IRC log for #asterisk on 20070701

00:02.28[TK]D-Fender~ygwypf
00:02.29jbotmethinks ygwypf is You Get What You Pay For.  If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.
00:02.33ovnicraftbut ti will work or not
00:02.43[TK]D-Fenderovnicraft, You you POSSESS it?
00:02.52[TK]D-FenderDo*
00:05.07*** join/#asterisk fnordus (n=dnall@24.84.160.227)
00:07.06ovnicraftNO i want to buy one,  and i wnat to know if is a good idea
00:07.32crimethinkerNo, it's not a good idea.  Buy a tdm11b instead.
00:33.38*** join/#asterisk javar (n=javar@69.79.134.24)
00:42.07nohopi just just found out something strange... i was looking at firewalling options and stuff, because on outbound calls the 'other' side couldn't hear me (i'm talking SIP phone here)... now i just found out there IS sound for about half a second before it dies...
00:56.29nohop... and now it actually worked for a couple of seconds before dying... which doesn't happen if i connect to my the sip server directly with my softphone, but does when asterisk is 'in between' then
00:56.31nohopthem
01:03.02*** join/#asterisk asdx (n=diego@adsl-133-38.click.com.py)
01:05.30asdxdamn, me and many people cant use voip, etc in this country :/  http://www.google.com/translate?u=http%3A%2F%2Fanteriores.lanacion.com.py%2Fnoticia-162455-2007%2F06%2F26.htm&langpair=es%7Cen&hl=en&ie=UTF8
01:06.13asdxis there a way to by-pass that?
01:06.24crimethinkersatellite
01:09.04nohopheh..
01:09.08nohopmy provider does that too
01:09.24nohopthat's why i'm routing my SIP packages over a pptp tunnel
01:14.08asdxi heard you can also route the packages to a vpn tunel, will that work?
01:14.21asdxtunnel*
01:14.53dc3aeswhat really bites the most w/ ISP/SIP/IAX blocking issues.. is the fact that I actually have to wonder if im being blocked, and that if I were I would not be surprised.. sort of sick
01:15.13dc3aesrather "interfered with" than blocked I should clarify
01:15.15asdxyeah
01:15.53asdxthe thing is that all isps goes throw a big telephone company here, and the telephone company is the one that blocks everything... so all the isps can't use SIP/IAX, etc
01:16.19dc3aesive got a pretty tight setup here w/ 8mbps broadband with the last docsis modem, new house, commercial area.. and the conversations are all good but ever 20 minutes or so it goes to hell for a few minutes.. so IAX outbound calls cannot be made professionaly
01:16.51dc3aesasdx, a vpn tunnel is effectively a pptp... in fact pptp = type of vpn
01:17.16dc3aesvpn can be encrypted.. most people assume that if its a vpn, its encrypted.. I could be wrong on this but from what I understand, vpn does not imply encryption
01:17.32dc3aesat least they couldnt analyze packet headers, and detect encapsulated SIP/IAX traffic
01:18.15nohopbut you'd need some point to tunnel to
01:18.33dc3aesour broadband company (Shaw Cable western canada) has a professional internet telephone product.. it has its own modem, backup battery, etc.. in it.  It plugs directly into the COAX instead of your LAN.. they have mental control over their own telephony now
01:19.07dc3aesnohop, i have a rack server in a datacenter with huge bandwidth.. im considering having it as an endpoint, at least for experimenting to see if its not just 'voip being tampered with'
01:19.15dc3aesthey screw with our torrents, this is confirmed 100%
01:19.52nohopahhh, i fixed my no-sound problem
01:19.54*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
01:19.59crimethinkerand people mock me for purchasing commercial internet service in my house.
01:20.27dc3aeshehe i would if i could afford it
01:20.46crimethinkersounds like you afforded a new house.
01:20.50asdxdc3aes: i see
01:20.53dc3aesim renting a "newer" house :P
01:21.01crimethinkerk
01:21.04dc3aesthat point was only to imply new wiring, etc..
01:23.01dc3aesi almost wonder if the fed govt. doesnt have a program in place for archiving sip traffic already for echelon-like reasons lol
01:24.30*** join/#asterisk saftsack (n=saftsack@pD9E07657.dip.t-dialin.net)
01:25.57polerindc3aes: if nothing else I bet they are doing basic metadata stuff :)
01:26.03nohopi signed up with secureix... they do tunneling for pretty little money
01:26.14*** join/#asterisk codejunky (n=jan@e177229148.adsl.alicedsl.de)
01:27.12dc3aesnohop, any indication as to its effect?
01:27.30nohopwhat do you mean by that /
01:28.18dc3aeswell.. have you had a chance to test it, and did it dramatically change call quality?
01:28.59nohopno, cause my ISP actually only blocks the SIP packets...
01:29.25nohopso i only route THOSE through the tunnel... the actual audio still uses my 'default gateway'
01:30.01dc3aesahh..
01:30.12dc3aeswild
01:30.29nohopwas some fiddling around to get that working with asterisk, but that's fine now...
01:30.51dc3aesits just amazing.. its like a car company selling you a car and saying well.. you cant drive it on the following roads, because well we operate a train station there so you should instead ride our trains hahaha
01:31.00nohophow do i tell asterisk that i want my SIP phones to authenticate to my asterisk server in order to let them be able to make outbound calls ?
01:31.09nohopdc3aes: exactly...
01:31.26nohopi called my isp about it... and they just DENIED they were blocking any ports...
01:31.27dc3aesnohop, no idea on the sip authentication yet ..
01:31.40nohopi was like "but... i have nmap output saying FILTERED here, right in front of me!"
01:31.56dc3aeswhere is this nohop?
01:31.56nohopand he just said it's not true, cause he asked he co-worker, and he said so. period.
01:32.07nohopi'm in holland
01:32.13nohopwhere shit like that should NOT happen :)
01:43.58`Seanguys is there any reason i should be getting this error because of Jun 30 21:44:52 NOTICE[19748]: chan_sip.c:3716 process_sdp: No compatible codecs!
01:46.34*** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
01:46.46*** join/#asterisk OloBola (n=not@74.95.13.57)
01:46.59rvhi0anyone uses RAGI? ruby agi? seems interesting, but not updated for more than 2 years
01:52.40mihinomenestthat seems pretty common.
01:54.15*** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com)
01:56.14*** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net)
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01:57.45polerinhmm
01:57.49nohopif i want my sip phones to authenticate... would that mean that i would have to declare each one of them once as peer, and once as user? instead of just friend ?
02:06.45*** join/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg)
02:07.30littleballhello, i try to destroy a zap channel. example, after i run zap destroy channel 63, how to get it back?
02:08.48JTThoMe: yes, bristuff is isdn bri, exactly what it sounds like
02:08.56*** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com)
02:09.30*** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
02:13.22*** join/#asterisk wedhorn (n=wedhorn@dsl-58-6-89-11.act.westnet.com.au)
02:15.02OloBolaI updated asterisk and now it won't start
02:16.04littleballhello, i found that some of my E1 channels the zap PRI Flags is always in Call status even after call hangup. And these channels cannot be used anymore until restart asterisk. who can give me suggestions?
02:16.23littleballzap show channel xxx
02:22.23[TK]D-Fenderlittleball, You should never use "destroy channel".  Only way to get it back is to reload Zaptel
02:23.03[TK]D-Fenderlittleball, thats a way to nuke out a channel "live".  You should use "sof hangup [channel]" to end calls.
02:24.41OloBolais anyone willing to ssh into my machine and upgrade from fedora 4 to 5, get asterisk running again (it was running fine before updating today) ? I can paypal for the help.
02:26.02littleball<[TK]D-Fender, my issue is to find out why the PRI Flags: Call
02:26.09littleballeven after the call hangup.
02:27.01[TK]D-FenderOloBola, Doesn't yum update bring you effectively up to dat automatically?
02:28.21Tako-sanCan you "yum update" from CentOS 4 to 5?  If so I am curious how to do that.
02:29.05OloBolayum is not working for some reason. I think I screwed up yum.conf.
02:30.23nohopahhhh
02:30.32nohopi got all the asterisk stuff working, finally
02:32.35nohopthanks once more for all the help, [TK]D-Fender before, and shido6 :)
02:32.47littleballhi, what will happen if there is an error in the dialplan? which extension will jump to?
02:33.14*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
02:33.17littleballexample, exten=>s,n,SayNumber(${allowedminutes}), if i forget to set the allowedminutes
02:33.33[TK]D-FenderOloBola, Aren't they at Fedora 7 yet anyways?  Sounds like a stop-gap.  What is the purpose of your "upgrade"?
02:34.09[TK]D-Fenderlittleball, it'll hang up, and no, you won't really know why.  You missed theis conversation earlier
02:35.28littleball[TK]D-Fender, total new question: what will happen if there is an error in the dialplan? which extension will jump to?
02:36.20[TK]D-Fenderlittleball, I think I heard that it might trigger "h"
02:36.59littleballhangup? hm.... does it hangup the channel if "h" doesnot do this?
02:37.10nohopi guess now it's time to add neat lil tricks to my config :)
02:37.26*** join/#asterisk RememberPOL (n=pol@adsl-75-34-10-66.dsl.chcgil.sbcglobal.net)
02:37.41nohopare there scripts or whatever available that can make asterisk act as a modem ?
02:39.10rvhi0i'd like to run a fastagi server. what's a good one?
02:40.59*** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net)
02:41.06*** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net)
02:41.21[TK]D-Fenderlittleball, if your app call fails the channel will terminate.
02:41.44[TK]D-Fendernohop, lookup "zapras" on the wiki
02:42.16*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net)
02:42.19BSD_Techhey
02:42.30BSD_Techanyone know what happen to IAXTEL
02:42.45BSD_Techfor a month now the website has ben gone
02:42.50BSD_Techand no service
02:43.36littleball[TK]D-Fender, i need to know exactly what happend on (1)app fail, (2)hangup cmd explicitely called, (3) h extension. example, if one of app fail, will it jump to h extension? before jumping, does the system called hangup() internally?
02:43.54littleballor i need to call hangup() in the h extension
02:43.57littleball?
02:44.04BSD_Techwatch the cli for output
02:44.05[TK]D-Fenderlittleball, Ok, first.... download the * source.... then completely REWRITE IT.
02:44.08[TK]D-Fender~wglwat
02:44.09jbotfrom memory, wglwat is well, good luck with all that
02:44.13*** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com)
02:44.31*** join/#asterisk samarora (i=minesh@203.88.149.165)
02:44.32rue_mohranyone know where there is a guide to setting gains in a system with a lot of them?
02:44.44[TK]D-Fenderlittleball, * will kill the call, and no it wasn't designed to give you any details.
02:45.09[TK]D-Fenderrue_mohr, Setting gains is not a normal thing.  I can imagine no such doc.
02:45.18samarorahi TK-D, I need your help regarding connecting two asterisk server.
02:45.20[TK]D-FenderBSD_Tech, Hey, just the guy!
02:45.29rue_mohrerm
02:45.33BSD_TechWhat did I do
02:45.40[TK]D-FenderBSD_Tech, Got that bit of time to assist with regards to my Adesso USB KB?
02:45.49BSD_TechI am still fixing dial plan issues
02:45.50rue_mohrbut I have gins in the fxo card, asterisk, and the fxs card
02:46.00BSD_Techsure
02:46.10rue_mohrI think there are a few more... in the channelbank for hte T1 er something
02:46.11samarorai wannt to share dial plan of the two different * servers...
02:46.13BSD_Techgive me 5 min
02:46.13[TK]D-FenderBSD_Tech, Ok, let me pull it out and plug it in.
02:46.18[TK]D-FenderBSD_Tech, np
02:46.26BSD_Techhave to go to the bathroom
02:46.30BSD_Techbrb
02:46.51[TK]D-Fendersamarora, lookup "asterisk dual servers" on the WIKI, and lookup the "switch" dialplan operator.
02:47.02[TK]D-Fenderrue_mohr, sounds ugly...
02:47.02rue_mohrhttp://about.telus.com/publicpolicy/bics_bc/main.html  << interesting for anyone who wants to know
02:47.08rue_mohrI have problems
02:47.19samarorai gone through but having some problem ...
02:47.27samaroracan u suggest any more documentation
02:47.29rue_mohrthe least of which being cbc being picked up when the fxo card picks up
02:49.03[TK]D-Fendersamarora, maybe you could actually describe your PROBLEM.
02:49.26*** part/#asterisk RememberPOL (n=pol@adsl-75-34-10-66.dsl.chcgil.sbcglobal.net)
02:49.48samarorayes
02:49.51rue_mohrI'd love to have a working set of gians for a fxo->t1->asterisk->t1->fxs  arrangement
02:49.59samarorai have two * boxes at two branches...
02:50.56samarorai would like to have sip as internal protocol and IAX protocol as inter-asterisk protocol....and moreover i am able to call to each sip client at both the side...pl suggest
02:52.18rue_mohroh oh oh, is there a utility i can use on the T1 to calibrate to a 1mW reference?
02:52.26rue_mohr???
02:52.48[TK]D-Fendersamarora, NO.  You have jsut repeated what you already told me you wanted to do.  I told you where to go to get info on doing this.  You then told me you had problems and are failing to provide details of the problem.  I am not going to go hunting for more docs when you can't describe the problems you had following the FIRST bunch
02:53.23[TK]D-Fenderrue_mohr, fxotune?
02:53.47rue_mohrok...
02:54.21rue_mohr958-2011  2100 Hz tone (toll-free in BC) I think thats 1mW
02:55.52rue_mohrfxotune is just for echo
02:56.00rue_mohrI need to check levels
02:56.34rue_mohrone of the sets I have has a REALLY loud dtmf gen, and on the new system it distorts, so something is too high
02:56.55rue_mohrI cant get a sip working to play with it in the middle of the path
02:57.33rue_mohris there no oscilloscope for T1s on an asterisk system?
02:57.43rue_mohrfor scoping a channel?
02:58.21[TK]D-Fenderztmonitor I believe
02:58.27rue_mohrthen again, I dont know I'd know what to do with the numbers, but I could tell saturation easy
02:59.41rue_mohrbetter
02:59.49rue_mohrmuch much closer to what I was talking about
03:01.08rue_mohrchanging either gain is asterisk is silly
03:01.14rue_mohrso i nee to adjust the channelbank
03:02.04rue_mohrnumbers would be nice, maybe I should add some eh?
03:03.57rue_mohrhmm, see, my rx levels options go from -3db to -10db
03:04.05[TK]D-Fenderrue_mohr, meaningless to me... but maybe someone else may know.  Keep in mind its the weekend, and on canada Day as well.... low odds
03:04.13rue_mohr:)
03:04.21*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:04.32samaroraTK, i have 1XX series dial plan at siteA and 2XX series dialplan at siteB and i want that from siteA i can call both to 1XX and 2XX...and I am having sip configured at each sites...and would like to use IAX to communicate between two boxes..
03:04.52samaroraand also i did :switch => IAX2/master:secret@iax-gw1.company.net/outbound
03:05.24rue_mohrI wonder if 1mW should be 0db?
03:05.31samarorabut do i need to register context defined in iax.conf...and if yes then where ?
03:06.08samaroraand through this statement am i able to call to sip at another * boxes...
03:08.17rue_mohrI dont get it
03:08.18[TK]D-Fendersamarora, No, you never need to register.
03:08.32rue_mohrI down _down_ the gain on the fxo card and it gets louder
03:08.44b1shop[TK]D-Fender: thanks again for your help!  we ran all week with no issues!  ;-)
03:08.56*** join/#asterisk waverly360 (n=waverly@209.149.58.214)
03:09.05rue_mohr0db is Louder than -7db
03:09.09rue_mohr!?!?
03:09.52Corydon76-homerue_mohr: correct
03:10.17rue_mohr!?!
03:10.32rue_mohr-7db should be attenuated, quieter ??
03:11.06[TK]D-Fenderb1shop, Ah, good to hear!
03:11.13[TK]D-Fenderb1shop, Happy with the setup?
03:11.27[TK]D-Fenderrue_mohr, More > Less :)
03:11.37b1shopVERY!  you can use me for a recommendation ANY TIME!
03:12.16rue_mohr-3db shoudl be 50% of the signal, 3db should be 200% signal, right?
03:12.33b1shopi'm prob gonna have a few questions for ya next week..
03:12.55shmaltzD-Fender, you in NJ? right?
03:15.14BSD_Techno he is on mars
03:19.17OloBola[TK]D-Fender: apparently I need to run Fedora Core 5 or 6 to run LumenVox Speech Starter Kit
03:19.42rue_mohrrx and tx levels at 5 points...
03:19.51[TK]D-FenderOloBola, Why don't you just reinstall or have you done a lot to your server?
03:20.14[TK]D-Fendershmaltz, Yes, the Montreal, PQ side ;)
03:20.14rue_mohroo 3 points, I cant adjust gain for the T1
03:20.37shmaltzsorry for that
03:23.17OloBola[TK]D-Fender: I'm lazy! I know someone other than myself can get it up and running in no time. I don't know linux very well is the real issue. I have to look everything up.
03:24.07polerinmeh
03:24.15[TK]D-Fendershmaltz, you keep asking every time you look to outsource your projects ;)
03:25.03shmaltz[TK]D-Fender, I'm not looking to outsource, but to hire
03:27.05[TK]D-FenderAh, well..... sorry, but I'm not heading down that way until GWB / DHS = gone currently....
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03:29.43polerinbleh,  Ok inband question
03:29.53polerinerr inbound question.  gods thats not a good start
03:29.54polerinlol
03:30.40[TK]D-FenderBRB
03:33.01polerinMy * box is also the NAT box.  * listens on the external interface.  I'm currently using broadvoice and can make outgoing just fine, but my incoming is ... twichy.  Internally I have two softphones that can dial the extensions for each other just fine, "normal" internal call.  When I try to use Dial() to connect an incoming BV call to one of these softphones the dial executes, I can see the incoming on the softphone and answer, but whe
03:33.10polerinthe call immediatly ends
03:33.34rue_mohrcan I make changes in zapata.conf effective without losing a call?
03:33.38polerinif I ignore the incoming on the softphone it will move to the next step
03:33.57*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
03:34.13rue_mohrthat OR can someone tell me what the definition of rx and tx is for zapata.conf
03:34.20[TK]D-Fender.
03:35.39shmaltzrue_mohr, rx=receive and tx=transmit
03:35.46rue_mohrduh :)
03:36.02polerinheh
03:36.09rue_mohrI need to hear louder...
03:36.12rue_mohr:)
03:36.28polerin[TK]D-Fender: btw, mind if I msg you the text of the question rather than spamming the channel with it for a second time?
03:36.57polerinyou've helped me in the past is the only reason I ask ;p
03:37.03Strom_Mrue_mohr: are you on an analog circuit or are you on t1/e1?
03:37.16shmaltzpolerin, in general you should leave it in the channel so others can help as well, for large amoutn of text use pb
03:37.16rue_mohrheh
03:37.19shmaltz~pb
03:37.20jbotA Pastebin is a place to paste your stuff without flooding the channel.  Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org
03:37.45polerinshmaltz: I already dumped in the channel... 4 seconds after he left ;)
03:38.02polerinand it's just one line... one long line.
03:38.16rue_mohrthis is in zapata.conf, I need to boost the signal from the channelbank (channel 7) thru asterisk to the channelbank (channel 2)
03:38.17[TK]D-Fenderpolerin, PB it FAST, I'm out in 3 mins
03:38.26polerinah don't worry then D
03:38.35polerinthis isn't pressing, it's just annoying
03:38.53polerinhave a nice night :)
03:38.59[TK]D-Fenderpolerin, Still ask fast, the answer may be that quick
03:39.11rue_mohrexperimant you say?
03:39.35*** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net)
03:40.36polerinhttp://pastebin.ca/598187
03:41.24polerinoh, one more thing.  If I don't answer the call and hit the ignore button, it moves to the next step rather than the call immediatly ending.
03:41.25rue_mohrhrm,
03:42.00[TK]D-Fenderpolerin, "canreinvite=no" in [general] , and each phones entry in sip.conf
03:42.44polerinawesome.  I'll give it a shot, thanks muchly
03:42.46[TK]D-Fenderpolerin, ignore jsut tells the phone to refust the call, and * won't care.  * does what * is TOLD to do.  Check your DIALSTATUS to see if the reason is triggerable
03:43.24polerinyeah the ignore sends a 402 to *
03:43.26[TK]D-Fenderok, times up! :)  Out for a while, back later.
03:43.47[TK]D-Fenderpolerin, Yeah, but chan_sip doesn't pass it back to the dialplan necessarily in a usable way
03:43.52[TK]D-Fenderwork on that a bit :)
03:43.59polerinnods. I'm not even checking on that
03:44.00polerinlol
03:44.06polerinI'm more worried about it picking up right
03:44.07polerinhehe
03:44.16[TK]D-Fenderok, keep it up, I'll be around.
03:44.20polerinkay kay
03:44.21polerinthanks
03:44.23[TK]D-Fendernp
03:44.52Strom__rue_mohr: what do you have connected to those two zap channels?
03:45.40rue_mohrone is an fxo card, the other fxs card, on a channelbank
03:46.32Strom__ok, and so essentially you're attempting to compensate for attenuation on the POTS circuit?
03:46.58rue_mohrwell see, funny thing, plugging the phone into the pots is fine
03:47.08rue_mohrbut I cant adjust all the system gains to 0db
03:47.24rue_mohrthe rx on the fxs card only goes to -3db
03:47.39Strom__rue_mohr: you shouldn't be dicking with gains on the FXS ports
03:47.42rue_mohrand I'm having all sorts of fun with distortion
03:47.51rue_mohrwhat shoudl they be?
03:47.59Strom__0 on FXS
03:48.16Strom__compensate for attenuation on the FXO port only
03:48.17rue_mohrfxs has rx and tx
03:48.23Strom__both should be 0
03:48.29rue_mohrdont know if I can
03:48.36Strom__uh, of course you can
03:48.43rue_mohrand 0 is quiet on the fso, -7db is louder?
03:49.02Strom__pastebin your zapata.conf
03:49.21rue_mohr:)
03:49.35rue_mohrit would be nice to have pastebin url in the topic like ##c
03:49.43Strom__www.pastebin.ca
03:49.55Strom__there's enough crap in the topic already
03:50.02Strom__it's assumed you know where a pastebin is
03:50.06Strom__or can type
03:50.07Strom__~pb
03:50.08jbotA Pastebin is a place to paste your stuff without flooding the channel.  Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org
03:51.07rue_mohrhttp://rafb.net/p/4gamyw42.html
03:51.37Strom__and after all that kvetching, you go and use a different one
03:51.41Strom__ugh
03:51.47rue_mohrI used the one from ##c
03:51.53rue_mohrcause I was already there
03:51.54Strom__don't kvetch about these things if you're not going to listen to the replies
03:52.21rue_mohrI was already copying that url in ##c when you triggered the bot...
03:52.34rue_mohranyhow
03:52.42Strom__anyway, you're not limited to one instance of the rxgain and txgain lines
03:52.48Strom__you can do something like
03:52.53rue_mohr?
03:52.56Strom__rxgain=0
03:53.00Strom__channel => 1
03:53.07Strom__rxgain=2
03:53.13Strom__channel => 2
03:53.14Strom__etc
03:53.17rue_mohrok
03:54.19*** join/#asterisk bmg505 (n=leon@196.209.179.147)
03:55.34*** join/#asterisk lowlevel (n=Stuart@bas4-sudbury98-1242384558.dsl.bell.ca)
03:55.55Barmalwhere can I download sip firmware for cisco 7970 ???
03:56.05Strom__cisco.com
03:56.08Strom__or your reseller
03:56.36Barmaldo not have account w/ cisco and reseller is ebay :)
03:56.46Strom__well, looks like you're out of luck :)
03:57.11Strom__contact a cisco reseller and purchase a license from them
03:57.22Barmalhow much does it cost?
03:57.27Strom__$7?
03:57.58Barmalany recommendation on firmware version?
03:58.06Strom__whatever the latest is?
03:58.57polerinmeh
03:59.08polerinok so careinvite=no isn't working
03:59.16Strom__should be canreinvite
03:59.22Strom__not careinvite
03:59.25polerin*headdesk*
03:59.36polerinthat's what I get for assuming i'm reading somethign right
03:59.38polerinsomething
03:59.43*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:59.44Strom__can re invite
04:00.05polerinyeah,
04:00.10Strom__(syllables)
04:00.17polerinironically just one letter off, but that'll do it
04:02.38Strom__though one wonders if there's a difference between a care invite and a care package
04:04.05polerinwell.  after adding the 'n' to it in  the proper place... it works
04:04.07polerinawesomeness
04:04.11Strom__woooooot
04:04.29polerinaparently * doesn't like caring
04:04.52*** part/#asterisk samarora (i=minesh@203.88.149.165)
04:05.16Strom__rue_mohr: are you still there?
04:12.30rue_mohryes
04:13.26rue_mohrthus far, I have not eliminated CBC from the fxo card, I'll think of it a little more
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04:15.02Strom__CBC?
04:15.14Strom__....the canadian broadcasting corporation?
04:15.24rue_mohryup
04:15.32Strom__what are they doing in your FXO card
04:15.33rue_mohrthe radio station i cant stand most
04:15.38rue_mohrgood question
04:15.39Strom__oh
04:15.54Strom__if you're picking up radio on your POTS line, you need to buy a filter
04:16.04Strom__you can't configure that away
04:16.05rue_mohrI dont know if its the fxo card, or when the card connects to the fxs card
04:16.11Strom__uh
04:16.15rue_mohrnaaa, cause I dont get it with just a phone
04:16.15Strom__it's the fxo card
04:16.25polerinthis is why I like coax.  If you are picking up ingress... Find where the break in your sheilding is :P
04:16.34rue_mohrthink I shoudl try a small cap or large resistor across the line?
04:16.38Strom__maybe your channel bank is a piece of crap
04:16.43rue_mohrheh
04:16.51rue_mohrmainstreet 3624
04:17.00Strom__never heard of that company
04:17.08rue_mohrI really need to try grounding it better though
04:17.08Strom__i only use Adtran
04:17.08rue_mohrnewbridge
04:17.11Strom__uh, yes
04:17.38polerinwait.. you didn't gro... and you are wonder... oh ... heh.
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04:18.32Strom__polerin: no, i'd take the disconnected ground wire and strangle people with it
04:18.34polerinTo be honest, it is VERY odd to find cable that isn't grounded whatsoever.
04:18.40polerin(CATV mind you)
04:18.52rue_mohrits grounded with both conductors of the brown pair at the demarc :)
04:19.00Strom__rue_mohr: uh, no no and no
04:19.08rue_mohrI know I know
04:19.10Strom__"the brown pair" is not ground
04:19.11polerinbecause it's grounded at the pole, usually is grounded through whatever equipment (long story there), and is bonded to the house ground at the side of the house
04:19.21Strom__polerin: didn't you work for cocks?
04:19.22Strom__er, cox
04:19.27rue_mohrI wanted to know if It would quiet the radio station a little, before I go to #6 cable
04:19.31polerinnope I work for comcrap
04:19.41Strom__it's comcrapular?
04:19.48polerinevery fucking day.
04:19.51rue_mohrits grounded, I grounded it at the demarc
04:20.01Strom__rue_mohr: grounded it how
04:20.05polerinthough keep your finger crossed, because I might be moving up the shitladder
04:20.07rue_mohrto the ground lug
04:20.11rue_mohr:)
04:20.19Strom__and that's fine and well, but your channel bank needs to be grounded too
04:20.33rue_mohrthats what the other end of the brown pair is conneted to
04:20.48Strom__rue_mohr: "the brown pair" is for telecom
04:20.53rue_mohrI know
04:20.57Strom__you're supposed to use real ground wire
04:21.03Strom__and know that brown/white is pair four
04:21.04rue_mohrI needed wire to try some sorta-grounding
04:21.08rue_mohrI know
04:21.36Strom__as opposed to brown/red or brown/black or brown/yellow or brown/violet
04:21.51rue_mohrits just a 4 pair to the demark
04:21.55Strom__demarc
04:21.58rue_mohrcat5e :)
04:21.58Strom__not demark
04:22.14rue_mohrsorry
04:22.20rue_mohrfingers keep doing that
04:22.24Strom__you /do/ know it's short for "demarcation point", right?
04:22.30rue_mohryup
04:22.36polerindemarcation ftw.  (spelling for the loose)  I wish comcast knew what that is
04:22.46polerinwe end up supporting way too much crap
04:22.50rue_mohr:)
04:23.14rue_mohrhow do I transfer a call on my nortel phone again :)
04:24.02rue_mohr**266344266344 and find erase all?
04:24.27Strom__is this #nortel?
04:24.32rue_mohrheh
04:24.37rue_mohrjust kidding...
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04:26.45polerinngiht all
04:27.05polerin...  night even.  I think we see why I"m going to sleep
04:27.15polerinrue_mohr: ground it
04:27.17polerin;P
04:27.20rue_mohr:)
04:27.36rue_mohrshall I ground the 66 strip to?
04:27.47rue_mohrcan I just ground my roommate?
04:28.18*** join/#asterisk nephfl (n=no@wsip-68-110-130-57.ga.at.cox.net)
04:28.35nephflhave any of you guys taken the dCAP exam?
04:28.43rue_mohrnot I
04:28.48rue_mohrunless I didn't notice
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04:29.33Strom__nephfl: I've taken it
04:29.57Strom__and passed with flying colors (colours)
04:30.45Strom__nephfl: por que?
04:31.00nephflI have been looking at certifications for our company.  We have set up a couple asterisk systems now, and I was wondering if it would be worthwhile to take the exam. (also polycom'
04:31.03nephfls exam)
04:31.32*** join/#asterisk Nugget (i=nugget@carrera.macnugget.org)
04:32.03nephflalso how difficult is it to prepare for the exam?
04:32.11Strom__nephfl: depends on your experience
04:32.20rue_mohrhmm
04:32.23Strom__the practical gives you 90 minutes to set up a system
04:32.31Strom__the written is a lot of asterisk and telecom theory
04:32.40rue_mohrshoudl I use the 66 frame I have or should I spend $25 and get a bix frame?
04:32.49Strom__66 blocks are fine
04:33.14rue_mohrthis one dosn't look like it was made int eh 60's either
04:33.20nephflhow much does the cert cost?
04:33.26Strom__nephfl: US$300
04:35.12rue_mohrhttp://www.siemon.com/e-catalog/ECAT_GI_page.aspx?GI_ID=66_field-terminated-m-series-s66-blocks
04:35.15rue_mohrI have about 6 of those
04:35.25nephfli have a ton of IT experience, but am new to asterisk and phone, trying to figure out the course to get into phone
04:36.04Strom__nephfl: it may be worth taking the asterisk bootcamp course also
04:36.48Strom__rue_mohr: yes, i work with S-66-M-50 blocks all the time
04:36.55Strom__can't remember how the hell they're hyphenated though
04:37.39rue_mohrI dont know what the cable path shoudl be
04:38.12Strom__?
04:38.21Strom__two sets of blocks
04:38.28rue_mohron a bix strip, you do things to leave the service loops
04:38.32Strom__one for your incoming cabling, one for your internal station wiring
04:38.49nephfli saw the bootcamp but the price is pretty steep on my budget
04:38.53Strom__fixed cabling routes in through the standoff block, round the side, and into the outer set of contacts
04:39.01rue_mohrdo they take the distribution cable up from in behind it?
04:39.18rue_mohrah
04:39.19Strom__then you run jumpers from the inner set of contacts on the one block to the inner set of contacts on the other block
04:39.24rue_mohrah
04:39.33Strom__whizzy-jizzy easy
04:39.48Strom__but i've seen it screwed up a lot too
04:39.48rue_mohrshould you be able to pull it off and flip it over?
04:39.54Strom__wha?
04:39.57Strom__pull what off
04:40.05rue_mohrno service loops/slack?
04:40.11rue_mohrhmm
04:40.12Strom__....no
04:40.21rue_mohrits just different :)
04:40.48Strom__lemme find photos
04:40.57rue_mohrso there is no "leave slack like this so you can move any pair to anywhere"
04:41.04rue_mohrhttp://www.siemon.com/share/products05/66_pre-wired-157-series_big.jpg
04:41.44Strom__http://tippenring.com/images/telecom/sto/DSCF0581.JPG
04:41.50Strom__rue_mohr: the pairs remain fixed in place
04:42.03rue_mohryar
04:42.05Strom__once you punch the pair down to the block, you should never have to move it again
04:42.10Strom__you move the jumpers
04:42.17Strom__because jumper wire is cheap and plentiful
04:42.17rue_mohrin bix, you can do a thing so you can move pairs around
04:42.23rue_mohrk
04:42.26Strom__that's brain-dead
04:43.15rue_mohrnortel is just that isn't it, north america
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04:43.31Strom__?
04:43.41rue_mohrbix
04:43.50Strom__we don't have bix in california
04:43.56rue_mohrooo
04:44.06rue_mohrmaybe its a telus thing
04:44.22*** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net)
04:44.23Strom__get your bix on route sixty-six
04:44.30rue_mohrheh
04:44.39rue_mohrall the old stuff here is bix, the reallly old stuff
04:44.55Strom__i should take a shower
04:45.07Strom__then go out for the evening
04:45.52nephflout, what is that?
04:46.16rue_mohris there something I'm in I shoudl be worried about?
04:47.05rue_mohrI think the fact this 66 block is cat5 disturbs me
04:51.27NuggetI didn't think that was possible. I thought you needed type110 for cat5 cable
04:53.08rue_mohranyone want to trade an S66M1-50 ( 50 pair) 66 block for a S66M4-12 (12 pair) block?
04:53.47rue_mohrbah, i can just get a bix frame ...
04:54.03Strom__rue_mohr: oh don't be scared of the damned 66 block
04:54.15rue_mohrbut but but
04:54.18Strom__it's really really really really simple
04:54.54Strom__now if only i had a scanner
04:55.02rue_mohrthts it see
04:55.13rue_mohrbix is more modern
04:55.19Strom__blah blah blah
04:55.26nephflso, the practical part of the exam is that a config with ip with sources including the confs that are normally included?
04:55.26Strom__excuses excuses
04:55.53nephflby ip i mean voip extensions and trunk
04:56.03Strom__nephfl: you get a PC, a tdm card, a t1 card, a pots circuit, an isdn pri, an analog phone, a sip phone, and 90 minutes
04:56.25nephflcool
04:56.28Strom__the pc already has linux and the asterisk source on it
04:59.14nephfli imagine with zaptel configured automatically most of the custom stuff is in the dialplan, right?
05:00.09Strom__pretty much
05:00.49rue_mohrhttp://eds.dyndns.org:81/~ircjunk/images/dscn9964.jpg
05:01.06rue_mohrmaybe my bix implementations isn't pretty, but man, its so much smaller than the 66
05:01.17Strom__well good for you then :)
05:01.23rue_mohrits not my fualt, all the cables were just a little short
05:01.41Strom__the nice thing about 66 is that you terminate the cables /once/
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05:01.54nephfli think i read you can get the company labeled an authorized partner for 6 months until you get your cert, seems like that would be plenty of time to get experience with the aspects of asterisk that I haven't hit so far
05:01.54Strom__if you keep re-terminating the same cable, you will eventually run out of cable
05:02.19rue_mohryea, I suppose you dont have to move it aroudn to add more distribution cables
05:04.19rue_mohrheh, I just missed the network port with the wire conmming out that say "T1 CROSSOVER! T1 CROSSOVER!"
05:04.38rue_mohr:)
05:04.52nephfldo most people still prefer/use the 66 panels, or do alot of people use patch panels?
05:05.02Strom__depends on the application
05:05.06rue_mohrhere its preffered to use bix
05:05.08Strom__for ethernet, use patch panels
05:05.11Strom__for phone, use 66
05:05.13rue_mohr66 is old school
05:05.13Strom__simple
05:05.27rue_mohroh I see, sorry, I missed the question
05:05.50rue_mohrI installed patch panel for telco for a fellow who wanted to be able to do his own rewiring
05:06.21rue_mohrhe has 1 port for each pstn line and 1 for each drop, he inserts splitters/patches as required
05:06.26nephflFrom reading i got the impression that most places were just wiring cat5/6 and using patch cables (since rj45/cat5/6 is backwards compatible and can provide phone over the extra pair)
05:06.31Strom__sounds like a fucking mess
05:06.48Strom__nephfl: you're thinking of TIA-568-A/B
05:07.04Strom__RJ-45 specifies a jack type; CAt5/6 specifies a cable type
05:07.06rue_mohrthe problem with using 8 conductor jacks for phone is that on SOME connectors, the outter pairs in the jack bend and dont work for network
05:07.36rue_mohrits not too bad, and he can rewire at will, no special tools
05:07.43dc3aesw/ the PAP2, do i need to clear out its dialplans or anything to use with *?  I can't seem to make *[number][number] extensions in my dialplan, the pap2 just freaks out with a fast busy tone
05:07.56dc3aesim suspecting that it is interpreting it locally in a dialplan within the pap2?
05:07.58Strom__dc3aes: those are called vertical service codes
05:08.06Strom__~vsc
05:08.13jbothmm... vsc is Vertical Service Codes such as *67, *69, *72, and *82.  These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments.  A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html
05:08.30*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
05:08.42dc3aesi was vaguely aware of this.. but figured i could sneak some in around the "officials"
05:08.49Strom__WRONG
05:08.58*** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com)
05:09.12Strom__(buzzing noise)
05:09.14dc3aeshaha
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05:14.10rue_mohrhaha I know why -that guy- was mounting the bix strips vertically, he was a 66 guy from wayback... NOW that makes sense
05:14.35nephflso, if you were to wire new construction, wouldnt you wire all connections with cat5/6 to 568 b to a patch panel and then just connect to switch/pstn/asterisk box by patch cable?(no 66 blocks)
05:14.36rue_mohrdoes that mean I shoudl mount my 66 strip horizontally }:)
05:14.53rue_mohrnot for telco
05:15.09rue_mohrall the telco would go to a punched interconnect
05:15.21rue_mohrand the network to patch panels
05:15.23Strom_Mnever ever mount 66 horizontally
05:15.38rue_mohr- in comes voip - out goes punched interconnects
05:15.41denonnever mount a 66 block at all
05:15.42denon110 :)
05:15.43Strom_Mor i will personally unmount the block and hit you with it
05:15.47rue_mohrStrom_M, thats just wrong?
05:15.53Strom_Mvery
05:15.56rue_mohrhah
05:16.13denonjust use 110 blocks, then you wont have to redo em later when analog is history
05:16.29rue_mohrthat sounds just a little more like what me and a telco guys said when we saw a vertically mounted bix frame
05:16.52rue_mohrI'm working somewhere between a budget and a timeline
05:17.05rue_mohrits $25 for a bix frame, which I have strips for
05:17.18rue_mohror the 66 I have in my lap
05:17.29rue_mohror one of the other ones in the box in the livingroom
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05:17.39denonjust so long as whoever is whining about timeframe understands that you cant run data over it later
05:17.48rue_mohrme :)
05:17.58rue_mohrand I used cat3 wire for hte phones :)
05:18.02nephflso 110 blocks then
05:18.02denonew
05:18.05[TK]D-Fenderdc3aes, You can work around them all you want.
05:18.12denonnephfl: scroll back.
05:18.28denonrue_mohr: why cat3? cat5 is cheap
05:18.29[TK]D-Fenderdc3aes, the ATA's features can be remapped and the dialplan made to accoutn for trather easily
05:18.48[TK]D-Fendercat3 = MISTAKE.  Cat5E + RJ45 EVERYWHERE
05:18.55rue_mohractaully, I'm not sure it even said cat3, its just red black green yellow in a jacket :)
05:19.05denonoh brother
05:19.07rue_mohrI didn't have enough cat5e
05:19.17denongo down to your local lumber yard and get more :)
05:19.22denoneveryone stocks it no
05:19.22rue_mohr$$
05:19.22denonnow
05:19.25rue_mohryea
05:19.29denonmost people think its actually phone cable
05:19.29rue_mohrits cheaper than cat3
05:19.48rue_mohrbut I have a few boxes of it from a commercial install
05:19.58rue_mohrhe was gonna throw out a few full boxes
05:20.00denonI talked to an electrician last week, that said "oh, you can run computer networks on cat 5 now too?"
05:20.06rue_mohrhaha
05:20.08denonas if it was some new invention, running computers on phone cable
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05:20.43denonhe also used those horrible rj45 plates with the screws on the back
05:20.47rue_mohrit would fit in the cabinet better horizontally....
05:20.50denoninstead of a punchdown block
05:20.53rue_mohrI HATE those
05:21.01denonI cant imagine you ever get anything over 1Mbps through a jack like that
05:21.04denonthere's no twist for miles
05:21.06rue_mohrthey buy them at work cause their cheap
05:21.08nephflso why a 110 panel and not a rj45 punch panel?
05:21.26denonnephfl: he's doing a 66, analog stuff
05:21.30denonIm just saying he should future-proof
05:21.33rue_mohrto be honest I'v never *seen* a 110 interconnect
05:21.39denonhe's saying he's cheap, and already has the gear
05:21.47rue_mohrnot with cat3 cable
05:22.02rue_mohrI have cat5 runs isntalled,
05:22.07rue_mohrits a robotics workshop
05:22.07denonor whatever the ma-bell specialty cable is that you have
05:22.21denonare there *any* twists? :)
05:22.50rue_mohrI suspect not, actually, i'm gonna have to opent eh wall back up to remember what cable I used
05:23.02nephfli have a silly question...is there a tool you are supposed to use with the 66 blocks?
05:23.04rue_mohrit might have been that stuff I said, but it might have been 3 pair cat3
05:23.11rue_mohryes
05:23.16denonnephfl: yes - a punchdown tool with a 66 bit
05:23.26denonpreferrably a cutting bit imho, much faster and cleaner
05:23.43denona *real* spring loaded punchdown tool
05:23.49rue_mohrhttp://eds.dyndns.org:81/~ircjunk/shop/dscn9924.jpg
05:23.50denonnot just some crappy plastic thing from home depot
05:24.10rue_mohrhttp://eds.dyndns.org:81/~ircjunk/shop/dscn9921.jpg
05:24.17denonrue_mohr: that pex coming through the floor?
05:24.23rue_mohryup
05:24.26rue_mohrin floor heating
05:24.34denonnod
05:24.40nephflive never used a 66 bit... do they have them at most hardware stores?
05:24.46denonjust got done plumbing my new house - pretty used to seeing those caps
05:24.48rue_mohrI did an experiment
05:24.57rue_mohrI burried all the comms cable
05:25.00denonnephfl: not a clue, sorry - they're not hard to find though
05:25.29rue_mohrI know right where it is, gonna use mod rings to pull it all out as I need
05:25.31denonnephfl: I would imagine home depot or such has em
05:25.48denonoh, that pex is for the comm?
05:25.59rue_mohrno
05:26.04denonah ok
05:26.05denonwas gonna say
05:26.06nephflguess i should get one of those...lol...
05:26.25denonnephfl: how would you do it without?
05:26.31denonneedlenose or somethin?
05:26.32nephfldo people still use those red phone handsets with the aligator clips?
05:26.37denonbuttsets
05:26.38denonyeah
05:26.39rue_mohrplase dont say a flat blade screwdriver
05:26.56nephflyeah needlenose and electricians scissors to trim
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05:27.02denonrue_mohr: I'd love to see someone do a screwdrive job in a 110 block
05:27.10denonnephfl: man, that's gotta be painful
05:27.15littleballhello, is it possible to reset a specific Zap channel?
05:27.19denonyou can punch down like a strand per second with a tool
05:27.27nephflit was just a small job, it wasnt worth looking for a tool
05:27.28denonlittleball: yes,  soft hangup zap/1
05:27.47denonnephfl: I dunno, I'd buy a tool for more than like 3 pair :)
05:27.53rue_mohrhttp://eds.dyndns.org:81/~ircjunk/shop/dscn9691.jpg you cn just see the concentration point there
05:28.16nephflyeah, now that i know there Is one...i guess ill have to pick one up
05:28.34denonnaughty, naughy .. using pre-cut insulation batts?
05:28.42rue_mohr?
05:28.44denonyou should use a roll, then split the insulation for your wires
05:28.56rue_mohrroll?
05:29.04denonyeah
05:29.05littleballdenon, the problem is that this channel is not actually in call status
05:29.10rue_mohrwe dont have rolls
05:29.11littleballzap/112 is not a known channel
05:29.19denonreally?
05:29.22rue_mohrno
05:29.27denonwhat kinda lame big box store are you buying at?
05:29.32nephfltoo bad i cant just ask what other very basic crap i dont know...lol
05:29.33rue_mohrheh
05:29.49rue_mohrgooberville building supplies
05:29.53denonheh
05:30.01denonwell, mansville makes the rolls too
05:30.10denonone big honkin roll, you cut it to exact length
05:30.11littleballdenon,        Request that a channel be hung up. The hangup takes effect
05:30.11littleball<PROTECTED>
05:30.15denonworks great, fits really tight
05:30.20rue_mohrk..
05:30.24rue_mohrprolly cheaper to
05:30.32denonyeah, slightly as I recall
05:30.42nephflwho uses dyndns anymore? dont we all have static ips?
05:30.47rue_mohrhttp://eds.dyndns.org:81/~ircjunk/shop/dscn9789.jpg R40 would suck
05:30.47denonnot huge, though - seems like the kraft-backed crap is cheapest when it's on sale
05:31.08denonand I really hate that kraft paper
05:31.09rue_mohrI dont have $$ for static ips
05:31.21denonI prefer doing a single continuous vapor barrier out of like 6mil
05:31.34nephflhere it is cheaper to get business service at home with static than residential service
05:31.42denonoh, see I never put that stuff in the ceiling - I prefer that white blown cellulose
05:32.02nephfland their competator is $5/mo for static
05:32.12denonplace will usually let you use their blower for free
05:34.36rue_mohrstaicis like $25/mo here
05:38.09*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:38.21[TK]D-Fenderlittleball, Zap/112 is not a channel.  Zap/112-1 would be.
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05:38.39[TK]D-Fenderlittleball, in terms of a CALL channel.
05:38.53[TK]D-Fenderanyways... way late, I'm off later all.
05:41.00littleballTK, is it possible to reset a PRI channel, no matter it is used or not. Zap/112-1 means that the channel is in use
05:41.02littleballright?
06:00.26*** join/#asterisk denon (n=denon@tooth.decay.org)
06:00.26*** mode/#asterisk [+o denon] by ChanServ
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06:25.29rue_mohrdoes it sound plausable to create an asterisk-asterisk link to antoher system with a console phone thats hooked to an amp for paging?
06:27.41rue_mohr"woudl one of you guys _PLEASE_ get off your butt and answer hte phone!?!"
06:27.47rue_mohr:)
06:28.04rue_mohrsecret extension number... 4 that would be
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08:45.59tsurkohello
08:46.04*** part/#asterisk chexum (i=chexum@gateway/tor/x-b7a274b2700c4a9a)
08:46.18tsurkosince which release of asterisk ael2 is included by default?
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09:00.07Strom_Ctsurko: 1.4
09:07.27*** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com)
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09:17.01EmleyMoorWhen I'm making calls using my softphone and a USB headset, I intermittently get a loud echo of my voice back, a moment after I actually spoke - this is very offputting - I am wondering if this is down to asterisk, ekiga or the driver for the headser
09:17.05EmleyMoorheadset
09:17.17EmleyMoor(it doesn't just affect ekiga, the same happens with kiax
09:18.53Strom__what kind of entrance facilities are you using?
09:19.12EmleyMoor"entrance facilities"?
09:19.20Strom__...think about it
09:19.25Strom__your PSTN interconnect method
09:19.58EmleyMoorAh, fairly good point - let me try a call that does not go out over my TDM400P
09:22.29EmleyMoorOnly affects calls over PSTN - over Internet it's fine
09:22.42Strom__is echo cancellation enabled on your zap channels?
09:23.08EmleyMoorYes
09:23.52Strom__just fyi, softphones are echoey, and the additional packetization delay between the softphone
09:23.53Strom__er
09:24.17Strom__just fyi, softphones are echoey, analog interfaces are echoey, and the additional packetization delay between the softphone and the tdm interface exacerbates the echo
09:24.37EmleyMoorExacerbates it exponentially?
09:24.52Strom__i would at this point like to kill whatever shithead at Apple thought it would be a fantastic idea to put an additional "enter" key next to the "left arrow" key
09:25.25Strom__i dont know about exponentially
09:25.32Strom__but to some degree, yes
09:26.37EmleyMoorWhat value is good for echotraining?
09:26.44Strom__I ike 800
09:28.24Strom__s/ike/like/
09:32.04EmleyMoorIf 800 is not great but better, would a higher or lower number be best next?
09:32.18Strom__how long is the delay on the echo?
09:32.31EmleyMoorAbout 500
09:33.37Strom__500 milliseconds of echo based on your estimation?
09:33.43EmleyMoorYes
09:34.39Strom__try 1000
09:34.51EmleyMoorJust did - it dropped to about 250
09:34.54Strom__also, how long is the copper loop back to the telco's CO?
09:35.08Strom__the echo shouldn't be varying in length
09:35.44EmleyMoorIt is somewhere between half a mile and three quarters, so I am told
09:37.25Strom__that shouldn't be problematic
09:37.36Strom__have you tried the HPEC?
09:38.06EmleyMoorNot sure how to do that
09:39.03Dovidcan anyone tell me what package i am missing ? Trying to install 1.2.10 on CentOS5
09:39.04Dovidhttp://pastebin.ca/598404
09:39.09DovidI have termcap installed on my box
09:41.39Strom__Dovid: ncurses5-dev
09:49.17DovidStrom__: I never needed it b4. is this something new ?
09:50.09uwei needed it ever since i stated using asterisk
09:51.33Strom__Dovid: you've always needed it
09:51.33Strom__sometimes it gets installed with the linux distro though
09:51.38*** part/#asterisk Strom__ (n=strom@netblock-66-159-243-60.dslextreme.com)
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09:52.04Strom__oops
09:52.06Dovidah ok
09:52.25Dovidand i forgot to show how many g729 channels I have installed on my box
09:52.29Dovidhow do i do that ?
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10:12.59EmleyMoorHaving used fxotune, should I notice anything running?
10:16.23EmleyMoorWhy would I get this: Jul  1 11:16:02 WARNING[14841]: chan_zap.c:1425 zt_train_ec: Unable to request echo training on channel 4
10:21.39EmleyMoorWhat echo cancellers should I try?
10:21.58EmleyMoorI have MG2 right now
10:23.04ThoMeJT: hey. :-)
10:23.11JThi
10:24.38ThoMeJT: works bristuff with digium cards? i have a bri 4 port card.
10:24.47JTdon't know
10:24.49ThoMeJT: and is bris* better as mISDN ?
10:25.01JTi'm very curious, never heard anyone testing the digium card with it
10:25.04JTyes
10:25.05JTmuch better
10:25.09ThoMeah ok. much? ohha.
10:25.18ThoMeJT: last question, do u know snom?
10:25.42JTi know what they are, i don't use them
10:25.53ThoMehmmm, ok.
10:26.17ThoMeJT: i have yesterday, 12 hours test it how i can use the led's if i recieve a call...
10:26.20ThoMebut.. no result :/
10:26.36JTi couldn't understand that
10:27.08ThoMehmm, the phones have led, for the lines.
10:27.27ThoMeand if i recieve a call i would like led on... and if call == done, then led off
10:28.58EmleyMoorDoes OSLEC require 1.2.13 or will it work with 1.2.11?
10:29.39Strom__EmleyMoor: jesus, we're on 1.2.20 now...
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10:30.13JT2 days ago, explains it
10:30.35EmleyMoorI'm not interested in "out-of-system maintaining asterisk - so I'm stuck with 1.2.11 for now
10:30.43Strom__?
10:31.16JTin english?
10:31.43EmleyMoorI am stuck with what Debian gives me - that needs enough work for now.
10:32.03Strom__EmleyMoor: it's ridiculously easy to install asterisk from source
10:32.20EmleyMoorEasy, yes. Also time consuming
10:32.25JTEmleyMoor: err just compile it from source
10:32.26Strom__uh
10:32.30JTwhat a straw hat excuse
10:32.30Strom__it takes five minutes
10:32.33JTthat is stupid
10:32.48Strom__"make clean; make install"
10:32.52JTit is indeed fast unless you're on a Pentium I
10:32.58KwakwaThe current asterisk build is 1.2.13 on debian atm
10:33.07JTon stable perhaps
10:33.13Strom__JT: 386DX!!!!
10:33.16EmleyMoorWell, why am I still on 11?
10:33.22JTKwakwa: it's generally advised to compile asterisk from source
10:33.27JTnot use the packaged version
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10:33.34GivemeloveStrom__ OpenWRT!
10:33.43Strom__uh, no
10:33.49Strom__that's recent
10:33.49Givemeloveohhh yes
10:34.01Givemeloveyeah, but what a pain to cross-compile it
10:34.05KwakwaJT: Aye, I'm about to compile 1.4.6 after reading the cdr / iax fixes I've been after
10:34.05EmleyMoorasterisk is at 1.2.13, yes - zaptel 1.2.11
10:35.26JTdo you have those in america?
10:35.34Strom__indeed
10:35.40Strom__this is US currency I'm fondling
10:35.40JTi know you have threse stupid things known as "1 dollar bills"
10:35.46KwakwaIs there any way to bridge two channels from the manager api? (i.e. IAX2 -> ZAP)
10:36.59*** join/#asterisk remmo (n=junk@203.62.147.3)
10:37.22EmleyMoorI'm more interested in fixing this echo than spending lots of time undoing lots of my hard work
10:39.08KwakwaI'm basically trying to automate a transfer using manager... call comes into a dynamic queue member, they answer it then by clicking a link (I handle that) and it originates a new call on their phone. They answer it an start a fake attended transfer. If the callee wants to take the call I want the agent to click another link which then bridges the two calls on their phone together.
10:41.28*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
10:49.59EmleyMoorIf I get echo in softphones when calling over FXO, will I get it in hardware IP phones too?
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10:56.10Strom_Mits a possibility
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10:59.18JTEmleyMoor: who suggested to undo your hard work?
11:04.38tsurkois the account name in a caller saved in any variable, or is there another way to get it in the dialplan?
11:05.48GivemeloveGuys, I've identified the Issue I was facing with ODBC_STORAGE
11:06.08Givemeloveit appears that it's the package MyODBC on CentOS 4.4 which is responsible
11:06.39GivemeloveWith a CentOS 5.0 and the package mysql-connector-odbc (which replaces MyODBC), it seems to work like a charm.
11:06.58Givemelove(this was for the guys who seeks in the logs)
11:07.25EmleyMoorJT: apart from such things as the dialplan and custom sounds, they who suggested a local compile
11:07.41JTa local compile would not undo hard work
11:07.43JTthat makes no sense
11:07.54*** join/#asterisk alrs (n=lars@pozug.com)
11:08.38JTusing software that is not 7+ releases old is called sensible practice
11:10.51EmleyMoorJT: Using software that works properly is also sensible practice - not that I am saying it wouldn't work properly if I compiled it but it didn't once before
11:12.56JTit's also not a good idea to run software that has known security vulnerabilities
11:13.29EmleyMoorJT: I am certain it doesn't - any that there were will have been fixed
11:14.29EmleyMoorIf someone can prove that a fix for the specific problem I am having lies in an upgrade, I will try it
11:17.13JTi am certain there were serious security issues around 1.2.12 era or so
11:17.20EmleyMoorWere
11:17.37JTerr you're at that era of asterisk are you not?
11:17.48JTand so far your excuses to not upgrade have been ones of lazyness
11:19.16EmleyMoorI assume one of the security problems to which you refer is the integere overflow in the Skinny channel driver?
11:19.32EmleyMoorAh, no, that's even older!
11:19.33KwakwaEmleyMoor: Unless you're in a production environment and your boss will hang you for any downtime to his/her callcentre, I really do suggest you install the latest version of asterisk. There's lots of fixes for problems you might come into in an older version.
11:19.35JTthat was one
11:21.16EmleyMoorI just want rid of the echo
11:21.49KwakwaHave you tried a more recent version to see if the echo still exists?
11:22.15EmleyMoorNot as yet - I will probably do so when time permits though
11:22.20*** join/#asterisk friedrich| (n=friedric@e177251223.adsl.alicedsl.de)
11:23.46EmleyMoorI doubt it will fix it, TBJ
11:23.49EmleyMoorTBH
11:24.20Strom_Memley: you could have tried it in the time youve spent kvetching, you know
11:25.01KwakwaI just compiled 1.4.6 while you've been on here tbh :)
11:26.21Kwakwawget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.6.tar.gz; apt-get install build-essential bison ncurses-dev libssl-dev libnewt-dev zlib1g-dev initrd-tools cvs procps; make; make install; make samples
11:26.26Kwakwaassuming you're running debian
11:26.27JTi takes abut 5mins to recompile everything
11:26.34JTon an average machine
11:27.38EmleyMoorYes - then there's all my configs to put into place... OK, fair enough, it doesn't take long, but when there's other equally productive and essential things getting in the way...
11:28.14Kwakwaback up your config files
11:28.27JTcp /etc/asterisk * /some/backup/location
11:28.29JTcompile
11:28.39JTcp /some/backup/location/* /etc/asterisk
11:28.49JTanyway, it doesn't usually overwrite them
11:30.31EmleyMoorAt least my dialplan will be safe (apart from perhaps needing some rewrites for custom sound location)
11:30.39EmleyMoorIt now has 862 priorities
11:30.43*** join/#asterisk alrs (n=lars@pozug.com)
11:31.07EmleyMoorIs there a list anywhere of all that 1.4 considers deprecated (or worse, no longer supports)?
11:31.22JTi was talking about upgrading to the latest 1.2
11:31.32EmleyMoorJT: Yes - and that will be tried first
11:31.46KwakwaYou could check the changes log
11:32.04JTUPGRADE.txt
11:32.05EmleyMoorKwakwa: Good point
11:32.09JTfor 1.4
11:32.18Kwakwahttp://ftp1.digium.com/pub/asterisk/ChangeLog-1.4.6
11:32.26EmleyMoorI'm not using anything 1.2 considers deprected, to the best of my knowledge
11:32.40JTKwakwa: there's a better file for the 1.2 to 1.4 transition
11:33.02JTanyway, if you have echo, and really want to get rid of it, look into hardware echo cancellation
11:33.04Kwakwaahh, my apologies
11:33.14k31thIf you guys where going to start using voip and needed pbx for your client base, how would you go about do it... Iv thought of just using Trixbox for them all this is by far the easy'st option... my other thought is perhaps CentOS + Asterisk and installing freepbx... only thing i would be missing is the phone manager? but i could do the provisioning by hand using info from the phone manufacture? and just drop the configs by tftp but would be more w
11:33.18EmleyMoorJT: That costs
11:33.38JTyes, good telecommunications equipment costs money :)
11:34.08KwakwaWe were quoted £10k just to upgrade to the latest software version of our intertel cab
11:34.17Kwakwathus we said "nah" and went for * :)
11:34.22EmleyMoorI wouldn't mind but if I want that I not only have to pay for that but for far more than I actually need
11:34.25Strom__k31th: avoid freepbx, trixbox, and the like
11:34.26JTk31th: trixbox/freepbx aren't really support here
11:34.38Strom__EmleyMoor: get the HPEC from digium
11:34.41JTk31th: btw, that line got cut off, try not to write so much at once.
11:34.43k31thStrom__: how would you do the admin then ?
11:34.52EmleyMoorStrom__: I have enquired
11:34.54Strom__it's free for as many channels as you have on your card
11:34.54k31thsorry yeah i was rambling a bit :p
11:34.56JTk31th: also, what is "easy'st"?
11:35.00Strom__k31th: vim
11:35.02Strom__or emacs :)
11:35.06JTk31th: ssh + vi
11:35.33k31thStrom__: you do it all by hand then
11:35.48JTk31th: which is what most people do
11:35.59k31thTakes more time?
11:36.02JTno
11:36.10Strom__gives you far more control
11:36.15Strom__and takes less time :)
11:36.23k31thInfact, I setup trixbox and iv done asterisk from scratch.
11:36.28JTmanagement tools waste far more time, when you get stuck with a problem you can't solve or a feature you want to implement
11:36.36JTmanagement tools like freepbx
11:36.37k31thand i got the asterisk from scratch done far quicker.
11:37.30k31thfreepbx just writes to the confs tho?
11:37.39Strom__no
11:37.59Strom__freepbx is the creeping thing from Hell that takes over your dialplan and makes debugging next to impossible
11:38.04k31thif this like the eq of a noob that knows nothing of unix using webmin ?
11:39.08k31thWat is the best distro for asterisk and do you guys use a packaged version? or src
11:39.31Strom__the best distro is whatever you can admin most effectively
11:39.35JTwhat distro are you most familiar with?
11:39.37Strom__and always compile from source
11:39.38JTi use debian
11:39.41k31thim a bit against installing src on distros like RHE or Debian / Ubuntu atm as it seems bad practice ?
11:39.55Strom__it's not bad practice
11:39.58k31thJT: iv used most.
11:40.00Strom__the packages lag
11:40.07alrsk31th: I've had good luck running packaged Asterisk on Debian.
11:40.24k31thDebian i would say has been most kind to me.
11:40.25Strom__security holes, bugs, etc etc etc etc
11:40.29Strom__i like debian
11:40.38JTk31th: you can always turn asterisk you compile into a package on your distro
11:40.40Strom__but i won't touch the asterisk package with a ten foot pole
11:40.52alrsStrom__: The Debian maintainer of Asterisk leaves holes open?
11:41.08k31thJT: yeah i would most likely do that if i go down that route.
11:41.26*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
11:41.29JTalrs: the version is just old
11:41.38Strom__alrs: the version of asterisk that comes with debian stable is 1.2.12.  we're on 1.2.20 now
11:41.40alrsJT: old is OK by me
11:41.42k31thDo you guys provision via tftp ?
11:41.44Strom__you tell me
11:41.48JTalrs: old = security holes
11:41.50JTnot so okay
11:42.00k31thI am thinking about getting Snom phones
11:42.04alrsJT: The Debian security team isn't patching holes?
11:42.06JTget polycom
11:42.24k31thJT: they are better?
11:42.28JTthe best
11:42.44k31thcan i work them with AD?
11:42.53k31thbe good for clients.
11:42.54JTad?
11:43.04k31thActive Directory
11:43.09JTthey have nothing to do with ad
11:43.13KwakwaJT: I have polycom 330's, do you have the latest sip firmware installed?
11:43.16JTnot sure what relation they'd have
11:43.16k31ththe phone book i mean.
11:43.24JTk31th: i don't have a 330 myself
11:43.28JTKwakwa: even
11:43.45KwakwaJT: Which Polycom phone(s) do you have?
11:43.48k31thsnom u can pull phone book form LDAP / AD
11:44.13JTKwakwa: 301, 430, 501
11:44.18Strom__http://www.jerkcity.com/jerkcity514.html
11:44.27k31thAnother thing, wat about click to call and or Outlook plugin?
11:44.34JTk31th: polycoms have a microbrowser, might do what you want
11:44.35KwakwaJT: That in a production environment or personal use?
11:44.41JTclick to call is phone independant
11:44.50k31thI know.
11:44.54JTKwakwa: that's just my ones i have for myself
11:44.59Kwakwaahh
11:45.02k31thbut im thinking how do i do that with straight asterisk.
11:45.09alrsJT: Which security advisory should I be looking at on 1.2.13?
11:45.17k31thalso the outlook plugin is handy.
11:45.25JTk31th: .call files
11:45.29JTor AMI originate
11:45.46k31thso u can right click contacts in outlook and select call.
11:46.01k31thtrixbox uses some thing called HUDlite.
11:46.08k31thnot sure if its FOSS tho.
11:46.24Strom__it's more like a flaming pile of dogshit than FOSS
11:46.35*** join/#asterisk KDan (n=KDan@wakiki.gotadsl.co.uk)
11:47.31k31thStrom__: lol
11:47.48k31thit does not feel to great but it does seem to work.
11:47.59k31thStrom__: know of an equiv ?
11:48.27Strom__polycom 601 sets with sidecars
11:48.35Strom__buttons > point and drool
11:49.04alrsStrom__: Which Polycom firmware do you need to be at for the sidecars to behave reliably?
11:50.01Strom__i just use the latest one
11:50.37Strom__which is 2.1.2 or somesuch right now
11:51.17k31thStrom__: thats a bit of hardware that sits next to the phone?
11:52.24Strom__k31th: it's an adjunct unit with 14 line appearance buttons
11:53.45k31thhttp://www.provu.co.uk/protalk.html is some thing i looked at but then i looked into it more looks like freepbx banged on an asterisk box on a solid state system.
11:55.40Strom__just build it yourself
11:55.51alrsI went to the HAM swap in west LA yesterday morning and went digging for crappy Intel 537 softmodems.
11:55.58alrsaka "Digium X100P"
11:56.07Strom__alrs: was that the TRW swapmeet?
11:56.13alrsYup
11:56.14Strom__alrs: don't waste your time
11:56.24Strom__that card works like junk
11:56.32alrsStrom__ I know they're terrible.
11:56.33Strom__damnit, i keep missing the swapmeet
11:56.38k31thI bought one of those lol... I got it and thought this looks like a fucking win modem ?
11:56.50k31thi paid like 8 quid of it.
11:56.50Strom__~cheap
11:57.09jbotcheap is probably 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.
11:57.09alrsI paid $2 this morning
11:57.09k31thfor*
11:57.10k31thlol
11:57.10alrsonly becuase I couldn't see haggling over it
11:57.13k31ththats like 1 UKP
11:57.35alrsThe clones are a lot more tolerable when you run them with oslec
11:57.44k31thoslec?
11:57.45alrsI'd never recommend anyone use one in a business
11:57.52alrsbut they're handy for providing a timing source
11:57.54k31thhaha
11:57.56drzedlittle supid question: (why) is it not possible to use a gsm-mobile e.g. w/ bluetooth as "gsm-gateway"? (why) is it neccesaary to by gsm-card costing arround 1k$?
11:58.14JTdrzed: you can
11:58.16alrsk31th: oslec is the shit
11:58.17JTchan_cellphone
11:58.18alrshttp://www.rowetel.com/ucasterisk/oslec.html
11:58.55k31thalrs: this is if my card does not have hardware echo cancel ?
11:59.22alrsk31th: yes
11:59.30drzedJT: without any (severe) disadvantages? would it be useable in a productive environment?
12:00.00JTdrzed: depends, are you going to have piles of phones? that doesn't sound ideal
12:00.35drzedJT: actually one (maybe two) would be certainly enough
12:00.53k31thJust installing debian now
12:01.11JTit wouldn't be as reliable as a gsm card or external gsm gateway imho, but should suffice for a lot of applications
12:01.35*** join/#asterisk gardo (n=gardo@121.97.211.162)
12:01.57alrsI've been digging through reams of Motorola PDFs trying to find a way to enable the PTT button using AT commands
12:02.09alrsI have a couple of boost handsets that have been flashed to enable mototalk
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12:05.32Strom__http://www.jerkcity.com/jerkcity190.html
12:05.43k31thone massivly shit thing i have noticed about trixbox is no raid support
12:05.45drzedJT: thx, i'll google arround for it a bit
12:08.05Strom__one massively shit thing i've noticed about trixbox is that it exists in the first place
12:10.27k31thlol Strom__ your pbx's on deb ?
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12:24.57k31thto get the latest stable source i take it i can get it from a mirror. i dont want to be pulling the cvs for a production box.
12:25.12JTasterisk.org
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12:26.31knarflyI just rebuilt using FreeBSD-6.2 and *-1.4.4
12:27.42knarflyLoaded all the options but I'm getting an error message about radius client. Can I turn this off somehow?
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13:39.18CpuIDhey ppls, any preferences for which zaptel echo canceller algorithm/engine to use?
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14:10.51monstertruckhi
14:11.02monstertruckis there a way to force a remote reboot on a spa3102?
14:11.19monstertruckhttp://x.x.x.x/admin/reboot will wait until the unit is idle
14:11.44k31thcan the linksys SPA phones have an xml address book ?
14:11.56k31thor a ldap address book ?
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14:16.42polerinmonstertruck: a hammer?  ;P
14:17.07monstertruckpolerin, remotely .. woud have to throw the hammer :)
14:20.51k31thcomcast are a cable company in the US?
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14:22.31JTk31th: yes, we call them spamcast here
14:22.40k31thlol
14:22.53JTso many owned pcs with spambots
14:23.03crimethinkerindeed
14:23.25k31thand botnets no doubt.
14:23.47k31thud think they would just block port 25 inbound ?
14:23.57polerin..
14:24.01polerinno generally not :P
14:24.07JToh, irc bots too
14:24.25polerinthey do block off port 25 if you are detected sending off too much
14:24.34k31thirc ahhh mainly for botnet stuff  no doubt
14:24.45polerinthen you get the fun of calling abuse
14:24.50polerinwhich is running 6 months behind
14:24.57rob0There's no use in blocking 25 inbound.
14:25.07polerinrob0: they block it outbound
14:25.10k31thmy isp blocks port 25 only port they do block u have to call them to get it opened and they do an open relay test.
14:25.13polerinrob0: but only if you abuse it :P
14:25.20k31throb0: they can just spam out i guess yeah
14:25.31rob0yeah, my outbound 25 still works.
14:25.39rob0last I checked
14:25.43polerink31th: comcast is one of the largest isp's if not THE largest isp in the US
14:25.57k31thare they good?
14:26.01polerinthat depends
14:26.05polerinwhat are you expecting from them
14:26.06polerin:P
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14:26.25crimethinkerthe largest isp is uunet.
14:26.26rob0Is there any "good" ISP in the USA?
14:26.41crimethinkerrob0: speakeasy, if you don't mind being a best buy customer.
14:26.56polerincrimethinker: ...  I thought UUnet was backbone only  **shrugs**
14:27.03rob0yeah, I've heard of speakeasy, but they're not anywhere near me.
14:27.20polerinthey are a Covad reseller
14:27.23crimethinkeryou asked about USA. you got a USA answer
14:27.41polerin(speakeasy that is)
14:28.07polerinanyway, comcast i can talk about, I work the the assholes :)
14:28.09rob0they're not national, or at least not in the areas where I need service.
14:28.24polerinspeakeasy is a consumer isp only .. no?
14:28.34polerinthey don't do sdsl or buisness turnaround
14:29.19polerincomcast is mostly a consumer ISP, but it does have a buisness product.  Main differentiation is the avalibility of a static and a supposid 4 hour turnaround
14:29.39polerin(supposid because apparently no customers actually read their contract and try to enforce it)
14:30.06rob0I would consider upgrading my home comcast to business, but I don't trust them to provide business-class service.
14:30.22polerinrob0: What are you expecting of them?
14:31.30rob0Well, I'm in a rural area with poor electric service, outages and flickers all the time. I would want the entire infrastructure of nodes to be on power backups.
14:31.39polerinhmm
14:31.48rob0As it is, when the power flickers, my connection dies.
14:31.52polerindo they advertise Digital Voice (heh) in your area?
14:31.58rob0not sure
14:32.01polerinok
14:32.09crimethinkerpolerin, speakeasy does sdsl.
14:32.17rob0Florence, AL if you're interested.
14:32.22polerinIn general they don't roll out DV untill all of the power supplies have battery backups
14:33.20polerinrob0: I wouldn't even know where to start asking about that info, and I wouldn't trust customer service to tell you a truthfull answer even if  they were local to your area.  (our call center handles part of AL,MI,and a couple other areas XD)
14:33.29polerinis your modem on a ups?
14:33.41rob0of course ... an enormous one.
14:33.48polerinjust making sure :p
14:33.50rob0along with the computer
14:34.00polerinthen yeah, it's likely that they don't have battery backup.
14:34.17rob0And you're right, they did lie to me last time I asked.
14:34.32polerinmost power flickers with backup power supplies would give you a moment of packet loss but not lost synch
14:34.44rob0CS is in Huntsville during the day, Nashville after hours.
14:35.02polerinnods.  What did you ask them?
14:35.43rob0hmmm, this was some time back, don't remember exactly.
14:36.11rob0MI is Michigan, perhaps you meant MS? :)
14:36.17polerinThere is alot of stuff the Call center is asked that they just don't have the training or access too, and they can't pass it along because it can be hard to find out who's actually responsible for that stuff (In metro nashville I would ask Engineering about power supplies, or possibly the PM/DM Line techs)
14:36.40polerinyeah MS ;P
14:36.51polerinok. actuall * related question.
14:36.59rob0The call center really sucks.
14:37.20polerindon't get me started on them.  I spend all day cleaning up their mess
14:37.38rob0but, usually I can get by with the "L" word, because I make it clear I'm not asking them to "support my OS."
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14:37.52polerinheh
14:39.02polerinI got freaked out, one of our outer county techs who I talk too a fair bit is a fedora fan..  This is weird because the techs I talk frequently are not generally tech savy enough to do more than plug the modem into the ether net port (occasinally plugging the usb in as well)
14:40.04polerinbut to be fair, the selfprov software doesn't run half the time, installs broadjump, and is flakey.  So he tends to call in to have me push the modem :P
14:40.18polerinanywayregister => phonnum@sip.broadvoice.com:passwerd:phonnum@sip.broadvoice.com
14:41.22polerinthis comes in at the right context, but with the phone number as the extention, even if I specify /s at the end
14:41.42rob0"Comcast Digital Voice service is not yet available at the address you entered." Heh, but I've been using VoIP from there for > 3 years. :)
14:42.02polerinyeah
14:42.21polerinit meens they don't have a switch there, and don't have the plant in good enough shape to do it yet
14:43.25polerinis it something in the peer setup?
14:43.43rob0So put the phone number in your dialplan?
14:44.04polerinI did.  Trying to figure out why it's doing that though, just for my understanding.
14:44.33polerinis s more for catching zap lines that don't have specific stuff in it?
14:44.35rob0I can't help there, I kind of lurk here to try to pick up bits of understanding as well.
14:44.41polerinlol
14:45.05rob0I've got basics working, would eventually like to do fancy things.
14:45.06polerinwell, I'm a complete newb to *, barely got the peer working last night
14:45.34polerinI've got in and out now, but there are quirks that I want to understand before I move on and get the real stuff set up
14:45.42rob0You can tell that in the 3+ years I haven't put a lot of time into it. :)
14:45.49polerinheh
14:46.09polerinI've been focusing on webdev stuff mostly.
14:46.48polerinbut my wife is starting her own buisness, so I figured there's no better time than now to get the phones working
14:47.45crimethinker"The OSU Medical Center is requesting proposals for network monitoring system to enhnace its network quality of service."  whoa boy, I wonder how many folks would just go install nagios for them and charge 50000
14:52.18polerinhmm, so is there any way to change what extension when including?
14:53.31polerins/what extension/the effective extension/
14:53.55polerinI like that :D
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15:03.14*** join/#asterisk linex (n=blah@124.82.105.208)
15:12.04linexok, so if my asterisk is behind a nat and my clients are also behind a nat. Is it doable to get connection ?
15:12.33linex.. and my clients are behind ANOTHER nat ....
15:13.40*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
15:13.53monstertrucklinex use a stun server
15:14.03*** join/#asterisk mikex_ (n=mikex@cpe-66-69-143-156.houston.res.rr.com)
15:14.29linexmonstertruck : that'll solve the situation ?
15:14.38monstertrucklinex, yup
15:15.12monstertruckhere: stun.fwdnet.org, that one is public
15:16.31monstertrucktheres also a setting in sip.conf you need to change if * is behind nat
15:16.37monstertruckuse externip=whatever
15:16.53monstertruckwhatever your outside ip is
15:17.34*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
15:17.36monstertrucklinex, and .. if you are using iax, disregard all that
15:18.16linexiax is better or sip is better ?
15:18.35mostyiax is a nicer protocol, but almost no hardware supports it
15:20.24linexstun.fwdnet.org - bad link
15:22.10linexI have spare machines, can I run my own stun server ?
15:23.03mostywhat are you trying to do?
15:25.23SuPrSluGzaptel 1.4.2.1 build error zaptel-base.c:787: error: `fcstab' undeclared (first use in this function) . kernel = 2.6.7   Any ideas?
15:26.11mostythat kernel is ancient, i'd recommend upgrading that first
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15:30.30linexmonstertruck: my asterisk is behind a nat and my clients are also behind a another nat. I'm trying to get them connected.
15:31.43monstertrucklinex, what are your clients using to make calls?
15:31.50mostylinex, do you have a linux machine at each end?
15:31.55monstertrucksoftphones? hardphones?
15:32.01linexsoftphones
15:32.07linexyes linux machines
15:32.11monstertrucklinex, then use iax
15:32.15monstertruckand forget about nat
15:32.39mostylinex, run asterisk at each end and use IAX between them. the sip softphones will talk to the local asterisk machine
15:32.59linexyou mean the problem is only with sip, with iax no problem . Is that correct ?
15:33.09monstertrucklinex, yes
15:33.38linexhmm interesting.
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15:34.03mostylinex, plus IAX uses less bandwidth than SIP
15:34.21linexany softphones that runs on windows that also can use iax ?
15:35.08mostylinex, probably not any good ones. just use sip between the softphone and a local asterisk box, and let the asterisk boxes talk using iax
15:35.57linexm
15:36.18linexmosty: u don't understand, I have a nats.
15:36.39linexI mean nat behind asterisk and another nat behind clients
15:37.01linexusing iax , this is a non-issue. So thats solved
15:37.18RypPnlinex: idefisk is an iax client for windows
15:37.30linexi run ekiga or idedisk on the linux machines clients
15:37.41linexoh ya
15:37.42SuPrSluGmosty: I'd be doing a remote kernel upgrade
15:37.49mostylinex, the clients are on the same lan as your linux box though right?
15:37.59linexidefisk has iax , yeh
15:38.09linexmosty nope
15:38.29linexclients behind one nat
15:38.37linexasterisk behind another nat
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15:39.07mostyahh ok
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15:39.17linexclients all behind one nat. not different nats
15:40.00linexwell actually ... some behind other nats
15:41.08linexso I may have clientA behind nat1 , clientB behind nat2 and the asterisk box behind nat3. So if I'm using iax, this is a non-issue, right ?
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15:41.45linexand if I use sip, I need to use a stun server. Ok I understand that perfectly.
15:43.11mostynormally you'd put the asterisk box on a public ip and use sip or iax whatever you prefer
15:43.44mostyif the asterisk box is behind nat then sip is a pain in the arse
15:44.23linexyeah I know but this is just something I'm playing with at home.
15:44.50linexI'm gonna have asterisk at home and have people use it to talk to each other
15:45.18mostyit would be simpler to just use FWD or something
15:45.44linexusing iax sounds simpler
15:45.53mostyFWD is even simpler
15:45.57mostyno asterisk box at all
15:46.23linexI'm sorry I have not read the manual. I don't know what FWD means
15:47.17linexis it like direct connection . softphone to softphone ?
15:47.20mostyhttp://www.freeworlddialup.com/
15:47.29linexthanks
15:49.25linexWOW
15:49.39*** part/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com)
15:51.31linexI bet those FWD guys using asterisks
15:51.47linexa whole bunch of asterisk servers
15:53.34mostyi would think it's ser or openser, not asterisk
15:53.53mostysince they handle much higher loads than asterisk can
15:57.45rob0Last time I tried with FWD, their SIP was broken. IAX worked fine. But this was not recent.
15:59.44*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
16:02.02linexSo when two client made a connection and start talking, the connection or talking is made thru asterisk or is direct like p2p apps works ?
16:05.42linexI mean if the conversation goes thru asterisk, then it is possible for asterisk to record the conversation or else recording can only be made on the client side.
16:11.38mostylinex, depends how asterisk is setup
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16:22.27rue_mohrwonder how you measure THD...
16:22.36rue_mohrdigitally that is
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16:55.27WindBack[TK]D-Fender,
16:55.30WindBackhello
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16:56.14Dovidis there jitter buffer for sip on 1.2.x ?
16:57.35WindBackwhat port I need to forward in my NAT router if I'm using Xlite in a client inside de LAN???
16:57.47WindBack5060 tcp and udp and???
16:58.04WindBacka range of UDP
16:58.11WindBackbut what range???
16:59.04[TK]D-Fender5060,10000-20000 *ALL UDP*
16:59.35[TK]D-FenderWindBack, thats for * being behind NAT.  if youa re talking about a REMOTE X-lite behind NAT you don't ahve to forward ANYTHING
17:00.36WindBack[TK]D-Fender, yes but in this case I have to use a stun server
17:00.47[TK]D-FenderWindBack, No, you don't
17:00.51WindBackor not??
17:00.57Dovidno stun with asterisk
17:01.07Dovid(or rather asterisk dosent have a setting in it for STUN)
17:02.38*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
17:03.10[TK]D-FenderAsterisk doesn't support STUN, and your client does not need it either in this case.
17:03.22WindBack[TK]D-Fender, If I'm using Xlite inside a LAN who have a NAT router, and the asterisk is outside this LAN... Do I need to forward anything for the clien???
17:03.55[TK]D-FenderWindBack, again : NO
17:05.44WindBack[TK]D-Fender, excuse me for say the same thing again and again, but I red in a lot of place that I have to forward ports
17:05.59[TK]D-FenderWindBack, Get over it.
17:09.31WindBack[TK]D-Fender, ok, thank you
17:09.51WindBackI'll be back in a minute
17:10.43Dovidis there jitter buffer for sip on 1.2.x ?
17:11.13[TK]D-FenderDovid, Go DL it and find out
17:11.15rue_mohrhttp://en.wikipedia.org/wiki/Decibel#Voltage
17:11.23rue_mohrphones are 600ohm aren't they?
17:11.44[TK]D-Fenderrue_mohr, Sounds like what I remember.
17:12.01[TK]D-Fenderrue_mohr, impedence mismatch is one cause of echo.
17:12.05rue_mohrso 1mW is .775Vrms
17:12.16rue_mohrI'm glad I dont have echo
17:13.21rue_mohrthere is a digital 1mW tone gen in asterisk isn't ehre?
17:14.43[TK]D-Fenderhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt
17:20.16rue_mohrdialing on my phone is interesting, "is it distorted, or does it SOUND distorted?"
17:20.47rue_mohrCAN YOU HEAR ME NOOWWW!!!???  }:)
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17:43.41JoelSolankihi all
17:45.31JoelSolankii have some requirement for call monitoing for asteris.
17:45.42JoelSolankii want to monitor all incoming and outgoing calls in asterisk and display in browser refresing every 30 seconds
17:45.51JoelSolankiis there any open source code available for this ?
17:45.58JoelSolankior if i want to create how can i get incoming/outoing data in real ?
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19:21.22BSD_Techok I found a issue I am baffeld over
19:21.34BSD_Techbrb
19:21.59Strom__it stands for "be right back" - i don't understand what's so baffling about that one
19:22.21dc3aesha
19:23.09BSD_Techok I found a issue with 1.4.5
19:23.56BSD_Techthat is starts reandoming going threw  the [app-name] when you try to use a the # key
19:24.08Strom__uh
19:24.11Strom__English please
19:24.32BSD_Techwhen I hit the # key
19:25.01BSD_Techit starts randomly going to diff [app-??????]
19:25.19Strom__under what conditions are you pressing #?
19:25.22BSD_Techinsted of what its mapped to wich is the company dir
19:25.33BSD_Techhti # dial
19:25.41BSD_Techit should go to the company dir
19:25.49Strom__oh christ, you're not actually using # as an extension all on its own, are you?
19:26.00BSD_Techbut its not it goes to random [app?????]
19:26.04Strom__didn't you listen to me when I said "don't use # as an extension"?
19:26.27Strom__# means "I am finished dialing - put the call through"
19:26.27BSD_TechI have always used # for coampany dir
19:27.00BSD_Technever had a issue till 1.4.5
19:27.08Strom__well, try 1.4.6 then
19:27.13BSD_Technow its screwing up
19:27.17Strom__see if that fixes the bug
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19:27.28BSD_Techwhen did i.4.6 come out
19:27.36Kwakwa29th
19:27.39Strom__read the topic
19:27.39BSD_Techok
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19:28.08BSD_Techgrrr
19:28.09BSD_Techok
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20:00.36DaminI just made some awesome chicken soup..
20:05.45rob0Aha! And to be on topic, you can SIP it now.
20:09.57BSD_Techok I found the issue and fixed it
20:10.09BSD_TechI removed app-
20:10.26BSD_Techand it stopped randomly choosing a applicaion
20:10.41*** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca)
20:10.49BSD_Techwhen I dialed #
20:11.37Strom_Cwhat "app-" are you talking about
20:16.15[TK]D-FenderStrom_C, he's working on a dialplan thats grown too big for its britches :)
20:17.29BSD_Techno it only broke on 1.4.5
20:17.30*** join/#asterisk foug1 (n=foug@cpe-24-28-152-69.satx.res.rr.com)
20:17.43BSD_TechI just tested it on a 1.4.4 box ad does not actup
20:17.49foug1hi, does asterisk have a respository to download from? i'm having trouble finding the depencies so i can use ,/configure and make
20:20.13_VoiceMeUp_COMlittle question for yall i really need a macro for VM with multiple extensions ? else the Voicemail ( su#@default) will not let me press 0 correect ?  so to access the VM admin from the number directly its macro to exten => o,1,VoiceMailMain
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20:21.54[TK]D-Fender_VoiceMeUp_COM, this has NOTHING to do with macros
20:22.22_VoiceMeUp_COMyes but the way to go around that is that ?
20:22.38_VoiceMeUp_COMor is there another way
20:22.49DovidTK: May I PM ?>
20:23.02[TK]D-Fender_VoiceMeUp_COM, Whereever you call Voicemail, you'll want to have an "o".
20:23.13[TK]D-FenderDovid, if you're looking to hire me, sure ;)
20:23.25Dovidlol
20:23.26_VoiceMeUp_COMyes.. but if i have 100 extensions.. with 100 different acocount #'s .. then i can only have 1 o
20:23.42_VoiceMeUp_COMor hmm let me try
20:23.43russellb_VoiceMeUp_COM: your nick drives me crazy
20:23.52Dovidi update the SpanDSP page. for the future if an1 asks for it on 1.4.X
20:23.54Dovidhttp://cgi.ebay.com.au/Mini-Asterisk-PBX-VoIP-PSTN-in-Linksys-WI-FI-router-USB_W0QQitemZ330140619065QQihZ014QQcategoryZ61841QQrdZ1QQcmdZViewItem
20:23.55Dovidoops
20:24.02[TK]D-Fender_VoiceMeUp_COM, You are not understading the basics of extenions.conf!  This isn't a VM issue!
20:24.03Dovidhttp://www.voip-info.org/wiki/view/spandsp
20:24.06russellbif you want to have that nick for customeres to find you, fine.  but *please* try to use something else for what you use for discussion ...
20:24.44Dovidrussellb: advertising ;) some people are advertising ******
20:24.44[TK]D-Fender_VoiceMeUp_COM, "o" needs to be accessable in the context that CALLS's VoiceMail.
20:24.44russellbright, and this is not a commercial chat room ... advertising is not welcome
20:24.46Dovidhehe
20:24.46_VoiceMeUp_COMhold on let me make russellb happy
20:24.51Dovid:)
20:25.00russellbthank you :)
20:25.04Voicemeupthat better ;)
20:25.29russellbwell, i'm not thrilled, but it's better :)
20:25.53Strom_CVoicemeup: presumably russell wants something that doesn't include the phrase "voicemeup"
20:26.05russellbi'll compromise on that
20:26.16russellbthe whole _COM thing was really what was over the edge for me :)
20:26.26[TK]D-Fenderrussellb, we don't negociate with TERRISTS!
20:26.32russellbhehe
20:26.33Strom_Cyow
20:26.35Strom_Cspelling
20:26.37Strom_Couch
20:26.59[TK]D-FenderStrom_C, its a GWB joke FFS....
20:27.18*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
20:27.22Strom_Cyeah, but "negociate"?
20:27.25Strom_Cplease :)
20:28.11[TK]D-Fenderme grabs whats left of Strom_C's sense of humour, puts it in a brown paper bag, stomps on it, lights it ablaze, douses it in horse-piss (translation : Budweiser), lights it up AGAIN, stomps it out and hands it back.
20:28.24Strom_Cthanks !!!
20:28.29[TK]D-FenderAnd darn it for not having a leading slash!
20:28.41russellbpwned
20:28.59[TK]D-Fenderrussellb, I did enough in there to negate reproach ;)
20:31.59BSD_Techok you say extensions ahould never start with # yet in features.conf #72 for autoparking
20:32.36russellbthat's not an extension :)
20:32.41BSD_Techand when enabled it goes right to a busy tone
20:32.45russellbthat's a call featuer ...
20:32.49russellbfeature*
20:32.54Strom_CBSD_Tech: busy and reorder tones are not the same thing
20:33.05Strom_Creorder tone is twice as fast as busy tone
20:33.40BSD_Techok then its getting a reorder tone
20:33.52Strom_Cbusy means "the party on the other end is on the phone" while reorder means "something is terribly wrong"
20:33.55russellbsee above ... that's not an extension
20:36.38russellbin the same way that the setting for what to dial to do a transfer, is also not an extension
20:48.54BSD_Techok try to put this in better terms
20:49.06BSD_Tech1 I just updated to 1.4.6 and same issue
20:49.35BSD_TechI changed all the #XX number exren I commented themm out
20:49.49BSD_Techso there is no exten wiht # at the beginning
20:50.30BSD_Techbut when you hit the # key and dial its juping to my disa setup
20:50.39BSD_Techand asking me fir username and exten
20:50.51BSD_Techextension and password to be correct
20:51.09Strom_CBSD_Tech: but /when/ are you dialing #
20:51.12Strom_Cat the dial tone?
20:51.15Strom_Cor at an IVR menu?
20:51.25BSD_Techso figure this out if no exten has  exten =#, why is it jumping to disa
20:51.55BSD_Techat dial tone
20:52.00Strom_Csigh
20:52.07BSD_Techand there is no ivr
20:52.15Strom_Cdespite the fact that you should never have any reason to dial # at the dial tone, i'll debug it
20:52.25Strom_Cpastebin your extensions.conf as it is right now
20:52.28BSD_Techand 3 alone does not mean end of dial
20:52.37Strom_CBSD_Tech: uh, it should
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20:52.51Strom_Cs/3/#/
20:52.54BSD_Tech3/#
20:53.07BSD_Technot on poly coms
20:53.14Strom_CBSD_Tech: yes on polycoms
20:53.19Strom_Cnow pastebin the dialplan please
20:53.22BSD_Techwith thier default dial plan
20:53.28BSD_Techbrb
20:53.57Strom_Cthis is why I keep saying you should understand telephony /before/ you touch asterisk, but no one ever listens to me of course
20:57.26BSD_TechI have been using asteriusk for over 5 1/2 yeasr and never had this issue
20:57.39BSD_TechI have writen many dial plans
20:57.53BSD_Techand never had this issue with the # key before
20:58.16BSD_Techpastebin.ca seems to not be responding
20:58.26Strom_Cwell then use one of the other pastebins
20:58.42Strom_Cand just because you know asterisk doesn't mean you have a firm grasp of numbering plan theory :)
21:00.20BSD_TechI am they are all being major slow
21:01.02*** join/#asterisk WindBack (n=Administ@host19.190-31-201.telecom.net.ar)
21:01.12BSD_TechI pasted on pastebin.com and its taking forever siorry
21:01.39Strom_Cpastebin.com is also screwed
21:01.42Strom_Cwhat about....
21:01.45WindBackI have just test IDEFISK
21:01.47WindBackwowww
21:01.51Strom_Chttp://pastie.caboo.se/
21:01.56WindBackgood softphone
21:01.58WindBack!!!
21:02.44BSD_Techhttp://pastebin.com/940122
21:03.03Strom_CBSD_Tech: what context do your phones live in?
21:03.24BSD_Techdefault right now
21:03.27Strom_Cok
21:03.34Strom_Cthis is taking forever to load
21:03.35BSD_Techits at the bottom
21:03.37[TK]D-FenderStrom_C, [couldnt-possibly-include-more-crap] ;)
21:03.44WindBackIDEFisk the best softphone
21:03.48BSD_Techits all documenting
21:03.54Strom_CWindBack: yes, you said that already
21:04.02[TK]D-FenderWindBack, No it isn't, but its probably the best FREE one.
21:04.19WindBack[TK]D-Fender, yesss
21:04.23WindBackfree
21:05.00BSD_Techhttp://pastie.caboo.se/75227
21:05.04BSD_Techthat went fast
21:05.15WindBack[TK]D-Fender, I saw that idefisk use a stun server
21:06.10Strom_CBSD_Tech: there isn't even a [default] context in this dialplan
21:06.15WindBack[TK]D-Fender, but before you tellme that I don't need it
21:06.16BSD_Techthe whole did not paste
21:07.12Strom_CBSD_Tech: well then upload as a text file to some site you host and link to that
21:07.46BSD_Techok
21:07.46WindBackSomebody could use IDEFisk in linux?? Because I prove it, but it didn't found in my debian
21:07.47BSD_Techbrb
21:08.35WindBackI did hear nothing
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21:09.16WindBackStrom_C,
21:09.23Strom_Chi
21:09.48WindBackStrom_C, colud you use idefisk in linux??
21:10.39Strom_CWindBack: well, gee, let me look on the site
21:10.41Strom_Chttp://www.asteriskguru.com/idefisk/free/
21:10.49Strom_Cdon't pay attention to that "LINUX" button
21:12.00WindBackStrom_C, I installed it in my debian, but it didn't found
21:12.12Strom_Cin English, please
21:13.01BSD_TechStrom__, https://goldenfeather.homeip.net/static/extensions.conf
21:13.04WindBackStrom_C, ok, you are not a good person whit me
21:13.15WindBackStrom_C, thank you
21:14.52Strom_CBSD_Tech: now pastebin CLI output of what happens when you press # on your phone
21:16.58BSD_Techhttp://pastie.caboo.se/75231
21:17.09BSD_Techits now jumping to app-wish
21:18.05Strom_CBSD_Tech: well here's what's happening
21:18.28Strom_Cwhen you press # on the polycom phone, the phone sends the call to asterisk, assuming you're indicating the completion of a dialing sequence
21:18.43Strom_Casterisk receives a call with no called party number
21:19.07Strom_Cso, having worked with asterisk for years and years now, you should know that under those conditions, asterisk will look for extension "s"
21:19.40Strom_Cand some of the app contexts you've included do have "s" extensions
21:20.16BSD_TechI see a double sided blade in the
21:20.37BSD_Techbecaus eit should look for a exten = # and if no exten the return busy
21:20.42Strom_Cno
21:20.46Strom_Cthe phone doesn't send the #
21:20.50BSD_Techits use to
21:20.57Strom_Cit shouldn't, by default
21:21.00BSD_Tech1.4.3 did and iin 1.2.XX it does
21:21.23Strom_CIF the phone sends "#" then asterisk will match on # and not s
21:21.37Strom_Cbut if the phone sends "" then asterisk will match on s
21:21.55BSD_Techthis ook so now to ceck my phone dial plan then
21:22.00Strom_CNO
21:22.04Strom_Ccheck your asterisk dialplan
21:22.16Strom_Cyou should never assume the phone will behave a certain way and build your dialplan around that
21:22.23BSD_Techwell polycome may have screwed thier ne firmware also
21:22.33Strom_Cyou should build your dialplan around established standards and then get your phones to conform to that
21:23.08Voicemeupwahts priority -1
21:23.09Voicemeupin ARA
21:23.23Strom_Cthe simple solution is to (a) never use # as an extension for anything, and (b) never include a context which contains an "s" context
21:23.40Voicemeupseems it s looking for that upon the rturn of a dial.. its equiv in ara of +101 ?
21:26.54EmleyMoorNever write a dialplan routine you can't be sure of without testing it
21:27.20EmleyMoor(in other words, either test everything or keep it so simple it doesn't need testing)
21:27.36Strom_CI even test the simple stuff
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21:28.07EmleyMoorStrom_C: Some of my dialplan looks complicated, but bears in mind that one of the phones on my system is rotary
21:28.17Strom_CEmleyMoor: that shouldn't make a difference
21:28.24Strom_Cdigits are digits to asterisk
21:28.45EmleyMoorWell, you can't use * on rotary phones - but other than that, yes, I agree
21:29.15Strom_CEmleyMoor: yes you can...pulse the hookswitch eleven times :)
21:29.23Strom_Cand for #, pulse it twelve times :)
21:29.27*** join/#asterisk ruied (n=ruied@bl7-218-132.dsl.telepac.pt)
21:29.30crimethinkerAnyone have any hints where to sell an antique phone at a fair price?
21:29.38EmleyMoorI might try that just to see
21:29.39Strom_Ccrimethinker: define "antique"
21:29.44Strom_Ci collect old phones :)
21:29.48EmleyMoorcrimethinker: What country is it from?
21:30.27Voicemeuphey i just realized my prob.. once you go to mysql ARA architecture from withitn static dialplan you cant go back.. its a one way ticket
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21:32.01*** part/#asterisk wedhorn (n=wedhorn@dsl-58-6-89-11.act.westnet.com.au)
21:33.13Strom_Ccrimethinker: best idea is to just take some photos and show me :)
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21:33.51EmleyMoorIn fact, what I really want is a mid-1970s Hull phone
21:35.01currachI have a late 1970s rotary dial - not sure if it is a "Hull" phone. It is kind of cream plastic.
21:35.15Voicemeuphttp://www.engadget.com/2006/08/14/the-500-gsm-rotary-phone/
21:35.22Strom_CVoicemeup: old old news
21:35.29Voicemeupyep
21:35.36Voicemeupjust racting an an old inquiry
21:35.39Voicemeupreacting
21:35.41EmleyMoorcurrach: If cream, it's probably GPO
21:35.56rudholmhey Strom_C can I use the fourth column digits for extensions?
21:36.06Strom_Crudholm: you know, i've never tested
21:36.14rudholmand are those mapped to pulses?  13, 14, 15, 16 pulses?  :)
21:36.17Strom_Ci don't have an autovon phone
21:36.25rudholmI do, actually...
21:36.27currachThat would be probably be right, issued by the Post And Telegraphs in Ireland.
21:36.31EmleyMoorrudholm: I believe you can - if you have a phone that supports. As for pulses, I pass
21:36.36rudholmI should plug it in to one of those Zap channels I just configured
21:36.40Strom_Crudholm: yes!
21:36.41Strom_Cdo it
21:36.42EmleyMoorcurrach: Ah, an Irish phone
21:36.45Strom_CPEER PRESSURE
21:36.51Strom_Cyou KNOW you want to plug it in
21:36.55rudholmhahaha
21:37.02rudholmcoin line!
21:37.06Strom_CEVERYONE'S using the fourth column now!!
21:37.11[TK]D-FenderBest part about my God Complex..... no peer pressure ;)
21:37.27ruiedwhat is the chan_misdn for?
21:37.35rudholmall phreaks have 1PC Coin Service Lines
21:37.45Strom_Cruied: ISDN
21:37.47rudholmI don't even know *why* they let you in to the 2600 meet-ups :)
21:37.53JTruied: it's to give you the shits
21:37.59JTalso for bri
21:38.00Strom_Crudholm: something about Kinko's
21:38.07rudholmblah blah blah
21:38.18Strom_Chehe
21:38.22rudholmit's all about the 1PC now
21:38.31rudholmand NI1 BRI on Asterisk.
21:38.47Strom_C<girl fight> DENTING YOUR LUNCHBOX!!   I SWEAR I WILL BITE THIS BENDY DOLL'S HEAD RIGHT OFF!! </girl fight>
21:39.13JTi'm curious as to whether the sangoma A500 will support NI1 BRI
21:39.25JTdrivers are yet to be released
21:39.34J4k3real men only use test sets
21:39.37J4k3to make calls
21:39.40rudholmI seem to recall something about accepting ABCD somewhere in asterisk-land.
21:39.48Qwellrudholm: it should work fine
21:39.52JuggieABCD are dtmf's
21:39.57J4k3who needs a cellphone when theres below ground phone wiring with peds every 150 meters? :D
21:40.02ruiedJT, to talk to asterisk right? I've already compiled successfully mISDN and mISDNuser... but I'can't compile chan_mISDN, it's reporting errors....
21:40.13currachNight everybody
21:40.18JTruied: what card are you using?
21:40.18Strom_CJ4k3: Harris/Dracon TS44A FTW
21:40.31rudholmStrom_C: I should tell people my phone number is "A"
21:40.35J4k3Strom_C: its only cool if you stole it from a telco guy, though :D
21:40.36Strom_Chahahahahahaha
21:40.41Strom_CJ4k3: mine says "SBC"
21:40.47J4k3Strom_C: sweet!
21:40.54ruiedJT, w6692pci
21:40.58ruiedwinbond
21:41.00JTok no idea what that is
21:41.04JTerr ok
21:41.08JTyou sure it's a bri card?
21:41.08rudholmMine says Pacific Telephone & Telegraph Co.
21:41.09Strom_Cgoes with my SBC hard hat and baseball cap
21:41.16ruiedJT yes
21:41.41J4k3Strom_C: do you have a white chevy truck with a grey konig bed with some random "AT&T" stickers? :D
21:41.57rudholmhe has a Honda with random Bell System stickers.
21:42.29Strom_Cyes
21:42.32J4k3I pissed off some verizon tower installers because I had a nicer harness than they did
21:42.35J4k3hehe
21:42.38ruiedJT: compiling chan_misdn ends up with: make: ** [chan_misdn.o] Error 1
21:42.52JTruied: ok, i'm not really into supporting misdn, btw.
21:43.06Strom_Chttp://www.flickr.com/photo_zoom.gne?id=616288871&size=o&context=set-72157600471576143
21:43.13Strom_C^^^^
21:43.26ruiedJT ok
21:43.47Strom_CJ4k3: double bonus points if you can tell me what the license plate refers to
21:44.39ruiedchan_misdn is to make asterisk "talking" with misdn driver right?
21:45.14J4k3hahaha you DO have a honda with a bell system sticker!
21:45.27Strom_Coh, you fail
21:45.34Strom_C(311) 555-2368
21:45.35J4k3wasn't ever much of a phreak tho.  kinda young, in too big of a town (so we had a real switch)
21:45.42Strom_Cthe standard bell system fake number
21:45.47J4k3hah
21:46.27J4k3http://www.intrastar.net/~jsuter/6-13-07/0613071534.jpg
21:46.31J4k3(awful cellphone camera)
21:46.53Strom_Cheh, nice
21:46.55Strom_Cwhere is that?
21:46.59Juggieyou know what sucks, some loser company called grandcentral that is probally not based on asterisk is going to get bought out for that service by google, when any one of us could have created that same service on asterisk :(
21:47.00J4k3east texas
21:47.16Strom_Cah cool
21:47.25ruiedI'm having problems compiling chan_misdn, the last line compilation error is: "make: ** [chan_misdn.o] Error 1"
21:47.57J4k3google spoogle.
21:50.06JTruied: that is a completely useless error, you're wasting your time
21:51.55ruiedJT: how or what should I look for?
21:52.08JTthe real compilation error
21:54.32ruiedok, here it is :" error: to few arguments to function 'ast_moh_start'"
22:00.32*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
22:01.21*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
22:04.24ruiedcan anyone give me a help with chan_misdn compilation?
22:11.41BSD_Techcome to find out it is the sip.cfg for my polycoms
22:12.12BSD_Techone of the other techs was editing it and changed things and removed the timeouts and the dial strings
22:12.44BSD_Techso the phone was not sending the right dial stirngs
22:15.51*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
22:16.23Strom_CBSD_Tech: my argument still stands
22:16.27Strom_Cyour dialplan is bonkers
22:16.29Strom_Cfix it
22:24.12apturaStrom. MY CID indicator is working. no idea why.
22:24.27apturaI mean vm indicator
22:29.04*** join/#asterisk aao_pwner (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net)
22:44.45*** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net)
22:58.10*** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com)
23:05.07*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
23:06.02*** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net)
23:06.21iPod-nanoOK, I need one very simple, straight-forward answer.
23:06.53JTAll we need is one very simple, straightforward question.
23:06.56russellbare you on IRC from an ipod?  that's neat
23:07.01iPod-nanoCan I use Asterisk to bridge between my piece of junk hardware ATA and another VoIP service?
23:07.10russellbyes
23:07.13JTyes, as long as the ata isn't locked
23:07.20iPod-nanoJT, it's not.
23:07.27iPod-nanoIt just sucks.
23:07.38iPod-nanoIt doesn't support STUN.
23:07.46JTwhat model is it?
23:07.47JTmeh
23:07.54JTstun, you rarely need to use
23:07.55iPod-nanoAnd I don't have the ability to connect it directly to the internet.
23:08.23iPod-nanoIt's a D-Link DVG-1120S (actually a firmware-flashed 1120M).
23:08.51*** join/#asterisk Mavvie (n=edwin@ppp121-44-63-246.lns2.syd6.internode.on.net)
23:09.10iPod-nanoI use Gizmo Project, and all this thing lets me do is connect to conference rooms, it seems.
23:09.25iPod-nanoI call my Gizmo number, nothing happens.
23:09.43iPod-nanoI try to call another Gizmo user, and it doesn't work properly.
23:09.58iPod-nanoOh, I can make outgoing to landlines, too.
23:10.41JTd-links are pretty sucky i guess
23:10.44iPod-nanoAnd I know nothing about Asterisk, nor have I used it before.
23:10.54JT~thebook
23:10.55jbotit has been said that thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
23:10.55iPod-nanoSo I had to ask before I went ahead and tried it.
23:10.58JTthe book has a lot of info
23:11.31iPod-nanoIt would be nice if I could make this thing ring when I get calls.
23:12.11iPod-nanoSo if Asterisk can let me do that, then I'll dedicate one of my computers to it.
23:14.01iPod-nanoI've been looking at these pre-built Linux distributions made for VoIP.
23:19.29*** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net)
23:20.35JTiPod-nano: no point to most of them
23:20.46knarflycan anyone clue me in on the GUI for 1.4.4? I run FreeBSD not Linux...
23:20.59knarflyWhere do I find it, how to install it?
23:21.21JTasterisk-gui
23:21.30JTyou can use it, but there won't be much support for it
23:21.36rob0Oh, I suppose the dream of these distros is that you can run a PBX without knowing anything about Linux/Unix administration.
23:22.13*** part/#asterisk petecat3 (n=petethec@c-69-253-246-202.hsd1.pa.comcast.net)
23:22.31JTrob0: but that's silly, unskilled people should not be building PBXes
23:22.46rob0Yeah, I agree.
23:23.03iPod-nanoI'm not unskilled, but I am new to Asterisk.
23:23.10iPod-nanoI know Linux, though.
23:23.23JTi didn't say you were :)
23:23.31rob0Then there's definitely no point in getting such a distro.
23:23.43iPod-nanoAll I want is to be able to use this junk.
23:23.51rob0Just build your * and start playing.
23:23.55JTrefering more to people who think a manager should be able to build a pbx, which is crazy ;)
23:24.06*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:25.08iPod-nanoJT, rob0, I do like the idea that with such a distro, I can just run it live and save my settings on a flash drive.  That would allow me to switch computers if need be.
23:25.31JTastlinux might do that, not sure
23:26.27iPod-nanoPlus these are minimal distributions, which are easier to play with under VMware.
23:26.55JTvmware isn't a very good place to run asterisk
23:27.21iPod-nanoWell, I'm lazy. :-P
23:27.30iPod-nanoAnd my real computer is across the room.
23:27.46JTssh was invented for a reason...
23:28.00rob0for lazy people ;)
23:28.11iPod-nanoNot if the computer isn't turned on!
23:28.22JTno remote power board/lom?
23:28.32iPod-nanoNot to mention that I'd have to reinstall Linux on it.  I was foolish to try and run FreeBSD.
23:28.33rob0Real computers are ALWAYS on! :)
23:31.27iPod-nanoI wish a third party could've wrotten better firmware for the stupid thing.
23:31.31iPod-nano8written
23:31.36iPod-nano*written
23:32.22BSD_Tech?
23:32.35iPod-nanoMy D-Link.
23:37.35iPod-nanoHow do you get a kernel panic from a LiveCD?
23:37.58rob0cool!
23:38.20rob0Perhaps they needed a bit more debugging before release.
23:38.59iPod-nanoOh... I don't want to have to get up!
23:39.13iPod-nanoAnd walk the five foot to my other computer.
23:39.17iPod-nano*feet
23:39.41rudholmiPod-nano: you get a kernel panic when something calls panic()
23:39.43rudholmduh
23:39.52rudholm:-p
23:46.25obnauticusROFL
23:46.32obnauticusSprint will change my CID to Emergency <911>
23:46.33obnauticuswow
23:46.44rudholmwhy would they do that?
23:47.03obnauticusDunno
23:47.05obnauticusit's legal though
23:47.05obnauticus:)
23:47.06obnauticusLoL.
23:47.21rudholmthis is via a PRI or something?
23:47.23obnauticusIt ensures pickup of the phone on the receiving end
23:47.28obnauticusNo, coustomer service
23:47.32obnauticusI'm speaking to a Guy.
23:47.36obnauticusthat's doing it
23:47.36obnauticusrofl.
23:47.48rudholmwhat kind of phone service?
23:48.05obnauticusNothing special
23:48.07obnauticusfamily plan
23:48.09obnauticusI'm 15
23:48.10obnauticuslol.
23:48.12rudholmwireless?
23:48.14rudholmwired?
23:48.17obnauticusYa
23:48.19obnauticuswireless
23:48.23obnauticusIt's a cell
23:48.23Qwellobnauticus: no, I'm pretty sure that's not legal
23:48.29obnauticusQwell yes it is
23:48.35obnauticusit is illegal to impersonate
23:48.36russellbpretty sure it isn't
23:48.39obnauticusCaller ID spoofing IS illega
23:48.40obnauticusl
23:48.52Qwellmisrepresenting yourself as some sort of official?
23:48.58obnauticusno
23:49.04obnauticusi wouldn't do that
23:49.06obnauticusjsut wanna change my CID
23:49.19rudholmI don't think a court would see the distinction.
23:49.32russellbto something that misrepresents yourself as emergency personel?
23:49.35russellbyeah, that won't go well.
23:49.42obnauticushttp://www.engadget.com/2007/06/29/congress-looking-to-make-caller-id-spoofing-illegal/
23:49.48obnauticusIt is TO BE made illegal
23:49.52Qwellcallerid spoofing is something COMPLETELY different
23:49.54obnauticusI just wanna try it before then
23:49.58obnauticus...
23:50.01obnauticusThen what ma I doing?
23:50.10rudholmother laws apply
23:50.10obnauticusI dunno
23:50.11QwellBy setting your CID as 911?
23:50.14*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
23:50.15obnauticusjust wanna mess with friends
23:50.15obnauticuslol.
23:50.31Qwellyeah, I've got $20 that says it's not legal to do that.
23:50.40obnauticusnot yet
23:51.23russellbQwell: i'll raise your $20 to $50
23:51.26rudholmobnauticus: I believe that you're intention is just to mess with friends, and personally, yeah, it's funny, but make no mistake, just because spoofing CID isn't illegal per se does not mean that you're in the clear.  other laws apply.
23:52.10rudholmthere are laws against impersonating police and emergency service personnel.  those laws would most definitely apply.
23:52.44rudholmdoesn't matter if you use a fake badge, fake uniform, verbal statements, CID, or anything else to do it.  it's illegal.
23:58.28*** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
23:58.28*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.20, 1.4.6 (June 29, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support.

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