irclog2html for #asterisk on 20051016

04:54.09*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
04:54.09*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 was a great success! Thanks to everyone who made it happen, as well as to everyone who made it out!
04:54.16X-Rob~pb
04:54.18jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca/
04:54.21X-Robgood jbot!
04:54.52ManxPower~manxpower
04:54.54jbotfrom memory, manxpower is someone you should hire for a job in BelgiumNetherlands, or someone you should send lots of money to because he helps so many people.
04:54.59Insanity5umm what does restart now do on th asterisk serveR?
04:55.04Insanity5restart the server or box?
04:55.13kshumard_homerestarts asterisk
04:55.14kshumard_home(not the box)
04:55.15Qwell[]Insanity5: asterisk
04:55.20Insanity5disconnected me from the box
04:55.21kshumard_homedoesn't wait for calls to end or anything
04:55.22Insanity5wtf
04:55.27Insanity5can't getb ack in
04:55.32Qwell[]Insanity5: asterisk -r
04:55.40Insanity5no, it went:
04:55.44ManxPowerInsanity5, restart now stops and then starts Asterisk
04:55.47Insanity5Disconnected from Asterisk server
04:55.47Insanity5PC:/etc/asterisk #
04:55.47websaeqwell: have you used AstBILL?
04:55.49Insanity5and now it's frozen
04:55.53Qwell[]websae: no
04:55.57*** join/#asterisk Lathos42 (n=Lathos42@adsl-68-255-76-52.dsl.lgtpmi.ameritech.net)
04:56.02Insanity5can ping it
04:56.13Igbothom_IIIhttp://pastebin.ca/25653
04:56.29Igbothom_IIIOct 16 14:52:45 NOTICE[3456]: chan_sip.c:7355 handle_request: Failed to authenticate user Some Git <sip:88883805@sip03.astrasip.com.au>;tag=3798092828
04:56.52*** join/#asterisk copantl (n=copantl@205.240.200.126)
04:56.54X-RobIgbothom_III - astrasip are, bluntly, cunts.
04:57.00X-Robuse faktortel.
04:57.02Igbothom_IIIreally?
04:57.07X-Robreally.
04:57.07copantlhelp?
04:57.07Igbothom_IIIcool - I'll look elsewhere  :)
04:57.18Insanity5can anyone get a login prompt by sshing to cainkar.com?
04:57.32Insanity5i think I murderedi t!
04:57.38X-Robfaktortel is about the same price, and they have really good support.
04:57.40Igbothom_IIIonly got an account there as they had an easy to get into website and I could sign up with $20 of pre-paid
04:57.46X-RobInsanity5 - looks like the box has broken.
04:57.48Igbothom_IIIhave NO issues moving elsewhere
04:57.53*** join/#asterisk abcbooze (i=abcbooze@70.153.193.166)
04:58.00copantli have a te110p card in my asterisk and i unable to place call
04:58.11abcboozeis there somethign wrong with cvs.digium.org ?
04:58.18X-RobIgbothom_III - faktortel is good. Pennytel is a bit cheaper, but their website is useless unless you're using IE on Windows.
04:58.18Igbothom_IIIin the mean time, if we can fix this now, it will also be "fixed" when I move (to Faktortel or whomever)
04:58.25Insanity5well crap
04:58.26copantlbut i able to  receive calls
04:58.28Insanity5I can still make calls out!
04:58.28Igbothom_IIII use IE on Windows  :)
04:58.29Qwell[]abcbooze: That domain isn't right
04:58.32Qwell[].com
04:58.37Igbothom_IIIand yes, not much diff between the two
04:58.38abcboozeerm
04:58.41abcboozemy typo
04:58.42abcboozebut yeah
04:58.44Insanity5no I can't :(
04:58.47abcboozeCannot connect to cvs.digium.com: Connection refused
04:58.47abcboozeWill retry at 01:01:33
04:58.51X-RobIgbothom_III - I can quite happily help you with faktortel, or, even better, THEY can support you with asterisk problems
04:59.02X-Robfaktortel quite happily support asterisk as a client.
04:59.04Igbothom_III'cept faktor has untimed .au calls
04:59.06Insanity5well yes I can, bizarre.
04:59.12kshumard_homeI just pulled cvs from digium.com 10 seconds ago....
04:59.34ManxPowerDigium CVs is (or was) on roundrobin DNS
04:59.38copantlanyone from digium?
04:59.48Insanity5Igbothom_III - well get this, I did restart now, it dropped me to a prompt, I ssh box to the box, but I can make a call through it's asteirsk server.
04:59.55kshumard_homecopantl, I'm from Digium. : )
04:59.59Insanity5can't ssh that is :)
05:00.03copantlhello
05:00.10X-RobInsanity5 - there's something wrong with your sshd
05:00.13abcboozecan someone dns cvs.digium.com for me
05:00.16abcboozemaybe my dns is slack
05:00.23copantli bought a card te110p
05:00.28Insanity5X-Rob - well it took out apache too -- postfix answers but just hangs.
05:00.30X-Robin fact, there's something wrong with your machine. Even smtpd is broken.
05:00.38X-Robit's a kernel issue
05:00.40Igbothom_IIIInsanity5; lost me in there somewhere
05:00.42X-Robyou will need to reboot it.
05:00.49copantland i tried to connected to a lucent switch with PRI
05:01.08X-RobIgbothom_III - I think he was confused as to whom he was talking 8)
05:01.23Igbothom_IIIme too  :)
05:01.32Insanity5Igbothom_III - Yes, it was odd,  I did quit asterisk, it dropped me to a unix prompt, dropped my connection, asterisk keeps running, but all other daemons died.
05:01.45Igbothom_IIIso, assuming I'll move from AstraCunts, any clues on my issue?
05:01.51X-RobInsanity5 - your machine needs someone to push reset
05:01.58ManxPowerInsanity5, you have some OHER problem
05:02.12copantland i unable to receive calls but able to place calls... what can be wrong??
05:02.18X-RobIgbothom_III - your need to have type=friend in your sip config.
05:02.23kshumard_homecopantl, possibly dialplan configuration
05:02.27Igbothom_IIIand Faktor tel are QUEENSLANDERS
05:02.33kshumard_homecopantl, what does the console say when you dial into it?
05:02.34copantlsorry i unable to place calls
05:02.35Igbothom_IIIit is in there
05:02.41websaeAstBILL anyone---anyone familiar with it---i got it setup, but to authorize sip connections it's not been so kind, any ideas?
05:02.45Igbothom_IIIhttp://pastebin.ca/25653
05:02.49copantlbut i can receive calls
05:03.05abcboozecan someone dns cvs.digium.com for me?
05:03.10copantland im using asterisk@home
05:04.01Igbothom_IIIX-Rob; I'll call Faktortel tomorrow - I emailed them yesterday anyway  :)
05:04.12copantlwhen a call is placed i receive a congestion answer
05:04.36copantlbut  i can receive calls any time
05:04.49X-RobIgbothom_III  - you need [88880426] type=friend context=incoming in your sip.conf
05:05.02Igbothom_IIIok - doing it now
05:05.06*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
05:05.11kshumard_homecopantl, what does the console say when you try to place a call?
05:05.20copantlwait
05:05.22Igbothom_IIIcontext=incoming   ?
05:06.14copantl<PROTECTED>
05:06.15copantl<PROTECTED>
05:06.15copantl<PROTECTED>
05:06.15copantl<PROTECTED>
05:06.15copantl<PROTECTED>
05:06.23X-Robor whatever context you want your incoming calls to go to
05:06.30abcboozehey
05:06.34abcboozecan someone type
05:06.49abcbooze/dns cvs.digium.com
05:06.52abcboozeand tell me the output
05:07.19X-Robabcbooze - learn to use linux dude.
05:07.23moralecvs.digium.com      A216.27.40.102
05:07.24moralecvs.digium.com      A66.225.202.81
05:07.25X-Robthe command is 'host cvs.digium.com'
05:07.32abcboozeyeah
05:07.33abcboozei know
05:07.37abcboozebut for wahtever reason
05:07.42*** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer)
05:07.42abcboozeim getting a bad dns
05:07.43Igbothom_IIIX-Rob; same error
05:07.46abcboozefrom my dns servers
05:07.49abcboozeso thanks
05:07.52abcboozemorale
05:08.27X-RobIgbothom_III - hurm.
05:08.32Igbothom_IIIyup  :)
05:08.58fiber0ptihow can you dial to extensions at a time?
05:09.15JerJermooo
05:09.23jdv79baaaaaa
05:09.35jdv79JerJer, do you have a cow?
05:09.49copantlkshumard  did you see my post?
05:10.03JerJeri eat cow
05:10.05kshumard_homecopantl yeah, wasn't too helpful... could I login to the box? PM me...
05:10.12jdv79so do i
05:10.12JerJerlots of it, actually
05:10.17jdv79you'
05:10.21ManxPowerJerJer, How was astricon/
05:10.25jdv79ll be hte first to contract mad cow!
05:10.45ManxPowerCows don't get mad.  Cows get even.
05:11.31Insanity5hehe
05:11.52JerJerManxPower: pretty good - only had to yell at a couple people
05:12.04Insanity5y?:)
05:12.12JerJercuz i can
05:12.31Juggiei had fun
05:12.34Juggielots of fun drinking :P
05:12.37Insanity5your guys?  or random people?
05:12.46Juggiejerjer what was the name of that tool
05:12.54JerJera hammer
05:12.55Juggiefor the secondary ip's on network interface
05:12.58Insanity5My box froze :(
05:13.03JerJerwackamole
05:13.08Insanity5and its 2k miles away :(
05:13.09Juggiekk, i have to look @ that
05:13.20Insanity5and asterisk did it.
05:13.34JerJerhttp://www.backhand.org/wackamole/
05:13.38fiber0ptiI'm setting up my polycoms to auto answer to immitate a paging system.. is there way to dial a bunch of extensions at once without disconnecting all but the one that answers first.
05:13.58Insanity5hehe
05:14.17JerJerfiber0pti: app_page
05:14.18Juggiefiber0pti, throw all the calls into a listen only meetme conference
05:14.21Insanity5speakers in the hallway?
05:14.22Juggieor that :P
05:14.43Insanity5and a soundcard?
05:14.57JerJer500 watt PA system
05:15.04fiber0ptiJuggie: How would you connect a bunch of extensions in a meetme conference while the phone calling can speak?
05:15.16JerJerfiber0pti:  app_page
05:15.22JerJerits in cvs -head as of like 3 days ago
05:15.23JerJeruse it
05:15.25JerJerit works
05:15.27JerJerdamn good
05:15.36Juggiefiber0pti, looks like app_page is just a auto meetme generator but none the less try it out
05:15.41*** part/#asterisk halogen8 (n=halogen8@ip68-8-18-103.sd.sd.cox.net)
05:15.49jdv79how is h323 support these days?
05:15.52fiber0ptihow do I determine if app_page is in the version I have? :S
05:15.58JerJerjdv79: crap
05:16.09Juggiefiber0pti, i dunno, its in your apps dir
05:16.13Juggiecheck your source
05:16.18jdv79what if i need it
05:16.28Juggiefind a way not to use h323
05:16.30Juggieits crap
05:16.32JerJerfind a way not to need it
05:16.35X-RobJuggie - *laugh*
05:16.35jdv79supposed a lot of our voip carriers still use it and demand it
05:16.40JerJerblah
05:16.42ManxPowerjdv79, there are at least FOUR H323 drivers for Asterisk
05:16.52jdv79uh, and there's one SIP, right?
05:16.58ManxPowerjdv79, yup
05:16.59JerJertwo of which are barely H.323v2 compliant
05:17.04abcboozeJuggie?
05:17.06Juggiei have no expirence with h323 so i cant say anything
05:17.07jdv79what's the deal then?
05:17.11abcboozefrom efnet?
05:17.15JerJerH.323 just plain sucks
05:17.20Juggiebut you can try the one in asterisk-addons
05:17.23Juggieabcbooze, the one and only
05:17.25jdv79oh
05:17.26JerJerwhy do you think other signaling methods where created?
05:17.35abcboozeheheh whats up Juggie
05:17.42jdv79just because of polititcs
05:17.47Juggiejust got back from astricon in anaheim
05:17.50Juggielike 2hrs ago
05:17.52JerJerif there is any telco out there today that only supports H.323, they do not deserve your business
05:17.53abcboozeno way!
05:17.55coppiceJerJer: in order to learn nothing from the drawbacks of H.323? :-\
05:17.56*** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net)
05:17.56abcboozeawesome!
05:18.07abcboozei <3 asterisk
05:18.07moralehas much been changed or fixed from 1.0.9 -> 1.2.0-beta1? im thinking it is almost worth a try upgrding
05:18.19JerJermorale:  everything
05:18.21Juggieh323 is an example of what happens when engineers with no programming expirence design a protol
05:18.24Juggie*protocol
05:18.29abcboozei have cisco 12SP+ ip phones running off asterisk right now
05:18.33abcboozeerm
05:18.36JerJermorale:  read upgrade.txt in the source tree
05:18.37abcboozei have 3*
05:18.38Juggieits nice and tight, but a bitch to code for
05:18.48fiber0ptiDoesn't look like I have app_page.. I'll see about getting the new version.. but in the mean time I've created a paging conference room.. how do I dial few extensions and automatically connect them?
05:19.00JerJerabcbooze:  sweet... which skinny driver?     I have a dozen or so 12SP's out there limping along on chan_skinny
05:19.05Juggiefiber0pti, update to cvs-head
05:19.12X-Robfiber0pti - are you using 1.2 or CVS HEAD?
05:19.25X-Robif you're using 1.2beta1 you can use http://aussievoip.com/allpage.agi
05:19.41Juggieits not really that complicated
05:19.42JerJerjust cvs up
05:19.45Lathos42JerJer: This isnt protocol related.. but who would you suggest getting a T1 from in our neck of the woods?
05:19.48Juggieeven without app_page
05:19.50fiber0ptix-Rob:1.0.7
05:19.55abcboozeJerJer, chan_sccp2
05:19.56JerJerLathos42:  i am biased
05:19.59JerJerabcbooze:  fun
05:20.02X-Robfiber0pti - see jerjer's comments - update.
05:20.06Juggiegenerate your calls, auto join them into a meetme conf, listen only
05:20.11abcboozecan't get hold to work right though =/
05:20.26Lathos42JerJer: Why, because you could sell me one? :)
05:20.31Juggiewhen all the calls join, play a beep, and then let the person who generated the page talk
05:20.37JerJerLathos42:  yes
05:20.55fiber0ptiI can't right now.. I need to talk with someone else before updating it and I can't do that until the morning.. there's no easy way to dial a couple of extensions and then put them in a meetme conference room?
05:21.05JerJernow
05:21.06JerJerno
05:21.14JerJerwhy you think an app was created?
05:21.24JerJermkdir new_directory
05:21.25Juggiefiber0pti, asterisk manager interface
05:21.30JerJercd new_diretory
05:21.38moralecan you specify a password on the cvs login commandline?
05:21.43JuggieAction: Originate
05:21.48Juggiecreate your calls, join them into meetme
05:21.51JerJercvs -d:pserver:anoncvs@cvs.digium:/usr/cvsroot co asterisk
05:21.52Lathos42JerJer: We were reminded of how vulnerable our internet connection was the other day when SBC lost a router
05:21.54*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
05:22.04JerJerfiber0pti:  then compile that tree
05:22.09JerJerbut do not make install
05:22.26JerJerthen very simply copy over app_page.so to /usr/lib/asterisk/modules
05:22.29alephcomHas anybody here done anything with the Asterisk POE perl libraries?  I'm looking at building something to do realtime billing using that.  Any comments?
05:22.36JerJerthen asterisk -r    and load app_page.so
05:22.50JerJerthat's precisely how i tried it on a production system for a customer
05:23.01JerJerIP from sbc - are you mad?
05:23.26Lathos42JerJer: I may have told you before, They signed the contract with SBC before I started working there.
05:23.37JerJerwonderful
05:23.39JerJerget out of it   :)
05:23.55Lathos42It'll be up in March or April
05:24.02JerJerthere ya go
05:24.18moralemmm.. i wonder when this coffee was brewed..
05:24.45Lathos42For the time being, I'd like to get a backup connection going
05:25.17JerJerLathos42:  you are down in like lansing or detoliet right?
05:25.34Lathos42Yeah, Eaton Rapids, Actually
05:25.50JerJerchocolate cake
05:26.17JerJerbetter yet - special brownies - its that easy   :)
05:27.05*** join/#asterisk docelm0 (n=docelm0@pool-71-243-242-151.tampfl.fios.verizon.net)
05:27.52Lathos42What would be a rough number that it would cost me for a T1?
05:28.47JerJercertainly less than $500 a month
05:28.50JerJerprolly a lot less
05:29.09JerJerlevel3 and uunet - all gig-e network
05:31.23Lathos42SBC is trying to push us towards their 3.0 service.. instead of saving us money by renegotiating our rates down from the $799 we pay now
05:31.41JerJeroh christ
05:31.47JerJerits now $600  :)
05:32.18*** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net)
05:32.30JerJerstatic !
05:32.38citatswhats happening?
05:32.45Lathos42Is that all that SBC has come down to in the past 3 years?
05:32.47JerJerjust got back from Cali
05:33.12citatsi'm sure a good time was had by all
05:33.21JerJersome had too good of a time
05:33.33citatsis that even possible? :)
05:33.43JerJerSomeone needs to ask Damin about "Sir...Sir, please don't urinate in the kitchen"   :)
05:33.56citatsheh
05:34.01Qwell[]JerJer: according to Damin, Del Taco is THE place to be at 2am
05:34.06Qwell[]the Del Taco parking lot even
05:34.10JerJeryeah
05:34.24JerJerhe was hangin out with some of the locals, i guess
05:34.36Qwell[]that guy is too much
05:34.45Lathos42JerJer: I'll have to get back with ya..  My ambien decided it would be a good time I went to sleep now.
05:34.46JerJeryeah Damin is great
05:35.08Lathos42My chat window keeps bouncing around my field of view.. its fun!
05:36.07Lathos42later gators
05:36.21jdv79anyone using a lucent TN with *?
05:36.32jdv79TNT i meant
05:36.34JerJerQwell[]: assignment size of 'Qwell' is not known
05:36.41websaeanyone here using ASTBILL?
05:36.47websaeanyone at all?
05:36.50JerJerjdv79:  yeah there are a few crazies out there running them
05:36.54websaelooking for some AstBILL help---it seems like a great package
05:36.58JerJerum no
05:37.01websaebut can't get it to work with sip phones
05:37.02Qwell[]JerJer: You aren't the first to make that joke. :p
05:37.05jdv79why are they crazy
05:37.11jdv79they're cheaper than ciscos
05:37.23jdv79we just need some gw boxes
05:37.24JerJerask me that IF you get one running
05:37.31JerJerit is not easy
05:37.41jdv79i've had them doing some app once
05:37.46jdv79but that was a long time ago
05:38.08jdv79JerJer, what would you use for a flexible GW solution?
05:38.11moralehaha, i made it work.
05:38.15JerJerjdv79:  Asterisk
05:38.15moralethis is so nifty.
05:38.22jdv79DS3 style
05:38.28JerJerAsterisk
05:38.36jdv79that's not possible
05:38.46JerJerbullfuckingshit
05:38.48jdv79it only does quad t1
05:38.53JerJerso?
05:38.58JerJerever heard of multiple boxes?
05:39.02JerJerand a ds-3 mux
05:39.15moraleOct 15 23:33:23 WARNING[10777]: chan_sip.c:3442 process_sdp: Unknown SDP media type in offer: video 21654 RTP/AVP 26 31 34 - what does that mean? its trying to do some funky video stuff... should i just disable that module?
05:39.18JerJerproblem solved
05:39.25JerJerTE411P baby!
05:39.26Qwell[]just get the ds3 card when it comes out, heh
05:39.51jdv79a ds3 mux, 7 quad t1 cards and...3 boxes?
05:39.58JerJerthere ya go thinkin again
05:40.08WilliamKJer, BAH
05:40.11WilliamK=)
05:40.11jdv79that's probably more than a cisco
05:40.22JerJerdefine more
05:40.27jdv79i can get a AS5something for 25K
05:40.39JerJerhow many CCIEs does it take to configure an as5400?
05:40.48Qwell[]oh, I love these jokes
05:40.51Qwell[]how many?
05:40.54jdv79i have some guys that do it in their sleep
05:41.02JerJerthen compare its flexibility to Asterisk
05:41.04jdv79that's not an issue for me
05:41.06WilliamKJer, none
05:41.16WilliamKlast I knew most didn't know howto leave the drugs alone
05:41.20jdv79how many quad cards will a box support?
05:41.23jdv792,3,4?
05:41.27JerJerone
05:41.36Qwell[]1 is realistic
05:41.36jdv79that sucks, no?
05:41.39JerJerno
05:41.49dasuberdavidyou can run two quad cards in a box
05:41.52jdv79a small server is $1500
05:41.56JerJerfind me another solution that I can solve my own problems with
05:42.00Qwell[]sure, can...
05:42.13WilliamK=)
05:42.17*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
05:42.17WilliamKor an APX8000
05:42.19WilliamK:)
05:42.20jdv79so that's 4K/box...
05:42.21JerJeri take it nobody has spent any time dealing with TAC
05:42.24jdv797 boxes
05:42.33jdv7928K - same as a cisco
05:42.35JerJerarguing with them about 'bugs'
05:42.45Qwell[]jdv79: huge difference
05:42.46WilliamKJer, I've spent plenty of time argueing with TAC about bugs
05:42.59JerJerwith Asterisk I don't argue - if i find something i don't like, i can very easily change that behavior
05:42.59jdv79well, there is no difference to me
05:43.06WilliamKquestion is do you give up before they do
05:43.17jdv79i simply need GW boxes - nothing more
05:43.27JerJerits not just the boxes
05:43.31JerJerits managing them
05:43.42jdv79AS5400s just work
05:43.46jdv79like a rock
05:43.58jdv79set and forget style
05:44.01JerJerif you say so
05:44.04citatshaha
05:44.06Qwell[]put that rock in water.  I bet it sinks
05:44.09jdv79my cisco guys do
05:44.11jdv79and i've seen it
05:44.24JerJerand they are pushing cisco products at you
05:44.34citatsa few reboots a week kinda rock
05:44.53jdv79cause they are cheaper than anything else and they're solid
05:45.01jdv79i want to use something else but i haven't found anything yet
05:45.02ManxPowerWe've never had problems with ciscos, but we also don't push them hard either.
05:45.21JerJerjdv79: operate it for a while and you'll see
05:45.37jdv79i'd like to avoid problems if i can:)
05:45.47jdv79why would you not use ciscos or lucents?
05:45.53citatsciscos data stuff is solid.  voice gear is a different software, its loaded with bugs
05:46.04JerJerflexibility
05:46.06JerJeradaptability
05:46.10Qwell[]jdv79: <JerJer> with Asterisk I don't argue - if i find something i don't like, i can very easily change that behavior
05:46.15jdv79yeah but a gw box is just a gw
05:46.23JerJerasterisk can `just be a gw`
05:46.39MikeJ[Laptop]I can be drunk.
05:46.54jdv79yeah but i have to manage 28 servers and a mux instead of 1 cisco box
05:46.59JerJerso yeah i get blamed for talkin shit, huh   :)
05:47.19jdv79power req++
05:47.27jdv79admin req++
05:47.29jdv79so why?
05:47.38jontowgoddamn aheeva and their prebuilt binaries
05:48.01JerJerlike i said, operate a proprietary solution for a while and you'll see
05:48.28jdv79ok, i kind of believe you but i'm really looking for explanable reasons
05:48.30Qwell[]tip: If you're going to make a radio with a "game" on the front panel...
05:48.42Qwell[]When you choose random numbers for the "game"...MAKE SURE THEY'RE RANDOM.
05:48.47JerJerasterisk is not perfect, but unlike a cisco or tnt box, at least I have a chance on working around issues, until a real solution is at hand
05:48.53jdv79how can i tell my telco guy that instead of one AS5400 we need 28 servers?
05:49.14JerJerhow do you get 28 servers?
05:49.16MikeJ[Laptop]why would you need 28 serers?
05:49.18Qwell[]001, 247, 483, 629, 865, ...001
05:49.25jdv79oh wait
05:49.33JerJertry 6-7
05:49.34jdv797 servers
05:49.36jdv79yeah
05:49.40JerJertrivial
05:49.44jdv7928K was my esitmated cose
05:49.52jdv79cost... i got mixed up:)
05:49.53JerJeryou net boot them all off the same fuckin image
05:49.59MikeJ[Laptop]yeah, but you can probably do it for way less cost than the 5400 still
05:50.03JerJerproblem solved
05:50.15jdv79CPU fans are still moving:)
05:50.23JerJersmnp
05:50.25JerJersnmp
05:50.31MikeJ[Laptop]you think there aren't fans on a 5400 :P
05:50.49jdv79there are but unlike a modern CPU they can deal for a bit
05:50.52*** join/#asterisk alphaque (n=alphaque@218.111.204.40)
05:50.54JerJerum
05:51.03MikeJ[Laptop]but with this solution, you can bounce boxes for maint one at a time
05:51.04jdv79unlike a PC where you got seconds until it frys
05:51.07JerJeras54 is not that much different from a 'modern cpu'
05:51.16JerJerits all microprocessors
05:51.24*** join/#asterisk without (n=dean_dav@CPE-60-226-180-177.qld.bigpond.net.au)
05:51.33Qwell[]MikeJ[Laptop]: yeah, I could just imagine losing one piece of $30k equipment...
05:51.39Qwell[]vs 1 of 7 servers.
05:51.40JerJeryeah, WHEN your as5400 does down, THE WHOLE DS-3 IS DOWN
05:51.43JerJernot if
05:51.48JerJerthey reboot a whole lot
05:51.57JerJertnt's also flake out entire spans too
05:52.16JerJerso does asterisk, but at least the scope of the problem is smaller on an Asterisk solution
05:52.16jdv79ah, now i have at least heresay
05:52.44jdv79asterisk flakes out?
05:52.47jdv79even stable?
05:52.50JerJerlike i keep saying, operate it for a while and you will have first hand experience of the pain
05:53.01JerJerstable is not 'without bugs and completely flaw free'
05:53.25coppiceever experienced the fault tolerance of a stratus? :-)
05:53.40moraleits pretty bad when your voip provider is 160ms lagged to you..
05:53.40JerJerit is a poor choice of some random point in time Mark (et all) decided to let Russel only commit 'bug fixes' to
05:53.47moralei need to find something closer.
05:53.54JerJermorale: yes you do
05:53.59jdv79hm?
05:54.06JerJeri'm 20ms and i hate ti
05:54.08JerJerit
05:54.20withoutI have a TE Pri card and i unload the zaptel drivers and the box stops resonding is this common ??
05:54.20JerJerpoor word
05:54.27morale15 hops - 15  pp01a.inphonex.com (208.239.76.157)  104.245 ms  105.079 ms  104.593 ms
05:54.29JerJerfor the branch name
05:54.30moralethats the last hop
05:54.38jdv79is russel a bottleneck or something?
05:54.46JerJereh?
05:54.58JerJerthe stable branch of code is crazy outdated
05:55.05jdv79uh huh
05:55.10jdv79why is that?
05:55.14JerJercvs -head  / 1.2rc1 is far superior
05:55.15ManxPower*shrug*  I'm 11ms from my gateway to the internet and it gets worse after that.
05:55.34jdv79i was gonna use the 1.2 release to go live with
05:55.35JerJerjdv79:  you obviously don't know much about software development cycles
05:55.36ManxPowerI don't even call is "stable" anymore.  It's 1.0.x
05:55.42ManxPowerSave the confustion.  it's 1.0.d
05:55.45jdv79JerJer, i certainly do
05:55.48X-Robit's not called 'stable'
05:55.51X-Robit's called '1.0'
05:55.57JerJergood
05:56.01JerJervery good
05:56.19JerJerlets keep reminding everyone of that
05:56.25*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
05:56.37ManxPowerThere's 1.0.x and there's the "developement version" commonly called 'CVS-HEAD"
05:56.39coppiceits called stable, because it is full of manure :-)
05:56.44jdv79the DS3 card is never coming, right?
05:57.13coppiceDS3 cards are a myth created to make people feel bigger
05:57.41moraleanyone know if comwave uses sip or mgcp?
05:57.41jdv79so you couldn't do 2 quad t1 card in one box?
05:58.16jdv79T1s to the carrier and IAX to the * iVR servers...
05:59.01JerJerif you are not doing transcoding, maybe
05:59.06jdv79t1s, ugh, i hate them
05:59.15jdv79no transcoding
05:59.27coppicewhat's wrong with T1's?
05:59.45jdv79you ever wire t1 frames up?
05:59.47jdv79oh god
06:00.01JerJerget the right tools
06:00.10jdv79its all those little wires and the colors...
06:00.15jdv79i used to do it all the time
06:00.19websaeanyone here using AstBILL at all--???
06:00.22coppiceT1s are a Cinch to wire up :-)
06:00.41JerJerwebsae: no one in their right mind would
06:00.54websaewhy's that?
06:01.02JerJercrap
06:01.11alphaqueJerJer: cvshead/1.2rc1 is superior in what sense, stability or features ?
06:01.12jdv79why not just cdr to a DB somewhere and build a little app on that
06:01.34ManxPowerYes, 25 pair cables are a bitch to punch down to a 66 block, but after a few days of practice you can get pretty good at it.
06:01.35jdv79be cool if there was a comprehensive test suite:)
06:01.44tzafrir_laptopIt's stable because its bugs remain constant
06:01.54jdv79no 66 blocks - those wire wrap things
06:01.59JerJeralphaque: yes
06:02.04jdv79i don't remember what's they're called
06:02.17jdv79they have the T1 plugs on the front and all...
06:02.17ManxPowerjdv79, I need to buy a wire wrap tool for one of out Tellabs channel banks
06:02.22jdv79same thing on the back of a T1 DACS
06:02.26coppiceyou wire wrap T1s? weird
06:02.27tzafrir_laptopanybody built res_sqlite with the system copy of sqlite rather than downloading sqlite?
06:02.37ManxPoweroff to bed
06:02.44jdv79i used to have the little gun wirewrap thing - it was really cool
06:03.13jdv79coppice, yeah dude - telco style hardcore
06:03.17jdv79i never want to do it again
06:03.20coppiceare you calling us from the 1970s, or something?
06:03.31jdv79i did it 5 years ago:)
06:03.46jdv79on the back of 12 tellabs 5500s DACSes
06:03.50jdv79ugh
06:03.52coppiceI haven't seen them wirewrapped for 20 years
06:04.02jdv79ok
06:04.41MikeJ[Laptop]Asterisk CVS HEAD built by mjerris@MIL-MJERRIS on a i686 running CYGWIN_NT-5.1 on 2005-10-16 04:55:39 UTC
06:04.43MikeJ[Laptop]:)
06:04.45jdv79oh wait - the DACS had huge connectors and then the connection pannels to other t1 stuff was the wirewrap one
06:04.52jdv79now i remember
06:04.52MikeJ[Laptop]almost working now...
06:05.02kshumard_homeon a PRI, is RELEASE COMPLETE a valid response to a SETUP message? I wouldn't think so...
06:05.16jdv79thanks for the advice all on the gw perdicament
06:05.53coppicekshumard_home: its valid. what do you thing is wrong with it?
06:06.18jontowmikej, hahah :D
06:06.50*** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
06:06.56kshumard_homecoppice, I can't make outgoing calls, I send SETUP and immediately get back RELEASE COMPLETE indicating 'normal clearing'
06:07.03kshumard_homecoppice inbound calls work fine though
06:07.41coppiceso, you have a service problem, or you are dialing a wrong number. RELEASE COMPLETE is a perfectly valid response
06:08.43kshumard_homecoppice do you know of a resource where I can find state diagrams or a list of valid responses to the various q931 messages?
06:08.49kshumard_homeI haven't had any luck on google
06:09.30coppiceyou don't need that right now. you need to look at your dialed number, your NPI and TON and possibly your originating number
06:09.49kshumard_homeI'm interested in it for personal education, though. : )
06:10.04kshumard_homeSo I don't have to hope there's someone around here that knows q931
06:10.11coppicebut the Q.931 response codes are all in the libpri header files. I put them there years ago
06:11.24kshumard_homehaha, right in front of my face, as it were. : )
06:11.58*** join/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net)
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06:15.20coppiceI must have created that stuff in 2000. time flies
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06:15.29kshumard_homewow
06:19.47*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
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06:20.32alephcomwebsae: I'll put a plug in for myself for billing www.aleph-com.net/astpp  I can't speak for astbill though.....
06:22.45joelsolankihello friends. i want to change the asterisk-adds on code so that i get START / ANSWER / END time in mysql cdr table. anybody have done this. by default asterisk-adds onlys gives START time :(
06:23.40Igbothom_IIIcan someone try to call my FWD number: 441658 please
06:23.54joelsolankii suppose  cdr_addon_mysql.c is the file which needs to be modified
06:24.04joelsolankican anybody gvie any suggestions please?
06:24.26Igbothom_IIInp - my rego was not successful
06:27.05moraleanyone use unlimitel?
06:27.40moralevoip providers are so hokey-pokey.
06:28.15alephcomjoelsanki: I don't have a reg but you should be able to calculate that if you have the answered time and and the billable seconds. as well as the start time.
06:28.27alephcomreg s/b patch.  sorry
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06:33.14moralewow.. 23 milliseconds away
06:34.13morale14ms seconds away on shaw cable..
06:37.23joelsolankialephcom: mysql cdr table has BILLSEC , DURATION, CALLDATE
06:38.07joelsolankialephcom: but i have to show the customers there cdrs like START TIME== answer of call and END TIME== end of call.
06:38.33jdv79quad T1 card - single CPU or dual?
06:39.18joelsolankialephcom: the cdr which get logs at /var/log/asterisk/cdr-csv/Master.csv has START ANSER END time.
06:39.21denondepends on the codec you want to use
06:39.29denonjdv79
06:39.49jdv79711 i think
06:39.57*** join/#asterisk dos000 (n=dos000@i216-58-62-73.cybersurf.com)
06:40.02jdv79straight though
06:40.03pauldyjoelsolanki, if it gives you start and duration youc an extrapolate end
06:40.03denonjdv79: you'll rarely regret going dual
06:40.05kshumard_homejdv79, you might be able to get away with 1 CPU if you also got the echo cancellation module on the card
06:40.15jdv79price point i may...:)
06:40.18jdv79but even still
06:40.19denonjdv79: but doing passthrough, your cpu reqs will be much lower
06:40.23pauldyanswer might be a bit of a proble,
06:40.41denonbut I'd still go dual -- if you're handling 4 PRIs, you certainly can justify a second cpu
06:40.44kshumard_homejdv79, will you be bridging voip calls? If not, there won't be any transcoding
06:40.51joelsolankipauldy: no but customers need the STARTTIME == call answer time and ENDTIME= call ended time.
06:41.04jdv79we have a IVR cluster and carriers:)
06:41.15jdv79and i have to mate the 2
06:41.24jdv79that's all
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06:48.27argos73joelsolanki: in my postgres cdr, you can "select calldate, duration,  calldate+(duration::reltime) from cdr;"
06:48.52argos73the ::reltime casts it as a unix timestamp, which can be used to calculate the ending time
06:49.16argos73mysql probably can do something similar, but I'd need to play a little to figure out how
06:50.19joelsolankiargos73: yes mysql has similar..it provides me calldate , duration , billsec on which i can do billing.
06:51.19argos73the trick is just to add calldate + duration - playing with syntax now
06:53.59joelsolankiargos73 i need to know is there any way to log START / ANSWER / END in to mysql cdrs. by default asterisk-addson logs START in mysql cdr.
06:54.17bmg505morning all
07:00.35argos73heh - easier than i thought...
07:00.49argos73just select calldate, duration, calldate+duration from cdr;
07:01.37abcboozeanyone know of some good wholesale voip termination providers?
07:02.27argos73without tweaking the source, it can't record endtime as it stands now..
07:04.28argos73denon: you ever catch last sunday's west wing?
07:04.49argos73started playing with encoding it, but other things got in the way
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07:16.20joelsolankiok got it Argos73 ..thanks :)
07:21.09argos73joelsolanki: cool - no prob
07:21.21joelsolanki:)
07:28.22pauldycalldate + duration = end, end - billsec = answer
07:34.29dasuberdavidfile[laptop] i for got my asterlink account info
07:34.33dasuberdavidforgot**
07:34.36*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
07:36.06desktopherohello everyone
07:40.18rvhitry to run /usr/sbin/asterisk, but it didn't go background, any suggestion?
07:40.39Igbothom_III!  <--- type that
07:42.17desktopherorvhi  -   did * not start at all,  r did you get an error message
07:42.37rvhiit starts well, running, processing calls
07:42.50Igbothom_IIIthen, as I said, type !"
07:42.51rvhiit just stays foreground
07:43.20rvhitype !? where?
07:43.38Igbothom_IIImaybe in the console?
07:43.41Igbothom_IIIWhere you are
07:43.52rvhiit didn't have a console
07:44.11Igbothom_IIIwhat, it ran and totally went blank?
07:44.11rvhii didn't use option -c
07:44.22rvhia bunch of output
07:44.28Igbothom_IIIand?
07:44.46rvhirunning ok, just didn' go background
07:44.50*** part/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net)
07:44.54rvhii have to leave the tty open
07:44.57Igbothom_IIIthen it is foreground, yes?
07:45.01rvhiyes
07:45.12desktopherowhat about asterisk - vvv &
07:45.40Igbothom_IIIthat's the bog standard Linux way to force a foreground app to the background
07:45.49rvhiwith & it goes background, but according to wiki, without option it should go background
07:45.50Igbothom_IIIshould work a treat
07:47.07desktopherofor * - i have used no other way.................
07:47.30rvhiyou mean ,with &?
07:47.31pooh_Hi guys, when I place an IAX2 call from 1 phone to the other, the displays says ' asterisk', how can I change this pls?
07:47.54Igbothom_IIIchange the "callerid=" in the context of the calling phone
07:47.58desktopherohow about 'SetCallerID' from the dialplan?
07:48.04pooh_I both did ...
07:48.27desktopheroyou did both what..?
07:48.47desktopherowhat version of *?
07:48.53Igbothom_IIIwhat about "usecallerid=yes for the receiving phone?
07:49.14pooh_* 1.0.9, hold on got something here
07:49.33pooh_ok, my bad. Dialplan (Macro too complex, typo!!!!)
07:49.48pooh_sorry for the noise
07:49.48Igbothom_III:)
07:50.08desktopherocomplex macro -- ohhh drive me crazy!
07:50.23pooh_I have hotdesking working, e.g. walk to a phone, logon and call
07:50.28pooh_and receive calls
07:50.50Igbothom_IIInice
07:50.57*** join/#asterisk Tommmo (n=tps@203.62.181.52)
07:51.01pooh_so a user is NOT bonded to 1 phone
07:51.06Igbothom_IIIexcellent
07:51.11pooh_kinda like computers or hotmail :-)
07:51.15Igbothom_III:)
07:51.19Tommmohi, im using asterisk Realtime, but when I try to register a user against mysql i get:
07:51.22Igbothom_IIIhow complex is the dialplan for that?
07:51.28TommmoOct 16 17:47:06 DEBUG[1235]: db.c:182 ast_db_get: Unable to find key '5140' in family 'SIP/Registry'
07:51.33Tommmoanyone know how to fix this?
07:51.41desktopherookay...my office is more about the phone being tied to a desk...than to a user
07:51.45pooh_Igbothom_III: pretty easdy actually
07:52.04Igbothom_III~pb if you are gonna paste anything
07:52.06jbotokay, Igbothom_III
07:52.15Igbothom_III~pb
07:52.16jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
07:52.23desktopheroTommo - huh?
07:52.33Igbothom_IIIg'day jbot
07:52.36Tommmodesktophero: ?
07:52.36pooh_no no, we have seperated the *thinking* that when you call a person you call a device
07:52.49pooh_so we have devices and usersnumbers
07:52.59pooh_the devicenumbers are not known to the users
07:53.00Igbothom_IIIyeah, hotdesking makes sense in a lot of cases
07:53.17pooh_e.g I am 504 wherev I am
07:53.25desktopheropooh - i get that concept - but have yet to see a viable concept about the "why"...
07:53.30Igbothom_IIIwhat distro u using?
07:53.36pooh_Centos 4.1
07:53.37Igbothom_III(or fresh build on Linux?)
07:53.37desktopheroat least in our environment...
07:53.47pooh_We have flexible workspaces and teleworkers
07:54.03Igbothom_IIIdesktophero; where you have 3 desks, 3 phones and 6 sales staff who are occas in the office  :)
07:54.21pooh_same issue when using device numbers normally
07:54.23desktopheropooh - we don't necessarily. Our users would not "get it"
07:54.34pooh_yeah
07:54.46desktopherosales staff have DIDs...they would not "get it"
07:54.54pooh_We even can hotdesk to with our cellphones
07:55.00Igbothom_III:)
07:55.07pooh_DID are easily integrated
07:55.20Igbothom_IIIassign the DID to the user, not the device
07:55.38pooh_nope, assign the DID to the device the user is at :-)
07:55.40desktopheroi can route a DID throughout the country - I DONT CARE ;)...the sales staff would complain about having to punch extra digits...;)
07:55.47Igbothom_IIIthat's what I meant
07:55.50Igbothom_IIIsame end result
07:55.53pooh_1 time login only, where they are
07:56.19pooh_and login is automatically logoff the 'old' device
07:56.30desktopheropooh - I know what you mean - not ALWAYS practical - know what I mean?
07:56.46pooh_Yes, so never log out :-)
07:57.09desktopherobut be able to go WHEREVER...and still get THEIR calls....ugh!!!!\
07:57.12af_which way could I debug a problem with dmtf?
07:57.20af_morning all
07:57.37pooh_desktophero: we are not in totally sync I guess ;-)
07:57.52desktopheropooh - what do you mean?
07:58.46pooh_I didn't understand your last comment
07:58.47desktopheropooh - calls route where they are...but they don't want to do ANYTHING for their calls to follow them....
07:59.14kshumard_homeaf_, voip or pstn?
07:59.45kshumard_homeaf_, you could start by turning on debug in logger.conf, then `logger rotate` in asterisk and make the call again
07:59.45desktopherothey LOGIN to phone A and then move to PHONE B...no login or logout they just want calls to follow them to phone b
07:59.48kshumard_homesee if asterisk sees it
08:00.31desktopherodoes that help pooh
08:00.34pooh_desktophero: some sort of mechanism should detect where the user is at, what do you suggest?
08:01.12desktopheropooh - they don't want login/logout...just have their extension "follow them"".......
08:01.25pooh_desktophero: I understand, but how?
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08:01.39pooh_the phones do not have eyes and can not see who is in front of them ;-)
08:01.47desktopherologin-logout...sure I can have calls follow that.
08:02.32pooh_there is only login, logout is automatically done by login
08:02.34desktopherobut if I can predict what phone (physical phone) that are that...I am going to go into the business of predicting the lotto numbers ;)
08:02.36pooh_1 less step ;-)
08:02.55desktopheroTHEY DON'T WANT TO LOGIN ;)_
08:03.08desktopheroi am dealing with
08:03.09kshumard_homeYou could rig something up with bluetooth to automate the login process based on presence...
08:03.14kshumard_homebut that might not be very feasible
08:03.15desktopheroSILLY users ;)
08:03.38desktopheroi am venting about the mindset...LOGIN only - i get it
08:03.39pooh_:-)
08:03.57kshumard_homeor script something on the system... are they always in front of a computer they have to login to?
08:04.04desktopheronow - we are on the same mindset...i think....darn users! ;) ;)
08:04.20pooh_:-)
08:04.53desktopheropooh -  have you done any work with IAX2 trunk between 2 * servers?
08:04.57pooh_Come to speak of that, we are working with SUN Microsystems to integrate our hotdesking into SunRay
08:05.26pooh_SUNRay is a kind of terminal server thing where the users have a smartcard to access the terminal
08:05.43pooh_slide in your card and you have your session AND your phonecalls ;-)
08:05.46desktopherohow about * with LDAP and use logins on a domain...HA!
08:05.55pooh_SUNRay does that :-)
08:06.22pooh_trunking is pretty easy, make sure that bith * servers have a trusted friend account
08:07.55desktopheropooh - how far are the 2 servers? and are u talking about LDAP...?
08:08.02JamesDotComanyone here have any experience with audiocodes mediant 2000?
08:08.22desktopherosorry James - no luck here...
08:08.24pooh_nope, talking about IAX.conf entries and VoIP is global ;-)
08:08.46desktopheropooh - how far are the two servers for the trunk?
08:09.13pooh_JamesDotCom: send audiocodes an e-mail, they are very helpfull. Better mail the israel office
08:09.36desktopheroall - what is audiocodes?
08:09.57pooh_very expensive but VERY HIGH QUALITY VoIP equipment
08:10.32desktopherodomestic us provider  at all....I like high qualiy ;)
08:10.53pooh_nope, Israelian corp
08:11.19desktopherodarn...
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08:13.51pooh_desktophero: what do you mean exactly by ' how far are the 2 servers for the trunk' ?
08:15.05JamesDotCompooh_: thanks might give that a shot, it sucks the lack of help around the net you can find for them
08:15.17pooh_JamesDotCom: np
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08:16.21desktopheropooh - i have two servers tht are connected over a WAN - one is over a business DSL line and about 1200 miles apart. the others are 2 feet in distance
08:16.38pooh_desktophero, it is all about the latency
08:16.45pooh_keep it below 100 ms
08:17.07moralei just cancelled my voip service, it was 190ms latent.
08:17.14pooh_yup
08:17.18JamesDotCompooh_: nm that, i managed to get it working... somehow
08:17.46JamesDotComthe only thing worse than not being able to figure out a problem is when its fixed and you're not quite sure how you did it
08:18.03pooh_tell me about it..... :-(
08:18.20desktopheroi check latency ALL the TIME  -  is that "status" when you "iax2 show peers" in the cli?? becuase a ping shows different ms times
08:18.47pooh_Yup, status is the thing to watch
08:19.03desktophero--agree with pooh about documents...ha!
08:20.08desktopheropooh -  normally, status ms is around 60ms. sometime it gets to ~300ms
08:21.14desktopheroi recompiles rtp.c with a 2056 max time skew - this removed some of the warnings and "chopper sound"
08:22.08pooh_ok...
08:23.02pooh_interesting RTP patch, got a URL for that pls?
08:24.24desktopheroi ended up finding that when searching for issues about data packets being reached prior to voice packets. it is a bug. and this was the solution;)
08:24.34desktopheroi can look it up
08:25.11pooh_thx
08:26.32desktopherohttp://bugs.digium.com/bug_view_page.php?bug_id=0001195
08:26.50desktopheroi think this is part .... see above
08:28.07pooh_thx!
08:28.35desktopherothis is the line in rtp.c in src : #define MAX_TIMESTAMP_SKEW
08:28.44moralewow this www.link2voip.com provider is looking pretty decent.
08:28.51morale23ms latency
08:28.55desktopherodefault is like 64...not enough over a WAN ;)
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08:29.42desktopheromorale - I wish I could get that over my WAN ;)
08:30.02desktophero58 is a wet dream ;)
08:30.28desktopherosorry if that offends anyone....but this has being going on for about 6 months
08:30.37moraleim only on a cable modem.. that is doing a traceroute to them, im not sure what its going to add up to once it starts encapsulating the sip/iax packets
08:32.01desktopheromy connections are a business DSL. my consumer cable modem gets better sound quality over SIP that the VPN connected IAX2 trunk
08:32.38desktopherodoes my last statement make sense??
08:32.52desktopheroup tooooooo late. ;)
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08:37.22pooh_VPN eats bandwidth too
08:37.27pooh_IP eats bandwidth
08:37.33pooh_and VoIP eats bandwidth
08:38.03pooh_next to latency we have the issue of bandwidth ;-)
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08:47.25tzafrir_laptopwhy would you need a VPN for IAX?
08:47.46pooh_Maybe a corp descision
08:53.58Dr_RayI think people equate vpn with security
08:54.25Dr_Raymy IAX box is on the outside, with a healthy set of iptables.
09:03.31tzafrir_laptopDr_Ray, iptables adds latency. build the firewall so that IAX2 packets go through a minimal number of rules
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09:11.14astcryzHi :-)
09:11.32*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
09:12.01astcryzI have a little problem. i have intsalled mpg123 and music on hold is running, but i dont get anything but scramling and wicked sounds, is that because i do not have a sound card in the asterisk machine? or what can it be?
09:12.06astcryzany help would be appreciated! :)
09:13.43moraledid you install mpg123 0.59r ?
09:13.55moralewhat version is it of mpg123
09:14.35BladeRunner05someone use chan_capi ?
09:15.14astcryz0.2.10 is the version
09:16.01moralethat is mpg321 not mpg123 i think.. try installing mpg123 0.59r
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09:17.43astcryzhm, i will try
09:17.45astcryzthank you
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09:35.03BladeRunner05I need a good extension.conf example to study, who have one ?
09:35.46moraleanyone know when a final release for 1.2.0 is supposed to be?
09:39.37tzafrir_laptopanybody know of a free software for call quality testing?
09:43.32Igbothom_IIIyour ear?
09:43.38Igbothom_IIIyour brain's soft  :)
09:43.46Igbothom_III(well, mine is, at least)  :)
09:44.05Igbothom_IIIBladeRunner05; me too - try http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
09:51.26BladeRunner05Why can call out with this: exten => _XXXXXXXXXX,2,Dial,CAPI/959010${CALLERIDNUM}:b${EXTEN:1})
09:51.35BladeRunner05sorry Why can't call out with this: exten => _XXXXXXXXXX,2,Dial,CAPI/959010${CALLERIDNUM}:b${EXTEN:1})
09:53.01BladeRunner05it say: Unable to create channel of type 'CAPI'
09:53.12BladeRunner05But it answer the call
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09:57.25tzafrir_laptopIgbothom_III, my ear is known to be very insensetive. Even compared to an "average person"
10:00.56snittyou mean insensitive?
10:02.53tzafrir_laptopinsensitive to typos as well, I figure
10:03.50snitt:)
10:03.51tzafrir_laptopIt is also a matter of memory: is the current sample better than the one I heard two minutes ago? or the one two days ago?
10:04.07snittcan i use Set(TIMEOUT(digit)=5) with * version < 1.2?
10:05.36snitti did not say that the current version is 1.2
10:05.38tzafrir_laptopsnitt, I figure you can, but it has a different name
10:05.44snittoh yes i know
10:05.48snittbut im downgrading now
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10:10.41tzafrir_laptopany idea how to implement a "for" loop in the dialplan without using an AGI? What I don't know how to do is to decrement the counter.
10:11.10pooh_GotoIf ?
10:11.33tzafrir_laptoppooh_, yes, but how do I decrement the counter?
10:11.41pooh_var1 =var - 1
10:11.47pooh_var1 = var1 -1
10:11.56tzafrir_laptopIs this syntax supported by SetVar?
10:12.00pooh_yes
10:12.51tzafrir_laptopnow any ideas which of the standard sound files takes exactly (or little less than) 1 second?
10:13.19snittsilence/1?
10:13.20pooh_exten => s,6,SetVar(current_loop=$[${current_loop} + 1])
10:14.08pooh_beep or maybe silence in different lenghts ?
10:14.15pooh_snitt: ;-) yes
10:14.25tzafrir_laptoppooh_, actually, it is: _23. => do_something_with_${EXTEN:2}
10:14.37pooh_so?
10:14.50pooh_what do you want with that what does it have to do?
10:15.14tzafrir_laptopwell, seems like what I need
10:15.44pooh_Still do not understand what you need
10:16.05pooh_'do something'  can be a awefull lot ;-)
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10:20.42snitt+++ killed by SIGKILL +++
10:20.43snitt:DD
10:21.33littleballhello, i have a conference setup by using meetme. When all parties disconnected, i still can see there is one user in the conference room. and one channel is still occupied. how to recover this channel?
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10:26.44tzafrir_laptoppooh_, I call 23xxx with a call file, and this gives me a timed-out "doing something". Just trying to load my server
10:27.08tzafrir_laptopI also use Wait and WaitMusicOnHold for other types of "doing"
10:28.57tzafrir_laptoplittleball, kick the user?
10:29.20tzafrir_laptop(or another solution that involves a boot)
10:31.34littleballin my case, i tested three users in one conference room. When each user leave the room (just disconnect the call), a deadagi will be called.
10:32.46littleballbut when one of the three users disconnected, it seems that the asterisk doesn't notice  such a disconnection.So. asterisk still occupy a zap channel until i restart the asterisk
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10:34.22littleballtzafrir_laptop, kick user doesn't work. (1) when my users use conference call, no way for me to know whether the conference is live or not. (2)it should be cleaned automatically
10:42.49tzafrir_laptoplittleball, so the problem is with chan_zap, and not with the meetme app
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10:43.36littleballit could be
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11:02.22astcryzRoyK`;)
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11:15.28astcryz:-)
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12:49.18cryzeckForbidden - wrong password on authentication for INVITE to
12:49.19cryzeckwhat is that?
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14:41.40puzzledhi all
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15:19.42InfraRedhi
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16:01.51*** join/#asterisk Xen^ (n=linux@202.63.195.71)
16:02.17Xen^hello all
16:02.20Xen^can some one tell me can i run real players files using asterisk ?
16:02.33Xen^like we have MP3Player() command
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16:07.35*** join/#asterisk mmmToop (n=chatzill@mtngprs7.mtn.co.za)
16:10.01SkramXXen^: Convert to GSM :)
16:10.17Xen^SkramX : how can i convert if streaming it live :)
16:10.23Xen^means real time :)
16:11.26SkramXWhat is the file ext.?
16:12.06Xen^file.rm
16:12.32file[laptop]you can't.
16:12.35*** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net)
16:13.05Insanity5whos treams in rm anymore anyways?
16:13.17SkramXheh
16:13.19file[laptop]people do.
16:13.25file[laptop]I know, it's crazy
16:13.36Qwell[]file[laptop]: You missed some good times last week. ;]
16:13.38Xen^we are doing it
16:13.40Insanity5It's really a crappy format that never scaled beyond dialup.
16:13.41Xen^here in Pakistan
16:13.43file[laptop]Qwell[]: I bet
16:13.51Xen^for earthquake
16:14.03Xen^so that people can dial and listen to news :)
16:14.04file[laptop]Xen^: asterisk can't deal with realplayer stuff, at all... so have fun
16:14.44Xen^:(
16:14.52InfraRednot sure it can deal with streamed news either
16:14.53supaigtrOnly way I know is to take output of soundcard and reformat on another machine mic input.
16:14.57InfraRedstreamed mp3
16:15.07file[laptop]restream it as mp3...
16:15.11InfraRedsupaigtr: no
16:15.11Insanity5play it out to a soundcard, loop it back in to microphone, and reencode.  It will sound like crap, but that's ok, telephone and rm alwready sound like crap.
16:15.16SkramXXen^: how do callers dial in if land line phones are out.. and I assume cell coverage is hay-wire aswell..
16:16.51Insanity5shortwave radios is the best solution :)
16:17.32Xen^SkramX : well dude
16:18.06InfraRedXen^: get shoutcast server running
16:18.16InfraRedit can take the input from wav/sound out
16:18.20InfraRedthen stream it as mp3
16:18.25InfraRedconnect to that and stream it
16:18.28Xen^InfraRed : well i have slimserver :)
16:18.33InfraRedtranscoding at its worst
16:18.36InfraRed:)
16:20.01Insanity5and then listening over phone lines
16:20.03Insanity5hehe
16:20.08*** join/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net)
16:20.11Insanity5one mroe transcode
16:20.24Xen^?
16:20.57InfraRedthis is xen reporting from the moon
16:21.24Xen^umm
16:22.09Insanity5It will transcode again from mp3 to g711u
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16:31.59Wi_Fiyo
16:32.02Wi_Fii set up FWD with AAH and get busy signal when dialing 393612
16:32.25Wi_Fiand nothing showing up in asterisk -rvvvvvv
16:35.29Wi_Fihehe
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16:41.00toresbehey guys
16:41.21toresbeAnyone here have a modem accessible with asterisk?
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16:41.35kFuQsup shido6
16:41.40shido6zzZzzz
16:41.43kFuQl0l
16:41.46shido6nicotine in the morning
16:41.49kFuQyah
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16:41.58toresbeI have a terminal here with an acoustic coupler, y'see, and I want to "dial up" a modem
16:41.59kFuQthe 'ine' drugs
16:42.30toresbewhistling only gets me so far :)
16:42.54InfraRedterminal with an acoustic coupler
16:42.57InfraRedhaha
16:43.05InfraRedyou should ebay it
16:43.11kFuQthat like 900baud or something ?
16:43.17kFuQl0l
16:43.22InfraRedprobably 2400
16:43.27InfraRedor 300 like the one i had
16:43.30toresbeIt's 300
16:43.31InfraRedwell, used
16:43.35InfraRed\o/
16:43.37toresbeTI Silent 700
16:43.44InfraRedget rid of it
16:43.46InfraRedold POS
16:43.48toresbeWTF
16:43.59toresbeDo you know how much I've searched for this?
16:44.10toresbeIt's my favorite classical piece of computing
16:44.16kFuQuse it for dialing on payphones lol
16:44.16InfraRed18 minu=tes?
16:44.46InfraRedtoresbe: someone is giving away a PDP-11 in london if you want it
16:44.49InfraRedfits well
16:45.43toresbeI already have a PDP-11 :)
16:45.51InfraRedheh
16:46.00InfraRed'and a nuclear power station to run it'
16:46.02toresbeInfraRed: maybe you should post on the ClassicCmp list?
16:46.11InfraRednot mine
16:46.11toresbeInfraRed: Mine doesn't spend that much juice
16:46.21InfraRedi am getting rid of stuff
16:46.25InfraRednot aquiring more
16:46.27InfraRed+c
16:46.28toresbeheh
16:46.39InfraRedi am down to 2 rooms missing from the house
16:46.44InfraRedi was missing 3 rooms
16:46.46InfraRed:)
16:46.53Insanity5you collect a lot of crap
16:46.57Insanity5:)
16:46.59InfraRedyes!
16:47.16InfraRedi am giving away 7u compaq proliant 3000R
16:47.16Insanity5Evidentely, you need to throw more stuff away.
16:47.23Insanity57u?  ouch
16:47.23InfraRedi am !
16:47.24Insanity5shipping
16:47.35InfraRedna, i emailed freecycle, some people want em
16:47.38*** join/#asterisk erickj_az (n=erickj_a@wsip-68-98-222-74.ph.ph.cox.net)
16:47.42Qwell[]heh, cheaper to buy a roundtrip flight
16:47.43InfraRedyou should subscribe to freecycle
16:47.52InfraRedit's the goodness
16:47.56Insanity5I miss the old freeboxen.com site
16:47.57BladeRunner05Hi sir, I have a problem, I can receive call, call an internal extension, but where I try to call out it return this error: Oct 16 18:43:43 NOTICE[589841]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = . you should check your config!
16:47.58BladeRunner05Oct 16 18:43:43 NOTICE[589841]: app_dial.c:742 dial_exec: Unable to create channel of type 'CAPI'
16:48.10InfraRedInsanity5: archive.org
16:48.15BladeRunner05and in the extension.conf I use: exten=>_0.,1,Dial(CAPI/@${4959530}:B${EXTEN}|30)
16:48.16erickj_azdoes anyone know how to lengthen the amount of time voicemailmain waits for a user to enter their extension?
16:48.32BladeRunner05where is the problem =?
16:48.33Insanity5InfraRed - lol, any drives in it?
16:48.35Qwell[]BladeRunner05: Is 4959530 a variable?
16:48.42InfraRedInsanity5: 3x 9GBs
16:48.46BladeRunner05Qwell[]: is the msn
16:48.48InfraRedand raid array card
16:48.49Insanity5InfraRed - well, it's no longer active after the maintained gave up.
16:48.56Insanity5InfraRed - stirp the ram and drives, ebay, toss the rest.
16:49.05InfraRedthinking of doing that tbh
16:49.29Qwell[]BladeRunner05: Is it a variable?
16:49.35Qwell[]If not, why is it in ${}?
16:49.45Insanity5But you'll still only get $10-20 unless it's got a crap-ton of ran.
16:50.07BladeRunner05Qwell[]: ok I remove ${}
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16:51.10BladeRunner05Oct 16 18:47:53 NOTICE[606225]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 4959530. you should check your config!
16:51.10BladeRunner05Oct 16 18:47:53 NOTICE[606225]: app_dial.c:742 dial_exec: Unable to create channel of type 'CAPI'
16:51.20BladeRunner05Now this is the error that appears
16:51.43erickj_azdoes anyone know how to lengthen the amount of time voicemailmain waits for a user to enter their extension?
16:52.18Qwell[]BladeRunner05: Did you check your config?
16:52.54BladeRunner05I use chan_capi and don't know what to check !?!?!?!
16:53.13Qwell[]~docs
16:53.14jbot[docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:53.16Qwell[]BladeRunner05: Start there
16:55.45InfraRedqurestion regarding (can)reinvite, once the call is estableshed and the media stream is now going between the phone and destination does the sip control command still go through the server?
16:56.04InfraRedor are they direct between the phone and the peer ?>
16:57.00erickj_azdoes anyone know how to lengthen the amount of time voicemailmain waits for a user to enter their extension?
16:57.09*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
16:58.19file[laptop]InfraRed: signalling still goes through server
16:58.30InfraRednice!
16:58.33InfraRedgood stuff
17:01.32erickj_azdoes anyone know how to lengthen the amount of time voicemailmain waits for a user to enter their extension?
17:01.52Qwell[]erickj_az: No need to ask over and over
17:02.15erickj_azI was told to do that last time I was here
17:02.18Qwell[]asking repeatedly basically means "Hey, I'm more important than everybody else.  Look at me!  Look at me!"
17:02.24Qwell[]erickj_az: sarcastically perhaps
17:02.48erickj_azI'm not more important than any one else here.
17:02.58Qwell[]Then please act like it...
17:03.06erickj_azI am wating for a lull in the chat before I ask again.
17:03.21Qwell[]What lull?  I can see all three times you've asked, in one screen
17:03.44erickj_azHence the tremendous voume of chating going on
17:03.54erickj_azThanks for the tip.
17:04.06Qwell[]tremendous volume...right
17:04.16Qwell[]because 15 lines is "tremendous"
17:04.21*** join/#asterisk mud (n=mud@bestekdsl.customer.sentex.ca)
17:04.26erickj_azPoint and match
17:04.30Qwell[]12, actually
17:04.58file[laptop]'tis rude and makes the most experienced of us not want to help you
17:05.21Qwell[]file[laptop]: Why weren't you at astricon? :P
17:05.32erickj_azIf no one here wants to help me, let me know and I'll go away.
17:05.35file[laptop]Qwell[]: because a plane ticket was $1000+
17:05.41Qwell[]we went to this bar...he would have been cool with you drinking there, heh
17:06.01file[laptop]but, like, I'm 19 and legal drinking age is 21! sillyness
17:06.14Qwell[]like I said...that bartender would have been cool with it. :p
17:06.19file[laptop]scary
17:06.25file[laptop]life goes on however
17:06.31Qwell[]next year ;]
17:07.44erickj_azJust a thought....
17:08.05fiber0ptiI'm trying to make it so if the user dials their own extension from their phone it takes them directly to their mailbox and asks for their password. Here's what I have and doesn't work: http://pastebin.com/395550
17:08.10erickj_azThere is a reason people buy products from companies like Microsoft.
17:08.19erickj_azIt's called support.
17:08.27Qwell[]erickj_az: Support from MS is *NOT* free
17:08.35Qwell[]This *IS* free.  Deal with it.
17:08.37MikeJ[Laptop]erickj_azm there is support for asterisk
17:08.46MikeJ[Laptop]you can pay for all the support you want.
17:08.53Qwell[]fiber0pti: what version?
17:08.57erickj_azYou all in the Asterisk world tout this open source environment
17:09.02fiber0ptiQwell: 1.0.9
17:09.03MikeJ[Laptop]free support however requires patience
17:09.08Qwell[]fiber0pti: try something like
17:09.23erickj_azI see.
17:09.24Qwell[]x${CALLERIDNUM} = x${EXTEN}
17:09.39Qwell[]fiber0pti: See what thats doing?
17:09.54MikeJ[Laptop]and no, I don't off the top of my head know.. .
17:09.54fiber0ptiactually no. what does the "x" do?
17:10.04erickj_azHopefully Lord Bill witll take an intrest in Asterisk
17:10.05MikeJ[Laptop]is there a config option in the sample conf for it/
17:10.05Qwell[]fiber0pti: its a literal x
17:10.09Qwell[]x500 = x500
17:10.30InfraRedsome company published asterisk book online as pdfs
17:10.31InfraRed\o/
17:10.35Qwell[]fiber0pti: and take it out of quotes.
17:10.41Qwell[]InfraRed: The authors did
17:10.47ManxPowerfiber0pti, It's an old shell scripting trick.  they could not test for an empty variable either.
17:10.52InfraRedcool
17:11.00Qwell[]InfraRed: O'Reilly let them publish it under the Creative Commons license.  You should still buy the book though, it's great
17:11.11file[laptop]the book rocks
17:11.12erickj_azI did.
17:11.19Qwell[]file[laptop]: Is yours signed?
17:11.24InfraRedi probably will
17:11.25file[laptop]Qwell[]: no, sadly not
17:11.29Qwell[]:(
17:11.32InfraRedi'll sign it
17:11.33Qwell[]Should've come to Astricon.  heh
17:11.34InfraRedfor a small fee
17:11.42file[laptop]yeah... if I had money :P
17:11.55fiber0ptiHrm.. that's putting me into voicemail right away..
17:11.56Qwell[]free book signing!  totally worth it. :D
17:12.29InfraRedits missing 2 appendixes tho
17:12.58Qwell[]file[laptop]: Ship it to all of the authors, and have them sign it by mail.
17:13.01fen[mobile]erickj_az: looks to be hardcoded to me in app_voicemail.c
17:13.34*** join/#asterisk ben_d (n=ben@cpe-66-66-209-96.rochester.res.rr.com)
17:14.33file[laptop]Qwell[]: excellent idea lol
17:14.48erickj_azfen[mobile]: OK then I should change the code.  Is there a way to just re-compile app_voicemail.c and install it.  I'm new to linux.
17:14.57InfraRedi need to hire a coder forsome asterisk voiodoo
17:15.21*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
17:16.28Qwell[]file[laptop]: Just put three prepaid envelopes and the book into one envelope, ship it to one, and put instructions on how to ship to the next. :p
17:17.35MikeJ[Laptop]erickj_az, if you change it in the code and just re-do the make and make install, it will just recompile the stuff it needs to
17:18.05erickj_azThank you.
17:18.41fiber0ptiQwell: Did you mean something like this? http://pastebin.com/395565
17:18.43erickj_az<PROTECTED>
17:19.08Qwell[]fiber0pti: something like that, yeah
17:19.22Qwell[]except
17:19.24fen[mobile]erickj_az: iirc look at vm_authenticate
17:19.26Qwell[]s|1000 ?
17:19.42Qwell[]What is _XXX,1000, and what is s,1,?
17:20.29fiber0ptiQwell: That makes it so if I call my own extension from my phone it just sits there without connecting and if I dial someone elses extension it says they are unavailable.. any ideas?
17:20.57fiber0ptiI'm trying to make it that if they don't match this exten does nothing.. is that not correct?
17:21.05Qwell[]no, wait, thats right...nevermind
17:23.01fiber0ptiQwell: Any idea why it's doing that?
17:23.13Qwell[]Whats it doing?
17:23.20Qwell[]oh
17:23.51*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
17:24.08fen[mobile]erickj_az: timeouts are passed to ast_readstring(), which you can find in channel.c
17:24.16Qwell[]fiber0pti: paste the output you see on the CLI
17:25.05fiber0ptiQwell: If i dial my own ext: -- Executing GotoIf("SIP/160-52a8", "x160 = x160?s|1000") in new stack
17:25.26*** join/#asterisk Ahrimanes (n=michael@195.137.237.81)
17:26.23Qwell[]Does it say its moving to s,1000?
17:26.23fiber0ptiQwell: If I dial another extension this is what I get: http://pastebin.com/395574
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17:28.25*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
17:28.58[hC]Hmm. I have an AgentCallbackLogin, that currently uses SIP/${CALLERIDNUM} to log you in, but of course its not always a SIP/ technology doing it. Is there a variable for which technology you were on, or should i just use Local?
17:31.52*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
17:33.40*** join/#asterisk Assid (n=assid@203.115.64.62)
17:33.41Assidheya
17:35.16erickj_azfen[mobile]: I got to  ast_waitfordigit Do you think that would be the right place to do it?
17:35.41fiber0ptiQwell: And where should i put this?
17:35.51Qwell[]put what?
17:36.08erickj_az<PROTECTED>
17:36.52erickj_az<PROTECTED>
17:36.57fiber0ptiQwell: The _XXX
17:37.08fen[mobile]erickj_az: iirc the timeout is in the call to ast_readstring in vm_authenticate - the 2000 and 10000 - try changing those to what you want
17:37.13Qwell[]You already have the _XXX
17:37.20Qwell[]<Qwell[]> Does it say its moving to s,1000?
17:37.26Qwell[]in the CLI
17:38.21erickj_az<PROTECTED>
17:40.56fiber0ptiohh.. didn't see that... if I dial my own ext no.. it jus has the one liner until timeout: Executing GotoIf("SIP/160-7208", "x160 = x160?s|1000") in new stack
17:43.23fiber0ptiqwell: and if i dial another ext it doesn't say moving to s,1000 it says "-- Executing VoiceMail("SIP/160-a571", "u200") in new stack" which is s,1000
17:43.41Qwell[]With the u?
17:43.48Qwell[]That wasn't in your pastebin
17:43.53Assidheya Qwell[]
17:43.58Qwell[]Assid: hi
17:44.00Assidhows u
17:44.09Qwell[]alright
17:44.15fiber0ptiqwell: it's in the second line of http://pastebin.com/395574
17:44.22Qwell[]fiber0pti: the u isn't
17:44.32Qwell[]erm...
17:44.37Qwell[]not in 395565
17:45.55fiber0ptiQwell: correct.. should it be? I just tried it and the behavior is the same.
17:46.10Qwell[]no, I'm saying...it's not s,1000, if the u is there
17:48.49JerJertraveling makes me so lazy just after i get bacl
17:48.53JerJerk
17:49.02*** join/#asterisk Eyecon (n=matt@jumala.plus.com)
17:49.12erickj_azDo you think I need to do a reload or should I stop Asterisk and re-start it after the compile
17:49.15Qwell[]JerJer: Don't pass out in the code zone again. :p
17:49.18Qwell[]erickj_az: restart it
17:49.31*** join/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
17:49.42JerJerpower naps
17:49.45fiber0ptiQwell: I guess I don't know what you mean by that.. :S
17:49.56Juxthello
17:50.05Qwell[]fiber0pti: VoicemailMain(${EXTEN}) is not the same as VoicemailMain(u${EXTEN})
17:50.08Eyeconhi all, I've been trying to get asterisk to work; I'm using kphone to connect to it and do the 1000 test; I hear the test and if I use DTMF # to quit then that works fine, but if I use the hangup button in kphone, asterisk segfaults
17:50.12Eyeconany ideas?
17:50.13Qwell[]the latter is being called in your dialplan
17:50.21Qwell[]the former is s,1000
17:50.40fiber0ptiQwell:
17:50.54fiber0ptiNod.. but that line doesn't seem to be called with the statement is true
17:50.59Qwell[]in fact, no
17:51.06Qwell[]Voicemail(u${EXTEN})
17:51.08Qwell[]not voicemailmain
17:51.31Qwell[]JerJer: my nap was taken during Keiths talk...
17:52.11JerJerhell yeah
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17:53.57*** part/#asterisk pa (n=Paolo@unaffiliated/pa)
17:55.14fiber0ptiQwell: So I changed it to Voicemail(u${EXTEN}) and it still doesn't connect when dialing my own extension and when I dial another extension it still says they are unavailable. I don't know what it's doing but it seems to be ignoring the Gotoif when it's true.
17:55.35Qwell[]fiber0pti: pastebin the important parts of th dialplan...
17:55.46Qwell[]two lines doesn't help at all
17:56.26fiber0ptiQwell: http://pastebin.com/395610
17:56.32fiber0ptiohh..
17:56.36fiber0ptiwell that's all it is though..
17:57.00Qwell[]no, that isn't all it is...
17:57.06Qwell[]voicemail wouldn't be getting called if that were the case
17:57.58*** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com)
17:59.29*** part/#asterisk brookshire (n=pfffft@esbrooks3.traveller.com)
17:59.33erickj_azIt looks like the first number (2000) is the length of time it waits for the next digit and the second number (10000) total time?
18:00.03Qwell[]erickj_az: Does it really take your users 10 seconds to put in a voicemail box?
18:00.57erickj_azYep...He is an MD and PhD
18:01.30erickj_azAn most likly smater than anyone one else I know
18:01.50Beirdoand hopefully a better speller than most of us :)
18:02.16erickj_azHe is also the guy who saved my life last year when I was diagnosed with a brain tumor.
18:02.30Beirdocool
18:02.44Qwell[]make him fix it then
18:02.57erickj_azSo, for all I care it could take him several hours to enter his mail box number and I'd wait for him.
18:03.15*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
18:05.15erickj_azSo, is the second number actually the total time or is it somthing else?
18:09.00erickj_azI want to thank you all for your help.  I'm excited that I actually edit the c code correctly.  I hope as time gos on I'll be able to make a greater contribution to the ASterisk commnity.
18:12.13*** part/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
18:15.00jdv79how could one load test using the quad t1 cards...
18:15.27Qwell[]jdv79: You could probably crossconnect them, and make 46/48 "outgoing" calls
18:15.42jdv79how about 2 boxes?
18:15.51jdv79i want to see if i can get 2 cards in one box
18:15.52Qwell[]do they both have 4 ports?
18:16.06jdv79so 2 boxes, 4 cards - 1:1?
18:16.07Qwell[]jdv79: 2 cards in a box isn't recommended
18:16.13jdv79i know it isn't but i want to try
18:16.18Qwell[]then sure, try it
18:16.24jdv79hence the load testing idea:)
18:16.26jdv79aha!
18:17.01jdv79with the new firmware and no transcoding i might have a shot;)
18:17.20Qwell[]better do echo can on the card then...heh
18:17.30jdv79yup, best cards all around
18:18.10Qwell[]What type of machines?
18:18.35jdv79dell 2850s
18:18.40Qwell[]specs?
18:18.49jdv79dual 3.0 xeon
18:18.54jdv792G ram...
18:18.55Qwell[]32?
18:19.04jdv7932 whats?
18:19.06Qwell[]bit
18:19.11jdv79uh, no
18:19.18jdv79they're all 64 it looks like
18:19.39jdv79i may end up running 32 bit linux though
18:19.45Qwell[]why?
18:19.48Qwell[]pointless
18:20.04Qwell[]severly hamper your performance
18:20.08jdv79i've heard that the 64 bit version can be unworthy
18:20.15jdv79i'm gonna try it first
18:20.16Qwell[]You heard very wrong.
18:21.14Qwell[]in that setup, I'd be using a Gentoo stage 1, with almost nothing installed
18:21.55jdv79installed & running at 2 diff things but ok
18:21.59jdv79are rather
18:22.09Qwell[]not with gentoo...
18:22.20Qwell[]USE="-mysql", for instance
18:22.53jdv79are you talking about stuff installed on the box or stuff compiled into asterisk or something?
18:22.59Qwell[]all of the above
18:23.12Qwell[]noload a shitton of modules in asterisk
18:23.24jdv79that's just memory though, right?
18:23.33jdv79still no point in not doing it
18:25.39jdv79hmm, would there be a marked difference in looping 2 cards in the same box against eachother or just having another box with 2 cards and doing a 1:1 between the 2 boxes...
18:26.00Qwell[]probably not.  try all three methods
18:26.12Qwell[]or if they're identical machines, two methods
18:26.20jdv79they are all identical
18:26.25jdv79clones, we have clones!
18:28.14*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
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18:32.03ronaldl79Hello.
18:32.42ronaldl79I've always wondered, but just what exactly are you getting with 'CVS HEAD'? Are these nightly builds? And if so, do they contain new features?
18:33.03ronaldl79I'm just curious, because although I'm running 1.2 beta, I want to be sure that I'm not missing out on something in the HEAD.
18:33.08Qwell[]ronaldl79: They are updated randomly throughout the day, whenever somebody commits a patch, its in CVS HEAD
18:33.51ronaldl79Thanks, Qwell.
18:34.10Qwell[]so yes, tons of new features
18:34.15Qwell[]and also new bugs
18:34.23ronaldl79Even those not available in 1.2 beta?
18:34.32Qwell[]yes
18:34.35ronaldl79Really....
18:34.50ronaldl79Qwell, are you using HEAD?
18:34.54Qwell[]I am
18:35.05ronaldl79Damn, I need more boxes to run these different versions.
18:35.20ronaldl79Sounds HEAD is more experimental, and I'm an experimenting type of guy. :)
18:35.31*** join/#asterisk KriS83 (n=KriS@212.202.141.92)
18:35.49ronaldl79Qwell, are these changes viewable via the CVS log?
18:35.59KriS83hi
18:36.02Qwell[]ronaldl79: They are
18:36.08ronaldl79Good deal.
18:36.11ronaldl79Thanks, Qwell.
18:36.14ronaldl79Hi, Kris83.
18:36.15Qwell[]ronaldl79: if you're interested, subscribe to the asterisk-cvs mailing list
18:36.31jontowronaldl79: CVS HEAD is the true moving target branch; its never tagged as a release, always just development
18:36.31ronaldl79I'll just view it...I don't do so too often.
18:36.46ronaldl79Interesting....'
18:36.55ronaldl79Where was 1.2 derived from?
18:36.58jontowit is the basis for new major-version releases, but it is "unrefined" as far as bugfixes go
18:37.13ManxPower1.2Beta is a snapshot of CVS-HEAD
18:37.14ronaldl79Jon, are those additions documented for implementation?
18:37.26jontowin code, and occasionally text otherwise ;)
18:37.36jontowmailing list archives are your best bet
18:37.40ronaldl79Really, so the new features in 1.2 beta....were already active in HEAD, eh?
18:37.41ManxPower~mailinglist
18:37.44jbotmailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
18:37.48Qwell[]ManxPower: its just a snapshot, and not a branch?
18:37.54ronaldl79Yeah, I'm subscribed to users.
18:37.57ManxPowerQwell[], correct.
18:38.12ManxPowerQwell[], once it's out of beta I believe it will be a branch
18:38.14ronaldl79How well is the HEAD working for you, Awell?
18:38.23jontowyeah, it'll be tagged as it's own branch when it hits -release
18:38.28Qwell[]ManxPower: So are features not currently being added to CVS?
18:38.31jontowthats the model as i understand it, anyway
18:39.18KriS83I need some help with Asterisk ;( What I need asterisk to do is pickup Phonecalls from MSN 27 (via AVM B1 ISDN Card) and Play some musik or Have a VoiceBox or whatever. This is supposed to be the first step. Has anybody got a tutorial for me on how to set this up? I am very confussed by the mass on configs included in Asterisk
18:39.18jontowqwell; i believe feature stabilization is the point of the beta.. its just bugfixes, iirc.
18:39.19ManxPowerQwell[], as I understand it, based on the guidelines from kpflemming, anything that was not aleady in the bug tracker as a feature addition after some specific date (aug 15?) will not be added to 1.2
18:39.34Qwell[]I see, okay.  I do recall him saying that not at the dev meeting
18:39.38Qwell[]now*
18:40.46jontowalternately, as i had no luck at ~3am -- anyone played with sqlite [3?] .. res_sqlite or cdr_sqlite specifically?
18:42.09ronaldl79So, to get CVS HEAD, would you simply tag it (-r) as 'CVS HEAD', or just the default checkout?
18:42.24jontowronaldl79: "cvs checkout asterisk asterisk-addons libpri zaptel ......."
18:42.29jontowno -r requiresd
18:42.31jontow-s
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18:42.38jontowholy crap its windy out, hm
18:42.51ronaldl79Oh, well, hell....I've been using CVS HEAD all along then....lol....
18:42.55jontow:P
18:43.08jontowsmooth.
18:43.08jontowhehe
18:43.20ronaldl79CVS is still new for me, so I'm still learning how the process works with Asterisk.
18:43.27*** join/#asterisk damned (n=Victor@prior.lanck.net)
18:43.42ronaldl79So, just to clarify once more, the latest CVS download has everything in 1.2 beta...plus any patches, correct?
18:46.29jontowand features
18:46.35jontowit is not tied to 1.2.0-beta1
18:46.49jontow"nothing is held back" in CVS-HEAD :)
18:46.56ronaldl79Good explanation! :)
18:47.09jontowthere are no restrictions except that the code must go through the peer review etc
18:47.19jontowcompletely experimental by definition
18:47.34jontowbut that doesn't necessarily mean its unstable,
18:47.45ronaldl79Well, it sounds like I'm limiting myself by download 1.2 beta .... I'll just continue with what I've always done...downloading CVS head
18:47.55jontowi've successfully used it in a stable manner in a production environment or two when some of the features were called for and were not available in 1.0.x
18:48.06ronaldl79Of course not, because for the past year, I always downloaded HEAD...and it worked fine.
18:48.31ronaldl79My confusion is coming to rest I think. :)
18:48.36jontowi know many people who swear by HEAD only, but .. i'll be honest -- if you're gonna use it, make sure you have a good test environment setup ;)
18:49.01jontowyou *will* find the occasional headache when something just isn't right
18:49.21jontowand the advice is generally "cvs update and rebuild.. you may have found a bad point in the tree"
18:49.34ronaldl79Yeah
18:49.45jontowspecifically, it is possible to catch a moment where something is half-committed or committed in a broken state and someone catches that, fixes in 5 minutes, and you've got the broken one.. ;)
18:49.46ronaldl79I've read instances where folks updated later that night, and the build was successful.
18:49.46*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
18:50.15jontowjust live by "make clean ; cvs update -dP"
18:50.29jontow;)
18:51.10ronaldl79hmmmm
18:51.14ronaldl79-dP ...
18:51.21ronaldl79I always clean....:P
18:51.38ronaldl79But those options doesn't ring a bell...cause I never use 'em.
18:54.26jontowheheh
18:54.33jontowwell, investigate 'em
18:54.35jontowmay come in handy :)
18:57.03ronaldl79Are you guys using inband or rfcxxx for DTMF in your sip.conf?
18:57.14Ahrimanesrfc..
18:57.19ronaldl79Same here
18:57.48ronaldl79I was testing my first IVR a few weeks ago, and people couldn't navigate the menus...until I switched from inband to rfc.
18:58.26ronaldl79The only thing I'm seeing is a delayed response to DTMF ... and I don't believe it's anything on my end.
18:58.32ManxPowerronaldl79, That would be expected.
18:58.55ronaldl79And sometimes, the DTMF tones aren't always detected....not cool.
19:00.16ManxPowerDTMF detection problems on Zap ports can frequently be traced down to an issue with rcgain/txgain
19:00.21ronaldl79I'm demoing * again tomorrow for the board of a client ... I hope they pee in their pants from the excitement over this PBX...telling everyone they know about the demo.
19:03.19ronaldl79My hope is to expand my clientele around VoIP and IT in general...we shall see.
19:03.42ManxPowerLast week I was having problems with IVR DTMF.  turned out that I needed to ingrease the rxgain on my Zap port by 3db.  today I had an issue with outbound calls.  turned out I had to increase txgain by 3db
19:03.54ronaldl79By the way, has anyone found any good comparison sheets for * vs. offerings from Nortel, Avaya with prices? I recall seeing one before, but don't remember the location.
19:03.59ManxPowerI hate this new keyboard
19:04.16ronaldl79Manx, I'm not using any hardware...
19:04.25ronaldl79What type of board did you buy?
19:04.28ManxPowerronaldl79, then you have a problem with your provider
19:04.39ManxPowerronaldl79, I was using X101P
19:04.42ronaldl79Bummer, don't tell me that...lol
19:04.50ronaldl79Maybe BroadVoice just sucks.
19:05.04ManxPowerbut I only use that board for local incoming/outgoing calls on my personal home Asterisk server
19:05.28ronaldl79Another thing I've noticed is a chirp just as * answers a call from the DID ... it's not *, but something on BV's end....
19:05.31ManxPowerronaldl79, Search the mailing list archives for "broadvoice DTMF" and you'll see just how much they suck
19:05.41ronaldl79Damn
19:05.59ManxPowerAnd people wonder why I use local Zap and PSTN connections.
19:07.13ronaldl79I really need to get into the hardware side of * ... I don't have any experience there.
19:07.55ManxPowerronaldl79, for prototyping you can use X100P or TDM400P cards, but for production I only use PRI and Zap T-1/PRI cards if I can.
19:10.16Dr_Rayronald - get the dev kit
19:26.29*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
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19:29.09infinity1haven't been here in awhile...
19:29.20infinity1hows the current HEAD? :)
19:29.45*** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com)
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19:36.54drbrownanyone had any success with snom phones and intercom??????
19:37.06drbrownspecifically w/ sipaddheader
19:38.59nexishas chanspy been pulled out of the cvs?
19:39.08filenope.
19:39.09jontowdrbrown; i had no problems making autoanswer-intercom work..
19:39.48drbrownjontow: what snom phones?? I am using snom 320's w/ the latest firmware
19:39.54jontow320, 4.1
19:40.12drbrownjontow: using sipaddheader???
19:40.13nexisodd, cas its not part of the cvs i grabbed.
19:40.16jontowno
19:40.19jontowi haven't touched sipaddheader
19:40.33jontowdidn't have a need to
19:41.01drbrownare you using auto answer??? on the phones themselves????
19:41.30jontowi setup a line (line 11, iirc) as an extension in the dialplan, setup auto-answer on that line
19:42.10jontowand thats..it
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19:43.00drbrownI would really like to use the sipaddheader method, would save a lot of typing, and then I could also use the allpage agi program
19:43.26infinity1wow. there is 1.2.0-beta1 now? so HEAD is finally stablizing?
19:43.40jontowheheheh
19:43.58jontowdrbrown: don't know what to say.. i setup a macro to do the dirty work and its.. doing quite well
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19:47.06docelm0Hola!
19:49.24infinity1hey
19:50.04docelm0Astricon was killer..  I have more pics to put up..  about 200 of them
19:51.51nexisim still trying to figure out why i dont have chanspy
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19:52.33nexisit should be in the src dir asterisk/apps/app_chanspy.c correct?
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19:55.23ManxPowerthey need astericon more often.  kram has done more updates to Asterisk CVS in the past week than he has done in the past 3 months
19:56.15infinity1ManxPower: becasue of astricon?
19:56.26ManxPowerinfinity1, I dunno, but astricon was last week
19:56.49infinity1ManxPower: ;) ... lots of people submitting patches eh
19:57.02Qwell[]ManxPower: Dallas close enough?
19:57.14ManxPowerQwell[], Yeah, or Texarkana
19:57.15docelm0yes.. I was sitting there besided him..  He was a coding fool..
19:57.21infinity1ManxPower: next one should be at Moscone Center in SF :)
19:57.33Qwell[]ManxPower: they were saying Astricon 2006 will be in Dallas, I believe
19:57.39docelm0Yes.
19:57.42ManxPowerSF would be cool too.  the problem is that conventions in big cities are expensive
19:57.42docelm0Dallas!
19:57.54infinity1ahhh ...San Jose!
19:58.11infinity1the only way i can go is if its driveable.
19:58.11docelm0Dude..  Its crazy..   Cost me almost 3000 for this trip
19:58.19infinity1i ain't driving to texas :)
19:58.25ManxPowerinfinity1, you're on the No Fly list?
19:58.45docelm0infi where you from?
19:58.59ManxPoweri'll be back later, need to rewire move the servers
19:59.11infinity1san jose. i saw digium at the linux world here a few months ago
19:59.39infinity1i don't use it enough to warrant the price to travel for it.
19:59.46docelm0ahh..  Sorry you were not there..   It was a fricken blast.
19:59.52infinity1sounds like it
19:59.53Qwell[]docelm0: Who were/are you?
20:00.09docelm0was @ Astricon last week now @ Tampa, FL
20:00.18Qwell[]who, not where :P
20:00.19WilliamKwhen is Astricon 2006?
20:00.28Qwell[]WilliamK: said we should know by Nov 15
20:00.30docelm0ohh Brian Fertig
20:01.11docelm0Qwell were you there?
20:01.14Qwell[]yep
20:01.31docelm0You know I bet we met and didnt know it.
20:01.43Qwell[]docelm0: likely.  I was hanging out with the IRC crowd most of the time
20:01.44docelm0Have you been to the astricon photo gallery yet?
20:01.53Qwell[]I saw the first pictures on Monday
20:01.56docelm0I am in there like 500 times
20:01.57Qwell[]thats it though
20:02.01Qwell[]got the link?
20:02.02docelm0check it out now
20:02.05*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
20:02.07docelm0astri2005.netdr.biz
20:02.36docelm0Astricon Photo Gallery is being hosted by NetDoctor Group, Inc.
20:02.37docelm0:)
20:03.04Juggieonly theres no pictures there
20:03.08Juggieor not enough
20:03.17Juggiebecause there are tons more
20:03.19docelm0Jug, I have like 600 picutes..
20:03.30docelm0I just couldnt upload them there cause of the fricken bandwidth
20:03.43Juggieit wasnt that bad
20:03.47Juggiei noticed no bw problems
20:03.53docelm0But I am gonna do an all call for more of them
20:03.57docelm0When I tried to upload
20:04.00docelm0download was good
20:04.09Juggieuse a better gallery
20:04.10Juggiethat one sucks
20:04.12Juggiehorribly
20:04.18docelm0Upload was like 20k if I was lucky
20:04.21docelm0I am going to recode it.
20:04.27docelm0sometime this or next week
20:04.41Juggiewhy recode it?
20:04.42Juggiehttp://gallery.menalto.com/
20:04.58docelm0Tried to install that..  Smarty kept crashing on me.
20:05.38docelm0Im gonna customize this one out so anyone can upload to it and add comments and such
20:05.45docelm0This is a basic shell of what I plan to do
20:05.50*** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
20:06.14infinity1what is AMPortal?
20:06.27*** join/#asterisk brookshire (n=pfffft@esbrooks3.traveller.com)
20:06.31Juggiewtf is smarty?
20:06.52brookshireit's a template engine for php
20:06.53Juggiea templat engine
20:06.56Qwell[]docelm0: 3623
20:06.57Juggiewhy are you worrying about that
20:07.03Juggiejust install gallery outside the engine
20:07.05Qwell[]I don't recall anybody taking a picture of me though, heh
20:07.06docelm03623 what?
20:07.17Qwell[]img_
20:07.33docelm0That was Junk-Y
20:07.39Qwell[]ahh, thats right
20:07.42*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
20:07.49brookshireomg.. never get drunk and decide to install gentoo
20:07.52brookshireheh
20:07.55mog_homelol
20:07.57mog_homebrooks
20:07.58InfraRedfreak
20:07.59InfraRed:)
20:08.02NuggetLinux is poo.
20:08.06mog_homeif i had a nickle for every time i did that...
20:08.12brookshirei misspelled all my compile options
20:08.19Qwell[]docelm0: Where is you in there?
20:08.26brookshireso all night it was compiling for ppc
20:08.26brookshirehaha
20:08.32docelm0Look for Docelmo
20:08.33Qwell[]brookshire: Thats awesome
20:08.34brookshireand it's a pentium3
20:08.43Qwell[]docelm0: ahh, okay
20:08.44docelm0I am with damin a few times
20:08.55Qwell[]yeah, I saw you a bunch of times
20:08.58kshumard_homenicely done, brooks
20:09.14brookshirei know right? :)
20:11.07brookshirewell at least it's replacing that fbsd install
20:11.14brookshirei hated fbsd.. heh
20:11.43docelm0What can I say..  Im popular
20:12.08infinity1why aren't there current debian packages for * ?  ...does everyone like compiling?
20:12.23Qwell[]infinity1: it's better that way, really
20:12.40infinity1Qwell[]: its a mega pain for someone that rarely does it :)
20:12.56Qwell[]it takes what, 5 minutes to compile?
20:12.57infinity1i always have to re-learn it
20:13.05brookshirei like the debian package for zaptel
20:13.08brookshireyou apt-get it
20:13.14brookshireand all it does is install the source code for you
20:13.20Qwell[]heh
20:13.33infinity1my biggest complaint is when you compile and install *, it installs all kinds of stuff in /usr/bin etc.
20:13.35Qwell[]dpkg --compile zaptel
20:13.39infinity1and not /usr/local/bin
20:13.47Qwell[]infinity1: You can change that...
20:13.57infinity1Qwell[]: how? ...:) ...
20:14.00Qwell[]Makefile
20:14.07infinity1hmmm ...i'll look.
20:16.20nexisis there any way to barge in on a call in progress?
20:16.41mog_homezapbarge
20:16.43mog_homeand chan spy
20:16.50fiber0ptiis it possible to get the status of a phone and utilize that information depending on what it is. Example: The user puts their phone status to "Out to lunch" and then have the dialplan use "Find me" functionality to go to the persons cell.
20:17.16nexismog_home, chanspy is not aval in the cvs i grabbed for some reason.
20:17.43brookshirenexis: did you grab head?
20:18.30Qwell[]okay...when you walk away for a minute...don't immediately read the last line of text in the channel.   ^^^
20:18.33nexisbrimstone, i dont play with cvs much, im more of a tar.gz guy, but i grabbed it with this. cvs checkout -r v1-0 asterisk blah
20:18.48brimstone?
20:19.02brimstonenexis: that's stable by the way
20:19.33nexisok, so i take it that chanspy is not in stable?
20:19.36brookshirechan spy is not in stable
20:19.52brookshirenexis: you can get 1.2 beta tar.gz from asterisk.org
20:20.08brookshirebrim???
20:20.23nexissorry, typed b and hit tab.
20:20.30brimstonethat's not the first time that someone's confused brimstone and brookshire
20:20.37mog_homelol
20:20.40fileMATTS!
20:20.45brookshireand we're both named matt
20:20.52nexisis 1.2 beta half way stable?
20:21.01brookshireto add to the confusion
20:21.02brookshirelol
20:21.02Qwell[]nexis: Thats almost literally what it is
20:21.05fileeep
20:21.28nexisok, so ill try that.
20:21.30brookshireit hasn't crashed on me yet, but adding chanspy to the mix.. i dunno
20:21.44nexisehh, whats the worst that can happen right?
20:21.54brookshirehey file, btw
20:22.10nexisalso, has a LOT changed in the configs from 1.0 to 1.2, or can i use the same config files that i spent a few hours, and a lot of mt. dew writing?
20:22.17infinity1Qwell[]: i see what i did last time. i used checkinstall to create a crappy package
20:22.26Qwell[]nexis: get some more Mt. Dew
20:22.47Qwell[]nexis: there is like UPGRADE.txt in there that tells you what changed
20:22.51brookshirehah...
20:24.46nexisthats good to know
20:25.02nexisto grab the 1.2 from cvs, i just drop -r v1-0 correct?
20:25.13infinity1rofl. i like how the changelog jumps from 1.0.10 to 1.2.0
20:25.20brookshirecvs co asterisk
20:25.21brookshire:)
20:25.24nexisk
20:25.27Qwell[]infinity1: because thats exactly what happened...
20:25.54brookshirewow.. they updated it
20:25.57brookshireheh
20:25.58*** join/#asterisk Icemaann (n=flimpy@204.228.197.139)
20:26.06brookshireit was like 1.0.3 to 1.2
20:26.30infinity1Qwell[]: yes. that is true.
20:26.50Qwell[]didn't think they released 1.0.10 though
20:26.52infinity1Qwell[]: i think you get my point though.
20:26.54nexisyou know, in playin with asterisk more and more, i realise that i place i worked for used it i think.
20:27.00Qwell[]infinity1: not really
20:27.16Icemaannin v1.2 im using Set(GROUP()=MYGROUP), and GROUP_COUNT(MYGROUP) is returning 0, is the 1st count 0 not 1?
20:27.58infinity1Qwell[]: scrolld own the changelog to asterisk .4.0. that kind of text would have been nice between 1.0.10 and 1.2, but there was never a version
20:28.12Qwell[]infinity1: link?
20:28.20infinity1ChangeLog in cvs
20:28.35brookshireis there a 1.0.10?
20:28.37brookshireheh
20:28.49brookshirei guess that would be stable cvs
20:28.54infinity1so the fact that there was never a version, makes it a little too convenient :)
20:28.56*** join/#asterisk file (n=jcolp@mctnnbsa31w-142166094161.nb.aliant.net)
20:29.13docelm0ARGH!  REGISTERFLY SUCKS!
20:29.32infinity1so does HEAD still exist? or has it been renamed to beta1?
20:29.42docelm0As far as I know its still around
20:29.43Qwell[]infinity1: HEAD is still HEAD
20:29.49docelm0Beta1 is 1.2 stable
20:29.59Qwell[]"HEAD" is just the most current version in CVS
20:30.18Qwell[]Thats a CVS thing..not really something the Asterisk devs decided on
20:31.23infinity1anyone running a recent HEAD? is it okay?
20:32.04docelm0I am
20:32.06docelm0runs fine
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20:34.21supaigtrHello.
20:38.23*** part/#asterisk logicalonline (n=Ken@209.242.52.25)
20:41.33docelm0ahh well..  Im off for a bit..  going to recode the site..  :)
20:43.03infinity1Qwell[]: this doesn't make sense.... can you help me out?
20:43.10infinity1in the makefile, ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
20:43.29infinity1so if INSTALL_PREFIX=/usr/local, i'll get stuff in /usr/local/usr/lib/asterks?
20:43.32infinity1that doesn't make sense.
20:43.50Qwell[]then change it to $(INSTALL_PREFIX)/usr/local/lib/asterisk
20:44.04Qwell[]no, I don't know
20:44.10Qwell[]that would probably work
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20:44.46infinity1i remember looking at it last time. it doesn't seem to have any logic.
20:45.57supaigtrAnyone using ser in front of *?  Or have ser working on basic level?
20:46.22infinity1supaigtr: yea. my old may build of head :)
20:46.36*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
20:47.51supaigtrinfinity1:  ser or openser?
20:52.29*** join/#asterisk xtr (i=01928375@S01060012174cc0e1.vf.shawcable.net)
20:55.51supaigtrinfinity1: Any pointers on maxfwd and what it is in relation to a sipura getting the 483 message (too many hops)?
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20:59.05nexisanyone had any problems with cvs head puking out on editline?
21:05.09brookshiresearch the bug tracker
21:05.19brookshirehttp://bugs.digium.com
21:09.20nexisyea, old version of libc6
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21:14.54supaigtrAny ser gurus around?
21:20.00docelm0maybe
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21:22.06supaigtr:)  I having troubles with openser and domains.  Getting a loop. But I havn't figured out debugging yet.
21:24.29docelm0ohh I dont knwo it that much..  I just wrote some code in it to pass it on to asterisk
21:24.33docelm0for load balancing
21:26.13supaigtrI'm just trying to get it to register.
21:27.46SplasPoodQuestion for you all...  I have a PRI plugged into zaptel hardware and when I dial out on any of the channels I connect to what sounds like a modem...  Any thoughts?
21:30.13pooh_Hi Guys, is there any way to dynamically change sip or iax conf entries ?
21:30.57pooh_not using realtime but variables (or changing variables) ?
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21:32.40docelm0hay Qwell got the new gallery running
21:33.27Qwell[]any new stuff uploaded?
21:34.25Qwell[]looks exactly the same
21:35.56docelm0Not there..  somewhere else..  working out bugs right now
21:38.36docelm0hay Q how do I make it so users can comment on this thing?   I cant find it anywhere
21:38.52Qwell[]got me...I don't use gallery
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21:39.01Vlathi
21:39.13Error_XWhat ports does IAX2 runs on?
21:43.49docelm0CHECK THE WIKI!  OR IAX2.CONF!
21:44.21docelm0dont ask lame questions without consulting:  WIKI, Goodgle, Userlists..  If all else fails..  come here
21:45.17ariel_Error_X, 4569
21:52.47Vlatwho was on the latest Astricon ?
21:55.00snitt#astricon
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21:55.14iqhi
21:55.24Vlathi
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22:00.54docelm0To all who is avlie..  Gallery2 is online..  check it out..  http://astri2005.netdr.biz
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22:06.26infinity1just upgraded to latest head
22:06.30infinity1fingers crossed
22:08.10ManxPowerinfinity1, be afraid, be very afraid.
22:09.23ManxPowerIn a little while i'll be uploading hires photos of my house and waveland MS after Katrina
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22:11.50infinity1hah. something already doesn't work as it use to .
22:12.16ManxPowerinfinity1, there are 2 bugs that I know of.
22:12.47ManxPowerpriorityjumping is off by default (it should be on by default) and enum lookups now REQUIRE a + prefix
22:12.48infinity1ManxPower: please tell me
22:13.29infinity1prio jumping. hmm ..i think thats the one i may have experienced.
22:13.43infinity1i dont use enum
22:13.54ManxPowerinfinity1, look at extensions.conf.sample in the astrisk source DIr and also UPGRADE.txt in the asterisk source dir
22:14.06infinity1ManxPower: isn't there an option in a conf file for prio jumping?
22:14.50ManxPowerinfinity1, that's why I referred you to extensions.conf.sample
22:15.33ManxPowerI never use priority jumping eve in 1.0.x so I didn't notice that problem right away
22:16.33infinity1ManxPower: hmmm ..sounds like they are going to change it to no by default in the future?
22:17.12ManxPowerinfinity1, I would assume yes for POST 1.2
22:18.03infinity1ah
22:18.13infinity1lets see what happens now. i added it to my extensions.conf
22:19.48infinity1hmm ...didn't solve it. i'll dig around. i haven't messed with * in months. i'm very rusty. once it worked ...i just had to reload the zap modules every few months.
22:20.08infinity1..hopefully that problem went away.
22:21.05infinity1...when calling into the zap interface, it says another zap interface is calling instead of the outside line.
22:21.11infinity1odd
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22:44.40*** join/#asterisk gorauskas (n=gorauska@adsl-68-127-104-56.dsl.pltn13.pacbell.net)
22:47.57nexiswhy would music on hold fade out if the party on hold is silent, like if they mute, the music on hold stops.
22:49.06*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
22:56.12*** join/#asterisk spiekey (n=spiekey@p549D23DD.dip0.t-ipconnect.de)
22:56.14spiekeyhello!
22:56.45spiekeyanyone here with a german isdn setup? ;)
22:57.02spiekeyi would like to put a asterisk box between my "Amt" and my TK
22:57.52*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
22:58.42docelm0Can anyone say fun with images?
23:00.35*** join/#asterisk JustinG (n=justin@line105-166.adsl.actcom.co.il)
23:00.51JustinGhi, anyone willing to help a frustrated fellow with a IAXlite/asterisk setup?
23:02.33*** join/#asterisk |cleric| (n=dacleric@p54829E97.dip0.t-ipconnect.de)
23:06.53*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
23:06.56Vlata question: why do the people choice IAX, not SIP ?
23:07.48kshumard_homeVlat, smaller headers = less bandwidth, no nat problems, simpler to implement
23:08.24*** part/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com)
23:09.21Vlatkshumard_home: and the sip nasty features like call hold, transfer, redirect ?
23:09.58kshumard_homeiax can do pretty much all of those as well
23:11.40Vlatkshumard_home: any IAX supported hardware ? I'm asking just for know, shall I test IAX in our network or not
23:11.45spiekeywhats "TK-Anlage" in english?
23:11.53spiekeyor "Amt"? ;)
23:12.53kshumard_homeDigium sells the IAXy, an ATA which speaks IAX
23:13.11ManxPowerIAX is mostly used for Server-server stuff
23:13.14kshumard_homeother than that, any box running asterisk can do iax, and there are several softphones like iaxcomm, diax, etc
23:13.27Vlatkshumard_home: the only thing i'm afraid for is the syncronization. Since IAX encapsulate RTP & Signalling in one port, it become the...hm...in-band coding isn't the proper definition for it
23:13.41Vlatbut the packet loss can vary the RTP IMHO
23:13.56kshumard_homeVlat, iax doesn't use RTP
23:14.13kshumard_homeVlat, it carries signalling and media both on the iax2 protocol, UDP port 4569
23:14.19kshumard_homethat's why it's nat-friendly
23:16.21spiekeywhats a BRI card good for? i somehow cant find any basic documentation about it.
23:16.33Vlatkshumard_home: another question. Is Asterisk is ready for a large-scale deployment ? I mean 1000+ customers at dual Xenon as primary gw
23:16.38Igbothom_IIIBRI - ISDN
23:16.59spiekeyah!
23:17.16Igbothom_IIIeach BRI line gives 2 phone lines
23:17.17Vlatkshumard_home: only sip, just routing, no rtp bypass, + media-backend features
23:18.10kshumard_homeVlat, you might look into SER
23:18.18VlatXenon-Xeon
23:18.25kshumard_homeVlat, SER + asterisk is being used by several people for large-scale sip networks
23:18.31Vlatkshumard_home: currently I'm using SER+asterisk :)
23:18.37spiekeyIgbothom_III: so a setup could look like this: telephone provider --> BRI/ISDN(asterisk)analog card --> phones  ?!
23:18.37Igbothom_IIIkshumard_home; SER runs on *?  or with *?
23:18.39kshumard_homehaha, alright
23:18.42Vlatkshumard_home: SER as router, asterisk as backend
23:18.42distortionkshumard_home: so you're saying by definition because voice goes over the same port in iax that its not "rtp
23:18.45distortion?
23:18.51Igbothom_IIIBRI isn't analog, it is digital
23:19.00Vlatkshumard_home: but ser's config messes my brain up :)
23:19.15kshumard_homertp is an ietf protocol which specifies how to carry a realtime media stream. It's one way of doing it
23:19.19kshumard_homeiax doesn't use it
23:19.36kshumard_homeiax carries its own media stream. So iax is a replacement for sip, and iax is also a replacement for rtp
23:19.39Igbothom_IIIlike IAX doesn't use smtp - another protocol available
23:19.39kshumard_homeit does both
23:20.17kshumard_homeVlat, I've never used SER, don't know much about it... just that it's the de facto choice for large scale asterisk + sip systems
23:20.22spiekeyIgbothom_III: well, yes. But it comes to my house digital, then i connect it to my bri card in my asterisk box, right?!
23:20.32*** join/#asterisk Katty (n=katrina@68.112.15.110)
23:20.39Kattymew.
23:20.50spiekeyfrom there i can use one of Digiums analog cards to connect my analog phones.
23:20.55spiekeycorrect?!
23:21.03Igbothom_IIIok - Telco---------BRI Card---*---IP Phones
23:21.21Vlatkshumard_home: we had tried *, it sucked at 300 clients. so it's the way we choiced * as m-back. I just wanted to know maybe the things imporved. :)
23:21.23kshumard_homespiekey, yeah, that would work
23:21.27Igbothom_IIIspiekey; yes, if you have some FXS ports, you can connect regular phones to the * server
23:21.28spiekeythanks!
23:22.00Igbothom_IIIor buy IP Phones, preferably with IAX support  :)
23:22.30Kattyanyone work in robotics?
23:22.51kshumard_homeI watched I, Robot last night....  : )
23:22.58Kattynot quite the same.
23:23.09kshumard_homeoh. sorry. : (
23:24.08Kattytwisted: stop spreading germs.
23:24.54*** join/#asterisk |cleric| (n=dacleric@p54829E97.dip0.t-ipconnect.de)
23:24.58*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
23:25.00Kattytwisted: the flat peoples won't let me have a kitty, so i'm going to build one.
23:25.32twistedbuild a flat? or a kitty?
23:25.41VlatI, Robot..hm... 2 day ago i read back Azimov memories. He wrote (in 1956) that several years later there'll be a global communication network, that will change the life of million people. There'll be such definition as remote-work, the people won't have to close each other to share the problemms, etc...
23:25.52Vlatand all it in 1956
23:26.07Kattytwisted: a kitty, obviously. like Necoro
23:26.22twistedahh
23:26.26twistedcool
23:26.49kshumard_homeI was surprised how different the movie was than the book.... it didn't even say "based on the I. Asimov book". They had to say "suggested by"
23:26.54Kattymust go to radio shack
23:27.11Vlatkshumard_home: Hollywood crap
23:27.29spiekeyi have a normal telephone setup right now which looks like this: Telco --> TK --> ISDN-Phones.
23:27.42Vlat"make it understandable to everyone, even if he/she have IQ 10)
23:27.55spiekeyand if i want to put Asterisk between it all it would look like this?!  Telco --> BRI
23:27.58*** join/#asterisk hanged (n=mefisto@195.244.159.7)
23:28.06kshumard_homeVlat, yeah, even a robot could understand it.  ; )
23:28.06spiekeyand if i want to put Asterisk between it all it would look like this?!  Telco --> BRI*BRI --> ISDN-Phones ?!
23:28.39Vlatkshumard_home: if the robot read i,robot it mean the robot read i,robot...so endless recursion :)
23:29.00spiekeyi would have to configure asterisk with two "interfaces", therefore * sits between it all.?!
23:29.44kshumard_homehaha
23:30.30*** join/#asterisk nobell (n=Jared@160.7.249.18)
23:31.14nexiswhy would music on hold fade out if the party on hold is silent, like if they mute, the music on hold stops.
23:31.17Igbothom_IIIspiekey; dunno if you can plug ISDN phones into * - didn't know there WERE ISDN phones, actually, it usually presents as 2 * analog interfaces if you have a box on the wall - Bri goes in, 2 * analog comes out
23:31.28Kattytwisted: looks like building a robotic kitty is going to cost me about 2000
23:31.54*** part/#asterisk nobell (n=Jared@160.7.249.18)
23:32.01Kattytwisted: mew :<
23:32.35docelm0KATTY!
23:32.51twistedKatty, :(
23:33.08Kattydocelm0: what?
23:33.16docelm0nothing.. Just hola..
23:33.39Kattyhi.
23:34.01Vlatnexis: vad ?
23:34.33nexisVlat, vad?
23:34.53kshumard_homevad: vad?
23:34.55kshumard_home: )
23:34.57Vlatvoice activity detection
23:35.14nexisdont think so, let me check
23:35.19kshumard_homenexis, turn off silence suppression on the phone if it's an option
23:35.26spiekeyIgbothom_III: the thing is, that i have a telephone setup here, and all i want to do is to put asterisk between it, to make the migration easier. But i have trouble understanding "where" the call comes in an out again, through asterisk.
23:35.30nexisit was doing it in stable, and still doing it in dev
23:35.34spiekeyand what cards i need for that
23:35.43Vlathm, yes...SS on non-cisco equipment :))))
23:35.52spiekeyis there no documentation about it?
23:36.07*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
23:36.10nexisvad is off
23:36.42nexiskshumard_home, tried on all 3 of my ATAs, and both of my outgoing trunks, still doin it.
23:37.58*** join/#asterisk nobell (n=Jared@160.7.249.18)
23:40.51Vlatnexis: MOH is mp3 ?
23:40.56Igbothom_IIIspiekey; Asterisk is SERIOUSLY lacking in documentation for pretty much every aspect of it
23:41.04spiekeyhehe
23:41.07*** join/#asterisk PMantis (n=pmantis@cpe-69-204-25-153.rochester.res.rr.com)
23:41.07nexisVlat, correct
23:41.10ManxPowernexis, the call comes into the context in extensions.conf that is defined in zaptel.conf (assuming the call is coming in via Zap)
23:41.19Igbothom_IIIok, let's go step-by-step  :)
23:41.27ManxPowerzaptel.conf contexts have NOTHING to do with outgoing calls
23:41.29Igbothom_III1. ISDN - does it go to a box on the wall?
23:41.45spiekeyIgbothom_III: yes
23:41.47Vlatnexis: mpg123 59r + silencesuppression=no in phone.conf
23:41.56Igbothom_IIIcool - and does this have 2 * phone ports on it?
23:41.58PMantisAnyone solve this issue with iaxcomm ?  "./iaxcomm: error while loading shared libraries: libtiff.so.3: cannot open shared object file: No such file or directory"
23:42.22ManxPowerPMantis, no, but the error is pretty obvious
23:42.24PMantislibtiff.so.4 exists, creating a symlink doesn't help
23:42.27nexisVlat, thats what i have.
23:42.35spiekeyIgbothom_III: yes, you _could_ plug in 2 ISDN phones there.
23:42.39ManxPowerPMantis, you didn't build from source.
23:42.43nexisPMantis, what distro?
23:42.54Igbothom_IIIISDN phones?  Sure they are not regular analog phones?
23:43.07PMantisManxPower, Right, I found no source archive. :(
23:43.12Vlatnexis: the thing going interesing...silencesuppression=yes? what would be the result ?
23:43.18PMantisnexis, Ubuntu AMD64
23:43.26Igbothom_IIIthat's what I've seen with BRI - Telco ---- NT1/NT2 == analog phone ports
23:43.28nexisPMantis, apt-cache search libtiff
23:43.32Vlatbtw, google say a lot about your problem
23:43.40nexisshould come up with libtiff4 or something, apt-get install it
23:43.41Vlatasterisk+vad
23:44.01Kattytwisted: the hard part is going to be making it not run into things. if front left whisker is on, stop, go backwards 2 feet, turn right, continue
23:44.16nexisVlat, ok, ill look into it some more, not that its really a problem, i may just ice the music on hold.
23:44.48Kattytwisted: do you have a drill press or a band saw i can borrow?
23:44.49spiekeyIgbothom_III: the line comes in from the provider and goes into a splitter, where it seperates ISDN and DSL
23:44.50*** join/#asterisk pussfeller (n=todd@12.150.129.170)
23:44.52Vlatnexis: never used it really, to be honest. it eat my bandwidth
23:45.00Vlatand nervous for users
23:45.01PMantisnexis, libtiff 4 is installed. I created a symlink for libtiff.so.3, didn't help
23:45.22ManxPowerPMantis, I doubt creating a symlink between major versions will help.
23:45.31ManxPowerBUt, did you run ldconfig after you did the symlink/
23:45.37spiekeythen the isdn cable goes to a box where i could plug two isdn phone in. But in my case there is only one big telephone box pluged in.
23:45.52nexisPMantis, is there a site like with debian for ubuntu?
23:46.01ManxPowerspiekey, ISDN phones use ISDN PRI.  Telephone systems frequently use ISDN PRI
23:46.03spiekeyand on that "big" telephone box, there are 5-8 analog phones plugged in.
23:46.14PMantisManxPower, Hmmm, no idn't run that
23:46.16Igbothom_IIIok - never seen it done this way - here in australyamate we have a box that breaks the BRI into 2 * POTS (analog) ports
23:46.18ManxPowerLets try that again
23:46.22ManxPowerspiekey, ISDN phones use ISDN BRI.  Telephone systems frequently use ISDN PRI
23:46.25Igbothom_IIIso, 1 BRI gives 2 * POTS lines
23:46.28ManxPowerBRI is totally different from PRI
23:46.38Igbothom_IIIyes
23:46.44ManxPowerBRI can handle two calls, PRI can handle 23 or 30 calls
23:46.53spiekeyi think i have BRI
23:46.54Igbothom_IIIPRI is digital all the way, BRI isn't necessarily
23:47.03ManxPowerIgbothom_III, that is incorrect.
23:47.08Igbothom_IIIPRI can handle 10, 20, 30 here in Australyamate
23:47.09Vlatthe "big telephone box" is the PRI router/splitter ?
23:47.33ManxPowerISDN (PRI OR BRI) is always digital to the local phone switch, from there you never know.
23:47.43Igbothom_IIIfair enough
23:48.04spiekeyVlat: "the big telephone box" is the box where you can set up the msns and scheduled calls/rules, etc...
23:48.14Igbothom_IIIwhat I SHOULD have said is, what I'm used to is BRI breaking into 2 * Analog, and PRI being delivered digitally direct into a PABX
23:48.16Igbothom_III:)
23:48.20PMantisnexis, Ubuntu.com? Yeah. Ubuntu is also based on debian.
23:48.28ManxPowerspiekey, until you know the version of ISDN your PBX uses, you are wasting your time.
23:48.34Igbothom_IIIexactly
23:48.44ManxPoweryou can't even know what interface card you need for Asterisk until you know if it's BRI or PRI
23:48.49Vlati just didn't get a glue :) i have a big telephone box at my table, it's the laaarge siemens phone :)
23:48.57spiekeyi am 99% sure its BRI!
23:49.09Igbothom_IIIthat;s 99% good enough  :)
23:49.17ManxPowerspiekey, no Digium card supports BRI
23:49.53PMantisnexis, Oh, WOW... I just found a deb for iaxcomm..
23:50.16spiekeyManxPower: wasnt there a 8-Port card for Bri?
23:50.23ManxPowerspiekey, not from digium.
23:50.29Igbothom_IIIyeah, but not a Digium card
23:50.34ManxPowerOctoBRI is from Kapejob's company
23:50.52Kattyyou know, i wonder what you oculd use an asterisk robot for
23:51.00Igbothom_IIIberoNET have some as well
23:51.06Kattyi guess it would sorta have to be stationary.
23:51.14Igbothom_IIIa paper robot?
23:51.17X-Robspiekey - you want chan_capi and bristuff
23:51.32KattyIgbothom_III: no. i'm trying to find some use for asterisk on a robot
23:51.32spiekeysounds reasonable
23:51.42KattyIgbothom_III: other than having the robot do something based on an incoming call.
23:51.56*** join/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net)
23:51.58KattyIgbothom_III: but even if it did, what would the robot do?
23:51.58Igbothom_IIIdistinctive dances?
23:52.10KattyIgbothom_III: locate the person by gps?
23:52.19Kattyit seems silly.
23:52.22Igbothom_IIIyes
23:52.45BrianR___is there any way to make asterisk change its codec negotiation preference order based on the peer's IP address being or not being on a particular list of subnets?
23:52.48Kattyi find it silly to have a robot track a person's GPS just because they get a call
23:53.22spiekeyso i come from my provider via my dsl/isdn splitter into my *, from there i go via the 8-Port BRI card. correct?
23:53.23Kattymy kitty must have some useful purrpoise though
23:53.44Kattymaybe it can function as a speakerphone.
23:54.20kshumard_homeBrianR___, You could define different peers for different subnets, each with its own codec preferences...
23:55.10Kattyhmm, text to speech posibilities.
23:55.28Vlattext2speech is FESTIVAL
23:55.30Kattycallerid reading mobile kitty?
23:55.37kshumard_homeor would that be text to meow?
23:55.54Kattykshumard_home: i think i'd rather have my robotic cat talking than meowing all the time
23:56.00*** join/#asterisk mooh_ (n=freaky@80.252.133.102)
23:56.01Kattykshumard_home: would simply be more useful.
23:56.07mooh_hi adepts
23:56.28kshumard_homeI don't know, though.... having a robotic cat talk to me might creep me out a bit
23:56.34spiekeyhas someone got me a link for a 8 port bri card?
23:56.42VlatOh, there's mooh_! The biggest asterisk adept :)
23:56.56Kattykshumard_home: making the cat seem life like is going to be difficult. while using sensors can keep you from running into things...i'm not sure how to put a wireless reciever on it and then make it say whatever the callerid is
23:57.03mooh_jo Vlat!
23:57.10*** join/#asterisk skrewtape (n=skrewtap@pool-71-112-240-56.sttlwa.dsl-w.verizon.net)
23:57.19skrewtape<PROTECTED>
23:57.26skrewtapeI was happy with broadvoice last year but they no longer have a presence in the area code I want to use.  my needs are (206) and dialin from (310) and (818)
23:57.29Kattykshumard_home: yes, but you could make it do all sorts of things with text to speech
23:57.32skrewtapeanyone awake that has opinions on the matter?
23:57.44Kattykshumard_home: temperature sensors, motion sensors,...
23:57.52Vlatspiekey: www.ipgw.net
23:57.56kshumard_homeKatty, It would be really amazing if you could make it do meow to speech. : )
23:57.58Kattykshumard_home: it could follow you around based on motion sensors.
23:58.07kshumard_homehaha, a robo-stalker kitty??
23:58.13Kattywhy not?
23:58.14skrewtapeyou'd just hear 'feed me pet me love me'
23:58.21skrewtapeover and over
23:58.29skrewtapere: meow to speech
23:58.31kshumard_homelike a neopet you don't have to carry
23:58.36Kattythere's also video modules in robotics
23:58.38Vlatkshumard_home: it would like 'MiU'
23:58.39kshumard_homeor whatever those things were called
23:58.44skrewtapewell depends on the laziness of the cat
23:58.55Vlatkshumard_home: or M e o Ou :)
23:58.55Kattybut how do you really turn a robotic cat into a phone?
23:58.57Kattyoh god.
23:59.01Kattya cat phone
23:59.12Kattya robotic cat phone! talk to your cat and it functions as the phone
23:59.19skrewtapeanyone? business voip providers in US that don't suck?
23:59.19kshumard_homehaha, you'd have bandwidth concerns because of course it could only speak mu-law.  ; )
23:59.22Kattyand the cat could follow you around based on motion sensors
23:59.25Kattyand be your MOTHER
23:59.26*** join/#asterisk hardwire (n=hardwire@209-112-136-78-cdsl-rb1.nwc.acsalaska.net)
23:59.33hardwireaww
23:59.37hardwireI missed astricon
23:59.49Kattythat shall be my project
23:59.57Kattya robotic cat which functions as a wireless phone
23:59.59spiekeyVlat: cant read that language :P

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