause thats were a t1 becomes cheaper
07:47.46Ast-Mmaliuta Well put ;)
07:48.20rue_mohrbut you still need land lines for 911 calls
07:48.40Maliutarue_mohr: or some sort of GSM interface
07:48.55HellBound~_^
07:49.04Maliutarue_mohr: that would allow you to route those calls over the mobile network
07:49.12rue_mohrit works if a) everyone has a sip phone b) its being used for cheating longdistance
07:49.29MaliutaAst-M: some people need to be bitchslapped into reality
07:49.33rue_mohrer isn't gsm satillite?
07:49.43jblackHellbound: People can make free sip to sip calls, and such. But calling phone numbers, that always costs.
07:49.51Maliutarue_mohr: no, GSM is mobile
07:49.59rue_mohrah, sorry
07:50.10rue_mohrok, I'm out of steam, gnight
07:50.20jblackHellBound: Sometimes, free sip ain't enough. This channel has a conference room that nobody ever uses. There's even a phone # hooked up to it
07:50.36Maliutaif someone has a [legal] way to connect to any pstn in the world for free I'm all ears
07:50.50HellBoundok
07:51.01Ast-MOk, so just to clear this up,  I still have to pay for my PSTN call somehow right? Darn
07:51.06HellBoundbut is there anyway to make it totally free
07:51.22HellBoundif yes i wan to give a try
07:51.23HellBound:D
07:51.48jblackAst-M: Right. Usually around 1.2 to 2.0 cents per minute.. which means a 1 hour call will cost less than a cup of coffee
07:52.20MaliutaHellBound: sure, just outlaw the worldwide use of any tech that isn't SIP/IAX based for voice. And make everyone move to 'net based voice comms
07:52.45HellBound~_^
07:52.57Maliutajblack: if that's landline you're being ripped
07:52.58HellBoundok ok :D
07:52.58jblackhellbound: You can get a no-cost incoming did from IPKall. That's incoming only.
07:53.08Ast-MHellBound: there are ways to make it fairly low cost at low volumes but I think your idea of setting up as some sort of free skype is probably not going to work, but if your offereing shares/stock in your startup for free I will take some just in case you find a way, I expect you will get a lot of subscribers but ofcourse they wont be paying anything ;)
07:53.25jblackMaliuta: That's voip.  Landline here is considerably more expensive.
07:53.53Maliutajblack: my VOIP-> landline is cheaper than that
07:54.12Maliutaand in .au nobody would dream of charging for incoming calls
07:54.13jblackthan 1.2 cents a minute?
07:54.21Maliuta8cents untimed
07:54.42Maliutaso my 3 hour call to my parents in canada costs me 8c
07:55.37Ast-MThis is getting silly, Im off the get *lunch*  I expect I will have to pay for said lunch as is not free either
07:55.47Maliutajblack: voip -> any landline in .au US, .uk, .ca ..... all $0.08AU  untimed
07:56.27Maliutajblack: and some of the destinations include mobiles (like .ca)
07:57.32*** join/#asterisk Nicolas\ (n=nicolas@91.176.96.4)
07:58.52jblackThat's australia, I presume?
07:59.08MaliutaI'm in australia
07:59.25jblackThat doesn't strike me as practical for americans.
07:59.32mchouMaliuta: which ITSP you have?
07:59.40Maliutayou can still signon to that provider, your packets just have to come over here first
07:59.52Maliutamchou: www.pennytel.com
07:59.58jblackThat's why it wouldn't be practical. The latency would be a pain.
08:00.15Maliutajblack: it's not _that_ bad
08:00.22mchouMaliuta: do they do iax?
08:00.31Maliutajblack: and it doesn't stop it beign "practical"
08:00.33*** join/#asterisk af_ (n=getsmart@88-149-230-104.dynamic.ngi.it)
08:00.34Maliutamchou: no
08:00.39Ast-MAnyone anyidea why asterisk-addons wont compile?   Is like it cant find the main asterisk source
08:01.31jblackWhat is the server? I'll check.
08:01.33mchouMaliuta: what their sip server address? Want to traceroute it to get idea of hops
08:01.38Ast-Mshould add wont compile *for me*
08:01.56Ast-Mjblack: debian
08:02.05jblackAst-M: Not you.
08:02.09Ast-Msorry
08:02.31Maliutamchou: sip.pennytel.com
08:02.31jblackMaliuta: If it's in the same data centre, then americans are looking at latency of 264ms. That's not very good.
08:03.08jblackYeah. About the same.
08:03.32Maliutamy latency over my DSL connection is more than that
08:03.51jblackare you serious?
08:03.55mchouMaliuta: whoa, # of hops isnt that bad
08:04.18mchouMaliuta: so what you say, all this is free?
08:04.51Maliutamchou: did you read what I actually said? or look at the website?
08:05.17Maliutajblack: ping says my average latency is up at around 270ms
08:05.18mchouMaliuta: nah, I just joined the channel, more or less
08:05.44jblackTo your isp's gateways? That's atrocious relative to what we see in the states.
08:06.15Maliutajblack: pennytel aren't an ISP they only do my SIP
08:06.39mchouIt bothers me that traceroute goes through alter.net though.....
08:06.43tzafrir_laptopAst-M, mind sharing with us the buil logs?
08:06.46Maliutajblack: .au is larger and more sparsely populated than the US
08:07.16jblackHeh. Australia is larger than the US
08:07.24mchoulooks like 3 hops in sydney
08:07.38tzafrir_laptopdo you use the package asterisk-dev?
08:08.04jblackHow exactly did you come up with "larger" ?
08:08.08mchouI dunno, not sure it would work out
08:08.48jblackmore sparse, I don't know whether I agree or not. Don't most aussies live in a city?
08:08.59mchousigh
08:09.17mchouany fool knows australia is more sparse than US
08:09.35mchouwhy even ask the question?
08:09.57mchougo ahead and get devoured by dingos in the outback :)
08:11.47Ast-Mtzafrir_laptop: me? is so no I used source tar
08:12.23tzafrir_laptopso please pastebin the build log
08:12.32tzafrir_laptopalso: what versions?
08:12.47jblackI don't see anything about population density at the factbook. Less land, but a lot less people.
08:13.20jblackI imagine most live near the coast though, which could lead to higher density.
08:14.09mchouthe whole population of australia is commensurate to NYC
08:14.42mchouthere's no WAY there'd be higher population density than the US
08:14.44Ast-Mtzafrir_laptop: http://www.pastebin.ca/1196257
08:15.22jblackit can't find the asterisk headers.
08:15.37Ast-MI thought that but not sure why,
08:15.42HellBound~_^
08:15.51tzafrir_laptopapp_addon_sql_mysql.c:19:22: error: asterisk.h: No such file or directory
08:15.51HellBoundgot dc
08:15.52jblackput the asterisk sources in /usr/src/asterisk That should take care of that, I think
08:15.57tzafrir_laptopthat's the error
08:15.59Ast-MI even ln -s asterisk.1.4.12 asterisk
08:16.07tzafrir_laptopit should be: /usr/include/asterisk.h
08:16.26Ast-Mah, so I have to make install asterisk first?
08:16.28tzafrir_laptopor maybe /usr/local/include/asterisk.h ? or whereever you installed asterisk to
08:16.31Ast-MDoh
08:16.48jblackWell, hold up. YOu said debian, right?
08:16.50tzafrir_laptopI wonder how the configure script let it go through
08:17.20jblackMaybe I'm getting confused with the zaptel drivers, which expect /usr/src/asterisk
08:17.24Ast-Mits currently running a bin asterisk and its live, so I wanted to get everything inplace before I make install asterisk.
08:17.46Ast-Mthe zaptel and librpri built fine.
08:18.13jblackDebian should have the mysql stuff module already.
08:18.13Ast-Mas did asterisk but I did not do the make install yet
08:18.15jblackWhy are you building it?
08:18.22mchouMaliuta: you familiar with the term "harden the fuck up?" :)
08:18.44*** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132)
08:18.48Ast-Mneed func-devstate
08:18.52mchouMaliuta: I forget the name of the comedian.....
08:19.10mchouMaliuta: but he sure is hilarious
08:19.31Ast-Malso might need to *hack* so stuff to stop tromboneing of call between two switches over IAX, unless it already does it.
08:20.11Ast-Mlikes source, you know where you are with the source.
08:20.21mchouAst-M: tromboneing??  what's that?
08:20.59Ast-Mlots of calls between switches that are not needed becuase the final two ends are actually on the same switch.
08:21.09mchouAst-M: whateve it is, it sounds kinky :)
08:21.22mchouwhatever*
08:21.44Ast-MImagine a trombone, moving in an out ;)  think of that as one call accross two servers
08:22.07Ast-MI think there is another term like *takeback*
08:22.20mchouhow about ping-pong :)
08:24.12Ast-Mwhen the person at site 2 joins the call to a person at site one, as the call is originally from site one there is no need for it to go to site two and back again.
08:24.33Ast-Mthe site one server could/should join them at site one.
08:25.32mchouAst-M: you referring to signalling or RTP?
08:25.59Ast-MBoth I expect.
08:26.17Ast-Mmaybe not so bad if is only signalling I guess.
08:26.34Ast-MBut really I want the user at site 2 to trigger a reinvite at site 1
08:26.39mchouRTP can be direct
08:27.16Ast-MThe call would be :  PSTN-IAX-SIP-IAX-SIP Would this cope?
08:27.41mchoulooks ugly
08:28.00mchoudepends on a whole lot of variables
08:28.04Ast-MWe receive the call from pstn via IAX from an ITSP at site 1
08:28.39Ast-MThat is sent to site 2 via IAX ( 2 channels used now ),  Then the server at site 2 presents this to a SIP extension.
08:28.47mchouwhy not just skip the SIP in the middle?
08:29.08mchouAst-M: IAX trunking
08:29.14Ast-MThey put it on hold ( at site 2 ), then make a call to an sip extension at site 1 via IAX between the two systems.
08:29.59Ast-MIf the person at site one want that call its bridged, but I think only at the site 2 asterisk server, when actually it could be bridged at site 1
08:30.10mchouI dunno.  best to look at the rtcp or sip debug
08:30.35mchoushould tell you reinvites and rtp is being set up
08:31.02Ast-Mbut can the sip reinvites get passed over the IAX link between the switches.
08:31.13Ast-MShould I put all the phones on the same asterisk server.,
08:31.23mchoudoubtful
08:31.37Ast-Mits 50 extensions at each site.
08:31.57mchoubet there is also funky transcoding going on at each iax
08:32.32Ast-MIts all g711a but probably not ideal,  not sure how else to do it really.
08:33.10Ast-Mmaybe all the reception phones on the same asterisk server regardless of site, route all incoming calls to that server and queue.
08:33.27mchouif it's all g711a then there's no transcoding (by definition) betw. boxes.
08:33.27tzafrir_laptopAst-M, ./configure --with-asterisk=/path/to/asterisk/root
08:33.45tzafrir_laptopyou might need to add some dummy symlinks there
08:34.05Ast-Mtzafrir_laptop: I did a make install now on /usr/src/asterisk,  I was just putting it off as its a live sysyem ;)
08:34.53Ast-MThanks for you help, builds now, maybe the ./configure in asteriks-addons could check for the source.
08:35.25Ast-Mtzafrir_laptop: you got any ideas about my two system configuration by any chance?
08:35.32*** join/#asterisk mandh (n=mandh@82.137.216.38)
08:35.38DarKnesS_WolFtzafrir_laptop: hello man :-) how are u ?
08:36.15tzafrir_laptopDarKnesS_WolF, still working on the why. Will get to the how some day
08:36.37tzafrir_laptop:-)
08:38.17DarKnesS_WolFtzafrir_laptop: mmm good for u :P
08:41.57mchouanyone have experience running asterisk 1.4 w/openwrt on a linksys wrt54g?  Wanna know if the HW has enough RAM w/o going int an OOM condition
08:42.53tzafrir_laptophow much memory do you have?
08:44.29DarKnesS_WolFtzafrir_laptop: u might be able to help me
08:45.20mchoutzafrir_laptop: wrt54g has 16M RAM, iirc
08:45.49tzafrir_laptophmmm... quite marginal
08:46.22mchoutzafrir_laptop: which is why I'm soliciting feedback here
08:47.14mchoudont want to install and have my home network disappear for a few measly phone calls
08:47.22mchouhehe
08:47.28DarKnesS_WolFtzafrir_laptop: i havfe one sip phone behind nat using G729,-------> internetl ------> asterisk using g729 -----> sip provider G729 --> connect are from ip phone to asterisk 256/64 and then from asterisk to sip provider 512/128, but the call gets soo much delay u think it is connection ? or what ?
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08:50.00mchouDarKnesS_WolF: this is sip to sip w/o going thrus PSTN?
08:52.15DarKnesS_WolFsip to sip
08:52.19DarKnesS_WolFpure voip
08:52.36mchoubut then you say 'sip provider'
08:52.56mchoumakes no sense
08:53.06jblackwhoah
08:53.07tzafrir_laptopDarKnesS_WolF, IIRC you can show sip connection stats
08:53.19jblack"The agency vehemently denies allegations officials knew the alleged perpetrator, a 7-year-old boy, had been accused of sexually assaulting other children prior to being placed in the relativeâ's home in 2000"
08:53.20tzafrir_laptop(well, at least for IAX you can)
08:53.40mchouDarKnesS_WolF: tzafrir_laptop is correct
08:55.31DarKnesS_WolFtzafrir_laptop: ok will do thx
08:55.37DarKnesS_WolFtzafrir_laptop: but not bandwidth issue ?
08:56.35tzafrir_laptopmaybe. do you have an idea if the capacity of the line is getting full?
08:56.51tzafrir_laptophow many calls are there on the connection?
08:57.09tzafrir_laptopcan you try to give voip traffic higher priority than other traffic?
09:01.28DarKnesS_WolFtzafrir_laptop: that what i think
09:01.38DarKnesS_WolFtzafrir_laptop: 2 max from ip phone to * then from * to sip provider
09:01.42DarKnesS_WolFso 2 G729
09:01.59DarKnesS_WolFand the ping time becom 4 scounds
09:01.59DarKnesS_WolFand lots of delay.
09:02.04DarKnesS_WolFtzafrir_laptop: this line is only for voip
09:02.10DarKnesS_WolFhow much teh G729 should take ?
09:02.42tzafrir_laptop~calculator
09:02.43jbotSimple Command Line Calculator. URL: http://www.mindspring.com/~joelgg/calc.html
09:02.50tzafrir_laptop~bandwidth
09:02.50jbothmm... bandwidth is This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus.
09:03.06tzafrir_laptop~bandwith calculator
09:04.05tzafrir_laptophttp://www.asteriskguru.com/tools/bandwidth_calculator.php
09:04.52tzafrir_laptop~bandwidth calculator
09:05.36tzafrir_laptopjbot, bandwidth calculator is http://www.asteriskguru.com/tools/bandwidth_calculator.php
09:05.37jbotokay, tzafrir_laptop
09:06.29*** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132)
09:07.41jblackThat's a nice calculator.
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09:08.24jblackI didn't realize iax2 saved so much bandwidth
09:11.53DarKnesS_WolFthx tzafrir_laptop ;)
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11:12.04meredyddAfternoon, all.
11:13.11meredyddIs there a generally-accepted method for getting Asterisk to authenticate clients with an external auth system rather than manually-configured users in the config file?
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11:20.56mvanbaakmeredydd: realtime
11:24.29meredyddmvanbaak:thanks!
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11:36.28zxdhi
11:36.36zxddoes asterisk come with a benchmark utlity
11:37.04zxdto generate lots of audio encoding/decoding
11:39.42mvanbaakzxd: you want to test sip ?
11:39.55DarKnesS_WolFzxd: for sip try sipp
11:39.58DarKnesS_WolFas far as i recall
11:40.05mvanbaakindeed
11:42.25zxdi want to simulate alot of calls so the cpu will get with the codecs encoding and decoding business
11:49.24nr4qanyone messed with alert info and polycom phones ?
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11:57.23hi365im having a weird situation where asterisk is acting slow... im using Read() and its taking forevet to respond to #, wating for the timeout. The same dialplan has worked befor, and there is no system stress. Any thouhgt?
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12:46.49luca`gervasiHallo
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13:41.04Alton2.
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14:39.13hasslerHi folks, anyone for a quick question regarding configuration for an analog and t1 card in the same system?
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14:47.56[TK]D-Fender~ask
14:47.57jboti heard ask is Questions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
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14:53.35luca`gervasihallo. I registered one number with my voip company into asterisk using "register => username:password@sip/inbound", the "inbound" dialplan just calls one of my interns. When i call my voip number from my landline, asterisk says:  chan_sip.c:13888 handle_request_invite: Failed to authenticate user <sip:_mylandline_@_someip_> ... where is the glitch ?
14:53.58*** join/#asterisk saftsack (n=saftsack@cm1011282-a.maast1.lb.home.nl)
14:55.29[TK]D-Fenderluca`gervasi: "registering" does not authenticate incoming calls.  You need a peer set up to auth the incoming call
14:55.51luca`gervasiuhm...like? i'm a newbie :D
14:56.07[TK]D-Fenderluca`gervasi: Go look at how every other ITSP does this.
14:56.33luca`gervasiuhm... ?
14:57.39[TK]D-Fenderluca`gervasi: [infromwherever] type=peer   host=myprovider.com , etc...........
14:58.08[TK]D-Fenderluca`gervasi: Go check your providers FAQs, read the book, and compare how every OTHER provider is set up with *
14:58.19luca`gervasitrying :)
15:02.16nr4qfender: (this is kj4acm from the other day) thought i'd let you know about the resolution to the problem I was having with the php permissions. not sure if you remember me or the problem i was having
15:03.46nr4qfender: (really a trixbox thing). installed trixbox 2.6.1 changed all the default passwords and got everything working. updated all the packages... apparently one of the updates changed the mysql password which broke the webinterface
15:04.21luca`gervasiIf "voip.eutelia.it:5060            09221830512        105 Registered", then i should be able to receive calls, right?
15:04.41[TK]D-Fenderluca`gervasi: no, that indicates that you have registered
15:04.50[TK]D-Fendernr4q: Congratulations
15:05.29[TK]D-Fenderluca`gervasi: Sorry, actually you are receiving calls... ecept that they are getting REFUSED because you are not autoherizing them
15:05.54[TK]D-Fenderwishes he could type, but that won't be for at least a week or two
15:06.04Wayhighlet's take a poll.. how many incoming/outgoing trunks do ya'll have registered?
15:06.19brodiemwhat's up fender, you break a finger?
15:06.21nr4qfender: i ended up reinstalling after becoming frustrated... which didnt get me anywhere because i updated and that's when i discovered that's what happened. since it's a freepbx thing it shouldn't be in here anyway though
15:06.27WayhighI've got 7/3 at the moment
15:06.30luca`gervasii have this: [eutelia]; type=peer; host=voip.eutelia.it;context=inboundroute0
15:06.37luca`gervasieach ";" means new line
15:07.14luca`gervasiisn't it sufficient to authorize each call from my voip provider? (eutelia)
15:07.32[TK]D-Fenderluca`gervasi: No.  Go look on the WIKI for a sample from your provider
15:07.34[TK]D-Fender~wikis
15:07.35jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
15:07.51luca`gervasithanks, looking :D
15:08.56[TK]D-Fenderbrodiem: Sword accident, plenty of blood
15:09.03nr4qfor a 2 DID system would ISDN be the preferred connection method over Zap?
15:09.41[TK]D-Fendernr4q: what kind of ISDN?  How many channels?
15:09.55Wayhigh<-- wishes he could find a DID that's local to my landline ratecenter and didn't want $35 to sign up
15:10.19nr4qFender: well the client has 2 DIDs.  could go with 2 POTS lines or a BRI
15:11.11nr4qjust haven't used any of the ISDN hardware. know that the cards like the TDM400 has a tendancy to echo sometimes
15:11.13[TK]D-Fendernr4q: POTS doesn't do DID's.  Each is pretty independant
15:11.45nr4qfender: okay they have 2 telephone numbers
15:12.17luca`gervasiWow It works!!!
15:12.34luca`gervasinow i need to understand why I can't authenticate my phones to my * :D
15:14.28nr4qfender: let me phrase it differently. client says "wow there's echo problems with this sytem" which is using a TDM400 card. would a possible solution be to switch to ISDN
15:14.33Wayhighanyone here have a recommendation for DIDs that isn't vitelity?
15:15.49[TK]D-Fendernr4q: What EC have they used?
15:16.14nr4qthis is hypothetical
15:16.37[TK]D-Fendernr4q: Didn't sounds hypothetical a moment ago....
15:17.10[TK]D-Fendernr4q: If they are teh client that seems to make you their service provider and you should know better.
15:17.25[TK]D-Fendernr4q: Go get familiar with the options out there and thier ups & downs
15:17.50luca`gervasi~pastebin
15:17.51jbot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
15:18.02nr4qit's a friend of mine. i give him free service... he doesn't expect much since he isn't paying for it
15:18.18Wayhigh~did
15:18.18jboti guess did is Direct Inward Dialing, or just a phone number
15:18.24[TK]D-Fendernr4q: Good, then I'm sure any choice you make will do :)
15:19.00nr4qfender: i've tried to research this a bit but haven't found it laid out in one spot
15:19.16nr4qi'll practice my google-fu more
15:19.36luca`gervasithis should allow my phone to register itself with asterisk... right? http://pastebin.ca/1196472
15:20.21[TK]D-Fenderluca`gervasi: looks normal enough, though you should specify codecs.
15:21.30luca`gervasidoesn't it get the ones defined on top? i disallowed all and allowed ulaw, alaw, gsm
15:21.46[TK]D-Fenderluca`gervasi: You should define them in EVERY peer you use
15:22.14[TK]D-Fenderluca`gervasi: And for a given peer you should normally only have 1 codec selected
15:22.55luca`gervasioooh... witch one should i use? :D
15:23.40luca`gervasithe hw phone says g.711, g.723, g.726, g.729a and g.729b
15:23.49luca`gervasiwitch one is better?
15:24.11luca`gervasihow can i see if my asterisk setup supports them?
15:24.16brodiem[TK]D-Fender: lol, FTW
15:25.08tzafrir_laptopluca`gervasi, on a LAN or a WAN?
15:25.20luca`gervasilan
15:25.35tzafrir_laptopg.711
15:26.11luca`gervasihow can i see if * supports it on my setup?
15:26.21Wayhighsweet.. looks like my AC is finally working again.. now if I only knew why the dang AC fairy took a crap on my AC for a week I'd be all kinds of happy
15:26.26tzafrir_laptopothers compress (more) and take less bandwidth but also reduced quality
15:26.41luca`gervasicore show codecs :D
15:26.41tzafrir_laptopon a LAN you don't really care about the bandwidth
15:26.59[TK]D-Fenderluca`gervasi: "alaw" <-
15:27.07luca`gervasialaw.. so be it :D
15:27.59luca`gervasimy phone still doesn't register...
15:28.28tzafrir_laptopdo you use g.711a (alaw) or g.711u (ulaw)?
15:29.00luca`gervasialaw
15:29.08luca`gervasibut i think the problem is elsewhere
15:29.24[TK]D-Fenderluca`gervasi: indeed it is.
15:29.26luca`gervasiasterisk writes nothing on the console
15:29.40[TK]D-Fenderluca`gervasi: then you should have enabled "sip debug"
15:29.45luca`gervasi(still i disabled firewall and so on)
15:29.50luca`gervasisip debug
15:29.57luca`gervasicore set verbose 999999999999999999
15:30.03luca`gervasicore set debug 99999999999999999
15:30.07[TK]D-Fenderluca`gervasi: And look at the actualy packets * is receiving
15:30.45[TK]D-Fenderluca`gervasi: If you see no packets with sip debug enabled, then either your firewall/networking is bad, or your phone isn't even pointed in the right direction
15:30.57tzafrir_laptopluca`gervasi, you don't get anything from debug/verbose of more than 10
15:31.55luca`gervasiuhm...
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15:51.41*** join/#asterisk eklof (i=jonas@trimix.eklof.eu)
15:51.44eklofHi,
15:52.09eklofIs it possible to have asterisk play a specific message beck to the caller if no telephones are loggin in ?
15:55.55tzafrir_laptopeklof, in a queue?
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15:59.11WimpManeklof: Check DIALSTATUS
15:59.30eklofok
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16:13.07outsider12hello - does anyone have the sip firmware for cisco 7975g?
16:14.30[TK]D-Fenderoutsider12: www.cisco.com probably does.
16:14.59mvanbaakoutsider12: you can try to use it with chan_skinny
16:15.02outsider12yea they do, but the particular phone requires smartnet
16:15.17outsider12mvanbank - chan_skinny?
16:15.30mvanbaakoutsider12: SCCP implementation in asterisk
16:16.13outsider12to be honest, I much rather upgrade to SIP - I have many of the 7940/7960 on sip and they work great, -- frankly i dont know anything about skinny
16:16.38mvanbaakoutsider12: then you have to get a smartnet account
16:17.20outsider12mvanbaak- i know, but i was just trying to avoice the hassle calling cisco (we all know how difficult they are).
16:17.29outsider12avoid*
16:17.59mvanbaakgetting a smartnet account is easy
16:18.09outsider12is it an instant process?
16:18.15[TK]D-Fenderoutsider12: Maybe you should have gotten a phone that doesn't require you licensing it first then.
16:18.17mvanbaakno
16:18.32mvanbaakbut almost instant
16:18.40mvanbaaktakes 2 hours or something like that
16:18.52mvanbaakyou can buy it online at many webshops
16:19.01mvanbaakyou register an account on cisco.com website
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16:19.25*** join/#asterisk crycos (n=paul@216.159.238.110)
16:19.27mvanbaakwait for the smartnet number to arrive in your mailbox, link it to your account at cisco.com and there you go
16:19.34mvanbaak1 year of downloads for your phone
16:21.01crycosI am on asterisk 1.4 . Does anyone know why the inbound fax is not working? it has to do with the zapata.conf file  I believe
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16:22.56jon_mvbank- do you have a smartnet account, and I would send paypal $$ for the help?
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16:23.35jon_my name just changed - from outsider12  :/
16:23.58jeevleif, are you awake
16:24.02jeevslaps lmadsen's ass
16:25.53mvanbaakjon_: I dont have a smartnet account anymore. I had one, but I did not renew it
16:26.17jon_mvanbaak - ok... bu thank you for the help ;)
16:27.44jon_mvanbaak - how do i configure the unit with skinny with asterisk? through the endpoint manager?
16:28.21mvanbaakhave a look at conf/skinny.conf
16:28.52mvanbaakand I dont know the 7975.
16:29.03mvanbaakmaybe you can config it using the keypad or a webbrowser
16:29.05DarKnesS_WolFpasta with tuna i made so bad :-s damn internet recpises
16:29.10mvanbaakI provision my phones using tftp
16:29.28jon_mvanbaak- I have only used tftp also with the 7940's and 0's
16:29.29jon_60's
16:29.43jon_this is the same, but i dont have the sip firmware
16:29.44mvanbaakyeah, I have 7960's and 7905's
16:29.52mvanbaakme neither
16:29.54jon_this is similar load to the 7911
16:29.57jon_(like 7905)
16:29.58mvanbaakall my phones are running skinny
16:30.21jon_really, ok i will try to get this up and running with skinny, I have the unit pointint to the TFTP
16:30.38rue_mohrat some point somone is gonna help somesone with something and i'm gonna learn something....
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16:32.48DarKnesS_WolFrussellb: weekends u come late ;-)
16:34.45jon_mvanbaak- is iax2 considered skinny?
16:35.47crycosyou using ?
16:36.51jon_asterisk / cisco 7975 phone on skinny - dont have sip firmware :(
16:36.53mvanbaakjon_: no
16:37.12jon_mavanbaak- ok i guess i'll do some more research on skinny/asterisk
16:37.15jon_thank y ou
16:43.22*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
16:52.10ManxPowerI might consider IAX2 a thin protocol, but not a skinny protocol.
16:53.10[TK]D-FenderManxPower: ba-dum-dum!
16:57.32gr0mitanyone from France here ?  Need some advice about calls to 0899 numbers!
16:58.27ManxPowerlet me know when you are interested in non-French opinions
16:59.07*** join/#asterisk lotho (n=lotho@valhalla.via-publica.de)
16:59.23[TK]D-Fender<PROTECTED>
16:59.56gr0mitmais oui!
17:00.15ManxPower[TK]D-Fender: If he wants to limit help to only people in France, I'm not going to argue with him.
17:00.24gr0mitis trying to find out some official pricing of calls within France
17:00.43ManxPowerI assume a 0899 number is a toll free, but it might be a premium.
17:00.45[TK]D-Fendergr0mit: I'm sure your provider will tell you how much they'll charge....
17:01.01gr0mitto 0899 numbers, which are listed on the French regulator as '0899 Autres tariffs'
17:01.07ManxPowerI could look up what kind of number it is.
17:01.16gr0mitit is premium rate
17:02.22gr0mitbut seems a wide variation in price from carrier to carrier
17:02.22gr0mitso not clear.
17:02.23jayteeI think the French base their per/minute rates for calls as a fraction of the price of a kilogram of butter. Something like .005 X Price of 1 Kilo of butter.
17:02.23ManxPowerIn France 0899 650 160 (calls charged at EUR 1.34 per call and EUR 0.34 per minute. Calls from mobile networks may be charged at a higher rate)
17:02.23ManxPowerfrom a web page.
17:02.34gr0mitooh - which one, manxpower?
17:02.45ManxPowerhttp://wizzair.com/about_us/contact_us/
17:02.56ManxPowernot an official rate, but at least there is some info there.
17:03.16gr0mitperfect -
17:03.20gr0mitthanks ManxPower
17:03.33gr0mitlooks outrageously expensive!
17:03.55gr0mitmakes a mental note not to call wizzair
17:04.03*** join/#asterisk Levonk (n=lk@adsl-76-237-14-170.dsl.lsan03.sbcglobal.net)
17:04.06gr0mitbbl.
17:04.12ManxPowerHere's another one http://www.frenchentree.com/france-indre/DisplayArticle.asp?ID=18076
17:04.53ManxPower"0899 – more than €1.21 per min"
17:05.19ManxPowergr0mit: your best bet is to block 0899 numbers.
17:05.41mvanbaakindeed
17:05.44*** join/#asterisk ujwal (n=ujwal@124.41.197.33)
17:09.32ManxPowerlooks like they are like the 900 or 976 numbers in the USA.
17:11.32jeevManxPower, so far so good with GRE.
17:11.56ManxPowerjeev: Cool.  You move to SIP as well?
17:12.42jeevyea
17:12.45jeevi couldn't test it fully
17:12.49*** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
17:12.50jeevi did it through xlite RDP from the office
17:12.56jeevsince i'm too lazy to go there
17:13.03*** join/#asterisk F00JIN (n=F00JIN@lns-bzn-58-82-251-197-99.adsl.proxad.net)
17:13.06jeevbut audio one way worked, DTMF the other way worked.. so i will test tomorrow.. by talking on it and seeing if ti works
17:13.07F00JINhi !
17:13.31drmessanoI wonder howX-Lite would work over Citrix
17:13.34ManxPowerjeev: one-way-audio is almost always a nat or filter problem.
17:13.59ManxPowerAh, that is only what you could test, not what was wrong.
17:14.16ManxPowerjeev: set up an extension to run Echo() or do a Record and a Playback
17:14.29jeevah
17:14.41jeevno, only reason why it's one way audio is cause RDP doesn't have microphone
17:14.43jeevso i couldn't test it.
17:14.49ManxPowerah, OK.
17:14.54jeevi'm sure it works both ways
17:15.14ManxPowerI hope it works for you.  Don't forget to try enabling reinvites once everything is stable.
17:15.22ManxPower(and working for a few days)
17:16.33ManxPowerjeev: on Cisco routers, you can enable Compressed RTP (saves quite a bit of bandwidth).  You might consider finding out if BSD supports cRTP..  I doubt it does, but you never know.  I'll be turning on cRTP on our network soon.
17:16.58*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
17:18.43*** join/#asterisk nofxx (n=nofxx@unaffiliated/nofxx)
17:19.36nofxxAny reason I`m getting not found  for modules.conf and logger.conf, even they both being on /etc/asterisk ?
17:19.50ManxPowernofxx: you installed from a package, didn't you?
17:20.03nofxxManxPower: yes , ports mac os x
17:20.28ManxPowernofxx: then you will have to ask the package builder where this build of Asterisk expects to find it's config files.
17:20.30[TK]D-Fendernofxx: wrong place, wrong spelling, or wrong permissions.  Pick one.
17:20.59nofxxhm... I think I got it... thanks!
17:20.59ManxPowerWe really don't have much love for packages here.
17:21.41nofxxManxPower: first time, just to play around... try adhearsion
17:21.54*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
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17:26.23*** mode/#asterisk [+o russellb] by ChanServ
17:32.52Wayhighanyone here used voip.ms?
17:35.33Wayhighbabbles to himself
17:38.10drmessanonope
17:39.28nofxxsoftphone suggestion for mac ?
17:39.40drmessanoX-lite
17:40.05nofxxgot gizmo, loudhush, zoiper...
17:40.08*** join/#asterisk N9URK (i=IceChat7@159.sub-70-223-201.myvzw.com)
17:40.13nofxxdrmessano: gonna try
17:41.22YournameWill someone be kind enough to help me get call forwarding operational? Like *22<phonenumber> will enable call forwarding, and *23 will disable it? Asterisk 1.4.21. Thanks.
17:42.44N9URKI am thinking about running * on a vps with 128MB ram on xen.    How well would 128MB work for this scenerio?  I want the VPS * setup to mainly handle the IVR/voice mail and then it will IAX transfer the call to the *box at the office.   I suspect that the *on the VPS will only have 3-5 concurrent calls at most.  Does this sound like a winning scenerio?
17:44.21DarKnesS_WolFN9URK: i think yes also depend on ur codec ur going to use
17:44.32N9URKbeen using ulaw mostly
17:44.45N9URKYourname:  http://www.voip-info.org/wiki-Asterisk+call+forwarding
17:44.53DarKnesS_WolFyes i think memory and cpu will be more than enough
17:45.04N9URKYourname: I think that will do it, but I only took a quick brief gander
17:45.15N9URKlet me know if that doesn't answer your question
17:45.28N9URKDarKnesS_WolF: awesome thanks
17:46.04*** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb)
17:46.40N9URKDarKnesS_WolF: since you bring up codec, which codec do you prefer?
17:46.52YournameN9URK: Thanks, I looked in it. Didn't really give me what I want for the way I need it.. :S
17:46.57DarKnesS_WolFN9URK: g729 cuz we don't have much bandwidth in egypt
17:47.05DarKnesS_WolFN9URK: gsm / speex are cool
17:47.18DarKnesS_WolFYourname: what u want ? to do ?
17:47.43YournameDarKnesS_WolF: Will someone be kind enough to help me get call forwarding operational? Like *22<phonenumber> will enable call forwarding, and *23 will disable it? Asterisk 1.4.21. Thanks.
17:47.55N9URKExample 3 on teh link I sent should be it
17:48.03[TK]D-FenderYourname: Get off your ass.
17:48.08ManxPowerYourname: So you basically want someone to write a call forwarding system for you for FREE?
17:48.24drmessanoIs he still spamming?
17:48.35[TK]D-FenderYourname: Its 1 stupid dialplan function and a a GotoIf.
17:48.41drmessanoYourname: Download Trixbox
17:48.42DarKnesS_WolFYourname: yes that link should do it
17:48.54drmessanoYourname: The magic inside Trixbox is made of chimps
17:48.57[TK]D-Fenderdrmessano: thats almost what I suggested he do :)
17:49.21drmessanoLittle chimps that write dialplans
17:49.37drmessanoYou want to forward: Spank the monkeys
17:49.52drmessanoYou want an auto attendant with a hot voice: Spank the monkeys
17:49.59drmessanoAll about the chimps
17:50.34drmessanoSo in essence, go download Trixbox and get busy spanking the monkeys
17:51.57*** join/#asterisk JenniferAkemi- (n=akemi@206-248-161-97.dsl.teksavvy.com)
17:52.51DarKnesS_WolFdrmessano: enough spanking the moneys :P they did nothing to u :D
17:53.34Yournamelol drmessano
17:53.56*** join/#asterisk Levonk (n=lk@adsl-76-243-67-245.dsl.lsan03.sbcglobal.net)
17:54.27YournameSince I'm getting a lot of attention today.. here's another question.
17:54.34drmessanoYourname: Seriosuly.. People in here do the kind of WORK you are asking for and get PAID to do it.. This is SELF HELP, not DO IT FOR ME.  You're almost being insulting here
17:54.37drmessanoSeriosuly
17:54.50drmessanoYou will get on the /ignore-help-vampire list very quickly
17:55.14drmessanoWow, and I spelled SIRIUSLY wrong twice
17:55.27YournameAnd a third time.
17:55.31YournameAgent101 is logged in via AgentLogin(). Suddenly his internet drops. However, Asterisk still thinks Agent101 is logged in. A few mins later, Agent101 regains internet connection and tries to log back in, but Asterisk says "Already logged in". Most of the time, if we wait a few more minutes, Asterisk finally somehow realizes that Agent101 is not really active and drops him out of the queue. Either that, or someone soft hangup that chann
17:57.24N9URKYourname what does that have to do with call forwarding?
17:57.45drmessanoN9URK: He
17:57.59drmessanoN9URK: He's trying to squeeze that into his $200/hr of free consulting
17:58.04drmessanoCut him some slack
17:58.24N9URK:)
17:58.39drmessanoDamn hams.. always trying to be so technical
17:58.51drmessanoThis isn't 75-meters
17:59.05N9URKReminds me of an aquaintance who once asked me for some help with *@H, which I should have known better
17:59.07N9URKbut I helped anyway
17:59.17N9URKand about halfway through this complicated fix
17:59.25Yournamelol
17:59.28N9URKhe says, hang one, I have a call from the "money man"
17:59.32YournameIt has nothing to do with call forwarding, N9URK.
17:59.37YournameIt's a separate question.
18:00.11drmessanoYourname: The net has not ackowledged your check-in
18:00.12N9URKthen I am like, "what do you mean money man"
18:00.13YournameAnd drmessano, this is not a frikkin consulting IRC channel. Open your own if you consider every question a supposed money maker. SIRIUSLY.
18:00.27drmessanoORLY?
18:00.31N9URKNet Control, thank you for the ack
18:00.37YournameIt's an open source community help channel. If someone chooses to help, they will. So stfu.
18:00.48filecome down all
18:01.11fileYourname: if using SIP there is an rtptimeout option in sip.conf which will disconnect a call if it does not get audio from the device within a settable period of time
18:01.41Yournamefile: Thanks, is that a per peer setting? Let me look in sip.conf
18:02.17drmessanoN9URK: You're 5-9-9 good buddy rubber duck
18:02.27N9URKdrmessano: so I tell him to shove off.  I am glad to help in anyway I can, and don't mind even if someone else is getting paid, but to get deeply immersed and then told he was being paid made me angry. lol
18:02.38drmessanolol
18:02.38N9URKthat's a big 4 roger
18:03.13N9URKI just spotted 4 bears harrasign a lot lizard at the pickle park at the 49 yardstick
18:03.42Yournamefile: It's not! Great, I'll use that as it looks like the one setting I'd need. Do you know what the default is if I don't set it explicitly?
18:04.19fileif you do not set the option it is disabled, and if you set it I'm pretty sure you have to give it a time
18:04.28drmessanoN9URK: It helps in here to be respectful of those that actually make a living in Asterisk.. To have free access for little tips and help here and there from people that put food on their tables coding dialplans is pretty slick to me.  I have NO tolerance for those that come in here with "Fix my pastebin so I don't get fired.  I R PAID company Akerisk Expurt"
18:04.58Yournamefile: Ok, cool, thanks a ton!
18:05.08*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
18:05.41drmessanoJust like that guy that posted the job survery questions on the Ubuntu forums for the Net Admin job
18:05.58drmessanoand the guy that send the applicant the questions happen to post there too and saw the list.. lol
18:06.04drmessanoThat was... total FAIL
18:06.22N9URKdrmessano: OMG that is hilarious
18:07.12drmessanoSure, we all use Google..
18:07.19N9URKdrmessano: there is a middle ground though.  I do hate RTFM answers, especially when they don't tell you where to look
18:08.14N9URKdrmessano: I have a friend who you ask him a question he will always say: "come on man, RTFM. this shit is in the man"  then I say, "where?  I looked and didn't see it" then he will be like, "I don't know, I never looked for it before"
18:08.21*** part/#asterisk HellBound (i=hellboun@should.have.tried.shellium.org)
18:09.22WimpManI have a friend who will usualyy answer RTFS.
18:09.31drmessanoMost people are like "The man? I don't have time to read, if I don't fix this, I R FIRED..  Help me please"
18:09.39WimpManAs in ...source
18:09.45drmessanoHell no, I won't help you.. Can you tell me where you work so I can apply for your job when they fire you?
18:09.53WimpMan"The manyally is probably outdated or incomplete anyway"
18:10.17drmessanoI always ask help-vampires that
18:10.21drmessano"So, who do you work for?"
18:10.29drmessano"Cool cool.. How long you been there"
18:10.48drmessano"Cool cool.. Think they would rehire your position when.. errr.. if you ever left?"
18:11.26N9URKI like that drmessano
18:11.28N9URK:)
18:12.54drmessanoI work in an environment where decisions have to be made fast.. fix whats there, or come up with a solution because that ones not working.. NOW..
18:13.06drmessanoNo time to get on IRC and put triggers in there for a week lol
18:13.52drmessanoI guess I have low tolerance for "I have been looking for the ANY key.. if I don't find it soon, I am gonna get fired.. it's been a week and I can't load Vista on my bosses new computer.  Help?"
18:14.14N9URK:)
18:14.33carrartake the time to build things right and you don't issues like that
18:15.20drmessanocarrar: Who builds things RIGHT anymore?  It's all halfass thrown together by clueless "experts" anymore.. It work "ok enough" is the new "done right"/
18:15.32carrarMaybe you do things half ass
18:15.41drmessanoUh yeah
18:15.46drmessanoThat's exactly what I was saying
18:16.17*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
18:16.56*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
18:17.00drmessanoDo you understand sarcasm?
18:17.00carrarIf your company won't let you do what it takes to build it right why work there?
18:17.14drmessanoMY company DOES
18:17.40outtoluncwho be sar carm?
18:18.09outtoluncgrr typo'd
18:19.08drmessanoMy point was that some people don't have the option of half-assing things and taking their sweet time to do so
18:19.40drmessanoSometimes you have to do things fast and correct and come up with solutions or else someone is going to take your seat
18:20.39ManxPowerAsterisk is a complex system that requires knowledge of Asterisk, SIP (and/or IAX2), RTP, NAT, networking, Linux, and telecom
18:20.53N9URKdrmessano: do you have a link to the job survey question debacle you were mentioning?
18:20.54ManxPowerFar too many people do not realize how complex it is.
18:20.56N9URKI would love to see that
18:21.32*** join/#asterisk F00JIN (n=F00JIN@lns-bzn-58-82-251-197-99.adsl.proxad.net)
18:21.40F00JINhi !
18:21.55ManxPower~ask
18:21.56jbotwell, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
18:21.56N9URKManxPower then add AGI, PHP, MySQL, C, C++, Cron, ad 50 other things you could think of :)
18:22.14ManxPowerN9URK: Yup.
18:22.29ManxPowerF00JIN: Did you have a question?
18:22.50N9URK~help
18:23.12carrarit's good to be assumptious :)
18:23.33N9URKhow do I get jbot's help/man file?
18:23.40N9URK~jbot
18:23.40jbotfrom memory, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass
18:23.53ManxPowerN9URK: /msg jbot help
18:28.50F00JINyes I have a question
18:29.09*** join/#asterisk xuser (i=jaood@unaffiliated/xuser)
18:29.13F00JINi'm trying to install asterisk 1.6 and asterisk-gui
18:29.22F00JINbut it doesn't work
18:29.31F00JINwith 1.4 no problem
18:31.10F00JINI don't know what i've done wrong
18:31.40*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
18:31.40*** mode/#asterisk [+o mog] by ChanServ
18:33.14YournameIs the CFIM key removed in 1.4.* or something? For some reason CFIM doesn't seem to be working, heck not even showing up in database show..
18:34.28Yournamesorry family
18:37.01*** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net)
18:38.20drmessanoN9URK
18:39.04*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
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18:47.15[TK]D-FenderYourname: pastebin <-
18:48.26*** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net)
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18:53.19*** join/#asterisk vk4akp (n=Ken@c58-111-159-31.ipswc2.qld.optusnet.com.au)
18:54.03vk4akpIs asterisk supposed to come with a script so I can do a rc-update to make it start at boot?
18:55.22*** part/#asterisk purple_v45 (n=archway@71-91-227-114.static.stls.mo.charter.com)
18:55.49Yourname[TK]D-Fender one sec
18:56.02jayteevk4akp, what distro are you running?
18:59.08Yourname[TK]D-Fender: http://pastebin.ca/1196622 -> I do database show and I see the CFIM in there now. However, when 121 is called, it still rings the phone where 121 is registered instead of forwarding it..
19:00.42vk4akpSabayon 3.5 Linux, Asterisk SVN 1.4xxxx
19:01.29vk4akpAsterisk SVN-branch-1.4-r135058M built by root @ localhost on a i686 running Linux on 2008-08-07 11:47:20 UTC
19:01.56RypPnvk4akp: no, add it to /etc/conf.d/local.start
19:02.04*** join/#asterisk jpastore (n=jpastore@69.65.65.40)
19:02.20vk4akpOK thanks.
19:04.00[TK]D-FenderYourname: Congratulations, you set up a framework to set and remove AstDB flags to indicate a status concerning  certain CID's related to other extensions.
19:04.16vk4akpIs there any scripts around for init.d ? I am thinking that maybe I should be doing it that way as I also need to start things like ZTCFG etc as well.
19:05.06[TK]D-Fendervk4akp: under the contrib folder.  Go look.
19:05.19vk4akpcontrib folder?
19:05.43*** join/#asterisk Xaviertoor (n=Xavierto@189-015-150-089.xd-dynamic.ctbcnetsuper.com.br)
19:05.45[TK]D-Fendervk4akp: Yes.
19:05.55Yourname[TK]D-Fender: For some reason wat you just said went straight over my head, lol
19:05.56vk4akpLOL funny. Where / what is that?
19:06.10Yournamevk4akp: Download asterisk to your computer, extract it, and you'll see it there.
19:06.12[TK]D-Fendervk4akp: Where have you considered looking for it.
19:06.30*** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net)
19:06.33vk4akpSo /usr/src/asterisk... ?
19:06.42[TK]D-FenderYourname: let me paraphrase : those AstDB values don't mean ANYTHING because you aren't USING THEM.
19:07.03[TK]D-Fendervk4akp: You're ASKING me where you considered looking?  Go show me some real thought or effort.
19:07.21Yourname[TK]D-Fender: ah? I thought the db automatically does something about it. :S Voipinfo's article didn't really mention anything about trying to use it explicitly..
19:07.34[TK]D-FenderYourname: Rand DB values mean nothing
19:07.47vk4akpThats what I love about you guys. Alwyas so friendly and caring! ;)
19:07.56[TK]D-FenderYourname: And now I know why that code looked familiar... Mr. Cut & Paste
19:08.12Yournamelolol
19:08.36tzafrir_laptopvk4akp, ztcfg is run from the init.d script in the zaptel package
19:08.38vk4akprc.gentoo.asterisk    <<--- Looks like it might be the go ?¿? :)
19:08.43Yourname[TK]D-Fender: Actually, after using Example 3, I had to change DrPut to Set(DB(.. and {CALLERID(num)}
19:09.03Yourname[TK]D-Fender: Then, I just saw the "comments" part, and it was done with something else and nicely formatted, so used that one.
19:11.30vk4akpUmm, I don't see any init.d in /usr/src/zaptel
19:12.07*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
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19:17.30*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
19:17.47vk4akpWhere should I be looking to find the init.d stuff for zaptel ?
19:17.56rue_mohr/etc/init.d
19:18.00rue_mohrits a folder
19:18.09rue_mohrthats the only init.d I'm aware of
19:18.16vk4akpYea it's not there.
19:18.24vk4akpMaybe I need to do a make config.
19:18.27rue_mohrso why are you looking for it?
19:18.34vk4akpBut if I do that it will scrub my existing configs hey?
19:18.37rue_mohrdo you have a directory called /etc/init.d?
19:20.11vk4akpIts ok.
19:20.17vk4akpIt sits here.   /etc/rc.d/init.d/zaptel
19:20.21*** part/#asterisk danalien (n=danalien@unaffiliated/danalien)
19:20.25vk4akpI dunno what the difference is but eyp.
19:21.04riddleboxis this a correct way to handle calls from callerid?
19:21.06riddleboxexten => s/7135551212,1,Goto(fax-ext,s,1)
19:21.48*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
19:22.05heedlyI don't think so :\
19:22.09rue_mohrvk4akp, /etc/init.d is the directory with all the programs that are run on startup
19:22.23heedlyriddlebox: what do you expect to accomplish with that?
19:22.28riddleboxheedly, you talking to me?
19:22.29vk4akpwhats rc.d  ?
19:23.01rue_mohrthe programs are run from a set of different directories that have a selection of startup programs based on the systems runlevel
19:23.15riddleboxheedly, I was reading through the forums and saw someone post that as a solution for someone else, and I had never seen that way of routing calls from caller id
19:23.38rue_mohrso /etc/rc2.d is full of the programs used when the system is started in runlevel 2
19:24.01*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
19:24.05outtoluncriddlebox:  it is the example of 'ex-girlfriend'
19:25.04vk4akpOK. So my rc-update line would not go to default for that then?
19:25.22riddleboxouttolunc, it is a clean way to do it, but I havent seen that way, is it in the updated book?
19:25.41vk4akpOr will the asterisk init.d run the zaptel startup for me?
19:25.43heedlyriddlebox: just 7135551212,1,Goto(fax-ext,s,1) seems like a better way.
19:25.55riddleboxheedly, thats what I thought
19:25.58heedlychanging fax-ext to what ever you want to jump to.
19:26.19[TK]D-Fenderriddlebox: Your way is perfectly valid.
19:27.15riddlebox[TK]D-Fender, it seems cleaner and easier to do it that way
19:27.37[TK]D-Fenderriddlebox: the best way depends on how much you need to do.
19:27.50[TK]D-Fenderriddlebox: Good for one scale, bad for another.
19:28.10riddlebox[TK]D-Fender, true, I was just reading the forums and saw that example in a post and was wondering about it
19:29.39outtoluncriddlebox: search 'ex-girlfriend' (you should see it was even in the first handbook-draft)
19:30.15[TK]D-Fenderouttolunc: EEK, don't even refer to that archaric crap...
19:30.35riddleboxouttolunc, I used it along time ago to keep my mom from calling me so early on weekends
19:36.49*** part/#asterisk jpastore (n=jpastore@69.65.65.40)
19:40.26*** join/#asterisk zamba (i=marius@sveigde.hih.no)
19:43.15*** join/#asterisk ManxPower (n=manxpowe@44.sub-75-203-133.myvzw.com)
19:44.57*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
19:44.59*** join/#asterisk blaylock (n=seth@c-68-57-177-235.hsd1.va.comcast.net)
19:51.25Wayhighgets grandcentral workin to his dialin
19:53.26riddleboxWayhigh, how are you using it?
19:56.04*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
19:56.04*** mode/#asterisk [+o lmadsen] by ChanServ
19:56.56Wayhighriddlebox: you know.. free calls from my folks ratecenter to my home..
19:56.59ManxPower"Wayhigh" sounds like a Radical Faerie name.
19:57.26ManxPowerI know one named "Change" and one named "By The Way" (yes, that's why you call him)
19:58.28drmessanoGrandcentral needs to dump the dial-1 thing
19:58.43riddleboxWayhigh, howd you get it working though, cause I just tell people to call my grandcentral number which has my cell and home number in it?
19:58.45WayhighRadical Faerie? Is that like a faerie that wants to secede from faeriedom by means of force?
19:59.14Wayhighriddlebox: passed the GC number to ipkall which is pointed to my sip number
19:59.29drmessanoI think ManxPower means a metrosexual man that wears white after labor day
20:00.25ManxPowerdrmessano: not even close.
20:00.56ManxPowerWayhigh: http://www.radfae.org/
20:01.54Wayhighahh interesting.. it's apparently a group of gay folk that want to redefine gay identity so they don't need to fit the stereotypical heterosexual role in order to fit into the culture that we live in. In essence, they want to redefine our culture itself.
20:02.39WayhighI'm actually totally cool with them redefining our culture so that they're accepted however they want to live as long as it isn't hurting another.
20:03.01drmessanoLooks like a cult to me
20:03.36drmessanoThis is why I hate Kool-Aid
20:03.43drmessanoCults ruined it for everyone
20:04.40Wayhighthen again.. you can always start a cult.. call it a religion.. and get wealthy like those folk from scientology
20:05.13heedlyhmm, how does gays show a stereotypical heterosexual role...
20:05.56Wayhighheedly: I think the idea is that if they don't act heterosexual then they are rejected by a large percentage of our society
20:06.40WayhighI'm hoping that percentage gets smaller as people go silently into the night
20:07.39*** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net)
20:09.22mvanbaak_I want ppl to forget about the 'difference' between hetrosexual and homosexual
20:09.23ManxPowerheedly: think "gay marriage".  There is nothing more hetrosexual than marriage.
20:09.30mvanbaak_they are all people
20:09.40ManxPowerthe whole monogamy thing as well.
20:09.48mvanbaak_ManxPower: here in .nl marriage is not hetrosexual
20:10.06mvanbaak_gay, lesbian, hetro, bisexual, whatever
20:10.13mvanbaak_you can marry the guy/gal you want
20:10.24carrarNice Wayhigh, look what you've started!!
20:10.39drmessanoMarriage is stupid anyway.. monogamy would actually lower the population
20:10.45drmessanoerr
20:10.49drmessanolack of..
20:11.01WayhighPersonally, I like how the thai people look at things..
20:11.10mvanbaak_drmessano: gheh, I'm married, but there will be totally no kids in my life
20:11.13ManxPowerI never could understand monogamists.
20:11.22drmessanoMost people make babies because they do what they need to do to have sex.. if we had sex all the time, there would be no need to do it to make babies, hence less kids
20:11.34heedlyManxPower: how many people are you with now?
20:11.42ManxPowerheedly: two
20:11.47heedlyright on ;)
20:11.55heedlyleft and right?
20:11.56heedlyhaha
20:12.33mvanbaak_drmessano: having sex has nothing to do with creating babies
20:12.51mvanbaak_drmessano: it's a 15 minute meeting in the hospital and you can never create kids anymore
20:12.56mvanbaak_it's brilliant
20:13.08WimpManreally hates modern medicine
20:13.27WimpManOh, that way round
20:13.47ManxPowermvanbaak_: I live in the USA, have visited Europe.  Most USAians don't really understand just how *DIFFERENT* European / Country culture is from USA culture.
20:14.06Wayhighit wasn't that long ago that the general thinking in the anglo-saxon world was that sex for the purpose of having sex was wrong
20:14.09drmessanomvanbaak_: Sex has everything to do with making babies when a lot of guys are stuck with women that think that's all it's good for.. So the guy does hat he has to do..
20:14.23mvanbaak_ManxPower: that's because most USAians think they own the truth
20:14.31mvanbaak_:)
20:14.42ManxPowerI was told that if a Belgian politician ended a speech with "God Bless!", he would be voted out of office.
20:14.45Wayhighmanx: I'm all kinds of down with swedish culture as commented on by Manswers
20:14.58mvanbaak_ManxPower: yes
20:15.09ManxPowerof course "in god we trust" was only added in the 1950's to use currency -- I believe as a response to the Godless Communists.
20:15.17ManxPoweruse == USA
20:15.35mvanbaak_ManxPower: here in .eu we try to do what we think is best. and rule #1 is politics and religion should never be mixed
20:15.47ManxPowerA gay man will be elected to the USA presidency before an Atheist will.
20:15.59mvanbaak_a politician should act on behalf of the people, and not let religion guide him/her
20:16.19heedlymvanbaak_: what if religion guides the people?
20:16.26ManxPowermvanbaak_: odd how USAians seem to ignore that, even though it is part of the constitution.
20:16.28mvanbaak_heedly: that's impossible
20:16.35heedlyoh?
20:16.47mvanbaak_heedly: there is no global religion
20:16.49heedlymany people let religion lead their lives.
20:16.55heedlythre is no global politican..
20:16.56ManxPowerheedly: The LAW should guide politicians.
20:16.57mvanbaak_heedly: everyone has their own opinion.
20:17.20heedlyI'm saying if >50% of a politican supports are religious.
20:17.23mvanbaak_heedly: that's why the ppl vote who they want to be their politician
20:17.36heedlyTheir majority "opinions" are based on religion.
20:17.44mvanbaak_heedly: that 50% will never have the same religion
20:18.09mvanbaak_heedly: it will be a mix of christianity, hindoism, islam, atheism etc
20:18.10ManxPowermvanbaak_: they would all be christian
20:18.14heedlywhat about the majority group then?
20:18.20mvanbaak_ManxPower: not here in .nl
20:18.37ManxPowermvanbaak_: non-christians are a tiny part of the population in the USA.
20:18.37mvanbaak_ManxPower: here in .nl 65% of the ppl are athiests
20:18.42heedlyruling by religion is no more worse or good that ruling by any other conviction.
20:18.48ManxPowermvanbaak_: sounds WONDERFUL
20:18.56mvanbaak_ManxPower: it sure is
20:19.13heedlyno if you could only find a place were no one cared about anything!
20:19.21heedlynow that would be wonderful.
20:19.26Qwelllike .nl
20:19.27mvanbaak_ManxPower: religion is opium for the masses. it's a louzy excuse to not be responsible for your own acts
20:19.34heedlyno .nl cares about people caring.
20:19.49ManxPowermvanbaak_: that statement could get you killed in some of the more rural parts of the USA
20:19.52heedlythose are the worse kind!
20:19.59mvanbaak_if you f*cked up you simply blame god
20:19.59ManxPower(I live near one of those areas)
20:20.37mvanbaak_ManxPower: I live in a very religious part of .nl. I see it happen every weekend here
20:21.07heedlymvanbaak_: I think they usually ask forgivness before they blame him.
20:21.15mvanbaak_young guys driving in a car with shitloads of beer in their system. they trash their car and die. and the paper tells you the next monday: "God has taken another soul"
20:21.19mvanbaak_that's just bullshit
20:21.42mvanbaak_I mean, as if God ordered this guy to drink all that beer and take the car and speed into a tree
20:21.47ManxPowermvanbaak_: I grew up in Holland, Michigan USA, that area of the USA was settled by people escaping religions prosecution in the Netherlands.   There's a church on every corner.   Pretty typical for many parts of the USA.
20:21.52heedlylol. you use an extreme case to insult extremists?
20:22.22mvanbaak_heedly: it's not extreme. It's like that in every part of .nl where the religious ppl gather
20:22.40heedlymvanbaak_: that sounds a bit stereotypical to me.
20:22.53heedlybut I have no facts or experience to deny it.
20:22.56mvanbaak_heedly: you should come live in holland for a couple of years
20:23.04jaytee"Have you found Jesus?" "Um, no. I didn't know he was lost and that I was supposed to be looking for him" is my favorite answer.
20:23.10mvanbaak_heedly: .nl is very small.
20:23.13heedlyThat's always peoples excuse.
20:23.14heedlylive here.
20:23.24ManxPowerI like the bumper sticker I saw.  "Got Religion?  Keep it to yourself!"
20:23.33heedlyI've live all over.. and it ultimate comes down to people are stupid and self centered.
20:23.36heedlyeverywere!
20:23.44mvanbaak_ManxPower: you know those silver fish stickers they put on cars ?
20:23.55ManxPowermvanbaak_: yes.
20:24.11mvanbaak_ManxPower: here in .nl we have those, with a grill under it ;)
20:24.14WayhighI'm all for people having religion.. I just think they people need to come up with their ideas/beliefs without being subject to outside forces like evangelism
20:24.15heedlymvanbaak_: on another note, how are the greens thar?
20:24.38WayhighIt's cool to tell someone about a religion.. not so cool to tell them they're going to hell if that's not the one they believe in
20:24.58ManxPowerWayhigh: but it would not be religion if that didnt happen
20:25.17mvanbaak_hhmm, is this about chan_religion.so ?
20:25.29jayteeexcuse my language please but I am so tired of these retarded "fuckwits" blaming all of America's problems on gays and lesbians.
20:25.57mvanbaak_Dial(Religion/<your_god>)
20:26.10mvanbaak_where <your_god> can be any god depending on your religion
20:26.29ManxPowerjaytee: before that it was communism, before that it was african american, before that the Catholics, before that the Jews.
20:26.40mvanbaak_in my case that will be: Dial(Religion/mvanbaak)
20:26.58mvanbaak_ManxPower: well, blaming the jews makes sense ;)
20:27.00mvanbaak_hides
20:27.10jayteeand the Westboro Baptist Church from Kansas is a perfect example of that. Protesting at the funerals of soldiers killed in Iraq because of their extreme homophobia is just so against any of the moral values I was raised with it makes me angry and nauseous at the same time.
20:27.10*** join/#asterisk Assid (n=assid@unaffiliated/assid)
20:27.12Assidhi
20:27.27Assidanyone know any decent providers around 1.2-1.5c/min for US48 ?
20:27.47ManxPower~itsp-us
20:28.00Assiditsp-us ?
20:28.04mvanbaak_teliax ?
20:28.09mvanbaak_I have no idea
20:28.23Assidteliax any good?
20:28.30Qwell~itsplist-us
20:28.30jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
20:29.10Assidaah thanks
20:29.32ManxPowerQwell: Your jbot foo is strong.
20:30.00ManxPowerI guess that would be jbot-fu
20:30.27jaytee"hot pockets and a supply of Xena tapes"
20:30.41mvanbaak_ha
20:30.58mvanbaak_teliax is the one I remembered, and now I see it's the most respected one
20:31.12mvanbaak_I'm going to try voipbuster again
20:31.14ManxPowerthey are not in "respect" order
20:31.22mvanbaak_you can laugh at me already, I know
20:31.24ManxPowerTeliax has a horrid verification process.
20:32.29AssidQwell: you know any good white labelled providers who can do number portability and give channel based unlimited incoming
20:32.40Assidlike voicepulse connect.. except white labelled
20:33.15jayteevoicepulse does both sip and iax termination
20:33.18WayhighI like voicepulse connect but their charges per month for a DID are ridiculous
20:33.56Assid11 bucks
20:34.06ManxPower$1.49/$1.95 a month (depending on the rate center) for a DID
20:34.18ManxPowerfrom Vitelity
20:34.34ManxPowerAssid: It MIGHT cost the ITSP 5/cents/min/number
20:34.38ManxPower..er..
20:34.48ManxPower5/cents/month per number
20:34.54Wayhighmanx: yeah.. my issue with vitelity is they made it impossible for me to tell that the ratecenter I wanted was backordered until I'd allowed them to charge me $35
20:35.02ManxPowerso charging $11 is quite a markup.
20:35.47jayteestill better than what most local telcos charge for DID
20:35.57Assidokay so vitelity not good
20:36.07Assidteliax is coming expensive
20:36.27Assid1.88c/min
20:36.35Assidneed it around 1.2-1.5
20:36.45jayteebut how do you determine the actual cost unless you know how reliable the call delivery is?
20:36.53ManxPowerjaytee: Huh?  we get 100 DIDs for $5/month for the entire block and we are not even a carrier
20:37.35AssidManxPower: dont you have to pay per min after that?
20:37.43ManxPowerAssid: no.
20:38.04Assid100 dids for $5 ???and unlimited incoming?
20:38.25ManxPowerWe use both AT&T (formerly BellSouth) and XFone (a regional carrier), have the numbers terminate on our PRI and we are DONE.
20:38.33Strom_Mboy, I remember when we paid 10c per minute and thought it was the biggest bargain in the universe
20:38.56drmessanoor 7 cents!
20:39.08ManxPowerAssid: VoIP is really the only telecom industry that charges for incoming calls.
20:39.22ManxPower(and cell phones in the USA, not most of the world)
20:39.42ManxPowermost of the world has free incoming cell calls
20:39.45Assidyeah.. i know.. we got unlimited incomign here on cell phones as well
20:40.09jayteeI remember reading about a woman in Maine that had been leasing her phone for like 3.95 a month for the last 35 years or so from her local telco. it was an old rotary dial phone even. Her son finally realized what was going on and cancelled the phone lease and got her a new phone.
20:40.11ManxPowerI think we have 200 DIDs on the PRI at HQ (only one PRI)
20:40.24jeevManxPower, friend went to office and called me
20:40.27jeevquality was excellent.
20:40.32jayteeyay!
20:40.34ManxPowerjeev: aresome!
20:40.34jeevso far so good
20:40.39AssidManxPower: sweet.. how many lines can your pri handle?
20:40.40jeevwe'll see what happens
20:40.44jeevso IAX just sucks or what ?
20:40.46ManxPowerjeev: all the real hard problems only happen under load.
20:40.51jeevyea
20:41.02ManxPowerjeev: nope.  Some people just have problems with it.
20:41.05jeevahh k
20:41.08jayteeManxPower, you should get more karma for that suggestion for GRE tunnelling
20:41.15ManxPowerI never understood what caused it, but I don't care, I needed it fixed.
20:41.21jeevgre tunneling was the coolest thing ever
20:41.25jayteehow do people accumulate karma in here anyways?
20:41.52ManxPowerpeople are always to send a paypal donation to the ManxPower Drinking Fund to eric@fnords.org
20:42.06*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
20:42.08jayteelol
20:42.20Assidokay so who else gives calls for 1.2-1.5
20:42.22ManxPowerjeev: I do FAR more networking consulting that I ever do Asterisk consulting
20:42.46Assidi need it for a business+call centre.. but none of those annoying ones that call you for stuff.. its more on the lines of people calling you for your flight tickets
20:42.47jeevah
20:42.55jeevnext time i donate, will be yours
20:43.11jeevregrets sending the case of redbull to digium.... or at least not adding anthrax to Qwell's.
20:43.18ManxPowerI'd starve if I only did Asterisk consulting
20:43.25jayteeManxPower, you mean you do far more network consulting FOR MONEY than you do Asterisk consulting. Cuz you're here helping just about every single day.
20:43.37ManxPowerjaytee: that is correct.
20:43.52jeevwtf man
20:43.53ManxPowerI'm outta here for a while
20:43.54Assidhrmm
20:43.55jeevmy girlfriend is coming to pick me up
20:43.56jeevshes like
20:44.00jeev'look cute'
20:44.04Assidlook cute?
20:44.08jeevyea
20:44.09Assidhow thehell do you look cute?
20:44.12jeevi dunno
20:44.17jeevi'm the 3rd hottest guy in the world...
20:44.19Assidkeep a teddy bear next to you
20:44.19jeevwhy do i have to look cute
20:44.29Assidwhat freaking mirror do you own?!?!?
20:44.34Assidthat gives you those stats
20:44.39ManxPowerAwayjeev: you are meeting some of her friends and she wants them to be jealous.
20:44.50jeevManxPowerAway, they already are.. believe me
20:44.54jeevone even stopped talking to her hahaha
20:44.59jeevcause she couldn't stand that she wasn't with me!
20:45.05jeevna, i sometimes leave the house looking homeless
20:45.10jeevi was wearing my friends logo shirt
20:45.11ManxPowerAwaypeople like that should be shot.
20:45.12jeevwent to olive garden
20:45.16jayteejeev, stick a rolled up pair of socks down your pants and comb your hair like a "bad boy" :-)
20:45.24jeevthe guy is like, "do you work at the W hotel?" i'm like "no, why?" he's like "your shirt"..
20:45.30jeevso then i paid with my american express centurion card
20:45.36jeevand he came back and said, "you could kill someone with this"
20:45.47Assidokay im finally gonna start getting paid !! yeay!
20:45.50jeevjaytee, i barely have hair, i always shave it
20:45.53Assidwhere voip is concerned atleast
20:46.23jayteejeev, sorry but I'm a bit slow today, what's the kill someone with this about with the card?
20:46.51jeevyea
20:46.55jeevthe black card is titanium
20:47.00jayteeah!
20:47.05jayteecool
20:47.34*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)
20:49.00Assidis there some kinda 14.2% tax or something
20:49.01Assidon voip?
20:51.40jeevok i gotta go
20:51.41jeevlol there is ?
20:51.43jeevbbiab
21:03.28*** join/#asterisk F00JIN (n=F00JIN@lns-bzn-56-82-255-222-4.adsl.proxad.net)
21:03.33F00JINhi!
21:04.04F00JINthis time my asterisk 1.6 and gui 2.0 are running
21:04.17ManxPowerAwaycreams in horror
21:04.21ManxPowerAway*sigh*.
21:04.26ManxPowerAwayscreams in horror.
21:06.58F00JINi'd like to have tips because i want to use ldap with asterisk
21:07.41*** join/#asterisk tuxd00d (n=tuxd00d@128.187.129.239)
21:09.28F00JINI'm looking for a tuto to explain ldap integration
21:13.12*** join/#asterisk Levonk (n=lk@adsl-76-230-110-253.dsl.lsan03.sbcglobal.net)
21:13.34[TK]D-FenderF00JIN: LDAP to do WHAT?
21:14.09F00JINto list users of asterisk
21:14.16Yournameexten => s,n,GotoIf($["${CFIM}"!=""]?s-CFIM,1:s-NoCFIM,1) -> Does this look normal for 1.4? because for some reason it's going to NoCFIM rather than CFIM.
21:14.33YournameThe square brackets weren't there before and it still didn't work
21:16.05ManxPowerAwayYourname: as the priority above that one add a Noop(CFIM is ${CFIM}) and make sure it was not screwed up at some point.
21:17.07ManxPowerAwayI thought spaces were optional around operators in 1.4.  try adding a space around the !=
21:18.15[TK]D-FenderYourname: Maybe you show us what you think that var HOLDS
21:18.41[TK]D-FenderYourname: Maybe that line is fine and the entire rest of your setup is trash
21:18.51[TK]D-FenderYourname: So how about backing it up?
21:19.15[TK]D-FenderF00JIN: go read teh realtime chapter in the BOOK
21:19.16[TK]D-Fender~book
21:19.17jbothmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
21:20.13Yournameok
21:21.13YournameCFIM does show up though, which is why it's weird. Since even though CFIM is present, it does the GotoIf to NoCFIM
21:21.41[TK]D-FenderYourname: PASTEBIN
21:23.11Yourname[TK]D-Fender: http://pastebin.ca/1196730
21:23.47[TK]D-FenderYourname: And the call?
21:24.22[TK]D-FenderYourname: And I already see serious errors in there
21:25.08[TK]D-FenderYourname: But do continue
21:26.01Yourname[TK]D-Fender: http://pastebin.ca/1196727
21:29.11*** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org)
21:29.15[TK]D-FenderYourname:  -- Executing [s@macro-stdexten:4] GotoIf("SIP/63.211.239.18-08db5b90", "$""!=""?s-NoCFIM|1") in new stack <- so nothing strikes you as wrong with this line?
21:29.46[TK]D-FenderYourname: exten => s,n,GotoIf($"${vmbox}"!=""?s-NoCFIM,1) <- or this dialplan line that generated that?
21:30.03ManxPowerAwayHe LIED to US?
21:30.12YournameManxPowerAway no lol!
21:30.19ManxPowerAwayIt's obvious he retyped the line rather than copy and paste.
21:30.28Yourname[TK]D-Fender: I'm making puppy eyes right now as in I don't know.. :S
21:30.43ManxPowerAway"$""!=""?s- that should not show up, you screwed up your BRACKETS
21:31.10YournameActually, frig.. I know what you guys are talking about! There's two GotoIfs in there..
21:31.42[TK]D-FenderYourname: No, its that you wouldn't seem to know how to format an expression if it ran up bit you in the ass :p
21:32.08YournameOk, so if I change exten => s,n,GotoIf($"${vmbox}"!=""?s-NoCFIM,1) to exten => s,n,GotoIf($["${vmbox}"!=""]?s-NoCFIM,1) would be nice ? :$
21:32.16Yourname[TK]D-Fender: shut it! I'm learning ok! :P
21:32.24ManxPowerAwayYourname:  a mistake like that in a production system could open up a major security hole in your system.
21:33.41[TK]D-FenderManxPowerAway: Guess I won't say anything about his giving arbitrary transfer rights....
21:33.44F00JINI've found what I want thx [TK]D-Fender
21:34.51ManxPowerAwayexten => _NXXNXXXXXX,1,GotoIf($"${AUTHENTICATED}!=""?auth,1)
21:35.43ManxPowerAway[TK]D-Fender: you gotta wonder how many Asterisk PBXs out there let toll calls be dialed from IVRs?
21:35.48[TK]D-FenderManxPowerAway: not to meantion the call itself isn't auth'd
21:35.51ManxPowerAway(unintentionally)
21:39.18YournameProbably because most people such as I copy/paste from the Wiki :(
21:41.06[TK]D-FenderYourname: What scares me is you do this for your job....
21:42.07Yourname[TK]D-Fender: It's not my job. :)
21:45.16*** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net)
21:45.43YournameShit, you just kinda gave me the shudders with that thought. What _IF_ it was?! lol
21:46.45*** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net)
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21:52.33DanskmandHowdy :-) - O know this channel is about asterisk, but maybe you know if capisuite works together with mISDN ?
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22:18.23Wayhighanyone here have a voice T1 or PRI that you mind telling me the cost of?
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22:42.37EI5GTBok guys, recommendations on a softphone for linux
22:44.03mchouEI5GTB: dont use softphone on any os
22:44.26EI5GTBoh....
22:44.31EI5GTBy?
22:44.32mchoubut if you must use one, twinke, ekiga, maybe even xlite
22:44.50mchouEI5GTB: softphones suck in general
22:45.13EI5GTBi see... its just for me to avoid paying lots of money for lots of hardware while i play with asterisk
22:45.44mchouheadphone, mic, computer & phone calls dont translate into convienience or  ease of use
22:46.10mchouEI5GTB: for "prrof of concept" softphones are OK
22:46.11EI5GTBactually.. in my case it would..
22:46.29EI5GTBim sitting beside 3 computers with a headset on whenever im here
22:46.31mchouotherwise invest in decent HW
22:47.00EI5GTBi have a system so that when i have a phonecall, i get my radio audio in one ear, and phone in the other
22:47.02mchouEI5GTB: I sit nect to 4 comps and I still use a regular phone
22:47.04mchou:)
22:47.07EI5GTB:P
22:47.09EI5GTBi know
22:47.17EI5GTBdepends if im at myradio bench or not
22:47.35mchouI think pap2s are "good enough"
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22:47.53mchouif you want a fancy phone the go polycom
22:48.06mchouall depends on what you want
22:48.13EI5GTByea
22:48.20EI5GTBi like things o look cool with lots of buttons
22:48.36mchoupap2s arent expensive
22:49.04mchouand they work sufficiently well except for some corner cases like MWI
22:49.16EI5GTBheh
22:49.21mchouwhich has some issue I havent figured out
22:50.12mchouin a pinch you can also go for the vonage/dlink vta/vr
22:50.32mchoubut I think that way sub par compared to pap2
22:50.40EI5GTByea
22:50.52mchouI've seen vonage vta-vr on clearance for $20
22:51.00EI5GTBanother reason for using a softphone is i have a 15" touchscreen here
22:51.16mchouforget the ouch screen man :)
22:51.22mchoutouch*
22:51.42mchouany modern decent phone has lcd and whatnot
22:51.58EI5GTByea
22:51.59mchouyou want touch screen go get an iphone :)
22:52.03EI5GTBbut touch screens are fun
22:52.11EI5GTBi dont want an i-phone
22:52.29mchouiphones are more phone than any of your touch screens
22:52.31mchou:)
22:52.44mchouor get a wii remote control
22:52.56mchoulemme find linky......
22:53.00EI5GTBi saw it
22:53.00EI5GTBsok
22:53.19EI5GTBi dont want to wear ir lights P
22:53.22mchouEI5GTB: you know what I'm talking about with wii??
22:53.24EI5GTBor reflective tape on my hands
22:53.27EI5GTByea
22:53.31EI5GTBthe chinese guy?
22:53.38mchoukorean, pls
22:53.54EI5GTBah, is that what he was?
22:53.56mchounot all friging asians look alike, you know :)
22:54.00EI5GTBbeen a while since i watched it
22:54.04mchoufriiging*
22:54.14mchoufrigging**
22:54.39DanskmandO.k.....let me ask in a different way.....Does misdn work together with asterisk, substituting I4L ?
22:54.41EI5GTB3rd time lucky :P
22:57.06mchouEI5GTB: keep in mind softphones have other drawbacks besides inconvenience.  like crappy sound quality and whatnot
22:57.19sakajawebeI4L ?
22:58.39DanskmandIsdn 4 (for) linux
23:00.00sakajawebeso are you asking then "does misdn replace I4L in asterisk" ?
23:00.15[TK]D-FenderIt doesn't
23:00.22[TK]D-FenderBecause I4L has no place in Asterisk
23:00.45[TK]D-FenderNow go look to see if your card IS supported by and ISDN interface written for *
23:01.16DanskmandWell, its a passive Fritzcard
23:01.27DanskmandI will look...
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23:10.08edwin_quijadasomebody has had any issue with Postgres and Postgres as CDR?
23:11.24jeevfender, know of a linux/bsd solution for cRTP ?
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